- sr-users - kamailio.org

[Kamailio-Users] Issue with dialog module
by Santiago Gimeno 19 Feb '10

19 Feb '10

Hi, I'm using kamailio 1.5 from the branch. In a conditional forwarding scenario with this call flow: Phone A Kamailio Phone B Phone C | | | | | | | | | | | | |INVITE | | | |------------->| | | |100 Trying | | | |<-------------| | | | |INVITE | | | |------------->| | | |100 trying | | | |<-------------| | | |180 Ringing | | | |<-------------| | |180 Ringing | | | |<-------------| | | | |486 Busy Here | | | |<-------------| | | |ACK | | | |------------->| | | |INVITE | | | |---------------------------->| | |100 trying | | | |<----------------------------| | |180 Ringing | | | |<----------------------------| |180 Ringing | | | |<-------------| | | | |200 OK | | | |<----------------------------| |200 OK | | | |<-------------| | | |ACK | | | |------------->| | | | |ACK | | | |---------------------------->| | | | | | | | | What I'm observing is that when Kamailio receives the 486 from B, the dialog between A and B is not destroyed, so when the call is finally established between A and C there are two active dialogs as I could check with `kamctl fifo get_statistics active_dialogs`. Shouldn't there be only one active dialog? Is this a problem in my script or a problem with the dialog module? Thanks in advance, Santi

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[Serusers] tell me module of ser for nideo streaming..
by city spider 18 Feb '10

18 Feb '10

i want to use ser for streaming . please tell me the specific module for streaming... in streaming me using RTSP (Real Time Streaming Protocol) suggest me the specific module that will be added to fulfil my purpose.

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[Kamailio-Users] Help me on 'No matching subscription dialog found'
by Asterisk User 18 Feb '10

18 Feb '10

Hi Group, I have one more problem regarding a SIP subscribe packet. When my phone registers first time, it gets OK for SUBSCRIBE sent from phone. But after couple of hours it(phone) keeps on sending subsequent SUBSCRIBE requests but the response from Kamailio is 481 Subscription does not exist. Here are my packets, =======================> For first SUBSCRIBE SUBSCRIBE sip:ABC.com SIP/2.0 Via: SIP/2.0/UDP 172.18.100.89:5060;branch=z9hG4bK-4082dbb From: DEF <sip:307@ABC.com>;tag=802bf23e23f4741 To: DEF <sip:307@ABC.com> Call-ID: 9f465e4b-2207f239(a)172.18.100.89 CSeq: 30004 SUBSCRIBE Max-Forwards: 70 Contact: DEF <sip:307@172.18.100.89:5060> Accept: application/simple-message-summary Expires: 2147483647 Event: message-summary User-Agent: Linksys/PAP2T-3.1.15(LS) Content-Length: 0 ------------------------------------------------------- SIP/2.0 202 OK Via: SIP/2.0/UDP 172.18.100.89:5060;branch=z9hG4bK-4082dbb;rport=5060;received=220.224.237.54 From: DEF <sip:307@ABC.com>;tag=802bf23e23f4741 To: DEF <sip:307@ABC.com>;tag=0cca86077d334bd53b00d2c8eab3d5fb.dfeb Call-ID: 9f465e4b-2207f239(a)172.18.100.89 CSeq: 30004 SUBSCRIBE Expires: 3600 Contact: <sip:192.168.94.30:5060> Server: Kamailio (1.5.0-notls (i386/linux)) Content-Length: 0 =======================> For subsequent SUBSCRIBEs SUBSCRIBE sip:ABC.com SIP/2.0 Via: SIP/2.0/UDP 172.18.100.89:5060;branch=z9hG4bK-69712d2e From: DEF <sip:307@ABC.com>;tag=5d3bd6c2989f0839 To: DEF <sip:307@ABC.com>;tag=0cca86077d334bd53b00d2c8eab3d5fb.dfeb Call-ID: 7f6d2c36-97569155(a)172.18.100.89 CSeq: 55947 SUBSCRIBE Max-Forwards: 70 Contact: DEF <sip:307@172.18.100.89:5060> Accept: application/simple-message-summary Expires: 2147483647 Event: message-summary User-Agent: Linksys/PAP2T-3.1.15(LS) Content-Length: 0 ------------------------------------------------------- SIP/2.0 481 Subscription does not exist Via: SIP/2.0/UDP 172.18.100.89:5060;branch=z9hG4bK-69712d2e;rport=5060;received=220.224.237.54 From: DEF <sip:307@ABC.com>;tag=5d3bd6c2989f0839 To: DEF <sip:307@ABC.com>;tag=0cca86077d334bd53b00d2c8eab3d5fb.dfeb Call-ID: 7f6d2c36-97569155(a)172.18.100.89 CSeq: 55947 SUBSCRIBE Server: Kamailio (1.5.0-notls (i386/linux)) Content-Length: 0 ---------------------------------------------------------------------------------------------------------------------------------------------- ===>The error which I get on kamailio is, ERROR:presence:get_database_info: No matching subscription dialog found in database ===>Have already loaded the module presence.so modparam("presence", "db_url", "mysql://XYZ:XYZ@192.168.90.12/openser") modparam("presence", "max_expires", 3600) modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "clean_period", 100) modparam("presence", "db_update_period", 100) modparam("presence", "to_tag_pref", 'pres') modparam("presence", "expires_offset", 10) modparam("presence", "server_address", "sip:192.168.94.30:5060") modparam("presence", "fallback2db", 1) modparam("presence", "subs_htable_size", 11) modparam("presence", "pres_htable_size", 11) loadmodule "presence_mwi.so" loadmodule "presence_xml.so" modparam("presence_xml", "force_active", 1) modparam("presence_xml", "db_url", "mysql://XYZ:XYZ@192.168.90.12/openser") --------------------------------------------------------------------------------------------------------------------------------------------- Please help me on this. --SM

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[Kamailio-Users] starting
by Andrew Matthews 17 Feb '10

17 Feb '10

I'm looking for a tutorial of some sort to help me with getting openser running. What I need from it, are a basic proxy with both IP and username/password authentication, and an lcr. I've searched the net and can't seem to find what I need. I'm new to openser, I came from the asterisk world. If anyone can point me in the right direction I'd appreciate it. Thanks, Exstatica

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Number Portability Functionality
by Asim Riaz 17 Feb '10

17 Feb '10

Hi list, Can someone explain me about PDB server? what I understand that PDB server have all the routing information for all the numbers and SIP-Router receive that information and route the call accordingly but how PDB server receive the routing data? Do we need to load this routing data manually? What happen if the routing information for a number is changed, do we need to update manually? Thanks, Asim

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[Kamailio-Users] Inbound call no way audio
by RAJNIKANT VANZA 17 Feb '10

17 Feb '10

Hi Friends, Outbound calling works fine. When I receive an inbound call, the phone rings, but when I pick up the phone, the conversation does not get established. I have call from inside NAT extension number to Outside NAT extension ( Outboud call ) then call get established and two-way audio work fine but when I have call from Outside NAT extension to Inside NAT extension (Inbound call ) then phone rings, but when pickup the phone, audio is not present. During inboud call RTP packets are not transmit whereas Outboud call RTP packets transmit successful. Sometimes, Inbound and Outbound call only one-way audio present. Can anybody help me, or point me to a previous thread where this issue was discussed and solved? Thanks in advance. -- Best Regards, Rajnikant Vanza

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[Kamailio-Users] What is the best (and secure) way to change FROM_URI using regexp?
by Fernando Frota M Morais 16 Feb '10

16 Feb '10

Before sending this message I looked for an answer and found some options, two of them are followed: 1) SUBST function, sample listed here: if (is_present_hf("User-Agent")) { remove_hf("User-Agent"); remove_hf("P-Asserted-Identity"); subst("/^From: \".+\" (<sip: *[a-zA-Z0-9_\.-]+ <at> .*>)/From: \1/ig"); subst("/^To: \".+\" (<sip: *[a-zA-Z0-9_\.-]+ <at> .*>)/To: \1/ig"); } 2-a) uac_replace: if (from_uri=~"^sip:10804007\@voip.mydomain") { xlog("L_ALERT", "TESTANDO - Original From is ($fu-$fn)\n"); # uac_replace_from("sip:031" . $fU . "\(a)voip.mydomain"); uac_replace_from("Frota10804007","sip:03110804007@voip.mydomain"); xlog("L_ALERT", "TESTANDO - From changed to ($fu-$fn)\n"); }; OR : 2-b) uac _replace #2: if (from_uri=~"^sip:10804007\@.*") { xlog("L_ALERT", "Route 21 - Original from is ($fu-$fn)\n"); $avp(from)=$fu; # $avp(to)=$ru; # avp_pushto("$ru", "$avp(from)"); avp_subst("$avp(from)", "/(sip:)/sip:031/"); uac_replace_from("$avp(from)"); # xlog("L_ALERT", "Route 21 - From changed to ($fu-$fn)\n"); # avp_pushto("$ru", "$avp(to)"); }; } OR Looks like uac_replace is safer then "subst" function, because it can be reverted, but I did not found a way to use concatenation or any other kind of string operation as a parameter for it. I don't want to start a discussion, I need only a few lines showing me how to change from_uri_from from "sip:108xxxx" to "sip:031108xxxx". Thanks in advance, -- Fernando Frota Machado de Morais Divisao de Redes de Comunicacao Centro de Computacao Universidade Federal de Minas Gerais Brasil Tel. +55(31)3409.4007 Fax. +55(31)3409.4004

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[Kamailio-Users] What is wrong with this NOTIFY packet
by Asterisk User 15 Feb '10

15 Feb '10

Hi Group, I know this issue has to do bit more with SIP and off from kamailio problems, but just wanted to know your opinions on this. I have my sip extensions registered to kamilio and Asterisk works as a voicemail server. Now to implement MWI i send NOTIFY packet from Asterisk to Kamailio and from kamailio to SIP extension (Grandstream GXP2000). here is the packet that travels from kamailio to extension and the response sent from extension to kamilio. =========================== NOTIFY sip:1111111@arzoo.com <sip%3A1111111(a)arzoo.com> SIP/2.0 Max-Forwards: 10 Record-Route: <sip:172.18.100.74;lr;nat=yes> Via: SIP/2.0/UDP 172.18.100.74;branch=z9hG4bK3c3.e1003f03.0 Via: SIP/2.0/UDP 127.0.0.1:45382 ;received=172.18.100.73;branch=z9hG4bK.4f83530d;rport=45382;alias Via: SIP/2.0/UDP 172.18.100.73:5060 From: <sip:111@arzoo.com <sip%3A111(a)arzoo.com>> To: <sip:1111111@arzoo.com <sip%3A1111111(a)arzoo.com>> Contact: <sip:1111111@arzoo.com <sip%3A1111111(a)arzoo.com>> Call-ID: 4d61cf9e505d40e905032a18329d61ec(a)172.19.100.42 CSeq: 1 NOTIFY User-Agent: VoiceMail Event: message-summary Content-Type: application/simple-message-summary Content-Length: 39 =========================== SIP/2.0 415 Unacceptable Content-Type Via: SIP/2.0/UDP 172.18.100.74;branch=z9hG4bK3c3.02003f03.0 Via: SIP/2.0/UDP 127.0.0.1:43416 ;received=172.18.100.73;branch=z9hG4bK.7be73901;rport=43416;alias Via: SIP/2.0/UDP 172.18.100.73:5060 Record-Route: <sip:172.18.100.74;lr;nat=yes> From: <sip:111@arzoo.com <sip%3A111(a)arzoo.com>> To: <sip:1111111@arzoo.com <sip%3A1111111(a)arzoo.com>>;tag=3c6ba3e4a93d6864 Call-ID: 4d61cf9e505d40e905032a18329d61ec(a)172.19.100.42 CSeq: 1 NOTIFY User-Agent: Grandstream GXP2000 1.1.4.14 Accept: application/sdp, application/simple-message-summary, application/octet-stream, application/dialog-info+xml, application/pidf+xml, message/sipfrag;version=2.0 Content-Length: 0 =========================== Please guide me in right direction to get rid of 'Unacceptable Content-Type' problem. Thanks in advance, --SM

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[Kamailio-Users] Problems with parallel forking to different users
by Anton Roman 15 Feb '10

15 Feb '10

Hi all, I'm trying to implement parallel forking to different users registered on a Kamailio server. I want to fork the call to a number of different users (not including the called user). To do this I added the piece of code below to the default 3.0.0 config file: ... # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting setflag(4); #dialog aware if(avp_db_load("$ru/username","$avp(s:fork)")) { avp_pushto("$ru/username","$avp(s:fork)/g"); }; } .... This is usr_preference table content: +----+------+----------+--------+-----------+------+-------+---------------------+ | id | uuid | username | domain | attribute | type | value | last_modified | +----+------+----------+--------+-----------+------+-------+---------------------+ | 4 | | 1000 | | fork | 0 | 1001 | | | 3 | | 1000 | | fork | 0 | 1005 | | +----+------+----------+--------+-----------+------+-------+---------------------+ Kamailio correctly forks the call to 1001 and 1005, but the Request-URI of the 1005 user is not correctly set, because the proxy IP is used instead of the IP where the user is registered (10.1.2.15). You can see the INVITEs below. Could anybody tell me what I'm doing badly? Thanks for your time, best regards Anton ---------------------------------------------------------------------------------------------------------------------------------------- # U +0.000328 10.1.20.103:5060 -> 10.1.3.18:5061 INVITE sip:1001@10.1.3.18:5061 SIP/2.0 Record-Route: <sip:10.1.20.103;lr=on;did=c34.3ed81d77> Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKc40b.ec3572a3.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-213aa77 From: "1002" <sip:1002@10.1.20.103 <sip%3A1002(a)10.1.20.103> >;tag=280da520d15d62cfo1 To: "1000" <sip:1000@10.1.20.103 <sip%3A1000(a)10.1.20.103>> Call-ID: 8973c34-427096cb(a)10.1.3.15 CSeq: 102 INVITE Max-Forwards: 69 Contact: "1002" <sip:1002@10.1.3.15:5061> Expires: 240 User-Agent: Linksys/SPA941-5.1.5 Content-Length: 202 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 2003696 2003696 IN IP4 10.1.3.15 ... # U +0.000052 10.1.20.103:5060 -> 10.1.20.103:5060 INVITE sip:1005@10.1.20.103 <sip%3A1005(a)10.1.20.103> SIP/2.0 Record-Route: <sip:10.1.20.103;lr=on;did=c34.3ed81d77> Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKc40b.ec3572a3.1 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-213aa77 From: "1002" <sip:1002@10.1.20.103 <sip%3A1002(a)10.1.20.103> >;tag=280da520d15d62cfo1 To: "1000" <sip:1000@10.1.20.103 <sip%3A1000(a)10.1.20.103>> Call-ID: 8973c34-427096cb(a)10.1.3.15 CSeq: 102 INVITE Max-Forwards: 69 Contact: "1002" <sip:1002@10.1.3.15:5061> Expires: 240 User-Agent: Linksys/SPA941-5.1.5 Content-Length: 202 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 2003696 2003696 IN IP4 10.1.3.15 ... ----------------------------------------------------------------------------------------------------------------------------------------

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[Kamailio-Users] Registration Timeout Retransmission
by Hemanshu Patel 13 Feb '10

13 Feb '10

Hello to all users, I am very new to kamailio and just started using it. I have installed it and my phones are getting register successfully. But sometimes i am seeing a scenario, which makes my phones unusable for sometime. please see the condition as below | phone |--------------| Kamailio | | REGISTER-----------------> <----------------------401 | REGISTER-----------------> Tricky condition here | timeout <--------------200 OK phone shows itself as unregistered | REGISTER----------------> <--------------------401 After several retries phone stops to retry till say next 20 seconds, during that time though there's an entry in location table in kamailio (i am using mysql) my phone shows itself as unregistered phone. Due to load or any other conditions Server takes a bit time to send 200OK back to phone and During that time timer expires and phone sends REGISTER once again. now phone will keep sending REGISTER request with same nonse and other value, and kamailio is configured to not to entertained same credentials with nonce if it is authenticated once. So kamailio generates new 401 Message with new nonce value to which phone understands as the previous reply thet he send for previous 401 was rejected, and the most probable reason is wrong user credentials..... Hope you guys have got my problem? i can not allow kamailio to let use the same credentials+ response with nonce if it has used it once for security reason. Then what could be the solution for this. I am very good in Programming, but just that i have just started to use kamailio and not very much aware about its architecture. -- Regards, Hemanshu Patel M: 09601295238

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[Serusers] stateful statistics
by Sabatella, Michael 12 Feb '10

12 Feb '10

I know somebody else posted this a while ago but I did not see any response. I tried to research it but could not come up with any answers and would like to get an understanding of this . On the stateful statistics what does 130 waiting and 6 current mean Stateful Statistics Current: 6 (130 waiting) Total: 3540496 (0 local) Replied localy: 3809867 Is it recommended to restart the process once a week or once a month? I am running ser 0.9.6 Thanks for the help. Michael ________________________________ DISCLAIMER: This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unintended recipients are prohibited from taking action on the basis of information in this e-mail.E-mail messages may contain computer viruses or other defects, may not be accurately replicated on other systems, or may be intercepted, deleted or interfered with without the knowledge of the sender or the intended recipient. If you are not comfortable with the risks associated with e-mail messages, you may decide not to use e-mail to communicate with IPC. IPC reserves the right, to the extent and under circumstances permitted by applicable law, to retain, monitor and intercept e-mail messages to and from its systems.

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[Kamailio-Users] Error in header - where and how?
by Anders 12 Feb '10

12 Feb '10

Hi, My carrier doesn't like the invites that come from two specific customers I have, who both use something called VoipSwitch. It returns from the carrier as a Bad Request. Below is the dialog - I'm not sharp enough to see what's wrong - anyone? Thanks! //Anders U 2010/02/01 17:53:42.355689 CustomerIP:5060 -> MyProxyIP:5060 INVITE sip:890912551128650136@MyProxyIP SIP/2.0. CSeq: 1 INVITE. Via: SIP/2.0/UDP CustomerIP:5060;branch=z9hG4bk01025810150801381470578. From: "SERVER BR" <sip:001184517324@CustomerIP:5060>;tag=010258101508381470562. Call-ID: GW6d4c945d2c573c27@CustomerIP. To: <sip:890912551128650136@MyProxyIP>. Contact: <sip:CustomerIP:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 250. Remote-Party-ID: "SERVER BR"<sip:69912390@customerDNS>;privacy=off;screen=no. . v=0. o=VoipSwitch 9298 9298 IN IP4 CustomerIP. s=VoipSIP. i=Audio Session. c=IN IP4 CustomerIP. t=0 0. m=audio 8298 RTP/AVP 18 101. a=rtpmap:18 G729/8000/1. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2010/02/01 17:53:42.355990 MyProxyIP:5060 -> CustomerIP:5060 SIP/2.0 100 Giving a try. CSeq: 1 INVITE. Via: SIP/2.0/UDP CustomerIP:5060;branch=z9hG4bk01025810150801381470578;rport=5060;received=187.45.218.17$ From: "SERVER BR" <sip:001184517324@CustomerIP:5060>;tag=010258101508381470562. Call-ID: GW6d4c945d2c573c27@CustomerIP. To: <sip:890912551128650136@MyProxyIP>. Server: Kamailio. Content-Length: 0. . U 2010/02/01 17:53:42.356010 MyProxyIP:5060 -> CarrierIP:5060 INVITE sip:84370551128650136@CarrierIP:5060 SIP/2.0. Max-Forwards: 10. Max-Forwards: 5. Record-Route: <sip:MyProxyIP;lr=on;nat=yes;did=e4c.e1f33ad5>. CSeq: 1 INVITE. Via: SIP/2.0/UDP MyProxyIP;branch=z9hG4bK139a.a74691c5.0. Via: SIP/2.0/UDP CustomerIP:5060;rport=5060;received=CustomerIP;branch=z9hG4bk0102581015080138147057$ From: "SERVER BR" <sip:001184517324@CustomerIP:5060>;tag=010258101508381470562. Call-ID: GW6d4c945d2c573c27@CustomerIP. To: <sip:890912551128650136@MyProxyIP>. Contact: <sip:CustomerIP:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 250. Remote-Party-ID: "SERVER BR"<sip:69912390@customerDNS>;privacy=off;screen=no. . v=0. o=VoipSwitch 9298 9298 IN IP4 CustomerIP. s=VoipSIP. i=Audio Session. c=IN IP4 CustomerIP. t=0 0. m=audio 8298 RTP/AVP 18 101. a=rtpmap:18 G729/8000/1. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2010/02/01 17:53:42.608917 CarrierIP:5060 -> MyProxyIP:5060 SIP/2.0 400 Bad Request. Via: SIP/2.0/UDP MyProxyIP;branch=z9hG4bK139a.a74691c5.0. Via: SIP/2.0/UDP CustomerIP:5060;rport=5060;received=CustomerIP;branch=z9hG4bk0102581015080138147057$ From: "SERVER BR" <sip:001184517324@CustomerIP:5060>;tag=010258101508381470562. To: <sip:890912551128650136@MyProxyIP>. Call-ID: GW6d4c945d2c573c27@CustomerIP. CSeq: 1 INVITE. Content-Length: 0.

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[Kamailio-Users] Drouting Question
by Sven Schulz 12 Feb '10

12 Feb '10

Hello, Ive been testing Kamailio 3.0 and drouting. In the documentation it states (Section 1.5 Routing Rule Processing) ³The module will execute serial forking for each address in chain. The next address in chain is used only if the previously has failed.² What exactly defines a failure? Does it use ping or OPTIONS to determine if a GW is busy or down? Id like to know exactly how drouting figures this out. Sven Schulz Penn State University Telecommunications and Network Services 814.865.6116 sip:sven@psu.edu

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[Kamailio-Users] acc module
by Denis Putyato 12 Feb '10

12 Feb '10

Hello everybody! I have a such problem with ACC module. Calls was accounted but only INVITE and BYE requests has appeared in acc table. ACK is absent. Kamailio 3.0 was installed from GIT. Thank you for any help loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "tm.so" loadmodule "sl.so" loadmodule "tmx.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "avpops.so" loadmodule "siptrace.so" loadmodule "dialplan.so" loadmodule "lcr.so" loadmodule "sqlops.so" loadmodule "carrierroute.so" loadmodule "drouting.so" loadmodule "kex.so" # ----- acc params ----- modparam("acc", "early_media", 0) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 1) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "failed_transaction_flag", 29) modparam("acc", "db_table_acc", "acc") modparam("acc", "db_table_missed_calls", "missed_calls") modparam("acc", "acc_method_column", "method") modparam("acc", "acc_from_tag_column", "from_tag") modparam("acc", "acc_to_tag_column", "to_tag") modparam("acc", "acc_callid_column", "callid") modparam("acc", "acc_sip_code_column", "sip_code") modparam("acc", "acc_sip_reason_column", "sip_reason") modparam("acc", "acc_time_column", "time") modparam("acc", "db_url", "xxxxxxxxxxxxxxxxxxxxxxxx") modparam("acc", "db_extra", "src_user=$avp(i:2);src_domain=$si;dst_ouser=$avp(i:1);dst_user=$rU;dst_doma in=$rd") route { sip_trace(); setflag(30); if (has_totag()) { if (loose_route()) { if (is_method("BYE")) { setflag(1); setflag(29); } if (is_method("ACK")) { xlog("L_NOTICE", "ACK detected"); # I see this line in syslog setflag(1); setflag(29); } route("relay"); exit; } else { if ( is_method("ACK") ) { if ( t_check_trans() ) { setflag(1); route("relay"); exit; } else { exit; } } } }

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[Kamailio-Users] acc & radius
by Kris Amy 12 Feb '10

12 Feb '10

Hi all, Is there anything special required to get radius support into the acc module? I have tried the following... make distclean make config modules_include="mediaproxy auth_radius group_radius uri_radius misc_radius" group_include="mysql standard" make deb But alas it still seems to not have it. I have confirmed that the radius libraries are installed aswell. ii libradiusclient-ng-dev 0.5.6-1 Enhanced RADIUS client library development f ii libradiusclient-ng2 0.5.6-1 Enhanced RADIUS client library Cheers, Kris

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[Kamailio-Users] Extra byte in Content-length header
by Vikram Ragukumar 11 Feb '10

11 Feb '10

Hello, I am testing a cellphone software that registers with an Asterisk server through an outbound proxy (Kamailio+rtpproxy). I am capturing network traffic using wireshark and i notice that the SIP Register message from the cellphone has an extra byte in the Content-Length header, which causes Kamailio to discard the message with the following error output : - ERROR:core:parse_content_length: parse error near char [13][^M] - ERROR:core:get_hdr_field: bad content_length header Could it be that this byte was appended somewhere downstream(possibly at the wireless router..) or can i be sure that the byte was added during SIP Message construction by the cellphone software ? Thanks in advance, Vikram.

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Re: [Kamailio-Users] Siremis Accounting
by Daniel-Constantin Mierla 11 Feb '10

11 Feb '10

Hello, please use public mailing list for such questions. Direct messages get long delayed or lost in most of the cases. Mailing list has plenty of developers and community users that can help you faster. Besides that, new people can benefit of older discussions. On 2/11/10 2:14 AM, Kris Amy wrote: > Hi Daniel, > > It would seem to me that the stored procedure on > http://siremis.asipto.com/install-accounting/ is for PostgreSQL. I'm > going to try and getting it working with MySQL. it is for mysql. Cheers, Daniel > > Just thougth you may want to update the page. > > -- Daniel-Constantin Mierla eLearning class for Kamailio 3.0.0 * http://www.asipto.com/

4 3

Re: [Kamailio-Users] Siptrace and Kamailio 3.0.0
by Daniel-Constantin Mierla 11 Feb '10

11 Feb '10

On 2/11/10 11:16 AM, Kris Amy wrote: > On Thu, Feb 11, 2010 at 8:09 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > ok, then the only point in mod_init of siptrace that causes > initialization error without printing a log message is when you > haven't set the trace_flag parameter: > http://kamailio.org/docs/modules/3.0.x/modules_k/siptrace.html#id2520001 > > Cheers, > Daniel > > > Hi Daniel, > > That fixed it. > > Perhaps there is a bug in that it's not picking up the default of 0? in the sources the default is -1, which means disabled -- docs needs to be updated. Thanks for pointing, Daniel -- Daniel-Constantin Mierla eLearning class for Kamailio 3.0.0 *http://www.asipto.com/

1 0

[Kamailio-Users] SDP IP address not rewritten
by Vikram Ragukumar 11 Feb '10

11 Feb '10

Hello, I am running Kamailio version 1.5 and rtpproxy version 1.2. I have enabled NAT support in Kamailio.cfg. I have a UA behind NAT (UA IP:192.168.2.101), and the UA registers with an Asterisk server (Public IP) through an outbound proxy running Kamailio and rtpproxy with Public IP. Registration succeeds, however when making an outbound call from the UA to a cellphone, there is only one sided voice. This is happening because Asterisk is addressing rtp streams to the UA's local IP address instead of its PublicIP:port. Why is the IP address not being rewritten in the SDP ? Is there an alternate way of invoking rtpproxy (with flags) such that the IP address in the SDP is rewritten ? Thanks and Regards, Vikram. PS : Below is the main routing section of my kamailio.cfg route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # NAT detection route(4); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) route(2); if (is_method("REGISTER")) { if (isflagset(5)) { setbflag(6); # setbflag(7); } if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { if (check_route_param("nat=yes")) { setbflag(6); } if (isflagset(5) || isbflagset(6)) { route(5); } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); setflag(6);#VR } setflag(5); } return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset(6)) { fix_nated_contact(); } } failure_route[1] { if (is_method("INVITE") && (isbflagset(6) || isflagset(5))) { unforce_rtp_proxy(); } if (t_was_cancelled()) { exit; } }

2 4

[Kamailio-Users] [kamailio-users] Method NOTIFY not supported ...
by Chandrakant Solanki 11 Feb '10

11 Feb '10

Hi below line is in my configuration... loadmodule "pua.so" modparam("pua", "db_url", "mysql:// kamailio:kamailio@172.16.0.10:3306/kamailio") modparam("pua", "db_table", "pua") ---- ---- if(is_method("NOTIFY") && src_ip==172.16.0.15) { xlog("L_INFO", "NOTIFY Method found\n"); if(!t_relay()){ sl_reply_error(); } } ---- ---- I found following error in log... Feb 10 15:14:40 kamailio_1 /sbin/kamailio[5597]: Method not supported - M=NOTIFY RURI=sip:172.16.0.15 F=sip:2222222@pbxusers.net<sip%3A2222222(a)pbxusers.net>T=sip:172.16.0.15 IP=172.18.100.89 ID= 17dd4bb9-81d132a0(a)172.18.100.89 -- Regards, Chandrakant Solanki

2 7

[Kamailio-Users] Fwd: Siptrace and Kamailio 3.0.0
by Kris Amy 11 Feb '10

11 Feb '10

---------- Forwarded message ---------- From: Kris Amy <kris(a)amy.id.au> Date: 2010/2/11 Subject: Re: [Kamailio-Users] Siptrace and Kamailio 3.0.0 To: Daniel-Constantin Mierla <miconda(a)gmail.com> On Thu, Feb 11, 2010 at 8:09 PM, Daniel-Constantin Mierla <miconda(a)gmail.com > wrote: > Hello, > > ok, then the only point in mod_init of siptrace that causes initialization > error without printing a log message is when you haven't set the trace_flag > parameter: > http://kamailio.org/docs/modules/3.0.x/modules_k/siptrace.html#id2520001 > > Cheers, > Daniel >

1 0

[Kamailio-Users] Siptrace and Kamailio 3.0.0
by Kris Amy 11 Feb '10

11 Feb '10

Hi, Just wondering if anyone is using Siptrace with Kamailio 3.0.0. The module fails to load and I have no idea why (or how to get more information than from debug = 99). 0(13453) DEBUG: <core> [sr_module.c:871]: DEBUG: init_mod: siptrace 0(13453) ERROR: <core> [sr_module.c:874]: init_mod(): Error while initializing module siptrace ERROR: error while initializing modules Cheers, Kris

2 4

[Kamailio-Users] kamailio crashes during connection attempt
by Dmitri 10 Feb '10

10 Feb '10

Hello list, I have fresh installation of Debian Squeeze on X86_64. I downloaded source tarball http://www.kamailio.org/pub/kamailio/latest/src/kamailio-3.0.0_src.tar.gz and did 'make deb' there. So, I got following debian packages: kamailio_3.0.0_amd64.deb, kamailio-berkeley-modules_3.0.0_amd64.deb, kamailio-carrierroute-modules_3.0.0_amd64.deb kamailio-cpl-modules_3.0.0_amd64.deb, kamailio-ldap-modules_3.0.0_amd64.deb, kamailio-memcached-modules_3.0.0_amd64.deb kamailio-mysql-modules_3.0.0_amd64.deb, kamailio-nth_3.0.0_amd64.deb, kamailio-perl-modules_3.0.0_amd64.deb kamailio-postgres-modules_3.0.0_amd64.deb, kamailio-presence-modules_3.0.0_amd64.deb, kamailio-purple-modules_3.0.0_amd64.deb kamailio-radius-modules_3.0.0_amd64.deb, kamailio-snmpstats-modules_3.0.0_amd64.deb, kamailio-tls-modules_3.0.0_amd64.deb kamailio-unixodbc-modules_3.0.0_amd64.deb, kamailio-utils-modules_3.0.0_amd64.deb, kamailio-xmlrpc-modules_3.0.0_amd64.deb kamailio-xmpp-modules_3.0.0_amd64.deb I installed packages: kamailio_3.0.0_amd64.deb, kamailio-mysql-modules_3.0.0_amd64.deb, kamailio-presence-modules_3.0.0_amd64.deb, kamailio-xmlrpc-modules_3.0.0_amd64.deb Created database, changed passwords for openser and openserro users in MySQL as well as in kamailio config files, enabled MySQL. DUMP_CORE=yes in /etc/default/kamailio, also added '-w /var/run/kamailio' to options. Created test user with kamctl. Started kamailio with init script: /etc/init.d/kamailio start: Feb 7 22:05:10 kamailio kamailio: INFO: <core> [tcp_main.c:4150]: init_tcp: using epoll_lt as the io watch method (auto detected) Feb 7 22:05:10 kamailio /usr/sbin/kamailio[2678]: INFO: usrloc [hslot.c:53]: locks array size 512 Feb 7 22:05:10 kamailio /usr/sbin/kamailio[2678]: INFO: <core> [udp_server.c:166]: INFO: udp_init: SO_RCVBUF is initially 124928 Feb 7 22:05:10 kamailio /usr/sbin/kamailio[2678]: INFO: <core> [udp_server.c:217]: INFO: udp_init: SO_RCVBUF is finally 262142 Feb 7 22:05:10 kamailio /usr/sbin/kamailio[2678]: INFO: <core> [udp_server.c:166]: INFO: udp_init: SO_RCVBUF is initially 124928 Feb 7 22:05:10 kamailio /usr/sbin/kamailio[2678]: INFO: <core> [udp_server.c:217]: INFO: udp_init: SO_RCVBUF is finally 262142 Feb 7 22:05:10 kamailio /usr/sbin/kamailio[2678]: INFO: <core> [udp_server.c:166]: INFO: udp_init: SO_RCVBUF is initially 124928 Feb 7 22:05:10 kamailio /usr/sbin/kamailio[2678]: INFO: <core> [udp_server.c:217]: INFO: udp_init: SO_RCVBUF is finally 262142 Feb 7 22:05:10 kamailio /usr/sbin/kamailio[2695]: INFO: ctl [io_listener.c:224]: io_listen_loop: using epoll_lt io watch method (config) Then I tried to connect to kamailio server with softphone Zoiper, and kamailio crashes immediately: Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2700]: : <core> [pass_fd.c:283]: ERROR: receive_fd: EOF on 21 Feb 7 22:05:20 kamailio kernel: [ 1051.407643] kamailio[2689]: segfault at c2 ip 00007fa8b5947b19 sp 00007fff48ff9700 error 4 in tm.so[7fa8b5917000+72000] Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2678]: ALERT: <core> [main.c:722]: child process 2689 exited by a signal 11 Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2678]: ALERT: <core> [main.c:725]: core was not generated Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2678]: INFO: <core> [main.c:737]: INFO: terminating due to SIGCHLD Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2691]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2699]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2698]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2687]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2697]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2690]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2696]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2688]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2695]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2680]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2694]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2693]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2692]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2686]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2685]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2681]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2684]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2682]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2700]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2683]: INFO: <core> [main.c:788]: INFO: signal 15 received Feb 7 22:05:20 kamailio /usr/sbin/kamailio[2678]: ERROR: ctl [ctl.c:379]: ERROR: ctl: could not delete unix socket /tmp/ser_ctl: Operation not permitted (1) Feb 7 22:05:20 kamailio kernel: [ 1051.432646] kamailio[2678]: segfault at 12 ip 00007fa8b592ae43 sp 00007fff48ff9f40 error 4 in tm.so[7fa8b5917000+72000] For unknown reason core was not generated. Any ideas ? Many thanks in advance, Dmitri

5 35

[Kamailio-Users] save() function
by Kris Amy 10 Feb '10

10 Feb '10

Hi All, It was to my understanding that this function was part of the core and thus didn't come from a module. The relevant code snippet is if(!save("location")) { xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); exit; } The error I'm getting is.. 0(20467) DEBUG: <core> [sr_module.c:516]: find_export_record: <save> not found 0(20467) DEBUG: <core> [sr_module.c:516]: find_export_record: <save> not found 0(20467) : <core> [cfg.y:3329]: parse error in config file /etc/kamailio/kamailio.cfg, line 656, column 21: unknown command, missing loadmodule? Cheers, Kris

2 2

[Kamailio-Users] Carrierroute issue
by michel freiha 10 Feb '10

10 Feb '10

Hi all, I'm using carrierroute module and ie seems that it's not working well...Suppose I have 2 entries in carrierroute table...Entry 1 with scan prefix 00 and prob 0 and a second entry with scan prefix 00 and prob=1...As soon as i make a call, this call will be forwarded to the entry with prob 0 because i guess it appears before the one with prob 1 in the table... When starting Kamailio I got the below: Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix 1, prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: domain 0 not found, add it Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_tree: tree default has 1 trees Jul 29 12:30:11 [23143] DBG:carrierroute:add_route_tree: tree 0Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: created route tree: 0, with id 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix , prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix 011, prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix 00, prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix 111, prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix 00, prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix , prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix 011, prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix 011, prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: adding prefix , prob -14931260.000000 Jul 29 12:30:11 [23143] INFO:carrierroute:get_route_tree: found domain 0 Jul 29 12:30:11 [23143] INFO:carrierroute:add_route: found route, now adding Jul 29 12:30:11 [23143] INFO:carrierroute:add_route_rule: no backed up rules It seems that kamailio is not reading probability entry in carrier route table Please find below carrierroute table structure: CREATE TABLE IF NOT EXISTS `carrierroute` ( `id` int(10) unsigned NOT NULL auto_increment, `carrier` int(10) unsigned NOT NULL default '0', `domain` varchar(64) NOT NULL default '', `scan_prefix` varchar(64) NOT NULL default '', `flags` int(11) unsigned NOT NULL default '0', `mask` int(11) unsigned NOT NULL default '0', `prob` float NOT NULL default '0', `strip` int(11) unsigned NOT NULL default '0', `rewrite_host` varchar(128) NOT NULL default '', `rewrite_prefix` varchar(64) NOT NULL default '', `rewrite_suffix` varchar(64) NOT NULL default '', `description` varchar(255) default NULL, PRIMARY KEY (`id`) ) ENGINE=MyISAM DEFAULT CHARSET=latin1 AUTO_INCREMENT=23 ; Can someone help me to find this issue? Regards

