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Lubomir Christov

2003-Mar-12 11:00 UTC

[Asterisk-Users] SIP/G723/iconnect with todays CVS version isn't working

Hello all,

I'm using iconnect with LineJACK/PhoneJACK/PhoneCARD and G723.1 codec 
from about 1 mount without any problems. The quality is ok and 
everything is OK (only some little problems sometime ... when the format 
in phone.conf isn't slinear, but format=g723.1 I have only ONE way audio 
(the other side is hearing ONLY strange sounds ....)).
But today morning, when I updated new CVS version of * I found that 
SIP(G723/ulaw) and iconnect aren't working anymore .... ???????

When I try to connect trough iconnect I receive this error message:

     -- Got SIP response 488 "Not Acceptable Media" back from 
213.137.73.178

here is my config:

sip.conf
[general]
port = 5060
;bindaddr = 0.0.0.0
context = incoming
disallow=all
allow=g723.1
;allow=ulaw
tos=lowdelay
tos=184

[iconnect]
type=friend
username=12345678
password=1234
host=213.137.73.178
callerid=1234567890

I have attached my todays sip debug output.
I'm sure that the problem is in todays CVS version only because when I 
download yesterdays version (cvs -z9 co -D "Mar 11 2003" asterisk)
there
wasn't such a problem and everything was OK.
I hope that iconnect will be back soon :)))

Lubo



-------------- next part --------------
*CLI> 
Sip read: 
SIP/2.0 488 Not Acceptable Media  
Via:  SIP/2.0/UDP 213.16.62.32:5060;branch=0efccd4b
From: "asterisk" <sip:asterisk@213.16.62.32>;tag=545f9b5c
To: sip:14158645225@213.137.73.178;tag=D7559CE0-3D4
Date: Wed, 12 Mar 2003 17:11:22 GMT
Call-ID: 20b05d914ee5138f43c79def2c87d12a@213.16.62.32
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Content-Length: 0


9 headers, 0 lines
Interface is ppp0
IP Address is 213.16.62.32

*CLI> 
*CLI> 
    -- Executing Dial("Phone/phone0",
"Sip/35929817675@iconnect||C") in new stack
Interface is ppp0
IP Address is 213.16.62.32
We're at 213.16.62.32 port 7562
Answering with preferred capability 1
10 headers, 6 lines
XXX Need to handle Retransmitting XXX:
INVITE sip:35929817675@213.137.73.178 SIP/2.0
Via: SIP/2.0/UDP 213.16.62.32:5060;branch=3e790dd7
From: "asterisk" <sip:asterisk@213.16.62.32>;tag=5d3317c9
Contact: <sip:asterisk@213.16.62.32>
To: <sip:35929817675@213.137.73.178>
Call-ID: 0bb13c9319498c833440fc9e192cb139@213.16.62.32
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 106

v=0
o=root 4008 4008 IN IP4 213.16.62.32
s=session
c=IN IP4 213.16.62.32
t=0 0
m=audio 7562 RTP/AVP
 (no NAT) to 213.137.73.178:5060
    -- Called 35929817675@iconnect
Sip read: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.16.62.32:5060;branch=3e790dd7
Call-ID: 0bb13c9319498c833440fc9e192cb139@213.16.62.32
From: "asterisk" <sip:asterisk@213.16.62.32>;tag=5d3317c9
To: <sip:35929817675@213.137.73.178>
CSeq: 102 INVITE
Content-Length: 0


7 headers, 0 lines
Sip read: 
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 213.16.62.32:5060;branch=3e790dd7
Call-ID: 0bb13c9319498c833440fc9e192cb139@213.16.62.32
From: "asterisk" <sip:asterisk@213.16.62.32>;tag=5d3317c9
To: <sip:35929817675@213.137.73.178>;tag=534dad39-5e7fe66
CSeq: 102 INVITE
Proxy-Authenticate: DIGEST realm="deltathree.com",
nonce="3e6f6a54", algorithm=MD5
Content-Length: 0


