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snacktime

2005-Apr-09 00:06 UTC

[Asterisk-Users] g726 > gsm not working with sipura

I have a DID from livevoip coming into * as SIP/gsm.    My phone is a
sipura 2000 and the sip.conf for the sipura only allows g726.  When I
dial the sipura on an incoming call to connect the channels the sipura
returns the error "Media Type Not Available".  If I set sip.conf to
allow ulaw for the sipura it works fine.

Following is the sip debug.  What am I missing here?  

Sip read: 
INVITE sip:8002703805@69.25.136.31 SIP/2.0
Via: SIP/2.0/UDP 217.160.244.186:5060;branch=z9hG4bK36dd7751;rport
From: "4258201020"
<sip:4258201020@217.160.244.186>;tag=as0163cfd3
To: <sip:8002703805@69.25.136.31>
Contact: <sip:4258201020@217.160.244.186>
Call-ID: 766d3cda76f3f9a16aff8fac778b210f@217.160.244.186
CSeq: 102 INVITE
User-Agent: Live VoIP CJ.M-250
Date: Sat, 09 Apr 2005 07:01:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 345

v=0
o=root 18516 18516 IN IP4 217.160.244.186
s=session
c=IN IP4 217.160.244.186
t=0 0
m=audio 13448 RTP/AVP 0 4 3 8 18 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

12 headers, 15 lines
Using latest request as basis request
Sending to 217.160.244.186 : 5060 (non-NAT)
Found no matching peer or user for '217.160.244.186:5060'
Found RTP audio format 0
Found RTP audio format 4
Found RTP audio format 3
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 217.160.244.186:13448
Found description format PCMU
Found description format G723
Found description format GSM
Found description format PCMA
Found description format G729
Found description format iLBC
Found description format telephone-event
Capabilities: us - 0x2 (gsm), peer - audio=0x50f
(g723|gsm|ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x2
(gsm)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
0x1 (g723)
Looking for 8002703805 in default
list_route: hop: <sip:4258201020@217.160.244.186>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.160.244.186:5060;branch=z9hG4bK36dd7751
From: "4258201020"
<sip:4258201020@217.160.244.186>;tag=as0163cfd3
To: <sip:8002703805@69.25.136.31>
Call-ID: 766d3cda76f3f9a16aff8fac778b210f@217.160.244.186
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8002703805@69.25.136.31>
Content-Length: 0


 to 217.160.244.186:5060
    -- Executing Wait("SIP/217.160.244.186-08688000", "3")
in new stack
    -- Executing Answer("SIP/217.160.244.186-08688000", "")
in new stack
We're at 69.25.136.31 port 12976
Answering with preferred capability 0x2 (gsm)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.160.244.186:5060;branch=z9hG4bK36dd7751
From: "4258201020"
<sip:4258201020@217.160.244.186>;tag=as0163cfd3
To: <sip:8002703805@69.25.136.31>;tag=as0c5f43fb
Call-ID: 766d3cda76f3f9a16aff8fac778b210f@217.160.244.186
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8002703805@69.25.136.31>
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 27417 27417 IN IP4 69.25.136.31
s=session
c=IN IP4 69.25.136.31
t=0 0
m=audio 12976 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

 to 217.160.244.186:5060
    -- Executing Ringing("SIP/217.160.244.186-08688000", "")
in new stack
    -- Executing Dial("SIP/217.160.244.186-08688000",
"SIP/chris|10")
in new stack
We're at 69.25.136.31 port 14790
Answering with capability 0x10 (g726)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:chris@192.168.1.47:5060 SIP/2.0
Via: SIP/2.0/UDP 69.25.136.31:5060;branch=z9hG4bK77aa2d9a;rport
From: "4258201020" <sip:4258201020@69.25.136.31>;tag=as179b5cc8
To: <sip:chris@192.168.1.47:5060>
Contact: <sip:4258201020@69.25.136.31>
Call-ID: 7b13ca2b503e0c2d199187832157b692@69.25.136.31
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 09 Apr 2005 07:02:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 223

v=0
o=root 27417 27417 IN IP4 69.25.136.31
s=session
c=IN IP4 69.25.136.31
t=0 0
m=audio 14790 RTP/AVP 111 101
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 (NAT) to 4.32.7.39:50350
    -- Called chris
catalog1*CLI> 