5 13

[Kamailio-Users] check_user_blacklist function
by Antonio Reale 10 Feb '10

10 Feb '10

Hi all, the function check_user_blacklist in kamailio 1.4 returns -1 (FALSE based on the doc) "if a match is found and it is not set to whitelist", otherwise it returns 1 (TRUE). So the function returns the same value if a match is found and is set to whitelist and if there are no matches. I need to differentiate the two situations. The userblacklist module in kamailio 3.0 has a new function "check_user_whitelist" that can solve my problem, but at this time I can't upgrade. I solved the problem, changing only the return value in case of whitelisted entry (userblacklist.c file), but I don't know if it is a safe (and suffice) modification. Below the piece of code in the userblacklist.c: if (dt_longest_match(dt_root, ptr, &whitelist) >= 0) { if (whitelist) { /* LM_ERR("whitelisted"); */ return 2; /* found, but is whitelisted */ <===== CHANGED HERE. IT WAS return 1 } } else { /* LM_ERR("not found"); */ return 1; /* not found is ok */ } Thanks. Regards. -- Antonio

2 2

[Kamailio-Users] LCR
by Kris Amy 10 Feb '10

10 Feb '10

Hi All, I've been ripping my hair out for a little with load_gws(). How do I know what the lcr_id is and the caller_uri? I would have thought that $fu for caller_uri would have been sufficient but it seems not. Cheers, Kris

3 2

[Kamailio-Users] kamctl and lcr table
by Kris Amy 10 Feb '10

10 Feb '10

Hi, Could we get the lcr-create.sql updated to have the ping field? Cheers, Kris

2 3

[Kamailio-Users] irc devel meeting
by Daniel-Constantin Mierla 09 Feb '10

09 Feb '10

Hello, I was thinking of having an IRC meeting to discuss next steps within sip router. Now that 3.0 branches are out for a while, would be good to plan a bit 3.1. For me Friday (or even Thursday if not too early for others to make arrangements) this week would be good, say 14:00 or 15:00UTC to allow fair time for US. If this date does not fit, please make proposals. The channel is #sip-router on freenode.net. Cheers, Daniel -- Daniel-Constantin Mierla eLearning class for Kamailio 3.0.0 * http://www.asipto.com/

1 0

[Kamailio-Users] problems with mi_xmlrpc command
by Anton Roman 09 Feb '10

09 Feb '10

Hi all, I'm trying to send xmlrpc commands to Kamailio with Siremis but I get the error below: Fault Code: 5 Fault Reason: Didn't receive 200 OK from remote server. (HTTP/1.1 400 Bad Request) I get the same error when I manually send the xmlrpc command with curl. On the other hand, mi_xmlrpc module seems to be correctly loaded, but nothing is written in the log file (set with modparam("mi_xmlrpc", "log_file", "/var/log/abyss.log")). Does anybody know where the problem might be? Best regards, thanks in advance. Anton Roman These are the traces of the xmlrpc message flow: T +11.569284 127.0.0.1:37821 -> 127.0.0.1:8080 [AP] POST HTTP/1.0 User-Agent: PEAR XML_RPC Host: 127.0.0.1 Content-Type: text/xml Content-Length: 135 <?xml version="1.0" encoding="UTF-8"?> <methodCall> <methodName>system.listMethods</methodName> <params> </params> </methodCall> ## T +0.000363 127.0.0.1:8080 -> 127.0.0.1:37821 [AP] HTTP/1.1 400 Bad Request # T +0.000146 127.0.0.1:8080 -> 127.0.0.1:37821 [AP] Content-type: text/html # T +0.001982 127.0.0.1:8080 -> 127.0.0.1:37821 [AFP] Connection: close Date: Mon, 08 Feb 2010 14:04:08 GMT Server: XMLRPC_ABYSS/1.06 <HTML><HEAD><TITLE>Error 400</TITLE></HEAD><BODY><H1>Error 400</H1><P>Bad Request</P><p><HR><b><i><a href="http://xmlrpc-c.sourceforge.net">ABYSS Web Server for XML-RPC For C/C++</a></i></b> version 1.06<br></p></BODY></HTML> ####

2 2

[Kamailio-Users] Stateless forward
by Antonio Reale 09 Feb '10

09 Feb '10

Hi all, I'm trying to setup a very basic scenario with kamailio 3.0. It is placed in front of a softswitch and it has to modify some headers before forward (stateless) messages to softswitch. If I use route { rewritehost("192.168.10.1"); t_relay(); } all work, but if I want to do that stateless and I use forward() or send() instead of t_relay(), I get a syntax error: "bad forward argument" or "bad send argument". If I use ... forward("udp:192.168.10.1:5060"); ... I see in the log /usr/local/sbin/kamailio[3675]: : <core> [proxy.c:271]: ERROR: mk_proxy: could not resolve hostname: "udp:192.168.10.1:5060" /usr/local/sbin/kamailio[3675]: ERROR: <core> [route.c:1026]: fixing failed (code=-478) at cfg:/usr/local/etc/kamailio/kamailio.cfg:77 Same error using the long format of send() function. I can't see any particular hint on the documentation, what I'm missing? Thank you. Best regards. -- Antonio

3 6

[Kamailio-Users] Install Documentation
by Kris Amy 09 Feb '10

09 Feb '10

Hi All, Just thought that you might want to put down that libpcre is required for lcr in the install document. (Since the other main modules have their requirements listed). Cheers, Kris

2 1

[Kamailio-Users] change domain at to header
by Adnan Çelikel 09 Feb '10

09 Feb '10

Dear All, I want to change a field (Sip to address host part) at sip message header. I think avpops will solve this.If anybody has a sample confg and send to me I will be very glad.Regards,Adnan. #---------------------------------------------------- #7777 CALL ROUTING #-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-* if(is_method("INVITE") && (uri=~"sip:7777")){ rewritehostport("192.168.1.10:5060"); route(1); exit; } #---------------------------------------------------- at this rule, the invite is going to 192.168.1.10 with the host part as abc.com, we want to send with another domain. Instead of abc.com domain, I want to use 192.168.1.10 domain ; To:<sip:7777@abc.com> To:<sip:7777@192.168.1.10> _________________________________________________________________ Windows Live Hotmail: Arkadaşlarınız Facebook'taki güncellemelerinizi doğrudan Hotmail®'den alır. http://www.microsoft.com/windows/windowslive/see-it-in-action/social-networ…

5 11

Routing Problems
by Andreas Rehbein 08 Feb '10

08 Feb '10

Hello, we use Kamailio 3.0 on a Red Hat EL5.4 system with openssl 0.9.8e (the current Red Hat OpenSSL version). We want to use Kamailio 3.0 in front of our Call Server (OpenSER 1.3.2) as a security gateway. So the Call Server should only deal tcp, while the Security Gateway terminates the TLS Data which he receives from the User Agents but forwards the SIP-Messages via TCP to the Call Server. UA ---SIP_over_TLS---> Security Gateway (Kam3.0) ---SIP_over_TCP---> Call Server (OpenSER1.3.2) UA<---SIP_over_TLS--- Security Gateway (Kam3.0) <---SIP_over_TCP--- Call Server (OpenSER1.3.2) Nearly everything works fine in this scenario: User Agents are able to register and when they are sending INVITES the callee receives it. The problem we have right now is: if Kamailio 3.0 receives BYE we get 477 Unfortunatly error on sending to next hop occured. It seems that the messages are ok... I attached a text file with the BYE message and the errors. Explanation for the text file: * UA1: 192.168.0.126 * UA2: 192.168.0.176 *Security Gateway (Kam3.0): 192.168.0.89 *Call Server (OpenSER1.3.2): 192.168.0.106 Do you have any suggestions? Thank you very much in advance! Regards Andreas Dipl.-Ing. (FH) Andreas Rehbein FH Frankfurt am Main - University of Applied Sciences FB 2 - Informatik und Ingenieurwissenschaften Forschungsgruppe für Telekommunikationsnetze Kleiststraße 3 60318 Frankfurt am Main Tel.: +49 (0) 69 1533 3613 Fax: +49 (0) 69 1539 5224 961 E-Mail: <mailto:rehbein@e-technik.org> rehbein(a)e-technik.org Internet: <http://www.e-technik.org/frame_forschung.htm> http://www.e-technik.org/frame_forschung.htm

2 2

[Kamailio-Users] Maximum retries exceeded on transmission
by Daniel-Constantin Mierla 06 Feb '10

06 Feb '10

On Sat, Feb 6, 2010 at 11:10 AM, Jon Farmer <viperdudeuk(a)gmail.com> wrote: > Sorry it was a problem I had with gmail. > It happens to me from time to time, but I couldn't find the real cause, whether is gmail or thunderbird. Thunderbird hangs a while, reports a delivery error, then message appears twice... Daniel > On 5 Feb 2010 12:21, "Iñaki Baz Castillo" <ibc(a)aliax.net> wrote: > > 2010/2/5 Jon Farmer <viperdudeuk(a)gmail.com>: > > Hi > > Please don't send the same smail twice. Thanks. > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > -- Daniel-Constantin Mierla http://www.asipto.com -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] Maximum retries exceeded on transmission
by Daniel-Constantin Mierla 06 Feb '10

06 Feb '10

Hello, On Fri, Feb 5, 2010 at 10:56 AM, Jon Farmer <viperdudeuk(a)gmail.com> wrote: > Hi > > I am getting this dreaded error on a asterisk box when call from one > UA to another using Asterisk as a media relay and openser as a SBC and > proxy. > > chan_sip.c:3779 retrans_pkt: Maximum retries exceeded on transmission > 1310650592(a)10.100.0.101 > > If I make a call from a UA to the PSTN through Asterisk all is > fine. This only happens when calling from one UA to another on the > same LAN. > > I have posted an example sip trace here http://pastebin.com/m2262b0ef > ACK is not accepted as I could spot quickly, 200ok being retransmitted. Probably there is something wrong with dialog attributes, after Fosdem I can check it better. Daniel > My openser.cfg is here http://pastebin.com/d5e84aaf7 > > I realise this is a issue with Asterisk but I have not been able to > find a fix. Therefore I figure there might be something I can do on > openser. The problem seems to be with a ACK not being returned but my > SIP knowledge is not good enough to know how to fix it. Any help is > greatly appreciated. > > Regards > > Jon > -- > Jon Farmer > Tel: 07795 118140 > Email: viperdudeuk(a)gmail.com > Twitter: @viperdudeuk > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > -- Daniel-Constantin Mierla http://www.asipto.com -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] Maximum retries exceeded on transmission
by Jon Farmer 06 Feb '10

06 Feb '10

Hi I am getting this dreaded error on a asterisk box when call from one UA to another using Asterisk as a media relay and openser as a SBC and proxy. chan_sip.c:3779 retrans_pkt: Maximum retries exceeded on transmission 1310650592(a)10.100.0.101 If I make a call from a UA to the PSTN through Asterisk all is fine. This only happens when calling from one UA to another on the same LAN. I have posted an example sip trace here http://pastebin.com/m2262b0ef My openser.cfg is here http://pastebin.com/d5e84aaf7 I realise this is a issue with Asterisk but I have not been able to find a fix. Therefore I figure there might be something I can do on openser. The problem seems to be with a ACK not being returned but my SIP knowledge is not good enough to know how to fix it. Any help is greatly appreciated. Regards Jon -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

2 2

[Serusers] RECEIVED column in SER.LOCATION table
by Bill McNamara 05 Feb '10

05 Feb '10

Question, when does a value get inserted into the ser.location table’s received column. We have some UA’s with a contact like value and some with null values – different devices (Polycom 331/450/55, Grandstream 286/486) behind the same firewall. I’m just curious to know why some have a received value and some don’t. Thanks, Bill SER 2.0 Linux 2.6.9-78 x86_64 MySQL 5.1.30

2 1

[Kamailio-Users] CRITICAL:dialog:log_next_state_dlg
by alex pappas 05 Feb '10

05 Feb '10

Dear users, I'm doing some stress test with kamailio 1.5. When I'm at 50 CPS a see some strange messages: /usr/sbin/kamailio[12754]: CRITICAL:dialog:log_next_state_dlg: bogus event 2 in state 3 for dlg 0x2b0b911831b0 [3345:993496027] with clid '297-19089@IP-address' and tags '19089SIPpTag00297' '15519SIPpTag01304' Does anyone before experienced this before? Thanks Alex

1 0

[Kamailio-Users] Maximum retries exceed on transmission
by Jon Farmer 05 Feb '10

05 Feb '10

Hi I am getting this dreaded error on a asterisk box when call from one UA to another using Asterisk as a media relay and openser as a SBC and proxy. If I make a call from a UA to the PSTN through Asterisk all is fine. This only happens when calling from one UA to another on the same LAN. I have posted an example sip trace here http://pastebin.com/m2262b0ef My openser.cfg is here http://pastebin.com/d5e84aaf7 I realise this is a issue with Asterisk but I have not been able to find a fix. Therefore I figure there might be something I can do on openser. The problem seems to be with a ACK not being returned but my SIP knowledge is not good enough to know how to fix it. Any help is greatly appreciated. Regards Jon -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

3 5

[Kamailio-Users] new: interactive debugger for config
by Daniel-Constantin Mierla 05 Feb '10

05 Feb '10

Hello, the new debugger module can be used to do runtime debugging of SIP Router config file, in similar manner of gdb. The module is controlled via the RPC interface, therefore investigation can be done from a remote site. That makes it very handy to use with sercmd as well. One features is to print a log message for each action executed in the config for a SIP message, showing the execution path. The option can be enabled/disabled per process. The other one is step by step execution of config file. When a SIP message message comes it, SIP Router process will stop at first action and wait for commands, like: execute the action, evaluate a pseduo-variable, print to syslog a pseudo-variable, remove the breakpoints and continue execution without interruption, ... See the readme for more: http://sip-router.org/docbook/sip-router/branch/master/modules/debugger/deb… It is still some stuff to complete, some known issues to remove, but the module is usable and may help some of you to troubleshoot migrations from older versions to 3.0 (use master branch until your config is updated and running fine, then use it with stable 3.0). Hope is going to be useful and waiting for feedback to improve. Of course, contributions are most welcome! Cheers, Daniel -- Daniel-Constantin Mierla eLearning class for Kamailio 3.0.0 Starting Feb 8, 2010 * http://www.asipto.com/

4 5

[Kamailio-Users] Kamailio/SIPRouter Training
by Bruce McAlister 05 Feb '10

05 Feb '10

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi All, Are there any planned dates for Kamailio/SIPRouter training events around Europe in the next couple months. Looking forward to your replies..... Thanks Bruce -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQEcBAEBAgAGBQJLa+kVAAoJEGJOkoTLLvBntMUIAJZEb7AaFDAdwKmTxlyokq30 A5eebX8mTUXk9fkzcHX0IoRRrl8eQKGqUbIl2312oQZtxJrffTksY3xiFFOdpBCp 78RDmKjLvmhAwq7jYPExk91BiV4no+dcAKuHsntwz8EfW9uHd34NRlp+TEc8Q2A8 QB7Ztfq8J+8KjGWJoRE27gqiHEIwZp7F7/rvn4N/VAuaTcaTtYhaiI52iIOA+aNM UTgkGMVzjjpPrwXh7C/DZERKThTifF01PzCx1DSlPiMczrGCGxHy8NEgAaHHsXb9 yrrtd/B4qWAbaYT8mRXisw9hNaGktdmCAA6Ksrq/xz/qjjVa84Tflo9d754PTGw= =kgI0 -----END PGP SIGNATURE-----

1 0

[Kamailio-Users] load_credentials problem
by Dubravko Caric 05 Feb '10

05 Feb '10

Hi all, i have a problem when using AVPs which are loaded on client registration by the "load_credentials". in the "subscriber" table i have "rpid" column (value 12345), "email address" (bob(a)example.com) and "acl" (value 3). when client registers, these three AVPs are loaded as you can see in debug. But when i want to use these AVPs it seems that they are empty. i don't understand if they are ready for use or do i need to load them and how? i'm not doing any actions on AVPs so i could by mistake delete or replace the original ones... version of kamailio is 1.5.1 Thanks Dubravko modparam("auth_db", "load_credentials", "$avp(i:66)=rpid;email_address;$avp(i:67)=acl") DBG:db_mysql:db_mysql_get_columns: allocate 16 bytes for RES_NAMES[0] at 0x783400 DBG:db_mysql:db_mysql_get_columns: RES_NAMES(0x783400)[0]=[password] DBG:db_mysql:db_mysql_get_columns: use DB_STRING result type DBG:db_mysql:db_mysql_get_columns: allocate 16 bytes for RES_NAMES[1] at 0x783420 DBG:db_mysql:db_mysql_get_columns: RES_NAMES(0x783420)[1]=[acl] DBG:db_mysql:db_mysql_get_columns: use DB_INT result type DBG:db_mysql:db_mysql_get_columns: allocate 16 bytes for RES_NAMES[2] at 0x783440 DBG:db_mysql:db_mysql_get_columns: RES_NAMES(0x783440)[2]=[email_address] DBG:db_mysql:db_mysql_get_columns: use DB_STRING result type DBG:db_mysql:db_mysql_get_columns: allocate 16 bytes for RES_NAMES[3] at 0x783460 DBG:db_mysql:db_mysql_get_columns: RES_NAMES(0x783460)[3]=[rpid] DBG:db_mysql:db_mysql_get_columns: use DB_STRING result type DBG:core:db_allocate_rows: allocate 16 bytes for rows at 0x783480 DBG:core:db_allocate_row: allocate 128 bytes for row values at 0x7834a0 DBG:core:db_str2val: converting STRING [1234] DBG:core:db_str2val: converting INT [3] DBG:core:db_str2val: converting STRING [bob(a)example.com] DBG:core:db_str2val: converting STRING [12345] DBG:auth_db:get_ha1: HA1 string calculated: 649766733fe475191f6c32d338aae51d DBG:auth:check_response: our result = 'f929dd9729d3c9447c4a4d632a762c85' DBG:auth:check_response: authorization is OK DBG:auth:post_auth: nonce index= 0 DBG:auth_db:generate_avps: set int AVP ""/67 = 3 DBG:auth_db:generate_avps: set string AVP "email_address"/0 = "bob(a)example.com" DBG:auth_db:generate_avps: set string AVP ""/66 = "12345" DBG:core:db_free_columns: freeing 4 columns ################# if (is_method("INVITE")) { xlog("L_INFO", "RPID AVP VALUE $avp(i:66)\n"); xlog("L_INFO", "EMAIL AVP VALUE $avp(s:email_address)\n"); xlog("L_INFO", "ACL AVP VALUE $avp(i:67)\n"); } ./kamailio[21687]: PRID AVP VALUE <null> ./kamailio[21687]: EMAIL AVP VALUE <null> ./kamailio[21687]: ACL AVP VALUE <null>

4 5

[Kamailio-Users] Kamailio v1.5.4 and v1.4.5 released
by Daniel-Constantin Mierla 05 Feb '10

05 Feb '10

Hello, Kamailio version 1.5.4 is out, built from latest svn version of SVN branch 1.5. All details about it are available at: http://www.kamailio.org/mos/view/News/NewsItem/Kamailio-v1.5.4-Released/ Also Kamailio v1.4.5 was released to mark the end of official maintenance for SVN branch 1.4. Details at: http://www.kamailio.org/mos/view/News/NewsItem/Kamailio-v1.4.5-Released/ If you are using a version from these release series, consider updating to the last one. Personally I recommend that you upgrade to the latest stable branch release v3.0.0 -- you get access lot of new features and improvements. Cheers, Daniel

3 5

Re: [Kamailio-Users] [kamailio-users] can't connect with remote database
by Chandrakant Solanki 05 Feb '10

05 Feb '10

On Thu, Feb 4, 2010 at 5:38 PM, Chandrakant Solanki < solanki.chandrakant(a)gmail.com> wrote: > > > On Thu, Feb 4, 2010 at 5:31 PM, marius zbihlei <marius.zbihlei(a)1and1.ro>wrote: > >> Chandrakant Solanki wrote: >> >>> Hi >>> >>> I have 3 server with 172.16.0.10/13/14 <http://172.16.0.10/13/14> IP >>> address. >>> >>> >>> My DB is located for 172.16.0.10. I tried to connect mysql from terminal, >>> it works great. >>> >>> But when I start kamailio it gives me following error.. >>> >>> Hello >> >> Just to double check Have you tried: >> >> mysql --user=openser -p --host=172.16.0.10 openser >> >> from 172.16.0.13 and it works? >> >> From localhost afaik mysql client might use Unix socks(/tmp/mysql.sock) . >> Use should double check that your MySQL server allows TCP connections >> >> Cheers >> >> Marius >> > > Hi mysql --user=openser -p --host=172.16.0.10 openser --socket=/var/lib/mysql/mysql. i have used this command from 172.16.0.13/14 it will works fine and connect to DB on 172.16.0.10 -- Regards, Chandrakant Solanki

1 0

[Kamailio-Users] OK, I've had enough "Ausente"
by Alex Balashov 04 Feb '10

04 Feb '10

Basta. Ya me has dicho que no estás. -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/

3 3

[Kamailio-Users] Openser with MySql Cluster
by Ricardo Coelho 04 Feb '10

04 Feb '10

Hi Right now I have a single openser server configured to work with a single mysql database. I want to enhance availability by adding another openser server and use Mysql Cluster. When failover occurs, despite the fact that he databases have the same data, on one of the softphones i receive a message saying "User not found" and on the other softphone i receive a message saying "Security check failed". Do I need to change something in openser's configuration to make it work with mysql cluster? Thanks in advanced

3 4

Re: [Kamailio-Users] break a call?
by BERGANZ Francois 04 Feb '10

04 Feb '10

Where can I activate it? -----Message d'origine----- De : Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Envoyé : jeudi 4 février 2010 14:09 À : BERGANZ Francois Objet : Re: [Kamailio-Users] break a call? probably the problem is that you can not use tm module functions in onreply_route. You have to handle the responses in the failure route (every final response >=300 triggers the failure_route (if activated)). regards klaus Am 04.02.2010 13:40, schrieb BERGANZ Francois: > Line 254 is t_reply("406", "Not Acceptable"); > > > > -----Message d'origine----- > De : Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Envoyé : jeudi 4 février 2010 13:08 > À : BERGANZ Francois > Objet : Re: [Kamailio-Users] break a call? > > > > Am 04.02.2010 12:37, schrieb BERGANZ Francois: >> I have that error: >> >> >> onreply_route[1] { >> if(t_check_status("3[0-9][0-9]")){ >> xlog("L_ERR", "3XX!!!!!!!!!!!\n"); >> t_reply("406", "Not Acceptable"); >> exit; >> } >> } >> >> Feb 4 12:36:41 [4584] CRITICAL:core:yyerror: parse error in config >> file, line 254, column 16-17: Command cannot be used in the block > > How will I know what line 254 is? > > The code I sent uses the failure-route, not the reply-route. > > klaus > > >> >> >> Any idea? >> >> Thank you >> >> >> >> -----Message d'origine----- >> De : Klaus Darilion [mailto:klaus.mailinglists@pernau.at] >> Envoyé : jeudi 4 février 2010 08:59 >> À : BERGANZ Francois >> Cc : users(a)lists.kamailio.org >> Objet : Re: [Kamailio-Users] break a call? >> >> e.g.: >> >> failure_route[1] { >> if(t_check_status("302")) { >> xlog("L_INFO","302 redirect received\n"); >> t_reply("403","received 302 from downstream client"); >> exit; >> } >> } >> >> klaus >> >> Am 04.02.2010 08:37, schrieb BERGANZ Francois: >>> Hello, >>> >>> I know that we can break a call with sl_reply(). >>> >>> But, in the on_reply route, how can I break a call if I receive a >>> 3XX (I cant use a sl_reply() function!) ? >>> >>> Thank you >>> >>> >>> >>> _______________________________________________ >>> Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> >

2 1

Routing Problems (Sorry, Text file is now included)
by Andreas Rehbein 04 Feb '10

04 Feb '10

Hello, we use Kamailio 3.0 on a Red Hat EL5.4 system with openssl 0.9.8e (the current Red Hat OpenSSL version). We want to use Kamailio 3.0 in front of our Call Server (OpenSER 1.3.2) as a security gateway. So the Call Server should only deal tcp, while the Security Gateway terminates the TLS Data which he receives from the User Agents but forwards the SIP-Messages via TCP to the Call Server. UA ---SIP_over_TLS---> Security Gateway (Kam3.0) ---SIP_over_TCP---> Call Server (OpenSER1.3.2) UA<---SIP_over_TLS--- Security Gateway (Kam3.0) <---SIP_over_TCP--- Call Server (OpenSER1.3.2) Nearly everything works fine in this scenario: User Agents are able to register and when they are sending INVITES the callee receives it. The problem we have right now is: if Kamailio 3.0 receives BYE we get "477 Unfortunatly error on sending to next hop occured". It seems that the messages are ok... I attached a text file with the BYE message and the errors. Explanation for the text file: * UA1: 192.168.0.126 * UA2: 192.168.0.176 *Security Gateway (Kam3.0): 192.168.0.89 *Call Server (OpenSER1.3.2): 192.168.0.106 No. Time Source Destination Protocol Info 101 7.964015 192.168.0.106 192.168.0.89 SIP Request: BYE sip:+497118113608227@192.168.0.89:37260;transport=TLS Frame 101 (1138 bytes on wire, 1138 bytes captured) Ethernet II, Src: AlliedTe_3e:b8:05 (00:15:77:3e:b8:05), Dst: Dell_a0:e2:3f (00:1c:23:a0:e2:3f) Internet Protocol, Src: 192.168.0.106 (192.168.0.106), Dst: 192.168.0.89 (192.168.0.89) Transmission Control Protocol, Src Port: 50445 (50445), Dst Port: sip (5060), Seq: 1234, Ack: 1, Len: 1072 Session Initiation Protocol Request-Line: BYE sip:+497118113608227@192.168.0.89:37260;transport=TLS SIP/2.0 Message Header Record-Route: <sip:192.168.0.106;transport=tcp;lr=on;ftag=ba3448a0> Record-Route: <sip:192.168.0.89;transport=tcp;r2=on;lr=on> Record-Route: <sip:192.168.0.89:5061;transport=tls;r2=on;lr=on> Via: SIP/2.0/TCP 192.168.0.106;branch=z9hG4bK854f.b305c913.0;i=19 Via: SIP/2.0/TCP 192.168.0.89;rport=37260;branch=z9hG4bK854f.dac34f77.0;i=a Via: SIP/2.0/TLS 192.168.0.126:62399;branch=z9hG4bK-d8754z-51f50d6cfa997da7-1---d8754z-;rport =1876 Max-Forwards: 68 Route: <sip:192.168.0.89;transport=tcp;r2=on;lr=on> Route: <sip:192.168.0.89:5061;transport=tls;r2=on;lr=on> Contact: <sip:+497118113608226@192.168.0.89:37260;transport=TLS;rinstance=972e46956fc 87f26> To: "+497118113608227"<sip:+497118113608227@192.168.0.106>;tag=15c25277 From: <sip:+497118113608226@192.168.0.106>;tag=ba3448a0 Call-ID: ZjdiMWY5NTYxMTMyN2Q5NTkwOWVmZWExOTEwMDhjYzQ. CSeq: 2 BYE User-Agent: Bosch Communicator release 2.5.4.2 GAC4 stamp 54836 Reason: SIP;description="User Hung Up" Content-Length: 0 P-hint: rr-enforced No. Time Source Destination Protocol Info 103 7.964791 192.168.0.89 192.168.0.106 SIP Status: 477 Unfortunately error on sending to next hop occurred (477/SL) Frame 103 (887 bytes on wire, 887 bytes captured) Ethernet II, Src: Dell_a0:e2:3f (00:1c:23:a0:e2:3f), Dst: AlliedTe_3e:b8:05 (00:15:77:3e:b8:05) Internet Protocol, Src: 192.168.0.89 (192.168.0.89), Dst: 192.168.0.106 (192.168.0.106) Transmission Control Protocol, Src Port: sip (5060), Dst Port: 50445 (50445), Seq: 1, Ack: 2306, Len: 821 Session Initiation Protocol Status-Line: SIP/2.0 477 Unfortunately error on sending to next hop occurred (477/SL) Message Header Via: SIP/2.0/TCP 192.168.0.106;branch=z9hG4bK854f.b305c913.0;i=19 Via: SIP/2.0/TCP 192.168.0.89;rport=37260;branch=z9hG4bK854f.dac34f77.0;i=a Via: SIP/2.0/TLS 192.168.0.126:62399;branch=z9hG4bK-d8754z-51f50d6cfa997da7-1---d8754z-;rport =1876 To: "+497118113608227"<sip:+497118113608227@192.168.0.106>;tag=15c25277 From: <sip:+497118113608226@192.168.0.106>;tag=ba3448a0 Call-ID: ZjdiMWY5NTYxMTMyN2Q5NTkwOWVmZWExOTEwMDhjYzQ. CSeq: 2 BYE Server: kamailio (3.0.0 (i386/linux)) Content-Length: 0 [truncated] Warning: 392 192.168.0.89:5060 "Noisy feedback tells: pid=10378 req_src_ip=192.168.0.106 req_src_port=50445 in_uri=sip:+497118113608227@192.168.0.89:37260;transport=TLS out_uri=sip:+497118113608227@192.168.0.89:37260;transport No. Time Source Destination Protocol Info 104 7.964845 192.168.0.89 192.168.0.106 SIP Status: 477 Unfortunately error on sending to next hop occurred (477/TM) Frame 104 (887 bytes on wire, 887 bytes captured) Ethernet II, Src: Dell_a0:e2:3f (00:1c:23:a0:e2:3f), Dst: AlliedTe_3e:b8:05 (00:15:77:3e:b8:05) Internet Protocol, Src: 192.168.0.89 (192.168.0.89), Dst: 192.168.0.106 (192.168.0.106) Transmission Control Protocol, Src Port: sip (5060), Dst Port: 50445 (50445), Seq: 822, Ack: 2306, Len: 821 Session Initiation Protocol Status-Line: SIP/2.0 477 Unfortunately error on sending to next hop occurred (477/TM) Message Header Via: SIP/2.0/TCP 192.168.0.106;branch=z9hG4bK854f.b305c913.0;i=19 Via: SIP/2.0/TCP 192.168.0.89;rport=37260;branch=z9hG4bK854f.dac34f77.0;i=a Via: SIP/2.0/TLS 192.168.0.126:62399;branch=z9hG4bK-d8754z-51f50d6cfa997da7-1---d8754z-;rport =1876 To: "+497118113608227"<sip:+497118113608227@192.168.0.106>;tag=15c25277 From: <sip:+497118113608226@192.168.0.106>;tag=ba3448a0 Call-ID: ZjdiMWY5NTYxMTMyN2Q5NTkwOWVmZWExOTEwMDhjYzQ. CSeq: 2 BYE Server: kamailio (3.0.0 (i386/linux)) Content-Length: 0 [truncated] Warning: 392 192.168.0.89:5060 "Noisy feedback tells: pid=10378 req_src_ip=192.168.0.106 req_src_port=50445 in_uri=sip:+497118113608227@192.168.0.89:37260;transport=TLS out_uri=sip:+497118113608227@192.168.0.89:37260;transport Do you have any suggestions? Thank you very much in advance! Regards Andreas

1 0

[Kamailio-Users] Sip MESSAGE
by Abdul Hakeem 04 Feb '10

04 Feb '10

Hello, Does anyone knows if Kamalito supports MESSAGE method & where I can find the documentation or API's ? Regards, Abdul Hakeem

6 18

[Kamailio-Users] new module: mqueue
by Elena-Ramona Modroiu 04 Feb '10

04 Feb '10

Hi, few details about the new module mqueue. It provides a generic message queue for SIP Router inter process communication. Items in the queue are pairs of (key, value), both string. The items are consumed in the order they were added in the queue (first in first out). More details at: http://sip-router.org/docbook/sip-router/branch/master/modules/mqueue/mqueu… http://sip-router.org/wiki/cookbooks/pseudo-variables/devel#mqueue_module_p… The module is handy to use for time consuming and/or blocking operations (eg, email notifications). Not to hold the SIP worker process, you can add an item in a queue that is going to be consumed by another process (like timer processes created via rtimer module). Best regards, Ramona

1 0

[Kamailio-Users] [kamailio-users] can't connect with remote database
by Chandrakant Solanki 04 Feb '10

04 Feb '10

Hi I have 3 server with 172.16.0.10/13/14 IP address. My DB is located for 172.16.0.10. I tried to connect mysql from terminal, it works great. But when I start kamailio it gives me following error.. *kamailio.cfg* loadmodule "usrloc.so" ----- ----- ----- modparam("usrloc", "socket_column", "socket") modparam("usrloc", "use_domain", 0) modparam("usrloc", "desc_time_order", 0) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "db_url", "mysql://openser:openserrw@172.16.0.10/openser ") *error.... *Feb 4 17:10:53 localhost /usr/local/sbin/kamailio[28792]: ERROR:core:db_do_init: could not add connection to the pool Feb 4 17:10:53 localhost /usr/local/sbin/kamailio[28792]: ERROR:usrloc:child_init: child(8): failed to connect to database Feb 4 17:10:53 localhost /usr/local/sbin/kamailio[28792]: ERROR:core:init_mod_child: failed to initializing module usrloc, rank 8 Feb 4 17:10:53 localhost /usr/local/sbin/kamailio[28792]: ERROR:core:main_loop: init_child failed for UDP listener Feb 4 17:10:53 localhost /usr/local/sbin/kamailio[28793]: ERROR:db_mysql:db_mysql_new_connection: driver error: Can't connect to MySQL server on '172.16.0.10' (4)* * -- Regards, Chandrakant Solanki

2 1

[Kamailio-Users] break a call?
by BERGANZ Francois 04 Feb '10

04 Feb '10

Hello, I know that we can break a call with sl_reply(). But, in the on_reply route, how can I break a call if I receive a 3XX (I cant use a sl_reply() function!) ? Thank you

2 2

[Kamailio-Users] xlog enhancements
by Elena-Ramona Modroiu 04 Feb '10

04 Feb '10

Hi, I added two new functions to xlog (modules_k): - xlogl(level, message) - xdbgl(message) They are similar to the existing functions, in addition they print the config line at the beginning of the message. The printed value is: "cfglineno:message" In this way you can spot easily in the logs where the function was used to print a particular message. http://sip-router.org/docbook/sip-router/branch/master/modules_k/xlog/xlog.… Regards, Ramona

3 5

[Kamailio-Users] Loose Route / Re-Invite
by Brandon Armstead 04 Feb '10

04 Feb '10

Hello everyone, I'm just curious as to see what some of you guys do in regards to handling a Re-Invite that comes back downstream to a NATTED UAC. For example, call scenario: UAC -> Kamailio (Fix Nated Contact) -> PSTN Re-Invite Occurs: PSTN -> Kamailio -> UAC UAC (200 OK w/ NAT RFC1918 contact) -> Kamailio (branch flags at this point are not notifying of NAT, due to the downstream direction of the INVITE, so RFC1918 address exists, but does not fix_nated_contact) -> PSTN PSTN does not appropriately ACK. So, what are your guys solutions for solving this problem? Is the best way to add an attribute onto the contact header sent in original INVITE? Are there other ways of handling? What is the best, cleanest method. Possible to handle with AVP's? Thanks in advanced for all of your input! Sincerely, Brandon

6 22

[Kamailio-Users] [OT] acabrera@sintys.gov.ar is away, did you know?
by Iñaki Baz Castillo 04 Feb '10

04 Feb '10

Hi, am I the only person receiving an auto-reply mail from acabrera(a)sintys.gov.ar with subject "Ausente" ("Away") for each mail sent to this list (and perhaps to others)? I've already sent him a direct mail. However, why do people use such auto- reply stuf in mail?? this is mail, an asynchronous communication, the receipt can read the mail whenever he prefers, not need to tell me "I'm away now". grrrr -- Iñaki Baz Castillo <ibc(a)aliax.net>

3 3

Re: [Kamailio-Users] Loose Route / Re-Invite (brandon@cryy.com)
by Brandon Armstead 03 Feb '10

03 Feb '10

Hello, Please refer to sip trace: http://pastebin.org/86010. On LINE 294, PSTN sends the Re-INVITE On LINE 378, REGISTRAR sends 200 OK to SIP SBC (with RFC 1918 Address, that is being relayed from the UAC that also sends this same address upon 200 OK of the Re-Invite). Let me know if this makes sense, this trace was taken on the SBC.

