Thanks Alex for you reply.
I understand the dispatcher module can be used to load balance inbound requests. But my questions is more related with outbound requests thru Kamailio.
Are you saying that in-dialog request can be routed automatically by Kamailio to the correct Asterisk instance on the back end ?
And this is included in the stock config ?
I was just looking for an example.
Thanks, --Jignesh
On 3/30/16, 6:00 AM, "sr-users on behalf of sr-users-request@lists.sip-router.org" <sr-users-bounces@lists.sip-router.org on behalf of sr-users-request@lists.sip-router.org> wrote:
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Today's Topics:
- Destination Timeout (ACK message) (Al S)
- Proxy configuration (Jignesh Gandhi)
- Re: Proxy configuration (Alex Balashov)
- Local Kamailio share a hosted Kamailio/Asterisk sip server (S?bastien Brice)
- SQlite DB Lock with Presence (Kristian Kielhofner)
- Kamailio local and hosted (S?bastien Brice)
- Re: Releasing Kamailio v4.4.0 (Daniel-Constantin Mierla)
- Re: SQlite DB Lock with Presence (Daniel-Constantin Mierla)
Message: 1 Date: Tue, 29 Mar 2016 09:27:56 -0400 From: Al S alirsm@outlook.com To: "sr-users@lists.sip-router.org" sr-users@lists.sip-router.org Subject: [SR-Users] Destination Timeout (ACK message) Message-ID: BAY177-W1513432E7A3AFDAAB1DE8DD4870@phx.gbl Content-Type: text/plain; charset="iso-8859-1"
Hi, The destination node is down. By adding the following code Kamailo re-routing INVITE to a new destination after getting 408 timeout (as below).
failure_route[MANAGE_FAILURE]{ if (t_branch_timeout()) { # change to new destination IP $rd = "10.10.10.10"; t_relay(); exit; }...} However, the ACK message coming from source is still being send to the old destination. How can I reroute other messages in the dialog to be sent to this new destination.. Thank, AS -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.sip-router.org/pipermail/sr-users/attachments/20160329/b91e2df1/attachment-0001.html
Message: 2 Date: Tue, 29 Mar 2016 14:24:01 +0000 From: Jignesh Gandhi Jignesh.Gandhi@MoviusCorp.com To: "sr-users@lists.sip-router.org" sr-users@lists.sip-router.org Subject: [SR-Users] Proxy configuration Message-ID: 011222F3-5820-41FB-A0D8-8D727EA121FA@moviuscorp.com Content-Type: text/plain; charset="utf-8"
Hello,
Is it possible to use kamailio to be a stateful outbound proxy ?
The use case is kamailio is an outbound proxy for multiple instances of Asterisk. Thus all responses need to go back to the same instance of Asterisk that generated them.
After the call is established. How can I ensure that the Re-Invites, BYEs, etc are sent to the same instance of Asterisk that generated the outbound Invite ?
Any help is appreciated.
Thanks, --Jignesh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.sip-router.org/pipermail/sr-users/attachments/20160329/5081718c/attachment-0001.html
Message: 3 Date: Tue, 29 Mar 2016 10:29:19 -0400 From: Alex Balashov abalashov@evaristesys.com To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] Proxy configuration Message-ID: 56FA913F.6080903@evaristesys.com Content-Type: text/plain; charset=windows-1252; format=flowed
Jignesh,
Everything you're asking for is part of Kamailio's core functionality:
- Most useful applications of Kamailio involve stateful relay, and the
stock included configurations are stateful.
- The loose-routing section of the configuration, and, more broadly,
RFC 3261 in general, take care of routing in-dialog requests (ACKs, reinvites, BYEs, etc.) to the appropriate dialog party.
- Load-balancing/distribution among Asterisk instances can be done with
the 'dispatcher' module; there are many examples floating around, as it's one of the most common applications for Kamailio.
-- Alex
-- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States
Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
Message: 4 Date: Tue, 29 Mar 2016 16:32:36 +0200 (CEST) From: S?bastien Brice sebastien.brice@jvs.fr To: sr-users sr-users@lists.sip-router.org Subject: [SR-Users] Local Kamailio share a hosted Kamailio/Asterisk sip server Message-ID: 2011796741.12246629.1459261956126.JavaMail.zimbra@jvs.fr Content-Type: text/plain; charset=utf-8
Hi everyone At my workplace there's plenty of hardphone directly connected to an hosted sbc in the cloud. As an added value i proposed to add locally Kamailio as a proxy server to 'federate' all the sip traffic.
ipphone behind nat (A) <---> sip proxy local 192.168.x.x (B)<----> sbc registrar (mydomain.com) and sip proxy with public IP (C)
when hardphone try to register, B rewrite (with rewritehost) the ip public of (C). B then uses exported function of TM to t_relay_to(C). to route INVITE?from A to C i use proxy_authenticate("mydomain.com") function (on B) from auth_db to pass the calls inbetween.
This works well, but i am wondering if that makes sense if i get several phones behind B. Should stateless forward works better in this kind of scenario?
In a near future i plan to offer disaster recovery, and i want all users binding to C (users@mydomain.com)instead bind on B (in case link B---C get down)and assume some kind of telephone functionnality Is that possible ? Do you have some sort of example Kamailio do it good On-premise/localy ? How to deal with nat since all hardphones have rfc1918?address?
thx you
PS: as an sbc (C) i did setup kamailio/asterisk in a real-time configuration.
