sr-users May 2009

sr-users@lists.kamailio.org

59 participants
114 discussions

by Siddhardha Garige

Hello all, Just wondering if anyone know any tutorial on setting up HA+DRBD solution for kamailio. Especially creating partitions, DRBD devices and mount points. Thanks in advance, -Sid "May the light be with you." ______________________________________________ Siddhardha Garige www.luminepixels.com

8 years, 9 months

[Kamailio-Users] AudioCodes + Kamailio : Problem in SIP Message Headers

by Samuel Muller

Hello all, I recently add a classical Audiocodes Mediant 2000 with 2x 8E1, the purpose is to have several interconnections with PSTN. I configured it like this : Audiocodes registers as a gateway to the Kamailio, using a dedicated port (5062). Registration seems to be OK, and the pstn gw uses OPTIONS method to ping the proxy. I can attack the Audiocodes with a SIP phone behind Kamailio, no pbm. But the audiocodes returns some errors about SIP headers sent by Kamailio : ( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time: 12:30:26] ( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol '0' in scheme. ALPHA expected Here you have the example of an INVITE from a SIP phone to the PSTN : ** audiocodes debug ** 4d:12h:30m:26s ( lgr_flow)(44730 ) ---- Incoming SIP Message from 77.246.81.132:5060 ---- INVITE sip:0323719001@77.246.81.136:5062;transport=udp SIP/2.0 Record-Route: <sip:77.246.81.132;lr=on;ftag=71078b346a20fb3eo0;nat=yes> Via: SIP/2.0/UDP 77.246.81.132;branch=z9hG4bKdace.1ab1d59.0 Via: SIP/2.0/UDP 192.168.0.113:5060;rport=15170;received=77.246.81.162;branch=z9hG4bK-b432f96 From: "Sam" <sip:0123451010@sip.720.fr <sip%3A0123451010(a)sip.720.fr>>;tag=71078b346a20fb3eo0 To: <sip:0323719001@sip.720.fr <sip%3A0323719001(a)sip.720.fr>> Call-ID: 944d8aec-27503ee6(a)192.168.0.113 CSeq: 102 INVITE Max-Forwards: 49 Contact: "Sam" <sip:0123451010@77.246.81.162:15170> Expires: 240 User-Agent: Linksys/SPA941-5.1.8 Content-Length: 281 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER Supported: 100rel, replaces Content-Type: application/sdp P-Asserted-Identity: <0123451010> Remote-Party-ID: <0123451010>;party=caller;privacy=none;screen=yes v=0 o=- 26933860 26933860 IN IP4 192.168.0.113 s=- c=IN IP4 77.246.81.133 t=0 0 m=audio 35038 RTP/AVP 18 0 8 101 a=rtpmap:18 G729a/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv a=nortpproxy:yes ( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time: 12:30:26] ( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol '0' in scheme. ALPHA expected ( sip_stack)(44734 ) !! [ERROR] Message type: INVITE [Time: 12:30:26] ( sip_stack)(44735 ) !! [ERROR] Source header: [Time: 12:30:26] ( sip_stack)(44736 ) !! [ERROR] Line: 17. Column: 23 [Time: 12:30:26] The outgoing INVITE from Kamailio is exactly the same received by the AudioCodes. When I searched over Google, I just found 2 answers about Asterisk / Audiocodes unsolved problem, but no more informations. I supposed that the problem is as indicated : " s=- " where source is empty in place of "NULL" / "0" or something like this ... Someone can confirm or already met the problem ? Many thanks all :) .Sam.

11 years, 7 months

[OpenSER-Users] uaCSTA support in openser

by Hugo Koblmueller

high.all! i'm wondering if there is any support of uaCSTA in openser (planned)? i'm just working on the integration of asterisk (*) environment to OCS 2007 environment, having openSER in the middle (mainly for TCP/UDP translation and smoothing out the protocol deficienes on both sides). in this setup the * having the openSER in front is talking to the OCS (and vice versa) via the OCS mediation server, which is moreorless sending standard SIP messages, which enables normal softphone (integration to *) of the office communicator. this configuration is already working... now i'm planning to go for the CTI integration, where there is no OCS mediation server in between OCS and openSER, doing the translation of SIP/CSTA to SIP. i'm thinking about using openSER for this task, that's why i'm looking for a CSTA module or perl programm, which is capable of this functionality. afaik for the CTI communication there isn't the full complexity of CSTA needed, just a subset mainly for call setup and call clearing. anyone having experience on this topic? thx & cheers -hugo Great Ideas for Small Devices Hugo Koblmueller Senior Staff Engineer Software Development COMNEON electronic technology GmbH & Co. OHG Freistaedter Strasse 400 4040 Linz Austria hugo.koblmueller(a)comneon.com tel: fax: mobile: Skype ID: +43 (5) 1777 - 15730 <http://www.plaxo.com/click_to_call?lang=en&src=jj_signature&To=%2B43+%285%2 9+1777+%2D+15730&Email=hugo(a)koblmueller.com> +43 (5) 1777 - 15810 +43 (676) 82051280 <http://www.plaxo.com/click_to_call?lang=en&src=jj_signature&To=%2B43+%28676 %29+82051280&Email=hugo(a)koblmueller.com> drhookson Want to always have my latest info? <https://www.plaxo.com/add_me?u=21475050628&src=client_sig_212_1_banner_join &invite=1&lang=en> Want a signature like <http://www.plaxo.com/signature?src=client_sig_212_1_banner_sig&lang=en> this?