4 5

[Kamailio-Users] Registration delays when using Kamailio proxy
by Vikram Ragukumar 03 Feb '10

03 Feb '10

Hello, I am using an x-lite softphone to register with my SIP domain, (a) Without an outbound proxy - Registration is successful and takes about 3 seconds (b) With an outbound proxy - Registration is successful but takes nearly 15 seconds I briefly describe below the sequence of SIP messages while using the outbound proxy Register softphone -> proxy Register proxy -> SIP server Register softphone -> proxy 407 Proxy authentication reqd. SIP server -> proxy Register softphone -> proxy 407 Proxy authentication reqd. proxy -> softphone Register softphone -> proxy (With Digest response) Register proxy -> SIP server (With Digest response) 200 OK Sip server -> proxy Register softphone -> proxy (With Digest response) I am attaching a log with wireshark captures that show SIP messages exchanged between the SIP server,SIP proxy and the x-lite softphone. The log shows several messages being duplicated (REGISTER, 200 OK) 1) Why does the softphone continue to send REGISTER messages after receiving a 200 OK response ? (almost 15 repetitions of REGISTER after receiving 200 OK) 2) What is the reason for the delay in registration times with/without the proxy ? Thanks and Regards, Vikram. No. Time Source Destination Protocol Info 1 0.000000 ab.cde.fgh.ijx(xlite) ab.cde.fgh.ijk(proxy) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 1 (611 bytes on wire, 611 bytes captured) Ethernet II, Src: 1a:1b:1c:1d:1e:1f (1a:1b:1c:1d:1e:1f), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: ab.cde.fgh.ijx(xlite) (ab.cde.fgh.ijx(xlite)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: 33374 (33374), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-c252300b156adb6c-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d> To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 3 0.839564 ab.cde.fgh.ijk(proxy) lm.nop.qrs.tu(server) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 3 (722 bytes on wire, 722 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: Unispher_40:b5:39 (00:90:1a:40:b5:39) Internet Protocol, Src: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)), Dst: lm.nop.qrs.tu(server) (lm.nop.qrs.tu(server)) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijk(proxy);branch=z9hG4bK31c8.d11a2c71.0 Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;received=ab.cde.fgh.ijx(xlite);branch=z9hG4bK-d8754z-c252300b156adb6c-1---d8754z-;rport=33374 Max-Forwards: 69 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d> To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 P-hint: outbound No. Time Source Destination Protocol Info 4 0.952097 lm.nop.qrs.tu(server) ab.cde.fgh.ijk(proxy) SIP Status: 407 Proxy Authentication Required (0 bindings) Frame 4 (555 bytes on wire, 555 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: lm.nop.qrs.tu(server) (lm.nop.qrs.tu(server)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header CSeq: 1 REGISTER Via: SIP/2.0/UDP ab.cde.fgh.ijk(proxy);branch=z9hG4bK31c8.d11a2c71.0 Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-c252300b156adb6c-1---d8754z-;rport=33374 From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. To: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=030240100052 Proxy-Authenticate: DIGEST realm="VoipSwitch", nonce="126513605203001203402405205305" Content-Length: 0 No. Time Source Destination Protocol Info 5 1.404504 ab.cde.fgh.ijk(proxy) lm.nop.qrs.tu(server) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 5 (722 bytes on wire, 722 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: Unispher_40:b5:39 (00:90:1a:40:b5:39) Internet Protocol, Src: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)), Dst: lm.nop.qrs.tu(server) (lm.nop.qrs.tu(server)) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijk(proxy);branch=z9hG4bK31c8.d11a2c71.0 Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;received=ab.cde.fgh.ijx(xlite);branch=z9hG4bK-d8754z-c252300b156adb6c-1---d8754z-;rport=33374 Max-Forwards: 69 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d> To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 P-hint: outbound No. Time Source Destination Protocol Info 6 1.517292 lm.nop.qrs.tu(server) ab.cde.fgh.ijk(proxy) SIP Status: 407 Proxy Authentication Required (0 bindings) Frame 6 (555 bytes on wire, 555 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: lm.nop.qrs.tu(server) (lm.nop.qrs.tu(server)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header CSeq: 1 REGISTER Via: SIP/2.0/UDP ab.cde.fgh.ijk(proxy);branch=z9hG4bK31c8.d11a2c71.0 Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-c252300b156adb6c-1---d8754z-;rport=33374 From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. To: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=030240100053 Proxy-Authenticate: DIGEST realm="VoipSwitch", nonce="126513605303001203402405305305" Content-Length: 0 No. Time Source Destination Protocol Info 7 1.522829 ab.cde.fgh.ijx(xlite) ab.cde.fgh.ijk(proxy) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 7 (611 bytes on wire, 611 bytes captured) Ethernet II, Src: 1a:1b:1c:1d:1e:1f (1a:1b:1c:1d:1e:1f), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: ab.cde.fgh.ijx(xlite) (ab.cde.fgh.ijx(xlite)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: 33374 (33374), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-c252300b156adb6c-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d> To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 No. Time Source Destination Protocol Info 8 2.522300 ab.cde.fgh.ijk(proxy) ab.cde.fgh.ijx(xlite) SIP Status: 407 Proxy Authentication Required (0 bindings) Frame 8 (492 bytes on wire, 492 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: 1a:1b:1c:1d:1e:1f (1a:1b:1c:1d:1e:1f) Internet Protocol, Src: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)), Dst: ab.cde.fgh.ijx(xlite) (ab.cde.fgh.ijx(xlite)) User Datagram Protocol, Src Port: sip (5060), Dst Port: 33374 (33374) Session Initiation Protocol Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header CSeq: 1 REGISTER Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-c252300b156adb6c-1---d8754z-;rport=33374 From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. To: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=030240100052 Proxy-Authenticate: DIGEST realm="VoipSwitch", nonce="126513605203001203402405205305" Content-Length: 0 No. Time Source Destination Protocol Info 9 2.730896 ab.cde.fgh.ijx(xlite) ab.cde.fgh.ijk(proxy) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 9 (801 bytes on wire, 801 bytes captured) Ethernet II, Src: 1a:1b:1c:1d:1e:1f (1a:1b:1c:1d:1e:1f), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: ab.cde.fgh.ijx(xlite) (ab.cde.fgh.ijx(xlite)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: 33374 (33374), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-4248775550708257-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d> To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Proxy-Authorization: Digest username="1234",realm="VoipSwitch",nonce="126513605203001203402405205305",uri="sip:lm.nop.qrs.tu(server)",response="2c0d7d2f3b4cc218792ebef30e0eb2e6",algorithm=MD5 User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 No. Time Source Destination Protocol Info 10 2.796109 ab.cde.fgh.ijk(proxy) ab.cde.fgh.ijx(xlite) SIP Status: 407 Proxy Authentication Required (0 bindings) Frame 10 (492 bytes on wire, 492 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: 1a:1b:1c:1d:1e:1f (1a:1b:1c:1d:1e:1f) Internet Protocol, Src: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)), Dst: ab.cde.fgh.ijx(xlite) (ab.cde.fgh.ijx(xlite)) User Datagram Protocol, Src Port: sip (5060), Dst Port: 33374 (33374) Session Initiation Protocol Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header CSeq: 1 REGISTER Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-c252300b156adb6c-1---d8754z-;rport=33374 From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. To: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=030240100052 Proxy-Authenticate: DIGEST realm="VoipSwitch", nonce="126513605203001203402405205305" Content-Length: 0 No. Time Source Destination Protocol Info 11 3.308767 ab.cde.fgh.ijx(xlite) ab.cde.fgh.ijk(proxy) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 11 (801 bytes on wire, 801 bytes captured) Ethernet II, Src: 1a:1b:1c:1d:1e:1f (1a:1b:1c:1d:1e:1f), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: ab.cde.fgh.ijx(xlite) (ab.cde.fgh.ijx(xlite)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: 33374 (33374), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-4248775550708257-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d> To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Proxy-Authorization: Digest username="1234",realm="VoipSwitch",nonce="126513605203001203402405205305",uri="sip:lm.nop.qrs.tu(server)",response="2c0d7d2f3b4cc218792ebef30e0eb2e6",algorithm=MD5 User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 No. Time Source Destination Protocol Info 12 4.099907 ab.cde.fgh.ijk(proxy) lm.nop.qrs.tu(server) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 12 (912 bytes on wire, 912 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: Unispher_40:b5:39 (00:90:1a:40:b5:39) Internet Protocol, Src: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)), Dst: lm.nop.qrs.tu(server) (lm.nop.qrs.tu(server)) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijk(proxy);branch=z9hG4bK01c8.2e6fe072.0 Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;received=ab.cde.fgh.ijx(xlite);branch=z9hG4bK-d8754z-4248775550708257-1---d8754z-;rport=33374 Max-Forwards: 69 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d> To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Proxy-Authorization: Digest username="1234",realm="VoipSwitch",nonce="126513605203001203402405205305",uri="sip:lm.nop.qrs.tu(server)",response="2c0d7d2f3b4cc218792ebef30e0eb2e6",algorithm=MD5 User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 P-hint: outbound No. Time Source Destination Protocol Info 13 4.216914 lm.nop.qrs.tu(server) ab.cde.fgh.ijk(proxy) SIP Status: 407 Proxy Authentication Required (0 bindings) Frame 13 (555 bytes on wire, 555 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: lm.nop.qrs.tu(server) (lm.nop.qrs.tu(server)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header CSeq: 2 REGISTER Via: SIP/2.0/UDP ab.cde.fgh.ijk(proxy);branch=z9hG4bK01c8.2e6fe072.0 Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-4248775550708257-1---d8754z-;rport=33374 From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. To: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=030240100055 Proxy-Authenticate: DIGEST realm="VoipSwitch", nonce="126513605503001203402405505305" Content-Length: 0 No. Time Source Destination Protocol Info 14 4.324392 ab.cde.fgh.ijx(xlite) ab.cde.fgh.ijk(proxy) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 14 (801 bytes on wire, 801 bytes captured) Ethernet II, Src: 1a:1b:1c:1d:1e:1f (1a:1b:1c:1d:1e:1f), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: ab.cde.fgh.ijx(xlite) (ab.cde.fgh.ijx(xlite)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: 33374 (33374), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-4248775550708257-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d> To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Proxy-Authorization: Digest username="1234",realm="VoipSwitch",nonce="126513605203001203402405205305",uri="sip:lm.nop.qrs.tu(server)",response="2c0d7d2f3b4cc218792ebef30e0eb2e6",algorithm=MD5 User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 No. Time Source Destination Protocol Info 15 4.704708 ab.cde.fgh.ijk(proxy) lm.nop.qrs.tu(server) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 15 (912 bytes on wire, 912 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: Unispher_40:b5:39 (00:90:1a:40:b5:39) Internet Protocol, Src: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)), Dst: lm.nop.qrs.tu(server) (lm.nop.qrs.tu(server)) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijk(proxy);branch=z9hG4bK01c8.2e6fe072.0 Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;received=ab.cde.fgh.ijx(xlite);branch=z9hG4bK-d8754z-4248775550708257-1---d8754z-;rport=33374 Max-Forwards: 69 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d> To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Proxy-Authorization: Digest username="1234",realm="VoipSwitch",nonce="126513605203001203402405205305",uri="sip:lm.nop.qrs.tu(server)",response="2c0d7d2f3b4cc218792ebef30e0eb2e6",algorithm=MD5 User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 P-hint: outbound No. Time Source Destination Protocol Info 16 4.822244 lm.nop.qrs.tu(server) ab.cde.fgh.ijk(proxy) SIP Status: 407 Proxy Authentication Required (0 bindings) Frame 16 (555 bytes on wire, 555 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: lm.nop.qrs.tu(server) (lm.nop.qrs.tu(server)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header CSeq: 2 REGISTER Via: SIP/2.0/UDP ab.cde.fgh.ijk(proxy);branch=z9hG4bK01c8.2e6fe072.0 Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-4248775550708257-1---d8754z-;rport=33374 From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. To: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=030240100056 Proxy-Authenticate: DIGEST realm="VoipSwitch", nonce="126513605603001203402405605305" Content-Length: 0 No. Time Source Destination Protocol Info 17 5.710292 ab.cde.fgh.ijk(proxy) ab.cde.fgh.ijx(xlite) SIP Status: 407 Proxy Authentication Required (0 bindings) Frame 17 (492 bytes on wire, 492 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: 1a:1b:1c:1d:1e:1f (1a:1b:1c:1d:1e:1f) Internet Protocol, Src: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)), Dst: ab.cde.fgh.ijx(xlite) (ab.cde.fgh.ijx(xlite)) User Datagram Protocol, Src Port: sip (5060), Dst Port: 33374 (33374) Session Initiation Protocol Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header CSeq: 2 REGISTER Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-4248775550708257-1---d8754z-;rport=33374 From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. To: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=030240100055 Proxy-Authenticate: DIGEST realm="VoipSwitch", nonce="126513605503001203402405505305" Content-Length: 0 No. Time Source Destination Protocol Info 18 5.724689 ab.cde.fgh.ijk(proxy) ab.cde.fgh.ijx(xlite) SIP Status: 407 Proxy Authentication Required (0 bindings) Frame 18 (492 bytes on wire, 492 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: 1a:1b:1c:1d:1e:1f (1a:1b:1c:1d:1e:1f) Internet Protocol, Src: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)), Dst: ab.cde.fgh.ijx(xlite) (ab.cde.fgh.ijx(xlite)) User Datagram Protocol, Src Port: sip (5060), Dst Port: 33374 (33374) Session Initiation Protocol Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header CSeq: 2 REGISTER Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-4248775550708257-1---d8754z-;rport=33374 From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. To: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=030240100055 Proxy-Authenticate: DIGEST realm="VoipSwitch", nonce="126513605503001203402405505305" Content-Length: 0 No. Time Source Destination Protocol Info 19 5.832474 ab.cde.fgh.ijx(xlite) ab.cde.fgh.ijk(proxy) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 19 (801 bytes on wire, 801 bytes captured) Ethernet II, Src: 1a:1b:1c:1d:1e:1f (1a:1b:1c:1d:1e:1f), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: ab.cde.fgh.ijx(xlite) (ab.cde.fgh.ijx(xlite)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: 33374 (33374), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-316786162159173a-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d> To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 3 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Proxy-Authorization: Digest username="1234",realm="VoipSwitch",nonce="126513605503001203402405505305",uri="sip:lm.nop.qrs.tu(server)",response="014a8ffc9e6cc551acc86b22f2c26c11",algorithm=MD5 User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 No. Time Source Destination Protocol Info 21 7.455368 ab.cde.fgh.ijk(proxy) lm.nop.qrs.tu(server) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 21 (912 bytes on wire, 912 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: Unispher_40:b5:39 (00:90:1a:40:b5:39) Internet Protocol, Src: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)), Dst: lm.nop.qrs.tu(server) (lm.nop.qrs.tu(server)) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijk(proxy);branch=z9hG4bK11c8.7f536395.0 Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;received=ab.cde.fgh.ijx(xlite);branch=z9hG4bK-d8754z-316786162159173a-1---d8754z-;rport=33374 Max-Forwards: 69 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d> To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 3 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Proxy-Authorization: Digest username="1234",realm="VoipSwitch",nonce="126513605503001203402405505305",uri="sip:lm.nop.qrs.tu(server)",response="014a8ffc9e6cc551acc86b22f2c26c11",algorithm=MD5 User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 P-hint: outbound No. Time Source Destination Protocol Info 22 7.457267 ab.cde.fgh.ijx(xlite) ab.cde.fgh.ijk(proxy) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 22 (801 bytes on wire, 801 bytes captured) Ethernet II, Src: 1a:1b:1c:1d:1e:1f (1a:1b:1c:1d:1e:1f), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: ab.cde.fgh.ijx(xlite) (ab.cde.fgh.ijx(xlite)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: 33374 (33374), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-316786162159173a-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d> To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 3 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Proxy-Authorization: Digest username="1234",realm="VoipSwitch",nonce="126513605503001203402405505305",uri="sip:lm.nop.qrs.tu(server)",response="014a8ffc9e6cc551acc86b22f2c26c11",algorithm=MD5 User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 No. Time Source Destination Protocol Info 23 7.574533 lm.nop.qrs.tu(server) ab.cde.fgh.ijk(proxy) SIP Status: 200 OK (1 bindings) Frame 23 (539 bytes on wire, 539 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: lm.nop.qrs.tu(server) (lm.nop.qrs.tu(server)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header CSeq: 3 REGISTER Via: SIP/2.0/UDP ab.cde.fgh.ijk(proxy);branch=z9hG4bK11c8.7f536395.0 Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-316786162159173a-1---d8754z-;rport=33374 From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. To: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=030240100059 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d>;expires=60 Expires: 60 Content-Length: 0 No. Time Source Destination Protocol Info 24 8.355247 ab.cde.fgh.ijk(proxy) lm.nop.qrs.tu(server) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 24 (912 bytes on wire, 912 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: Unispher_40:b5:39 (00:90:1a:40:b5:39) Internet Protocol, Src: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)), Dst: lm.nop.qrs.tu(server) (lm.nop.qrs.tu(server)) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijk(proxy);branch=z9hG4bK11c8.7f536395.0 Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;received=ab.cde.fgh.ijx(xlite);branch=z9hG4bK-d8754z-316786162159173a-1---d8754z-;rport=33374 Max-Forwards: 69 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d> To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 3 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Proxy-Authorization: Digest username="1234",realm="VoipSwitch",nonce="126513605503001203402405505305",uri="sip:lm.nop.qrs.tu(server)",response="014a8ffc9e6cc551acc86b22f2c26c11",algorithm=MD5 User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 P-hint: outbound No. Time Source Destination Protocol Info 25 8.472891 lm.nop.qrs.tu(server) ab.cde.fgh.ijk(proxy) SIP Status: 200 OK (1 bindings) Frame 25 (539 bytes on wire, 539 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: lm.nop.qrs.tu(server) (lm.nop.qrs.tu(server)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header CSeq: 3 REGISTER Via: SIP/2.0/UDP ab.cde.fgh.ijk(proxy);branch=z9hG4bK11c8.7f536395.0 Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-316786162159173a-1---d8754z-;rport=33374 From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. To: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=030241100000 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d>;expires=60 Expires: 60 Content-Length: 0 No. Time Source Destination Protocol Info 26 9.310268 ab.cde.fgh.ijk(proxy) ab.cde.fgh.ijx(xlite) SIP Status: 200 OK (1 bindings) Frame 26 (476 bytes on wire, 476 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: 1a:1b:1c:1d:1e:1f (1a:1b:1c:1d:1e:1f) Internet Protocol, Src: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)), Dst: ab.cde.fgh.ijx(xlite) (ab.cde.fgh.ijx(xlite)) User Datagram Protocol, Src Port: sip (5060), Dst Port: 33374 (33374) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header CSeq: 3 REGISTER Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-316786162159173a-1---d8754z-;rport=33374 From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. To: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=030240100059 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d>;expires=60 Expires: 60 Content-Length: 0 No. Time Source Destination Protocol Info 29 10.939788 ab.cde.fgh.ijx(xlite) ab.cde.fgh.ijk(proxy) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 29 (796 bytes on wire, 796 bytes captured) Ethernet II, Src: 1a:1b:1c:1d:1e:1f (1a:1b:1c:1d:1e:1f), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: ab.cde.fgh.ijx(xlite) (ab.cde.fgh.ijx(xlite)), Dst: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)) User Datagram Protocol, Src Port: 33374 (33374), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;branch=z9hG4bK-d8754z-075267174a75f03a-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d>;expires=0 To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 4 REGISTER Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Proxy-Authorization: Digest username="1234",realm="VoipSwitch",nonce="126513605503001203402405505305",uri="sip:lm.nop.qrs.tu(server)",response="014a8ffc9e6cc551acc86b22f2c26c11",algorithm=MD5 User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 No. Time Source Destination Protocol Info 30 11.097051 ab.cde.fgh.ijk(proxy) lm.nop.qrs.tu(server) SIP Request: REGISTER sip:lm.nop.qrs.tu(server) Frame 30 (907 bytes on wire, 907 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: Unispher_40:b5:39 (00:90:1a:40:b5:39) Internet Protocol, Src: ab.cde.fgh.ijk(proxy) (ab.cde.fgh.ijk(proxy)), Dst: lm.nop.qrs.tu(server) (lm.nop.qrs.tu(server)) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:lm.nop.qrs.tu(server) SIP/2.0 Message Header Via: SIP/2.0/UDP ab.cde.fgh.ijk(proxy);branch=z9hG4bKe0c8.73270ff2.0 Via: SIP/2.0/UDP ab.cde.fgh.ijx(xlite):50648;received=ab.cde.fgh.ijx(xlite);branch=z9hG4bK-d8754z-075267174a75f03a-1---d8754z-;rport=33374 Max-Forwards: 69 Contact: <sip:1234@ab.cde.fgh.ijx(xlite):50648;rinstance=0f34c912798e872d>;expires=0 To: "1234"<sip:1234@lm.nop.qrs.tu(server)> From: "1234"<sip:1234@lm.nop.qrs.tu(server)>;tag=cb57d146 Call-ID: Zjg5MTdhZjMyYTZlMzlmZTA2NjY4MDBhMjNhYjkyMDU. CSeq: 4 REGISTER Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Proxy-Authorization: Digest username="1234",realm="VoipSwitch",nonce="126513605503001203402405505305",uri="sip:lm.nop.qrs.tu(server)",response="014a8ffc9e6cc551acc86b22f2c26c11",algorithm=MD5 User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 P-hint: outbound

4 5

[Kamailio-Users] protocol/port mismatch on BYE from Calle to Caller
by Arafath-uz-zaman khan 03 Feb '10

03 Feb '10

Hello All, I need a working example for sending all request to another sip server using kamailio. X-lite-->TLS(7161)-->Kamailio-->UDP(5060)-->SIP Server-->PSTN Number. i have attached my cfg file. The problem is when i dialed any pstn number everything working fine except the by msg. If the caller send BYE callee get hangup but if Calle send BYE caller never hangup the call. i got the following msg on my log file Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_msg: method: <BYE> Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_msg: uri: <sip:91131@192.168.10.2:5060> Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_msg: version: <SIP/2.0> Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_headers: flags=2 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:get_hdr_field: cseq <CSeq>: <1> <BYE> Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bk010213101521011481381407>; state=16 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_via: end of header reached, state=5 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_headers: via found, flags=2 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_headers: this is the first via Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:receive_msg: After parse_msg... Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:receive_msg: preparing to run routing scripts... Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_headers: flags=100 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_to_param: tag=b23c1531 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_to: end of header reached, state=29 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_to: display={"91131"}, ruri={sip:91131@192.168.10.2:5060} Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:get_hdr_field: <To> [51]; uri=[sip:91131@192.168.10.2:5060] Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:get_hdr_field: to body ["91131"<sip:91131@192.168.10.2:5060>] Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:get_hdr_field: content_length=0 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:get_hdr_field: found end of header Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:maxfwd:is_maxfwd_present: max_forwards header not found! Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:check_via_address: params 192.168.10.2, 192.168.10.2, 0 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_headers: flags=80 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:siputils:has_totag: totag found Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_headers: flags=200 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:rr:is_preloaded: is_preloaded: No Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:grep_sock_info: checking if host==us: 12==13 && [192.168.10.2] == [192.168.10.1] Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:grep_sock_info: checking if host==us: 12==13 && [192.168.10.2] == [192.168.10.1] Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:grep_sock_info: checking if port 7161 matches port 5060 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:grep_sock_info: no match for: [192.168.10.2:5060] Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:grep_aliases: no match for: [0:192.168.10.2:5060] Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:check_self: host != me Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:grep_sock_info: checking if host==us: 13==13 && [192.168.10.1] == [192.168.10.1] Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:grep_sock_info: match found for: [192.168.10.1:5060] Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:check_self: host == me Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:rr:after_loose: Topmost route URI: 'sip:192.168.10.1;r2=on;lr=on;nat=yes' is me Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:grep_sock_info: checking if host==us: 13==13 && [192.168.10.1] == [192.168.10.1] Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:grep_sock_info: checking if port 7161 matches port 7161 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:grep_sock_info: match found for: [192.168.10.1:7161] Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_headers: flags=200 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:get_hdr_field: found end of header Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:rr:find_next_route: No next Route HF found Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:rr:after_loose: no next URI found Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:rr:check_route_param: params are <;r2=on;lr=on;nat=yes> Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_headers: flags=40 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_to_param: tag=01021310150514813644546133 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_to: end of header reached, state=29 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_to: display={"911317434"}, ruri={sip:911317434@192.168.10.1:5060} Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:siputils:has_totag: totag found Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:tm:t_newtran: transaction on entrance=0xffffffff Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_headers: flags=ffffffffffffffff Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_headers: flags=78 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:tm:t_lookup_request: start searching: hash=850, isACK=0 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:tm:t_lookup_request: proceeding to pre-RFC3261 transaction matching Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:tm:t_lookup_request: no transaction found Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:tm:run_reqin_callbacks: trans=0xb615b5f8, callback type 1, id 0 entered Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:parse_headers: flags=78 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:mk_proxy: doing DNS lookup... *Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:get_send_socket: force_send_socket of different proto (1)! Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: WARNING:core:get_send_socket: protocol/port mismatch* Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:tm:set_timer: relative timeout is 500000 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:tm:insert_timer_unsafe: [4]: 0xb615b744 (46600000) Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:tm:set_timer: relative timeout is 30 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:tm:insert_timer_unsafe: [0]: 0xb615b760 (76) Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:tm:t_relay_to: new transaction fwd'ed Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:destroy_avp_list: destroying list (nil) Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8808]: DBG:core:receive_msg: cleaning up Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8809]: DBG:tm:utimer_routine: timer routine:4,tl=0xb615b744 next=(nil), timeout=46600000 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8809]: DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0xb615b5f8, BYE sip:9 ... ) Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8809]: DBG:tm:set_timer: relative timeout is 1000000 Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8809]: DBG:tm:insert_timer_unsafe: [5]: 0xb615b744 (47600000) Feb 1 04:27:29 ser /usr/local/sbin/kamailio[8809]: DBG:tm:retransmission_handler: retransmission_handler : done Feb 1 04:27:30 ser /usr/local/sbin/kamailio[8809]: DBG:tm:utimer_routine: timer routine:5,tl=0xb615b744 next=(nil), timeout=47600000 Feb 1 04:27:30 ser /usr/local/sbin/kamailio[8809]: DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0xb615b5f8, BYE sip:9 ... ) Feb 1 04:27:30 ser /usr/local/sbin/kamailio[8809]: DBG:tm:set_timer: relative timeout is 2000000 Feb 1 04:27:30 ser /usr/local/sbin/kamailio[8809]: DBG:tm:insert_timer_unsafe: [6]: 0xb615b744 (49600000) Feb 1 04:27:30 ser /usr/local/sbin/kamailio[8809]: DBG:tm:retransmission_handler: retransmission_handler : done

2 3

[Kamailio-Users] the best editor?
by BERGANZ Francois 03 Feb '10

03 Feb '10

Hello all, what is the best edit tool to see quickly the right syntax ... ? which language to choose?

5 6

[Kamailio-Users] KAMAILIO 1.4.3 no TLS Stop processing calls when Radius server goes down
by Omar 03 Feb '10

03 Feb '10

I have set it up the Radius for Kamailio, it works fine, until the Radius server goes down it did not go down for a while, when tis happes the Kamailio server stops processing any calls. I do not think is a normal behaviour, probably a bug, can anybody please tell me if this is expected behave in kamailio. or was already noticed before and was already fixed in a later release. Is possible to point to more than 2 radius servers in parallel? NOTE: i was running radius and DB in parallel with no issues, until the RADIUS server goes down. __________________________ O

3 2

[Kamailio-Users] dialog and pua_dialoginfo: Issue regarding vanished PUBLISH messages
by Martin Koenig 03 Feb '10

03 Feb '10

Hello all, I've run into an issue of Mar 2009: http://lists.kamailio.org/pipermail/users/2009-March/022143.html This seems to be unresolved in v1.5.4, the then presented work-around via modparam("pua_dialoginfo", "override_lifetime", 600) still works fine. The question is whether this should work differently nowadays, or if notice should be put into the docs? Best regards, Martin

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[Kamailio-Users] alias_db: append_branches and dialog
by Martin Koenig 03 Feb '10

03 Feb '10

Hello all, two questions regarding alias_db: 1) How do I make use of the append_branches function of alias_db? Trying to insert two or more aliases is not possible in the current db layout ( UNIQUE KEY `alias_idx` (`alias_username`,`alias_domain`) ). The idea was to enable simple db-based parallel branching ("group call") via alias_db by inserting one or more users to one alias. This seems not possible. 2) Use of alias_db "breaks" presence_dialog, because dialog matching is based on from / to header (instead from / final AoR after alias_db_lookup() ), thus presence_dialog events cannot be created and PUBLISHed to end devices. I was wondering whether it would be sufficient to uac_replace_from in branch_route[] before the final t_relay work around this, or if there is another solution out there? Many thanks and best regards, Martin

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Re: [Kamailio-Users] append_rpid_hf add privacy?
by Daniel-Constantin Mierla 03 Feb '10

03 Feb '10

Hello, On 2/2/10 1:18 PM, alex pappas wrote: > Hi Daniel, > > I'm capturing with tcpdump the packets when they live the server and I > don't see the extra header added. > are you having the rpid value in subscriber table? Is it loaded via load_credentials of auth_db or do you set the right avp in script? Have you tried append_hf()? Seems you want to add r-uri to rpid, which is easier via append_hf() then. Cheers, Daniel > Cheers > Alex > > > > On Tue, Feb 2, 2010 at 12:24 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hi Alex, > > > On 2/2/10 11:04 AM, alex pappas wrote: > > Dear friends, > > I'm trying to give to a user the option to have privacy in a > call. I'm trying to do that by adding an extra header with: > append_rpid_hf("sip", > ";party=calling;id-type=subscriber;privacy=off;screen=no"); > > I'm using this header in the begining of the kamailio.cfg > > if(($rU=="99999") || ($rU=="77777")) > { > append_rpid_hf("sip", > "$ru;party=calling;id-type=subscriber;privacy=off;screen=no"); > xlog("alx ------- We have rU match > --------"); > } > > When I see the sip messages I don't see this extra header > added into it. > > Does anyone had experience in this ? > > the header is visible when the message is forwarded. To see it > immediately, but only in 3.0.0, you can use msg_apply_changes(). > > Cheers, > Daniel > > > -- > Daniel-Constantin Mierla > eLearning class for Kamailio 3.0.0 > Starting Feb 8, 2010 > * http://www.asipto.com/ > > -- Daniel-Constantin Mierla eLearning class for Kamailio 3.0.0 Starting Feb 8, 2010 * http://www.asipto.com/

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[Kamailio-Users] preparing kamailio 1.5.4 and 1.4.5 releases
by Daniel-Constantin Mierla 02 Feb '10

02 Feb '10

Hello, soon I will start preparing the 1.5.4 and 1.4.5. The only issue reported and not clarified yet is the tcp failover that I tried to reproduce today without luck, might be dependent of (triggered faster on) particular architecture. It can be fixed once gathering more details or being able to reproduce. If you are aware of something else, please report to devel list. 1.4.5 will mark the official support for 1.4.x series, although some fixes may go back there from time to time, but I don't think will be another release of that branch. I recommend to update to newer version or stick to svn for updates. Cheers, Daniel -- Daniel-Constantin Mierla eLearning class for Kamailio 3.0.0 Starting Feb 8, 2010 * http://www.asipto.com/

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[Kamailio-Users] PV bug in modules_k/acc/acc_extra.c in K 3.0.0
by Alex Balashov 02 Feb '10

02 Feb '10

I ran into a very interesting problem last night with 'acc' that took me a few hours to solve, so I am posting my workaround for collective benefit. It is probably not the cleanest and most appropriate solution because I am not extensively familiar with the 'acc' code; this is my first dive into it. Thus, I do not presume to submit an actual patch. The problem I had was that PVs read from the db_extra parameter that had an internal type of integer (TYPE_INT) because of their lexical scope were not being correctly evaluated, resulting in bizarre values being put in the database. I am using K 3.0.0 on 64-bit Xen guest running CentOS with db_postgres. I did not check to see if this problem occurs with db_mysql as I do not have MySQL anywhere nearby, but I don't believe it is happening at the database layer anyhow. I also know this problem was not present in 1.5.x, or several of my installations would be non-functional; this must have something to do with use of new 'srdb1' stuff in the database side of 'acc', I hypothesise. Anyway, the actual problem went like this. Let us say, for simplicity of example, that I am using 'acc' with Postgres backing and have something like this: modparam("acc", "db_extra", "account_id=$avp(s:account_id)") ... modparam("acc", "db_flag", 1) What was happening was that if I did this in a request route: $avp(s:account_id) = 42; ... setflag(1); if(!t_relay()) sl_reply_error(); exit; Very strange and occasionally bizarre values, mostly duplicates of other values I have in db_extra, would be written to acc.account_id (a column of type 'integer') - at any rate, certainly not 42. I enabled SQL statement logging on my PostgreSQL database to ensure that the data being inserted was literally not 42, rather than this being some sort of database-layer type conversion / representation problem. Sure enough, it was true, although I also did notice that 'acc' encloses all inserted values in single quotes to be treated as string literals. But this is immaterial, because RDBMs automatically cast such input: INSERT INTO acc (..., account_id) VALUES(..., '12'); Now, I could fix the problem by simply assigning the AVP a string literal instead: $avp(s:account_id) = '42'; Now the data was inserted normally. I also tested with user variables ($var(...)) and the same was true, so the problem is not AVP-specific. Also, all other evaluative contexts for PVs with these values work fine, such as printing them. The problem does not seem to be with 'db_postgres', neither with 'pv', nor with 'avpops' -- it is particular to 'acc'. After some hours of investigation I found that what was happening is that the type of the PV being referenced is not being correctly packed into the various arrays populated by extra2strar() in modules_k/acc/acc_extra.c, and thus was being misread when iterated in acc_db_request() in modules_k/acc/acc.c. Specifically, this check was inadequate: if (value.rs.s+value.rs.len==static_detector) { val_arr[n].s = int_buf + r*INT2STR_MAX_LEN; val_arr[n].len = value.rs.len; memcpy(val_arr[n].s, value.rs.s, value.rs.len); r++; LM_INFO("New int tenant: %s\n", val_arr[n].s); } else { val_arr[n] = value.rs; LM_INFO("New str tenant: %s\n", value.rs); } In my debugging, all PV values were showing up as "New str tenant" regardless of whether they were natively integers, and regardless of what the PV spec type was actually set to. The type was set correctly on the PV spec, so of course the integer value got stored correctly in int_var[] below: if (value.flags&PV_VAL_INT) { int_arr[n] = value.ri; type_arr[n] = TYPE_INT; } else { type_arr[n] = TYPE_STR; } ... but since 'acc' wants to treat all extra values as strings, this was not sufficient, because what was failing was this condition: if (value.rs.s+value.rs.len==static_detector) I guess value.rs.s in case of the PV value being PV_VAL_INT is not null terminated properly? This I truly do not know. In any event, I just added a check on the PV flags to see if it is an integer value and this fixed my problem: if (value.rs.s+value.rs.len==static_detector || value.flags & PV_VAL_INT) { ... however, I do not know that this was the most insightful solution. This is rather critical I think, because it means anyone trying to store PV values that are initialised with a natively numerical type in script (or extracted as a numerical value from a database, e.g. sql_query("SELECT account_id::integer FROM table ..."), I found this to be a problem as well) cannot store them in the database reliably using 'db_extra'. Thanks! -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/

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[Kamailio-Users] isup in invite
by Eric Hiller 02 Feb '10

02 Feb '10

I have an issue where my sip endpoint is sending ISUP information in the invite. IE: sip:+1xxxxxxxxxx;npdi=yes;cic=+1-0110@xxxx:5060 and when I run the enum query I am getting: 0(24410) enum_query(): uri user is not an E164 number This is most likely due to the fact that $rU=+17794294010;npdi=yes;cic=+1-0110 Does openser not read the user properly? Is there a way to correct this? I am running openser 1.2.2 notls Thanks for any help! -Eric _________________________________________________________________ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. http://clk.atdmt.com/GBL/go/196390709/direct/01/

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[Kamailio-Users] append_rpid_hf add privacy?
by alex pappas 02 Feb '10

02 Feb '10

Dear friends, I'm trying to give to a user the option to have privacy in a call. I'm trying to do that by adding an extra header with: append_rpid_hf("sip", ";party=calling;id-type=subscriber;privacy=off;screen=no"); I'm using this header in the begining of the kamailio.cfg if(($rU=="99999") || ($rU=="77777")) { append_rpid_hf("sip", "$ru;party=calling;id-type=subscriber;privacy=off;screen=no"); xlog("alx ------- We have rU match --------"); } When I see the sip messages I don't see this extra header added into it. Does anyone had experience in this ? Thank you Alex

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[Kamailio-Users] new feature: remote user registration
by Daniel-Constantin Mierla 01 Feb '10

01 Feb '10

Hello, if you follow the sr-dev mailing list, you may have noticed some new features added in master branch (for the 3.1.0). I will send more details about each, now: remote user registration. The uac module can register in behalf of users to remote servers. The user details (credentials, local and remote username/domain) are loaded from database. The module sends registration at start-up and then refreshes just before expiration time. If the REGISTER is challenged for authentication, uac will re-send the REGISTER with proper credentials. A typical scenario is when users have accounts to many voip providers and they want to receive calls to those accounts via your sip server (inbound traffic). Of course, the user has to give details such as username/password to you. Other use case can be dynamic interconnection/peering. you can imagine a scalable solution where you have a traffic dispatcher and worker servers. When an worker becomes active, it registers to traffic dispatcher for receiving traffic... and examples can continue. Hope will help you in various cases! Readme at: http://sip-router.org/docbook/sip-router/branch/master/modules_k/uac/uac.ht… SQL to create the mysql table is in utils/kamctl/mysql/uac-create.sql Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] new module: mtree
by Daniel-Constantin Mierla 01 Feb '10

01 Feb '10

Hello, the master branch got a new module named mtree. It is able to load database records in the form of (key,value) and index them in shared memory trees based on key. One db table is loaded in one mtree, you can define as many trees as you need. The typical use case is to load arbitrary data related to DIDs or prefixes, offering very fast lookup even with large number of records. The value can be string or digested in a runtime structure, so far being able to digest dstid:weight values, useful for prefix matching on weight rather than longest match. The trees are read only, with option to reload via mi command (rpc to be added soon). Online readme at: http://sip-router.org/docbook/sip-router/branch/master/modules/mtree/mtree.… Cheers, Daniel -- Daniel-Constantin Mierla eLearning class for Kamailio 3.0.0 Starting Feb 8, 2010 * http://www.asipto.com/

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[Kamailio-Users] kamailio 1.4 cpc
by Asim Riaz 01 Feb '10

01 Feb '10

Hi List, Is it possible to add diffrent CPC(calling party catagory) for diffrent type of the number in the INVITE before sending out to an upstream SIP server? INVITE ----> Kamailio(add CPC) -------> SIP_Proxy_2 Thanks, Asim

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[Kamailio-Users] Fosdem 2010 - reminder
by Daniel-Constantin Mierla 01 Feb '10

01 Feb '10

Hello, a reminder for those that already received (also sorry for cross-posting) and invitation for new ones, Fosdem 2010 happens this weekend in Brussels and we will have the traditional SIP routing dinner on Saturday Feb 6, trying to get all of us up-to-date regarding projects such as ser, kamailio (openser), sip-router, sems, asterisk, freeswitch, a.s.o. If you plan to attend, please confirm till Wednesday (send me an email), we need to have a fair estimation of attendees number to make the restaurant reservation. See more at: http://sip-router.org/2010/01/31/fosdem-2010-meeting-in-brussels/ See you at Fosdem, Daniel -- Daniel-Constantin Mierla eLearning Class for Kamailio 3.0.0 Starts Feb 8, 2010 * http://www.asipto.com/

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[Kamailio-Users] project updates
by Daniel-Constantin Mierla 01 Feb '10

