sr-users March 2009

sr-users@lists.kamailio.org

92 participants
149 discussions

[Kamailio-Users] HA+DRBD setup
by Siddhardha Garige 09 Jul '15

09 Jul '15

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

11 24

[Kamailio-Users] AudioCodes + Kamailio : Problem in SIP Message Headers
by Samuel Muller 26 Sep '12

26 Sep '12

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.

5 5

[OpenSER-Users] uaCSTA support in openser
by Hugo Koblmueller 09 Feb '11

09 Feb '11

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?

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[Kamailio-Users] uac_replace_from check
by jh＠tutpro.com 05 Nov '09

05 Nov '09

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

7 37

[Kamailio-Users] HOWTO rewrite to: header
by alexander merkulov 12 Aug '09

12 Aug '09

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

6 21

Re: [Kamailio-Users] [LCR] About ping
by Iñaki Baz Castillo 04 Jul '09

04 Jul '09

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>

2 6

[Kamailio-Users] Profile takes some time to be cleared
by mayamatakeshi 20 Apr '09

20 Apr '09

Hello, I'm using the set_dlg_profile function from the dialog module (kamailio 1.4). I use it to control limit of simultaneous calls to clients. I'm noticing that after this function is called, even if the call terminates immediately, it takes some 3 to 4 seconds for the profile to be cleared. Is there any reason for this? I can see even if I call unset_dlg_profile on failure_route, the profile will take that time span to be cleared. regards, mayama

2 9

[Kamailio-Users] another instance of kamailio on the same machine
by Juan Perez 03 Apr '09

03 Apr '09

hi To install another instance of kamailio listening in another IP, does one need to do something in particular aside from install it in another directory? Also, when it is time to run the script to create the mysql tables, how can I tell the script to create a db named "openser2" to avoid deleting the existing one named by default "openser" ? Just by changing the line DBNAME=openser in kamctlrc to DBNAME=openser2? thank you jp

8 22

[Kamailio-Users] Processing ACK.
by Carlos A. Alvarez 01 Apr '09

01 Apr '09

I am having an issue while setting up Kamailio. I using the carrierroute modules to process calls between carriers. The configuration works great. However I have and issue. On the SIP messages We are forcing all the media through an RTP proxy, however my issue has to do with the Contact field on the Message Header. While I am able to use force_rtp_proxy to replace the contact and connection fields on the SDP part of the message, I can seem to modify the Contact on the Header part. I tried using subst to change the Contact:, and while I was partly successful, it seems the ACKs going from one carrier to another are getting messed up and throwing Kamailio into a loop. We are trying to avoid providing Contact: information to our carriers. Basically keeping them from know about each other. Any ideas/help will be Greatly appreciated.

2 1

[Serusers] running rtpproxy on quad core processors
by Mundiline 01 Apr '09

01 Apr '09

Hi, I would like to run rtpproxy 1.2.0 on a server with Quad-core dual Xeon processor and load balance the load across the 4 processors. Do you know how do I enable this feature pls. ( I am not an expert in this field so keep it simple pls. thanks) Thanks in advance Regards, Tony

2 1

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sr-users March 2009