Hello,
I've successfully installed Asterisk from last CVS and configured it
for using with DLINK-DG104S as mgcp CPE and PGW2200 as external sip
server.
All are work fine at G711 codecs, but then I disable all codecs except
g729 some calls failed (Not all calls. Some calls passed at g729
succesfully).
All my devices configred to use only g729 and I don't see other codecs
at mgcp or sip messages, but I see strange string at asterisks log:
NOTICE[196633]: File rtp.c, Line 418 (ast_rtp_read): Unknown RTP codec
123 received
NOTICE[196633]: File channel.c, Line 1478 (ast_set_read_format): Unable
to find a path from ALAW to G729A
NOTICE[196633]: File channel.c, Line 1448 (ast_set_write_format): Unable
to find a path from G729A to ALAW
WARNING[196633]: File chan_mgcp.c, Line 757 (mgcp_write): Asked to
transmit frame type 8, while native formats is 256 (read/write 256/256)
WARNING[196633]: File app_dial.c, Line 279 (wait_for_answer): Unable to
forward frame
I find similary posts at Asteris-Users mailing list, but don't find how
to resolve this trouble. Is this a bug or some misconfiguration at my
configs ?
sip.conf:
[general]
port = 5060
bindaddr = 0.0.0.0
context = local
disallow = all
allow = g729
mgcp.conf
[general]
port = 2427
bindaddr = 0.0.0.0
disallow = all
allow = g729
[DLINK]
context=local
host=Y.Y.Y.Y
threewaycalling=yes
transfer=yes
line => aaln/1
line => aaln/2
line => aaln/3
line => aaln/4
extension.conf
[local]
ignorepat => 9
exten => _9XXXXXXX,1,Dial,SIP/${EXTEN:1}@IP.IP.IP.IP
Some logs from Asterisk:
First mgcp CRCX after hang up:
Posting Request:
CRCX 323 aaln/1@DLINK MGCP 1.0
v=0
o=root 23577 23577 IN IP4 X.X.X.X
s=session
c=IN IP4 X.X.X.X
t=0 0
m=audio 14548 RTP/AVP 18
a=rtpmap:18 G729/8000
After that I enter phone number and sent call to sip server:
-- Executing Dial("MGCP/aaln/1@DLINK-0",
"SIP/3632034@IP.IP.IP.IP")
in new stack
INVITE sip:3632034@IP.IP.IP.IP SIP/2.0
<skip>
v=0
o=root 16078 16078 IN IP4 X.X.X.X
s=session
c=IN IP4 X.X.X.X
t=0 0
m=audio 18480 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Then I receive reply from SIP server:
Sip read:
SIP/2.0 100 Trying
<skip>
Sip read:
SIP/2.0 183 Session Progress
<skip>
v=0
o=- 0 0 IN IP4 Z.Z.Z.Z
s=-
c=IN IP4 Z.Z.Z.Z
t=0 0
m=audio 49640 RTP/AVP 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=X-sqn: 0
a=X-cap: 1 image udptl t38
a=sqn: 0
a=cdsc: 1 image udptl t38
After this message sometimes Asterisk make error message at log and drop
call:
-- SIP/IP.IP.IP.IP-b782 is making progress passing it to
MGCP/aaln/1@DLINK-1
srv-5*CLI> NOTICE[196633]: File rtp.c, Line 418 (ast_rtp_read): Unknown
RTP codec 123 received
NOTICE[196633]: File channel.c, Line 1478 (ast_set_read_format): Unable
to find a path from ALAW to G729A
NOTICE[196633]: File channel.c, Line 1448 (ast_set_write_format): Unable
to find a path from G729A to ALAW
WARNING[196633]: File chan_mgcp.c, Line 757 (mgcp_write): Asked to
transmit frame type 8, while native formats is 256 (read/write 256/256)
WARNING[196633]: File app_dial.c, Line 279 (wait_for_answer): Unable to
forward frame
Reliably Transmitting:
CANCEL sip:3632034@IP.IP.IP.IP:5060 SIP/2.0
Sip read:
SIP/2.0 487 Request Cancelled
....