01 Feb '10

Hello, I do not know how many noticed so far, for several days the management web page was update to reflect an enlarged group of people that represent Kamailio project. Since the collaboration within SIP Router project proved fruitful, it is clear that the future development is related to that project, therefore devel decisions will happen there (join sr-dev(a)lists.sip-router.org to be in touch). The last year and half was filled with heavy integration work and testing, most of it not very challenging, but had to be done, so we wanted to acknowledge the help given by our most active community contributors. As a project policy, we maintain officially the latest two stable releases, now being 1.5.x and 3.0.x. The efforts of the group will be directed in advocating and promoting Kamailio and SIP Router projects world wide. Also some of them sponsor and take care of maintaining resources such as servers, dns and web pages. http://www.kamailio.org/mos/view/Management/ Given the big amount of work done for 3.0.0, the feedback so far is very good, in these tree weeks we discovered few issues, most relevant affecting sip_trace and uac modules, with a pretty active discussions pool about this version. 3.0.0 branches are run by some open voip providers for quite some time now (e.g., voipuser.org and iptel.org), servicing thousands of active users, ensuring that the core modules (e.g., acc, tm, auth, nat traversal, group, dialog, presence, dispatcher, pv, textops, ...) are working smooth. As a side note, I have it in other several production systems. While I encourage everyone to send feedback about 3.0.0, worth to mention that devel version (git master branch to become 3.1.0) has already quite a lot of new features, so if you are looking for adventures or need Lua API, geoip location, in memory tree indexed data, Python API, user registrations to remote SIP servers (for inbound calls), check-out git master and give it a try. Note that master is as stable as 3.0.0 branches so far regarding old code, new additions haven't touched other parts, being stand alone. Looking forward to 3.1.0 as introducing new features but also providing more tight integration with SIP Router, especially on some common used modules. Cheers, Daniel

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[Kamailio-Users] help me to find the problem
by shaon＠opennets.net 31 Jan '10

31 Jan '10

Hello I am new with kamailio. here is my problem X-lite -->TLS-->Kamailio(TLS+RegisterForwrd)-->UDP-->RegisterServer-->PSTN/VOIPCall I am able to forward my register request to the register server, and make call. Able to talk from both side. But when i hangup the call from termination side call never hangup from origination side. I attached my config file. I got following log on my kamailio.log file ############# Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:tm:t_lookup_request: start searching: hash=53225, isACK=0 Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:tm:t_lookup_request: proceeding to pre-RFC3261 transaction matching Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:tm:t_lookup_request: no transaction found Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:tm:run_reqin_callbacks: trans=0xb61945d0, callback type 1, id 0 entered Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:core:mk_proxy: doing DNS lookup... Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:core:get_send_socket: force_send_socket of different proto (1)! Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: WARNING:core:get_send_socket: protocol/port mismatch Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:tm:set_timer: relative timeout is 500000 Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:tm:insert_timer_unsafe: [4]: 0xb619471c (40400000) Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:tm:set_timer: relative timeout is 30 Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:tm:insert_timer_unsafe: [0]: 0xb6194738 (69) Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:tm:t_relay_to: new transaction fwd'ed Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:core:destroy_avp_list: destroying list (nil) Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7579]: DBG:core:receive_msg: cleaning up Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7581]: DBG:tm:utimer_routine: timer routine:4,tl=0xb619471c next=(nil), timeout=40400000 Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7581]: DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0xb61945d0, BYE sip:9 ... ) Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7581]: DBG:tm:set_timer: relative timeout is 1000000 Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7581]: DBG:tm:insert_timer_unsafe: [5]: 0xb619471c (41400000) Jan 31 15:54:45 ser /usr/local/sbin/kamailio[7581]: DBG:tm:retransmission_handler: retransmission_handler : done Jan 31 15:54:46 ser /usr/local/sbin/kamailio[7581]: DBG:tm:utimer_routine: timer routine:5,tl=0xb619471c next=(nil), timeout=41400000 Jan 31 15:54:46 ser /usr/local/sbin/kamailio[7581]: DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0xb61945d0, BYE sip:9 ... ) Can anybody help me to solve the issue? Shaon

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[Kamailio-Users] ERROR:tm:w_t_reply failed to send a t_reply to a message for which no transaction-ate has been established
by AnthonyD 31 Jan '10

31 Jan '10

Dear List, I am running Kamailio (1.5.3-notls (i386/linux)) with the following modules: loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "avpops.so" loadmodule "lcr.so" loadmodule "dialog.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "domain.so" I get a lot of these errors in my logs: ERROR:tm:w_t_reply failed to send a t_reply to a message for which no transaction-ate has been established Can someone give me some info on these errors and how to trouble shoot the cause? Thanks Anthony Drozdek

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[Kamailio-Users] DTLS with kamailio
by mustafa samara 31 Jan '10

31 Jan '10

Hello I want to try DTLS with kamailio, I am working on minisip it support DTLS, but what a bout the kamailio side? is there any thing to do? also i would ask about IPSec support is there any thing to do on kamailio side, or it just on the kernel level? any help please? thanks -- Eng.Mustafa Al-Samara

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[Kamailio-Users] tcp_blocking_connect related errors and Kamailio using 100% CPU
by Pascal Maugeri 29 Jan '10

29 Jan '10

Hi Using Kamailio 1.5 we periodically get the following errors: Jul 1 09:54:41 localhost /usr/local/sbin/openser[3836]: ERROR:core:tcp_blocking_connect: poll error: flags 18 Jul 1 09:54:41 localhost /usr/local/sbin/openser[3836]: ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR (110) Connection timed out Jul 1 09:54:41 localhost /usr/local/sbin/openser[3836]: ERROR:core:tcpconn_connect: tcp_blocking_connect failed Jul 1 09:54:41 localhost /usr/local/sbin/openser[3836]: ERROR:core:tcp_send: connect failed Jul 1 09:54:41 localhost /usr/local/sbin/openser[3836]: ERROR:tm:msg_send: tcp_send failed They are several emails in this mailing list explaining we should play with tcp parameters. I will try to change my configuration and see if we can solve this issue. But when this kind of error happen we observe that the kamailio processes are using 100% of CPU (top). Our client is using SIP over TCP only (no UDP). Do you have any hint on how to configure Kamailio to improve TCP support ? Is it normal to kamailio processes to overload CPU ? Regards, Pascal

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Re: [SR-Users] TLS problems
by Andreas Rehbein 29 Jan '10

29 Jan '10

Hi, I solved the problem within snom370 registration with TLS: It was necessary to set the TLS-Modul Parameter "cipher_list" to RSA. Now the snom 370 registers via TLS. Thanks Klaus! regards Andreas

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Re: [SR-Users] TLS problems
by Andreas Rehbein 29 Jan '10

29 Jan '10

Hello Klaus, thank you very much! I really appreciate your help! I followed your instructions and I hope that the attached file includes the necessary informations. regards Andreas -----Ursprüngliche Nachricht----- Von: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Gesendet: Freitag, 22. Januar 2010 10:50 An: Andreas Rehbein Betreff: Re: AW: [SR-Users] TLS problems Andreas Rehbein schrieb: > Hi Klaus! > > Yes, the crash happens everytime (and immediately) when I push the > "Register"-Button in snoms web-gui. > > I attached the backtrace-file, but until now I did not rebuild kamailio with > -DDBG_QM_MALLOC. Please let me know if it's necessary. the log only shows the last function call. Please start gdb again and at the gdb prompt generate a backtrace by entering bt, e.g: (gdb) bt regards klaus > > regards > Andreas > > -----Ursprüngliche Nachricht----- > Von: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Gesendet: Freitag, 22. Januar 2010 09:35 > An: Andreas Rehbein > Cc: sr-users(a)lists.sip-router.org > Betreff: Re: [SR-Users] TLS problems > > Hi Andreas! > > Maybe this is the same bug I reported yesterday. Unfortunately I can not > reproduce the crash anymore. > > Does the crash happen everytime? > > Kamailio produced a core dump: > > 0(6257) ALERT: <core> [main.c:725]: core was generated > > So please send a the backtrace: > # gdb kamailio /path/to/corefile > > bt > > Location of the core file is usually / or /tmp or the kamailio's working > directory. Use > # cat /proc/sys/kernel/core_pattern > to find the exact location of the core dump. > > Further, it might be useful to enable memory debugging: > - in Makefile.defs use -DDBG_QM_MALLOC instead of -DF_MALLOC and > rebuild/reinstall kamailio > - set higher loglevels in kamailio.cfg > memlog=3 > memdbg=3 > debug=3 > modparam("tls", "tls_log", 3) > > Note: this excessive logging makes kamailio start/stop real slow and > logfile will increase rapidly (might fill up your hard disk) > > regards > klaus > > > > Andreas Rehbein schrieb: >> Hi, >> >> >> >> Ive installed Kamailio 3.0 on RHEL5 and activated the mysql backend and >> tls support. Everything works fine with Soft User Agent Phoner lite. >> My Problem is: If I try to register a snom 370 Kamailio crashes >> immediatly. What is wrong? Any suggestions? >> >> Debug: >> >> 17(6290) DEBUG: <core> [ip_addr.c:116]: tcpconn_new: new tcp connection: >> 192.168.0.222 >> >> 17(6290) DEBUG: <core> [tcp_main.c:1052]: tcpconn_new: on port 1430, type > 3 >> 17(6290) DEBUG: <core> [tcp_main.c:1351]: tcpconn_add: hashes: >> 978:1690:1400, 1 >> >> 17(6290) DEBUG: <core> [io_wait.h:361]: DBG: io_watch_add(0x8222fe0, 31, >> 2, 0xb613a2b8), fd_no=24 >> >> 17(6290) DEBUG: <core> [io_wait.h:588]: DBG: io_watch_del (0x8222fe0, >> 31, -1, 0x0) fd_no=25 called >> >> 17(6290) DEBUG: <core> [tcp_main.c:3627]: tcp: DBG: sendig to child, >> events 1 >> >> 17(6290) DEBUG: <core> [tcp_main.c:3336]: send2child: to tcp child 0 >> 13(6283), 0xb613a2b8 >> >> 13(6283) DEBUG: <core> [tcp_read.c:884]: received n=4 con=0xb613a2b8, fd=8 >> >> 13(6283) DEBUG: tls [tls_server.c:109]: Using TLS domain TLSs<default> >> >> 17(6290) DEBUG: <core> [tcp_main.c:2826]: DBG: handle_tcp_child: dead >> tcp child 0 (pid 6283, no 13) (shutting down?) >> >> 17(6290) DEBUG: <core> [io_wait.h:588]: DBG: io_watch_del (0x8222fe0, >> 24, -1, 0x0) fd_no=24 called >> >> 17(6290) : <core> [pass_fd.c:283]: ERROR: receive_fd: EOF on 4 >> >> 17(6290) DEBUG: <core> [tcp_main.c:3038]: DBG: handle_ser_child: dead >> child 13, pid 6283 (shutting down?) >> >> 17(6290) DEBUG: <core> [io_wait.h:588]: DBG: io_watch_del (0x8222fe0, 4, >> -1, 0x0) fd_no=23 called >> >> 0(6257) ALERT: <core> [main.c:722]: child process 6283 exited by a >> signal 11 >> >> 0(6257) ALERT: <core> [main.c:725]: core was generated >> >> 0(6257) INFO: <core> [main.c:737]: INFO: terminating due to SIGCHLD >> >> 17(6290) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 15(6286) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 16(6289) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 12(6281) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 14(6285) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 10(6278) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 6(6269) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 2(6261) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 11(6280) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 1(6260) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 9(6276) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 8(6272) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 3(6262) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 7(6270) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 5(6267) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 4(6265) INFO: <core> [main.c:788]: INFO: signal 15 received >> >> 0(6257) DEBUG: tm [t_funcs.c:122]: DEBUG: tm_shutdown : start >> >> 0(6257) DEBUG: tm [t_funcs.c:125]: DEBUG: tm_shutdown : emptying hash > table >> 0(6257) DEBUG: tm [t_funcs.c:127]: DEBUG: tm_shutdown : removing > semaphores >> 0(6257) DEBUG: tm [t_funcs.c:129]: DEBUG: tm_shutdown : destroying tmcb >> lists >> >> 0(6257) DEBUG: tm [t_funcs.c:132]: DEBUG: tm_shutdown : done >> >> 0(6257) DEBUG: tls [tls_init.c:621]: tls module final tls destroy >> >> 0(6257) DEBUG: <core> [mem/shm_mem.c:236]: shm_mem_destroy >> >> 0(6257) DEBUG: <core> [mem/shm_mem.c:239]: destroying the shared memory >> lock >> >> 0(6257) DEBUG: <core> [main.c:741]: terminating due to SIGCHLD >> >> >> >> tia >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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[Kamailio-Users] kamailio 1.5.4
by Daniel-Constantin Mierla 29 Jan '10

29 Jan '10

Hello, should we package kamailio 1.5.4? 1.5.3 was in october last year, there were some commits around since then. Next week monday? 1.4.x got out of maintenance period, should we do the 1.4.5 (just release tarballs) to mark that? Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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Re: [SR-Users] Request for a special parallel + serial forking scenario
by Klaus Feichtinger 29 Jan '10

29 Jan '10

> > Klaus Feichtinger schrieb: > > > >>> Hi, > >>> > >>> I have a special question regarding a mixture of parallel and serial > >>> forking, operated by Kamailio-3.0.0. > >>> > >>> First of all I will give you a short background information, what the > >>> target will be: > >>> > >>> I have a SIP-server on place A, 3 gateways on place B and 3 gateways > >>> on place C). These gateways are registered on SIP server (A) with > >>> different Q-values (e.g. GW-B1=1.0, GW-C1=1.0, GW-B2=0.8, GW-C2=0.8, > >>> GW-B3=0.6, GW-C3=0.6). The target is, that a call for a gateway MUST > >>> be signalled on place B and C in parallel (forking) until the call is > >>> finally established over one of the two involved gateways on place B > >>> or C. When the prime gateway(s) (with the highest Q-value) fail (e.g. > >>> they do not send a provisional response within a timeout or send a > >>> negative response), the next gateway(s) should be addressed (with the > >>> next lower Q-value), a.s.o. > >>> > >>> The configuration works well in case that both gateways on place B > >>> and C fail at the same time (BOTH do not send a response or BOTH send > >>> a negative response or one does not send a response and the other one > >>> sends a negative response). However, in case that only ONE of the two > >>> (always in parallel) addressed gateways fails, it does not work as > >>> expected. > >> > >> What is expected? > >> > >> Do you want to trigger failover to lower priority even if one the > >> gateways is still working fine? You can not do that with sip-router. > >> sr implements parallel forking according to RFC3261, this means it > >> will not generate a failure response as long as one of the branches is > >> still active, maybe sending 200 ok. > >> > > The ideal behaviour would be following: > > - try reaching a gateway on place B and C until one gateway on every > > place (B AND C) sends back a positive (provisional) response - e.g. 180 > > ringing, 183 session progress > > > > - in case that one of the two parallel addressed gateways is reachable > > and sends back a provisional response (= "it is alive"), keep it > > > > - in case that the other one (of the two parallel addressed gateways) > > has a problem (negative response 480 / 404 / 486 ...) or is unreachable > > (= |fr_inv_timer_next| timeout), try the next gateway on this specific > place > > > I'm still not sure if I understand it correct. May examples help: > > the call is forwarded to B1 and C1. B1 does not send back anything, C1 > sends back 100 trying. What is the expected behavior? > - CANCEL C1 and send the call to B2 and C2? > - keep C1 and send another call to B2? > - stay with B1 unreachable and possible C1 answering the call? > The second variant with "keep C1 and send another call to B2" is the expected behaviour. > > - the parallelism is necessary, because calls over these gateways are > > addressed to a usergroup which is split to place B and C behind a > > redundant "non-SIP" PBX system with two active halfs); you can not > > suspect if a user on place B or a user on place C will take the call; > > therefore the call has to be sent to both places. > > Ok. > > > However, the favored behaviour is not standard conform - I know. I guess > > therefore I would need a B2BUA, which is able to establish two > > independent (serially forked) preliminary connections to gateways on > > place B and C and interconnects the successful call to the call > > originator. I fear that no possibility will be available with Kamailio / > > a standard SIP server. > > Maybe you can achieve the wanted behavior by modifying tm module to > CANCEL all branches if a single branch fail (similar to processing of > 6xx error response, which also cancels all branches). > > Depending on the exact needed behavior you might be able to do it with > Asterisk. > > regards > klaus > an alternative solution I think about (but have not yet tested) might be following: - SIP server on place A fork the call to SIP servers on place B and C (both with a fix usrloc contact => parallel forking to two fix destinations) - SIP server on place B fork the call to the gateways on place B serial - SIP server on place C fork the call to the gateways on place C serial The advantage of this solution might be that not one server is responsible for making the mixture of serial AND parallel forking. Each server is responsible for making EITHER parallel (= place A) OR serial (= places B and C) forking. Could this be a practicable alternative? regards Klaus F. -- GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01

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[Kamailio-Users] new feature: geoip location in cfg
by Daniel-Constantin Mierla 29 Jan '10

29 Jan '10

Hello, a new module was uploaded to master branch to lookup ip addresses against a geoip database. The module supports many lookups at the same time and returns the details in pseudo-variables. Therefore a new pv class: $gip(pvc=>key). pvc is the id of the container where to store the results of a geoip search, key is an id for the attribute you want to receive. See details on wiki: http://sip-router.org/wiki/cookbooks/pseudo-variables/devel#geoip_module_ps… Readme of the module: http://sip-router.org/docbook/sip-router/branch/master/modules/geoip/geoip.… Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Kamailio-Users] documentation team
by Daniel-Constantin Mierla 29 Jan '10

29 Jan '10

Hello, following some discussions on users mailing list, seems we may get a group of people willing to digest documentation and contribute to modules' readme or wiki pages. Hopefully Alex Balashov (who volunteered so far) would have some time to coordinate contributions to git docs, for wiki being easier, as anyone has access. I am willing to help as much I can, but time is always a constraint for a dev. I invite you to join IRC channel #sip-router hosted on freenode.net to plan an synchronize easier. Thanks, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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Re: [Kamailio-Users] /usr/local/sbin/kamdbctl create error
by Daniel-Constantin Mierla 29 Jan '10

29 Jan '10

please cc the mailing list, otherwise your emails will be ignored. do you have the mysql client installed? Can you connect to the mysql server from terminal with 'mysql' command? Cheers, Daniel On 25.11.2009 19:42 Uhr, YÜKSEL wrote: > Dear Daniel, > > Thanks for your response : Actually I am getting an error below when > I try to create DB with `/usr/local/sbin/kamdbctl create` > > debian:/usr/local/src/kamailio-1.5.0/sip-server# > /usr/local/sbin/kamdbctl create > MySQL password for root: > INFO: test server charset > /usr/local/lib/kamailio/kamctl/kamdbctl.mysql: line 105: mysql: > command not found > /usr/local/lib/kamailio/kamctl/kamdbctl.mysql: line 106: mysql: > command not found > Usage: grep [OPTION]... PATTERN [FILE]... > Try `grep --help' for more information. > /usr/local/lib/kamailio/kamctl/kamdbctl.mysql: line 112: [: =: unary > operator expected > INFO: creating database openser ... > /usr/local/lib/kamailio/kamctl/kamdbctl.mysql: line 71: mysql: command > not found > ERROR: Creating core database and grant privileges failed! > > I`ve applied the instructor one by one but I didn`t understand whay I > am getting this error. > > Regards, > > Yuksel > > > --- On *Wed, 11/25/09, Daniel-Constantin Mierla /<miconda(a)gmail.com>/* > wrote: > > > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [Kamailio-Users] /usr/local/sbin/kamdbctl create error > To: "lazturk" <lazturk(a)yahoo.com> > Cc: users(a)lists.kamailio.org > Date: Wednesday, November 25, 2009, 12:33 PM > > Hello, > > > On 25.11.2009 17:59 Uhr, lazturk wrote: > > Hello, > > > > How did you solve this problem bcz I am getting a similar error ; > > > > which one? the one related to no database engine specified or the > one related to mysql socket file? > > First: updated the kamctlrc file -- see the comments inside. > > Second: check if mysql server is running and if yes, check the > mysql config to see where the server is listening and where the > mysql client tries to connect to. They must be the same. > > Cheers, > Daniel > > > > > Thanks for your feedback. > > > > > > vivi-8 wrote: > > > >> Hi all > >> I have specify the wanted db type (DBENGINE=MYSQL) in the > >> /usr/local/etc/kamailio/kamctlrc > >> then I using "/usr/local/sbin/kamdbctl create" to create MySQL > database, > >> but I got this error: > >> > >> > >> > >> ERROR: database engine not specified, please setup one in the > config > >> script > >> > >> root@acer:/usr/local/src/kamailio-1.5.0/sip-server# vim > >> /usr/local/etc/kamailio/kamctlrc > >> > >> root@acer:/usr/local/src/kamailio-1.5.0/sip-server# > >> /usr/local/sbin/kamdbctl > >> create > >> > >> MySQL password for root: > >> > >> INFO: test server charset > >> > >> ERROR 2002 (HY000): Can't connect to local MySQL server through > socket > >> '/var/run/mysqld/mysqld.sock' (2) > >> > >> ERROR 2002 (HY000): Can't connect to local MySQL server through > socket > >> '/var/run/mysqld/mysqld.sock' (2) > >> > >> Usage: grep [OPTION]... PATTERN [FILE]... > >> > >> Try `grep --help' for more information. > >> > >> /usr/local/lib/kamailio/kamctl/kamdbctl.mysql: line 112: [: =: > unary > >> operator expected > >> > >> INFO: creating database openser ... > >> > >> ERROR 2002 (HY000): Can't connect to local MySQL server through > socket > >> '/var/run/mysqld/mysqld.sock' (2) > >> > >> ERROR: Creating core database and grant privileges failed! > >> > >> > >> > >> Cheers, > >> > >> vivi > >> > >> > >> _______________________________________________ > >> Kamailio (OpenSER) - Users mailing list > >> Users(a)lists.kamailio.org </mc/compose?to=Users(a)lists.kamailio.org> > >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > >> > >> > > > > > > -- Daniel-Constantin Mierla > * http://www.asipto.com/ > > -- Daniel-Constantin Mierla * http://www.asipto.com/

3 3

Request for a special parallel + serial forking scenario
by Klaus Feichtinger 29 Jan '10

29 Jan '10

Hi, I have a special question regarding a mixture of parallel and serial forking, operated by Kamailio-3.0.0. First of all I will give you a short background information, what the target will be: I have a SIP-server on place A, 3 gateways on place B and 3 gateways on place C). These gateways are registered on SIP server (A) with different Q-values (e.g. GW-B1=1.0, GW-C1=1.0, GW-B2=0.8, GW-C2=0.8, GW-B3=0.6, GW-C3=0.6). The target is, that a call for a gateway MUST be signalled on place B and C in parallel (forking) until the call is finally established over one of the two involved gateways on place B or C. When the prime gateway(s) (with the highest Q-value) fail (e.g. they do not send a provisional response within a timeout or send a negative response), the next gateway(s) should be addressed (with the next lower Q-value), a.s.o. The configuration works well in case that both gateways on place B and C fail at the same time (BOTH do not send a response or BOTH send a negative response or one does not send a response and the other one sends a negative response). However, in case that only ONE of the two (always in parallel) addressed gateways fails, it does not work as expected. It does nothing and/or waits for an eventually negative response or the call setup timeout. Configuration is done as described in the README of the TM module. It is a mixture of the main route and a failure_route in combination with the functions t_load_contacts() and t_next_contacts(). Does anybody have an idea how this problem could be solved or any alternative solution for this special requirement of parallel signalisation in any case? Thanks in advance! Regards, Klaus -- GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01

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[Kamailio-Users] rtpproxy session is not automatically deleted if INVITE transaction fails?
by Iñaki Baz Castillo 28 Jan '10

28 Jan '10

Hi, I'm realizing that rtpproxy session is not automatically terminated when an outgoing INVITE transaction fails ([3456]XX response received from downstream), so the rtpproxy session dies after a while due to time expiration. So I've added the following code to my on_reply_route: ------------ onreply_route[X] { if is_method("INVITE") && status =~ "[3456][0-9][0-9]" unforce_rtp_proxy(); ------------ It seems to do the job. However I expected this to be done automatically by rtpproxy module when an INVITE transaction for which a rtpproxy session has been created fails. Do I miss something? or is it the expected behavior? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] new bunch of PVs
by Daniel-Constantin Mierla 28 Jan '10

28 Jan '10

Hello, I added a bunch of new variables to deal easier with next hop address and be able to retrieve micro-seconds in the config file. Check wiki at: http://sip-router.org/wiki/cookbooks/pseudo-variables/devel#timeval http://sip-router.org/wiki/cookbooks/pseudo-variables/devel#next_hop_address Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Kamailio-Users] Problems with snmpstats in Kamailio-3.0.0
by Santiago Gimeno 28 Jan '10

28 Jan '10

Hi, I'm trying to migrate from K-1.5 to K-3.0.0 and I'm having problems loading the snmpstats module in kamailio-3.0.0. The error is: ERROR: <core> [sr_module.c:390]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/snmpstats.so>: /usr/local/lib/kamailio/modules_k/snmpstats.so: undefined symbol: module_loaded By checking the code it looks like module_loaded function is not implemented in sr_module.c anymore. Could this be the problem? Thanks. Best regards, Santi

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Re: [Kamailio-Users] Bad URI
by Jon Farmer 28 Jan '10

28 Jan '10

On 28 January 2010 13:41, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > > > On 1/28/10 2:36 PM, Jon Farmer wrote: >>> >>> is the call going via route[2]? >>> >>> Try to print the whole sip message $mb before and after executing your >>> php >>> script. >>> >> >> Yes route[2] >> >> Here is a link to the $mb before and after http://pastebin.com/d4b7ca03 >> > > k, it is good, however $mb is always received message so it is the same > before and after -- just wanted to check if got corrupted. > > I want to see now the r-uri before and after: $ru > > Daniel >> Aha found it... my PHP is returning error when URI has ";user=phone". Daniel you are a star thank you for your help and patience on this one. -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

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[Kamailio-Users] Request for a special parallel serial forking scenario
by Klaus Feichtinger 28 Jan '10

28 Jan '10

Hi Kamailio users, I have a special question regarding a mixture of serial and parallel forking, operated by Kamailio-3.0.0. First of all I will give you a short background information, what the target will be: I have a SIP-server on place A, 3 gateways on place B and 3 gateways on place C). All (6) gateways are registered with the same username on SIP server (A), but with different Q-values (e.g. GW-B1=1.0, GW-C1=1.0, GW-B2=0.8, GW-C2=0.8, GW-B3=0.6, GW-C3=0.6). The target is, that a call for a gateway MUST be signalled on gateway(s) of place B and C in parallel (forking) until the call is finally established over one of the two involved gateways on place B or C. When (one of) the prime gateway(s) (with the highest Q-value) fail (e.g. they do not send a provisional response within a timeout or send a negative response), the next gateway(s) should be addressed (with the next lower Q-value), a.s.o. The parallelism for signalisation on place B and C is necessary, because of redundancy / safety reasons. Configuration for serial forking is done as described in the README of the TM module. It is a mixture of the main route and a failure_route in combination with the functions t_load_contacts() and t_next_contacts(). The loaded contacts for serial forking differ (from standard scenarios) in that way, that each sequential target consists of a pair of "users" instead of a single user. This configuration works well in case that both gateways on place B and C fail at the same time (BOTH do not send a response or BOTH send a negative response or one does not send a response and the other one sends a negative response). However, in case that only ONE of the two (always in parallel) addressed gateways fails, it does not work as expected. It does nothing and/or waits for an eventually negative response or the call setup timeout. Only then the next gateway-pair will be addressed. => is the behaviour of TM correct/"as designed" to make an AND-relation with both branches and access the failure_route only then, when both branches fail? Does anybody have an idea how this problem could be solved or any alternative solution for this special requirement of parallel signalisation in any case? Thanks in advance! Regards, Klaus -- GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01

1 0

[Kamailio-Users] TM module
by Denis Putyato 28 Jan '10

28 Jan '10

Hello everybody! ERROR: tm [tm_load.c:47]: tm:load_tm: Module not initialized yet, make sure that all modules that need tm module are loaded after tm in the configuration file Kamailio 3.0 was installed from GIT. Kamailio starts success. But the error appears. I think because of this I have a problem with making a call through kamailio. Calls were failed because kamailio send CANCEL or 487 code immediately after send INVITE to some outgoing gateways (after t_relay() has been made). Is this really need load all modules for tm working properly? I read documentation and fount that almost all modules that need tm loaded do not need in my config. Thank you for any help.

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Re: [Kamailio-Users] Bad URI
by Jon Farmer 28 Jan '10

28 Jan '10

2010/1/27 Iñaki Baz Castillo <ibc(a)aliax.net>: > El Miércoles, 27 de Enero de 2010, escribió: >> 2010/1/27 Iñaki Baz Castillo <ibc(a)aliax.net>: >> > El Miércoles, 27 de Enero de 2010, Jon Farmer escribió: >> >> Hi >> >> >> >> Does anyone know why I am getting >> >> >> >> ERROR: parse_uri: bad host in uri (error at char >> >> >> >> from this ( i hidden the domain and IP) >> > >> > How does the domain look? To show it replace the letters but keep the >> > symbols (as underscore and so). >> >> Hi, >> >> Sorry I am not entirely sure what you mean could you give me an example >> please. >> > > Hi, please keep the thread in the maillist. > > I just mean that, to keep your domain hidden, just change the letters but no > the symbols. Example: > > - Real domain: mycompany-cool-voip.com > - Showed domain: lalalaala-lolo-lili.com > > -- > Iñaki Baz Castillo <ibc(a)aliax.net> > Sorry about the wrong reply address, blame gmail :-) Here you go INVITE sip:01684299007@mydomain.net:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.20.240:5060;branch=z9hG4bK-d8754z-ed473d0ab724045f-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:500@78.32.132.193:5060>. To: <sip:01684299007@mydomain.net:5060>. From: "Michele "<sip:500@mydomain.net:5060>;tag=fd5b583a. Call-ID: NzU1MTZiMjZjZGM0ZTY5ZmFhMWRjNDlkNGQxZTZmODM.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO. Content-Type: application/sdp. User-Agent: 3CXPhoneSystem 7.0.4708.0. Authorization: Digest username="500",realm="mydomain.net",nonce="4b606dcc81900a95d9962d3c737cc570e624d409",uri="sip:01684299007@mydomain.net:5060",response="1ac466e73c744ed5936179152b5b27d4",algorithm=MD5. Content-Length: 278. . v=0. o=3cxPS 444025798656 5368709121 IN IP4 78.32.132.193. s=3cxPS Audio call. c=IN IP4 78.32.132.193. t=0 0. m=audio 9000 RTP/AVP 0 8 3 101. c=IN IP4 78.32.132.193. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

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Re: [Kamailio-Users] ACC on Failure route
by Daniel-Constantin Mierla 28 Jan '10

28 Jan '10

On 1/25/10 2:10 PM, alex pappas wrote: > > What exactly is iirc? common abbreviation in net-slang, helping to save the planet, think green, by sending less bits on net :-): http://www.internetslang.com/IIRC.asp Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Kamailio-Users] problem with ACC module
by Denis Putyato 28 Jan '10

28 Jan '10

Hello everybody! I have a such problem with ACC module. Calls was accounted but only INVITE and BYE requests has appeared in acc table. ACK is absent. Kamailio 3.0 was installed from GIT. Thank you for any help # ----- acc params ----- modparam("acc", "early_media", 0) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 1) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "failed_transaction_flag", 29) modparam("acc", "db_table_acc", "acc") modparam("acc", "db_table_missed_calls", "missed_calls") modparam("acc", "acc_method_column", "method") modparam("acc", "acc_from_tag_column", "from_tag") modparam("acc", "acc_to_tag_column", "to_tag") modparam("acc", "acc_callid_column", "callid") modparam("acc", "acc_sip_code_column", "sip_code") modparam("acc", "acc_sip_reason_column", "sip_reason") modparam("acc", "acc_time_column", "time") modparam("acc", "db_url", "xxxxxxxxxxxxxxxxxxxxxxxx") modparam("acc", "db_extra", "src_user=$avp(i:2);src_domain=$si;dst_ouser=$avp(i:1);dst_user=$rU;dst_doma in=$rd") route { sip_trace(); setflag(30); if (has_totag()) { if (loose_route()) { if (is_method("BYE")) { setflag(1); setflag(29); } if (is_method("ACK")) { xlog("L_NOTICE", "ACK detected"); # I see this line in syslog setflag(1); setflag(29); } route("relay"); exit; } else { if ( is_method("ACK") ) { if ( t_check_trans() ) { setflag(1); route("relay"); exit; } else { exit; } } } }

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[Kamailio-Users] Dynamic setting of dispatcher route set
by Alex Balashov 27 Jan '10

27 Jan '10

Unfortunately, ds_select_dst()/ds_select_domain() do not seem to take PVs for their first argument (the route set ID from which to choose). Is there any workaround for this, e.g. when the route set is indeterminate until runtime? It would be very inconvenient to have to work around this by implementing own rollover logic when the dispatcher module is so good... perhaps the possibility of a quick and easy patch to the git trunk? Thanks in advance! -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/

2 2

[Kamailio-Users] Bad URI
by Jon Farmer 27 Jan '10

27 Jan '10

Hi Does anyone know why I am getting ERROR: parse_uri: bad host in uri (error at char from this ( i hidden the domain and IP) INVITE sip:xxxxxxxx@mydomain.net:5060;user=phone SIP/2.0. Via: SIP/2.0/UDP xx.xxx.xxx.xxx:5060;branch=z9hG4bKea312c9f5673C67A. From: "200" <sip:200@mydoamin.net>;tag=D6C112A0-D880344D. To: <sip:xxxxxxxx@mydomain.net;user=phone>. CSeq: 2 INVITE. Call-ID: f1f44bbc-58cbed36-2ea0cdab(a)xx.xxx.xxx.xxx. Contact: <sip:200@xx.xxx.xxx.xxx:5060>. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.3.0067. Supported: 100rel,replace. Allow-Events: talk,hold,conference. Authorization: Digest username="200", realm="mydomain.net", nonce="4b605fcd30e54ca4d6b52ffc9fc6128bc355cf71", uri="sip:xxxxxxxx@mydomain.net:5060;user=phone", response="7c82f9511843dc55aeaed27f468efe39", algorithm=MD5. Max-Forwards: 70. Content-Type: application/sdp. Content-Length: 254. . v=0. o=- 1264546283 1264546283 IN IP4 xx.xxx.xxx.xxx. s=Polycom IP Phone. c=IN IP4 xx.xxx.xxx.xxx. t=0 0. a=sendrecv. m=audio 10010 RTP/AVP 0 8 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

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Re: [Kamailio-Users] [sr-dev] Rtpproxy transcoder patch
by Henning Westerholt 27 Jan '10

27 Jan '10

On Tuesday 26 January 2010, Vikram Ragukumar wrote: > I came across a post by Stefan Sayer regarding an rtpproxy transcoder > patch http://lists.iptel.org/pipermail/serdev/2008-July/012794.html. > We are interested in having rtpproxy perform transcoding, both in > software and using DSP array cards, and are currently running rtpproxy > version 1.2.1. > Is this link http://user.cs.tu-berlin.de/~sayer/transcoder/ that is > mentioned on the post our starting point or is there an updated > version ? Hi Vikram, I don't know about this particiular case, so I can't give you more informations in this regards. But i think Stefan read this list as well, if you don't get a reply here perhaps you can contact him direcly by mail, if you have not tried this yet. Cheers, Henning

2 1

[Kamailio-Users] Wrong 1.5 wiki Cookbook page?
by Iñaki Baz Castillo 27 Jan '10

27 Jan '10

Hi, under 1.5 cookbook I see some stuf related to Kamailio 3.0. For example "debug" values are those for SR (if I'm not wrong) and the appearing command to change debug leves uses "sercmd" rather than "kamctl": http://kamailio.org/dokuwiki/doku.php/core-cookbook:1.5.x#debug Is it correct? -- Iñaki Baz Castillo <ibc(a)aliax.net>

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Re: [Kamailio-Users] LCR module in Kamailio 3.0
by Daniel-Constantin Mierla 27 Jan '10

27 Jan '10

For reference: http://kamailio.org/dokuwiki/doku.php/install:1.5.x-to-3.0.0 It helps to upgrade to 3.0.0, feel free to add to the page. In this particular case, kex initializes mi engine ad registers core commands. Still don't see the dr_reload, probably the command is not registered by module's mod init. I will check. Cheers, Daniel On 1/27/10 6:43 AM, Denis Putyato wrote: > > After load kex: > > root@ubuntutest:/kamailio# kamctl fifo which > > shv_get > > shv_set > > sip_trace > > dp_reload > > dp_translate > > lcr_reload > > lcr_gw_dump > > lcr_lcr_dump > > cr_reload_routes > > cr_dump_routes > > cr_replace_host > > cr_deactivate_host > > cr_activate_host > > cr_add_host > > cr_delete_host > > uptime > > version > > pwd > > arg > > which > > kill > > ps > > debug > > get_statistics > > reset_statistics > > *From:* Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* Tuesday, January 26, 2010 9:00 PM > *To:* Denis Putyato > *Cc:* users(a)lists.kamailio.org > *Subject:* Re: [Kamailio-Users] LCR module in Kamailio 3.0 > > > > On 1/26/10 1:01 PM, Denis Putyato wrote: > > root@ubuntutest:/kamailio# kamctl fifo which > > 500 command 'which' not available > > > do you have the kex module loaded? > > Any fifo command that works for you? > > Cheers, > Daniel > > > root@ubuntutest:/kamailio# > > *From:* Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* Tuesday, January 26, 2010 2:55 PM > *To:* Denis Putyato > *Cc:* users(a)lists.kamailio.org <mailto:users@lists.kamailio.org> > *Subject:* Re: [Kamailio-Users] LCR module in Kamailio 3.0 > > > > On 1/26/10 12:35 PM, Denis Putyato wrote: > > I do not use lcr routing in config (only gw table is used). > > ok > > > > Not in kamailio 1.5 nor 3.0. I use carrierroute (in 1.5) and now > drouting (in 3.0). > > By the way. Recently you helped me with drouting. Can I ask one more > question about this module? > > When I try to do "kamctl fifo dr_reload" I receive "500 command > 'dr_reload' not available". > > > do: > > kamctl fifo which > > Should list available MI commands, is dr_reload there? > > Cheers, > Daniel > > > > *From:* Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* Tuesday, January 26, 2010 12:43 PM > *To:* Denis Putyato > *Cc:* users(a)lists.kamailio.org <mailto:users@lists.kamailio.org> > *Subject:* Re: [Kamailio-Users] LCR module in Kamailio 3.0 > > > > On 1/26/10 10:38 AM, Denis Putyato wrote: > > Daniel thank you. Now it's working > > > welcome! Be sure you understand the idea behind the instances. Not > being a user of lcr don't know exactly the internal architecture, so > as I understood from discussions on mailing lists, you can have more > lcr instance on same kamailio grouped under same value of lcr_id (many > gws and lcr rules with same lcr_id). > > Cheers, > Daniel > > > > > *From:* Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* Tuesday, January 26, 2010 12:31 PM > *To:* Denis Putyato > *Cc:* users(a)lists.kamailio.org <mailto:users@lists.kamailio.org> > *Subject:* Re: [Kamailio-Users] LCR module in Kamailio 3.0 > > Hello, > > On 1/26/10 7:33 AM, Denis Putyato wrote: > > Hello everybody! > > I have a such problem with LCR module > > When I add a new gw in gw table via kamctl addgw or direct via mysql > command, never mind, after doing lcr reload, in memory loaded only gw > with lcr_id = 1. > > All other gateways, with lcr_id > 1, not loaded into memory, so I > cannot use them. > > (After lcr reload I do lcr dump and only gateways whis lcr_id = 1 can > be seen) > > Thank you for any help. > > P.S. lcr table, where lcr_id is present too, is empty. It does not > need for processing in my kamailio. Only gw table is used. > > Kamailio 3.0 was installed from GIT. > > maybe is related to lcr_count parameter: > > http://kamailio.org/docs/modules/3.0.x/modules/lcr.html#id2495949 > > AFAIK, there were some changes in lcr by adding instances, which are > hold in lcr_id (gws to same instance have same id). > > Cheers, > Daniel > > > > > -- > Daniel-Constantin Mierla > *http://www.asipto.com/ > > > > > > -- > Daniel-Constantin Mierla > *http://www.asipto.com/ > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > > > > -- > Daniel-Constantin Mierla > *http://www.asipto.com/ > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > > > -- > Daniel-Constantin Mierla > *http://www.asipto.com/ -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] LCR module in Kamailio 3.0
by Denis Putyato 27 Jan '10

27 Jan '10

Hello everybody! I have a such problem with LCR module When I add a new gw in gw table via kamctl addgw or direct via mysql command, never mind, after doing lcr reload, in memory loaded only gw with lcr_id = 1. All other gateways, with lcr_id > 1, not loaded into memory, so I cannot use them. (After lcr reload I do lcr dump and only gateways whis lcr_id = 1 can be seen) Thank you for any help. P.S. lcr table, where lcr_id is present too, is empty. It does not need for processing in my kamailio. Only gw table is used. Kamailio 3.0 was installed from GIT.

