sr-users June 2014

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86 participants
145 discussions

route [Location] Documentation
by aawaise 30 Jun '14

30 Jun '14

I want to know that how route [Location] works when an INVITE packet approaches kamailio server. Basically route [Location] gathers information whether callee is a valid user or not. Secondly if it is online or not. I want to understand this whole procedure. Any help will be appreciated. Thanks. -- View this message in context: http://sip-router.1086192.n5.nabble.com/route-Location-Documentation-tp1286… Sent from the Users mailing list archive at Nabble.com.

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Voice quality issues when registering to Kamailio
by Travis Dillon 30 Jun '14

30 Jun '14

Hello all, I have an environment setup as follows: VoIP routers -> Kamailio 4.0 -> Asterisk 11.6 When I register the phones to Asterisk the call quality is excellent. So to be clear, the call comes in the VoIP router, routes through Kamailio to the phone registered to Asterisk. When I register the phone to Kamailio the call quality is significantly reduced. So clarity here, the call comes in the VoIP router, routes through Kamailio to the phone registered to Kamailio. Why would the call quality diminish when registered to Kamailio? Where can I start looking for this issue? Thank you, Travis R. Dillon

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UAC module refresh registration, could not reuse callid/fromtag !!!
by nevious 30 Jun '14

30 Jun '14

Hello all! I make kamailio1 remotely register to kamailio2. Here I set K1 modparam("uac", "reg_timer_interval", 15), and expires in uacreg as 30s. Then K1 should refresh registration after 15s. However the callid and from tag of every registration are different, and the Cseq just 10 or 11,never increased.In RFC3261, when UAC want refresh bindings, A UA SHOULD use the same Call-ID for all registrations during a single boot cycle. Is there any way to reuse the callid and fromtag during refresh registration ? Many thanks! And how to make the Cseq become 12 or more?

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Strange application hang querying DB
by Tim Chubb 27 Jun '14

27 Jun '14

Hi, I'm currently wracking my brains trying to deduce where an application hang can be coming from. I have ported the PostGres schema to be compatible with MS Sql Server and have had no issues getting a working dev environment working. However when I try and use SQL Azure as the backend db as opposed to SQL Server 2012 (running locally on my dev machine) Kamailio will seem to hang after about 10 - 15 mins of flawless operation. I have recreated the SQL Azure schema from the working schema running on my dev machine and copied all data in a 1:1 manner so I can be certain it's not a data quality issue. I am using the same FreeTDS/UnixODBC config that I am using for Asterisk realtime (which doesn't exhibit this behaviour). The really strange thing is that I can see the queries being sent by Kmailio in the freetds trace log, and the response from SQL Azure, however its as if the db response handler is deadlocked/hung. The kamailio process continues to work fine, and as long as its not required to commiunicate with the DB all appears well i.e. no error messages in the log file. This behaviour only happens when im using SQL Azure as opposed to SQL Server, so there must be something causing the issue, that's unique to SQL azure as opposed to SQL Server. As far as I can tell this is not something I can create a backtrace for as no core dumps are produced, as far as I can tell there is no easy way debug what the lock is. If anyone has any suggestions I would be very grateful to hear them. One thing that occurs to me is that the db_unixodbc module doesn't have half as many parameters as its equilent in Asterisk, and that connection pooling may be in use, this could be a bit of a show stopper AFAIK you have to disable connection pooling with SQL server, certainly any of the guides I have come across configuring Asterisk to work with SQL server suggests that. Tim Chubb Developer [VS-logo-email] The Coach House, Heywood House, Park Lane, Heywood, Westbury, BA13 4NA Direct: +44 (0)333 0110164 Email: tim.chubb(a)voicesimplified.com<mailto:tim.chubb@voicesimplified.com> Web: www.VoiceSimplified.com<http://www.voicesimplified.com/> Voice Simplified Ltd (registered in England & Wales: 07171825) registered office: Curzon House, 2nd Floor, 24 High Street, Banstead, Surrey. SM7 2LJ. The information in this email is confidential and may be legally privileged. It is intended solely for the addressee. Access to this email by anyone else is unauthorised. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. If you are not the intended recipient, please return the message to the sender and delete it from your records.

