sr-users August 2015

sr-users@lists.kamailio.org

79 participants
123 discussions

by Daniel-Constantin Mierla

Hello, Kamailio SIP Server v4.2.6 stable release is out. This is a maintenance release of the previous stable branch, 4.2, that includes fixes since release of v4.2.5. There is no change to database schema or configuration language structure that you have to do on installations of v4.2.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.2.6 (or to 4.3.x series). For more details about version 4.2.6 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/07/kamailio-v4-2-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 4.3, at this moment with its latest release v4.3.1. See more details about it at: * http://www.kamailio.org/w/kamailio-v4-3-0-release-notes/ Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/miconda - http://www.linkedin.com/in/miconda

9 years, 3 months

Kamailio Redis support

by Solutions Solutions

Hi All, i want to replace MySQL with redis database in kamalio,but in kamalio supports transactions with MySQL ,ie saving data in MySql DB.but while trying to run kamailio with redis ,transaction are not happening,is there any module that i have to enable in kamalio for No SQL databases transaction support.

9 years, 3 months

Re: [SR-Users] Remote SIP Registration

by Keith

Hi, I have a carrier who require registration, I have looked at the uac module but can't see how to do this. Anyone point me in the right direction? Thanks Keith

9 years, 3 months

Difference between corex module's 'alias_subdomains' and 'alias' core parameter

by Iwan Aucamp

>From the descriptions these seem similar: corex alias_subdomains ( http://kamailio.org/docs/modules/devel/modules/corex.html#idp18017160): Register a domain and all its sub-domains to match the “myself” condition. It can be set many times. Its full format is: 'proto:domain:port', allowing to set restrictions on protocol and port as well. Protocol and port are optional. alias core parameter ( http://www.kamailio.org/wiki/cookbooks/devel/core#alias): Parameter to set alias hostnames for the server. It can be set many times, each value being added in a list to match the hostname when 'myself' is checked. Is the only difference that corex alias_subdomains can include 'proto' ? -- Iwan Aucamp

9 years, 3 months

Question about how location server_id is supposed to be used.

by Ding Ma

Looking for some doc that describes how the new columns in 4.3.x location table is supposed to be used. Can find some email discussions about this, but seems no formal doc exists. Would appreciate if someone can provide an explanation. Thanks

9 years, 3 months

Re: [SR-Users] R: Mutiple branch forking and voicemail

by Daniel-Constantin Mierla

Indeed usually there are few options for same functionality, good to hear you found a solution. Cheers, Daniel On 03/08/15 14:53, Marino Mileti wrote: > Thanks Daniel, > > I've found another way using a seturi for mediaserver that will be executed > after all branch (using t_load_contacts and t_next_contacts) > In this way I can complete my scenario in the right way > > -----Messaggio originale----- > Da: sr-users [mailto:sr-users-bounces@lists.sip-router.org] Per conto di > Daniel-Constantin Mierla > Inviato: lunedì 3 agosto 2015 12:37 > A: Kamailio (SER) - Users Mailing List > Oggetto: Re: [SR-Users] Mutiple branch forking and voicemail > > Hello, > > On 31/07/15 11:31, Marino Mileti wrote: >> Hello, >> >> I would like to implement a voicemail services but i'm trying >> everything without good results :( >> >> My scenario is: >> >> - A make a parallel forking to B(1001) and C(1002) >> - If timeout, call is routed to voicemail >> >> I've done something like this...very simple >> >> seturi("sip:1001@172.20.40.103"); >> >> append_branch("1002(a)172.20.40.103"); >> route(RELAY) >> >> on failure_route.. >> if (t_check_status("486|408")) { >> $ru = "sip:8000@172.20.40.103:5090"; >> >> route(RELAY); >> >> exit; >> >> } >> >> The scenario is working but on log there are some error... >> t_should_relay_response >> >> I would like that when a branch reach the failure route, all other >> branches will be cancelled..and call is routed to VM. >> Can anyone help me? >> >> > When the failure_route is executed, no branch is active anymore. > > To be able to assist further, you have to send here the errors you get in > the log, to understand what is wrong. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Book: SIP Routing With Kamailio - http://www.asipto.com > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > --- > Questa e-mail è stata controllata per individuare virus con Avast antivirus. > https://www.avast.com/antivirus > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

