I am using Asterisk 1.4 beta 3. I am calling from 3G to SIP. As I see from the trace Asterisk is not supporting the clear chanel codec (a=rtpmap:125 G.nX64/8000) used by the PGW.
Am I right or the problem is somewhere else? Please take a look of my config and the trace of the asterisk cli.
sip.conf
[general]
videosupport=yes
disallow=all ; First disallow all codecs
allow=alaw ; Allow codecs in order of preference
allow=h263
allow=h263p
allow=h261
[32515901]
type=friend
secret=phone1
host=dynamic
;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
mailbox=1000 ; Mailbox for message waiting indicator
context=sip
videosupport=yes
maxcallbitrate=128
callerid= "test" <32515901>
allow=alaw
allow=speex
allow=gsm
allow=h261
allow=h263
allow=h263p
<------------->
--- (14 headers 13 lines) ---
Using INVITE request as basis request - 3f7a5361-3057ba40-5f7fc171-25@my.domain.com (3f7a5361-3057ba40-5f7fc171-25@my.domain.com)
Found peer 'test'
Found RTP audio format 125
Peer audio RTP is at port 85.118.195.7:18010
Found description format G.nX64 for ID 125
Found description format G.nX64 for ID 125
Capabilities: us - 0x1c0008 (alaw|h261|h263|h263p), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Nov 15 23:15:53] NOTICE[22446]: chan_sip.c:4996 process_sdp: No compatible codecs, not accepting this offer!
<--- Reliably Transmitting (no NAT) to my.domain.com:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP my.domain.com:5060 ;branch=z9hG4bKterm-30-myphone-32515901-17145;received=my.domain.com
From: myphone <sip:myphone@my.domain.com ([email]sip:myphone@my.domain.com[/email]);user=phone>;tag=1763495500
To: 32515901 <sip:32515901@172.18.10.100 ([email]sip:32515901@172.18.10.100[/email]);user=phone>;tag=as11b984c5
Call-ID: 3f7a5361-3057ba40-5f7fc171-25@my.domain.com (3f7a5361-3057ba40-5f7fc171-25@my.domain.com)
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
I am using Asterisk 1.4 beta 3. I am calling from 3G to SIP. As I see from
the trace Asterisk is not supporting the clear chanel codec (a=rtpmap:125
G.nX64/8000) used by the PGW.
Am I right or the problem is somewhere else? Please take a look of my config
and the trace of the asterisk cli.
sip.conf
[general]
videosupport=yes
disallow=all ; First disallow all codecs
allow=alaw ; Allow codecs in order of preference
allow=h263
allow=h263p
allow=h261
[32515901]
type=friend
secret=phone1
host=dynamic
;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
mailbox=1000 ; Mailbox for message waiting indicator
context=sip
videosupport=yes
maxcallbitrate=128
callerid= "test" <32515901>
allow=alaw
allow=speex
allow=gsm
allow=h261
allow=h263
allow=h263p
<------------->
--- (14 headers 13 lines) ---
Using INVITE request as basis request -
3f7a5361-3057ba40-5f7fc171-25@my.domain.com
Found peer 'test'
Found RTP audio format 125
Peer audio RTP is at port 85.118.195.7:18010
Found description format G.nX64 for ID 125
Found description format G.nX64 for ID 125
Capabilities: us - 0x1c0008 (alaw|h261|h263|h263p), peer - audio=0x0
(nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0
(nothing), combined - 0x0 (nothing)
[Nov 15 23:15:53] NOTICE[22446]: chan_sip.c:4996 process_sdp: No compatible
codecs, not accepting this offer!
