sr-users February 2009

sr-users@lists.kamailio.org

76 participants
97 discussions

[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.

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[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] 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

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[Kamailio-Users] Problem login Siremis
by Alberto Furtado 30 Mar '09

30 Mar '09

Hi, I am a new user, just installed Kamailio 1.4.3. The Siremis interface will be of great help but I am getting this error when I try to login. [2009-02-13 18:22:53 (GMT)] An exception occurred while executing this script: Error message: #256, No Database information found in the config file. Script name and line number of error: /var/www/siremis-0.9.1/openbiz/bin/Configuration.php:42 I created the ob_users Table, inserted a user (admin,admin), but I can't seem to figure out what I did wrong. Maybe its not finding the Config.xml ??? Thanks in advance for any help, Alberto Furtado No virus found in this outgoing message. Checked by AVG - http://www.avg.com Version: 8.0.176 / Virus Database: 270.10.23/1951 - Release Date: 13/2/2009 06:51

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[Serusers] Deadly embrace with rtpproxy - Is it necessary? Can it be turned off?
by Frank Durda IV 20 Mar '09

20 Mar '09

I'm running into a problem with rtpproxy on this point, quoting from the README: - - - - - - - - - - - - after the session has been created, the proxy listens on the port it has allocated for that session and waits for receiving at least one UDP packet from each of two parties participating in the call. Once such packet is received, the proxy fills one of two ip:port structures associated with each call with source ip:port of that packet. When both structures are filled in, the proxy starts relaying UDP packets between parties; - - - - - - - - - - - However, a number of clients frequently fail to emit any audio when originating a call until they hear something from the TDM gateway, such as ring-back or the called party answering. So although rtpproxy is receiving a stream of audio, such as a voice mail menu robot, the calling party can't hear any of it unless they happen to make some noise or randomly and blindly press a DTMF key. This seems to be made worse on links with silence suppression, so there is no background noise to trigger two-way audio. This is being encountered between Class 4 carriers, so we don't have the option to get someone to adjust their phone/PBX settings or have them breathe heavier. Is there a setting adjustment to get rtpproxy to just pass the RTP packets from directed calling and called sources even if one party hasn't happened to make noise yet? I personally don't understand why this requirement for seeing audio from both sides before starting the flow in either direction if audio starts coming in even exists. It seems to have no benefit but is bound to cause this deadly embrace problem in many situations that may be beyond the control of the owners of the equipment passing traffic along to the site where rtpproxy is in use. Suggestions? Fix? I have looked at the latest snapshot of rtpproxy and the README is unchanged since 1.1 so apparently this behavior is still the same. Thanks in advance!

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[Kamailio-Users] LDAP Module and TLS/SSL
by Sven Schulz 15 Mar '09

15 Mar '09

HI, Ive tested the LDAP authentication using Kamailio 1.4.2 and it works great. However, I'd like prevent the passwords being sent back "in the clear" from the LDAP server. Is there a way to use the Kamailio ldap module with TLS/SSL? I already have openssl and openldap-client on the kamailio box and can successfully use the ldapsearch command line tool via TLS. Sven Schulz Penn State University

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[Kamailio-Users] How to change request uri with Header value?
by Ramu 02 Mar '09

02 Mar '09

Hi All, I have situation where I need to change the request uri with To header value. Scenario: OpenServ will receive INVITE as follows INVITE sip:2000@open-ser-ip SIP/2.0. To: "2000"<sip:2000@sip.mydomain.com <sip%3A2000(a)sip.mydomain.com>>. Now I would like to change the request uri as follows: INVITE sip:2000@sip.mydomain.com <sip%3A2000(a)sip.mydomain.com> SIP/2.0. To: "2000"<sip:2000@sip.mydomain.com <sip%3A2000(a)sip.mydomain.com>>. and will do a lookup location. Ramu

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[Serusers] fix_nated_contact onreply
by samuel 02 Mar '09

02 Mar '09

I have seen lots of default config files where in the reply route only after checking the message (client_nat_test(1)) fix_nated is called. Why is not called when the NAT flag is set upong lookup_XX? In ser-oob I think the reply_route should include the case of a user called behind a NAT and the reply is not fixed due to some router in the middle. Will it hurt including fix_nated_contact in the case of checking the flag? Friday thoughts... Samuel

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[Kamailio-Users] how to know if my gw servers are active ?
by BERGANZ François 02 Mar '09

02 Mar '09

hello, I have a list of servers in my database. I need that openser send OPTIONS to servers. for example, I could have a col 'active' to know if each server is active or not. In fact, I need to do an other style of dispatch, and, I cant use the dispatcher module! is there a solution for that ? Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

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[Kamailio-Users] dial the prefix #31# to mask the number
by ALAEDDINE abbech 01 Mar '09

01 Mar '09

Hi When i dial #31# 011111with a sip phone i have an INVITE in OPENSER with a number %2331%23 011111. Who know how can i send an INVITE whith prefix #31# using a phone or softphone. thanks,

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sr-users February 2009