Re: [SR-Users] How to cleanup unconfirmed dialog in memory?
I am unable to recreate the problem. I will keep eye on it. If it happen again, I will report back here. gary On Fri, Jul 5, 2013 at 3:52 AM, Charles Chance charles.cha...@sipcentric.com wrote: Hi Daniel, If someone can try it and confirm is working fine for sl replied dialogs as well as for those forwarded, I will backport. Seems to work fine for us in our test environment. So far, no unconfirmed dialogs stuck in memory for statelessly replied transactions. Cheers, Charles On 3 July 2013 17:25, Charles Chance charles.cha...@sipcentric.comwrote: Hi Daniel, Sounds perfect! I will try to test here tomorrow and let you know. Cheers, Charles On 2 July 2013 22:44, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, I pushed a patch in master that detects when the dialog is created by not getting to transaction due to a stateless reply. http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commitdiff;h=fa0339b1906690f009786fc9ed92c73a8c9e6520;hp=b4682cac2e2f151288a411018da077b6d1526eca If someone can try it and confirm is working fine for sl replied dialogs as well as for those forwarded, I will backport. Cheers, Daniel On 7/2/13 4:22 PM, Carlos Ruiz Díaz wrote: Now that you mention it, it makes perfect sense since this function sends BYE to both legs which only makes sense with confirmed dialogs. I run out of alternatives. Maybe patching the module is the only solution. Regards, Carlos On Tue, Jul 2, 2013 at 10:16 AM, Charles Chance charles.cha...@sipcentric.com wrote: That doesn't work I'm afraid. Also from http://www.kamailio.org/docs/modules/3.1.x/modules_k/dialog.html#id2524783: *Note: Works only for confirmed dialogs.* It's something I've been meaning to look further into for a while, but haven't yet had chance. I suspect a small patch will be required though. Regards, Charles On 2 July 2013 15:11, Carlos Ruiz Díaz carlos.ruizd...@gmail.comwrote: Ok, I haven't noticed that, although I can't tell for sure whether it will work or not. If you can dump the dialogs using xmlrpc or rpc interface, maybe you could parse the info and tear down those unconfirmed dialogs using dlg_end_dlg: http://www.kamailio.org/docs/modules/3.1.x/modules_k/dialog.html#id2524783 On Tue, Jul 2, 2013 at 10:02 AM, Charles Chance charles.cha...@sipcentric.com wrote: The parameter does not work for us. We have modparam(dialog, default_timeout,7200), but it only has an effect on confirmed dialogs. As you can see from my previous example, there is not even a timestart value on the unconfirmed dialog, so how can Kamailio know when timeout is reached? On 2 July 2013 14:51, Carlos Ruiz Díaz carlos.ruizd...@gmail.comwrote: There is, but for all dialogs, not only the unconfirmed ones. http://www.kamailio.org/docs/modules/3.3.x/modules_k/dialog.html#default-timeout-id On Tue, Jul 2, 2013 at 9:49 AM, Charles Chance charles.cha...@sipcentric.com wrote: Hmm, I don't think there is even a timeout value set on unconfirmed dialogs in memory. Example (Kamailio 3.3.3): dialog:: hash=1791:10106 state:: 1 ref_count:: 1 timestart:: 0 timeout:: 0 ... Whereas: dialog:: hash=2963:2808 state:: 4 ref_count:: 2 timestart:: 1372772302 timeout:: 114829207 ... Therefore, the unconfirmed dialogs never get cleared automatically, in my experience at least. I hope I'm wrong though :) Cheers, Charles On 2 July 2013 14:31, Henning Westerholt h...@kamailio.org wrote: Am Dienstag, 2. Juli 2013, 14:23:25 schrieb Charles Chance: I don't think this will help at all, as regardless of DB mode, unconfirmed dialogs are not stored in DB anyway. The important thing to remember is that if you are calling dialog_manage() in your config, to only do it once you are ready to forward the request. If you call it but then exit for some reason without actually forwarding, you will probably end up with a stuck dialog. Maybe someone else can suggest other possible causes? To my knowledge, there is no existing way to clear these without restarting. Hello, AFAIK these stale dialogs are cleaned up after the dialog timeout. There are module parameter and also dialog specific parameter to control this variable. This stale dialogs needs a bit of memory, but are otherwise harmless. Best regards, Henning www.sipcentric.com Follow us on twitter @sipcentric http://twitter.com/sipcentric Sipcentric Ltd. Company registered in England Wales no. 7365592. Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham B7 4EJ. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Carlos http://caruizdiaz.com +595981146623
Re: [SR-Users] Load database table in memory
I am using mtree for the same purpose and it works great. It never expire. I had loaded about 7 millions records in mtree table and there is no performance issue at all. Gary On Wed, Jul 3, 2013 at 7:57 AM, Alex Balashov abalas...@evaristesys.comwrote: Sure, you can do it that way. However, htable and mtree accommodate relatively primitive data sets. Given sufficient table complexity and/or size, your best bet is to use an in-memory storage backend on the database side itself. In MySQL, this is called: http://dev.mysql.com/doc/**refman/5.7/en/memory-storage-**engine.htmlhttp://dev.mysql.com/doc/refman/5.7/en/memory-storage-engine.html -- Alex On 07/03/2013 05:41 AM, Grant Bagdasarian wrote: Hello, I need to query a database for every SIP request coming into Kamailio, but I want this to be handled as fast as possible, so I was thinking of loading the data I need in memory using the HTABLE or MTREE modules. When the SIP request is coming from one of our carriers, the called number ($rU) must be used to get the data for this called number. Normally, I would query the database, using sqlops, and pass the value of $rU as the parameter and get the column values using the $dbr variable. Can this be accomplished with HTABLE or MTREE? Also, the autoexpire parameter in HTABLE; once expired, will the data be reloaded again from the database? Thanks, Grant __**_ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**usershttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ __**_ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**usershttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] How to cleanup unconfirmed dialog in memory?