<--- Reliably Transmitting (no NAT) to my.domain.com:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP my.domain.com:5060
;branch=z9hG4bKterm-30-myphone-32515901-17145;received=my.domain.com
From: myphone <sip:myphone@my.domain.com;user=phone>;tag=1763495500
To: 32515901 <sip:32515901@172.18.10.100;user=phone>;tag=as11b984c5
Call-ID: 3f7a5361-3057ba40-5f7fc171-25@my.domain.com
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
Appreciate any help,
Niko
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Posted: Thu Nov 16, 2006 10:02 am Post subject: [Asterisk-video] 3G to SIP problem
Thanks Andrey,
the call is actually a video call and the video is comming from the softswitch as unrestricted digital G.nX64/8000). I guess asterisk in general is not supporting unrestricted digital.
I am using Asterisk 1.4 beta 3. I am calling from 3G to SIP. As I see from
the trace Asterisk is not supporting the clear chanel codec (a=rtpmap:125
G.nX64/8000) used by the PGW.
Am I right or the problem is somewhere else? Please take a look of my config
and the trace of the asterisk cli.
sip.conf
[general]
videosupport=yes
disallow=all ; First disallow all codecs
allow=alaw ; Allow codecs in order of preference
allow=h263
allow=h263p
allow=h261
[32515901]
type=friend
secret=phone1
host=dynamic
;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
mailbox=1000 ; Mailbox for message waiting indicator
context=sip
videosupport=yes
maxcallbitrate=128
callerid= "test" <32515901>
allow=alaw
allow=speex
allow=gsm
allow=h261
allow=h263
allow=h263p
<------------->
--- (14 headers 13 lines) ---
Using INVITE request as basis request -
3f7a5361-3057ba40-5f7fc171-25@my.domain.com (3f7a5361-3057ba40-5f7fc171-25@my.domain.com)
Found peer 'test'
Found RTP audio format 125
Peer audio RTP is at port 85.118.195.7:18010
Found description format G.nX64 for ID 125
Found description format G.nX64 for ID 125
Capabilities: us - 0x1c0008 (alaw|h261|h263|h263p), peer - audio=0x0
(nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0
(nothing), combined - 0x0 (nothing)
[Nov 15 23:15:53] NOTICE[22446]: chan_sip.c:4996 process_sdp: No compatible
codecs, not accepting this offer!
<--- Reliably Transmitting (no NAT) to my.domain.com:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP my.domain.com:5060
;branch=z9hG4bKterm-30-myphone-32515901-17145;received= my.domain.com
From: myphone < sip:myphone@my.domain.com ([email]sip:myphone@my.domain.com[/email]);user=phone>;tag=1763495500
To: 32515901 < sip:32515901@172.18.10.100 ([email]sip:32515901@172.18.10.100[/email]);user=phone>;tag=as11b984c5
Call-ID: 3f7a5361-3057ba40-5f7fc171-25@my.domain.com (3f7a5361-3057ba40-5f7fc171-25@my.domain.com)
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
Appreciate any help,
Niko
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This is a probably a proprietary (non-standard) SDP content generated
by your 3G phone. I remember I read from some RFC (dont remember which
one already), that prefix X- is normally used to indicate that the
following is proprietary and understood only by the device, which
generated it. The rest of the devices not aware of this extension,
will simply ignore it.
Best of luck,
Andrey.
On 11/16/06, Nikolay Milovanov <n.milovanov@gmail.com> wrote:
Quote:
Thanks Andrey,
the call is actually a video call and the video is comming from the
softswitch as unrestricted digital G.nX64/8000). I guess asterisk in general
is not supporting unrestricted digital.
Posted: Thu Nov 16, 2006 11:04 am Post subject: [Asterisk-video] 3G to SIP problem
Hi!
I guess these lines are irrelevant, because the m line offers only codec
125.
regards
klaus
Nikolay Milovanov wrote:
Quote:
Thanks Andrey,
the call is actually a video call and the video is comming from the
softswitch as unrestricted digital G.nX64/8000). I guess asterisk in
general
is not supporting unrestricted digital.
Posted: Thu Nov 16, 2006 11:16 am Post subject: [Asterisk-video] 3G to SIP problem
Appart of that, no video media is specified in the sdp.
