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42 participants
37518 discussions

why a register error?
by Olvera, Victor 09 Jan '17

09 Jan '17

As I attempt to use X-lite I get a 504 error. On the CSCFs I get the following: pcscf 0(2066) INFO: <script>: [REGISTER] from [sip:bob@epc.mnc001.mcc001.3gppnetwork.org] to [sip:bob@epc.mnc001.mcc001.3gppnetwork.org] 0(2066) ERROR: <script>: REGISTER (sip:bob@epc.mnc001.mcc001.3gppnetwork.org (192.168.56.1:55880) to sip:bob@epc.mnc001.mcc001.3gppnetwork.org, 81563ZDg1Y2MwZDExYWJiYWJjNDZmNzY0ZWEwYjE1ODNkYmM) icscf 11(2150) INFO: cdp [receiver.c:932]: peer_connect(): peer_connect(): Peer hss.epc.mnc001.mcc001.3gppnetwork.org<http://mcc001.3gppnetwork.org>:3868 connected 0(2138) INFO: <script>: [REGISTER] from [sip:bob@epc.mnc001.mcc001.3gppnetwork.org] to [sip:bob@epc.mnc001.mcc001.3gppnetwork.org] 0(2138) ERROR: <script>: REGISTER (sip:bob@epc.mnc001.mcc001.3gppnetwork.org (192.168.56.76:4060) to sip:bob@epc.mnc001.mcc001.3gppnetwork.org, 81563ZDg1Y2MwZDExYWJiYWJjNDZmNzY0ZWEwYjE1ODNkYmM) 0(2138) INFO: ims_icscf [cxdx_uar.c:71]: create_uaa_return_code(): created AVP successfully : [uaa_return_code] 6(2145) INFO: ims_icscf [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_server_name: Failed finding avp 6(2145) INFO: ims_icscf [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_result_code: Failed finding avp 6(2145) INFO: ims_icscf [cxdx_uar.c:71]: create_uaa_return_code(): created AVP successfully : [uaa_return_code] 0 scscf 11(2047) WARNING: cdp [receiver.c:864]: peer_connect(): peer_connect(): Trying to connect to 192.168.56.78 port 3868 11(2047) INFO: cdp [receiver.c:932]: peer_connect(): peer_connect(): Peer hss.epc.mnc001.mcc001.3gppnetwork.org<http://mcc001.3gppnetwork.org>:3868 connected 9(2045) ERROR: cdp [peerstatemachine.c:635]: I_Snd_CER(): I_Snd_CER(): Error on finding local host address > Socket operation on non-socket Great regards, Victor Olvera victor.olvera(a)verizonwireless.com<mailto:victor.olvera@verizonwireless.com> mobile # 001-214-923-9405

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Re: [SR-Users] BYE issue kamailio + rtpproxy "481 call leg transaction doesn't exists".
by Nihar Ranjan Deb 09 Jan '17

