On Thu, Nov 5, 2009 at 8:23 AM, Kevin P. Fleming kpflem...@digium.comwrote:
We need to see how you are originating the calls; it's up to the
originator to specify the formats that will be allowed for that call. In
spool files, for example, there is a header that can be included to
specify
Tom Browning wrote:
vi call.txt
Channel: SIP/f...@bar.com mailto:f...@bar.com
CallerID: testcall
Context: default
Extension: demo
Codecs: siren14
cp call.txt /var/spool/asterisk/outgoing/
-- Attempting call on SIP/f...@bar.com mailto:f...@bar.com for
On Tue, Nov 10, 2009 at 1:20 PM, Kevin P. Fleming kpflem...@digium.com wrote:
Please run this test with the 'debug' level enabled for the 'console'
channel in logger.conf, and then ensure that you have 'core set verbose
10' and 'core set debug 10' before attempting the outbound call. This
Tom Browning wrote:
The only difference between the call attempt that actually sends the
INVITE and the call attempt that complains is 'ulaw' vs 'siren14' in
the sip.conf allow= and spol file Codecs: header.
Clearly those codec choices are not treated the same to build an
outbound INVITE.
On Tue, Nov 10, 2009 at 3:39 PM, Kevin P. Fleming kpflem...@digium.com wrote:
They are, but we won't be able to know what is happening unless you post
a detailed console log like I suggested in my previous reply.
-- Attempting call on SIP/f...@bar.com for d...@default:1 (Retry 1)
[Nov 10
Tom Browning wrote:
On Tue, Nov 10, 2009 at 3:39 PM, Kevin P. Fleming kpflem...@digium.com
wrote:
They are, but we won't be able to know what is happening unless you post
a detailed console log like I suggested in my previous reply.
-- Attempting call on SIP/f...@bar.com for
Continuing the siren14 usage thread:
sip.conf has:
disallow=all ; First disallow all codecs
allow=siren14;
Should I be able to originate an outbound call with siren14 as my only
codec?
When I try originate using either the spool file or a CLI originate
What are you reaching out to exactly? It would need to be a Siren14
capable. Also, do you have the Siren codec binary installed? It's not
part of the Asterisk distribution.
Also, you should know that all Siren14 calls are presently downsampled
to 16 KHz, so are effectively Siren7.Asterisk doesn't
What are you reaching out to exactly? It would need to be a Siren14
capable. Also, do you have the Siren codec binary installed? It's not part
of the Asterisk distribution.
Inbound calls to Asterisk work (from a platform that supports both Siren14
and G.711). Leaving ulaw out of the allow
I've got a fresh (1 day old) svn trunk release SVN-trunk-r225360 of Asterisk
and I'm trying to get it to accept a SIREN14 call from Polycom's softphone.
Having trouble with SDP negotiation, I want to only allow SIREN14 and
nothing else. I also want to record and playback files, any tips on what
-Commercial Discussion
Subject: [asterisk-users] SIREN14 call setup and record/playback
I've got a fresh (1 day old) svn trunk release SVN-trunk-r225360 of Asterisk
and I'm trying to get it to accept a SIREN14 call from Polycom's softphone.
Having trouble with SDP negotiation, I want to only allow
Tom Browning wrote:
I've got a fresh (1 day old) svn trunk release SVN-trunk-r225360 of
Asterisk and I'm trying to get it to accept a SIREN14 call from
Polycom's softphone. Having trouble with SDP negotiation, I want to
only allow SIREN14 and nothing else. I also want to record and
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