Greetings
Sergio
-----Original Message-----
From: asterisk-video-bounces@lists.digium.com [mailto:asterisk-video-bounces@lists.digium.com] On Behalf Of Klaus Darilion
Sent: jueves, 16 de noviembre de 2006 12:02
To: Development discussion of video media support in Asterisk
Subject: Re: [Asterisk-video] 3G to SIP problem
Hi!
I guess these lines are irrelevant, because the m line offers only codec 125.
regards
klaus
Nikolay Milovanov wrote:
Quote:
Thanks Andrey,
the call is actually a video call and the video is comming from the
softswitch as unrestricted digital G.nX64/8000). I guess asterisk in
general is not supporting unrestricted digital.
_______________________________________________
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I guess these lines are irrelevant, because the m line offers only codec 125.
regards
klaus
Nikolay Milovanov wrote:
Quote:
Thanks Andrey,
the call is actually a video call and the video is comming from the
softswitch as unrestricted digital G.nX64/8000). I guess asterisk in
general is not supporting unrestricted digital.
>
On 11/16/06, Andrey Kuprianov < andrey.kouprianov@gmail.com (andrey.kouprianov@gmail.com)> wrote:
>
> Yup,
>
> It looks like Asterisk does not support your codec. That's what your
> SDP
> says:
>
> a=rtpmap:125 G.nX64/8000
> a=rtpmap:101 /8000
> a=rtpmap:100 /8000
>
> And that's what you have in config file:
>
> allow=alaw
> allow=speex
> allow=gsm
>
> Try switching codec to one of these listed in your sip.conf.
>
>
> On 11/16/06, Nikolay Milovanov < n_milovanov@mail.bg (n_milovanov@mail.bg)> wrote:
> > Hi Guys,
> >
> > My Scenario is
> >
> > 3Gphone -> (3G network provider)->(Softswitch Cisco
> > PGW)->SIP<-Asterisk<-SIP->SIP phone
> >
> > I am using Asterisk 1.4 beta 3. I am calling from 3G to SIP. As I
> > see
> from
> > the trace Asterisk is not supporting the clear chanel codec
> (a=rtpmap:125
> > G.nX64/8000) used by the PGW.
> >
> > Am I right or the problem is somewhere else? Please take a look of
> > my
> config
> > and the trace of the asterisk cli.
> >
> >
> > sip.conf
> >
> > [general]
> >
> > videosupport=yes
> > disallow=all ; First disallow all codecs
> > allow=alaw ; Allow codecs in order of preference
> > allow=h263
> > allow=h263p
> > allow=h261
> >
> > [32515901]
>> > type=friend
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You must not, directly or indirectly, use, disclose, distribute, print, or copy any
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Posted: Thu Nov 16, 2006 1:34 pm Post subject: [Asterisk-video] 3G to SIP problem
Uff!!
Then you'll need a H324M gateway in the middle to handle the negotiation and decode both streams, at least
until someone finally develope it for asterisk at last.. :)
Greetings
Sergio
From:asterisk-video-bounces@lists.digium.com [mailto:asterisk-video-bounces@lists.digium.com] On Behalf Of Nikolay Milovanov
Sent: jueves, 16 de noviembre de 2006 13:51
To: Development discussion of video media support in Asterisk
Subject: Re: [Asterisk-video] 3G to SIP problem
Hi Sergio,
I guess that's because of the clear channel. For me that means that both are encoded in it.
Thanks for the help.
Niko
On 11/16/06, Sergio García Murillo <Sergio.Garcia@ydilo.com (Sergio.Garcia@ydilo.com)> wrote:
Quote:
Appart of that, no video media is specified in the sdp.
I guess these lines are irrelevant, because the m line offers only codec 125.
regards
klaus
Nikolay Milovanov wrote:
Quote:
Thanks Andrey,
the call is actually a video call and the video is comming from the
softswitch as unrestricted digital G.nX64/8000). I guess asterisk in
general is not supporting unrestricted digital.
