[asterisk-users] ISDN- E1 PRI module in network side signaling
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 situation? Thanks in advance. Dimitar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN- E1 PRI module in network side signaling
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 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 situation? Thanks in advance. Dimitar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN- E1 PRI module in network side signaling
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 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 situation? 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. And you may need to make an E1 crossover cable. These are different from Ethernet crossover cables. You need to cross pair 1-2 with pair 4-5. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN- E1 PRI module in network side signaling
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 ddimit...@consult.bg wrote: 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 situation? 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. And you may need to make an E1 crossover cable. These are different from Ethernet crossover cables. You need to cross pair 1-2 with pair 4-5. Cheers Tony -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Feature request: Need to INVITE to peer with other domain without peer domain addition
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: num...@domain2.com@domain1.com I need: num...@domain2.com I can't use SIP uri dial, i need authorization (peer1) I think asterisk can't do that. Is where work around? Dmitriy Serov. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SRTP woes
-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,
my SIP clients return: SIP 488 Not acceptable Here
I'm really stumped on this one, any ideas?
srtp module is loaded:
*CLI module show like res_srtp.so
Module Description
Use Count
res_srtp.soSecure RTP (SRTP)
0
1 modules loaded
extensions.conf extract
exten = 1002,1,Set(_SIPSRTP=${SIPPEER(1002,srtpcapable)})
exten = 1002,n,Set(CHANNEL(secure_bridge_signaling)=1)
exten = 1002,n,Dial(SIP/exten1002,20)
exten = 1002,n,Hangup()
sip.conf extract:
[exten1002]
type=friend
host=dynamic
secret=averygoodone
context=users
nat=force_rport,comedia
encryption=yes
transport=tls
Thanks,
Regards,
John
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Re: [asterisk-users] Feature request: Need to INVITE to peer with other domain without peer domain addition
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 host=domain1.com fromdomain=domain1.com As a result in SIP packet uri: num...@domain2.com@domain1.com I need: num...@domain2.com I can't use SIP uri dial, i need authorization (peer1) I think asterisk can't do that. Is where work around? Would it work if you created a sip peer [domain2.com] and set outboundproxy= domain1.com then sent the call to SIP/num...@domain2.com ? -Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 104, Issue 53
Roberto estoy en uruguay en estoos momentos. Recien lllego el miercoles El mar 31, 2013 1:59 p.m., asterisk-users-requ...@lists.digium.com escribió: Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to asterisk-users-requ...@lists.digium.com You can reach the person managing the list at asterisk-users-ow...@lists.digium.com When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. ISDN- E1 PRI module in network side signaling (Dimitar Dimitrov) 2. Re: ISDN- E1 PRI module in network side signaling (Mitul Limbani) 3. Re: ISDN- E1 PRI module in network side signaling (Tony Mountifield) 4. Re: ISDN- E1 PRI module in network side signaling (Dimitar Dimitrov) -- Message: 1 Date: Sun, 31 Mar 2013 09:54:17 +0300 From: Dimitar Dimitrov ddimit...@consult.bg Subject: [asterisk-users] ISDN- E1 PRI module in network side signaling To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 5157dd99.4090...@consult.bg Content-Type: text/plain; charset=iso-8859-1; Format=flowed 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 situation? Thanks 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 Limbani mi...@enterux.in Subject: Re: [asterisk-users] ISDN- E1 PRI module in network side signaling To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: caaogpgr4y2vyndtu3nsnlct_6t1vdojsgouh634ec+zfogp...@mail.gmail.com Content-Type: text/plain; charset=utf-8 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 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 situation? Thanks in advance. Dimitar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20130331/a54a9e7c/attachment-0001.htm -- Message: 3 Date: Sun, 31 Mar 2013 08:15:19 + (UTC) From: t...@softins.co.uk (Tony Mountifield) Subject: Re: [asterisk-users] ISDN- E1 PRI module in network side signaling To: asterisk-users@lists.digium.com Message-ID: kj8ran$kje$1...@softins.clara.co.uk 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 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 situation? 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. And you may need to make an E1 crossover cable. These are different from Ethernet crossover cables. You need to cross pair 1-2 with pair 4-5. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- Message: 4 Date: Sun, 31 Mar 2013 13:55:39 +0300 From: Dimitar
Re: [asterisk-users] [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled
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 calling process ended. I check the /var/log/asterisk/messages got the following log: [Dec 17 14:54:11] WARNING[11471][C-] chan_sip.c: Received SAVPF profle in audio offer but AVPF is not enabled: audio 52760 RTP/SAVPF 103 104 0 8 107 106 105 13 126 [Dec 17 14:54:11] WARNING[11471][C-] chan_sip.c: Received SAVPF profle in video offer but AVPF is not enabled: video 52760 RTP/SAVPF 100 101 102 [Dec 17 14:54:11] WARNING[11471][C-] chan_sip.c: Insufficient information in SDP (c=)... As the warning states - you haven't enabled AVPF support. This is generally done on a per-peer basis using avpf=yes in the configuration. I would suggest you follow https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support since there may be other things you have missed. 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 Asterisk is too old (it is 1.8 from Debian). Can you confirm which version is needed to parse a media descriptor with SAVPF? Do I need to upgrade all the way to v11 with WebRTC support, or was avpf support added in some intermediate version? Also, I'm using a SIP proxy and it takes care of handling all the WebRTC connections and proxying the requests into a normal TCP/TLS connection to Asterisk. I was hoping to avoid opening up WebRTC access directly on Asterisk. One effect this has is that I can't control the `avpf=yes' setting on a per-peer basis, as the proxy is carrying requests from various types of peer, some public, some private. Is there any outright reason Asterisk can't support (S)AVPF on demand? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled
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 Asterisk is too old (it is 1.8 from Debian). Can you confirm which version is needed to parse a media descriptor with SAVPF? Do I need to upgrade all the way to v11 with WebRTC support, or was avpf support added in some intermediate version? Asterisk 1.8 does not have any knowledge of AVPF, and since it's a new feature it was only added to Asterisk 11. You could try to backport the changes but chan_sip has changed quite a bit, so it could be rough. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