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[Kamailio-Users] Rtpproxy transcoder patch
by Vikram Ragukumar 26 Jan '10

26 Jan '10

Hello, I came across a post by Stefan Sayer regarding an rtpproxy transcoder patch http://lists.iptel.org/pipermail/serdev/2008-July/012794.html. We are interested in having rtpproxy perform transcoding, both in software and using DSP array cards, and are currently running rtpproxy version 1.2.1. Is this link http://user.cs.tu-berlin.de/~sayer/transcoder/ that is mentioned on the post our starting point or is there an updated version ? Thanks and Regards, Vikram.

1 0

[Kamailio-Users] core:proxy.c problem
by marius zbihlei 26 Jan '10

26 Jan '10

Hello, There seems to be a design problem in proxy.c mk_proxy(str* name, unsigned short port, unsigned short proto, int is_sips) function. The purpose of this function is to create a proxy and return a pointer to the created structure. The issue arises from the fact that the name (type struct str) member of the proxy structure is not deep copied from the given parameters(refer to the str* name ) (the hostent structure is instead deep copied). This isn't a problem for now but I have worked on a small patch that caches proxies (using add_proxy() and find_proxy()) and ,because of this shallow copy, things are broken. I said that this is a design problem because we can let the shallow copy happen (performance is better), and when needed the caller should provide a buffer that doesn't change (let him do the copy instead). This is not clearly documented but done from /modules(_k)/utils/conf.c. But it this way we may have a memory leak when the proxy is deallocated, because I doubt that the caller keeps track of the allocated buffers(the code in proxy.c doesn't take ownership of the given pointer). This affects functions mk_proxy and mk_shm_proxy in both kamailio(1.5 to speak of) and sip-router. I have created a patch that also does a deep copy of the name, thus removing the need for the caller to bother about the lifetime of the name buffer. Any ideas?! Cheers Marius

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Re: [Kamailio-Users] ACC on Failure route
by alex pappas 26 Jan '10

26 Jan '10

> Hi Daniel, > > This is my ACC configuration: > > # ----- acc params ----- > /* what sepcial events should be accounted ? */ > modparam("acc", "early_media", 1) > modparam("acc", "report_ack", 1) > modparam("acc", "report_cancels", 1) > /* by default ww do not adjust the direct of the sequential requests. > if you enable this parameter, be sure the enable "append_fromtag" > in "rr" module */ > modparam("acc", "detect_direction", 0) > /* account triggers (flags) */ > *modparam("acc", "failed_transaction_flag", 3)* > modparam("acc", "log_flag", 1) > modparam("acc", "log_missed_flag", 2) > modparam("acc", "log_extra", > > "src_user=$fU;src_domain=$fd;dst_to_user=$tU;dst_user=$rU;dst_domain=$rd;diversion_uri=$di;destination_uri=$du;display_name=$fn;orig_src_domain=$od;orig_req_uri=$ou;orig_username=$oU;prefered_identity=$pd;display_name_pref_ident=$pn;proxy_ip=$Ri;proxy_port=$Rp;src_ip=$si;dst_dom_uri=$td;user_agent=$ua") > /* uncomment the following lines to enable DB accounting also */ > modparam("acc", "db_flag", 1) > modparam("acc", "db_missed_flag", 2) > modparam("acc", "db_url", > "mysql://openser:openserrw@localhost/openser10") > modparam("acc", "db_extra", > > "src_user=$fU;src_domain=$fd;dst_to_user=$tU;dst_user=$rU;dst_domain=$rd;diversion_uri=$di;destination_uri=$du;display_name=$fn;orig_src_domain=$od;orig_req_uri=$ou;orig_username=$oU;prefered_identity=$pd;display_name_pref_ident=$pn;proxy_ip=$Ri;proxy_port=$Rp;src_ip=$si;dst_dom_uri=$td;user_agent=$ua") > > I run kamailio-1.5.3 no TLS > > What exactly is iirc? > > When i call from an user to an other and the call is passing in case of > failure from failure route1 then I get an extra cdr from the failure route1 > BUT when I call pstn and the call goes to Failure route2 then I don't get > the extra CDRs. One important diference between Failure route1 and 2 is that > in the Failure route2 I have an extra *append_branch();* > > Failure Route 1 (default) > > failure_route[1] { > xlog("alx ------- Failure Route 1 -------"); > #n# if (is_method("INVITE") > #n# && (isbflagset(6) || isflagset(5))) { > #n# unforce_rtp_proxy(); > #n# } > > if (t_was_cancelled()) { > exit; > } > > # uncomment the following lines if you want to block client > # redirect based on 3xx replies. > ##if (t_check_status("3[0-9][0-9]")) { > ##t_reply("404","Not found"); > ## exit; > ##} > > # uncomment the following lines if you want to redirect the failed > # calls to a different new destination > ##if (t_check_status("486|408")) { > ## sethostport("192.168.2.100:5060"); > ## append_branch(); > ## # do not set the missed call flag again > ## t_relay(); > ##} > } > > Failure Route2 > > > failure_route[2] { > > if(t_was_cancelled()) { > exit; > } > > if(t_check_status("4[0-9][0-9]|5[0-9][0-9]")) > { > > #xlog("ACCOUNTING: > src_user=$fU;src_domain=$fd;dst_to_user=$tU;dst_user=$rU;dst_domain=$rd;diversion_uri=$di;destination_uri=$du;display_name=$fn;orig_src_domain=$od;orig_req_uri=$ou;orig_username=$oU;prefered_identity=$pd;display_name_pref_ident=$pn;proxy_ip=$Ri;proxy_port=$Rp;src_ip=$si;dst_dom_uri=$td;user_agent=$ua;sip_replay_reson=$rr;sip_replay_status=$T_reply_code"); > > if(ds_next_domain()) > { > xlog("alx ------------------------------------- The final RU is > $rU and the cust_prefix:$avp(s:cust_prefix) --------"); > xlog("alx ------- [ FAILURE ROUTE 2 ] First part of failure with > rd = $rd and prefix = $(ru{uri.param,prefix}) ---"); > # more destinations from same dispatcher group > > # add prefix if set and remove the param from R-URI > if($(ru{uri.param,prefix})!=null) > { > $ru = "sip:" + $avp(s:cust_prefix) + > $(ru{uri.param,prefix}) + $avp(s:user) + "@" + $rd; > } else { > $ru = "sip:" + $avp(s:cust_prefix) + $avp(s:user) + "@" + > $rd; > } > #rewriteport(5065); > #setport("5065"); > t_on_failure("2"); > *append_branch();*##################################################################### This > is different !!!!!!!!!!!!!!!!!! > t_relay(); > exit; > } else { > # go to next dispatcher group > xlog("alx ------- This is the next DST group = $avp(s:dstgrp) > -------"); > > > if($avp(s:dstgrp)!=null) > { > # select destination > if(ds_select_domain("$avp(s:dstgrp)", "4")) > { > xlog("alx ------- [ FAILURE ROUTE 2 ] Second part of > failure with rd = $rd and prefix = $(ru{uri.param,prefix}) ---"); > xlog("alx ------------------------------------- The > final RU is $rU and the cust_prefix:$avp(s:cust_prefix) --------"); > # add prefix if set and remove the param from R-URI > if($(ru{uri.param,prefix})!=null) > { > $ru = "sip:" + $avp(s:cust_prefix) + > $(ru{uri.param,prefix}) + $avp(s:user) + "@" + $rd; > } else { > $ru = "sip:" + $avp(s:cust_prefix) + $avp(s:user) > + "@" + $rd; > } > $avp(s:dstgrp) = null; > xlog("alx ------- This is the next DST group after > NULL = $avp(s:dstgrp) -------"); > xlog("alx ------- The final RURI is $ru ------- "); > t_on_failure("2"); > *append_branch();*##################################################################### > This is different !!!!!!!!!!!!!!!!!! > t_relay(); > exit; > } > } else { > t_reply("444", "No more tries for you!"); > } > } > } > > > > Thanks > Alex > > > > > On Mon, Jan 25, 2010 at 12:39 PM, Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > >> Hi Alex, >> >> >> On 1/25/10 11:03 AM, alex pappas wrote: >> >> Hi Daniel, >> >> We make a call and the call tries from the Failure Route 5 times new >> gateway (with dispatcher) but WITHOUT success in termination. In every try >> we get back 503 Congestion. In this case how many CDRs we should see in our >> LOG. >> >> We have the modparam("acc", "failed_transaction_flag", 6) and we >> setflag(6) for every INVITE. >> >> iirc, at some point was an option to get all failed legs accounted. I >> don't remember using it so I have to check the sources (you are on 1.5.x, >> right?). >> >> For next version I plan to make acc functions available on rest of the >> routes, should be few updates to take sip method from cseq header and ignore >> r-uri if it is a reply processed, in failure route should work by default, I >> see no reasons now what could be the problem. >> >> Cheers, >> Daniel >> >> >> On Fri, Jan 22, 2010 at 3:33 PM, Daniel-Constantin Mierla < >> miconda(a)gmail.com> wrote: >> >>> Hello, >>> >>> >>> On 1/22/10 1:08 PM, alex pappas wrote: >>> >>>> Dear Friends, >>>> >>>> In case of a call failure I'm trying from failure route to send the call >>>> through other gateways with Dispatcher module. >>>> I would like to have ACC for all failures that a call can possibly pass. >>>> If the call fails in the 3 first gateways I want them also in my mysql ACC >>>> table. I tried that with AVPs in the syslog and it works fine BUT is there >>>> any way that I can simply setflag(1) for accounting and I can have ACC for >>>> every try? >>>> >>>> I tryied to add in failure route : >>>> >>>> setflag(1); # do accounting ... >>>> setflag(3); # ... even if the transaction fails >>>> >>>> but I'm getting ACC only the last try. >>>> >>>> last failed try? Or what you mean by "the last try"? IIRC, failed >>> transaction flag should be set for each leg if you want all failures. >>> >>> Cheers, >>> Daniel >>> >>> -- >>> Daniel-Constantin Mierla >>> * http://www.asipto.com/ >>> >>> >> >> _______________________________________________ >> Kamailio (OpenSER) - Users mailing list >> Users@lists.kamailio.orghttp://lists.kamailio.org/cgi-bin/mailman/listinfo/usershttp://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> >> >> -- >> Daniel-Constantin Mierla >> * http://www.asipto.com/ >> >> >

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[Kamailio-Users] Manipulate Berkeley DB
by Santiago Soares 26 Jan '10

26 Jan '10

Hello, I'd like to use dispatcher module with Berkeley DB. How can I manipulate the dispatcher table, I mean, insert or delete gateways? kamctl seems not to work: [root@vnsiprtest ~]# kamctl dispatcher addgw 1 '1.2.3.4' 1 'test' /usr/local/sbin/kamctl: line 1458: insert into dispatcher ( setid, destination, flags, description ) VALUES (1,'1.2.3.4',1,'test');: command not found ERROR: dispatcher - SQL Error Thank you. Santiago

2 1

[Kamailio-Users] Proxy caching patch
by marius zbihlei 26 Jan '10

26 Jan '10

Hello, I have worked lately on the patch that tries to mitigate one performance problem in core:action.c file. I am talking about the multiple calls to mk_proxy(), and the creation of temporary proxies for each FORWARD_T action that the daemon executes. I think this behaviour can be improved by caching certain proxies. My idea is this: * If the next_hop matches a name in the alias list (for example we have an alias for example.com and the next_hop is proxy1.example.com), then cache the proxy by calling add_proxy instead of mk_proxy. If the proxy is already cached a pointer is returned, if not it is created and added (the functionality already exists in proxy.c) * For all other hosts a temporary proxy is made(same as now). This whole logic can be turned off(default)/on by a parameter in .cfg. I have run some tests and this can lead to big improvemets (because mk_proxy mean some pkg_malloc - fast but also some DNS resolving - slow). This might benefit some medium-large setups where you have some servers forwarding request to other application servers, etc. The patch is very small and it's already done and finished. I will provide it tomorrow if somebody thinks this idea is good Cheers Marius

1 0

[Kamailio-Users] 30x Redirect / Forward, even questions
by Samuel Muller 26 Jan '10

26 Jan '10

Hello all, I've questions 'bout the redirect/forward with 30x requests. Actually, it does not work, it seems the uac redirect module doesn' t use the new contact for the new invite. simple schema : pstn gw -> kamailio -> linksys -> call forward unconditionnal to another pstn number -> kamailio -> gw pstn the 302 as seen with ngrep tool : #SIP/2.0 302 Moved Temporarily. #To: <sip:B@sip_proxy;user=phone>;tag=91515579fff6b8d1i0. #From: <sip:A@pstn_gw1;user=phone>;tag=16627-UI-00020cf3-54eea92c4. #Call-ID: 16627-RO-00020cf2-4186e9762@pstn_gw1. #CSeq: 81990 INVITE. #Via: SIP/2.0/UDP sip_proxy;branch=z9hG4bK0f0c.22989b87.1. #Via: SIP/2.0/UDP pstn_gw1:5060;rport=5060;received=ip_pstn_gw1;branch=z9hG4bK-25EA-243F3. #Record-Route: <sip:sip_proxy;lr=on;ftag=16627-UI-00020cf3-54eea92c4;did=f65.bcc44f06>. #Contact: <sip:C@sip_proxy>. #Diversion: "B" <sip:B@sip_proxy>;reason=unconditional. #Server: Linksys/SPA942-6.1.3(a). DBG:uac_redirect:shmcontact2dset: adding contact <sip:C@sip_proxy> dest set : Contact: sip:B@ip_B, <sip:C@sip_proxy>;q=0.01 contact $ct: <sip:pstn_gw1:5060> the second pstn gateway received a request to B, and not C (so it creates a loop through pstn ...). in my config : loadmodule "uac.so" modparam("uac", "rr_store_param", "vsf") modparam("uac", "from_restore_mode", "auto") loadmodule "uac_redirect.so" modparam("uac_redirect", "default_filter", "accept") modparam("uac_redirect","bflags", 1) loadmodule "diversion.so" failure_route[2] { setdebug(4); if (t_check_status("301|302")) { route(26) # acc fwd $avp(s:acc_status) = "cfu"; if (!get_redirects("2:1")) { xlog("L_NOTICE", "failed to extract contact info from 30x header"); route(10); # stop rtp t_reply("480", "Temporarily Unavailable"); exit; } # flag 3xx status, open a new branch and back to find the new callee setflag(28); append_branch(); t_on_branch("1"); route(3); # search the new callee setdebug(); exit; } t_on_branch("1"); setdebug(); route(3); # find callee route(12); # local outbound route(10); # stop rtp } I'm using lcr module to route calls, kamailio v. 1.5.3. I've another question, is the 30x always considered as a failure_route ? Any help and/or explanations about the tricks would be really appreciated ... thanks, -- Samuel MULLER sml(a)l33.fr

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[Serusers] Questions about the command : serctl fifo
by inge 26 Jan '10

26 Jan '10

Hi, I am using SER 0.9.9. I have questions about two commands "serctl fifo" : - serctl fifo t_stats : What does mean the "current", "waiting" and "total" values returned by the command? - serctl fifo ul_stats : What does mean the two values returned on the line "location"? Regards, David.

1 0

[Kamailio-Users] Global constants
by Alex Balashov 26 Jan '10

26 Jan '10

Daniel, What is the recommended way (if any) to have a "global constant" that exists outside of any particular routes, in K3? My objective is to define a global DB_URL that can be used by all modules everywhere, but I can't just 'and' the modules together, e.g. modparam("mod1|mod2|mod3|...", "db_url", "...") because the modules are dynamically included (thanks to include_file!) and I do not know the list of all loaded modules easily at runtime. So instead, db_url is configured on a per-module basis. I thought preprocessor constants (e.g. #!define and friends) would allow me to set values for them, but they don't, so that's out. What other options are available that do not involve an event/local route, and operate instead in the completely global namespace? Right now, I am using a Perl script that manually substitutes the value of the db_url modparam everywhere in a large directory tree. :/ What am I not thinking of? Cheers, -- Alex -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/

2 1

[Kamailio-Users] Active dialogs waaaay to high
by Anders 26 Jan '10

26 Jan '10

Hi, When I run 'kamctl fifo get_statistics all calls', it tells me that the the number of dialogs is about 2000 times higher than what it really is. Actually it seems like it has added a couple of thousand to the number. How do I 'reset' this number? A restart of kamailio didn't help... Thanks! //Anders

2 1

[Kamailio-Users] "Permanent" AVP
by Antonio Reale 26 Jan '10

26 Jan '10

Hi all, I'd like to simulate in kamailio 1.4.4, the loading of a global setting (like a parameter that is loaded only at startup), using AVPs. So, I want to load some AVP only at proxy startup. Probably it is not possible because, if I'm not wrong, AVPs are transaction persistent. The problem is that I don't need to load those AVPs for every transaction. There is a why to implement such configuration? Thanks. Best regards. -- Antonio

3 5

[Kamailio-Users] 477 Send failed
by vragukumar＠signalogic.com 25 Jan '10

25 Jan '10

Hello, I have setup Kamailio and rtpproxy on a server, that acts as a SIP Proxy. An asterisk server sits behind this proxy server. I have noticed that on some occasions when the softphone(x-lite) tries to register with the asterisk through the proxy,it receives a 477 send failed. If i restart Kamailio, the problem seems to disappear. I have attached a logfile showing Kamailio's error output. What could be the reason for such behavior ? Thanks and Regards, Vikram.

3 2

[Serusers] how to compile the docs?
by Chump Chumpster 25 Jan '10

25 Jan '10

Hi, I'm sure this is a simple question and hopefully a simple answer. I've downloaded and am experimenting with SER 2.0.x. I can't find any instructions on how to compile the docs included with the source, into a usable format. They're pretty difficult to use as XML without a stylesheet :-) I appreciate any advice to get me going in the right direction....

1 0

[Kamailio-Users] Fwd: TLS Multiple records not read
by Jijo Jose 25 Jan '10

25 Jan '10

Thanks for the response. The SIP server is reading only the first record(first sip message), the _tls_read() function's log shows the length of first sip message which is matching with length of first record in wireshark.. This means SSL_read returns only the first record. I did the test using TCP which is working fine as it is reading two sip messages on a single read. Please find the log and attached print screen of wireshark traces. Jan 22 20:41:37 REGS-1a0240 openser[936]: io_wait_loop_sigio_rt: siginfo: signal=35 (35), si_code=1, si_band=0x41, si_fd=37 Jan 22 20:41:37 REGS-1a0240 openser[936]: TCPCONN: handle_io: fd map 0x1011d73c (37): {37, 2, 0x30294da0} Jan 22 20:41:37 REGS-1a0240 openser[936]: tls_update_fd: New fd is 37 Jan 22 20:41:37 REGS-1a0240 openser[936]: _tls_read: 559 bytes read Jan 22 20:41:37 REGS-1a0240 openser[936]: read= 559 bytes, parsed=559, state=4, error=1 Jan 22 20:41:37 REGS-1a0240 openser[936]: tcp_read_req: last char=0x0A, parsed msg= BYE sip:551138175007@21.21.81.11:6061;transport=tls SIP/2.0^M To: SIP5007_TLS_SIMPLEX <sip:551138175007@41.41.0.230<sip%3A551138175007(a)41.41.0.230>>;tag=91c2c894c0^M From: <sip:551138175008@41.41.0.230:5061;transport=tls>;tag=snl_G84KdX4MLT^M Call-ID: 0ad3e453326a4160^M CSeq: 1 BYE^M Route: <sip:21.21.27.10:5061;transport=tls;ftag=91c2c894c0;lr=on>^M Via: SIP/2.0/TLS 41.41.0.230:5061;branch=z9hG4bK_brancha_41.41.0.230_KonTU5DMPx^M Accept-Language: en;q=0.0^M Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER^M Date: Fri, 22 Jan 2010 20:41:36 GMT ^M Max-Forwards: 69^M Content-Length: 0^M ^M Jan 22 20:41:37 REGS-1a0240 openser[936]: tcp_read_req: end of header part Jan 22 20:41:37 REGS-1a0240 openser[936]: - received from: port 5061 Jan 22 20:41:37 REGS-1a0240 openser[936]: - received from: ip 41.41.0.230 Jan 22 20:41:37 REGS-1a0240 openser[936]: tcp_read_req: headers: BYE sip:551138175007@21.21.81.11:6061;transport=tls SIP/2.0^M To: SIP5007_TLS_SIMPLEX <sip:551138175007@41.41.0.230<sip%3A551138175007(a)41.41.0.230>>;tag=91c2c894c0^M From: <sip:551138175008@41.41.0.230:5061;transport=tls>;tag=snl_G84KdX4MLT^M Call-ID: 0ad3e453326a4160^M CSeq: 1 BYE^M Route: <sip:21.21.27.10:5061;transport=tls;ftag=91c2c894c0;lr=on>^M Via: SIP/2.0/TLS 41.41.0.230:5061;branch=z9hG4bK_brancha_41.41.0.230_KonTU5DMPx^M Accept-Language: en;q=0.0^M Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER^M Date: Fri, 22 Jan 2010 20:41:36 GMT ^M Max-Forwards: 69^M Content-Length: 0^M ^M . Jan 22 20:41:37 REGS-1a0240 openser[936]: tcp_read_req: content-length= 0 Jan 22 20:41:37 REGS-1a0240 openser[936]: tcp_read_req: body: Jan 22 20:41:37 REGS-1a0240 openser[936]: calling receive_msg(0x30294e14, 559, ) Jan 22 20:41:37 REGS-1a0240 openser[936]: SIP Request: Jan 22 20:41:37 REGS-1a0240 openser[936]: method: <BYE> Jan 22 20:41:37 REGS-1a0240 openser[936]: uri: <sip:551138175007@21.21.81.11:6061;transport=tls> Jan 22 20:41:37 REGS-1a0240 openser[936]: version: <SIP/2.0> Thanks Jijo On Sun, Jan 24, 2010 at 5:53 AM, Klaus Darilion < klaus.mailinglists(a)pernau.at> wrote: > I would verify if the received fragment really contains 2 complete SIP > messages (e.g. 2xCRLF at the end of the SIP headers and if Content-Length > header is correct). > > For debugging, TLS is PITA. > > Maybe the SIP server shows the same behavior when using TCP. Otherwise you > can try to configure the NULL:CIPHER on both servers - then you see the > plaintext SIP message in the TLS packets. > > regards > klaus > > Jijo Jose wrote: > >> Hi All, >> We have a SIP server which enabled NAGLE algorithm and proxy as openser >> ver 1.1 >> SIP Server send a TLS multiple records( 2 SIP messages) in a packet to >> openser. Openser is reading only the first record( first SIP message). The >> second TLS record(sip message) is read only when the next message recieved >> by openser from the SIP Server. >> I would like to know anybody observed this issue. I have compared the >> codebase of kamailio 1.4 and openser 1.1 but didin't find any diffrence for >> TLS. >> I have looked at the API _tls_read() in openser which is same as that of >> kamilio. >> Don't we need to use SSL_pending() after SSL_read() to verify any buffer >> is left in the SSL layer ? >> Do you think is it due to the poll method error? The poll method we use >> is POLL_SIGIO_RT >> Please let me know your comments. >> Thanks in advance. >> Jijo >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Kamailio (OpenSER) - Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> > >

2 4

[Kamailio-Users] load drouting
by Denis Putyato 25 Jan '10

25 Jan '10

Hello everybody! Such error happens when I try start kamailio Jan 25 10:52:57 ubuntutest kamailio: ERROR: <core> [sr_module.c:390]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/drouting.so>: /usr/local/lib/kamailio/modules_k/drouting.so: undefined symbol: db_bind_mod Jan 25 10:52:57 ubuntutest kamailio: : <core> [cfg.y:3326]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 42, column 12-24: failed to load module ls /usr/local/lib/kamailio/modules_k/ acc.so db_text.so group.so nathelper.so pv.so sl.so uac_redirect.so alias_db.so dialog.so htable.so nat_traversal.so qos.so sms.so uac.so auth_db.so dispatcher.so imc.so path.so ratelimit.so speeddial.so uri_db.so auth_diameter.so diversion.so kex.so pdt.so registrar.so sqlops.so userblacklist.so auth.so domainpolicy.so maxfwd.so permissions.so rr.so sst.so usrloc.so benchmark.so domain.so mi_datagram.so pike.so rtimer.so statistics.so xlog.so call_control.so drouting.so mi_fifo.so presence_mwi.so siptrace.so textops.so cfgutils.so exec.so msilo.so pua_mi.so siputils.so tmx.so In config file: loadpath "/usr/local/lib/kamailio/modules:/usr/local/lib/kamailio/modules_k" loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "avpops.so" loadmodule "siptrace.so" loadmodule "dialplan.so" loadmodule "lcr.so" loadmodule "sqlops.so" loadmodule "carrierroute.so" loadmodule "drouting.so" Kamailio ver 3.0 was installed from GIT. Thank you for any help

2 4

[Kamailio-Users] Outgoing registrations
by Andrey Utkin 25 Jan '10

25 Jan '10

How do you make outgoing registrations to sip providers' accounts? I haven't found Kamailio functionality like this, so looks this requires some b2bua usage? Is there some other way?

3 6

[Kamailio-Users] Option ac_extra_hdrs to tm in K 3.0.x
by Alex Balashov 25 Jan '10

25 Jan '10

What exactly is the appropriate syntax for the value of the ac_extra_hdrs option in K3's 'tm'? As described here, it is not very clear: http://www.kamailio.org/docs/modules/3.0.x/modules/tm.html#ac_extra_hdrs modparam("tm", "ac_extra_hdrs", "myfavoriteheaders-") What does this mean? Where do I put the header name, and what is the delimeter? Is it a semicolon or colon-separated list? What if, for example, I want to pass X-Alex-Status and X-Alex-Cause in CANCELs and negative ACKs from the original INVITE? Do I use ... modparam("tm", "ac_extra_hdrs", "X-Alex_Status;X-Alex-Cause") If so, what is the significance of the '-' in the above example? Thanks! -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/

2 3

[Kamailio-Users] ACC on Failure route
by alex pappas 25 Jan '10

25 Jan '10

Dear Friends, In case of a call failure I'm trying from failure route to send the call through other gateways with Dispatcher module. I would like to have ACC for all failures that a call can possibly pass. If the call fails in the 3 first gateways I want them also in my mysql ACC table. I tried that with AVPs in the syslog and it works fine BUT is there any way that I can simply setflag(1) for accounting and I can have ACC for every try? I tryied to add in failure route : setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails but I'm getting ACC only the last try. Thanks Alex

3 11

[Kamailio-Users] TLS Multiple records not read
by Jijo Jose 24 Jan '10

24 Jan '10

Hi All, We have a SIP server which enabled NAGLE algorithm and proxy as openser ver 1.1 SIP Server send a TLS multiple records( 2 SIP messages) in a packet to openser. Openser is reading only the first record( first SIP message). The second TLS record(sip message) is read only when the next message recieved by openser from the SIP Server. I would like to know anybody observed this issue. I have compared the codebase of kamailio 1.4 and openser 1.1 but didin't find any diffrence for TLS. I have looked at the API _tls_read() in openser which is same as that of kamilio. Don't we need to use SSL_pending() after SSL_read() to verify any buffer is left in the SSL layer ? Do you think is it due to the poll method error? The poll method we use is POLL_SIGIO_RT Please let me know your comments. Thanks in advance. Jijo

2 1

[Kamailio-Users] state field on dialog table
by Panagiotis Skoulikaritis 23 Jan '10

23 Jan '10

Hello I have a question about the state field on the dialog table. I guess it is more sip protocol oriented than kamailio deployment, I assume that the values (1-5) that are logged are the state of the dialog correct ? and if the above are correct the dialog states are the following ? 1. waiting for UAC deciscion 2. Initiate dialog 3. Establish dialog 4. Stop Media 5. Destroy dialog I apologize for my sip ignorance Trying to build up my knowledge. :) I' m asking this because I' m trying to export statistics about the calls that are handled from the kamailio and the state of the call for specific users,IP's, destinations. Thanks Panagiotis

4 5

[Kamailio-Users] mysql loadbalancing
by BERGANZ François 22 Jan '10

22 Jan '10

Hello, I need to know if Kamailio can change Mysql Server if there is a problem with a mysql server ? Can Kamailio do loadbalancing between 2 mysql servers which are duplicate ? Thank you François BERGANZ

5 17

TLS problems
by Andreas Rehbein 22 Jan '10

22 Jan '10

Hi, I've installed Kamailio 3.0 on RHEL5 and activated the mysql backend and tls support. Everything works fine with Soft User Agent "Phoner lite". My Problem is: If I try to register a "snom 370" Kamailio crashes immediatly. What is wrong? Any suggestions? Debug: 17(6290) DEBUG: <core> [ip_addr.c:116]: tcpconn_new: new tcp connection: 192.168.0.222 17(6290) DEBUG: <core> [tcp_main.c:1052]: tcpconn_new: on port 1430, type 3 17(6290) DEBUG: <core> [tcp_main.c:1351]: tcpconn_add: hashes: 978:1690:1400, 1 17(6290) DEBUG: <core> [io_wait.h:361]: DBG: io_watch_add(0x8222fe0, 31, 2, 0xb613a2b8), fd_no=24 17(6290) DEBUG: <core> [io_wait.h:588]: DBG: io_watch_del (0x8222fe0, 31, -1, 0x0) fd_no=25 called 17(6290) DEBUG: <core> [tcp_main.c:3627]: tcp: DBG: sendig to child, events 1 17(6290) DEBUG: <core> [tcp_main.c:3336]: send2child: to tcp child 0 13(6283), 0xb613a2b8 13(6283) DEBUG: <core> [tcp_read.c:884]: received n=4 con=0xb613a2b8, fd=8 13(6283) DEBUG: tls [tls_server.c:109]: Using TLS domain TLSs<default> 17(6290) DEBUG: <core> [tcp_main.c:2826]: DBG: handle_tcp_child: dead tcp child 0 (pid 6283, no 13) (shutting down?) 17(6290) DEBUG: <core> [io_wait.h:588]: DBG: io_watch_del (0x8222fe0, 24, -1, 0x0) fd_no=24 called 17(6290) : <core> [pass_fd.c:283]: ERROR: receive_fd: EOF on 4 17(6290) DEBUG: <core> [tcp_main.c:3038]: DBG: handle_ser_child: dead child 13, pid 6283 (shutting down?) 17(6290) DEBUG: <core> [io_wait.h:588]: DBG: io_watch_del (0x8222fe0, 4, -1, 0x0) fd_no=23 called 0(6257) ALERT: <core> [main.c:722]: child process 6283 exited by a signal 11 0(6257) ALERT: <core> [main.c:725]: core was generated 0(6257) INFO: <core> [main.c:737]: INFO: terminating due to SIGCHLD 17(6290) INFO: <core> [main.c:788]: INFO: signal 15 received 15(6286) INFO: <core> [main.c:788]: INFO: signal 15 received 16(6289) INFO: <core> [main.c:788]: INFO: signal 15 received 12(6281) INFO: <core> [main.c:788]: INFO: signal 15 received 14(6285) INFO: <core> [main.c:788]: INFO: signal 15 received 10(6278) INFO: <core> [main.c:788]: INFO: signal 15 received 6(6269) INFO: <core> [main.c:788]: INFO: signal 15 received 2(6261) INFO: <core> [main.c:788]: INFO: signal 15 received 11(6280) INFO: <core> [main.c:788]: INFO: signal 15 received 1(6260) INFO: <core> [main.c:788]: INFO: signal 15 received 9(6276) INFO: <core> [main.c:788]: INFO: signal 15 received 8(6272) INFO: <core> [main.c:788]: INFO: signal 15 received 3(6262) INFO: <core> [main.c:788]: INFO: signal 15 received 7(6270) INFO: <core> [main.c:788]: INFO: signal 15 received 5(6267) INFO: <core> [main.c:788]: INFO: signal 15 received 4(6265) INFO: <core> [main.c:788]: INFO: signal 15 received 0(6257) DEBUG: tm [t_funcs.c:122]: DEBUG: tm_shutdown : start 0(6257) DEBUG: tm [t_funcs.c:125]: DEBUG: tm_shutdown : emptying hash table 0(6257) DEBUG: tm [t_funcs.c:127]: DEBUG: tm_shutdown : removing semaphores 0(6257) DEBUG: tm [t_funcs.c:129]: DEBUG: tm_shutdown : destroying tmcb lists 0(6257) DEBUG: tm [t_funcs.c:132]: DEBUG: tm_shutdown : done 0(6257) DEBUG: tls [tls_init.c:621]: tls module final tls destroy 0(6257) DEBUG: <core> [mem/shm_mem.c:236]: shm_mem_destroy 0(6257) DEBUG: <core> [mem/shm_mem.c:239]: destroying the shared memory lock 0(6257) DEBUG: <core> [main.c:741]: terminating due to SIGCHLD tia

2 4

[Kamailio-Users] Asterisk 403 Forbidden error with port translation
by Vikram Ragukumar 21 Jan '10

21 Jan '10

Hello, I have made some progress since my previous post, but not enough :). ------------- -------- --- -------- |Sip Softphone|-------|Internet|--------|F.W|-----|Asterisk| ------------- -------- --- -------- IP addresses: a.b.c.d q.w.e.r The SIP softphone(x-lite) is configured to register with the asterisk server through port 9090 (Domain q.w.e.r:9090).Firewall(F.W) is setup as the outbound proxy for the softphone(Outbound proxy a.b.c.d:9090). Authentication credentials for the softphone match the user registered in asterisk's sip.conf. F.W runs Kamailio and rtpproxy, with Kamailio listening on port 5060. The asterisk server is setup to listen on port 5060. The Firewall(F.W), uses a libnetfilter_queue based program to : (a) Rewrite the destination port 9090 as 5060, and rewrite all other occurrences of 9090 as 5060 in the SIP message, for packets from the softphone to the asterisk server. (b) Rewrite the source port 5060 as 9090, and rewrite all other occurrences of 5060 as 9090 in the SIP message, for packets from the asterisk server to the softphone. The following exchange of SIP messages take place -Sip softphone sends a REGISTER message to asterisk -Asterisk responds with a 401 UNAUTHORIZED -Sip softphone replies with a REGISTER message containing auth. info. -Asterisk responds with a 403 FORBIDDEN : BAD AUTHORIZATION The above setup works when the softphone uses port 5060, so there problem here does not have anything to do with Authorization credentials. Is it possible i might be modifying parts of the packet that shouldn't be modified or i might not be modifying some relevant parts of the packet ? Thanks in advance, Vikram.