_______________________________________________
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asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-video
--------------------------------------------------------------------------------------
This message and any files transmitted with it are confidential and intended solely
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or privilege is waived or lost by any wrong transmission.
If you have received this message in error, please immediately destroy it and kindly
notify the sender by reply email.
You must not, directly or indirectly, use, disclose, distribute, print, or copy any
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Uff!!
Then you'll need a H324M gateway in the middle to handle the negotiation and decode both streams, at least
until someone finally develope it for asterisk at last.. :)
Greetings
Sergio
From:asterisk-video-bounces@lists.digium.com [mailto:asterisk-video-bounces@lists.digium.com] On Behalf Of Nikolay Milovanov
Sent: jueves, 16 de noviembre de 2006 13:51
To: Development discussion of video media support in Asterisk
Subject: Re: [Asterisk-video] 3G to SIP problem
Hi Sergio,
I guess that's because of the clear channel. For me that means that both are encoded in it.
Thanks for the help.
Niko
On 11/16/06, Sergio García Murillo <Sergio.Garcia@ydilo.com (Sergio.Garcia@ydilo.com)> wrote:
Quote:
Appart of that, no video media is specified in the sdp.
I guess these lines are irrelevant, because the m line offers only codec 125.
regards
klaus
Nikolay Milovanov wrote:
Quote:
Thanks Andrey,
the call is actually a video call and the video is comming from the
softswitch as unrestricted digital G.nX64/8000). I guess asterisk in
general is not supporting unrestricted digital.
_______________________________________________
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asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-video
--------------------------------------------------------------------------------------
This message and any files transmitted with it are confidential and intended solely
for the use of the individual or entity to whom they are addressed. No confidentiality
or privilege is waived or lost by any wrong transmission.
If you have received this message in error, please immediately destroy it and kindly
notify the sender by reply email.
You must not, directly or indirectly, use, disclose, distribute, print, or copy any
part of this message if you are not the intended recipient. Opinions, conclusions and
other information in this message that do not relate to the official business of
Ydilo Advanced Voice Solutions, S.A. shall be understood as neither given nor endorsed by it.
--------------------------------------------------------------------------------------
_______________________________________________
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Uff!!
Then you'll need a H324M gateway in the middle to handle the negotiation and decode both streams, at least
until someone finally develope it for asterisk at last.. :)
I guess these lines are irrelevant, because the m line offers only codec 125.
regards
klaus
Nikolay Milovanov wrote:
Quote:
Thanks Andrey,
the call is actually a video call and the video is comming from the
softswitch as unrestricted digital G.nX64/8000). I guess asterisk in
general is not supporting unrestricted digital.
On 11/16/06, Andrey Kuprianov < andrey.kouprianov@gmail.com (andrey.kouprianov@gmail.com)> wrote:
>
> Yup,
>
> It looks like Asterisk does not support your codec. That's what your
> SDP
> says:
>
> a=rtpmap:125 G.nX64/8000
> a=rtpmap:101 /8000
> a=rtpmap:100 /8000
>
> And that's what you have in config file:
>
> allow=alaw
> allow=speex
> allow=gsm
>
> Try switching codec to one of these listed in your sip.conf.
>
>
> On 11/16/06, Nikolay Milovanov < n_milovanov@mail.bg (n_milovanov@mail.bg)> wrote:
> > Hi Guys,
> >
> > My Scenario is
> >
> > 3Gphone -> (3G network provider)->(Softswitch Cisco
> > PGW)->SIP<-Asterisk<-SIP->SIP phone
> >
> > I am using Asterisk 1.4 beta 3. I am calling from 3G to SIP. As I
> > see
> from
> > the trace Asterisk is not supporting the clear chanel codec
> (a=rtpmap:125
> > G.nX64/8000) used by the PGW.