We need also have a way to timeout all the dead unconfirmed dialog. I have a situation that the SIP message was perfect fine with Bye signal, yet the dialog is still hang with state 1. Gary On Wed, Jul 3, 2013 at 12:25 PM, Charles Chance charles.cha...@sipcentric.com wrote: Hi Daniel, Sounds perfect! I will try to test here tomorrow and let you know. Cheers, Charles On 2 July 2013 22:44, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, I pushed a patch in master that detects when the dialog is created by not getting to transaction due to a stateless reply. http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commitdiff;h=fa0339b1906690f009786fc9ed92c73a8c9e6520;hp=b4682cac2e2f151288a411018da077b6d1526eca If someone can try it and confirm is working fine for sl replied dialogs as well as for those forwarded, I will backport. Cheers, Daniel On 7/2/13 4:22 PM, Carlos Ruiz Díaz wrote: Now that you mention it, it makes perfect sense since this function sends BYE to both legs which only makes sense with confirmed dialogs. I run out of alternatives. Maybe patching the module is the only solution. Regards, Carlos On Tue, Jul 2, 2013 at 10:16 AM, Charles Chance charles.cha...@sipcentric.com wrote: That doesn't work I'm afraid. Also from http://www.kamailio.org/docs/modules/3.1.x/modules_k/dialog.html#id2524783: *Note: Works only for confirmed dialogs.* It's something I've been meaning to look further into for a while, but haven't yet had chance. I suspect a small patch will be required though. Regards, Charles On 2 July 2013 15:11, Carlos Ruiz Díaz carlos.ruizd...@gmail.comwrote: Ok, I haven't noticed that, although I can't tell for sure whether it will work or not. If you can dump the dialogs using xmlrpc or rpc interface, maybe you could parse the info and tear down those unconfirmed dialogs using dlg_end_dlg: http://www.kamailio.org/docs/modules/3.1.x/modules_k/dialog.html#id2524783 On Tue, Jul 2, 2013 at 10:02 AM, Charles Chance charles.cha...@sipcentric.com wrote: The parameter does not work for us. We have modparam(dialog, default_timeout,7200), but it only has an effect on confirmed dialogs. As you can see from my previous example, there is not even a timestart value on the unconfirmed dialog, so how can Kamailio know when timeout is reached? On 2 July 2013 14:51, Carlos Ruiz Díaz carlos.ruizd...@gmail.comwrote: There is, but for all dialogs, not only the unconfirmed ones. http://www.kamailio.org/docs/modules/3.3.x/modules_k/dialog.html#default-timeout-id On Tue, Jul 2, 2013 at 9:49 AM, Charles Chance charles.cha...@sipcentric.com wrote: Hmm, I don't think there is even a timeout value set on unconfirmed dialogs in memory. Example (Kamailio 3.3.3): dialog:: hash=1791:10106 state:: 1 ref_count:: 1 timestart:: 0 timeout:: 0 ... Whereas: dialog:: hash=2963:2808 state:: 4 ref_count:: 2 timestart:: 1372772302 timeout:: 114829207 ... Therefore, the unconfirmed dialogs never get cleared automatically, in my experience at least. I hope I'm wrong though :) Cheers, Charles On 2 July 2013 14:31, Henning Westerholt h...@kamailio.org wrote: Am Dienstag, 2. Juli 2013, 14:23:25 schrieb Charles Chance: I don't think this will help at all, as regardless of DB mode, unconfirmed dialogs are not stored in DB anyway. The important thing to remember is that if you are calling dialog_manage() in your config, to only do it once you are ready to forward the request. If you call it but then exit for some reason without actually forwarding, you will probably end up with a stuck dialog. Maybe someone else can suggest other possible causes? To my knowledge, there is no existing way to clear these without restarting. Hello, AFAIK these stale dialogs are cleaned up after the dialog timeout. There are module parameter and also dialog specific parameter to control this variable. This stale dialogs needs a bit of memory, but are otherwise harmless. Best regards, Henning www.sipcentric.com Follow us on twitter @sipcentric http://twitter.com/sipcentric Sipcentric Ltd. Company registered in England Wales no. 7365592. Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham B7 4EJ. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Carlos http://caruizdiaz.com +595981146623 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- *Charles Chance* Managing Director t. 0121 285 4400m. 07932 063 891 www.sipcentric.com Follow us
Re: [SR-Users] How to cleanup unconfirmed dialog in memory?