3 3

[Serusers] Hello , Help, and Thanks
by Justin 21 Jan '10

21 Jan '10

Hi, I have just begun to set up my soft phone, and am currently reading through your How to's Ser text's ,,, but I keep getting error 401-403-404-408 all doing with I am entering the wrong DOMAIN and PROXY for the Iptel.org SIP service. If you could contact me with a solution. Much Thanks (dark4ajaguar(a)iptel.org)

1 0

[Kamailio-Users] Script variables Question
by alex pappas 21 Jan '10

21 Jan '10

Dear friends, When a sip 4rU is entering my kamailio script I'm cutting down the prefix of this customer and added to a script variable. In the same same route after i do some checks then i add back the customer prefix which I had kept in the script variable. The problem: Some times the call goes to failure Route for rerouting and then i try to add the prefix again but the is empty. How can I transfer a variable from route to route and be a unique variable for each call? Thank you Alex

5 5

[Kamailio-Users] concurrent dialogs
by alex pappas 21 Jan '10

21 Jan '10

Dear Friends, I have the following question: When I'm running the command *kamctl fifo dlg_list* I'm getting the concurrent dialogs in that moment. When I disable the sip_trace module(which I don't see how they are related) I get NO output when I run the same command. Does anyone knows on why this is happening ? Alex

2 2

[Kamailio-Users] kamailio and sip-router at FOSDEM 2010 - social event
by Henning Westerholt 21 Jan '10

21 Jan '10

Hello, in about two and a half week the annual open source developer conference FOSDEM (http://fosdem.org/2010/) takes place in Brussels. As in the last year we would like to meet here for a social event, probably a dinner on Saturday evening, 6th February. This developers already confirmed their participance: - Daniel-Constantin Mierla - he is also giving a talk about Kamailio 3.0.0 on Saturday: http://fosdem.org/2010/schedule/events/kamailio_sip_server - Elena-Ramona Modroiu - Marius Zbihlei - 1&1 kamailio and sip-router development Bucharest - Timo Reimann - 1&1 kamailio and sip-router development Karlsruhe - Henning Westerholt Participation at the event is free of charge, everybody pays for its own expenses. In order to organize a restaurant and do a reservation please send me a short mail if you would like to attend. Suggestions for a nice place with good food and beer are also welcome. Looking forward to see you in Brussels. Best regards, Henning P.S.: Photos of the last year event can be found on our web site: http://www.kamailio.org/events/2009-FOSDEM/FOSDEM2009-Album/ -- Henning Westerholt - Development Consumer Products / Consumer Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

4 6

[Kamailio-Users] Question about Timer, i need milliseconds
by Marco Bungalski 20 Jan '10

20 Jan '10

Hello all :-) I have a question about how to get a time-stamps with milliseconds. To create my own CDR's, i write to my databases timestamps, with a command like this: sql_query("ca", "UPDATE active_calls SET state = 'BILLING', end_time=$Ts WHERE call_id='$ci'", "ra"); I use the "$Ts"-variable, but with this i can get only "full" seconds. But i need milliseconds for my billing-system. Otherwise i have differences between my CDR's and my carrier's one. Example: A call has a duration of 0.2 seconds. In my CDR start- and endtime is the same (=0 seconds), to my carrier a have 1 second to pay ;-) Is there a possibility to get a time-stamp or time with milliseconds? Cheers, Marco Marco Bungalski GmbH Traversale 5 27283 Verden Telefon: +49 4231 - 776 9999 Fax: +49 4231 - 776 9998 Mobil: +49 172 4204774 e-mail: Marco(a)Bungalski.de web: www.bungalski.de Geschäftsführender Gesellschafter: Marco Bungalski Sitz der Gesellschaft: D-27283 Verden, AG Walsrode HRB 120586

3 2

[Kamailio-Users] 3G+ and LTE forum - meeting in Berlin
by Daniel-Constantin Mierla 20 Jan '10

20 Jan '10

Hello, I will be in Berlin for few days (Jan 20-22), attending 3G+ and LTE Forum. If anyone is around and want to meet, feel free to drop me an email. Next event I will talk about Kamailio and SIP Router is FOSDEM in Brussels -- more details will follow soon. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Kamailio-Users] BYE packet is not received in siptrace module !!
by Dinesh Gautam 20 Jan '10

20 Jan '10

Dear all, I am using siptrace module and getting all the packets except the BYE. when the call disconnected BYE packet is not received. I am also getting the packets OK, REGISTER, INVITE two times in sip_trace table. Can anyone help me to sort out this problem?? -- Dinesh Gautam !!

4 14

[Serusers] deactivate the Via field
by "Martínez García, Rebeca" 20 Jan '10

20 Jan '10

Hello all, I know it might sound strange but, is there any way to deactivate that SER puts its Via on the SIP message when forwarding it to other entity? Thanks in advance, Rebeca Martinez

1 0

Re: [Kamailio-Users] [kamailio-users] Implement kamailio sever failover
by Klaus Darilion 20 Jan '10

20 Jan '10

ram schrieb: > in that case we are going to loose registration right ? not if you do database replication of "location" table > and ongoing calls will be dropped right ? No. only ongoing transactions may be affected. regards klaus PS: please always Cc the mailing list > > Ram > > > > On Sat, Jan 16, 2010 at 2:58 AM, Klaus Darilion > <klaus.mailinglists(a)pernau.at <mailto:klaus.mailinglists@pernau.at>> wrote: > > Hi! > > For failover there are several possibilities. The simplest on, I > guess the one you are referring to, is IP takeover. > > This is: 2 servers. On both is Kamailio and the database (e.g. > mysql) installed. The SIP service uses a virtual IP address which is > either configured at server 1 or server 2 (never on both). Kamailio > process is only running on the server which has the virtual IP > address provisioned. > > Then you need some other software to monitor the SIP service and > trigger the failover from one server to the other server (e.g. > either write some shell scripts or use existing software like > hearbeat, ultramonkey, ...) > > Further - depending on your scenario - you might need some database > replication (e.g. mysql cluster or mysql with master-master > replication). > > regards > klaus > > RAJNIKANT VANZA wrote: > > Hi, > > I want to implement kamailio server failover. > so, anybody know about how to configure kamailio server failover? > > e.g. 2 kamailio severs run and when any one server fail then > other server take over thats resources. > > > -- > Best Regards, > > Rajnikant Vanza > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > >

3 2

[Kamailio-Users] no answer to invite
by Anders 19 Jan '10

19 Jan '10

Hi, I have a problem that I don't understand at all (yet another one ;-)). One client has Asterisk (like many others) from where he sends calls to our Kamailio. It worked fine, until it suddenly stopped for no apparent reason on Friday. No changes in configuration on his or our side. We can ping his server just fine, and vice versa. On any INVITE he sends, he gets no, zero, zip response from Kamailio on our server. I wanted to figure out why, so I put an 'ngrep' on port 5060 for his IP on our Kamailio server, and I got...nothing. No communication at all on this port containing this IP. Why? I have no clue. The ping should show that we can communicate, and a 'ngrep' on his server shows that he sends plenty of INVITEs, but gets absolute zero response from our Kamailio box. Any ideas where to look? Thanks a lot in advance!

3 3

[Kamailio-Users] siremis on sourceforge.net
by Elena-Ramona Modroiu 19 Jan '10

19 Jan '10

Hi, the sources of siremis are in sourceforge.net svn. If any of you has developed (plan to develop) extensions, contact me to grant devel access. Also, patches should be sent against latest svn trunk (no branches so far): http://sourceforge.net/projects/siremis/ The svn repo does not store the openbiz core framework -- you have to get it from siremis-1.0.0.tgz (the openbiz dir) or download it from openbiz project and apply the patch from misc directory (note that you also have to download zend and others according to openbiz guides). Once you fetch siremis svn and add openbiz, run make to create the symlinks required for web dirs and you are set. Regards, Ramona

1 0

[Kamailio-Users] SIREMIS v1.0.0 Released
by Elena-Ramona Modroiu 18 Jan '10

18 Jan '10

Hi, I have just released SIREMIS 1.0.0 - web admin interface for Kamailio (OpenSER) 3.0.0 and SIP Router. Among new features: * presence service management – watchers, active watchers and presentity * sip trace view – access sip_trace module records, highlight headers and group by call-id * user location statistics charts – online user agents, supported SIP methods, contacts and nat related charts * update of the core components to latest stable versions * update to work with latest Kamailio (OpenSER) v3.0.0 * patch to work with older Kamailio (OpenSER) v1.5.x * time-based X-axis for charts print labels as HH:MM * charts types supported: line, line dot and area Some screenshots specific for this version: http://www.asipto.com/gallery/v/siremis/siremis_22.png.html http://www.asipto.com/gallery/v/siremis/siremis_23.png.html http://www.asipto.com/gallery/v/siremis/siremis_24.png.html Download and installation steps: http://siremis.asipto.com/install/ More screenshots: http://www.asipto.com/gallery/v/siremis/ Demo site (it works on a database with random data, username: admin, password: admin): http://siremis.asipto.com/demo/ Regards, Ramona

1 0

[Kamailio-Users] setbflag not working?
by Ricardo Martinez 18 Jan '10

18 Jan '10

Hello. I have the next situation with kamailio 3.0.0 In my config file I have something like this : route If INVITE then route(AUTH_REQUEST) . . . route[AUTH_REQUEST] . . . if (client_nat_test("3")) { append_hf("P-hint: route(AUTH_REQUEST)|setflag7,forcerport,fix_contact\r\n"); if (setbflag("7")) { xlog("L_INFO","[$ci]:[AUTH_REQUEST] : setbflag(7)"); } force_rport(); fix_contact(); xlog("L_INFO","[$ci]:[AUTH_REQUEST] : CLIENT_NAT_TEST-3"); }; . . . route(IN_TO_IN) route[IN_TO_IN] . . . route(LOCAL_CALL) . . . route[LOCAL_CALL] . . . if (isbflagset("7") ) { xlog("L_INFO","[$ci]:[LOCAL_CALL] : flag 7 SET begin"); } else { xlog("L_INFO","[$ci]:[LOCAL_CALL] : flag 7 NO SET begin"); } if ( !lookup("location") ) { if ( avp_check("$avp(s:4)", "eq/s:1/g") ) { if( !avp_pushto("$br", "$avp(s:5)/g") ) { sl_send_reply("403","Forbidden - Forward Number NULL"); exit; }; prefix("222"); route(TO_VOICEMAIL_NODISP); exit; }; if ( avp_check("$avp(s:3)", "eq/s:1/g") ) { xlog("L_INFO", "[$ci]:[LOCAL_CALL]: $rm - From:$fu ; To:$ru ; Call-ID:$ci ; Desde:$si - VOICEMAIL ON\n"); route(TO_VOICEMAIL_NODISP); exit; } else { sl_send_reply("480", "Usuario fuera de Servicio"); xlog("L_INFO", "[$ci]:[LOCAL_CALL]: $rm - From:$fu ; To:$ru ; Call-ID:$ci ; Desde:$si - VOICEMAIL OFF\n"); exit; }; }; if (isbflagset("7") ) { xlog("L_INFO","[$ci]:[LOCAL_CALL] : flag 7 SET end"); } else { xlog("L_INFO","[$ci]:[LOCAL_CALL] : flag 7 NOT SET end"); } . . . For some reason in the route[LOCAL_CALL] the flagb "7" is changed from "SET" to "NOT SET". According to my debugs this unset is made in the "lookup(location)" part. But I don't understand why. Can someone help me here? Thanks in advance, Ricardo Martinez.-

2 2

[Serusers] SER 0.9.7-pre4 and RPID
by Steven C. Blair 18 Jan '10

18 Jan '10

I'm trying to append an RPID header on my SER 0.9.7-pre4 system. The config has not changed and this use to work in SER 0.9.7-pre3. The header does not get appended and I cannot see why. The use_rpid parameter or the auth_db module does not seem to exist in this release. Is there a different way to append the header? Thanks,Steve

1 0

[Kamailio-Users] automatic packaging building of master
by Henning Westerholt 18 Jan '10

18 Jan '10

Hi, we've updated the automatic packaging building process on the devel machine. This builds now every night from the git master packages for debian lenny, squeeze and sid. Hints about your archive configuration in order to use the packages can be found on http://devel.kamailio.org/ They are build from the control files in pkg/debian, so at the moment some packages with special dependencies are missing, and they are named with 'ser-$name'. The only thing that is modified at the moment is the version number. This packages are of course only for tests and not for production use. Please let me know if there are any problems with the building service. Henning

1 0

[Kamailio-Users] %23 remove from rU
by alex pappas 18 Jan '10

18 Jan '10

Dear Friends, Do you have any idea on how to remove from $rU the %23 when is send in to my network? The scenario is: The customer will send (prefix # number). This is translated in sip in (prefix %23 number). I need to remove this in order to check against PDT module. Thanks Alex

4 12

[Kamailio-Users] Kamailio 3.0 Crash when load with module siptrace.
by Quang Tri 18 Jan '10

18 Jan '10

Hello, When i try with module siptrace.so This is loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "dialog.so" #!ifdef WITH_CALLCONTROL loadmodule "call_control.so" #!endif #!ifdef WITH_MEDIAPROXY loadmodule "mediaproxy.so" #!endif #!ifdef WITH_SIPTRACE loadmodule "siptrace.so" #!endif .... and with # ----- sip trace ------- #!ifdef WITH_SIPTRACE modparam("siptrace", "traced_user_avp", "$avp(s:traced_user)") modparam("siptrace", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_flag", 2) #!endif and this is error log 8(7163) DEBUG: tm [t_lookup.c:764]: DEBUG: t_lookupOriginalT: searching on hash entry 60285 8(7163) DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed 8(7163) DEBUG: tm [t_lookup.c:854]: DEBUG: t_lookupOriginalT: no CANCEL matching found! 8(7163) DEBUG: tm [t_lookup.c:856]: DEBUG: t_lookupOriginalT completed 8(7163) DEBUG: tm [tm.c:952]: lookup_original: t_lookupOriginalT returned: (nil) 8(7163) DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) 8(7163) DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) 8(7163) DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) 8(7163) DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) 8(7163) DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) 8(7163) DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) 8(7163) DEBUG: <core> [receive.c:280]: receive_msg: cleaning up *17(7186) : <core> [pass_fd.c:283]: ERROR: receive_fd: EOF on 14* 17(7186) DEBUG: <core> [tcp_main.c:3038]: DBG: handle_ser_child: dead child 5, pid 7160 (shutting down?) 17(7186) DEBUG: <core> [io_wait.h:588]: DBG: io_watch_del (0x81ede80, 14, -1, 0x0) fd_no=22 called 0(7151) ALERT: <core> [main.c:722]: child process 7160 exited by a signal 11 0(7151) ALERT: <core> [main.c:725]: core was generated 0(7151) INFO: <core> [main.c:737]: INFO: terminating due to SIGCHLD 6(7161) INFO: <core> [main.c:788]: INFO: signal 15 received 7(7162) INFO: <core> [main.c:788]: INFO: signal 15 received 16(7185) INFO: <core> [main.c:788]: INFO: signal 15 received 15(7183) INFO: <core> [main.c:788]: INFO: signal 15 received 8(7163) INFO: <core> [main.c:788]: INFO: signal 15 received 14(7179) INFO: <core> [main.c:788]: INFO: signal 15 received 9(7164) INFO: <core> [main.c:788]: INFO: signal 15 received 10(7165) INFO: <core> [main.c:788]: INFO: signal 15 received 13(7176) INFO: <core> [main.c:788]: INFO: signal 15 received 1(7155) INFO: <core> [main.c:788]: INFO: signal 15 received 17(7186) INFO: <core> [main.c:788]: INFO: signal 15 received 2(7156) INFO: <core> [main.c:788]: INFO: signal 15 received 3(7158) INFO: <core> [main.c:788]: INFO: signal 15 received 4(7159) INFO: <core> [main.c:788]: INFO: signal 15 received 11(7171) INFO: <core> [main.c:788]: INFO: signal 15 received 12(7174) INFO: <core> [main.c:788]: INFO: signal 15 received 0(7151) DEBUG: tm [t_funcs.c:122]: DEBUG: tm_shutdown : start 0(7151) DEBUG: tm [t_funcs.c:125]: DEBUG: tm_shutdown : emptying hash table 0(7151) DEBUG: tm [t_funcs.c:127]: DEBUG: tm_shutdown : removing semaphores 0(7151) DEBUG: tm [t_funcs.c:129]: DEBUG: tm_shutdown : destroying tmcb lists 0(7151) DEBUG: tm [t_funcs.c:132]: DEBUG: tm_shutdown : done 0(7151) DEBUG: <core> [mem/shm_mem.c:236]: shm_mem_destroy 0(7151) DEBUG: <core> [mem/shm_mem.c:239]: destroying the shared memory lock 0(7151) DEBUG: <core> [main.c:741]: terminating due to SIGCHLD How can fix it. Thank you.

3 6

[Kamailio-Users] Start/Stop Kamailion
by juergen＠glowka.de 18 Jan '10

18 Jan '10

Hi Dinesh do as Daniel already told You, or use the start/stop-script kamailion.init in the folder debian/ after unpacking the sourcepacket and change it according to Your system. Copy/rename it to /etc/init.d/kamailion check for errors: /usr/loca/sbin/kamailio -c Juergen (who likes india:)

2 1

add_contact_alias
by Klaus Darilion 17 Jan '10

17 Jan '10

Hi Juha! Does the alias functions also support IPv6 addresses? regards klaus

1 0

[Kamailio-Users] [kamailio-users] Implement kamailio sever failover
by RAJNIKANT VANZA 16 Jan '10

16 Jan '10

Hi, I want to implement kamailio server failover. so, anybody know about how to configure kamailio server failover? e.g. 2 kamailio severs run and when any one server fail then other server take over thats resources. -- Best Regards, Rajnikant Vanza

2 1

[Kamailio-Users] How to enable all sip_trace messages?
by Dinesh gautam 16 Jan '10

16 Jan '10

Dear all, I have enabled the sip_trace module and able to log the messages ACK, INVITE, REGISTER , SUBSCRIBE. I want to log other messages like RING, OK, BYE etc. Please help me out how can I enable theses messages. -- With Regards, Dinesh Kumar Gautam

1 0

Re: [Serusers] configure SER to work as a gateway
by "Martínez García, Rebeca" 15 Jan '10

15 Jan '10

Authentication in my upstream server is a MUST in my case and I cannot disable it, sorry. In fact I would be going against the standards for what I am configuring if I disabled authentication. I only wanted SER for the special case of internal calls inside a private LAN, but I am realising it is not as simple as I though it would be. In any case, thanks for the support. Regards, Rebeca Martínez -----Mensaje original----- De: rupert.organ(a)bt.com [mailto:rupert.organ@bt.com] Enviado el: vie 15/01/2010 12:06 Para: Martínez García, Rebeca Asunto: RE: [Serusers] configure SER to work as a gateway Not that I know of... Anyway, we had the same problem....but solved it by disabling authentication on the upstream server. After all...you are now in a "trusted zone"....you have successfully "auth challenged" on the SER...so no need to do it twice....SER "IS" the Registrar, thats what it is designed to be....an architecture with 2 Registrars is not good. If you can turn off auth on the upstream server then this is best. When you t_replicate Registers SER just ignores the 200ok back from the upstream server....after all it is just sending the register as a courtesy gesture. Rupert ________________________________ From: "Martínez García, Rebeca" [mailto:remartinez@indra.es] Sent: 15 January 2010 10:54 To: Organ,RJE,Rupert,DRK R Cc: serusers(a)lists.iptel.org; "Huertas García, Victor" Subject: RE: [Serusers] configure SER to work as a gateway Hi all, Rupert you were right, SER does not permit to save the location based on the reg of a different upstream server; which means that I have to make "save_contacts("location") when SER receives the REGISTER. However, as it then sends directly to my client a 200 OK, when the client receives the "401 Unauthorized" error from my server, it does not authenticate with it because it has received a previous 200 OK (from SER...). So, here is my question: Is it possible to make SER not send a 200 OK when saving the location of the user? (I am only asking because I do not know). Thanks in advance, Rebeca Martinez -----Mensaje original----- De: rupert.organ(a)bt.com [mailto:rupert.organ@bt.com] Enviado el: lun 11/01/2010 12:37 Para: Martínez García, Rebeca Asunto: RE: [Serusers] configure SER to work as a gateway Rebeca, i am not an expert, but if I had to guess...then i would guess it would be because there is no contacts in the 200 to a Register. In our ser.cfg we save("location"); after testing if (method=="REGISTER") { but I can see that you are making registration and saving of info into location dependent on successful reg on a different upstream server. I do not think that is possible because you need to make the decision to save into location based on the reg that SER has received...and not based on the response of an upstream element....but if I am wrong I would be interested in the answer... Rupert ________________________________ From: "Martínez García, Rebeca" [mailto:remartinez@indra.es] Sent: 11 January 2010 11:21 To: Organ,RJE,Rupert,DRK R Cc: serusers(a)lists.iptel.org; "Huertas García, Victor" Subject: RE: [Serusers] configure SER to work as a gateway Hi all, Thanks Rupert for your answer, I realise that my explanation was not very clear but the scenario I am working into is a little bit complex. Finally, I have decided to change my configuration and make the client sends its REGISTER message to SER and then it will forward the same message to my Server. However, the Server carries out the authentication process (I can not change it), so SER has to save the location contact after the client is authenticated in the Server. My function for the REGISTER message in SER is the following one: route[REGISTER] { t_newtran(); # send the register to the proxy rewritehostport("open-ims.test:4060"); t_on_reply("REGISTER_reply"); #t_on_failure("REGISTER_failure"); if (!t_relay()) { sl_send_reply("500","Error forwarding to P-CSCF"); break; } } onreply_route[REGISTER_reply] { if (t_check_status("401")){ log(-1, "A response from the P-CSCF!"); break; } if (t_check_status("200")){ log(-1, "User registered in the IMS Core!"); save_contacts("location"); break; } } But SER does not start. It shows this error: cng@cng:~$ sudo /etc/init.d/ser start -P /var/run/ser/ser.pid Starting ser: ser 0(2545) parse error (147,19-20): Command cannot be used in the block I have checked where is the error and it complains about the "save_contacts("location"))", but I do not know why I can not use it inside the onreply_route. Thanks in advance, Rebeca Martinez -----Mensaje original----- De: rupert.organ(a)bt.com [mailto:rupert.organ@bt.com] Enviado el: vie 08/01/2010 15:03 Para: Martínez García, Rebeca Asunto: RE: [Serusers] configure SER to work as a gateway Hi Rebeca, I think I understand what you are trying to do, albeit you describe it in a strange way. SER is designed to be a Registrar / SIP Proxy. It is not designed to proxy REGISTER messages. However I had to interface it to a SIP Application Server that wanted to see the REGISTER messages. However SER was my REGISTRAR and performed my security / authentication. Therefore I filter on the REGISTER messages using if (method=="REGISTER") { after doing the auth challenge in the ser.cfg, and after doing an eNum lookup and some user agent field checks I simply t_replicate the REGISTER. i.e. t_replicate("rebeca.node.es.com", "5060"); or if you do not use DNS t_replicate("10.1.x.y", "5060"); Note this replication does not include the auth challenge....but it satisfys my SIP app server functionality nicely, Also you can overwrite the useragent field if you wish... e.g. if (subst_uri("/;userAgent=rupert")) { subst('/^User-Agent:.*$/User-Agent: Rebeca/ig'); Good luck Rupert ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of "Martínez García, Rebeca" Sent: 08 January 2010 11:52 To: serusers(a)lists.iptel.org Cc: "Huertas García, Victor" Subject: [Serusers] configure SER to work as a gateway Hello all, Solved the installation problem I had, now I am trying to configure SER in a certain way. My configuration is the following one: Client ------------------------- SER ---------------------------- Server/Proxy 192.168.x.x 192.168.x.y 10.1.x.x 10.1.x.y Unlike the normal case where SER is used as the Registrar, my client (in a private LAN) must register in a Server/Proxy and SER has to act as a SIP gateway. However, at the same time, SER has to register locally this client (for internal calls). So, put in other words, SER should forward the REGISTER message (directed from the client to the Server/Proxy) and save locally the location of that client. Once registered the user, SER should act as a transparent proxy. But in the practice, it does nothing if I put that the client registers in the Server/Proxy. The messages reach SER but it does not forward them, nor it registers locally the clients. So I tried configuring the client to register in SER instead of the Server/Proxy and then SER started to work. Is a listening problem? What I have to include to the ser.cfg so SER handles also messages which are not directed to it but to the Server/Proxy? I hope somebody could help me. Thanks in advance, Rebeca Martinez

1 0

Re: [Serusers] configure SER to work as a gateway
by "Martínez García, Rebeca" 15 Jan '10

15 Jan '10

Hi all, Rupert you were right, SER does not permit to save the location based on the reg of a different upstream server; which means that I have to make "save_contacts("location") when SER receives the REGISTER. However, as it then sends directly to my client a 200 OK, when the client receives the "401 Unauthorized" error from my server, it does not authenticate with it because it has received a previous 200 OK (from SER...). So, here is my question: Is it possible to make SER not send a 200 OK when saving the location of the user? (I am only asking because I do not know). Thanks in advance, Rebeca Martinez -----Mensaje original----- De: rupert.organ(a)bt.com [mailto:rupert.organ@bt.com] Enviado el: lun 11/01/2010 12:37 Para: Martínez García, Rebeca Asunto: RE: [Serusers] configure SER to work as a gateway Rebeca, i am not an expert, but if I had to guess...then i would guess it would be because there is no contacts in the 200 to a Register. In our ser.cfg we save("location"); after testing if (method=="REGISTER") { but I can see that you are making registration and saving of info into location dependent on successful reg on a different upstream server. I do not think that is possible because you need to make the decision to save into location based on the reg that SER has received...and not based on the response of an upstream element....but if I am wrong I would be interested in the answer... Rupert ________________________________ From: "Martínez García, Rebeca" [mailto:remartinez@indra.es] Sent: 11 January 2010 11:21 To: Organ,RJE,Rupert,DRK R Cc: serusers(a)lists.iptel.org; "Huertas García, Victor" Subject: RE: [Serusers] configure SER to work as a gateway Hi all, Thanks Rupert for your answer, I realise that my explanation was not very clear but the scenario I am working into is a little bit complex. Finally, I have decided to change my configuration and make the client sends its REGISTER message to SER and then it will forward the same message to my Server. However, the Server carries out the authentication process (I can not change it), so SER has to save the location contact after the client is authenticated in the Server. My function for the REGISTER message in SER is the following one: route[REGISTER] { t_newtran(); # send the register to the proxy rewritehostport("open-ims.test:4060"); t_on_reply("REGISTER_reply"); #t_on_failure("REGISTER_failure"); if (!t_relay()) { sl_send_reply("500","Error forwarding to P-CSCF"); break; } } onreply_route[REGISTER_reply] { if (t_check_status("401")){ log(-1, "A response from the P-CSCF!"); break; } if (t_check_status("200")){ log(-1, "User registered in the IMS Core!"); save_contacts("location"); break; } } But SER does not start. It shows this error: cng@cng:~$ sudo /etc/init.d/ser start -P /var/run/ser/ser.pid Starting ser: ser 0(2545) parse error (147,19-20): Command cannot be used in the block I have checked where is the error and it complains about the "save_contacts("location"))", but I do not know why I can not use it inside the onreply_route. Thanks in advance, Rebeca Martinez -----Mensaje original----- De: rupert.organ(a)bt.com [mailto:rupert.organ@bt.com] Enviado el: vie 08/01/2010 15:03 Para: Martínez García, Rebeca Asunto: RE: [Serusers] configure SER to work as a gateway Hi Rebeca, I think I understand what you are trying to do, albeit you describe it in a strange way. SER is designed to be a Registrar / SIP Proxy. It is not designed to proxy REGISTER messages. However I had to interface it to a SIP Application Server that wanted to see the REGISTER messages. However SER was my REGISTRAR and performed my security / authentication. Therefore I filter on the REGISTER messages using if (method=="REGISTER") { after doing the auth challenge in the ser.cfg, and after doing an eNum lookup and some user agent field checks I simply t_replicate the REGISTER. i.e. t_replicate("rebeca.node.es.com", "5060"); or if you do not use DNS t_replicate("10.1.x.y", "5060"); Note this replication does not include the auth challenge....but it satisfys my SIP app server functionality nicely, Also you can overwrite the useragent field if you wish... e.g. if (subst_uri("/;userAgent=rupert")) { subst('/^User-Agent:.*$/User-Agent: Rebeca/ig'); Good luck Rupert ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of "Martínez García, Rebeca" Sent: 08 January 2010 11:52 To: serusers(a)lists.iptel.org Cc: "Huertas García, Victor" Subject: [Serusers] configure SER to work as a gateway Hello all, Solved the installation problem I had, now I am trying to configure SER in a certain way. My configuration is the following one: Client ------------------------- SER ---------------------------- Server/Proxy 192.168.x.x 192.168.x.y 10.1.x.x 10.1.x.y Unlike the normal case where SER is used as the Registrar, my client (in a private LAN) must register in a Server/Proxy and SER has to act as a SIP gateway. However, at the same time, SER has to register locally this client (for internal calls). So, put in other words, SER should forward the REGISTER message (directed from the client to the Server/Proxy) and save locally the location of that client. Once registered the user, SER should act as a transparent proxy. But in the practice, it does nothing if I put that the client registers in the Server/Proxy. The messages reach SER but it does not forward them, nor it registers locally the clients. So I tried configuring the client to register in SER instead of the Server/Proxy and then SER started to work. Is a listening problem? What I have to include to the ser.cfg so SER handles also messages which are not directed to it but to the Server/Proxy? I hope somebody could help me. Thanks in advance, Rebeca Martinez

1 0

[Serusers] Max-Forwards missing in SER CANCEL during forking
by Andres 15 Jan '10

15 Jan '10

Hi, We encountered an issue in 0.9.6 when SER forks requests because there are several endpoints registered with the same account (ie office and home phones). When one of the endpoints answers the call, SER sends a CANCEL to the other endpoints so they can stop further call processing. It turns our that this SER generated CANCEL does not contain the "Max-Forwards" header which apparently violates RFC3261. Here is an example: U SER IP:5060 -> 10.10.1.50:36679 CANCEL sip:3055551234@10.10.1.50:36679;rinstance=C926AD943D258534F5F62913216C50ADC9D895D2; transport=udp SIP/2.0..Via: SIP/2.0/UDP 10.10.1.21:5065;branch=z9hG4bK1c1e.32eeb702.1.. From: "9545551234" <sip:9545551234@10.10.1.20:5066>;tag=as7d1243c9.. Call-ID: 4385c66230a310e714f020926aa2028c(a)10.10.1.20.. To: <sip:3055551234@X.X.X.X:5065>..CSeq: 102 CANCEL.. User-Agent: Sip EXpress router(0.9.6 (i386/linux)).. Content-Length: 0.... Is there any fix or workaround for this? How can we get that "Max-Forwards" header in the Message if it is internally generated by SER and no something that flows through the config script. Thanks, Andres http://www.telesip.net

2 3

[Kamailio-Users] mk_proxy: could not resolve hostname
by kévin l'heveder 14 Jan '10

14 Jan '10

Hi, I'm using kamailio V3.0 In the kamailio.cfg when I use the "send" function with an IP inside (as send("127.0.01:51001"), I see the following error : mk_proxy: could not resolve hostname: "127.0.01:51001" At first, kamailio was sending request to the DNS server which define in resolv.conf to resolve "127.0.01:51001" The DNS could not resolve this name then the error occured So I removed this defined DNS and now no request are sending but still have the same error. I tried to set dns and use_dns_search_list parameters to no, but that's change nothing Anybody have any ideas? Regards, Koon

3 2

[Serusers] ser scalability
by Sabatella, Michael 14 Jan '10

14 Jan '10

I just wanted to run this by the group. I did a complete overhaul on one of my ser environments and upgraded the physical server and all software around my sip proxies. I am also maintaining a complete front end for provisioning and billing that I also moved(PHP/mysql) I am running this on a Virtual machine (vmware esxi). If I need more proxies I can clone the VM and build in new service/capacity pretty fast. I am running 0.9.6 on Redhat ES 5. I have 4 cores dedicated to this server running Xeon 2.93GHz processers and 4 GIG memory. I am running the 32 bit OS. I hardly see the processor tick up to 1%. The machine I am testing has about 300 registered endpoints in the proxy db. My questions are: 1. What is the scalability of ser processing calls? 2. Are there any limits on the number of users that can be registered at one time. Is the registration memory based and am I severely limited with 4GB of memory? 3. Is there anyone that has 1000s of registration to the proxy and if so I would like to hear how you are scaling it. What type of server are you using? 4. What are some good metrics to look at on the sip proxy to gauge its performance? 5. Is there a way to capture calls per minute? Is anyone doing this? I have call accounting set up. Thanks, Michael ________________________________ DISCLAIMER: This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unintended recipients are prohibited from taking action on the basis of information in this e-mail.E-mail messages may contain computer viruses or other defects, may not be accurately replicated on other systems, or may be intercepted, deleted or interfered with without the knowledge of the sender or the intended recipient. If you are not comfortable with the risks associated with e-mail messages, you may decide not to use e-mail to communicate with IPC. IPC reserves the right, to the extent and under circumstances permitted by applicable law, to retain, monitor and intercept e-mail messages to and from its systems.

2 1

[Kamailio-Users] 64kbit, quad/dual core support
by mbsip 14 Jan '10

14 Jan '10

Hi all, I am wondering if kamailio do support 64-bit OS systems and dual/quad core cpu platforms? The question is raised because i am about to prepare hardware specification for kamailio server. Thanks in advance, Maciej.

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Re: [Kamailio-Users] about kamailio with sctp
by Daniel-Constantin Mierla 14 Jan '10

14 Jan '10

Hi Bruce, On 1/14/10 12:54 PM, Bruce McAlister wrote: > Hi Daniel, > > >> you may ask this one on sr-dev(a)lists.sip-router.org since the devel of >> SCTP use to be more active there. >> >> > Thanks, I will ask there too. > > >> When I implemented in in old kamailio, it was no option for me than >> using kamailio-to-kamailio SCTP channel. >> >> > When you did the testing between the two kamailio servers, how did you > enforce SCTP as the transport? > > Does t_relay offer a way to specify UDP/TCP/SCTP as the transport to relay > the messages between servers? > > if you set dst uri with transport parameter then t_relay() will use it, e.g.: $du = "sip:10.0.0.1;transport=sctp"; t_relay(); Or you can use t_relay_to_sctp() or t_relay_to() with appropriate parameters: http://kamailio.org/docs/modules/3.0.x/modules/tm.html Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

Re: [Kamailio-Users] about kamailio with sctp
by Daniel-Constantin Mierla 14 Jan '10

14 Jan '10

Hello, On 1/14/10 2:25 AM, Bruce McAlister wrote: > Hi Klaus, > > >> I do not know. For kamailio 1.5 I would just grep for SCTP in the >> sources to get the answer. >> For kamailio 3.0 (based on sip-router with a new (I guess much better) >> SCTP implementation there is probably another method needed. A cite from >> another email from Andrei: >> >> >>> Yes, make cfg SCTP=1 and then make all will compile with SCTP support. >>> I'm currently pondering whether or not I should make the SCTP support >>> automatic (if the needed *.h files are installed compile it >>> > automatically). > > I am compiling Kamailio 3.0.0 on Solaris 10 which does have SCTP support > inherent in the OS. I did a 'grep -i sctp Makefile*' and saw that there were > checks for SCTP in the makefiles. > > I ended up building Kamailio 3.0.0 with the following command line: > > make prefix=/opt/kamailio \ > SCTP=1 \ > CC_EXTRA_OPTS=-I/usr/gnu/include \ > group_include="standard postgres presence" \ > include_modules="snmpstats perl tls" \ > all > > The build went through successfully and I can see that sctp_server.o has > been linked into the kamailio binary when the build is running. > > Does anyone know of any tool that I can use to test Kamailio using the SCTP > protocol? > you may ask this one on sr-dev(a)lists.sip-router.org since the devel of SCTP use to be more active there. When I implemented in in old kamailio, it was no option for me than using kamailio-to-kamailio SCTP channel. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

2 1

[Kamailio-Users] running Qjsimple with kamailio 1.5 + tls
by mustafa samara 14 Jan '10

14 Jan '10

Hello i am trying Qjsimple 0.6.3 with Kamailio 1.5 with tls support, i make the user register successfully but when i am trying to invite another user i got the following error: " transport_srtp Failed generating random key: unspecified failure pjsua_call.c Error initializing media channel: Unknown error 259800 [status=259800] Dialog destroyed QjSimple: Error calling buddy: Unknown error 259800 [status=259800] " the sip setting as the following: protocol: tcp srtp : mandatory srtp requirements: tls also i got the same error if the sip setting as the following: protocol: tls srtp : mandatory srtp requirements: tls or: protocol: tls srtp : mandatory srtp requirements: sips any body help me please? -- Eng.Mustafa Al-Samara

2 2

Re: [Kamailio-Users] about kamailio with sctp
by Klaus Darilion 14 Jan '10

14 Jan '10

Bruce McAlister schrieb: > Hi All, > > I am wanting to play with Kamailio 3.0.0 and SCTP as well. I just want to > verify that I am building it in to Kamailio properly. When I build Kamailio > do I need to have USE_SCTP=1 as an option on the make command line to enable > SCTP support? > > I can see there are updates in the ChangeLog for SCTP but I cannot find how > to enable support for it in the INSTALL/README-MODULES documentation. > > Can someone point me in the right direction please. I do not know. For kamailio 1.5 I would just grep for SCTP in the sources to get the answer. For kamailio 3.0 (based on sip-router with a new (I guess much better) SCTP implementation there is probably another method needed. A cite from another email from Andrei: > Yes, make cfg SCTP=1 and then make all will compile with SCTP support. > I'm currently pondering whether or not I should make the SCTP support > automatic (if the needed *.h files are installed compile it automatically). regards klaus > > Thanks > Bruce > > -----Original Message----- > From: users-bounces(a)lists.kamailio.org > [mailto:users-bounces@lists.kamailio.org] On Behalf Of Daniel-Constantin > Mierla > Sent: 11 January 2010 09:57 > To: mustafa samara > Cc: users(a)lists.kamailio.org > Subject: Re: [Kamailio-Users] about kamailio with sctp > > Hello, > > On 1/11/10 1:37 AM, mustafa samara wrote: >> Hello >> i do not know how to try sctp, imake kmailio 1.5 with sctp=1 >> could you help me? > if you want to play with sctp then I recommend to use 3.0.0 (to be > released today). It has far more improvements and new features than > older versions to SCTP. You can take it from git now: > http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git > > Some highlights about sctp and 3.0.0: > http://by-miconda.blogspot.com/2010/01/best-of-new-in-kamailio-300-15-sctp.h > tml > > Cheers, > Daniel >

2 1

[Kamailio-Users] Rtpproxy not recording media stream
by Vikram Ragukumar 13 Jan '10

13 Jan '10

Hello, I have setup Kamailio(v1.5) and Rtpproxy(v1.2.1) on a Centos5.4 linux box. Kamailio's config file has been edited to include Nat and mysql support. With Kamailio running, i started rtpproxy using rtpproxy -l ip_addrs -s udp:127.0.0.1:7722 -a -r /var/log -F However rtpproxy does not seem to be recording any session. I read that the -a switch can be used to record all sessions. Am i missing a switch/doing something wrong while invoking rtpproxy ? Regards, Vikram.

