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Hello, Got a question regarding custom announcements in Asterisk. My goal is to allow my users record their own queue announcements and choose which announcements they want to use in each queue. I have several Asterisk servers and a Kamailio server which dispatches call traffic between the Asterisks. Question is, is it possible to have something like a NSF disk shared between several asterisk

Hello, I'm stuck with getting cdr records stored in MySql database. I have a working realtime environment and have verified that the db connection works fine when used via res_config_mysql.conf. I'd appreciate Your help on how to get the odbc connector working as I think there's something wrong with its configuration. The problem presented itself as an error when making a call that

On 6 February 2015 at 07:54, Olli Heiskanen <ohjelmistoarkkitehti at gmail.com> wrote: > > Hello, > > Got a question regarding custom announcements in Asterisk. > > My goal is to allow my users record their own queue announcements and > choose which announcements they want to use in each queue. I have several > Asterisk servers and a K...

...work with 'MySQL-asterisk' as the DSN instead of simply 'MySQL' ? I have machines that use /etc/odbc.ini and machines that use /usr/local/etc/odbc.ini depending on if I used a package to instal ODBC or if I compiled ODBC myself. On Tue, Feb 3, 2015 at 1:35 AM, Olli Heiskanen < ohjelmistoarkkitehti at gmail.com> wrote: > > Hello, > > I'm stuck with getting cdr records stored in MySql database. I have a > working realtime environment and have verified that the db connection works > fine when used via res_config_mysql.conf. I'd appreciate Your help on how > to g...

Hello, I have a problem with a call between 2 webrtc clients. Asterisk removes the ice-related lines from the sdp when it sends the INVITE out, and the called webrtc client rejects the INVITE due to the missing ice lines. Both webrtc clients are defined exactly the same way, same values in all fields except the number of the peer. There's probably something I've changed that causes this

Hello all, I've been testing a Kamailio Asterisk Realtime integration, and found a strange situation. My problem is that when using the integration, everything seems ok but Asterisk does not see the clients as registered. Kamailio and the clients report registered clients. Also calls fail. In Asterisk cli sip show peers shows nothing but for example realtime load sipusers name 660 shows the

Hello, I'm trying to get calls working between websocket clients and sip clients. For clients I have sip.js based clients on chrome, Zoipers and a Grandstream phone. Challenge here is I'd like to have Kamailio and rtpengine to handle the bridging between different rtp profiles but Asterisk changes them in the sdp bodies along the way. I'm using Asterisk 11.11.0. Is there a way to

Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as

Hello all, I have an Asterisk installation with Kamailio using realtime integration. I have gotten my clients to register, but there is something odd about the sip message flow with some of my clients. My clients are Zoiper and Asterisk is 11.10.2. When I set 'Subscribe to MWI' value to 'both', after a normal, successful registration Asterisk begins to send REGISTER messages to

Hello, I was testing with sdp and something came up worth asking: While calling from a webrtc client to another (chrome, sip.js) Asterisk receives the following sdp and rejects it with 488 Not Acceptable. Why does this happen, what's wrong with the sdp? The second sdp body below is accepted instead. Both have rtp profile RTP/SAVPF, difference is that the second one was produced by rtpengine,

Hello all, Just wondering on a behavior I noticed while testing with realtime sip peers with names like 111.222 at mydomain.com. Using Kamailio as outbound proxy, it sends Asterisk a sip message where To header value is < sip:111.222 at mydomain.com> and From header has value "username" < sip:111.333 at mydomain.com;transport=UDP>;tag=fc609171. When Asterisk sends out the

Hello, I noticed a strange thing while testing my Asterisk-Kamailio Realtime setup. In an INVITE the From and To headers contain the same number when calling through a Realtime integration setup. This happens when the INVITE leaves Asterisk. Can you guys tell me what might be causing this? I have 660 at testers.com as a websocket client and 700 at testers.com (caller) using a Zoiper client (db

Hello, I'd appreciate your comments on the following problem I'm having, please forgive me if this is something obvious, I've been scratching my head on this for a while: I have Asterisk+Kamailio setup where I'm currently testing inbound calls from outside. I have both webrtc and sip clients, where webrtc peers are defined according to sip.js instructions (