Thanks for suggestion. I will give it try. Since every call will have to check database, will this effect the performance of Kamailio for high volume server? Gary On Tue, Jul 2, 2013 at 8:55 AM, I.Pavlov i...@izhnet.ru wrote: Hi, Gary. I had same problem. Try to use “db_mode” parameter of dialog module with 1 value. ** ** *1 - REALTIME* - any dialog information changes will be reflected into the database immediatly. ** ** Then you can kill the dead dialogs through database. ** ** *From:* sr-users-boun...@lists.sip-router.org [mailto: sr-users-boun...@lists.sip-router.org] *On Behalf Of *Gary Chen *Sent:* Tuesday, July 02, 2013 4:14 PM *To:* Kamailio (SER) - Users Mailing List *Subject:* [SR-Users] How to cleanup unconfirmed dialog in memory? ** ** ** ** Kamailio 3.3.3 I am using dialog module to do the concurrent call limit. Once a while I got a dead unconfirmed dialog hung in memory. The only way I know to cleanup this is to restart kamailio. So my questions are: 1) Does anybody know a better way to cleanup the dead dialog without having to restart the kamailio server? This is a production server. I really really hate to restart the server. 2) How easy to modify the source code to timeout the dead unconfirmed dialog in memory? Gary ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] LCR in Kamailio 4.0
First thing to remember that lcr_id field is the one to link all three tables together. Let's say that you want to create a lcr to route international call starting with 011: 1) Create a row in lcr_gw table with id = 4, lcr_id = 3 and gateway IP etc. 2) Then create a row in lcr_rule table with id =2, lcr_id = 3 and prefix = 011 3) Then create a row in lcr_rule_target table to glue the gateway and prefix together like this: lcr_id =3, gw_id=4 (Match the id value in lcr_gw table), rule_id = 2 (Match the id in lcr_rule table) also rest of the fields like priority etc. Hope this help. Gary On Fri, Jun 28, 2013 at 5:02 PM, Geoffrey Mina geoffreym...@gmail.comwrote: Greetings, I am migrating some 1.5 servers to 4.0 and I have some questions about how the LCR module works now. I am familiar with the concept of the gw table and the lcr table. This was pretty straight forward. In the new version it looks like we have: LCR Gateway List LCR Rule List LCR Target List I read through the module documentation and it doesn't really speak to what the new architecture is intended to accomplish. Anyone have a quick overview they would like to share which would help me understand the intent of the data structure? Thanks, Geoff ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] LCR in Kamailio 4.0
lcr_rule_target is like a glue to link lcr_gw and lcr_rule tables together through rule_id and gw_id as well as lcr_id. In this case if you have several sets of lcr_id, you will end up with several duplicate gateway and rulls. It is not the best design and is hard to use. To reload lcr you do this: sercmd lcr.reload Gary Chen On Fri, Jun 28, 2013 at 6:50 PM, Geoffrey Mina geoffreym...@gmail.comwrote: What is the value of the target table? Previously the lcr rule table was directly tied to the gateway. Is this simply to enable N:1 relationships between lcr rules and gateways. Also - any idea how to reload the LCR configuration without restarting kamailio? I previously issued kamctl fifo lcr_reload, but that doesn't appear to work any longer. Thanks! On Fri, Jun 28, 2013 at 4:44 PM, Gary Chen gchen3...@gmail.com wrote: First thing to remember that lcr_id field is the one to link all three tables together. Let's say that you want to create a lcr to route international call starting with 011: 1) Create a row in lcr_gw table with id = 4, lcr_id = 3 and gateway IP etc. 2) Then create a row in lcr_rule table with id =2, lcr_id = 3 and prefix = 011 3) Then create a row in lcr_rule_target table to glue the gateway and prefix together like this: lcr_id =3, gw_id=4 (Match the id value in lcr_gw table), rule_id = 2 (Match the id in lcr_rule table) also rest of the fields like priority etc. Hope this help. Gary On Fri, Jun 28, 2013 at 5:02 PM, Geoffrey Mina geoffreym...@gmail.comwrote: Greetings, I am migrating some 1.5 servers to 4.0 and I have some questions about how the LCR module works now. I am familiar with the concept of the gw table and the lcr table. This was pretty straight forward. In the new version it looks like we have: LCR Gateway List LCR Rule List LCR Target List I read through the module documentation and it doesn't really speak to what the new architecture is intended to accomplish. Anyone have a quick overview they would like to share which would help me understand the intent of the data structure? Thanks, Geoff ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] lcr.dump_gws problem
Kamailio 3.2.0 When testing Kamailio 3.2.0, I notice that lcr.dump_gws command does not display ip_addr correctly. Here is my output: lcr_id: 3 gw_id: 15 gw_index: 2 gw_name: gateway_1 scheme: sip ip_addr: 1273060816.0.0.0 hostname: port: 5060 params: strip: 0 prefix: tag: flags: 0 defunct_until: 0 LCR is still working correctly. It just does not display IP in the right form. Does any body also has the same problem? Gary ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] lcr.dump_gws problem
Sorry, it should be Kamailio 3.3.0 not 3.2.0. From: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Gary Chen Sent: Tuesday, July 24, 2012 3:39 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: [SR-Users] lcr.dump_gws problem Kamailio 3.2.0 When testing Kamailio 3.2.0, I notice that lcr.dump_gws command does not display ip_addr correctly. Here is my output: lcr_id: 3 gw_id: 15 gw_index: 2 gw_name: gateway_1 scheme: sip ip_addr: 1273060816.0.0.0 hostname: port: 5060 params: strip: 0 prefix: tag: flags: 0 defunct_until: 0 LCR is still working correctly. It just does not display IP in the right form. Does any body also has the same problem? Gary ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] How to use regular expression for $var ?
Kamailio 3.3.0 I have a variable $var(s:dst). It can store either a number or IP. How do I check to determine whether it is a number of IP? I tried the following and it did not work: If ($var(s:dst) =~ ^\d+\.\d+\.\d+\.\d+$){ It is a IP. }else{ It is a Number } Thanks Gary ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] ACK and BYE messages uses wrong socket.