09 Jan '17

Anyone any help on this. Regards, Nihar From: Nihar Ranjan Deb [mailto:nihar.ranjan@microtalkgroup.com] Sent: Wednesday, December 28, 2016 5:42 PM To: 'sr-users(a)lists.sip-router.org' Subject: BYE issue kamailio + rtpproxy "481 call leg transaction doesn't exists". Hi All, I am a new user of kamailio. We have setup a kamilio server with RTP proxy. Most of the features are working fine except BYE. Say User A and B are registered in kamailio server. A (caller) connected to B(called), call established. 1. If Caller (A) disconnects the call, everything goes fine. B gets disconnected and call releases. 2. But if Called party (B) disconnects the call, A never gets disconnected. I checked trace as well, in this case caller (A) gets a BYE with B's callid. So A rejects call with "481 call leg transaction doesn't exists". Basically A party and B party has separate dialog established with Kamilio. So they have two separate callids . All messages are exchanged correctly. Only when B party disconnects call, kamailio sends BYE to A party with B party callid. Please help me to resolve this issue. Do let me know if any more information required. ------ INVITE-------------------------- INVITE sip:966268@*** SIP/2.0 Via: SIP/2.0/UDP 192.168.1.36:63238;branch=z9hG4bK-524287-1---e64d171ec3b0b54e;rport Max-Forwards: 70 Contact: <sip:334031@***:8906;rinstance=ea9b5f48eb342a98> To: <sip:966268@***> From: "334031"<sip:334031@****>;tag=4406552c Call-ID: 82158NzljY2NiYTA5YmI3M2U1MzQyODgwMWYwYTI3ZWJjNzc CSeq: 2 INVITE Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE Content-Type: application/sdp Proxy-Authorization: Digest username="334031",realm="*****",nonce="****",uri="sip:966268@***",response=" ***",algorithm=MD5 Supported: replaces User-Agent: X-Lite release 4.9.6 stamp 82158 Content-Length: 334 v=0 o=- 13127393523589215 1 IN IP4 192.168.1.36 s=X-Lite release 4.9.6 stamp 82158 c=IN IP4 192.168.1.36 t=0 0 m=audio 57116 RTP/AVP 9 8 120 0 84 101 a=rtpmap:120 opus/48000/2 a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000 a=rtpmap:84 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv -------------------------BYE-------------------------------- BYE sip:334031@192.168.1.36:63238;rinstance=ea9b5f48eb342a98 SIP/2.0 Via: SIP/2.0/UDP ***:5060;branch=z9hG4bKa28b.a2ac8c26802cdcc8c619784643f82000.0 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKsr-HqsDUwx8-phRByVr-9VwUlbcUlPcIpKo-0noY0rjI95gmMhg10 qgY9mc-w5He0EJbTbkb3AZt2Yna6n4uPmqbJWEYwVKYlDTYlH* From: <sip:966268@***>;tag=1665200125 To: "334031" <sip:334031@***>;tag=4406552c Call-ID: !!:xePve9XKtEJ4NJJT-2hM9lsXe6RemqZPIRAa1PZf1wJ4DRx5ePRce0OwbHAY-RD4Hlya-HRwe MH4N6RemHYMI9Y0tV** CSeq: 3 BYE Contact: <sip:10.0.0.1;line=sr-mcsjIlJcYlxcIPVo-0-8Ylb8-9bKUlPjIlxEYwbw> Max-Forwards: 69 User-Agent: Yealink SIP-T20P 9.72.0.80 Content-Length: 0 -------------------------- 481--------------------- SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP ***:5060;branch=z9hG4bKa28b.a2ac8c26802cdcc8c619784643f82000.0;received=35.1 64.228.12 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKsr-HqsDUwx8-phRByVr-9VwUlbcUlPcIpKo-0noY0rjI95gmMhg10 qgY9mc-w5He0EJbTbkb3AZt2Yna6n4uPmqbJWEYwVKYlDTYlH* To: "334031" <sip:334031@***>;tag=4406552c From: <sip:966268@***>;tag=1665200125 Call-ID: !!:xePve9XKtEJ4NJJT-2hM9lsXe6RemqZPIRAa1PZf1wJ4DRx5ePRce0OwbHAY-RD4Hlya-HRwe MH4N6RemHYMI9Y0tV** CSeq: 3 BYE Accept-Language: en Content-Length: 0 Regards, Nihar

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VERY IMPORTANT: deciding when to remove the MI code
by Daniel-Constantin Mierla 06 Jan '17

06 Jan '17

Hello, we started discussing about removing MI (so called management interface) for very long time, more or less since 2008. The RPC should remain the control interface, given its better structure for commands, parameters, etc ... MI is custom protocol using a line-oriented communication via fifo or socket file with kamailio (e.g., implemented mi_fifo and mi_datagram modules). RPC is the alternative, a more standardized concept, with better structured format. I think it's time to set a clear roadmap for doing the removal. Overall, it will be easier to maintain the code, right now being duplicated code for doing the same operation over MI or RPC, and MI shows its limitations (or complexity to deal with) for advanced needs (see the discussions about how to provide multi-line value parameters over MI). So, I want to know if there are many relying on MI directly and they still want to keep it, what would be the expected duration they need for upgrading their tools to work with RPC interface, other relevant aspects people have in favour of mi vs rpc. I am even willing to do the removal in time befire freezing the 5.0 branch. We will ensure a clean start of 5.x series. The main concern from my point of view is kamctl -- but I think we can preserve the compatibility for kamctl commands and parameters (so command line execution of kamctl will be the same), but the output might be different. That's because it should be easy to updated it to communicate with jsonrpc-s module, but then it will get json-formatted results. To summarize, two big questions to answer: a) Are you ok to remove the MI code/commands? b) If yes to a), are you ok to be done for v5.0? Not providing feedback will be considered as 'yes' for both questions, so **speak up if you want MI to be kept or delay it removal**. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

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presentity tabla, faster update when drop TCP/TLS connection
by João Resende 05 Jan '17

05 Jan '17

I'm testing kamailio with a SIP client with presence support, when I get a connection drop from a client (i.e when the client lost the access to the Internet connection) I do not receive the change in the status to offline immediately, I just received when the timeout from the presentity database finishes. Exists a way to define this? or it is a bug? or should be a new feature? I know that I'm able to change the expiry parameter from the presentity table, but that increases my battery consumption and the data exchange, the same happens if I reduce the publish interval on the client side.

