sr-users January 2009

sr-users@lists.kamailio.org

85 participants
110 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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[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] 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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[OpenSER-Users] [OT] How to handle different DID's in incoming calls for a registered client?
by Iñaki Baz Castillo 27 Feb '09

27 Feb '09

Hi, maybe this question is a bit off-topic so I'm sorry for that. My question is about SIP providers using OpenSer that associate PSTN numbers to their local clients (SIP accounts): Usually the client must register to OpenSer in order to receive calls. Then it will appear in location table with "Contact=sip:clientX@IP". Suppose clientX has two PSTN numbers associated in a ENUM entry: +34999000111 +34999000222 When anyone in PSTN world calls to +34999000222 the call will arrive to the OpenSer from a gateway in an INVITE like: INVITE sip:+34999000222@gateway SIP/2.0 To: <sip:+34999000222@gateway> The OpenSer will do the lookup in location table and finally send this INVITE to the clientX: INVITE sip:clientX@IP_clientX SIP/2.0 To: <sip:+34999000222@gateway> The info about the called PSTN number is just available in "To" header, so a way to get different behaviour for each associated PSTN number is matching "To" URI. Is common to do it? which other alternatives are there? Thanks for any comment. Regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[Kamailio-Users] Looking for a highly available / load balanced setup
by John D 11 Feb '09

11 Feb '09

I currently have three Asterisk servers in my environment. None of them speak to each other just yet. The idea is to install Kamailio and be able to load balance/distribute SIP calls to any of the three asterisk boxes. If an entire box plug gets pulled, the operation continues. If a Kamailio box gets the plug pulled (and due to limited resources, I plan at first to run kamailio on all three of the asterisk boxes), the operation will continue. Does anyone have good documentation or insight in order to facilitate this sort of setup? Thanks.

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[Kamailio-Users] location table
by joy yue 05 Feb '09

05 Feb '09

Hi, Is there a limitation on the number of rows in location table? In my rig, whatever number of users I use (>2million users), I notice the number of users in location table is 343707, which is far less than the number of users in my test. I thought previously that location table saves all the users in my test. Also when I use large number of users (>2million), openser pops up an error: ERROR: registrar:update_contacts: invalid cseq for aor <xxxx>. Has anyone saw this before? I am using openser1.3.2. Thanks, -Joy

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[Serusers] Error loading "trusted" table (permissions.so)
by caio 05 Feb '09

05 Feb '09

Hello all, Does anyone knows about this "permission.so" module error when try to load "trusted" table (mysql)? If "trusted" table is empty, there is no error, but whan at least there is 1 record, the error appears. for example: mysql> select * from trusted; +---------------+-------+--------------+ | src_ip | proto | from_pattern | +---------------+-------+--------------+ | 10.124.35.253 | any | ^sip:.*$ | +---------------+-------+--------------+ and when I try lo restart ser: Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 966737 daemon.info] permissions - initializing Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 414293 daemon.warning] Default allow file (/usr/local/etc/ser/permissions.allow) not found => empty rule set Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 844440 daemon.warning] Default deny file (/usr/local/etc/ser/permissions.deny) not found => empty rule set Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 895822 daemon.error] ERROR: permissions: trusted_reload(): Database problem Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 650101 daemon.crit] init_trusted(): Reload of trusted table failed Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 345573 daemon.error] Error while initializing allow_trusted function Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 660978 daemon.error] init_mod(): Error while initializing module permissions or when executing serctl fifo cmd: root@delcolimba /usr/local # sbin/serctl fifo trusted_dump 200 OK root@delcolimba /usr/local # sbin/serctl fifo trusted_reload 400 Trusted table reload failed root@delcolimba /usr/local # tail /var/adm/messages Jan 15 15:45:37 delcolimba /usr/local/sbin/ser[717]: [ID 895822 daemon.error] ERROR: permissions: trusted_reload(): Database problem Jan 15 15:45:37 delcolimba /usr/local/sbin/ser[717]: [ID 450600 daemon.error] ERROR: fifo_server: command (trusted_reload) processing failed Thank you, Caio

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[Kamailio-Users] New module : PURPLE - IM/Presence multi-protocol gateway
by Eric PTAK 04 Feb '09

04 Feb '09

Hello everyone, Let me introduce to you a new module, available in trunk since yesterday, so it will be included in 1.5.0 release : >From http://kamailio.org/docs/modules/devel/purple.html: *Purple module is a multi-protocol instant-messaging gateway module which use libpurple, the library on bottom of Pidgin (ex-gaim).* *This plugin was wrote in order to proof the concept of gateway between SIP and MSN/Live networks.* *[...]* *It's no really adapted to a production environment with a huge user count but can be used in order to demonstrate the concept or to use it in SOHO environment.* I hope you'll enjoy purple module, and I'll welcome all feedbacks. Eric.

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sr-users January 2009