9 years, 3 months

Maintaining order of messages in a sip dialog on forwarding

by Daniel Tryba

Since switching to 4.2.5 (from 3.x) a customer has problems with a buggy user agent (IMHO). What I am seeing now is: provider kamailio customer INVITE1-> INVITE1-> <-Trying <-Trying <-200 OK <-200 OK ACK1-> INVITE2-> INVITE2-> ACK1-> ACK1 arrives 0.007s before INVITE2, but INVITE2 is send 0.004s before ACK1. It looks like this wasn't happening with Kamailio 3.x, it worked fine then (apparently). Can this reordering be (gracefully) prevented somehow (ignoring the fact that during transport reordering can happen)? A Q&D hack looks to be just usleep the reinvite a little. The endpoint is essentially ignoring the reinvite before (and after) the ACK. After a timeout INVITE2 is canceled and after that INVITE1 will be ended with a BYE from provider. kamailio-customer communication is UDP provider-kamailio is tcp when message >= 1300b. INVITE1 is TCP, INVITE2 and ACK1 are UDP.

9 years, 3 months

VIA header question.

by Chad

Hi list, I need a little help, I am a business owner trying to get Kamailio up and running as a SIP load balancer. I hired a Kamailio consultant to help me do so, but Kamailio is not working and I am getting conflicting information. My Kamailio consultant and my VOIP provider are telling me 2 different things and I don't know which one is right. Kamailio sends SIP traffic to the VOIP provider with 2 VIA headers like this (in this order): Via: SIP/2.0/UDP 10.10.10.254;branch=z9hG4bK7291.6a0bbd2e8fd639a47d7d2de606779e47.0. Via: SIP/2.0/UDP 209.170.201.25:5060;received=10.10.10.102;branch=z9hG4bK742dc03d;rport=5060. The VOIP provider says that is incorrect because they are supposed to reply back to the topmost VIA header so they reply to the 10.10.10.254 IP (which is not public) and the call ends. The VOIP provider says Kamailio should send the VIA headers like this instead: Via: SIP/2.0/UDP 209.170.201.25:5060;received=10.10.10.102;branch=z9hG4bK742dc03d;rport=5060. Via: SIP/2.0/UDP 10.10.10.254;branch=z9hG4bK7291.6a0bbd2e8fd639a47d7d2de606779e47.0. My Kamailio consultant says the way we are sending it is right and that the VOIP provider is processing the call incorrectly. I read that the SIP proxy is supposed to remove the internal header from the 1st example above based on this RFC: https://tools.ietf.org/html/rfc3261#section-16.7 Item: 3. Via "The proxy removes the topmost Via header field value from the response." If that applies to this situation (which I don't know if it does) then Kamailio should be removing the 10.10.10.254 VIA line and only sending 1 VIA header like this: Via: SIP/2.0/UDP 209.170.201.25:5060;received=10.10.10.102;branch=z9hG4bK742dc03d;rport=5060. Which would sort of make the VOIP provider right in that the topmost VIA line would then be the external IP, but how they said to fix it (reversing the VIA lines) is wrong. Does anyone know what the right answer is here? Please let me know. THANKS for your help. -- ^C

9 years, 3 months

MSILO: Archiving every message

by SamyGo

Hi, Tried finding anything as weird as I am trying to do on mailing list but couldn't. The idea is to save a copy of the messages on database. MSILO module can keep the message as long as the destination user is offline, or doesn't support method:MESSAGE or expiry timeout. Any ideas will be appreciated. Best Regards, Sammy

9 years, 3 months

Re: [SR-Users] Mutiple branch forking and voicemail

by Daniel-Constantin Mierla

Hello, On 31/07/15 11:31, Marino Mileti wrote: > Hello, > > I would like to implement a voicemail services but i'm trying everything > without good results :( > > My scenario is: > > - A make a parallel forking to B(1001) and C(1002) > - If timeout, call is routed to voicemail > > I've done something like this...very simple > > seturi("sip:1001@172.20.40.103"); > > append_branch("1002(a)172.20.40.103"); > route(RELAY) > > on failure_route.. > if (t_check_status("486|408")) { > $ru = "sip:8000@172.20.40.103:5090"; > > route(RELAY); > > exit; > > } > > The scenario is working but on log there are some error... > t_should_relay_response > > I would like that when a branch reach the failure route, all other branches > will be cancelled..and call is routed to VM. > Can anyone help me? > > When the failure_route is executed, no branch is active anymore. To be able to assist further, you have to send here the errors you get in the log, to understand what is wrong. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

9 years, 3 months

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sr-users August 2015