Re: [OpenSIPS-Users] mailing list reply-to : list or sender

I read many list many years and OpenSER-based software lists (OpenSER, OpenSIPS) are only lists where I send reply to poster when I press reply button. If I send my questions into list than I expect to read answers at list but not as my personal email. IMHO On 05.08.2009 17:58, Brett

[OpenSIPS-Users] ENUM Issue while using with opensips + asterisk ?

Hello, I am using OpenSIPs-1.5.1 and asterisk-1.4 for my call flow. I want to use ENUM with openSIPs. For that I gone through the ENUM module of OpenSIPs. I have done required modification in my opensips.cfg file. Do I need to register my domain or sip number for ENUM ?? Do I need to do any

Re: [OpenSIPS-Users] register performance with sipp

On Mon, Jun 22, 2009 at 3:17 AM, Iñaki Baz Castillo i...@aliax.net wrote: El Domingo, 21 de Junio de 2009, Uwe Kastens escribió: Hello, Anybody experience with measuring REGISTER performance with sipp? I made some tests and I was wondering how many requests should be possible with

Re: [OpenSIPS-Users] mailing list reply-to : list or sender

Hey, The 'reply-to-all' button in Thunderbird will cause it to be sent to the list and the person who wrote the message. David Iñaki Baz Castillo wrote: 2009/8/11 Victor Gamov v...@lipetsk.ru: I read many list many years and OpenSER-based software lists (OpenSER, OpenSIPS) are only

Re: [OpenSIPS-Users] [Kamailio-Users] dialog info + Grandstreams = freezing

Hey, I am not good at diagrams, but here goes : [ Asterisk] [ Asterisk ] [ MySQL ] / \/ \ [ Kamailio ] /\ / \/ \/ \ [ Extension ] [ Extension ] [

[OpenSIPS-Users] OpenSIPS 1.5 SVN branch is NOT STABLE

Hi, Again, the OpenSIPS 1.5 branch I have built has stopped responding to any and all SIP requests. In fact, it has even stopped recording anything in the log. This has happened after maybe 2-3 days of very light use. Only 8 phones are using this OpenSIPS as a UA. Unfortunately the BLF fixes that

[OpenSIPS-Users] no ringback

Hi, There: I am using opensips/kamailio in front of asterisk pool. my user register on the opensips, and pstn call are routed out via asterisk. what I find out is that when the caller calls callee, some of the UA doesn't generate ring back. for example, if I use xlite, the ring back works

[OpenSIPS-Users] SerMyAdmin for provisioning users

Hi, I have some questions regarding SerMyAdmin. 1) Is this the only web interface tool to provision users? 2) Is a user able to submit a registration and reset his password via this tool? 3) Is there a document how to build it? 4) Does it only compatible with

[OpenSIPS-Users] redirect to IVRs

hi list, I'm using opensips as a redirect server based on the TO or R-URI header. If the user is in the IVR group the proxy will respond with a redirect. (is_user_in(To, ivr)) { xlog(The call will be redirect to calling card server); rewritehostport(x.x.x.x:5060);