[SR-Users] Inter domain routing

Hello, I have two kamailio servers running independently with different domain names. I want to route calls between those two servers. Can it be done ? If yes then How ? Thanks. ___ SIP Express Router (SER) and

Re: [SR-Users] Reminder: about private emails

On 12/03/14 10:25, jay binks wrote: I apologise if I was one of the offenders. Not really an offend, just wanted everyone is aware of really small chance to get any answer via private emails, so there won't be blames about lack of reaction. I do however have a question in relation to this t

Re: [SR-Users] Bizarre dialog profile tracking behaviour

Do you happen to have any additional information about this? I've referenced the documentation for both dialog and dialog_ng, and there's no mention of reuse that fits this description. We've confirmed that calling dlg_manage() on a new authenticated invite does result in a second entry in th

Re: [SR-Users] trixbox or asterisk as a media server for kamailio

Thanks, I will look into the link and yes I got plain Asterisk also. Thanks for your response. Abdul Date: Wed, 12 Mar 2014 09:51:15 -0600 From: carlos.ruizd...@gmail.com To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] trixbox or asterisk as a media server for kamailio Hi, have you l

Re: [SR-Users] trixbox or asterisk as a media server for kamailio

Hi, have you looked this [1] already? I would recommend dropping trixbox is favor of a plain Asterisk installation. It makes things easier to configure. [1] http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb Regards, On Wed, Mar 12, 2014 at 9:47 AM, malik sherif wrot

[SR-Users] trixbox or asterisk as a media server for kamailio

I am looking documentation as to how to integrate kamailio and Asterisk, I am not sure if I use the correct term "media server" but I would like to configure 3-way call, call waiting, call transfer , and call forwarding. Your help is greatly appreciated. Thanks Abdul

Re: [SR-Users] Reminder: about private emails

A variety of people and entities on the mailing list provide paid support. Please see the Kamailio Business Directory on the Web site. On 12 March 2014 05:25:35 GMT-04:00, jay binks wrote: >I apologise if I was one of the offenders. >I do however have a question in relation to this thread. > >D

Re: [SR-Users] too many recursive routing table lookups

On 11/03/14 15:05, Camila Troncoso wrote: Daniel, I have maximum 40 gateways to try. That’s why this error is strange. Indeed the error is presenting in calls having no more than 12 retries ( because of channel exceed or failure messages) before they could be relay (see the attached file)

[SR-Users] dialog is not deleted when BYE received

Hello, I disable the rr module and only use SIP elements for dialog match modparam("dialog", "dlg_match_mode", 2) Upon receipt of a BYE request, the request is correctly forwarded and the call ended but the dialog still exists with ref_count = 2 kamctl fifo dlg_list dialog:: hash=731:5167 st

Re: [SR-Users] Routing based on destination and source

Hi Yeah I do already use the lcr module for routing to external carriers but is probably a little overkill for what I need to route that traffic internally. Is that my only real option? Cheers Keith ___ SIP Express Router (SER) and Kamailio (OpenSER) -

Re: [SR-Users] Reminder: about private emails

I apologise if I was one of the offenders. I do however have a question in relation to this thread. Do you offer paid support for kamailio ? if so is there a web page with information about this option ( prices, whats on offer etc ). thanks Jay On 12 March 2014 19:10, Daniel-Constantin Mierla

Re: [SR-Users] Kamailio Proxy for Freeswitch

BTW... Daniel, you were 100% right... I had missed a record_route in part of my config sigh.. just me being a complete noob with kamailio .. thanks for your assistance. Jay On 12 March 2014 14:13, jay binks wrote: > I have record_route and loose_route in my config ( record route , wher

[SR-Users] Reminder: about private emails

A message sent almost one year ago, but based on volume of private emails related to free support on SIP, Kamailio or other applications, perhaps it is time to refresh, with couple of extra bits for clarification: - http://lists.sip-router.org/pipermail/sr-users/2013-April/077700.html >>> I

Re: [SR-Users] (no subject)

The support page of siremis project give details where and what should be addressed: - http://siremis.asipto.com/support/ People should understand that time is a valuable asset and it is impossible to offer free support to everyone. Moreover, any message to business coming from someone that d

Re: [SR-Users] Bizarre dialog profile tracking behaviour

On 11/03/14 18:44, Alex Balashov wrote: We appear to have fixed this problem by calling dlg_manage() before doing any set_dlg_profile() manipulations. The documentation is not clear on whether dlg_manage() needs to be called first before doing this. But it makes me wonder: if dlg_manage() is

Re: [SR-Users] Bizarre dialog profile tracking behaviour

Are you using the latest 4.1.x? There was a fix related to the counting of the dialog in profile, by removing it from profiles when the call gets to terminated state. The dialog is still kept for a bit in memory after termination and was blocking new calls as the user appear to still have anoth

Re: [SR-Users] Routing based on destination and source

Keith writes: > I am looking for the best module to route calls based on the authenticated > user that call came from and the destination number. Is there any current > module to do this? lcr module allows selection of gateways based on those two entities. see readme. -- juha __

Re: [SR-Users] Routing based on destination and source

Hi all, I am looking for the best module to route calls based on the authenticated user that call came from and the destination number. Is there any current module to do this? Many thanks Keith ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-us