13 years, 2 months

[Kamailio-Users] tcp_blocking_connect related errors and Kamailio using 100% CPU

by Pascal Maugeri

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

14 years, 2 months

[Kamailio-Users] tm retransmits and times out even though reponse is received

by Alex Hermann

Hello, I'm seeing an increasing amount of spurious timeouts on my proxies. Even though the proxy receives a "100 Trying" response immediately after relaying the request, tm will send retransmits and times out eventually. Below is a commented log snippet. The timeout occurs on the proxy named 'balancer1'. Both are on the same host. I verified that the 100 Trying response is received in a t_on_reply block, so it is recognized by tm as belonging to a transaction. When needed, I'll provide a debug log when it occurs again. Meanwhile, any idea what is causing this? This is on kamailio 1.5+svn5768 fr_timer = 10 fr_timer_avp = 2 # Initial request received on balancer May 15 12:32:01 balancer1[24170]: [2 INVITE] - New request. <sip:uac@domain> => <sip:uas@domain> May 15 12:32:01 balancer1[24170]: [2 INVITE] Relaying # Initial request received on proxy May 15 12:32:01 proxy1[23803]: [2 INVITE] - New request. <sip:uac@domain> => <sip:uas@domain> # Balancer receives 100 response May 15 12:32:01 balancer1[24166]: [2 INVITE] - Reply. 100 Trying # Proxy relays request May 15 12:32:01 proxy1[23803]: [2 INVITE] Relaying # Proxy receives retransmission from balancer ??? May 15 12:32:02 proxy1[23804]: [2 INVITE] - New request. <sip:uac@domain> => <sip:uas@domain> # Balancer times out ??? May 15 12:32:02 balancer1[24171]: [2 INVITE] Failure (Status: 408) May 15 12:32:02 balancer1[24171]: [2 INVITE] No more gateways May 15 12:32:02 proxy1[23804]: [2 CANCEL] - New request. <sip:uac@domain> => <sip:uas@domain> -- Greetings, Alex Hermann

14 years, 4 months

[Kamailio-Users] uac_replace_from check

by jh＠tutpro.com

if i have understood correctly, if uac_replace_from() is called in "auto" mode, encoded from uri is added into an extra param of r-r header when the request is forwarded towards uas. does the module check that reply to the request from uas really includes the added parameter in its r-r header or is correct operation of in-dialog requests at the mercy of the uas? -- juha

14 years, 5 months

[Kamailio-Users] Load Balancing Asterisk using Kamailio

by Kurt Weasel

Hi, I searched around the web to load balance asterisk servers and found Kamailio for possible solution. Let's say I have two identical asterisk servers with same dialplan and configuration and I want both servers look like they have same IP address from clients. Based on my understanding, we need 1 Kamailio as load balancer and 2 Asterisk servers as the real servers. Let's say the setup is : Kamailio load balancer 192.168.2.1 Asterisk Server #1 192.168.2.2 Asterisk Server #2 192.168.2.3 My question is, X-Lite softphone Configuration should be set to domain 192.168.2.1, right? I also want to know the step by step configuration to set kamailio as load balancer. I have not used Kamailio before. However after searching the documentation, the step (based on my understanding) is somewhat like this : 1. Install Kamailio. I will use the step by step here : http://kamailio.org/dokuwiki/doku.php/install:kamailio-1.5.x-from-svn 2. Then, using dipatcher module, I will configure it using this guide here : http://kamailio.org/dokuwiki/doku.php/asterisk:load-balancing-and-ha Then modify the dispatcher.list file to match the IP address of my asterisk servers : *1 sip:192.168.2.2:5060 1 sip:192.168.2.3:5060* Am I missing some steps? Do I also need to configure dialplan or any other file at Kamailio load balancer? Or those two steps basically done it all for simple load balancing configuration? Thanks for your responses. Best Regard, Kurt Weasel.

14 years, 8 months

[Kamailio-Users] HOWTO rewrite to: header

by alexander merkulov

here is correct reqwest and reqwest via kamailio is there any way to rewrite To: field in kamailio because now it nto work correct reqwest(asterisk) RECEIVING FROM: 69.70.173.195:5060 CANCEL sip:34249@193.110.78.12:8484 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK4c97bd4a From: "Unknown" <sip:Unknown@voip1.bravotelecom.com<sip%3AUnknown(a)voip1.bravotelecom.com> >;tag=as366a0ba0 To: <sip:34249@193.110.78.12:8484> Call-ID: 322db75c71be4ba45a61c7666b93e6be(a)voip1.bravotelecom.com CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 incorrect reqwest(multi-homed kamailio, asterisk originate to internal eth) RECEIVING FROM: 69.70.173.195:5060 CANCEL sip:34249@193.110.78.12:8484 SIP/2.0 Via: SIP/2.0/UDP 69.70.173.195;branch=z9hG4bK1b1b.3c42cea6.0 From: "Unknown" <sip:Unknown@69.70.173.195 <sip%3AUnknown(a)69.70.173.195> >;tag=as6075df60 Call-ID: 4d784fa972c4e9c9795032417f46cb26(a)voip1.bravotelecom.com To: <sip:34249@192.168.2.170 <sip%3A34249(a)192.168.2.170>> CSeq: 102 CANCEL Max-Forwards: 70 asterisk Content-Length: 0 -- Merkulov Alexander