> >
> > Am I right or the problem is somewhere else? Please take a look of
> > my
> config
> > and the trace of the asterisk cli.
> >
> >
> > sip.conf
> >
> > [general]
> >
> > videosupport=yes
> > disallow=all ; First disallow all codecs
> > allow=alaw ; Allow codecs in order of preference
> > allow=h263
> > allow=h263p
> > allow=h261
> >
> > [32515901]
>> > type=friend
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for the use of the individual or entity to whom they are addressed. No confidentiality
or privilege is waived or lost by any wrong transmission.
If you have received this message in error, please immediately destroy it and kindly
notify the sender by reply email.
You must not, directly or indirectly, use, disclose, distribute, print, or copy any
part of this message if you are not the intended recipient. Opinions, conclusions and
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Ydilo Advanced Voice Solutions, S.A. shall be understood as neither given nor endorsed by it.
--------------------------------------------------------------------------------------
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Posted: Thu Nov 16, 2006 2:25 pm Post subject: [Asterisk-video] 3G to SIP problem
Could you tell us which are your plans about it? Schedule, licensing, pricing..
From:asterisk-video-bounces@lists.digium.com [mailto:asterisk-video-bounces@lists.digium.com] On Behalf Of Ramtin Amin
Sent: jueves, 16 de noviembre de 2006 14:56
To: Development discussion of video media support in Asterisk
Subject: RE: [Asterisk-video] 3G to SIP problem
Uff!!
Then you'll need a H324M gateway in the middle to handle the negotiation and decode both streams, at least
until someone finally develope it for asterisk at last.. :)
Greetings
Sergio
From:asterisk-video-bounces@lists.digium.com [mailto:asterisk-video-bounces@lists.digium.com] On Behalf Of Nikolay Milovanov
Sent: jueves, 16 de noviembre de 2006 13:51
To: Development discussion of video media support in Asterisk
Subject: Re: [Asterisk-video] 3G to SIP problem
Hi Sergio,
I guess that's because of the clear channel. For me that means that both are encoded in it.
Thanks for the help.
Niko
On 11/16/06, Sergio García Murillo <Sergio.Garcia@ydilo.com (Sergio.Garcia@ydilo.com)> wrote:
Quote:
Appart of that, no video media is specified in the sdp.
I guess these lines are irrelevant, because the m line offers only codec 125.
regards
klaus
Nikolay Milovanov wrote:
Quote:
Thanks Andrey,
the call is actually a video call and the video is comming from the
softswitch as unrestricted digital G.nX64/8000). I guess asterisk in
general is not supporting unrestricted digital.
_______________________________________________
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asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-video
--------------------------------------------------------------------------------------
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for the use of the individual or entity to whom they are addressed. No confidentiality
or privilege is waived or lost by any wrong transmission.
If you have received this message in error, please immediately destroy it and kindly
notify the sender by reply email.
You must not, directly or indirectly, use, disclose, distribute, print, or copy any
part of this message if you are not the intended recipient. Opinions, conclusions and
other information in this message that do not relate to the official business of
Ydilo Advanced Voice Solutions, S.A. shall be understood as neither given nor endorsed by it.
--------------------------------------------------------------------------------------
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> Hi!
>
> I guess these lines are irrelevant, because the m line offers only codec
> 125.
>
> regards
> klaus
>
> Nikolay Milovanov wrote:
> > Thanks Andrey,
> >
> > the call is actually a video call and the video is comming from the
> > softswitch as unrestricted digital G.nX64/8000). I guess asterisk in
>> > general is not supporting unrestricted digital.