Yes, it is available when kamailio started. I tried the latest version (3.2.3) and it did the same thing. If I removed listen=udp:x.x.130.34:5060 (primary IP on the NIC) from configuration file, it will report error when trying to relay() the message like this: ERROR: core [forward.c:220]: ERROR: get_out_socket: no socket found It seems that kamailio does not populate default socket with secondary IP (floating IP) and you have to use force_send_socket() to send the message. Gary From: Daniel-Constantin Mierla [mailto:mico...@gmail.com] Sent: Tuesday, July 03, 2012 12:23 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Cc: Gary Chen Subject: Re: [SR-Users] ACK and BYE messages uses wrong socket. Hello, is .36 address available on the network interface when kamailio is started? Also, use at least v3.2.3 (the latest in 3.2 series) to be sure you get all the fixes since 3.2.0 and this is not related to some issue already fixed. Cheers, Daniel On 7/3/12 4:52 PM, Gary Chen wrote: Yes, I did the same thing as you mentioned and it still doing the same thing. Here is my setup: mhomed=1 listen=udp:x.x.130.36:5060 # external IP listen=udp:x.x.130.34:5060 # external IP listen=udp:10.200.1.31:5060 # internal IP If I removed .130.34, I got error saying no socket found. Gary Chen From: sr-users-boun...@lists.sip-router.orgmailto:sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Vitaliy Aleksandrov Sent: Tuesday, July 03, 2012 10:27 AM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: Re: [SR-Users] ACK and BYE messages uses wrong socket. Have you specified interfaces with listen command ? I had a problem as you described and have fixed it by moving a listen directive with a floating ip to the top of the list. So you can try to specify interfaces you will use for SIP and set a virtual ip at the top of that list. Kamailio 3.2.0 I am trying to setup kamailio to do the sip trunking. It receive the sip traffic from customer and then send it to carrier. I have two NIC interface's assigned with three IP's: Interface 1: ( Public IP's) x.x.130.34 x.x.130.36 floating IP interface 2: (private IP's) 10.10.1.31 .36 is a floating IP assigned by Linux-HA (heartbeat/pacemaker). I only want to use .36 to receive and send sip traffic. I uses force_send_socket() to send INVITE with .36 IP. But the ACK message always want to use .34 IP even the Route header has .36 in it unless I force it with force_send_socket() . How can I fix this problem? See below for the SIP messages: (x.x.128.205 is customer IP, x.x.129.200 is PSTN gateway IP) U x.x.128.205:51694 - x.x.130.36:5060 INVITE sip:5033441174@x.x.130.36:5060mailto:sip:5033441174@x.x.130.36:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9hG 4bK1D3CD1..From: sip:5024427578@x.x.128.205mailto:sip:5024427578@x.x.128.205;tag=24513088-D59..To: sip:5033441174@x.x.130.3mailto:sip:5033441174@x.x.130.3 6..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.20mailto:1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.20 5..Supported: 100rel,timer,replaces..Min-SE: 1800..Cisco-Guid: 411443261-3293254113-3191264919-256 0877466..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, CO MET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..CSeq: 101 INVITE..Max-Forwards: 70..Remote-P arty-ID: sip:5024427578@x.x.128.205mailto:sip:5024427578@x.x.128.205;party=calling;screen=no;privacy=off..Timestamp: 1341322661 ..Contact: sip:5024427578@x.x.128.205:5060mailto:sip:5024427578@x.x.128.205:5060..Expires: 180..Allow-Events: telephone-event..Conte nt-Type: application/sdp..Content-Length: 366v=0..o=CiscoSystemsSIP-GW-UserAgent 9094 579 IN IP 4 x.x.128.205..s=SIP Call..c=IN IP4 x.x.128.205..t=0 0..m=audio 19312 RTP/AVP 125 0 18 100 10 1..c=IN IP4 x.x.128.205..a=rtpmap:125 X-CCD/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:18 G729/8000..a =fmtp:18 annexb=yes..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194..a=rtpmap:101 telephone-event/8000 ..a=fmtp:101 0-16.. # U x.x.130.36:5060 - x.x.128.205:51694 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9 hG4bK1D3CD1;rport=51694..From: sip:5024427578@x.x.128.205mailto:sip:5024427578@x.x.128.205;tag=24513088-D59..To: sip:502244117 4@x.x.130.36mailto:4@x.x.130.36..Call-ID: 1887588d-c44b11e1-be38d697-98a3e...@x.x.128.205..cseqmailto:1887588d-c44b11e1-be38d697-98a3e...@x.x.128.205..cseq: 101 INVITE..Se rver: LVS Proxy 1.0..Content-Length: 0 # U x.x.130.36:5060 - x.x.129.200:5060 INVITE sip:15033441174@x.x.129.200:5060mailto:sip:15033441174@x.x.129.200:5060 SIP/2.0..Record-Route: sip:x.x.130.36;lr=on..Via: S IP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0..Via: SIP/2.0/UDP x.x.128.205:5060;rport =51694;branch=z9hG4bK1D3CD1
[SR-Users] ACK and BYE messages uses wrong socket.