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Re: [SR-Users] using kamailio to forward ipv6 to ipv4
by anfecora 05 Jan '17

05 Jan '17

Yes i am planning on proxing media. The case is that client will only talk ipv6 but the registrar only have ipv4 connectivity. Therefore need to create a proxy to translate this dialogs. Thi thnking on using kamailio to receive the registration modify the pckets and forward registrations in ipv4. I was thinking in using the asterisk kamailio example but instead of asterisk behind will be an sbc. Therefore kamailio will listen in ipv6 and translate over to ipv4 including media of course via rtpproxy. On Jan 5, 2017 9:43 AM, "Frank Carmickle" <frank(a)carmickle.com> wrote: > On Jan 5, 2017, at 12:24 PM, anfecora <anfecora(a)gmail.com> wrote: > > Hi guys hope you have a great new year. > I would appreciate if anyone can point me in to the right direction . > > I need to build a proxy to translate from ipv6 to ipv4, but kamailio should not process the registers or invites, only translate from ipv6 to ipv4 and forward all registrations over. A bit more information about the situation would be helpful. What exactly are you trying to do? > is that possible? 6 to 4 translation is possible, but until we understand the situation I’m not sure we can be that helpful. Are these phones? If so are they dual stack? Is the underlying network dual stack? There’s a lot of work to do to get this right in a Kamailio config. You’ll need to run rtpengine to proxy the media if the endpoints are not dual stack. Who keeps the mapping of the v6 to v4 and v4 to v6 translations? How is that mapping determined? I’d recommend ds-lite if you are in a position to alter the network and the phones/systems allow for it. HTH —FC > thanks. > > > _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - 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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CDR - acc_cdrs
by Annus Fictus 05 Jan '17

05 Jan '17

Hello, I'm trying to save Kamailio CDR on acc_cdrs table without success. My relevant configuration look like: ----- #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 ----- #!define WITH_ACCDB ---- loadmodule "dialog.so" loadmodule "acc.so" modparam("dialog", "dlg_flag", 0) ---- # ----- acc params ----- modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_log_enable", 1) modparam("acc", "cdr_on_failed", 1) modparam("acc", "cdrs_table", "acc_cdrs") modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$td") #!endif Any hint is really appreciate. Regards

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unexpected char - status 6
by AL S 05 Jan '17

05 Jan '17

Hi, Kamailio is throwing error message from "parse_to.c" with status=6. The source code shows status 6 as URI_ENCLOSED. However, tcpdump of the call does not show any enclosed angle bracket is missing. (tcpdump pcap file is attached.) Kamailio log Jan 4 10:18:21 orl-ar-openser01 /usr/sbin/kamailio[2374]: ERROR: <core> [parser/parse_to.c:626]: ERROR: parse_to : unexpected char [#015] in status 6: <<<sip:2057030997@72.31.12.161;tag=gK04001cd3>> . FROM and TO in INVITE received by Kamailio From: <sip:7272697443@71.44.15.84>;tag=gK00d6c4e0 To: <sip:2057030997@72.31.12.161>;tag=gK04001cd3 FROM and TO in INVITE forward by Kamailio for the same call From: <sip:7272697443@71.44.15.84>;tag=gK00d6c4e0 To: <sip:2057030997@72.31.12.161\000\000>;tag=gK04001cd3 PS: this error message is happening on all 4 kamailio servers (3.3.0) PS: call failure only happens in re INVITE (call stays active longer than 15 min) Your help would be appreciated. Thanks, AS

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RTPProxy with VPN
by Rodrigo Moreira 05 Jan '17

05 Jan '17

Hello, I am having problems with RTPPROXY. I need it to work as a bridge. On one side will have a VPN tunnel called a tun0, on the other an eth0 interface. Is it possible for RTPPROXY to run correctly in this scenario? How can I test if RTPPROXY is running correctly? Is there any command? What would the configuration look like inside /etc/default/rtpproxy Please help me. Best regards -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

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Re: [SR-Users] Regarding usage of Redis Module in Kamailio core
by Vicente Hernando 05 Jan '17