--
Antonio
Have you bought G.729a from Digium which cost $10/channel? At 02:04 25/12/03 +0300, you wrote:>Hello, >I've successfully installed Asterisk from last CVS and configured it >for using with DLINK-DG104S as mgcp CPE and PGW2200 as external sip >server. >All are work fine at G711 codecs, but then I disable all codecs except >g729 some calls failed (Not all calls. Some calls passed at g729 >succesfully). > All my devices configred to use only g729 and I don't see other codecs >at mgcp or sip messages, but I see strange string at asterisks log: > >NOTICE[196633]: File rtp.c, Line 418 (ast_rtp_read): Unknown RTP codec >123 received >NOTICE[196633]: File channel.c, Line 1478 (ast_set_read_format): Unable >to find a path from ALAW to G729A >NOTICE[196633]: File channel.c, Line 1448 (ast_set_write_format): Unable >to find a path from G729A to ALAW >WARNING[196633]: File chan_mgcp.c, Line 757 (mgcp_write): Asked to >transmit frame type 8, while native formats is 256 (read/write >256/256) >WARNING[196633]: File app_dial.c, Line 279 (wait_for_answer): Unable to >forward frame > >I find similary posts at Asteris-Users mailing list, but don't find how >to resolve this trouble. Is this a bug or some misconfiguration at my >configs ? > >sip.conf: >[general] >port = 5060 >bindaddr = 0.0.0.0 >context = local >disallow = all >allow = g729 >mgcp.conf >[general] >port = 2427 >bindaddr = 0.0.0.0 >disallow = all >allow = g729 >[DLINK] >context=local >host=Y.Y.Y.Y >threewaycalling=yes >transfer=yes >line => aaln/1 >line => aaln/2 >line => aaln/3 >line => aaln/4 >extension.conf >[local] >ignorepat => 9 >exten => _9XXXXXXX,1,Dial,SIP/${EXTEN:1}@IP.IP.IP.IP > >Some logs from Asterisk: > >First mgcp CRCX after hang up: >Posting Request: >CRCX 323 aaln/1@DLINK MGCP 1.0 >v=0 >o=root 23577 23577 IN IP4 X.X.X.X >s=session >c=IN IP4 X.X.X.X >t=0 0 >m=audio 14548 RTP/AVP 18 >a=rtpmap:18 G729/8000 > >After that I enter phone number and sent call to sip server: > > -- Executing Dial("MGCP/aaln/1@DLINK-0", "SIP/3632034@IP.IP.IP.IP") >in new stack > >INVITE sip:3632034@IP.IP.IP.IP SIP/2.0 ><skip> >v=0 >o=root 16078 16078 IN IP4 X.X.X.X >s=session >c=IN IP4 X.X.X.X >t=0 0 >m=audio 18480 RTP/AVP 18 101 >a=rtpmap:18 G729/8000 >a=rtpmap:101 telephone-event/8000 >a=fmtp:101 0-16 > >Then I receive reply from SIP server: >Sip read: >SIP/2.0 100 Trying ><skip> > >Sip read: >SIP/2.0 183 Session Progress ><skip> >v=0 >o=- 0 0 IN IP4 Z.Z.Z.Z >s=- >c=IN IP4 Z.Z.Z.Z >t=0 0 >m=audio 49640 RTP/AVP 18 101 >a=rtpmap:101 telephone-event/8000 >a=fmtp:101 0-15 >a=X-sqn: 0 >a=X-cap: 1 image udptl t38 >a=sqn: 0 >a=cdsc: 1 image udptl t38 > >After this message sometimes Asterisk make error message at log and drop >call: > > -- SIP/IP.IP.IP.IP-b782 is making progress passing it to >MGCP/aaln/1@DLINK-1 >srv-5*CLI> NOTICE[196633]: File rtp.c, Line 418 (ast_rtp_read): Unknown >RTP codec 123 received >NOTICE[196633]: File channel.c, Line 1478 (ast_set_read_format): Unable >to find a path from ALAW to G729A >NOTICE[196633]: File channel.c, Line 1448 (ast_set_write_format): Unable >to find a path from G729A to ALAW >WARNING[196633]: File chan_mgcp.c, Line 757 (mgcp_write): Asked to >transmit frame type 8, while native formats is 256 (read/write >256/256) >WARNING[196633]: File app_dial.c, Line 279 (wait_for_answer): Unable to >forward frame > >Reliably Transmitting: >CANCEL sip:3632034@IP.IP.IP.IP:5060 SIP/2.0 > >Sip read: >SIP/2.0 487 Request Cancelled >.... > >-- >Antonio >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-usersPeter Brown CEO IP Telephonics