Kamailio 3.2.0 I am trying to setup kamailio to do the sip trunking. It receive the sip traffic from customer and then send it to carrier. I have two NIC interface's assigned with three IP's: Interface 1: ( Public IP's) x.x.130.34 x.x.130.36 floating IP interface 2: (private IP's) 10.10.1.31 .36 is a floating IP assigned by Linux-HA (heartbeat/pacemaker). I only want to use .36 to receive and send sip traffic. I uses force_send_socket() to send INVITE with .36 IP. But the ACK message always want to use .34 IP even the Route header has .36 in it unless I force it with force_send_socket() . How can I fix this problem? See below for the SIP messages: (x.x.128.205 is customer IP, x.x.129.200 is PSTN gateway IP) U x.x.128.205:51694 - x.x.130.36:5060 INVITE sip:5033441174@x.x.130.36:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9hG 4bK1D3CD1..From: sip:5024427578@x.x.128.205;tag=24513088-D59..To: sip:5033441174@x.x.130.3 6..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.20 5..Supported: 100rel,timer,replaces..Min-SE: 1800..Cisco-Guid: 411443261-3293254113-3191264919-256 0877466..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, CO MET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..CSeq: 101 INVITE..Max-Forwards: 70..Remote-P arty-ID: sip:5024427578@x.x.128.205;party=calling;screen=no;privacy=off..Timestamp: 1341322661 ..Contact: sip:5024427578@x.x.128.205:5060..Expires: 180..Allow-Events: telephone-event..Conte nt-Type: application/sdp..Content-Length: 366v=0..o=CiscoSystemsSIP-GW-UserAgent 9094 579 IN IP 4 x.x.128.205..s=SIP Call..c=IN IP4 x.x.128.205..t=0 0..m=audio 19312 RTP/AVP 125 0 18 100 10 1..c=IN IP4 x.x.128.205..a=rtpmap:125 X-CCD/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:18 G729/8000..a =fmtp:18 annexb=yes..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194..a=rtpmap:101 telephone-event/8000 ..a=fmtp:101 0-16.. # U x.x.130.36:5060 - x.x.128.205:51694 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9 hG4bK1D3CD1;rport=51694..From: sip:5024427578@x.x.128.205;tag=24513088-D59..To: sip:502244117 4@x.x.130.36..Call-ID: 1887588d-c44b11e1-be38d697-98a3e...@x.x.128.205..cseq: 101 INVITE..Se rver: LVS Proxy 1.0..Content-Length: 0 # U x.x.130.36:5060 - x.x.129.200:5060 INVITE sip:15033441174@x.x.129.200:5060 SIP/2.0..Record-Route: sip:x.x.130.36;lr=on..Via: S IP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0..Via: SIP/2.0/UDP x.x.128.205:5060;rport =51694;branch=z9hG4bK1D3CD1..From: sip:5024427578@x.x.128.205;tag=24513088-D59..To: sip:50224 41174@x.x.130.36..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3 E39A@x.x.128.205..Supported: 100rel,timer,replaces..Min-SE: 1800..Cisco-Guid: 411443261-3293254 113-3191264919-2560877466..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANC EL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..CSeq: 101 INVITE..Max-Forw ards: 69..Remote-Party-ID: sip:5024427578@x.x.128.205;party=calling;screen=no;privacy=off..Tim estamp: 1341322661..Contact: sip:5024427578@x.x.128.205:51694..Expires: 180..Allow-Events: tel ephone-event..Content-Type: application/sdp..Content-Length: 375v=0..o=CiscoSystemsSIP-GW-UserA gent 9094 579 IN IP4 10.200.1.51..s=SIP Call..c=IN IP4 10.200.1.51..t=0 0..m=audio 20464 RTP/AVP 12 5 0 18 100 101..c=IN IP4 10.200.1.51..a=rtpmap:125 X-CCD/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:18 G7 29/8000..a=fmtp:18 annexb=yes..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194..a=rtpmap:101 telephone- event/8000..a=fmtp:101 0-16..a=nortpproxy:yes.. U x.x.129.200:5060 - x.x.130.36:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0,SIP/2.0/UDP 216.4 9.128.205:5060;rport=51694;branch=z9hG4bK1D3CD1..From: sip:5024427578@x.x.128.205;tag=24513088 -D59..To: sip:5033441174@x.x.130.36;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Cal l-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Timestamp: 1341322661..Server: Cisco-SIPG ateway/IOS-12.x..CSeq: 101 INVITE..Allow-Events: telephone-event..Content-Length: 0 # U x.x.129.200:5060 - x.x.130.36:5060 SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0,SIP/2.0/ UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D3CD1..From: sip:5024427578@x.x.128.205;ta g=24513088-D59..To: sip:5033441174@x.x.130.36;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:4 1 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Timestamp: 1341322661..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 101 INVITE..Require: 100rel..RSeq: 6708..Allow: INVITE, OPTIONS, B YE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..Allow-Events: tele phone-event..Contact: sip:15033441174@x.x.129.200:5060..Record-Route:
Re: [SR-Users] ACK and BYE messages uses wrong socket.
Yes, I did the same thing as you mentioned and it still doing the same thing. Here is my setup: mhomed=1 listen=udp:x.x.130.36:5060 # external IP listen=udp:x.x.130.34:5060 # external IP listen=udp:10.200.1.31:5060 # internal IP If I removed .130.34, I got error saying no socket found. Gary Chen From: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Vitaliy Aleksandrov Sent: Tuesday, July 03, 2012 10:27 AM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: Re: [SR-Users] ACK and BYE messages uses wrong socket. Have you specified interfaces with listen command ? I had a problem as you described and have fixed it by moving a listen directive with a floating ip to the top of the list. So you can try to specify interfaces you will use for SIP and set a virtual ip at the top of that list. Kamailio 3.2.0 I am trying to setup kamailio to do the sip trunking. It receive the sip traffic from customer and then send it to carrier. I have two NIC interface's assigned with three IP's: Interface 1: ( Public IP's) x.x.130.34 x.x.130.36 floating IP interface 2: (private IP's) 10.10.1.31 .36 is a floating IP assigned by Linux-HA (heartbeat/pacemaker). I only want to use .36 to receive and send sip traffic. I uses force_send_socket() to send INVITE with .36 IP. But the ACK message always want to use .34 IP even the Route header has .36 in it unless I force it with force_send_socket() . How can I fix this problem? See below for the SIP messages: (x.x.128.205 is customer IP, x.x.129.200 is PSTN gateway IP) U x.x.128.205:51694 - x.x.130.36:5060 INVITE sip:5033441174@x.x.130.36:5060mailto:sip:5033441174@x.x.130.36:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9hG 4bK1D3CD1..From: sip:5024427578@x.x.128.205mailto:sip:5024427578@x.x.128.205;tag=24513088-D59..To: sip:5033441174@x.x.130.3mailto:sip:5033441174@x.x.130.3 6..