05 Jan '17

Hello Hemanth, this minimal kamailio cfg file works for me to test ndb_redis module. #!KAMAILIO debug=4 log_stderror=yes children=4 fork=yes #disable_tcp=yes mpath="modules/" loadmodule "ndb_redis.so" # An already configured redis server is needed here modparam("ndb_redis", "server", "name=srv1;addr=127.0.0.1;port=6379;db=0") request_route { redis_cmd("srv1", "SET foo bar", "r"); exit; } Changing the parameters to suit your configuration, you can use it to debug your system. Regards, Vicente. On 01/05/2017 07:55 AM, hemanth(a)advaitamtech.com wrote: > > Hello Vicente, > > Thanks for your response, > > In response to your query, > > Yes i have Built ndb_redis module and i can see ndb_redis.so in /usr/local/lib64/kamailio/modules folder, I have installed hiredis depenency also, > > I have also modified the kamailio.cfg file loadmodule and modparam list, > > I feel like the way i am using the redis_cmd is wrong, I think i need to add it in route section of kamailio.cfg file, Can you please guide me regarding how to use redis db in order to update user information to DB on registration of each user. > > Thanks & Regards > Hemanth > > > Message: 8 > Date: Wed, 4 Jan 2017 14:41:42 +0530 (IST) > From: "hemanth(a)advaitamtech.com" <hemanth(a)advaitamtech.com> > To: sr-users(a)lists.sip-router.org > Subject: [SR-Users] Regarding usage of Redis Module in Kamailio core > Message-ID: <1483521102.18025940(a)apps.rackspace.com> > Content-Type: text/plain;charset=UTF-8 > > Hello, > > I am using Kamailio as a SIP proxy, > > I want to store Registered user information in Redis DB, > > I am able to configure REDIS DB, But i am not able to run the redis command from > > Kamailio core using the function > > "redis_cmd("srvN","SET key 1", "r")" > > I am getting undefined reference error as below > > "/usr/local/src/kamailio-4.4/kamailio/receive.c:208: undefined reference to `redis_cmd'" > > can anyone please help me to fix this issue > > Thanks & Regards > Hemanth > > > > > ------------------------------ > > Message: 9 > Date: Wed, 4 Jan 2017 10:50:05 +0100 > From: Vicente Hernando <vhernando(a)systemonenoc.com> > To: sr-users(a)lists.sip-router.org > Subject: Re: [SR-Users] Regarding usage of Redis Module in Kamailio > core > Message-ID: <b003d2aa-ddd0-574d-b9e3-9b91e44976f1(a)systemonenoc.com> > Content-Type: text/plain; charset=utf-8; format=flowed > > Hello Hemanth, > > just a quick response, I would check if you compiled ndb_redis module > correctly (also if you have hiredis library dependency). > > Also I would check if you loaded ndb_redis module in your configuration > file. > > > Regards, > Vicente. > > On 01/04/2017 10:11 AM, hemanth(a)advaitamtech.com wrote: >> Hello, >> >> I am using Kamailio as a SIP proxy, >> >> I want to store Registered user information in Redis DB, >> >> I am able to configure REDIS DB, But i am not able to run the redis command from >> >> Kamailio core using the function >> >> "redis_cmd("srvN","SET key 1", "r")" >> >> I am getting undefined reference error as below >> >> "/usr/local/src/kamailio-4.4/kamailio/receive.c:208: undefined reference to `redis_cmd'" >> >> can anyone please help me to fix this issue >> >> Thanks & Regards >> Hemanth >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > ------------------------------ > > Subject: Digest Footer > > _______________________________________________ > sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > ------------------------------ > > End of sr-users Digest, Vol 140, Issue 3 > **************************************** > > > >

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Kamailio cancel branch only after receiving replies
by Aqs Younas 05 Jan '17

05 Jan '17

Greetings list, I am forking a call to multiple destinations and want to keep the only branch which sends quicker first 180/183 reply and cancel the remaining branches. Below is my related cfg snippet. modparam("tm", "failure_reply_mode", 3) modparam("tm", "fr_timer", 30000) modparam("tm", "fr_inv_timer", 120000) modparam("tm", "cancel_b_method", 2) ... route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); append_branch(); append_branch(); route(RELAY); exit; } onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { xlog("L_INFO","Received $rs (IP:$si:$sp)\n"); if(status=~"18[03]"){ t_cancel_branches("others"); xlog("L_INFO","cancelled all other branches\n"); } route(NATMANAGE); } } ... But I see kamailio does not instantly send CANCEL to other branches after it has received 180/180 from any branch. Kamailio is sending CANCEL to other branches after they start sending 180/183 one by one. How can I cancel all other branches instantly? I am sure there is something wrongly configured in my configuration. Any suggestion is much appreciated. Best Regards.

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