Quote:
> >
> > Could somebody explain to me what is that means:
> >
> > a=X-cpar: a=rtpmap:100 X-NSE/8000
> > a=X-cpar: a=fmtp:100 192-194,200-202
> >
> >
> > BR,
> > Niko
> >
> > On 11/16/06, Andrey Kuprianov < andrey.kouprianov@gmail.com (andrey.kouprianov@gmail.com) > wrote:
> >>
> >> Yup,
> >>
> >> It looks like Asterisk does not support your codec. That's what your
> >> SDP
> >> says:
> >>
> >> a=rtpmap:125 G.nX64/8000
> >> a=rtpmap:101 /8000
> >> a=rtpmap:100 /8000
> >>
> >> And that's what you have in config file:
> >>
> >> allow=alaw
> >> allow=speex
> >> allow=gsm
> >>
> >> Try switching codec to one of these listed in your sip.conf.
>> >>
Quote:
> >>
> >> On 11/16/06, Nikolay Milovanov <n_milovanov@mail.bg (n_milovanov@mail.bg)> wrote:
> >> > Hi Guys,
> >> >
> >> > My Scenario is
> >> >
> >> > 3Gphone -> (3G network provider)->(Softswitch Cisco
> >> > PGW)->SIP<-Asterisk<-SIP->SIP phone
> >> >
> >> > I am using Asterisk 1.4 beta 3. I am calling from 3G to SIP. As I
> >> > see
> >> from
> >> > the trace Asterisk is not supporting the clear chanel codec
> >> (a=rtpmap:125
> >> > G.nX64/8000) used by the PGW.
> >> >
> >> > Am I right or the problem is somewhere else? Please take a look of
> >> > my
> >> config
> >> > and the trace of the asterisk cli.
> >> >
> >> >
> >> > sip.conf
> >> >
> >> > [general]
> >> >
> >> > videosupport=yes
> >> > disallow=all ; First disallow all codecs
> >> > allow=alaw ; Allow codecs in order of
> preference
> >> > allow=h263
> >> > allow=h263p
> >> > allow=h261
> >> >
> >> > [32515901]
> >> > type=friend
> >> > secret=phone1
> >> > host=dynamic
> >> > ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
> >> > mailbox=1000 ; Mailbox for message waiting indicator context=sip
> >> > videosupport=yes
> >> > maxcallbitrate=128
> >> > callerid= "test" <32515901>
> >> > allow=alaw
> >> > allow=speex
> >> > allow=gsm
> >> > allow=h261
> >> > allow=h263
> >> > allow=h263p
> >> >
> >> >
> >> > Debug
> >> >
> >> > <--- SIP read from my.domain.com:5060 ---> INVITE
> >> > sip:32515901@172.18.10.100 ([email]sip:32515901@172.18.10.100[/email]);user=phone SIP/2.0
> >> > Via: SIP/2.0/UDP my.domain.com:5060
> >> > ;branch=z9hG4bKterm-30-myphone-32515901-17145
> >> > From: myphone < sip:myphone@my.domain.com ([email]sip:myphone@my.domain.com[/email]);user=phone>;tag=1763495500
> >> > To: 32515901 < sip:32515901@172.18.10.100 ([email]sip:32515901@172.18.10.100[/email]) ;user=phone>
> >> > Call-ID: 3f7a5361-3057ba40-5f7fc171-25@my.domain.com (3f7a5361-3057ba40-5f7fc171-25@my.domain.com)
> >> > CSeq: 1 INVITE
> >> > Supported: timer
> >> > Session-Expires: 1800
> >> > Min-SE: 1800
> >> > Contact: <sip:myphone@my.domain.com:5060>
> >> > Allow:
> >> > INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE
> >> > Max-Forwards: 70
> >> > Content-Type: application/sdp
> >> > Content-Length: 317
> >> >
> >> > v=0
> >> > c=IN IP4 85.118.195.7
> >> > m=audio 18010 RTP/AVP 125
> >> > a=rtpmap:125 G.nX64/8000
> >> > a=X-pc-codec: 125 101 100
> >> > a=rtpmap:125 G.nX64/8000
> >> > a=rtpmap:101 /8000
> >> > a=rtpmap:100 /8000
> >> > a=X-sqn:0
> >> > a=X-cap: 1 audio RTP/AVP 100
> >> > a=X-cpar: a=rtpmap:100 X-NSE/8000
> >> > a=X-cpar: a=fmtp:100 192-194,200-202
> >> > a=X-cap: 2 image udptl t38
> >> >
> >> > <------------->
> >> > --- (14 headers 13 lines) ---
> >> > Using INVITE request as basis request -
> >> > 3f7a5361-3057ba40-5f7fc171-25@my.domain.com (3f7a5361-3057ba40-5f7fc171-25@my.domain.com)