6 9

[Kamailio-Users] Logical operators
by Antonio Reale 13 Jan '10

13 Jan '10

Hi, I'm trying this: if(is_avp_set("$avp(s:avp)") && !isflagset(1) || is_avp_set("$avp(s:avp)") && !isflagset(2)) { ...; } The first condition before OR operator is never matched. Could you please suggest the correct syntax? Thank you very much. Regards. -- Antonio

2 4

[Kamailio-Users] SIP proxy is expecting "Route:" header to be filled information from the last "Contact:" header ?
by Julien Chavanton 13 Jan '10

13 Jan '10

Hi all, first thank you for maintaining this list and free/Open source SIP proxy software. I am working on one interco problem with a remote SIP proxy running on Kamailio. Cisco(U1) --> Opensips(P1) --> Internet --> Kamailio(P2) --> Asterisk(U2) The remote proxy(P2) is expecting us to fill the "Route:" header taking information from the "Contact:" header, according to my research the "Route:" header as to be filled using the information found in the last "Record-route:" header. Note that the "Contact:" header IP address is the one of U2, as it is the destination UA. I think our partner using Kalamailio only interconnected UA not Proxy, since they never faced this problem. Of course I do not suspect Kalamailio to be doing something wrong, I just want to validate what we should expect, and investigate further from there. I will have to contact the technicien that as configured the Kalamailio server to validate why face this behavior. -------------------------------------------------------------------------- " 16.12.1.3 Rewriting Record-Route Header Field Values " http://www.ietf.org/rfc/rfc3261.txt -------------------------------------------------------------------------- We found a temporary solution to make it worko, where we hardcoded the replacement of the IP address in the "Route:" header with the one of the U2 subst('/^Route: <sip:P2;(.*)$/Route: <sip:U2;\1/');

3 7

[Kamailio-Users] Does uac_auth work for you?
by Andrey Utkin 13 Jan '10

13 Jan '10

Hi all. Please share your experience. If you ever tried authenticating outgoing calls with Kamailio, did it work for you? Which devices or servers you know do/don't accept such authentication? Do you use it in some tricky way to get it working?

4 3

[Kamailio-Users] about kamailio with sctp
by mustafa samara 13 Jan '10

13 Jan '10

Hello i do not know how to try sctp, imake kmailio 1.5 with sctp=1 could you help me? thnaks -- Eng.Mustafa Al-Samara

3 2

Kamailio (OpenSER) v3.0.0 Released
by Daniel-Constantin Mierla 12 Jan '10

12 Jan '10

Hi everybody, this ML was forgotten yesterday to be cc-ed ...for the sake of archiving, here it is the announcement. After ten months since previous major release 1.5.0, I am glad to announce that Kamailio (OpenSER) v3.0.0 is out. It is a special release, being the first based on SIP Router project. Along with new things developed during the past months, you can blend features and modules provided by previous versions of Kamailio (OpenSER) and SIP Express Router (SER) in same configuration file. Given these, 3.0.0 offers an impressive number of new features and improvements, read release notes at: http://www.kamailio.org/mos/view/Kamailio-OpenSER-v3.0.0-Release-Notes Wiki page with more details about what is new: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.0.x The tarball with sources: http://www.kamailio.org/pub/kamailio/latest/src/ Binary packages for several distributions: http://www.kamailio.org/pub/kamailio/latest/bin/ http://www.kamailio.org/pub/kamailio/latest/packages/ Modules' documentation: - http://www.kamailio.org/docs/modules/3.0.x/ Cookbooks and more wiki docs: - http://www.kamailio.org/dokuwiki/ Migration guidelines: - http://www.kamailio.org/dokuwiki/doku.php/install:1.5.x-to-3.0.0 From Kamailio project point of view, this is the first major milestone accomplished within SIP Router project, proving its viability. The work done so far, to offer the features brought out by this release, together with the development teams working together, make the future of the SIP Router project very solid. I would like to thank to everybody that contributed to get SIP Router project at this stage and I am eager to get to next major release from it source tree -- the work will continue in order to offer a more tight integration of Kamailio (OpenSER) and SIP Express Router (SER) projects, of course, along with development of new features. Cheers, Daniel

1 0

[Kamailio-Users] Checking the destination IP after dispatcher
by Geoffrey Mina 12 Jan '10

12 Jan '10

What pseudo variable would i check after running ds_next_dst() to check the IP we are about to forward the INVITE to? Basically I want to have a check for $si == [something] to ensure I am not about to send the INVITE to the same UI which just requested it. While I have your attention, I may as well see if anyone has a better solution for what I am trying to accomplish: I am doing this because asterisk can't handle a hairpin scenario and I need to generate two distinct calls in my network, so I figure i'll just forward the request to another server in the network to deal with this. INVITE from PSTN --> Kamailio --> Asterisk#1 [Asterisk#1 needs to create a new session, but stay on-net] --> INVITE from Asterisk#1 to Kamailio --> [Dispatcher] --> INVITE from Kamailio to Asterisk#2 This would basically create 3 legs of the call, which is what I want PSTN --> Asterisk#1 --> Asterisk#2 I would prefer if Asterisk#1 could just INVITE to LocalHost, but that doesn't seem to be allowed. Not sure if that's a limitation of SIP or Asterisk... Thanks!

3 8

[Kamailio-Users] job offer: VoIP developer, team Bucharest
by Henning Westerholt 12 Jan '10

12 Jan '10

Job offer: Interested in working with one of the biggest Voice over IP networks in europe? We're hiring and have one additional VoIP developer position open in the team in Bucharest, Romania at the moment. For more informations (requirements, how to apply..) please refer to the opening on the job web page. Some facts about our VoIP system can be found at e.g. [1], please feel free to contact me per mail as well. http://cariere.1and1.ro/xml/JobsDetail?job_id=2142 Best regards, Henning Westerholt [1] http://www.kamailio.org/events/2009-linuxtag/presentation.pdf -- Henning Westerholt - Development Consumer Products / Consumer Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

1 0

[Kamailio-Users] This has probably been asked a thousand times before...
by Robert B 12 Jan '10

12 Jan '10

Regardless, despite my Googlage, I can't seem to find what I am looking for. I've looked at the Wiki and the documentation. While I do have a programming background, I still am having a hard time grasping some of these tutorials. Many of them seem very specific and not at all relevant to what I am trying to do. I feel like I'm left without a good starting point. My goal is to have Kamailio setup as an ITSP would. That is, I have three sipxecs PBX systems here, and rather than have each one configured with umpteen SIP trunks that I'll need to manage, I want to have (for the time being) just one -- that one would be my personal Kamailio server. When a user makes a call, it first goes to my Kamailio server which then distribute the calls to my ITSPs (like Bandwidth.com, voip.ms, etc). I want the ACLs to be setup based on IP addresses and not user registrations. I want nothing to do with the media or RTP streams. First step would be termination (outbound). The obvious next step would be setting it up for origination (inbound). I've downloaded and installed the SIREMIS web UI for OpenSER 1.4.x, but as I said my lack of understanding of the terminology's context prohibits me from really exploring anything. Some areas look familiar, others seem redundant, etc. The reason I started with the web UI was because it's obviously easier and it also produces output that is syntactically correct for my own examination and learning purposes. What would the appropriate terminology for this kind of setup be? Can anyone point out a Wiki article which I may have missed? To me, this seems like the most basic of basic configurations and a seemingly obvious launching point for anyone getting into the software... Thanks! -- Robert

3 2

[Kamailio-Users] Kamailio (OpenSER) v3.0.0 Released
by Daniel-Constantin Mierla 11 Jan '10

11 Jan '10

Hi everybody, after ten months since previous major release 1.5.0, I am glad to announce that Kamailio (OpenSER) v3.0.0 is out. It is a special release, being the first based on SIP Router project. Along with new things developed during the past months, you can blend features and modules provided by previous versions of Kamailio (OpenSER) and SIP Express Router (SER) in same configuration file. Given these, 3.0.0 offers an impressive number of new features and improvements, read release notes at: http://www.kamailio.org/mos/view/Kamailio-OpenSER-v3.0.0-Release-Notes Wiki page with more details about what is new: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.0.x The tarball with sources: http://www.kamailio.org/pub/kamailio/latest/src/ Binary packages for several distributions: http://www.kamailio.org/pub/kamailio/latest/bin/ http://www.kamailio.org/pub/kamailio/latest/packages/ Modules' documentation: - http://www.kamailio.org/docs/modules/3.0.x/ Cookbooks and more wiki docs: - http://www.kamailio.org/dokuwiki/ Migration guidelines: - http://www.kamailio.org/dokuwiki/doku.php/install:1.5.x-to-3.0.0 From Kamailio project point of view, this is the first major milestone accomplished within SIP Router project, proving its viability. The work done so far, to offer the features brought out by this release, together with the development teams working together, make the future of the SIP Router project very solid. I would like to thank to everybody that contributed to get SIP Router project at this stage and I am eager to get to next major release from it source tree -- the work will continue in order to offer a more tight integration of Kamailio (OpenSER) and SIP Express Router (SER) projects, of course, along with development of new features. Cheers, Daniel

3 2

[Kamailio-Users] preparing Kamailio 3.0.0 release
by Daniel-Constantin Mierla 11 Jan '10

11 Jan '10

Hello, as announced previously, today will package the kamailio 3.0.0 release. Please commit any change to kamailio_3.0 branch before 14:00UTC to avoid overlappings. If you are aware of new bugs, just drop an email to devel mailing list. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 1

Re: [Kamailio-Users] Users Digest, Vol 56, Issue 19
by mustafa samara 11 Jan '10

11 Jan '10

thank you very much i will see kamailio 3.0 but is there any voip soft phone which support sctp, please? thanks

2 1

Re: [Serusers] configure SER to work as a gateway
by "Martínez García, Rebeca" 11 Jan '10

11 Jan '10

Hi all, Thanks Rupert for your answer, I realise that my explanation was not very clear but the scenario I am working into is a little bit complex. Finally, I have decided to change my configuration and make the client sends its REGISTER message to SER and then it will forward the same message to my Server. However, the Server carries out the authentication process (I can not change it), so SER has to save the location contact after the client is authenticated in the Server. My function for the REGISTER message in SER is the following one: route[REGISTER] { t_newtran(); # send the register to the proxy rewritehostport("open-ims.test:4060"); t_on_reply("REGISTER_reply"); #t_on_failure("REGISTER_failure"); if (!t_relay()) { sl_send_reply("500","Error forwarding to P-CSCF"); break; } } onreply_route[REGISTER_reply] { if (t_check_status("401")){ log(-1, "A response from the P-CSCF!"); break; } if (t_check_status("200")){ log(-1, "User registered in the IMS Core!"); save_contacts("location"); break; } } But SER does not start. It shows this error: cng@cng:~$ sudo /etc/init.d/ser start -P /var/run/ser/ser.pid Starting ser: ser 0(2545) parse error (147,19-20): Command cannot be used in the block I have checked where is the error and it complains about the "save_contacts("location"))", but I do not know why I can not use it inside the onreply_route. Thanks in advance, Rebeca Martinez -----Mensaje original----- De: rupert.organ(a)bt.com [mailto:rupert.organ@bt.com] Enviado el: vie 08/01/2010 15:03 Para: Martínez García, Rebeca Asunto: RE: [Serusers] configure SER to work as a gateway Hi Rebeca, I think I understand what you are trying to do, albeit you describe it in a strange way. SER is designed to be a Registrar / SIP Proxy. It is not designed to proxy REGISTER messages. However I had to interface it to a SIP Application Server that wanted to see the REGISTER messages. However SER was my REGISTRAR and performed my security / authentication. Therefore I filter on the REGISTER messages using if (method=="REGISTER") { after doing the auth challenge in the ser.cfg, and after doing an eNum lookup and some user agent field checks I simply t_replicate the REGISTER. i.e. t_replicate("rebeca.node.es.com", "5060"); or if you do not use DNS t_replicate("10.1.x.y", "5060"); Note this replication does not include the auth challenge....but it satisfys my SIP app server functionality nicely, Also you can overwrite the useragent field if you wish... e.g. if (subst_uri("/;userAgent=rupert")) { subst('/^User-Agent:.*$/User-Agent: Rebeca/ig'); Good luck Rupert ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of "Martínez García, Rebeca" Sent: 08 January 2010 11:52 To: serusers(a)lists.iptel.org Cc: "Huertas García, Victor" Subject: [Serusers] configure SER to work as a gateway Hello all, Solved the installation problem I had, now I am trying to configure SER in a certain way. My configuration is the following one: Client ------------------------- SER ---------------------------- Server/Proxy 192.168.x.x 192.168.x.y 10.1.x.x 10.1.x.y Unlike the normal case where SER is used as the Registrar, my client (in a private LAN) must register in a Server/Proxy and SER has to act as a SIP gateway. However, at the same time, SER has to register locally this client (for internal calls). So, put in other words, SER should forward the REGISTER message (directed from the client to the Server/Proxy) and save locally the location of that client. Once registered the user, SER should act as a transparent proxy. But in the practice, it does nothing if I put that the client registers in the Server/Proxy. The messages reach SER but it does not forward them, nor it registers locally the clients. So I tried configuring the client to register in SER instead of the Server/Proxy and then SER started to work. Is a listening problem? What I have to include to the ser.cfg so SER handles also messages which are not directed to it but to the Server/Proxy? I hope somebody could help me. Thanks in advance, Rebeca Martinez

1 0

[Kamailio-Users] oracle sqlops URL
by alex pappas 11 Jan '10

11 Jan '10

Hi all, happy 2010 for all of us! Dear friends I'm trying to setup a sqlops connection to an Oracle db. I loaded the db_oracle module with (loadmodule "db_oracle.so") without parameters. The connection string that I gave in sqlops module is : modparam("sqlops","sqlcon","ca2=>user/pass@ip_addres:1521:db_name") When i try to start kamailio I get the following errors and kamailio does NOT start: ERROR:core:db_check_api: module db_user/pass@ip does not export db_use_table function ERROR:core:init_mod_child: failed to initializing module sqlops, rank 2 ERROR:core:main_loop: init_child failed for UDP listener Thank in advance and merry Christmas Alex

4 12

[Kamailio-Users] about kamailio and tls
by mustafa samara 11 Jan '10

11 Jan '10

i am mustafa samara master degree student. i try to test qjsimple with kamailio openser is it possible (to test the tls support) ? also i want to ask about ( in sip preferences) what is the deference when we you use tls as a protocol or when we use (tls or sips) as a SRTP requirements. clould you help me please? i wat to include the result in my thesis -- Eng.Mustafa Al-Samara

3 2

[Kamailio-Users] PDD
by Anders 09 Jan '10

09 Jan '10

Hi, Anyone with a smooth solution for tracking PDD of every call? Thanks!

5 6

[Serusers] configure SER to work as a gateway
by "Martínez García, Rebeca" 08 Jan '10

08 Jan '10

Hello all, Solved the installation problem I had, now I am trying to configure SER in a certain way. My configuration is the following one: Client ------------------------- SER ---------------------------- Server/Proxy 192.168.x.x 192.168.x.y 10.1.x.x 10.1.x.y Unlike the normal case where SER is used as the Registrar, my client (in a private LAN) must register in a Server/Proxy and SER has to act as a SIP gateway. However, at the same time, SER has to register locally this client (for internal calls). So, put in other words, SER should forward the REGISTER message (directed from the client to the Server/Proxy) and save locally the location of that client. Once registered the user, SER should act as a transparent proxy. But in the practice, it does nothing if I put that the client registers in the Server/Proxy. The messages reach SER but it does not forward them, nor it registers locally the clients. So I tried configuring the client to register in SER instead of the Server/Proxy and then SER started to work. Is a listening problem? What I have to include to the ser.cfg so SER handles also messages which are not directed to it but to the Server/Proxy? I hope somebody could help me. Thanks in advance, Rebeca Martinez

3 2

[Kamailio-Users] Unreadable script
by Bruno Machado 08 Jan '10

08 Jan '10

Hi all I'm using the version 1.5.2 and I'd like to know from you if it is possible to 'hide' the config file. Using other words, becoming the file unreadable for us, humans. There are people that have root access to the machine and I don't want to let them read the script. Does Kamailio have a function to do this kind of job? tks, Machado ____________________________________________________________________________________ Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com

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[Kamailio-Users] Problems with avp_printf in Kamailio-3.0.0
by Santiago Gimeno 07 Jan '10

07 Jan '10

Hi, I'm getting errors when trying to use avp_printf in K-3.0.0. The script line, for example, is: avp_printf("$avp(i:905)", "$rU"); The error is: kamailio: DEBUG: <core> [sr_module.c:516]: find_export_record: <avp_printf> not found I've tried pv_printf with no luck either. This was working correctly in K-1.5 Could this be a bug? Best regards, Santi

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[Kamailio-Users] openser/kamailio subscribe / MWI
by Joel Webster 07 Jan '10

07 Jan '10

I am looking to MWI messages openser/kamailio and send the MWI message back based on a database lookup or other means within a routing block, when the users register though, i get errors like: Jan 3 12:07:20 ser1 /usr/local/sbin/kamailio[20317]: ERROR:presence:get_database_info: No matching subscription dialog found in database Jan 3 12:07:20 ser1 /usr/local/sbin/kamailio[20317]: ERROR:presence:handle_subscribe: getting stored info and when I disable the DB setup I had, I get a similar error: Jan 4 16:59:48 ser1 /usr/local/sbin/kamailio[5211]: ERROR:presence:get_stored_info: record not found in hash_table Jan 4 16:59:48 ser1 /usr/local/sbin/kamailio[5211]: ERROR:presence:handle_subscribe: getting stored info

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Re: [Kamailio-Users] [kamailio-users] Kamailio - Polycom Sound Point
by Ramu 07 Jan '10

07 Jan '10

Hi, Your configuration looks fine. Can you try with 2 X-Lite phones on different machines just to narrow down the problem? On Wed, Jan 6, 2010 at 4:29 PM, Chandrakant Solanki < solanki.chandrakant(a)gmail.com> wrote: > Hi > > Thanks for reply > > Here is kamailio.cfg file > > ================================================================================================ > listen = 192.168.94.30 > port = 5060 > mpath = "/lib/kamailio/modules" > debug = 9 > fork = yes > disable_tcp = no > log_facility = LOG_LOCAL0 > log_stderror = no > dns = no > rev_dns = no > > loadmodule "usrloc.so" > modparam("usrloc", "user_column", "username") > modparam("usrloc", "domain_column", "domain") > modparam("usrloc", "contact_column", "contact") > modparam("usrloc", "expires_column", "expires") > modparam("usrloc", "q_column", "q") > modparam("usrloc", "callid_column", "callid") > modparam("usrloc", "cseq_column", "cseq") > modparam("usrloc", "methods_column", "methods") > modparam("usrloc", "flags_column", "flags") > modparam("usrloc", "user_agent_column", "user_agent") > modparam("usrloc", "received_column", "received") > modparam("usrloc", "socket_column", "socket") > modparam("usrloc", "use_domain", 0) > modparam("usrloc", "desc_time_order", 0) > modparam("usrloc", "timer_interval", 60) > modparam("usrloc", "db_url", "mysql:// > database:database@192.168.90.12/openser") > modparam("usrloc", "db_mode", 1) > modparam("usrloc", "matching_mode", 0) > modparam("usrloc", "cseq_delay", 20) > modparam("usrloc", "nat_bflag", 6) > > loadmodule "pv.so" > > loadmodule "registrar.so" > modparam("registrar", "default_expires", 3600) > modparam("registrar", "min_expires", 60) > modparam("registrar", "max_expires", 0) > modparam("registrar", "default_q", 0) > modparam("registrar", "append_branches", 1) > modparam("registrar", "case_sensitive", 0) > modparam("registrar", "received_param", "received") > modparam("registrar", "max_contacts", 100) > modparam("registrar", "retry_after", 0) > modparam("registrar", "method_filtering", 0) > modparam("registrar", "path_mode", 2) > modparam("registrar", "path_use_received", 0) > modparam("registrar", "received_avp", "$avp(i:801)") > > loadmodule "rr.so" > modparam("rr", "enable_full_lr", 0) > modparam("rr", "append_fromtag", 1) > modparam("rr", "enable_double_rr", 1) > modparam("rr", "add_username", 0) > > loadmodule "tm.so" > modparam("tm", "fr_timer", 3) > modparam("tm", "fr_inv_timer", 35) > modparam("tm", "wt_timer", 5) > modparam("tm", "delete_timer", 2) > modparam("tm", "ruri_matching", 1) > modparam("tm", "via1_matching", 1) > modparam("tm", "unix_tx_timeout", 2) > modparam("tm", "restart_fr_on_each_reply", 1) > modparam("tm", "pass_provisional_replies", 0) > modparam("tm", "fr_inv_timer_avp", "$avp(s:callee_fr_inv_timer)") > > loadmodule "xlog.so" > modparam("xlog", "buf_size", 4096) > modparam("xlog", "force_color", 0) > > loadmodule "mi_fifo.so" > modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") > > loadmodule "domain.so" > modparam("domain", "db_url", "mysql:// > database:database@192.168.90.12/openser") > modparam("domain", "db_mode", 1) > modparam("domain", "domain_table", "domain") > modparam("domain", "domain_col", "domain") > > loadmodule "nathelper.so" > modparam("nathelper", "natping_interval", 60) > modparam("nathelper", "ping_nated_only", 1) > modparam("nathelper", "sipping_method", "OPTIONS") > modparam("nathelper", "received_avp", "$avp(i:801)") > > loadmodule "textops.so" > > loadmodule "sl.so" > modparam("sl", "enable_stats", 1) > > loadmodule "maxfwd.so" > modparam("maxfwd", "max_limit", 256) > > loadmodule "db_mysql.so" > modparam("db_mysql", "ping_interval", 300) > modparam("db_mysql", "auto_reconnect", 1) > > loadmodule "auth.so" > modparam("auth", "nonce_expire", 300) > modparam("auth", "rpid_suffix", > ";party=calling;id-type=subscriber;screen=yes") > modparam("auth", "rpid_avp", "$avp(s:rpid)") > > loadmodule "auth_db.so" > modparam("auth_db", "db_url", "mysql:// > database:database@192.168.90.12/openser") > modparam("auth_db", "user_column", "username") > modparam("auth_db", "domain_column", "domain") > modparam("auth_db", "password_column", "password") > modparam("auth_db", "password_column_2", "ha1b") > modparam("auth_db", "calculate_ha1", 1) > modparam("auth_db", "use_domain", 1) > modparam("auth_db", "load_credentials", "$avp(s:caller_uuid)=uuid") > > loadmodule "uri_db.so" > modparam("uri_db", "db_url", "mysql:// > database:database@192.168.90.12/openser") > > loadmodule "avpops.so" > modparam("avpops", "db_url", "mysql:// > database:database@192.168.90.12/openser") > modparam("avpops", "avp_table", "usr_preferences") > modparam("avpops", "use_domain", 1) > modparam("avpops", "uuid_column", "uuid") > modparam("avpops", "username_column", "username") > modparam("avpops", "domain_column", "domain") > modparam("avpops", "attribute_column", "attribute") > modparam("avpops", "value_column", "value") > modparam("avpops", "type_column", "type") > > loadmodule "lcr.so" > modparam("lcr", "db_url", "mysql://database:database@192.168.90.12/openser > ") > modparam("lcr", "gw_table", "gw") > modparam("lcr", "gw_name_column", "gw_name") > modparam("lcr", "ip_addr_column", "ip_addr") > modparam("lcr", "port_column", "port") > modparam("lcr", "uri_scheme_column", "uri_scheme") > modparam("lcr", "transport_column", "transport") > modparam("lcr", "grp_id_column", "grp_id") > modparam("lcr", "lcr_table", "lcr") > modparam("lcr", "strip_column", "strip") > modparam("lcr", "prefix_column", "prefix") > modparam("lcr", "from_uri_column", "from_uri") > modparam("lcr", "priority_column", "priority") > modparam("lcr", "flags_avp", "$avp(i:712)") > modparam("lcr", "gw_uri_avp", "$avp(i:1400)") > modparam("lcr", "ruri_user_avp", "$avp(i:1402)") > modparam("lcr", "fr_inv_timer_avp", "$avp(s:fr_inv_timer_avp)") > modparam("lcr", "fr_inv_timer", 90) > modparam("lcr", "fr_inv_timer_next", 30) > modparam("lcr", "rpid_avp", "$avp(s:rpid)") > > loadmodule "alias_db.so" > modparam("alias_db", "db_url", "mysql:// > database:database@192.168.90.12/openser") > modparam("alias_db", "user_column", "username") > modparam("alias_db", "domain_column", "domain") > modparam("alias_db", "alias_user_column", "alias_username") > modparam("alias_db", "alias_domain_column", "alias_domain") > modparam("alias_db", "use_domain", 0) > > loadmodule "uac.so" > modparam("uac", "rr_store_param", "vsf") > modparam("uac", "from_restore_mode", "auto") > modparam("uac", "from_passwd", "s1p:Wiz4rd!") > > loadmodule "acc.so" > modparam("acc", "db_url", "mysql://database:database@192.168.90.12/openser > ") > modparam("acc", "db_flag", 1) > modparam("acc", "db_missed_flag", 2) > modparam("acc", "log_flag", 1) > modparam("acc", "log_missed_flag", 2) > modparam("acc", "log_extra", > "src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") > modparam("acc", "db_extra", > "src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") > > loadmodule "mi_datagram.so" > modparam("mi_datagram", "socket_name", "udp:127.0.0.1:8080" ) > modparam("mi_datagram", "children_count", 3 ) > modparam("mi_datagram", "unix_socket_mode", 0600 ) > modparam("mi_datagram", "unix_socket_group", "root" ) > modparam("mi_datagram", "unix_socket_user", "root" ) > modparam("mi_datagram", "socket_timeout", 2000 ) > > loadmodule "mediaproxy.so" > modparam("mediaproxy", "mediaproxy_socket", > "/var/run/mediaproxy/dispatcher.sock") > modparam("mediaproxy", "mediaproxy_timeout", 500) > modparam("mediaproxy", "signaling_ip_avp", "$avp(s:nat_ip)") > modparam("mediaproxy", "media_relay_avp", "$avp(s:media_relay)") > > loadmodule "nat_traversal.so" > modparam("nat_traversal", "keepalive_interval", 90) > modparam("nat_traversal", "keepalive_method", "OPTIONS") > modparam("nat_traversal", "keepalive_from", "sip:keepalive@mydomain.com<sip%3Akeepalive(a)mydomain.com> > ") > > route[0] > { > xlog("L_DBG", "New Request - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > force_rport(); > > if(msg:len > max_len) > { > xlog("L_DBG", "Message too big - M=$rm RURI=$ru F=$fu T=$tu > IP=$si ID=$ci\n"); > sl_send_reply("513", "Message Too Big"); > exit; > } > if (!mf_process_maxfwd_header("10")) > { > xlog("L_DBG", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu > IP=$si ID=$ci\n"); > sl_send_reply("483", "Too Many Hops"); > exit; > } > if(!is_method("REGISTER")) > { > if(nat_uac_test("19")) > { > record_route(";nat=yes"); > } > else > { > record_route(); > } > } > if (is_method("SUBSCRIBE")) { > # challenge/response > if (!www_authorize("", "subscriber")) { > www_challenge("", "0"); > exit; > } > exit; > } > > if(is_method("CANCEL") || is_method("BYE")) > { > end_media_session(); > } > if(loose_route()) > { > if(nat_uac_test("19") || search("^Route:.*;nat=yes")) > { > fix_nated_contact(); > setbflag(6); > } > route(8); > } > if(is_method("REGISTER")) > { > route(7); > } > if(is_method("INVITE")) > { > use_media_proxy(); > route(9); > } > if(is_method("CANCEL") || is_method("ACK")) > { > route(13); > } > route(14); > } > > ######################################################################## > # Request route 'clear-usr-preferences-caller' > ######################################################################## > route[1] > { > xlog("L_DBG", "Clear caller preferences - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > avp_delete("$avp(s:caller_cli)/g"); > avp_delete("$avp(s:clir)/g"); > } > > ######################################################################## > # Request route 'clear-usr-preferences-callee' > ######################################################################## > route[2] > { > xlog("L_DBG", "Clear callee preferences - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > avp_delete("$avp(s:callee_fr_inv_timer)/g"); > avp_delete("$avp(s:cfu)/g"); > avp_delete("$avp(s:cfc)/g"); > } > > ######################################################################## > # Request route 'usr-preferences-caller' > ######################################################################## > route[3] > { > route(1); > xlog("L_DBG", "Load caller preferences for uuid > '$avp(s:caller_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > # load caller avps > avp_db_load("$avp(s:caller_uuid)", "*"); > avp_copy("$avp(s:cli)", "$avp(s:caller_cli)/d"); > if(is_avp_set("$avp(s:clir)/n") && avp_check("$avp(s:clir)", > "eq/i:1")) > { > # mark for anonymization > setflag(28); > } > } > > ######################################################################## > # Request route 'usr-preferences-callee' > ######################################################################## > route[4] > { > xlog("L_DBG", "Load callee preferences for uuid > '$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > # load callee avps > avp_db_load("$avp(s:callee_uuid)", "*"); > if(is_avp_set("$avp(s:cfu)/s")) > { > xlog("L_DBG", "Call-forward-unconditional to '$avp(s:cfu)' > found - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > avp_delete("$avp(s:caller_uuid)/g"); > avp_copy("$avp(s:callee_uuid)", "$avp(s:caller_uuid)/d"); > avp_pushto("$ru", "$avp(s:cfu)"); > > route(3); > route(10); > exit; > } > if(is_avp_set("$avp(s:ringtimeout)/n")) > { > xlog("L_DBG", "Setting ring timeout to $avp(s:ringtimeout) > secs - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > avp_copy("$avp(s:ringtimeout)", > "$avp(s:callee_fr_inv_timer)/d"); > } > } > > ######################################################################## > # Request route 'clir' > ######################################################################## > route[5] > { > if(isflagset(28) && !isflagset(27)) > { > setflag(27); > > xlog("L_DBG", "Anonymize caller - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > > uac_replace_from("Anonymous","sip:anonymous@anonymous.invalid"); > if(is_present_hf("Privacy")) > { > remove_hf("Privacy"); > } > append_hf("Privacy: id\r\n"); > } > } > > ######################################################################## > # Request route 'stop-rtp-proxy' > ######################################################################## > route[6] > { > if(isflagset(22)) > { > end_media_session(); > } > } > > ######################################################################## > # Request route 'base-route-register' > ######################################################################## > route[7] > { > sl_send_reply("100", "Trying"); > if(!www_authorize("", "subscriber")) > { > xlog("L_DBG", "Register authentication failed - M=$rm > RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); > www_challenge("", "0"); > exit; > } > if(!check_to()) > { > > xlog("L_DBG", "Spoofed To-URI detected - M=$rm RURI=$ru > F=$fu T=$tu IP=$si ID=$ci\n"); > sl_send_reply("403", "Spoofed To-URI Detected"); > exit; > } > consume_credentials(); > if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) > { > fix_nated_register(); > setbflag(6); > } > > if(!save("location")) > { > xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > sl_reply_error(); > exit; > } > xlog("L_DBG", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu > IP=$si ID=$ci\n"); > exit; > } > > ######################################################################## > # Request route 'base-outbound' > ######################################################################## > route[8] > { > if(is_present_hf("P-Asserted-Identity")) > { > remove_hf("P-Asserted-Identity"); > } > if(is_present_hf("Remote-Party-ID")) > { > remove_hf("Remote-Party-ID"); > } > if(is_avp_set("$avp(s:caller_cli)/s")) > { > if(!isflagset(28)) > { > xlog("L_DBG", "Set caller CLI '$avp(s:caller_cli)' > - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > append_hf("P-Asserted-Identity: > <$avp(s:caller_cli)>\r\n"); > } > } > > route(5); > if(isbflagset(6)) > { > if(!isflagset(22) && !search("^Content-Length:[ ]*0")) > { > setflag(22); > # use_media_proxy(); > } > t_on_reply("2"); > } > else > { > t_on_reply("1"); > } > if(!isflagset(21)) > { > t_on_failure("2"); > } > if(isflagset(29)) > { > append_branch(); > } > if(is_present_hf("Proxy-Authorization")) > { > # consume_credentials(); > } > > xlog("L_DBG", "Request leaving server, D-URI='$du' - M=$rm RURI=$ru > F=$fu T=$tu IP=$si ID=$ci\n"); > # no 100 (we already sent it) and no DNS blacklisting > if(!t_relay("0x05")) > { > sl_reply_error(); > if(is_method("INVITE") && isbflagset(6)) > { > end_media_session(); > } > } > exit; > } > > ######################################################################## > # Request route 'base-route-invite' > ######################################################################## > route[9] > { > sl_send_reply("100", "Trying"); > if(from_gw()) > { > $avp(s:caller_uuid)="0"; > xlog("L_DBG", "Call from PSTN' - M=$rm RURI=$ru F=$fu T=$tu > IP=$si ID=$ci\n"); > setflag(23); > $avp(s:ringtimeout)=1000; > > route(3); > if (nat_uac_test("3")) > { > xlog("L_DBG", "Inside NATed INVITE Processing\n"); > fix_nated_contact(); > > fix_nated_sdp("3"); > > t_on_reply("1"); > } > else > { > t_on_reply("1"); > } > > if(nat_uac_test("19")) > { > setbflag(6); > } > > route(51); > } > else > { > if(!proxy_authorize("", "subscriber")) > { > xlog("L_DBG", "Proxy authentication failed - M=$rm > RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); > proxy_challenge("", "0"); > exit; > } > if(!check_from()) > { > xlog("L_DBG", "Spoofed From-URI detected - M=$rm > RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); > sl_send_reply("403", "Spoofed From-URI Detected"); > exit; > } > > route(3); > if(nat_uac_test("19")) > { > fix_nated_contact(); > setbflag(6); > } > route(10); > } > } > > ######################################################################## > # Request route 'invite-find-callee' > ######################################################################## > route[10] > { > if(alias_db_lookup("dbaliases")) > { > xlog("L_DBG", "Callee was aliased - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > } > > route(2); > if(!is_domain_local("$rd")) > { > setflag(20); > $avp(s:callee_uuid) = "0"; > route(12); > } > avp_delete("$avp(s:callee_uuid)"); > avp_db_query("select uuid from subscriber where username = '$rU'", > "$avp(s:callee_uuid)"); > if(is_avp_set("$avp(s:callee_uuid)/s")) > { > xlog("L_DBG", "Callee is local, uuid='$avp(s:callee_uuid)' > - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > route(11); > } > else > { > $avp(s:callee_uuid) = "0"; > xlog("L_DBG", "Callee is not local - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > # > # *86 - sending user direct to access voicemail from sip > client > # > if($rU=="*86") > { > route(12); > } > else > { > if($(rU{s.len})<=7) > { > route(11); > } > else > { > route(12); > } > } > } > exit; > } > > ######################################################################## > # Request route 'invite-to-internal' > ######################################################################## > > route[11] > { > route(4); > > > revert_uri(); > seturi("sip:*89@openser.org"); > > rewritehostport("voicemailserver.databasepbx.com:5060"); > route(8); > exit; > } > > ######################################################################## > # TEST ROUTE 51 > ######################################################################## > route[51] > { > if(alias_db_lookup("dbaliases")) > { > xlog("L_DBG", "Callee was aliased - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > } > > route(2); > if(!is_domain_local("$rd")) > { > setflag(20); > $avp(s:callee_uuid) = "0"; > route(12); > } > avp_delete("$avp(s:callee_uuid)"); > avp_db_query("select uuid from subscriber where username = '$rU'", > "$avp(s:callee_uuid)"); > if(is_avp_set("$avp(s:callee_uuid)/s")) > { > xlog("L_DBG", "Callee is local, uuid='$avp(s:callee_uuid)' > - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > route(52); > } > else > { > $avp(s:callee_uuid) = "0"; > xlog("L_DBG", "Callee is not local - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > route(12); > } > exit; > } > > ######################################################################## > #TEST 52 > ######################################################################## > route[52] > { > route(4); > if(!lookup("location")) > { > xlog("L_DBG", "Local user offline - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > sl_send_reply("404", "User Offline"); > } > else > { > xlog("L_DBG", "Local user online - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > avp_delete("$avp(s:callee_srcip)"); > avp_db_query("select received from location where username > = '$rU'", "$avp(s:callee_srcip)"); > avp_printf("$avp(i:13)", "$(du{s.select,1,:})" ); > avp_printf("$avp(i:14)", > "$(avp(s:callee_srcip){s.select,1,:})" ); > xlog("L_DBG", "source IP [$avp(i:13)] destination IP > [$avp(i:14)]\n"); > if (avp_check("$avp(i:13)","eq/$avp(i:14)/i")) > { > xlog("L_INFO", "Detected Two Clients Behind the > Same NAT - Disabling Mediaproxy"); > # Do not use mediaproxy as the clients seem to be > behind the same NAT > xlog("L_DBG", "source IP is the same as > destination IP \n"); > resetflag(2); > resetflag(3); > } > avp_delete("$avp(i:13)/g"); > avp_delete("$avp(i:14)/g"); > route(8); > } > exit; > } > > ######################################################################## > # Request route 'invite-to-external' > ######################################################################## > route[12] > { > if(isflagset(20)) > { > xlog("L_DBG", "Call to foreign domain - M=$rm RURI=$ru > F=$fu T=$tu IP=$si ID=$ci\n"); > route(8); > exit; > } > if(!isflagset(23)) > { > if(uri =~ "^sip:\*86@.*") > { > xlog("L_DBG", "User accessing Voicemail - M=$rm > RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); > rewritehostport(" > voicemailserver.databasepbx.com:5060"); > route(8); > } > # don't allow calls relaying from PSTN to PSTN, if not > explicitely forwarded > else if(uri =~ "^sip:[0-9]+@") > { > # only route numeric users to PSTN > if(!load_gws()) > { > xlog("L_ERR", "Error loading PSTN gateways > - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > sl_send_reply("503", "PSTN Termination > Currently Unavailable"); > exit; > } > if(!next_gw()) > { > xlog("L_ERR", "No PSTN gateways available - > M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > sl_send_reply("503", "PSTN Termination > Currently Unavailable"); > exit; > } > setflag(21); > t_on_failure("1"); > route(8); > } > } > > xlog("L_DBG", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu > IP=$si ID=$ci\n"); > sl_send_reply("404", "User Not Found"); > exit; > } > > ######################################################################## > # Request route 'base-route-local' > ######################################################################## > route[13] > { > t_on_reply("1"); > if(t_check_trans()) > { > xlog("L_DBG", "Request leaving server - M=$rm RURI=$ru > F=$fu T=$tu IP=$si ID=$ci\n"); > } > else > { > xlog("L_DBG", "Dropping mis-routed request - M=$rm RURI=$ru > F=$fu T=$tu IP=$si ID=$ci\n"); > exit; > } > if(!t_relay()) > { > sl_reply_error(); > } > exit; > } > > > ######################################################################## > # Request route 'base-route-generic' > ######################################################################## > route[14] > { > xlog("L_DBG", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu > IP=$si ID=$ci\n"); > sl_send_reply("501", "Method Not Supported Here"); > exit; > } > > ######################################################################## > # Request route 'base-filter-failover' > ######################################################################## > route[15] > { > if(!t_check_status("408|500|503")) > { > xlog("L_DBG", "No failover routing needed for this response > code - M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > route(6); > exit; > } > } > > > ######################################################################## > #Send the call out > ######################################################################## > > route[16] > { > #send the call outward > if(method=="INVITE" && !isflagset(10)) > { > t_on_failure("2"); > } > > if (!t_relay()) > { > xlog("L_WARN", "ERROR: t_relay failed"); > sl_reply_error(); > } > } > > ######################################################################## > # Request route 'Voicemail and Asterisk for further processing' > ######################################################################## > failure_route[3] > { > if(!t_was_cancelled()) > { > revert_uri(); > rewritehostport("voicemailserver.databasepbx.com:5060"); > append_branch(); > #PREVENT SOME CRAZY VOICEMAIL LOOP > xlog("L_DBG", "No Answer Sending the call to Voicemail - > M=$rm RURI=$ru F=$fu T=$tu IP=$si > ID=$ci\n"); > setflag(10); > route(8); > } > } > > ######################################################################## > # Reply route 'base-standard-reply' > ######################################################################## > onreply_route[1] > { > xlog("L_DBG", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); > if (nat_uac_test("1")) > { > fix_nated_contact(); > } > exit; > } > > ######################################################################## > # Reply route 'base-nat-reply' > ######################################################################## > onreply_route[2] > { > xlog("L_DBG", "NAT-Reply - S=$rs D=$rr F=$fu T=$tu IP=$si > ID=$ci\n"); > if(nat_uac_test("1")) > { > fix_nated_contact(); > xlog("L_DBG", "NAT-Reply==[Fix_nated_contact] - S=$rs D=$rr > F=$fu T=$tu IP=$si ID=$ci\n"); > } > > if(isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") > { > if(!search("^Content-Length:[ ]*0")) > { > use_media_proxy(); > xlog("L_DBG", "NAT-Reply==[Force Rtp proxy] - S=$rs > D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); > } > } > exit; > } > > ######################################################################## > # Failure route 'pstn-failover' > ######################################################################## > failure_route[1] > { > xlog("L_DBG", "Failure route for PSTN entered - M=$rm RURI=$ru > F=$fu T=$tu IP=$si ID=$ci\n"); > route(15); > if(!next_gw()) > { > xlog("L_ERR", "Failed to select next PSTN gateway - M=$rm > RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); > route(6); > exit; > } > > t_on_failure("3"); > route(8); > } > > ######################################################################## > # Failure route 'base-standard-failure' > ######################################################################## > failure_route[2] > { > if(!t_was_cancelled()) > { > revert_uri(); > route(10); > } > if(t_check_status("422|481|487")) > { > xlog("L_DBG", "Final reply - M=$rm RURI=$ru F=$fu T=$tu > IP=$si ID=$ci\n"); > route(6); > exit; > } > > t_on_branch("1"); > xlog("L_DBG", "Redirect from UAC intercepted - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > route(10); > if($avp(s:cfc) != NULL) > { > avp_pushto("$ru", "$avp(s:cfc)"); > setflag(29); > append_branch(); > > t_on_branch("1"); > xlog("L_DBG", "CFC detected - M=$rm RURI=$ru F=$fu T=$tu > IP=$si ID=$ci\n"); > route(10); > } > > route(15); > route(6); > } > > ######################################################################## > # Branch route 'cfc-drop-local' > ######################################################################## > branch_route[1] > { > if(is_domain_local("$rd")) > { > xlog("L_DBG", "Dropping local branch - M=$rm RURI=$ru F=$fu > T=$tu IP=$si ID=$ci\n"); > drop(); > } > } > > =============================================================================================== > > MediaProxy's config.ini > > ; > ; Configuration file for MediaProxy > ; > > [Relay] > relay_ip = 192.168.94.30 > dispatchers = 192.168.94.30 > port_range = 8000:20000 > log_level = DEBUG > on_hold_timeout = 1800 > stream_timeout = 90 > > [Dispatcher] > socket = /var/run/mediaproxy/dispatcher.sock > listen = 192.168.94.30 > listen_management = 192.168.94.30 > log_level = DEBUG > management_use_tls = no > > [TLS] > certs_path = tls > verify_interval = 300 > > [OpenSIPS] > max_connections = 100 > > Is there any problem in config file then please suggest me. This > configuration was running with other phone but not with * > PolycomSoundPointIP-SPIP_600-UA/3.1.3.0439*. > > > Thanks in advance > -- > Regards, > > Chandrakant Solanki > > On Wed, Jan 6, 2010 at 3:38 PM, Ramu <contactramu(a)gmail.com> wrote: > >> Hi, >> >> You have to do NATing if the UAs are behind NAT when REGISTER and INVITE >> packet comes to Kamailio. >> >> You have to call use_media_proxy() function for INVITE and 200 OK >> messages. >> >> Please send your kamailio.cfg file to resolve the issues. >> >> >> On Wed, Jan 6, 2010 at 11:12 AM, Chandrakant Solanki < >> solanki.chandrakant(a)gmail.com> wrote: >> >>> Hello >>> >>> I have installed kamailio - 1.5.2 with MediaProxy. >>> >>> I registered 101(a)domain.com and 102(a)domain.com from * >>> PolycomSoundPointIP-SPIP_600-UA/3.1.3.0439*. >>> >>> But when I look for location table into database, I show received field >>> is NULL, while from other phone like X-Lite it shows some value in same >>> field like sip:X.X.X.X:5061. >>> >>> When I call from 101 to 102 using *PolycomSoundPointIP *phone, The phone >>> rings on the other extension but when I pick up no voice is heard, the >>> caller phone keeps hearing the wait-for-other party. >>> >>> Is there any changes required into kamailio.cfg or not. ?? >>> >>> Thanks in advance >>> >>> -- >>> Regards, >>> >>> Chandrakant Solanki >>> >>> _______________________________________________ >>> Kamailio (OpenSER) - Users mailing list >>> Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>> >> >> >> Thanks, >> Ramu >> > > -- Ramu

