Hi,
I'm having problems with an asterisk server that's not offering Codecs
for ulaw and alaw as it should.
I've three servers in total: a1, a2 and "b"
A1 and A2 have pretty much the same config files, except IP address info
changes
Server B is configured to accept all inbound invites.
Calls from A1 to B, all work fine, and in a sip debug session I can see
A1 is offering codecs:
[May 6 16:43:19] WARNING[25404]: channel.c:720 ast_best_codec: Don't
know any of 0x4000 formats
Audio is at <IP HIDDEN> port 14958
Adding codec 0x2000 (amr) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
But when A2 makes the same call to B, it only offers amr:
[May 6 16:38:44] WARNING[20408]: channel.c:720 ast_best_codec: Don't
know any of 0x4000 formats
Audio is at <IP HIDDEN> port 15554
Adding codec 0x2000 (amr) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Its not building ulaw or alaw into its list. Server B doesn't support
AMR, so rejects the call.
(I've no idea about the 0x4000 error - but I see it on both the good and
bad servers, so I don't think its related).
The odd thing is that the sip.conf files for A1 and A2 are exactly the
same (save IP info).
The build of the Asterisk server is from a 1.4.15 private build to add
AMR, but, it's the same source built on both A1 and A2.
I'm trying to figure out why A2 isnt offering ulaw and alaw.
The codec seems ok, and is listed in the show codecs:
4 (1 << 2) (0x4) audio ulaw (G.711 u-law)
8 (1 << 3) (0x8) audio alaw (G.711 A-law)
8192 (1 << 13) (0x2000) audio amr (AMR)
But I cant see why its not transcoding across to ulaw/alaw.
Thanks,
Adrian
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Forgot to add: sip.conf for both A1 and A2 has the following global
codec definitions:
disallow=all
allow=clear
allow=amr
allow=ulaw
allow=alaw
The Asterisk build is a private build that adds the clear and AMR codec
setups.
The two servers are running Fedora, though A1s on 6 and A2s on 10. I
cant see why that would make a difference though.
________________________________
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Adrian
Marsh
Sent: 06 May 2009 17:53
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Understanding Codecs
Hi,
I'm having problems with an asterisk server that's not offering Codecs
for ulaw and alaw as it should.
I've three servers in total: a1, a2 and "b"
A1 and A2 have pretty much the same config files, except IP address info
changes
Server B is configured to accept all inbound invites.
Calls from A1 to B, all work fine, and in a sip debug session I can see
A1 is offering codecs:
[May 6 16:43:19] WARNING[25404]: channel.c:720 ast_best_codec: Don't
know any of 0x4000 formats
Audio is at <IP HIDDEN> port 14958
Adding codec 0x2000 (amr) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
But when A2 makes the same call to B, it only offers amr:
[May 6 16:38:44] WARNING[20408]: channel.c:720 ast_best_codec: Don't
know any of 0x4000 formats
Audio is at <IP HIDDEN> port 15554
Adding codec 0x2000 (amr) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Its not building ulaw or alaw into its list. Server B doesn't support
AMR, so rejects the call.
(I've no idea about the 0x4000 error - but I see it on both the good and
bad servers, so I don't think its related).
The odd thing is that the sip.conf files for A1 and A2 are exactly the
same (save IP info).
The build of the Asterisk server is from a 1.4.15 private build to add
AMR, but, it's the same source built on both A1 and A2.
I'm trying to figure out why A2 isnt offering ulaw and alaw.
The codec seems ok, and is listed in the show codecs:
4 (1 << 2) (0x4) audio ulaw (G.711 u-law)
8 (1 << 3) (0x8) audio alaw (G.711 A-law)
8192 (1 << 13) (0x2000) audio amr (AMR)
But I cant see why its not transcoding across to ulaw/alaw.
Thanks,
Adrian
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I got very excited when I read the title of this email - I was hoping someone had learnt to speak g729. Ah well. PaulH Adrian Marsh wrote:> > Hi, > > I?m having problems with an asterisk server that?s not offering Codecs > for ulaw and alaw as it should. > > I?ve three servers in total: a1, a2 and ?b? > > A1 and A2 have pretty much the same config files, except IP address > info changes > > Server B is configured to accept all inbound invites. > > Calls from A1 to B, all work fine, and in a sip debug session I can > see A1 is offering codecs: > > [May 6 16:43:19] WARNING[25404]: channel.c:720 ast_best_codec: Don't > know any of 0x4000 formats > > Audio is at <IP HIDDEN> port 14958 > > Adding codec 0x2000 (amr) to SDP > > *Adding codec 0x4 (ulaw) to SDP* > > *Adding codec 0x8 (alaw) to SDP* > > Adding non-codec 0x1 (telephone-event) to SDP > > But when A2 makes the same call to B, it only offers amr: > > [May 6 16:38:44] WARNING[20408]: channel.c:720 ast_best_codec: Don't > know any of 0x4000 formats > > Audio is at <IP HIDDEN> port 15554 > > Adding codec 0x2000 (amr) to SDP > > Adding non-codec 0x1 (telephone-event) to SDP > > Its not building ulaw or alaw into its list. Server B doesn?t support > AMR, so rejects the call. > > (I?ve no idea about the 0x4000 error ? but I see it on both the good > and bad servers, so I don?t think its related). > > The odd thing is that the sip.conf files for A1 and A2 are exactly the > same (save IP info). > > The build of the Asterisk server is from a 1.4.15 private build to add > AMR, but, it?s the same source built on both A1 and A2. > > I?m trying to figure out why A2 isnt offering ulaw and alaw. > > The codec seems ok, and is listed in the show codecs: > > 4 (1 << 2) (0x4) audio ulaw (G.711 u-law) > > 8 (1 << 3) (0x8) audio alaw (G.711 A-law) > > 8192 (1 << 13) (0x2000) audio amr (AMR) > > But I cant see why its not transcoding across to ulaw/alaw. > > Thanks, > > Adrian > > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users