> >> > Found peer 'test'
> >> > Found RTP audio format 125
> >> > Peer audio RTP is at port 85.118.195.7:18010 Found description
> >> > format G.nX64 for ID 125 Found description format G.nX64 for ID 125
> >> > Capabilities: us - 0x1c0008 (alaw|h261|h263|h263p), peer -
> >> > audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
> >> > Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer -
> >> > 0x0 (nothing), combined - 0x0 (nothing) [Nov 15 23:15:53]
> >> > NOTICE[22446]: chan_sip.c:4996 process_sdp: No
> >> compatible
> >> > codecs, not accepting this offer!
> >> >
> >> > <--- Reliably Transmitting (no NAT) to my.domain.com:5060 --->
> >> > SIP/2.0 488 Not acceptable here
> >> > Via: SIP/2.0/UDP my.domain.com:5060
> >> > ;branch=z9hG4bKterm-30-myphone-32515901-17145;received=
> >> > my.domain.com
> >> > From: myphone <
> >> > sip:myphone@my.domain.com ([email]sip:myphone@my.domain.com[/email]);user=phone>;tag=1763495500
> >> > To: 32515901 <
> >> > sip:32515901@172.18.10.100 ([email]sip:32515901@172.18.10.100[/email]);user=phone>;tag=as11b984c5
> >> > Call-ID: 3f7a5361-3057ba40-5f7fc171-25@my.domain.com (3f7a5361-3057ba40-5f7fc171-25@my.domain.com)
> >> > CSeq: 1 INVITE
>> >> > User-Agent: Asterisk PBX
Quote:
> >> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> >> > Supported: replaces
> >> > Content-Length: 0
> >> >
> >> > Appreciate any help,
> >> >
> >> > Niko
> >> >
> >> >
> >> > _______________________________________________
> >> > --Bandwidth and Colocation provided by Easynews.com --
> >> >
> >> > asterisk-video mailing list
> >> > To UNSUBSCRIBE or update options visit:
> >> >
> >> > http://lists.digium.com/mailman/listinfo/asterisk-video
> >> >
> >> >
> >> >
> >> _______________________________________________
> >> --Bandwidth and Colocation provided by Easynews.com --
> >>
> >> asterisk-video mailing list
> >> To UNSUBSCRIBE or update options visit:
> >> http://lists.digium.com/mailman/listinfo/asterisk-video
> >>
> >
> >
> > ----------------------------------------------------------------------
> > --
> >
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-video mailing list
> > To UNSUBSCRIBE or update options visit:
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>
>
> --
> Klaus Darilion
> nic.at
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
>> asterisk-video mailing list
Quote:
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-video
>
> --------------------------------------------------------------------------------------
>
> This message and any files transmitted with it are confidential and
> intended solely
> for the use of the individual or entity to whom they are addressed. No
> confidentiality
> or privilege is waived or lost by any wrong transmission.
> If you have received this message in error, please immediately destroy it
> and kindly
> notify the sender by reply email.
> You must not, directly or indirectly, use, disclose, distribute,
> print, or
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> part of this message if you are not the intended recipient. Opinions,
> conclusions and
> other information in this message that do not relate to the official
> business of
> Ydilo Advanced Voice Solutions, S.A. shall be understood as neither given
>> nor endorsed by it.
Quote:
>
> --------------------------------------------------------------------------------------
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
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