Sip read: 
SIP/2.0 100 Trying
To: <sip:chris@192.168.1.47:5060>
From: "4258201020" <sip:4258201020@69.25.136.31>;tag=as179b5cc8
Call-ID: 7b13ca2b503e0c2d199187832157b692@69.25.136.31
CSeq: 102 INVITE
Via: SIP/2.0/UDP 69.25.136.31:5060;branch=z9hG4bK77aa2d9a
Server: Sipura/SPA2000-2.0.13(g)
Content-Length: 0


8 headers, 0 lines
catalog1*CLI> 

Sip read: 
ACK sip:8002703805@69.25.136.31 SIP/2.0
Via: SIP/2.0/UDP 217.160.244.186:5060;branch=z9hG4bK30b231db;rport
From: "4258201020"
<sip:4258201020@217.160.244.186>;tag=as0163cfd3
To: <sip:8002703805@69.25.136.31>;tag=as0c5f43fb
Contact: <sip:4258201020@217.160.244.186>
Call-ID: 766d3cda76f3f9a16aff8fac778b210f@217.160.244.186
CSeq: 102 ACK
User-Agent: Live VoIP CJ.M-250
Content-Length: 0


9 headers, 0 lines
catalog1*CLI> 

Sip read: 
SIP/2.0 488 Not Acceptable Here
To: <sip:chris@192.168.1.47:5060>;tag=e6a7e87f58deccb7i1
From: "4258201020" <sip:4258201020@69.25.136.31>;tag=as179b5cc8
Call-ID: 7b13ca2b503e0c2d199187832157b692@69.25.136.31
CSeq: 102 INVITE
Via: SIP/2.0/UDP 69.25.136.31:5060;branch=z9hG4bK77aa2d9a
Warning: 304 spa "Media type not available"
Server: Sipura/SPA2000-2.0.13(g)
Content-Length: 0


9 headers, 0 lines
    -- Got SIP response 488 "Not Acceptable Here" back from 4.32.7.39
Transmitting: 
ACK sip:chris@192.168.1.47:5060 SIP/2.0
Via: SIP/2.0/UDP 69.25.136.31:5060;branch=z9hG4bK77aa2d9a;rport
From: "4258201020" <sip:4258201020@69.25.136.31>;tag=as179b5cc8
To: <sip:chris@192.168.1.47:5060>;tag=e6a7e87f58deccb7i1
Contact: <sip:4258201020@69.25.136.31>
Call-ID: 7b13ca2b503e0c2d199187832157b692@69.25.136.31
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (NAT) to 4.32.7.39:50350
  == No one is available to answer at this time
Destroying call '7b13ca2b503e0c2d199187832157b692@69.25.136.31'

Brian Capouch

2005-Apr-09 00:14 UTC

head link

[Asterisk-Users] g726 > gsm not working with sipura

snacktime wrote:> I have a DID from livevoip coming into * as SIP/gsm.    My phone is a
> sipura 2000 and the sip.conf for the sipura only allows g726.  When I
> dial the sipura on an incoming call to connect the channels the sipura
> returns the error "Media Type Not Available".  If I set sip.conf
to
> allow ulaw for the sipura it works fine.
> 
> Following is the sip debug.  What am I missing here?  
The fix for this was posted on the mailing list on Thursday.

Look for the #define for the DEPRECATED g726 codec ID.  Enable that 
define and recompile.

B.

snacktime

2005-Apr-09 00:28 UTC

head link

[Asterisk-Users] g726 > gsm not working with sipura

On Apr 9, 2005 12:14 AM, Brian Capouch <brianc@palaver.net>
wrote:> snacktime wrote:
> > I have a DID from livevoip coming into * as SIP/gsm.    My phone is a
> > sipura 2000 and the sip.conf for the sipura only allows g726.  When I
> > dial the sipura on an incoming call to connect the channels the sipura
> > returns the error "Media Type Not Available".  If I set
sip.conf to
> > allow ulaw for the sipura it works fine.
> >
> > Following is the sip debug.  What am I missing here?
> 
> The fix for this was posted on the mailing list on Thursday.
> 
> Look for the #define for the DEPRECATED g726 codec ID.  Enable that
> define and recompile.
Yep that was it.  Thanks

asterisk users - Apr 2005 - g726 > gsm not working with sipura

[Asterisk-Users] g726 > gsm not working with sipura

[Asterisk-Users] g726 > gsm not working with sipura

[Asterisk-Users] g726 > gsm not working with sipura