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.20mailto:1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.20 5..Supported: 100rel,timer,replaces..Min-SE: 1800..Cisco-Guid: 411443261-3293254113-3191264919-256 0877466..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, CO MET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..CSeq: 101 INVITE..Max-Forwards: 70..Remote-P arty-ID: sip:5024427578@x.x.128.205mailto:sip:5024427578@x.x.128.205;party=calling;screen=no;privacy=off..Timestamp: 1341322661 ..Contact: sip:5024427578@x.x.128.205:5060mailto:sip:5024427578@x.x.128.205:5060..Expires: 180..Allow-Events: telephone-event..Conte nt-Type: application/sdp..Content-Length: 366v=0..o=CiscoSystemsSIP-GW-UserAgent 9094 579 IN IP 4 x.x.128.205..s=SIP Call..c=IN IP4 x.x.128.205..t=0 0..m=audio 19312 RTP/AVP 125 0 18 100 10 1..c=IN IP4 x.x.128.205..a=rtpmap:125 X-CCD/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:18 G729/8000..a =fmtp:18 annexb=yes..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194..a=rtpmap:101 telephone-event/8000 ..a=fmtp:101 0-16.. # U x.x.130.36:5060 - x.x.128.205:51694 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9 hG4bK1D3CD1;rport=51694..From: sip:5024427578@x.x.128.205mailto:sip:5024427578@x.x.128.205;tag=24513088-D59..To: sip:502244117 4@x.x.130.36mailto:4@x.x.130.36..Call-ID: 1887588d-c44b11e1-be38d697-98a3e...@x.x.128.205..cseqmailto:1887588d-c44b11e1-be38d697-98a3e...@x.x.128.205..cseq: 101 INVITE..Se rver: LVS Proxy 1.0..Content-Length: 0 # U x.x.130.36:5060 - x.x.129.200:5060 INVITE sip:15033441174@x.x.129.200:5060mailto:sip:15033441174@x.x.129.200:5060 SIP/2.0..Record-Route: sip:x.x.130.36;lr=on..Via: S IP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0..Via: SIP/2.0/UDP x.x.128.205:5060;rport =51694;branch=z9hG4bK1D3CD1..From: sip:5024427578@x.x.128.205mailto:sip:5024427578@x.x.128.205;tag=24513088-D59..To: sip:50224 41174@x.x.130.36mailto:41174@x.x.130.36..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3 E39A@x.x.128.205..Supportedmailto:E39A@x.x.128.205..Supported: 100rel,timer,replaces..Min-SE: 1800..Cisco-Guid: 411443261-3293254 113-3191264919-2560877466..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANC EL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..CSeq: 101 INVITE..Max-Forw ards: 69..Remote-Party-ID: sip:5024427578@x.x.128.205mailto:sip:5024427578@x.x.128.205;party=calling;screen=no;privacy=off..Tim estamp: 1341322661..Contact: sip:5024427578@x.x.128.205:51694mailto:sip:5024427578@x.x.128.205:51694..Expires: 180..Allow-Events: tel ephone-event..Content-Type: application/sdp..Content-Length: 375v=0..o=CiscoSystemsSIP-GW-UserA gent 9094 579 IN IP4 10.200.1.51..s=SIP Call..c=IN IP4 10.200.1.51..t=0 0..m=audio 20464 RTP/AVP 12 5 0 18 100 101..c=IN IP4 10.200.1.51..a=rtpmap:125 X
[SR-Users] Question about using LCR wit multiple gateways
Kamilio 3.2.0 I'd like to use one gateway as primary gateway and the another gateway as backup for failover. I could not make it to work. Here is my table entries: Lcr_gw: +++-+-+--+--+++---+---++--+---+-+ | id | lcr_id | gw_name | ip_addr | hostname | port | params | uri_scheme | transport | strip | prefix | tag | flags | defunct | +++-+-+--+--+++---+---++--+---+-+ | 1 | 2 | gateway1 | 10.10.1.1 | NULL | 5060 | NULL | 1 | 0 | NULL | NULL | NULL | 0 |NULL | | 2 | 2 | gateway2 | 10.10.1.2 | NULL | 5060 | NULL | 1 | 0 | NULL | NULL | NULL | 0 |NULL | Lcr_rule: +---+++--+-+- | id| lcr_id | prefix | from_uri | stopper | enabled +---+++--+-+ | 1 | 2 | 1 | NULL | 0 | 1 | 2 | 2 | 011| NULL | 0 | 1 +---+++--+-+ Lcr_rule_target: +---++-+---+--++ | id| lcr_id | rule_id | gw_id | priority | weight | +---++-+---+--++ | 1 | 2 | 1| 1 |9 | 1 | | 2 | 2 | 1 | 2 |8 | 1 | | 3 | 2 | 2 | 1 |9 | 1 | | 4 | 2 | 2 | 2 |8 | 1 | When making call , it only uses the first gateway. If first gateway failed, it could not find second gateway. What is the correct table entry for this to work? Thanks. Gary ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Problem with adding extra field in acc table.
I using Mysql DB. I need to add extra fields in acc table. The field having problem is the last one 'account' Here is my code in kamailio.cfg: ... modparam(acc, db_extra, src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;user_agent=$hdr(User-Agent);account=$var(s:account_number)) ... route { dp_translate(3, $fU/$avp(s:dst)) $v(s:account_number) = $avp(s:dst); } By looking at the xlog message, $v(s:account_number) has correct value. But the value in acc table is not always correct. Sometime account field populated with 0 and sometime it populated with wrong account number. It seems that $v(s:account_number) sometime corrupted. Am I doing something wrong? Gary ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] What is the valid values for flags field in dispatcher table?
