Re: [SR-Users] Is ACK required in a re-invite?

2016-08-15 Thread Jay Li
ov wrote: On 08/15/2016 05:42 PM, Jay Li wrote: > Dear All, > > I just have a basic SIP question on re-invite. I wonder if "ACK" is > required in a re-invite scenario, like re-invite -> 200 OK -> ACK. What > I've seen is for some reason, my PSTN doesn't

[SR-Users] Is ACK required in a re-invite?

2016-08-15 Thread Jay Li
Dear All, I just have a basic SIP question on re-invite. I wonder if "ACK" is required in a re-invite scenario, like re-invite -> 200 OK -> ACK. What I've seen is for some reason, my PSTN doesn't send a "ACK" back to me after my 200 OK response to their re-invite. I don't know if not sending an

Re: [SR-Users] Video conferencing with Kamailio

2016-07-12 Thread Jay Li
Fred, Thanks a lot your detailed explanation. About the media server addition to Kamailio, do you have any suggestions I should look into besides Jitsi and FreeSWITCH? Thanks. Regards,Jay On Friday, July 8, 2016 7:32 AM, Fred Posner wrote: On 07/08/2016 12:36 AM, Jay Li wrote: > D

[SR-Users] Video conferencing with Kamailio

2016-07-07 Thread Jay Li
Dear All, I'm curious if anybody has set up an infrastructure for video conferencing utilizing Kamailio as a proxy (like NAT support and so so). I found a kind of old tutorial "Run you own Skype-like service in less than one hour"  kamailio:skype-like-service-in-less-than-one-hour [Asipto - SIP a

[SR-Users] A parameter with a life span of the entire dialog

2016-07-07 Thread Jay Li
Dear All, I'm trying to set a few parameters for an "INVITE", and access their values in the failure routine if the call fails. $var() doesn't seem suitable for my purpose as it only lasts for one process. I think if I set a $var() in "INVITE", it will be gone when I receive the failure response

Re: [SR-Users] re-Invite in the failure route

2016-07-03 Thread Jay Li
e_route: route {     ...     t_on_failure("FAILURE");     if(!t_relay())       sl_reply_error(); } failure_route[FAILURE] {     if(t_is_canceled())       exit;     $ru = "":     t_on_failure("FAILURE");     t_relay(); } -- Alex On 07/03/2016 09:14 PM, Jay Li wrote: > Dear All,

[SR-Users] re-Invite in the failure route

2016-07-03 Thread Jay Li
Dear All, Hopefully I'll be able to get some help here on re-INVITE in the failure route. For example I have a couple of PSTN gateway options. I'd like to try them by priorities. In case the one with top priority couldn't go through(e.g. returned 5XX on INVITE), I'd like to re-INVITE through ano

[SR-Users] Add a user through a http request

2016-06-21 Thread Jay Li
Dear All, I wonder if Kamailio has the functionality to add a subscriber dynamically. For example, if I use event_route[xhttp:request] to expect a http get request, then based on the received request, add a user in the DB. Not sure what the best approach would be. Any suggestion is appreciated.

[SR-Users] HTTP query at the end of a phone call

2016-06-14 Thread Jay Li
Dear All, I'd like to send a HTTP call to a HTTP server whenever a call routed through a Kamailio server ends. By looking at the document, seems I should use http_query() function in the utils module. I wonder which route function I should put the http_query()  in assuming we use the default kam

[SR-Users] The book of "SIP Routing with Kamailio"

2016-04-21 Thread Jay Li
Dear all, I'm looking for a document/book that explains the kamailio.cfg in detail. I've gone over the suggested study material "SER-Getting Started" which is such a nice book that explains ser.cfg line by line. I know Asipto has published a book named "SIP Routing with Kamailio". Just wondering

[SR-Users] Compatibility between Siremis 4.2 and Kamailio 4.4

2016-04-07 Thread Jay Li
Dear all, I got into the Kamailio world recently, and just wanted to get some info here about the compatibility between Siremis and Kamailio. From what I read,   Siremis is wonderful web management tool for Kamailio. However the latest Siremis to date is 4.2.0 which was released more than a year