No, I did't bought any license from Digium. But as I say at my previous post, only _some part_ of my g729 calls are failed ! I think if I need license for G729 at Asterisk then all of my calls must to fails. Is it right ? -- Antonio> -----Original Message----- > From: Peter Brown [mailto:peterabrown@froggy.com.au] > Sent: Thursday, December 25, 2003 2:50 AM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] G729 troubles > > Have you bought G.729a from Digium which cost $10/channel? > At 02:04 25/12/03 +0300, you wrote: > >Hello, > >I've successfully installed Asterisk from last CVS and > configured it > >for using with DLINK-DG104S as mgcp CPE and PGW2200 as external sip > >server. > >All are work fine at G711 codecs, but then I disable all > codecs except > >g729 some calls failed (Not all calls. Some calls passed at g729 > >succesfully). > > All my devices configred to use only g729 and I don't see > other codecs > >at mgcp or sip messages, but I see strange string at asterisks log: > > > >NOTICE[196633]: File rtp.c, Line 418 (ast_rtp_read): Unknown > RTP codec > >123 received > >NOTICE[196633]: File channel.c, Line 1478 > (ast_set_read_format): Unable > >to find a path from ALAW to G729A > >NOTICE[196633]: File channel.c, Line 1448 (ast_set_write_format): > >Unable to find a path from G729A to ALAW > >WARNING[196633]: File chan_mgcp.c, Line 757 (mgcp_write): Asked to > >transmit frame type 8, while native formats is 256 (read/write > >256/256) > >WARNING[196633]: File app_dial.c, Line 279 > (wait_for_answer): Unable to > >forward frame > > > >I find similary posts at Asteris-Users mailing list, but > don't find how > >to resolve this trouble. Is this a bug or some > misconfiguration at my > >configs ? > > > >sip.conf: > >[general] > >port = 5060 > >bindaddr = 0.0.0.0 > >context = local > >disallow = all > >allow = g729 > >mgcp.conf > >[general] > >port = 2427 > >bindaddr = 0.0.0.0 > >disallow = all > >allow = g729 > >[DLINK] > >context=local > >host=Y.Y.Y.Y > >threewaycalling=yes > >transfer=yes > >line => aaln/1 > >line => aaln/2 > >line => aaln/3 > >line => aaln/4 > >extension.conf > >[local] > >ignorepat => 9 > >exten => _9XXXXXXX,1,Dial,SIP/${EXTEN:1}@IP.IP.IP.IP > > > >Some logs from Asterisk: > > > >First mgcp CRCX after hang up: > >Posting Request: > >CRCX 323 aaln/1@DLINK MGCP 1.0 > >v=0 > >o=root 23577 23577 IN IP4 X.X.X.X > >s=session > >c=IN IP4 X.X.X.X > >t=0 0 > >m=audio 14548 RTP/AVP 18 > >a=rtpmap:18 G729/8000 > > > >After that I enter phone number and sent call to sip server: > > > > -- Executing Dial("MGCP/aaln/1@DLINK-0", > >"SIP/3632034@IP.IP.IP.IP") in new stack > > > >INVITE sip:3632034@IP.IP.IP.IP SIP/2.0 > ><skip> > >v=0 > >o=root 16078 16078 IN IP4 X.X.X.X > >s=session > >c=IN IP4 X.X.X.X > >t=0 0 > >m=audio 18480 RTP/AVP 18 101 > >a=rtpmap:18 G729/8000 > >a=rtpmap:101 telephone-event/8000 > >a=fmtp:101 0-16 > > > >Then I receive reply from SIP server: > >Sip read: > >SIP/2.0 100 Trying > ><skip> > > > >Sip read: > >SIP/2.0 183 Session Progress > ><skip> > >v=0 > >o=- 0 0 IN IP4 Z.Z.Z.Z > >s=- > >c=IN IP4 Z.Z.Z.Z > >t=0 0 > >m=audio 49640 RTP/AVP 18 101 > >a=rtpmap:101 telephone-event/8000 > >a=fmtp:101 0-15 > >a=X-sqn: 0 > >a=X-cap: 1 image udptl t38 > >a=sqn: 0 > >a=cdsc: 1 image udptl t38 > > > >After this message sometimes Asterisk make error message at log and > >drop > >call: > > > > -- SIP/IP.IP.IP.IP-b782 is making progress passing it to > >MGCP/aaln/1@DLINK-1 > >srv-5*CLI> NOTICE[196633]: File