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Error when install Kamailio 4.1.4 on Mac OS 10.9.
by Son Truong 27 Jun '14

27 Jun '14

Hi Kamailio, - I meet an error when I am trying install Kamailio 4.1.4 on Mac OS 10.9. The error: touch /usr/local/lib64/kamailio/modules/usrloc.so install -m 755 usrloc.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xhttp.so install -m 755 xhttp.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xhttp_rpc.so install -m 755 xhttp_rpc.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xlog.so install -m 755 xlog.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xprint.so install -m 755 xprint.so /usr/local/lib64/kamailio/modules /bin/sh: -c: line 1: syntax error: unexpected end of file make: *** [install-cfg] Error 2 How can I fix it? I am new in Kamailio, I want to learn Kamailio. -- Thanks and best regards, Phạm Sơn Trường Android, iOS Team Lead at ME Corp. Email1: truonguit2010(a)gmail.com Email2: truongps(a)mecorp.vn Mobile: 01218816208

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Processing of Invite Packet
by aawaise 27 Jun '14

27 Jun '14

When an invite packet is recieved by kamailio server, what processing is done on it ? Which modules and functions are called in its processing and in which sequence ? Thanking in advance. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Processing-of-Invite-Packet-tp12867… Sent from the Users mailing list archive at Nabble.com.

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Transaction Stateful Message Handling
by AliReza Khoshgoftar Monfared 26 Jun '14

26 Jun '14

Hi, I am trying to configure a SIP proxy architecture that is stateful with regard to transaction but not the dialogues, i.e. If an INVITE takes a specific route, the corresponding 180 Ringing, 200 OK and ACK shall traverse the same set of proxies, but the corresponding BYE can take another route (and obv it's own 200 OK takes the same route again) Which modules can help me with that? In specific which functions / modules are recommended to handle requests and responses? Currently, I use "t_relay()" from TM module to forward requests and "loose_route()" from RR module to handle replies, but it looks like that if a BYE ends in a proxy which its corresponding INVITE has not traversed, it is not forwarded by that proxy. Am I making a mistake in my routing? Isn't there a simple way to do transaction stateful routing in Kamailio? Thanks, Alireza

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How to stop kamailio from send 408?
by Allen Zhang 26 Jun '14

26 Jun '14

Hi, when t_relay() times out, calling sl_reply_error() will send 408 to the client. But if I don't want to send 408, how do I interpret the error code t_relay() set? Regards, Allen

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How to find whether or not certain header exists?
by Allen Zhang 26 Jun '14

26 Jun '14

Hi, If I normally receive a INVITE that has a header 'aaa' with value 'bbb', but it's possible that some INVITEs don't have this header. How do I determine in the config file if the header exists? To check the value of the header I do if($hdr(aaa) == "bbb"). But what if the header doesn't exist? Cheer, Allen

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Kamailio Freepbx Integration Dropping Calls
by Carlos Rangel 26 Jun '14

26 Jun '14

Hello I have successfully (I believe) implemented Kamailio 4.1.4 integration with Freepbx 5.2.11 taking as a guide Daniel's tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb. I just did not create the voicemail tables because voice mail is handled by Freepbx. I installed the system in a separate box for testing and connected to the Freepbx Production server via IAX trunk. The system is behind a Cisco Firewall and looks like this Remote User Internet Internal network Cisco 7960 ----DSL router ---|------Internet --------|-----Cisco ASA 5500 FW--------------Kamailio/Freepbx (Same Box)------IAX Trunk----------Freepbx Production Server --------|------ PSTN I have configured the FW to allow UDP and TCP traffic from the corresponding IP as well as tfpt that is needed for the Ciscos to pick up the configuration from the server. I have a few remotes Cisco 7960 phones that can register remotely in Kamailio as long as the user is added with kamctl add user password and as long as the extension is created in Freepbx. The problem that I have is when try to make a call from the remote Ciscos the call is dropped after 30 or 40 seconds. I can see from the logs that the problem appears to be that the server is not receiving responses from the phone 06-25 10:57:30] WARNING[1814] chan_sip.c: Retransmission timeout reached on transmission 000653dc-39400006-2579bbcd-13d9adcb(a)192.168.0.22 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32001ms with no response [2014-06-25 10:57:30] WARNING[1814] chan_sip.c: Hanging up call 000653dc-39400006-2579bbcd-13d9adcb(a)192.168.0.22 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). Is this something that we can adjust in kamailio or could it be related to the FW configuration?? Sorry but I am very new to kamailio and sip. Thanks Carlos

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sr-users June 2014