[asterisk-users] ISDN- E1 PRI module in network side signaling

2013-03-31 Thread Dimitar Dimitrov
Hello everyone. I am looking for a E1 PRI card which supports network side signaling not CPE. The main idea is to connect an plain old E1 compliant PBX which doesn't have an VoIP module to the newly created VoIP infrastructure. Could we use a Digium TE122P or something other to resolve this

Re: [asterisk-users] ISDN- E1 PRI module in network side signaling

2013-03-31 Thread Mitul Limbani
Use pri_net as signalling mode inside chan_dadhi.conf in /etc/asterisk/ folder. You can set this up using any pri card thats supported on Asterisk. Mitul On Mar 31, 2013 12:25 PM, Dimitar Dimitrov ddimit...@consult.bg wrote: Hello everyone. I am looking for a E1 PRI card which supports

Re: [asterisk-users] ISDN- E1 PRI module in network side signaling

2013-03-31 Thread Tony Mountifield
In article caaogpgr4y2vyndtu3nsnlct_6t1vdojsgouh634ec+zfogp...@mail.gmail.com, Mitul Limbani mi...@enterux.in wrote: On Mar 31, 2013 12:25 PM, Dimitar Dimitrov ddimit...@consult.bg wrote: Hello everyone. I am looking for a E1 PRI card which supports network side signaling not CPE. The

Re: [asterisk-users] ISDN- E1 PRI module in network side signaling

2013-03-31 Thread Dimitar Dimitrov
Thank you guys for the fast response. I will try that. Thanks. Dimitar On 03/31/2013 11:15 AM, Tony Mountifield wrote: In article caaogpgr4y2vyndtu3nsnlct_6t1vdojsgouh634ec+zfogp...@mail.gmail.com, Mitul Limbani mi...@enterux.in wrote: On Mar 31, 2013 12:25 PM, Dimitar Dimitrov

[asterisk-users] Feature request: Need to INVITE to peer with other domain without peer domain addition

2013-03-31 Thread Dmitriy Serov
Hi, asterisk admin and users. I need to SIP INVITE uri with domain via peer. And uri domain differ then peer domain. dialplan: exten = s,n,Dial(SIP/peer1/num...@domain2.com,60,r) [peer1] type=friend host=domain1.com fromdomain=domain1.com As a result in SIP packet uri:

[asterisk-users] SRTP woes

2013-03-31 Thread John Cahill
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I'm running Asterisk 11.3.0 on wheezy. I'm trying to do TLS +SRTP with blink SIP clients as shown here https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial. TLS is fine and I can call between clients. SRTP is a different matter,

Re: [asterisk-users] Feature request: Need to INVITE to peer with other domain without peer domain addition

2013-03-31 Thread Barry Flanagan
On 31 March 2013 18:11, Dmitriy Serov serov@gmail.com wrote: Hi, asterisk admin and users. I need to SIP INVITE uri with domain via peer. And uri domain differ then peer domain. dialplan: exten = s,n,Dial(SIP/peer1/number@**domain2.com num...@domain2.com ,60,r) [peer1] type=friend

Re: [asterisk-users] asterisk-users Digest, Vol 104, Issue 53

2013-03-31 Thread Kanuvar
in advance. Dimitar -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20130331/ef4a5743/attachment.html -- Message: 2 Date: Sun, 31 Mar 2013 12:50:09 +0530 From: Mitul

Re: [asterisk-users] [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled

2013-03-31 Thread Daniel Pocock
On 17/12/12 13:34, Joshua Colp wrote: Barco You wrote: Dear All, Hola, I use sipml5 to register two users from browser and the two clients are successfully connected. But when I made a call from one of the users, the other user doen'st have call notification and for a while the

Re: [asterisk-users] [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled

2013-03-31 Thread Joshua Colp
Daniel Pocock wrote: I'm trying to call from DruCall to Asterisk and I get this error: WARNING[11021]: chan_sip.c:8687 process_sdp: Error in codec string 'F 103 104 111 0 8 107 106 105 13 126' == Problem setting up ssl connection: error::lib(0):func(0):reason(0) I'm guessing my