rtp.c, Line 418 > (ast_rtp_read): Unknown > >RTP codec 123 received > >NOTICE[196633]: File channel.c, Line 1478 > (ast_set_read_format): Unable > >to find a path from ALAW to G729A > >NOTICE[196633]: File channel.c, Line 1448 (ast_set_write_format): > >Unable to find a path from G729A to ALAW > >WARNING[196633]: File chan_mgcp.c, Line 757 (mgcp_write): Asked to > >transmit frame type 8, while native formats is 256 (read/write > >256/256) > >WARNING[196633]: File app_dial.c, Line 279 > (wait_for_answer): Unable to > >forward frame > > > >Reliably Transmitting: > >CANCEL sip:3632034@IP.IP.IP.IP:5060 SIP/2.0 > > > >Sip read: > >SIP/2.0 487 Request Cancelled > >.... > > > >-- > >Antonio > >_______________________________________________ > >Asterisk-Users mailing list > >Asterisk-Users@lists.digium.com > >http://lists.digium.com/mailman/listinfo/asterisk-users > > Peter Brown > CEO > IP Telephonics > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
I'm going to take a stab at this, so someone correct me if I'm wrong! If you're calling one g729 device from another, the call is actually handed off without any decoding done, therefore the licensing isn't needed. If * has to connect the g729 call to another format, then the licensing comes in to play. And it could be that even though you've configured the disabling of the codec at one location, it still is enabled elsewhere? Close? Anyone? Sean -----Original Message----- From: Anton V Kirichenko [mailto:akirichenko@bsh.ru] Sent: Wednesday, December 24, 2003 7:04 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] G729 troubles No, I did't bought any license from Digium. But as I say at my previous post, only _some part_ of my g729 calls are failed ! I think if I need license for G729 at Asterisk then all of my calls must to fails. Is it right ? -- Antonio> -----Original Message----- > From: Peter Brown [mailto:peterabrown@froggy.com.au] > Sent: Thursday, December 25, 2003 2:50 AM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] G729 troubles > > Have you bought G.729a from Digium which cost $10/channel? > At 02:04 25/12/03 +0300, you wrote: > >Hello, > >I've successfully installed Asterisk from last CVS and > configured it > >for using with DLINK-DG104S as mgcp CPE and PGW2200 as external sip > >server. > >All are work fine at G711 codecs, but then I disable all > codecs except > >g729 some calls failed (Not all calls. Some calls passed at g729 > >succesfully). > > All my devices configred to use only g729 and I don't see > other codecs > >at mgcp or sip messages, but I see strange string at asterisks log: > > > >NOTICE[196633]: File rtp.c, Line 418 (ast_rtp_read): Unknown > RTP codec > >123 received > >NOTICE[196633]: File channel.c, Line 1478 > (ast_set_read_format): Unable > >to find a path from ALAW to G729A > >NOTICE[196633]: File channel.c, Line 1448 (ast_set_write_format): > >Unable to find a path from G729A to ALAW > >WARNING[196633]: File chan_mgcp.c, Line 757 (mgcp_write): Asked to > >transmit frame type 8, while native formats is 256 (read/write > >256/256) > >WARNING[196633]: File app_dial.c, Line 279 > (wait_for_answer): Unable to > >forward frame > > > >I find similary posts at Asteris-Users mailing list, but > don't find how > >to resolve this trouble. Is this a bug or some > misconfiguration at my > >configs ? > > > >sip.conf: > >[general] > >port = 5060 > >bindaddr = 0.0.0.0 > >context = local > >disallow = all > >allow = g729 > >mgcp.conf > >[general] > >port = 2427 > >bindaddr = 0.0.0.0 > >disallow = all > >allow = g729 > >[DLINK] > >context=local > >host=Y.Y.Y.Y > >threewaycalling=yes > >transfer=yes > >line => aaln/1 > >line => aaln/2 > >line => aaln/3 > >line => aaln/4 > >extension.conf > >[local] > >ignorepat => 9 > >exten => _9XXXXXXX,1,Dial,SIP/${EXTEN:1}@IP.IP.IP.IP > > > >Some logs from Asterisk: > > > >First mgcp CRCX after hang up: > >Posting Request: > >CRCX 323 aaln/1@DLINK MGCP 1.0 > >v=0 > >o=root 23577 23577 IN IP4 X.X.X.X > >s=session > >c=IN IP4 X.X.X.X > >t=0 0 > >m=audio 14548 RTP/AVP 18 > >a=rtpmap:18 G729/8000 > > > >After that I enter phone number and sent call to sip server: > > > > -- Executing Dial("MGCP/aaln/1@DLINK-0", > >"SIP/3632034@IP.IP.IP.IP") in new stack > > > >INVITE sip:3632034@IP.IP.IP.IP SIP/2.0 > ><skip> > >v=0 > >o=root 16078 16078 IN IP4 X.X.X.X > >s=session > >c=IN IP4 X.X.X.X > >t=0 0 > >m=audio 18480 RTP/AVP 18 101 > >a=rtpmap:18 G729/8000 > >a=rtpmap:101 telephone-event/8000 > >a=fmtp:101 0-16 > > > >Then I receive reply from SIP server: > >Sip read: > >SIP/2.0 100 Trying > ><skip> > > > >Sip read: > >SIP/2.0 183 Session Progress > ><skip> > >v=0 > >o=- 0 0 IN IP4 Z.Z.Z.Z > >s=- > >c=IN IP4 Z.Z.Z.Z > >t=0 0 > >m=audio 49640 RTP/AVP 18 101 > >a=rtpmap:101 telephone-event/8000 > >a=fmtp:101 0-15 > >a=X-sqn: 0 > >a=X-cap: 1 image udptl t38 > >a=sqn: 0 > >a=cdsc: 1 image udptl t38 > > > >After this message sometimes Asterisk make error message at log and > >drop > >call: > > > > -- SIP/IP.IP.IP.IP-b782 is making progress passing it to > >MGCP/aaln/1@DLINK-1 > >srv-5*CLI> NOTICE[196633]: File rtp.c, Line 418 > (ast_rtp_read): Unknown > >RTP codec 123 received > >NOTICE[196633]: File channel.c, Line 1478 > (ast_set_read_format): Unable > >to find a path from ALAW to G729A > >NOTICE[196633]: File channel.c, Line 1448 (ast_set_write_format): > >Unable to find a path from G729A to ALAW > >WARNING[196633]: File chan_mgcp.c, Line 757 (mgcp_write): Asked to > >transmit frame type 8, while native formats is 256 (read/write > >256/256) > >WARNING[196633]: File app_dial.c, Line 279 > (wait_for_answer): Unable to > >forward frame > > > >Reliably Transmitting: > >CANCEL sip:3632034@IP.IP.IP.IP:5060 SIP/2.0 > > > >Sip read: > >SIP/2.0 487 Request Cancelled > >.... > > > >-- > >Antonio > >_______________________________________________ > >Asterisk-Users mailing list > >Asterisk-Users@lists.digium.com > >http://lists.digium.com/mailman/listinfo/asterisk-users > > Peter Brown > CEO > IP Telephonics > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
In my case I see only g729 codec request from CPE (see mgcp CRCX) and only g729 from PGW2200 (see debug of sip messages) and I don't need and transcoding from one codec format to another codec format. Could you expain to me why asterisk starts transcoding process from g729 to alaw ? -- antonio> -----Original Message----- > From: Sean Cheesman [mailto:scheesman@gdsworks.com] > Sent: Thursday, December 25, 2003 3:34 AM > To: 'asterisk-users@lists.digium.com' > Subject: RE: [Asterisk-Users] G729 troubles > > I'm going to take a stab at this, so someone correct me if > I'm wrong! If you're calling one g729 device from another, > the call is actually handed off without any decoding done, > therefore the licensing isn't needed. If * has to connect > the g729 call to another format, then the licensing comes in > to play. And it could be that even though you've configured > the disabling of the codec at one location, it still is > enabled elsewhere? Close? Anyone? > > Sean > > -----Original