14 years, 8 months

Re: [Kamailio-Users] [LCR] About ping

by Iñaki Baz Castillo

2009/3/23 Alexandr Dubovikov <shurik(a)start4.info>: > On Mon, Mar 23, 2009 at 11:15:52AM +0100, Iñaki Baz Castillo wrote: >> 2009/4/1 Alexandr Dubovikov <shurik(a)start4.info>: >> > On Fri, Mar 20, 2009 at 10:21:20PM +0100, Andreas Heise wrote: >> >> Hello Iñaki, > > Hi Inaki, > >> > >> > Hi all, >> >> >> >> you should ask Alexandr he has introdused this feature with rev5452, >> >> but I'm not sure if he is on the lists all the time, so I'll forward your >> >> question >> >> to him... >> > >> > sorry, I am currently offline and couldn't answer directly to the list. >> > >> > anyway. I agreed, 180 seconds it's too big ping interval, but it couldn't be >> > also less 32 seconds. >> > >> > >> > http://www.ietf.org/rfc/rfc3261.txt >> > >> > 17.2.2 Non-INVITE Server Transaction >> > >> > >> > T1 = 500 ms. >> > >> > >> > Timer J 64*T1 for UDP Section 17.2.2 Wait time for >> > 0s for TCP/SCTP non-INVITE request >> > >> > so, the timer for "completed" is 32 seconds and couldn't be less. >> > >> > >> > of course, you can use less value, but on own risk :-) >> > RFC 3261 are not recommendet it :) >> >> Thanks for pointing it out. >> However, if a gw takes so long time (~32 seconds) to respond I prefer >> to consider it as "offline" :) > > don't forget to disable ping for this gateway, because finaly you will have > a DDOS effect of the "offline" gw. :-) Well, If the host is already offline then it doesn't matter to kill it more XD Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

14 years, 9 months

[Kamailio-Users] Firewall and auth

by Andreas Granig

Hi all, I just tried a setup like [UA] --> [pub][Firewall][priv] --> [priv][Kam] where the Firewall maps the public IP reachable by UAs to a private IP where Kamailio is listening. If I run sipsak on the Kamailio-machine, I can register fine, but as soon as the request goes via Firewall, authentication stops working. So how does the IP of Kamailio actually influence authentication? Do I have to set something special on Kamailio to make this work? Here's the Register after a 401 and the resulting 401 again, and it looks pretty well to me (1.2.3.4 is the public Firewall IP, which is configured as outbound proxy on the UA, 172.17.10.50 is the private Kamailio-IP and is also used as domain for user sipwise1, which is trying to register). Trace is taken on client-side, but looks the same on the Kamailio server (NAT seems to be handled fine): U 192.168.123.150:50600 -> 1.2.3.4:5060 REGISTER sip:1.2.3.4 SIP/2.0. Via: SIP/2.0/UDP 192.168.123.150:50600;rport;branch=z9hG4bK906580090. From: <sip:sipwise1@172.17.10.50>;tag=1631756043. To: <sip:sipwise1@172.17.10.50>. Call-ID: 1235449552. CSeq: 4 REGISTER. Contact: <sip:sipwise1@192.168.123.150:50600;line=e779ddd40d3251b>. Authorization: Digest username="sipwise1", realm="172.17.10.50", nonce="4a06e2820000000a80c173db2d166fedb7d8d1e933c97855", uri="sip:1.2.3.4", response="de645a701a7c507c47a5278923bce54b", algorithm=MD5. Max-Forwards: 70. User-Agent: Linphone/2.1.1 (eXosip2/3.1.0). Expires: 900. Content-Length: 0. U 1.2.3.4:5060 -> 192.168.123.150:50600 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.123.150:50600;rport=50600;branch=z9hG4bK906580090;received=213.47.175.165. From: <sip:sipwise1@172.17.10.50>;tag=1631756043. To: <sip:sipwise1@172.17.10.50>;tag=a49efde55ae28efd11dc5969af09c5db.b607. Call-ID: 1235449552. CSeq: 4 REGISTER. WWW-Authenticate: Digest realm="172.17.10.50", nonce="4a06e2820000000b2bd307dd3e71c80e3d6549ccc2b28269". Server: Sipwise registrar. Content-Length: 0. So the only thing referring to the public Firewall IP is in the R-Uri of the registration and in the Authorization-uri-token. Is this token also used to calculate the auth hashes somehow? Username looks fine in the Authorization header, and so does Realm. Any ideas? Andreas

14 years, 10 months

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sr-users May 2009