S?bastien BRICE VoIP, Support et Int?gration
Message: 5 Date: Tue, 29 Mar 2016 20:50:14 -0400 From: Kristian Kielhofner kris@kriskinc.com To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" sr-users@lists.sip-router.org Subject: [SR-Users] SQlite DB Lock with Presence Message-ID: CAKDfjge4kLhdLjV5V7gsOOsLsdByeLgtwnqE-EEXHhxbVeV8Ag@mail.gmail.com Content-Type: text/plain; charset=UTF-8
Hello everyone,
I'm trying to use BLF with the config here (except with SQlite):
http://kb.asipto.com/kamailio:presence:k43-blf
BLF seems to work perfectly except after the call is ended I get "sqlite commit failed: database is locked" error messages as seen here:
BLF is then broken at this point.
kamailio -v version: kamailio 4.3.5 (x86_64/linux) 950657 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_PTHREAD_MUTEX, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 950657 compiled on 01:33:01 Mar 29 2016 with x86_64-openwrt-linux-gnu-gcc 4.8.3
Has anyone ever seen this before?
Thanks!
-- Kristian Kielhofner
Message: 6 Date: Tue, 29 Mar 2016 14:38:40 +0200 (CEST) From: S?bastien Brice sebastien.brice@jvs.fr To: sr-users sr-users@lists.sip-router.org Subject: [SR-Users] Kamailio local and hosted Message-ID: 1194659979.12170808.1459255120868.JavaMail.zimbra@jvs.fr Content-Type: text/plain; charset=utf-8
Hi everyone At my workplace there's plenty of hardphone directly connected to an hosted sbc in the cloud. As an added value i proposed to add locally Kamailio as a proxy server to 'federate' all the sip traffic.
ipphone behind nat (A) <---> sip proxy local 192.168.x.x (B)<----> sbc registrar (mydomain.com) and sip proxy with public IP (C)
when hardphone try to register, B rewrite (with rewritehost) the ip public of (C). B then uses exported function of TM to t_relay_to(C). to route INVITE?from A to C i use proxy_authenticate("mydomain.com") function (on B) from auth_db to pass the calls inbetween.
This works well, but i am wondering if that makes sense if i get several phones behind B. Should stateless forward works better in this kind of scenario?
In a near future i plan to offer disaster recovery, and i want all users binding to C (users@mydomain.com)instead bind on B (in case link B---C get down)and assume some kind of telephone functionnality Is that possible ? Do you have some sort of example Kamailio do it good On-premise/localy ? How to deal with nat since all hardphones have rfc1918?address? , thx you
PS: as an sbc (C) i did setup kamailio/asterisk in a real-time configuration.
S?bastien BRICE VoIP, Support et Int?gration
Message: 7 Date: Wed, 30 Mar 2016 10:02:02 +0200 From: Daniel-Constantin Mierla miconda@gmail.com To: "Kamailio (SER) - Users Mailing List" sr-users@lists.sip-router.org, sr-dev sr-dev@lists.sip-router.org, "business@lists.kamailio.org" business@lists.kamailio.org Subject: Re: [SR-Users] Releasing Kamailio v4.4.0 Message-ID: 56FB87FA.8080304@gmail.com Content-Type: text/plain; charset=utf-8
Hello,
the final packaging for 4.4.0 is expected to start after 14:00UTC today. Backports to branch 4.4 can be done up to 12:00UTC, after that let's sync on sr-dev mailing list, because I will start testing how it builds on different platforms and it is good to have the a consistent source code.
Cheers, Daniel
On 29/03/16 09:04, Daniel-Constantin Mierla wrote:
Hello,
refreshing about the major release 4.4.0 planned for tomorrow -- if any dev has patches to backport to branch 4.4 should do it as soon as possible.
Cheers, Daniel
On 18/03/16 12:16, Daniel-Constantin Mierla wrote:
Hello,
with no major issues reported lately, I think we are on a good track for releasing stable v4.4.0. Next week a lot of people prepare for Easter, therefore I propose to do the release on Wednesday, March 30, 2016. If new things pop up, the exact date can be adjusted a bit before or after the current proposal.
Any new discovered issue should be reported to bug trucker:
Help to complete the upgrade guide would be very appreciated:
A draft list of what is new in 4.4 is already available at:
Cheers, Daniel
-- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com
Message: 8 Date: Wed, 30 Mar 2016 11:37:39 +0200 From: Daniel-Constantin Mierla miconda@gmail.com To: "Kamailio (SER) - Users Mailing List" sr-users@lists.sip-router.org Subject: Re: [SR-Users] SQlite DB Lock with Presence Message-ID: 56FB9E63.7040104@gmail.com Content-Type: text/plain; charset=windows-1252
Hello,
not using sqlite here, but could be related to presence modules doing db transactions and not releasing the locks. Do you get the issue after the first call, or it runs for a while and at some point it breaks?
Cheers, Daniel
On 30/03/16 02:50, Kristian Kielhofner wrote:
Hello everyone,
I'm trying to use BLF with the config here (except with SQlite):
http://kb.asipto.com/kamailio:presence:k43-blf
BLF seems to work perfectly except after the call is ended I get "sqlite commit failed: database is locked" error messages as seen here:
BLF is then broken at this point.
kamailio -v version: kamailio 4.3.5 (x86_64/linux) 950657 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_PTHREAD_MUTEX, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 950657 compiled on 01:33:01 Mar 29 2016 with x86_64-openwrt-linux-gnu-gcc 4.8.3
Has anyone ever seen this before?
Thanks!
-- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com
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