Message----- > From: Anton V Kirichenko [mailto:akirichenko@bsh.ru] > Sent: Wednesday, December 24, 2003 7:04 PM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] G729 troubles > > > No, I did't bought any license from Digium. But as I say at > my previous post, only _some part_ of my g729 calls are failed ! > I think if I need license for G729 at Asterisk then all of my > calls must to fails. Is it right ? > > -- > Antonio > > > -----Original Message----- > > From: Peter Brown [mailto:peterabrown@froggy.com.au] > > Sent: Thursday, December 25, 2003 2:50 AM > > To: asterisk-users@lists.digium.com > > Subject: Re: [Asterisk-Users] G729 troubles > > > > Have you bought G.729a from Digium which cost $10/channel? > > At 02:04 25/12/03 +0300, you wrote: > > >Hello, > > >I've successfully installed Asterisk from last CVS and > > configured it > > >for using with DLINK-DG104S as mgcp CPE and PGW2200 as > external sip > > >server. > > >All are work fine at G711 codecs, but then I disable all > > codecs except > > >g729 some calls failed (Not all calls. Some calls passed at g729 > > >succesfully). > > > All my devices configred to use only g729 and I don't see > > other codecs > > >at mgcp or sip messages, but I see strange string at > asterisks log: > > > > > >NOTICE[196633]: File rtp.c, Line 418 (ast_rtp_read): Unknown > > RTP codec > > >123 received > > >NOTICE[196633]: File channel.c, Line 1478 > > (ast_set_read_format): Unable > > >to find a path from ALAW to G729A > > >NOTICE[196633]: File channel.c, Line 1448 (ast_set_write_format): > > >Unable to find a path from G729A to ALAW > > >WARNING[196633]: File chan_mgcp.c, Line 757 (mgcp_write): Asked to > > >transmit frame type 8, while native formats is 256 (read/write > > >256/256) > > >WARNING[196633]: File app_dial.c, Line 279 > > (wait_for_answer): Unable to > > >forward frame > > > > > >I find similary posts at Asteris-Users mailing list, but > > don't find how > > >to resolve this trouble. Is this a bug or some > > misconfiguration at my > > >configs ? > > > > > >sip.conf: > > >[general] > > >port = 5060 > > >bindaddr = 0.0.0.0 > > >context = local > > >disallow = all > > >allow = g729 > > >mgcp.conf > > >[general] > > >port = 2427 > > >bindaddr = 0.0.0.0 > > >disallow = all > > >allow = g729 > > >[DLINK] > > >context=local > > >host=Y.Y.Y.Y > > >threewaycalling=yes > > >transfer=yes > > >line => aaln/1 > > >line => aaln/2 > > >line => aaln/3 > > >line => aaln/4 > > >extension.conf > > >[local] > > >ignorepat => 9 > > >exten => _9XXXXXXX,1,Dial,SIP/${EXTEN:1}@IP.IP.IP.IP > > > > > >Some logs from Asterisk: > > > > > >First mgcp CRCX after hang up: > > >Posting Request: > > >CRCX 323 aaln/1@DLINK MGCP 1.0 > > >v=0 > > >o=root 23577 23577 IN IP4 X.X.X.X > > >s=session > > >c=IN IP4 X.X.X.X > > >t=0 0 > > >m=audio 14548 RTP/AVP 18 > > >a=rtpmap:18 G729/8000 > > > > > >After that I enter phone number and sent call to sip server: > > > > > > -- Executing Dial("MGCP/aaln/1@DLINK-0", > > >"SIP/3632034@IP.IP.IP.IP") in new stack > > > > > >INVITE sip:3632034@IP.IP.IP.IP SIP/2.0 <skip> v=0 o=root > 16078 16078 > > >IN IP4 X.X.X.X s=session c=IN IP4 X.X.X.X t=0 0 m=audio > 18480 RTP/AVP > > >18 101 > > >a=rtpmap:18 G729/8000 > > >a=rtpmap:101 telephone-event/8000 > > >a=fmtp:101 0-16 > > > > > >Then I receive reply from SIP server: > > >Sip read: > > >SIP/2.0 100 Trying > > ><skip> > > > > > >Sip read: > > >SIP/2.0 183 Session Progress > > ><skip> > > >v=0 > > >o=- 0 0 IN IP4 Z.Z.Z.Z > > >s=- > > >c=IN IP4 Z.Z.Z.Z > > >t=0 0 > > >m=audio 