Trying to figure out the values for flags field. where can I find the meaning of flags for each value like: flags 1 = ? flags 2 = ? flags 3 = ? flags 4 = ? flags 5 = ? flags 6 = ? flags 7 = ? flags 8 = ? Gary ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Download Kamailio Debian package problem.
I am trying to download GPG key as following and got connection timeout: wget http://deb.kamailio.org/kamailiodebkey.gpg It seems that the server is down. Where else can I download this same Debian package? Gary ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Questino about dialplan module
Kamailio 3.1.2 I am testing out dialplan module. Every time when I issue 'kamctl dialplan show', it shows what is in the database even if I have not yet issue kamctl dialplan reload' after I changed data in database. I thought that dialplan is stored in the memory and you have to issue the reload before you can see the change. Why the ' kamctl dialplan show' display the data directly from mysql database? Does that mean that dialplan data is not stored in the memory? Gary ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Questino about dialplan module
), 1 (str:int), 2 (int:str), 3 (int:int)) - sip-id must be an AoR (username@domain) - uuid must be a string but not AoR -- command 'alias_db' - manage database aliases alias_db show alias .. show alias details alias_db list sip-id . list aliases for uri alias_db add alias sip-id .. add an alias (*) alias_db rm alias remove an alias (*) alias_db help .. help message - alias must be an AoR (username@domain) - sip-id must be an AoR (username@domain) -- command 'domain' - manage local domains domain reload ... reload domains from disk domain show . show current domains in memory domain showdb ... show domains in the database domain add domain . add the domain to the database domain rm domain .. delete the domain from the database -- command 'cisco_restart' - restart CISCO phone (NOTIFY) cisco_restart uri restart phone configured for uri -- command 'online' - dump online users from memory online . display online users -- command 'monitor' - show internal status monitor show server's internal status -- command 'ping' - ping a SIP URI (OPTIONS) ping uri . ping uri with SIP OPTIONS -- command 'ul|alias' - manage user location or aliases ul show [username] show in-RAM online users ul show --brief. show in-RAM online users in short format ul rm username [contact URI] delete user's usrloc entries ul add username uri introduce a permanent usrloc entry ul add username uri expires .. introduce a temporary usrloc entry -- command 'fifo' fifo ... send raw FIFO command -- command 'cisco_restart' - restart CISCO phone (NOTIFY) cisco_restart uri restart phone configured for uri -- command 'online' - dump online users from memory online . display online users -- command 'monitor' - show internal status monitor show server's internal status -- command 'ping' - ping a SIP URI (OPTIONS) ping uri . ping uri with SIP OPTIONS -- command 'ul|alias' - manage user location or aliases ul show [username] show in-RAM online users ul show --brief. show in-RAM online users in short format ul rm username [contact URI] delete user's usrloc entries ul add username uri introduce a permanent usrloc entry ul add username uri expires .. introduce a temporary usrloc entry -Original Message- From: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Juha Heinanen Sent: Friday, June 03, 2011 3:50 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing List Subject: [SR-Users] Questino about dialplan module Gary Chen writes: I am testing out dialplan module. Every time when I issue 'kamctl dialplan show', it shows what is in the database even if I have not yet issue kamctl dialplan reload' after I changed data in database. I thought that dialplan is stored in the memory and you have to issue the reload before you can see the change. Why the ' kamctl dialplan show' display the data directly from mysql database? Does that mean that dialplan data is not stored in the memory? there is no mi function to show dialplan rules in memory. i have no idea, what 'show' does. please read readme of dialplan module before asking questions about the module. -- juha ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Problem using dispatcher.
-2c94..Call-ID: 00082166-efcb0013-692f44d9-6760a...@10.10.1.144..cseq: 101 INVITE..Server: kamailio (3.1.2 (x86_64/linux))..Content-Length: 0 # U 10.10.1.144:51030 - 10.10.1.230:5060 ACK sip:1...@fs2000.testnet.net;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d..From : Line1 sip:5022441...@fs2000.testnet.net;tag=00082166efcb002153bc0a31-20004ffd..To: sip:1...@fs2000.testnet.ne t;user=phone;tag=a6a1c5f60faecf035a1ae5b6e96e979a-2c94..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..D ate: Thu, 02 Jun 2011 19:14:14 GMT..CSeq: 101 ACK..Content-Length: 0 Gary Chen ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Problem using dispatcher.
Thanks for the quick reply. How can I fix this? We are using Kamailio as single point of entry for all our phones. Our phones all registered with Freeswitch. Kamailio only route the SIP traffic. Is there any way I can make Kamailio not treat 407 as failure? I also have the same problem with REGISTER messages when Freeswitch reply with '401 Unauthorized'. I end up using Forward() instead of t_relay() to make the phone1 register with Freeswitch. Any suggestion? -Original Message- From: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Alex Balashov Sent: Thursday, June 02, 2011 3:30 PM To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] Problem using dispatcher. On 06/02/2011 03:26 PM, Gary Chen wrote: the dispatcher always get into FAILURE route although the log shows that kamailio did send INVITE to 10.10.1.222 and received '407 - Proxy Authentication Required' from Freexwitch1 and then dispatcher send INVITE to 10.10.1.223 and also received '407 - Proxy Authentication Required' from Freexwitch2 and eventually failed. That's because any reply = 300 is considered a failure. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] How to get rid of this error message.
I am using prefix2domain(2) in PDT module like this: $var(found) = prefix2domain(2); and I got this error: ERROR: core [lvalue.c:411]: assignment failed at pos: (701,56-701,56) Apparently when prefix2domain(2) returns an value to $var(found), it does not like it. But it still work. The $var(found) still contains the return value (-1) if no prefix found. And all my configuration is working. It just generate a lot of this kind of error in my log file. Am I doing something wrong? How can I fix it? Gary ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] How to get rid of this error message.
