[SR-Users] Where do I place the logic for modifying auth and call requests?

2013-11-08 Thread erik
Hello everyone. Some of my confusion probably comes from me coming from the web development world and not software dev, but I would greatly appreciate any response to the following question: I use the MySQL auth for my subscribers which is working great. However, let's say that I add the

Re: [SR-Users] Where do I place the logic for modifying auth and call requests?

2013-11-08 Thread Grant Bagdasarian
Take a look at: http://www.kamailio.org/wiki/cookbooks/4.0.x/pseudovariables It contains a list of variables you can use inside the kamailio.cfg script. The modules page also provides a lot of information: http://www.kamailio.org/docs/modules/4.0.x/ From: sr-users-boun...@lists.sip-router.org

[SR-Users] 1 via header

2013-11-08 Thread phillman25
Dear List I am trying to interconnect with a Cisco Call manager express, i am sending the INVITE message with the below format: INVITE sip:22775019@81.21.39.153 SIP/2.0 Record-Route: sip:81.21.38.33;lr=on;ftag=685329680;did=671.51a2 Via: SIP/2.0/UDP

Re: [SR-Users] Where do I place the logic for modifying auth and call requests?

2013-11-08 Thread erik
That's exactly what I'm after! Thanks a lot. 2013-11-08 01:03 skrev Grant Bagdasarian: Take a look at: http://www.kamailio.org/wiki/cookbooks/4.0.x/pseudovariables [2] It contains a list of variables you can use inside the kamailio.cfg script. The modules page also provides a lot

Re: [SR-Users] 1 via header

2013-11-08 Thread Daniel-Constantin Mierla
Hello, On 11/8/13 10:05 AM, phillman25 wrote: Dear List I am trying to interconnect with a Cisco Call manager express, i am sending the INVITE message with the below format: INVITE sip:22775019@81.21.39.153 mailto:sip%3A22775019@81.21.39.153 SIP/2.0 Record-Route:

Re: [SR-Users] kamailio 4.0.4 '408' response

2013-11-08 Thread Daniel-Constantin Mierla
I haven't seen any 408 response. Maybe you can run kamailio with debug=3 in config to get verbose output to syslog. Cheers, Daniel On 11/8/13 4:15 AM, kamai...@aaronlux.com wrote: Below please find a filtered SIP packet capture showing a problem I'm having with callee '408' responses in

Re: [SR-Users] #TM: timing problem in serial forking (INVITE vs. CANCEL)

2013-11-08 Thread Klaus Darilion
I think it would be nice if the CANCELs are sent before the INVITE. But this will never ensure the order how they are received at the client side. E.g. there can be packet loss which drops the CANCEL but not the INVITE, or with load balancing the INVITE can overtake the CANCEL. And if the

[SR-Users] UUID generation

2013-11-08 Thread Grant Bagdasarian
Hello, Any plans for building a module which generates a UUID or building it into the core? Daniel once told me to use $sruid, which basically returns a unique value. It works good. I haven't had any duplicates yet, even under heavy load tests. ___

Re: [SR-Users] PERMISSIONS module issue

2013-11-08 Thread Daniel Tryba
On Thursday 07 November 2013 19:53:47 Samuel Ware wrote: I having issue updating my allow list for the PERMISSIONS module. I added an address to the ADDRESS table. I have tried to do a service restart, kamctl address reload, and kamcmd permissions.addressReload. The kamctl address show

Re: [SR-Users] #TM: timing problem in serial forking (INVITE vs. CANCEL)

2013-11-08 Thread Klaus Feichtinger
I think it would be nice if the CANCELs are sent before the INVITE. But this will never ensure the order how they are received at the client side. E.g. there can be packet loss which drops the CANCEL but not the INVITE, or with load balancing the INVITE can overtake the CANCEL. And if the

[SR-Users] kamailio dialplan

2013-11-08 Thread Joli Martinez
I am new to Kamailio and am having an issue with the dialplan setup. I have Kamailio setup as an SBC to handle all user authentication and call routing. I need freeswitch to handle all conferences and voicemails. When I dial 433001 I would like to be transferred to freeswitch for conferences.

Re: [SR-Users] kamailio dialplan

2013-11-08 Thread Fred Posner
When you dial 43 you get a prompt or 41? Also, do you see anything in the freeswitch logs or have a sip capture/ Fred Posner | The Palner Group direct: 503-914-0999 | fax: 954-472-2896 On 11/08/2013 06:04 PM, Joli Martinez wrote: I am new to Kamailio and am having an issue with the dialplan

Re: [SR-Users] kamailio dialplan

2013-11-08 Thread Joli Martinez
43 get 407 proxy authentication required. This never hits the FS side. 41 does and I get the VM prompt. thanks, On Nov 8, 2013, at 6:13 PM, Fred Posner f...@palner.com wrote: When you dial 43 you get a prompt or 41? Also, do you see anything in the freeswitch logs or have a sip capture/