1 0

[Kamailio-Users] [kamailio-users] Kamailio - Polycom Sound Point
by Chandrakant Solanki 06 Jan '10

06 Jan '10

Hello I have installed kamailio - 1.5.2 with MediaProxy. I registered 101(a)domain.com and 102(a)domain.com from * PolycomSoundPointIP-SPIP_600-UA/3.1.3.0439*. But when I look for location table into database, I show received field is NULL, while from other phone like X-Lite it shows some value in same field like sip:X.X.X.X:5061. When I call from 101 to 102 using *PolycomSoundPointIP *phone, The phone rings on the other extension but when I pick up no voice is heard, the caller phone keeps hearing the wait-for-other party. Is there any changes required into kamailio.cfg or not. ?? Thanks in advance -- Regards, Chandrakant Solanki

2 1

[Kamailio-Users] memlog and mem_dbg : kamailio 3.0.0
by Ricardo Martinez 05 Jan '10

05 Jan '10

Hello. I have a doubt about the memlog and memdbg log levels. I have in my configuration file the next lines : debug=2 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) log_facility=LOG_LOCAL0 children=4 port=5060 memdbg=0 memlog=0 I was expecting not to see any memory debugs, instead I have this lines in the log file : Jan 5 16:20:14 - /usr/local/sbin/kamailio[16672]: WARNING: <core> [mem/q_malloc.c:406]: qm_malloc(0x826db60, 8) returns address 0x82f9408 frag. 0x82f93f0 (size=8) on 1 -th hit Jan 5 16:20:14 - /usr/local/sbin/kamailio[16672]: WARNING: <core> [mem/q_malloc.c:428]: qm_free(0x826db60, 0x82f9408), called from nathelper: nathelper.c: nh_timer(3252) Jan 5 16:20:14 - /usr/local/sbin/kamailio[16672]: WARNING: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x82f93f0 alloc\'ed from nathelper: nathelper.c: nh_timer(3143) Is this ok? How can I remove such debug? Regards, Ricardo Martinez

2 1

[Kamailio-Users] t_replicate in Kamailio-3.0.0 ?
by Santiago Gimeno 05 Jan '10

05 Jan '10

Hi, I'm trying to use t_replicate in K-3.0.0 with no luck. t_replicate("sip:192.168.1.1:5060"); The logs report "unknown command" and by looking at the documentation of the TM module I've seen that t_replicate is no longer in the TM module. Is this correct? Any workaround? Thanks in advance. Best regards, Santi

2 3

[Kamailio-Users] odd query
by Anders 05 Jan '10

05 Jan '10

Hi, Using SIREMIS, there is a column called dst_domain in both the acc and cdrs tables. This is supposedly the ID of the gw used for the call. However, this seems to be messed up when there is more than one gw that has the same IP. I have gws with the same IP but different prefixes, but then it just seems to put into dst_domain whatever the ID is of the first gw with that IP. Not so good, since I need the info of which gw used to be able to find out how much the call cost me. Anyone knows about how to handle this? Is there a fix, hatch or work-around? Thanks!

1 0

[Kamailio-Users] multiple kamctl configurations
by Klaus Darilion 04 Jan '10

04 Jan '10

Hi! I have several Kamailio servers on a single server running. Is there a way to tell kamctl to use a different configuration (e.g. for fifo)? regards klaus

2 1

[Kamailio-Users] re-invite mid call?
by Antonio Goméz Soto 04 Jan '10

04 Jan '10

Hi, maybe I should go to the -dev list for this, but thought I'd ask here first. Is it possible to let openSER send a SIP reinvite in the middle of a call, to redirect the sound stream to some rtpproxy? Maybe using the MI? Or should OpenSER be modified to do this? Thanks, Antonio

5 25

[Kamailio-Users] Kamailio 3.0.0 - Jan 11, 2010
by Daniel-Constantin Mierla 04 Jan '10

04 Jan '10

Hello, it was a bit longer way than usual with the testing, but the achievements pay the bill - you can run kamailio and ser modules at the same time, as well as most of the core features from both applications. I propose to release Kamailio (OpenSER) 3.0.0 on 11th of January, by that time we should fix the small issues discussed on the devel list lately. If you are aware of other issues, report them to: sr-dev [at] lists.sip-router.org I encourage everyone to contribute to wiki pages for what is new and migration from 1.5.x: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.0.x http://www.kamailio.org/dokuwiki/doku.php/install:1.5.x-to-3.0.0 To get it from GIT and test, see: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git I started to detail some of the new features in blog posts, see ToC at: http://bit.ly/4qy9cn A great 2010 for kamailians and sip router users! Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Serusers] problems when starting ser
by "Martínez García, Rebeca" 04 Jan '10

04 Jan '10

Hello all, I have installed ser in Ubuntu Karmic by means of Synaptic without any problems or errors during the installation. But before starting to configure the ser.cfg, I decided firstly to try the configuration by default and I got this error (I only uncommented the "log_stderror=yes"): cng@cng:/etc/ser$ ser D E Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 10.1.3.207 [10.1.3.207]:5060 udp: 192.168.50.10 [192.168.50.10]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 10.1.3.207 [10.1.3.207]:5060 tcp: 192.168.50.10 [192.168.50.10]:5060 Aliases: tcp: cng.local:5060 tcp: cng:5060 tcp: localhost:5060 udp: cng.local:5060 udp: cng:5060 udp: localhost:5060 0(2956) init_tcp: using epoll_lt as the io watch method (auto detected) cng@cng:/etc/ser$ 0(2958) Maxfwd module- initializing 0(2958) ERROR: fifo_server.c:857: Can't create FIFO: No such file or directory (mode=384) 0(2958) ERROR: ctl: mod_init: init ctrl. sockets failed 0(2958) init_mod(): Error while initializing module ctl ERROR: error while initializing modules It seems that it starts ok but suddenly finds an error and stops. However if I run: cng@cng:/etc/ser$ sudo /etc/init.d/ser start Starting ser: serListening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 10.1.3.207 [10.1.3.207]:5060 udp: 192.168.50.10 [192.168.50.10]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 10.1.3.207 [10.1.3.207]:5060 tcp: 192.168.50.10 [192.168.50.10]:5060 Aliases: tcp: cng.local:5060 tcp: cng:5060 tcp: localhost:5060 udp: cng.local:5060 udp: cng:5060 udp: localhost:5060 0(3016) init_tcp: using epoll_lt as the io watch method (auto detected) . 0(3018) unable to create pid file /var/run/ser/ser.pid: No such file or directory It only complains about the ser.pid file. I am a little bit confused. Where is the problem? Thanks in advance, Rebeca Martinez

3 4

[Kamailio-Users] New install woes - 1.5.3 w/ mysql
by Graham Wooden 02 Jan '10

02 Jan '10

Hi all, Apparently I am missing something fundamentally here; trying to get mysql functions working with the kamdbctl command, but it seems that while it locates the kamdbctl.mysql file, it can¹t locate/use the the locate_tool() function from the kamctl.base¹ (which is called at the top of kamdbctl.mysql). I have been going back through trunk/INSTALL file and not seeing anything that is standing out. Neither when I ³strace² the kamdbctl create¹ command. But seems that my environment is missing something. I don¹t recall such issues with my production 1.4.2 machine. It¹s a fresh install CentOS 5.4 with most items (e.g. mysql server/client) installed via the standard yum repo¹s. I tried 1.5.3 from both building it from source as as well as the linux/i386 binary set. Still the same results. Thoughts? [root@misfits kamailio]# kamctl start database engine 'MYSQL' loaded Control engine 'FIFO' loaded INFO: Starting Kamailio : INFO: started (pid: 9006) [root@misfits kamailio]# kamdbctl create /usr/local/lib/kamailio/kamctl/kamdbctl.mysql: line 25: locate_tool: command not found error: 'mysql' tool not found: set MYSQL variable to correct tool path

2 5

[Kamailio-Users] A great 2010 for everyone!
by Daniel-Constantin Mierla 31 Dec '09

31 Dec '09

Hello, before starting the party, I want to wish everyone all the best for 2010. Many thanks for those helping during 2009 with Kamailio and SIP Router projects, with code, documentation, advocating and support on mailing list. It was a lot of work done and I am confident you will enjoy Kamailio 3.0.0 and its great improvements and new features. I started a series of posts trying to highlight the best of it from my point of view, link to first is here: http://by-miconda.blogspot.com/2009/12/best-of-new-in-kamailio-300-1-includ… Not finished, expect more before Kamailio 3.0.0 is out, planned to happen before mid of January 2010. The draft of new features is available on wiki: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.0.x A fantastic night for all! Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Kamailio-Users] avpops OR sqlops
by alex pappas 31 Dec '09

31 Dec '09

Hi all, I'm trying to make an sql query through my kamailio script in order to check for number portability. i have a table with 2 col (phonenumber,prefix). I'm trying both modules avpops and sqlops but I'm not able to retreive any data. modparam("avpops", "db_url","mysql://openser:openserrw@localhost/openser") avp_db_query("SELECT prefix FROM ported WHERE phonenumber='$rU'","$avp(s:lnp_prefix)") ;The query run perfectly in command line. The return code is always 1 but when I print $avp(s:lnp_prefix) I get no result. I have 2 questions : 1. Which is the best way to check portability in my case 2. What I'm doing wrong and I'm not getting any result from the query. Thank you ALL Alex

5 46

[Kamailio-Users] 1.5.3 dialog module: dialogs stay in database when in "shutdown only" mode
by Nathan Angelacos 30 Dec '09

30 Dec '09

Hi, It appears that in "shutdown only" mode (modparm db_mode = 3 ) dialogs are not cleared from the database on startup: 1. kamctl start initiate call 2. kamctl stop 3. kamctl start terminate call (dialog is terminated correctly) 4. kamctl db show dialog shows the dialog record 5. kamctl stop 6. kamctl db show dialog shows the (now very dead) dialog record Perhaps the dialog table should be cleared right after step 3 or at the beginning of step 5 when the dialog module is in "shutdown only" mode.

2 1

[Kamailio-Users] Problems with Sip Redirect Client
by Daniel Cardoso Alves 30 Dec '09

30 Dec '09

Hello... I have a SIP Redirect Serve with a kamailio and it works fine with others system(NexTone, Asterisk, etc), but I have a problem with a SIP redirect client with Kamailio. I am using http://kamailio.org/docs/modules/1.5.x/uac_redirect.html to try manipulate the 3xx message. I would like to send one destiny at a time, but the get_redirects("10:2","redirect") send only 2 contacts and at teh same time. I get from my SIP Redirect Server this message: SIP/2.0 301 Moved Permanently..Via: SIP/2.0/UDP 2XX.XX.XXX.XXX:5080;branch=z9hG4bK633e.a3e32ff2.0;rport=5080;received=2XX.XX.XXX.XXX..Via : SIP/2.0/UDP 192.168.0.133:7191;received=201.XX.X.XXX;branch=z9hG4bK-d87543-254f7f3a1058e232-1--d87543-;rport=7191..To: <sip:05433330000 @2XX.XX.XXX.XXX:5080>;tag=609b19d62bd6886602a6c902875525b0.c8b2..From: "1938"<sip:1938@2XX.XX.XXX.XXX:5080>;tag=187fec3c..Call-ID: 4959bc 623e556571@ZGVzZW4tcGMtMDguZ292b2lwLmNvbS5icg....CSeq: 1 INVITE..Contact: sip:55341#5433330000@192.168.0.2, q=1, sip:123#5433330000@1 92.168.0.3, q=2, sip:55341#5433330000@192.168.0.4, q=3, sip:55341#5433330000@192.168.0.5, sip:55341#5433330000@192.168.0.6, sip:553 41#5433330000(a)192.168.0.7, sip:55341#5433330000@192.168.0.8, sip:55341#5433330000@192.168.0.9, sip:55341#5433330000@192.168.0.10. .Server: Test Redirect..Content-Length: 0.... How I can configure to send fist to sip:55341#5433330000@192.168.0.10 and after if I not sucess it sends to sip:55341#5433330000@192.168.0.9, etc...? Thanks... ____________________________________________________________________________________ Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com

3 6

[Kamailio-Users] [kamailio-users] Kamailio 1.5.0 + MediaProxy 2.3.8
by Chandrakant Solanki 30 Dec '09

30 Dec '09

Hi Kamailio - 1.5.0 MediaProxy - 2.3.8 CentOS 5.2 - 2.6.18-92.el5PAE My kamailio is running on *192.168.X.Y* ** *Here is my /etc/mediaproxy/config.ini* ------------------------------------------------- ; Configuration file for MediaProxy ; [Relay] dispatchers = 192.168.X.Y relay_ip = <public ip> port_range = 8000:10000 log_level = DEBUG stream_timeout = 90 [Dispatcher] socket = /var/run/mediaproxy/dispatcher.sock listen = 192.168.X.Y listen_management = 192.168.X.Y log_level = DEBUG [TLS] certs_path = tls verify_interval = 300 [OpenSIPS] max_connections = 100 =========================================================== Start media-dispatcher / media-relay # media-dispatcher --no-fork # media-relay --no-fork When I see log it show dispatcher to relay and relay to dispatcher connected.. But when I call 101 to 102 it gives me error, as show below *Media Dispatcher error* error: No suitable relay found *Media Relay error* warning: Cannot use port pair 8684/8685 warning: Cannot use port pair 8686/8687 -- Regards, Chandrakant Solanki

3 2

Re: [Kamailio-Users] Caller and Callee behind the same NAT
by juergen＠glowka.de 29 Dec '09

29 Dec '09

On Tuesday 29 December 2009 01:55:27 juergen(a)glowka.de wrote: > Hello, > > I'm new here. So I say "hello" to everybody. > > Maybe, somebody can help me or push me in the right direction. > > If there are Caller and Callee behind the same NAT, so it should > be better, they connect directly (e.g. 192.168.x.1 <-->192.168.x.2), > instead of using the rtp-proxy (this causes a jitter noise) . > > > I'm using Kamailio 1.5.3 with RTPPROXY 1.2.1. > > Has anybody a code snip or a sample kamailio.cfg for me? > I would be thankful for it. Better than giving you the solution, try to guest it ... Think a little ... what have in common a REQUEST from UAC A to UAC B if they are behing THE SAME NAT ROUTER ? -- Raúl Alexis Betancor Santana Dimensión Virtual ____________________________________________ Dear Mr. Santana Thanks for trying to help. oh, I guess, they've got the same IP-Address ;-) I found this code-snip in : dokuwiki/doku.php/examples:caller-callee-behind-same-nat ______________________________________ else if ( isflagset(2) and isflagset(3) ) { log(1, "Both Clients are behind NAT"); # Store the destination domain into an AVP avp_printf("$avp(i:450)", "$dd"); if ( avp_check("i:450", "eq/$src_ip/g") ) { xlog("L_INFO", "2 Clients Behind the Same NAT - Disabling Mediaproxy"); # Do not use mediaproxy as the clients seem to be behind the same NAT resetflag(2); resetflag(3); } } _______________________________________ But this Code-snip contains many mistakes so I changed it to : _______________________________________ if ( isflagset(5) and isbflagset(6) ) { lookup("location"); log(1, "Both Clients are behind NAT"); # Store the destination domain into an AVP avp_printf("$avp(i:450)", "$dd"); if ( avp_check("$avp(i:450)", "eq/$src_ip/g") ) { xlog("L_INFO", "2 Behind the Same NAT - Disabling Mediaproxy"); # Do not use mediaproxy as the clients seem to be behind the same NAT resetflag(5); resetbflag(6); } } _______________________________________ This seems to work, but I still have got this looping, jittering feeping noise witch is getting louder and louder ... Two or three words I can talk, then it sounds, as if a group of Navajo-Indians gone attack me .... :-) Could it be the client? I use 2 PDAs, Windows Mobile, Portsip Ver. 2.6 Best Wishes, also for me Juergen Glowka

3 2

[Serusers] using ser to involve nat
by WangZhenzhong 29 Dec '09

29 Dec '09

Hi every body I am using ser to solve the nat problem, In my environment: invite A-->SER-->B 1. SER receive the invite message REQA from A(invite B) 2. SER judge the use B, if the B is in public network, will fix nat in the REQA message if judge the B is in private network , will not fix anything and I want to know, how to judge the B's network, when the SER receive the INVITE from A? Is there any examle code used in ser.cfg thank you very much 2009-12-29 姓名：王振中公司：艾诺通信系统(苏州)有限公司地址：苏州工业园区苏虹西路81号苏虹工业坊A8单元二楼

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[Kamailio-Users] Caller and Callee behind the same NAT
by juergen＠glowka.de 29 Dec '09

29 Dec '09

Hello, I'm new here. So I say "hello" to everybody. Maybe, somebody can help me or push me in the right direction. If there are Caller and Callee behind the same NAT, so it should be better, they connect directly (e.g. 192.168.x.1 <-->192.168.x.2), instead of using the rtp-proxy (this causes a jitter noise) . I'm using Kamailio 1.5.3 with RTPPROXY 1.2.1. Has anybody a code snip or a sample kamailio.cfg for me? I would be thankful for it. Best wishes Juergen Glowka _______________________________________________ Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

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Re: [Kamailio-Users] adding inbound and outbound routes seperately
by Klaus Darilion 28 Dec '09

28 Dec '09

please always Cc: the list.... On 23.12.2009 16:24, Rodney wrote: > Well could you explain how to actually get peering functioning > between Kamailio on inbound and outbound? > There is no "peering functioning" in kamailio. You self have to define how you want to peer. For example if you want to peer with somebody sending you inbound requests, you have to define at least: - number format - authentication (IP-based, TLS, digest) One you have defined those parameters, you can implement it, e.g. digest authentication or IP based authentication. The same for outbound route - you have to configure Kamailio so that outbound requests are sent in the proper format. And Kamailio can not register to any other SIP service. If you need such a feature, you better use Asterisk. regards klaus > > -----Original Message----- From: Klaus Darilion > [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, December 23, > 2009 8:49 AM To: Rodney Cc: users(a)lists.kamailio.org Subject: Re: > [Kamailio-Users] adding inbound and outbound routes seperately > > > On 22.12.2009 22:35, Rodney wrote: >> Ok, I have kamailio and siremis, and I need to know how to set up >> a separate in bound and outbound route. >> >> Peering with inbound and registering with outbound >> > > Kamailio is a sip proxy, thus it can not register to other SIP > proxy. > > regards klaus > >> Any ideas guys? >> >> R >> >> >> >> _______________________________________________ Kamailio (OpenSER) >> - Users mailing list Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

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[Kamailio-Users] Newbie questions
by Antonio Goméz Soto 28 Dec '09

28 Dec '09

Hi, I have used asterisk before, but now am looking at kamailio for scaling up. I've read 'Building Telephony Systems with OpenSER', but I still have lots of questions.... so, here it goes: - I could not find a way to reboot a phone remotely, can it send SIP NOTIFY to all phones? - does it have something like the Asterisk Manager Interface, so I can connect/create an operator panel for it? Does that already exist somewhere? Something like FOP where operators can transfer calls, and check availability? - according to the docs it is not a BTBUA. What does this mean? I know that asterisk is a BTBUA, but what can a BTBUA do that a SIP proxy cannot? Is it only the moh and conferences? - lastly, is there a good tutorial on load-balancing somewhere? Thanks, Antonio.

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[Serusers] About iptel 400 missing cookie
by Dennis.Cheng 28 Dec '09

28 Dec '09

Hi folks, I used a Draytek Vigor 2820PBX to register iptel, and I found 400 missing cookie issue. When my pbx send the reinvite to hold another side, iptel always responded pbx 400 missing cookie. But I checked invite Via field, its branch included magic cookie "z9hG4bK". I attached wireshark packets. BR Dennis

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[Kamailio-Users] Merry Christmas and a Happy New Year!
by Elena-Ramona Modroiu 25 Dec '09

25 Dec '09

Hi everybody, before you leave the computers and go on holidays, I want to thank you all for the good work and support of the Kamailio and SIP Router projects. I wish you a Merry Christmas and a Happy New Year! As the project name is a Hawaiian word - the "Kamailian" greeting is: "Mele Kalikimaka me ka Hau'oli Makahiki Hou!" How would it be in your language? May the new year bring you joy, peace, good health and prosperity! Enjoy the winter holidays! Ramona

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[Kamailio-Users] Transparent bridge mode
by Vikram Ragukumar 24 Dec '09

24 Dec '09

Hello All, I am trying to setup a test scenario, where i have Kamailio and rtpproxy running on one CentOS box (Server1) and i have Asterisk running on another CentOS box (Server2). Server1 has 2 NIC's eth0 and eth1 that are both assigned Public IP's. There is a transparent bridge br0 connecting eth0 and eth1 which also has its own Public IP. Finally eth0 on Server2 also has a Public IP. Server2 must be assigned a Public IP. My goal is to modify rtpproxy so that i can intercept packets traveling to Server2, process them and let them resume along their original path. I would like to know if there is another way of setting this up so that i dont use as many Public IP's ? Do any of you see a problem with this setup, things that may not work eventually, or any other concerns ? Thanks, Vikram.

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[Kamailio-Users] Server Internal Error with sample config file : Kamailio 3.0 rc3
by Ricardo Martinez 24 Dec '09

24 Dec '09

Hello list. I'm using the sample configuration that comes with Kamailio RC3. (version: kamailio 3.0.0-rc3 (i386/linux)). I have two IP phones registered in the proxy. If I call from one to another and the call is cancelled a "500 Server Internal Error" message is send back to the caller (please see the debug attached) by the proxy. This is not happening with Kamailio 1.4 or Opensips 1.6 (using the sample configurations). Could be some problem? 10.0.0.220 - Caller 10.0.0.211 - Proxy 10.0.0.152 - Callee U 10.0.0.220:5060 -> 10.0.0.211:5060 INVITE sip:5501122334@10.0.0.211:5060 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. Max-Forwards: 70. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO. Allow-Events: talk, hold, conference, LocalModeStatus. Contact: "5501234567" <sip:5501234567@10.0.0.220:5060;transport=udp>;+sip.instance="<urn:uuid: 00000000-0000-1000-8000-00085D1B6C7A>". Supported: gruu, path, timer, 100rel, replaces. User-Agent: Aastra 57i/2.5.2.1010. Content-Type: application/sdp. Content-Length: 285. . v=0. o=MxSIP 0 0 IN IP4 10.0.0.220. s=SIP Call. c=IN IP4 10.0.0.220. t=0 0. m=audio 3000 RTP/AVP 18 0 8 101. a=rtpmap:18 G729/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=silenceSupp:off - - - -. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 10.0.0.211:5060 -> 10.0.0.220:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. Server: kamailio (3.0.0-rc3 (i386/linux)). Content-Length: 0. Warning: 392 10.0.0.211:5060 "Noisy feedback tells: pid=9830 req_src_ip=10.0.0.220 req_src_port=5060 in_uri=sip:5501122334@10.0.0.211:5060 out_uri=sip:5501122334@10.0.0.152:5062 via_cnt==1". . U 10.0.0.211:5060 -> 10.0.0.152:5062 INVITE sip:5501122334@10.0.0.152:5062 SIP/2.0. Record-Route: <sip:10.0.0.211;lr=on>. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. Max-Forwards: 69. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO. Allow-Events: talk, hold, conference, LocalModeStatus. Contact: "5501234567" <sip:5501234567@10.0.0.220:5060;transport=udp>;+sip.instance="<urn:uuid: 00000000-0000-1000-8000-00085D1B6C7A>". Supported: gruu, path, timer, 100rel, replaces. User-Agent: Aastra 57i/2.5.2.1010. Content-Type: application/sdp. Content-Length: 285. . v=0. o=MxSIP 0 0 IN IP4 10.0.0.220. s=SIP Call. c=IN IP4 10.0.0.220. t=0 0. m=audio 3000 RTP/AVP 18 0 8 101. a=rtpmap:18 G729/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=silenceSupp:off - - - -. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 10.0.0.152:5062 -> 10.0.0.211:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. User-Agent: T20 9.41.0.70. Content-Length: 0. . U 10.0.0.152:5062 -> 10.0.0.211:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. Record-Route: <sip:10.0.0.211;lr=on>. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=1211757599. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. Contact: <sip:5501122334@10.0.0.152:5062>. User-Agent: T20 9.41.0.70. Allow-Events: talk,hold,conference,refer,check-sync. Content-Length: 0. . U 10.0.0.211:5060 -> 10.0.0.220:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. Record-Route: <sip:10.0.0.211;lr=on>. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=1211757599. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. Contact: <sip:5501122334@10.0.0.152:5062>. User-Agent: T20 9.41.0.70. Allow-Events: talk,hold,conference,refer,check-sync. Content-Length: 0. . U 10.0.0.220:5060 -> 10.0.0.211:5060 CANCEL sip:5501122334@10.0.0.211:5060 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. Max-Forwards: 70. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>. Call-ID: c8e55b952ba85701. CSeq: 13486 CANCEL. Supported: gruu, path. User-Agent: Aastra 57i/2.5.2.1010. Content-Length: 0. . U 10.0.0.211:5060 -> 10.0.0.152:5062 CANCEL sip:5501122334@10.0.0.152:5062 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. Max-Forwards: 69. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>. Call-ID: c8e55b952ba85701. CSeq: 13486 CANCEL. Content-Length: 0. . U 10.0.0.211:5060 -> 10.0.0.220:5060 SIP/2.0 200 canceling. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-cd 47. Call-ID: c8e55b952ba85701. CSeq: 13486 CANCEL. Server: kamailio (3.0.0-rc3 (i386/linux)). Content-Length: 0. Warning: 392 10.0.0.211:5060 "Noisy feedback tells: pid=9831 req_src_ip=10.0.0.220 req_src_port=5060 in_uri=sip:5501122334@10.0.0.211:5060 out_uri=sip:5501122334@10.0.0.211:5060 via_cnt==1". . U 10.0.0.152:5062 -> 10.0.0.211:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=1211757599. Call-ID: c8e55b952ba85701. CSeq: 13486 CANCEL. User-Agent: T20 9.41.0.70. Content-Length: 0. . U 10.0.0.152:5062 -> 10.0.0.211:5060 SIP/2.0 487 Request Terminated. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=1211757599. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. User-Agent: T20 9.41.0.70. Content-Length: 0. . U 10.0.0.211:5060 -> 10.0.0.152:5062 ACK sip:5501122334@10.0.0.152:5062 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.211;branch=z9hG4bKfee8.a17fc93.0. Max-Forwards: 69. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=1211757599. Call-ID: c8e55b952ba85701. CSeq: 13486 ACK. Content-Length: 0. . U 10.0.0.211:5060 -> 10.0.0.220:5060 SIP/2.0 500 I'm terribly sorry, server error occurred (1/TM). Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-cd 47. Call-ID: c8e55b952ba85701. CSeq: 13486 INVITE. Server: kamailio (3.0.0-rc3 (i386/linux)). Content-Length: 0. Warning: 392 10.0.0.211:5060 "Noisy feedback tells: pid=9830 req_src_ip=10.0.0.220 req_src_port=5060 in_uri=sip:5501122334@10.0.0.211:5060 out_uri=sip:5501122334@10.0.0.152:5062 via_cnt==2". . U 10.0.0.220:5060 -> 10.0.0.211:5060 ACK sip:5501122334@10.0.0.211:5060 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bKf9043df6290194056.e3af2c48f0fba0802. Max-Forwards: 70. From: "5501234567" <sip:5501234567@10.0.0.211:5060>;tag=dbaa8e4386. To: "5501122334" <sip:5501122334@10.0.0.211:5060>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-cd 47. Call-ID: c8e55b952ba85701. CSeq: 13486 ACK. User-Agent: Aastra 57i/2.5.2.1010. Content-Length: 0. . Ricardo.-

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[Kamailio-Users] kamailio/ sip-router at 26c3
by Henning Westerholt 24 Dec '09

24 Dec '09

Hi, I'll (like in the last years) participate in the annual congress of the CCC in Berlin next week - the 26c3. More informations can be found on their site: http://events.ccc.de/congress/2009/wiki/Welcome So if you are in Berlin as well in this week (not necessarly on this event) and are interested in meeting for some discussions, please drop me a mail. Best regards, Henning

1 0

[Serusers] How to remove npdi & rn parameters from INVITE messages?
by Frank Durda IV 24 Dec '09

24 Dec '09

I have another large carrier that I am trying to send calls to, and they have a requirement that you not send them the npdi and rn=nnnnnnnn parameters for all calls. If you do, the call handling will be botched and the call will fail, even if the LRN information is exactly correct. Apparently this carrier (who once owned the Death Star) is incapable of filtering or ignoring this information on their side, and the sending switch is incapable of not sending this information because they thought RFC 4694 said they MUST ALWAYS send npdi/rn on every SIP call. Of course, the RFC does not say this. Don't get me started on which party is dumber. Anyway, I was hoping that SER could act as diplomat or interpreter and with some rewrite-something() calls, SER could remove the npdi/rn, or alter them to the point that the receiving switch no longer recognized the parameters as rn/npdi and so happily does it's own thing, and the calls don't die. However, based on what documentation exists, none of the various rewrite functions appear to operate on this part of the INVITE message and replace doesn't appear to work on the first line of a message. So there seems to be a spot whose contents can't be reached or manipulated by the existing functions. Am I correct in thinking this? So, is there a way to do something with rn=NNNNNNNNNN and npdi/npdi=yes parameters without having to write a custom function to do it? Thanks in advance!

2 1

[Serusers] How to drop a redundant/broken SIP reply message?
by Frank Durda IV 24 Dec '09

24 Dec '09

I have a large carrier that I am trying to send calls to, and their major brand SIP/ISUP switch (whose name begins with a "S") has a quirk where it emits a 183 Session Progress message with no SDP payload immediately after 100 Trying, and before ACM has actually been received. When ACM is received, it emits either a 180 Ringing with a SDP payload, or a 183 Session Progress with a SDP payload. So INVITE 100 Trying 183 Session Progress (no SDP payload) 180 Ringing (or 183 Session Progress) (with SDP payload) 200 OK (with SDP payload) is what you normally get, with the two 18x messages coming one or two seconds apart. However, if the called number is busy, you get 100 Trying 183 Session Progress (no SDP payload) 486 Busy Here What the calling party experiences for that is a second or two of ring-back commencing with the 183 Session progress, then a busy signal. That's very non-compliant. So far, the carrier has been unable to get this switch to stop emitting the first 183 Session Progress message. What I would like to do is have SER detect the redundant 183 Session Progress message (which I can already detect), and then have SER discard it, eg not send that one message back to whomever the INVITE came from. So far, I have been unable to get SER to do this. While "drop" is allowed in onreply_route sections, it is silently ignored, and you get a reply anyway. I tried putting a route(666) in the onreply_route section and putting the drop in that route["666"] section, but that had no obvious effect either, although some other functions can now be used in route["666"] that so far aren't solving the problem either. It occurred to me that I could change the 183 Session Progress of the offending message to something else in the "so-what" department, such as a "181 Your Call Is Being Forwarded", but "replace()" doesn't seem to work on the first line of replies, but can certainly alter other things in the reply message. There also doesn't seem to be any way (that actually works) to change the numeric SIP Response Code value to something else, even though clearly SER has it in the variable "status". I even thought of changing the port number or IP address via a rewrite command and send the message off to nowhere, but those don't work on replies or don't appear to. So, any suggestions on how to simply drop a chosen reply message, or misdirect it or malform it to the point that the call originator doesn't recognize it anymore? Yeah, kludgy but I only need it to work for two months. Thanks in advance!

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