49640 RTP/AVP 18 101 > > >a=rtpmap:101 telephone-event/8000 > > >a=fmtp:101 0-15 > > >a=X-sqn: 0 > > >a=X-cap: 1 image udptl t38 > > >a=sqn: 0 > > >a=cdsc: 1 image udptl t38 > > > > > >After this message sometimes Asterisk make error message > at log and > > >drop > > >call: > > > > > > -- SIP/IP.IP.IP.IP-b782 is making progress passing it to > > >MGCP/aaln/1@DLINK-1 > > >srv-5*CLI> NOTICE[196633]: File rtp.c, Line 418 > > (ast_rtp_read): Unknown > > >RTP codec 123 received > > >NOTICE[196633]: File channel.c, Line 1478 > > (ast_set_read_format): Unable > > >to find a path from ALAW to G729A > > >NOTICE[196633]: File channel.c, Line 1448 (ast_set_write_format): > > >Unable to find a path from G729A to ALAW > > >WARNING[196633]: File chan_mgcp.c, Line 757 (mgcp_write): Asked to > > >transmit frame type 8, while native formats is 256 (read/write > > >256/256) > > >WARNING[196633]: File app_dial.c, Line 279 > > (wait_for_answer): Unable to > > >forward frame > > > > > >Reliably Transmitting: > > >CANCEL sip:3632034@IP.IP.IP.IP:5060 SIP/2.0 > > > > > >Sip read: > > >SIP/2.0 487 Request Cancelled > > >.... > > > > > >-- > > >Antonio > > >_______________________________________________ > > >Asterisk-Users mailing list > > >Asterisk-Users@lists.digium.com > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > Peter Brown > > CEO > > IP Telephonics > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
I am a new asterisk user. I have had a box up and running for a couple of months and been very happy with it. Last night I came up with a question that I have not been able to find an answer too. I purchased 5 licenses for the G729 codec from digium. My source is current from CVS as of late last night. Here are messages I'm getting from Asterisk. Can anybody help me? [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) Feb 28 08:47:49 WARNING[-1084458064]: asterisk.c:260 listener: Select retured er ror: Interrupted system call Feb 28 08:47:49 WARNING[-1084458064]: asterisk.c:260 listener: Select retured er ror: Interrupted system call == Detected 5 licensed G.729 transcoders Feb 28 08:47:49 WARNING[-1084456832]: translate.c:219 calc_cost: Translator 'g72 9tolinb' does not produce sample frames. == Registered translator 'g729tolinb' from format G729A to SLINR, cost 99999 == Registered translator 'lintog729b' from format SLINR to G729A, cost 26 Thanks so much, Darren Wiebe dkwiebe@hagenhomes.com
I just purchased a G729 codec yesterday and am having the same issue. Does anyone have a solution to this? Thanks for your input. Regards, Anthony
Anthony Law [anthonyl@accessv.com] wrote:> I just purchased a G729 codec yesterday and am having the same issue. Does > anyone have a solution to this? Thanks for your input. >Perhaps a little context will help. Same issue as whom? Solution to what? etc. -- _/ _/ _/_/_/_/ _/ _/ _/_/_/ _/ _/ _/_/_/ _/_/ _/ _/ _/ _/_/ _/ K e v i n W a l s h _/ _/ _/ _/ _/ _/ _/ _/_/ kevin@cursor.biz _/ _/ _/_/_/_/ _/ _/_/_/ _/ _/
Hi, What I mean is when I start asterisk with -vvvc. I got [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) == Detected 1 licensed G.729 transcoders Mar 23 13:46:12 WARNING[1024]: translate.c:219 calc_cost: Translator 'g729tolinb' does not produce sample frames. == Registered translator 'g729tolinb' from format G729A to SLINR, cost 99999 == Registered translator 'lintog729b' from format SLINR to G729A, cost 37 If I start if with daemon /etc/rc.d/init.d/asterisk start Asterisk crashed and cannot be started. Regards, Anthony