Thanks. This fixed error. Is $rc the same as $retcode ? Gary -Original Message- From: Daniel-Constantin Mierla [mailto:mico...@gmail.com] Sent: Friday, April 29, 2011 1:01 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Cc: Gary Chen Subject: Re: [SR-Users] How to get rid of this error message. Hello, On 4/29/11 4:32 PM, Gary Chen wrote: I am using prefix2domain(2) in PDT module like this: $var(found) = prefix2domain(2); and I got this error: ERROR: core [lvalue.c:411]: assignment failed at pos: (701,56-701,56) Apparently when prefix2domain(2) returns an value to $var(found), it does not like it. But it still work. The $var(found) still contains the return value (-1) if no prefix found. And all my configuration is working. It just generate a lot of this kind of error in my log file. Am I doing something wrong? How can I fix it? I will look at it, it might be that the -1 return code is usually associated with an error case. Meanwhile try to replace that line with the following: prefix2domain(2); $var(found) = $rc; Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Error when using dispatcher module
Kamailio 3.1.2 I am using Kamailio in front of Freeswitch as simple sip router. To do multi-tenant in Freeswitch I created SIP domain (fs2000.ly.net) and used as sip domain inside my sip phone. This domain does not map to any real IP and no entry in our DNS server. If I sent the call directly to Freeswitch, it works fine and registered with Freeswitch. When I tried to go through Kamailio using dispatcher to route the call to freeswitch, I got this error: ERROR: core [resolve.c:1540]: ERROR: sip_hostport2su: could not resolve hostname: fs2000.ly.net The strange thing is that sometime the call did go through kamailio and registered with freeswitch. Whenever I restart Kamailio, I got above error and then the error went away by itself. Sometime took a minute and sometime took several hours. And sometime the error come back without restart. Does anybody know why? Why dispatcher want to resolve the hostname when it only lookup dispatcher for destination IP? Gary ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] ERROR: failover functions used, but AVPs paraamters required are NULL
kamailio version 3.1.2 I am trying to setup dispatcher to use its failover feature. Here is dispatcher part of configure file: loadmodule dispatcher.so modparam(dispatcher, db_url, DBURL) modparam(dispatcher, table_name, dispatcher) modparam(dispatcher, ds_ping_interval, 30) modparam(dispatcher, ds_probing_threshhold, 10) modparam(dispatcher, ds_ping_reply_codes, class=2;class=4) modparam(dispatcher, ds_probing_mode, 1) modparam(dispatcher, ds_ping_from, sip:lb1 at m-lab-ca805-sig.kd-lab.de) modparam(dispatcher, dst_avp, $avp(dsdst)) modparam(dispatcher, grp_avp, $avp(dsgrp)) modparam(dispatcher, cnt_avp, $avp(dscnt)) modparam(dispatcher, attrs_avp, $avp(dsattrs)) modparam(dispatcher, dstid_avp, $avp(dsdstid)) modparam(dispatcher, flags, 2) But it is still give the following error: 0(1917) ERROR: dispatcher [dispatcher.c:624]: failover functions used, but AVPs paraamters required are NULL -- feature disabled Does anybody know why? Gary ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] ERROR: failover functions used, but AVPs paraamters required are NULL
Never mind. I used wrong kamailio.cfg file. From: Daniel-Constantin Mierla [mailto:mico...@gmail.com] Sent: Thursday, February 17, 2011 8:51 AM To: Gary Chen Cc: sr-users@lists.sip-router.org Subject: Re: [SR-Users] ERROR: failover functions used, but AVPs paraamters required are NULL Hello, can you send all log messages printed by: kamailio -E -ddd Thanks, Daniel On 2/17/11 2:42 PM, Gary Chen wrote: kamailio version 3.1.2 I am trying to setup dispatcher to use its failover feature. Here is dispatcher part of configure file: loadmodule dispatcher.so modparam(dispatcher, db_url, DBURL) modparam(dispatcher, table_name, dispatcher) modparam(dispatcher, ds_ping_interval, 30) modparam(dispatcher, ds_probing_threshhold, 10) modparam(dispatcher, ds_ping_reply_codes, class=2;class=4) modparam(dispatcher, ds_probing_mode, 1) modparam(dispatcher, ds_ping_from, sip:lb1 at m-lab-ca805-sig.kd-lab.de) modparam(dispatcher, dst_avp, $avp(dsdst)) modparam(dispatcher, grp_avp, $avp(dsgrp)) modparam(dispatcher, cnt_avp, $avp(dscnt)) modparam(dispatcher, attrs_avp, $avp(dsattrs)) modparam(dispatcher, dstid_avp, $avp(dsdstid)) modparam(dispatcher, flags, 2) But it is still give the following error: 0(1917) ERROR: dispatcher [dispatcher.c:624]: failover functions used, but AVPs paraamters required are NULL -- feature disabled Does anybody know why? Gary ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Error when tring to load lcr module.
I am using Kamailio 3.1.2. I added lcr module in my kamailio.cfg file. After run this: ./kamctl -E I got this error: 0(23744) ERROR: lcr [lcr_mod.c:1235]: lcr_gw params at row 0 does not start with ';' 0 I have set up some gateway information in lcr_gw , lcr_rule and lcr_rule_target. From log it seems retrieve all the information but then got above error. Here is my lcr part of config: loadmodule lcr.so modparam(lcr, db_url, DBURL) modparam(lcr, gw_uri_avp, $avp(i:709)) modparam(lcr, ruri_user_avp, $avp(i:500)) modparam(lcr, flags_avp, $avp(i:712)) Gary ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users