8 headers, 0 lines
XXX Need to handle Retransmitting XXX:
ACK sip:35929817675@213.137.73.178 SIP/2.0
Via: SIP/2.0/UDP 213.16.62.32:5060;branch=3e790dd7
From: "asterisk" <sip:asterisk@213.16.62.32>;tag=5d3317c9
To: <sip:35929817675@213.137.73.178>;tag=534dad39-5e7fe66
Call-ID: 0bb13c9319498c833440fc9e192cb139@213.16.62.32
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 213.137.73.178:5060
We're at 213.16.62.32 port 7562
Answering with preferred capability 1
XXX Need to handle Retransmitting XXX:
INVITE sip:35929817675@213.137.73.178 SIP/2.0
Via: SIP/2.0/UDP 213.16.62.32:5060;branch=3e790dd7
From: "asterisk" <sip:asterisk@213.16.62.32>;tag=5d3317c9
Contact: <sip:asterisk@213.16.62.32>
To: <sip:35929817675@213.137.73.178>
Call-ID: 0bb13c9319498c833440fc9e192cb139@213.16.62.32
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="12345678",
realm="deltathree.com", algorithm="MD5",
uri="sip:35929817675@213.137.73.178", nonce="3e6f6a54",
response="8863cae6c56b333a2c09b76e2a3013b3"
Content-Type: application/sdp
Content-Length: 106

v=0
o=root 3530 3530 IN IP4 213.16.62.32
s=session
c=IN IP4 213.16.62.32
t=0 0
m=audio 7562 RTP/AVP
 (no NAT) to 213.137.73.178:5060
Sip read: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.16.62.32:5060;branch=3e790dd7
Call-ID: 0bb13c9319498c833440fc9e192cb139@213.16.62.32
From: "asterisk" <sip:asterisk@213.16.62.32>;tag=5d3317c9
To: <sip:35929817675@213.137.73.178>
CSeq: 103 INVITE
Content-Length: 0


7 headers, 0 lines
Sip read: 
SIP/2.0 488 Not Acceptable Media  
Via:  SIP/2.0/UDP 213.16.62.32:5060;branch=3e790dd7
From: "asterisk" <sip:asterisk@213.16.62.32>;tag=5d3317c9
To: sip:35929817675@213.137.73.178;tag=6761889C-1316
Date: Wed, 12 Mar 2003 17:11:48 GMT
Call-ID: 0bb13c9319498c833440fc9e192cb139@213.16.62.32
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Content-Length: 0


9 headers, 0 lines
    -- Got SIP response 488 "Not Acceptable Media" back from
213.137.73.178
XXX Need to handle Retransmitting XXX:
ACK sip:35929817675@213.137.73.178 SIP/2.0
Via: SIP/2.0/UDP 213.16.62.32:5060;branch=3e790dd7
From: "asterisk" <sip:asterisk@213.16.62.32>;tag=5d3317c9
To: <sip:35929817675@213.137.73.178>;tag=534dad39-5e7fe66
Call-ID: 0bb13c9319498c833440fc9e192cb139@213.16.62.32
CSeq: 103 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 213.137.73.178:5060
WARNING[245774]: File app_dial.c, Line 271 (wait_for_answer): Unable to forward
voice
Sip read: 
SIP/2.0 488 Not Acceptable Media  
Via:  SIP/2.0/UDP 213.16.62.32:5060;branch=3e790dd7
From: "asterisk" <sip:asterisk@213.16.62.32>;tag=5d3317c9
To: sip:35929817675@213.137.73.178;tag=6761889C-1316
Date: Wed, 12 Mar 2003 17:11:48 GMT
Call-ID: 0bb13c9319498c833440fc9e192cb139@213.16.62.32
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Content-Length: 0


9 headers, 0 lines
Interface is ppp0
IP Address is 213.16.62.32
Sip read: 
SIP/2.0 488 Not Acceptable Media  
Via:  SIP/2.0/UDP 213.16.62.32:5060;branch=3e790dd7
From: "asterisk" <sip:asterisk@213.16.62.32>;tag=5d3317c9
To: sip:35929817675@213.137.73.178;tag=6761889C-1316
Date: Wed, 12 Mar 2003 17:11:48 GMT
Call-ID: 0bb13c9319498c833440fc9e192cb139@213.16.62.32
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Content-Length: 0


9 headers, 0 lines
Interface is ppp0
IP Address is 213.16.62.32
Sip read: 
SIP/2.0 488 Not Acceptable Media  
Via:  SIP/2.0/UDP 213.16.62.32:5060;branch=3e790dd7
From: "asterisk" <sip:asterisk@213.16.62.32>;tag=5d3317c9
To: sip:35929817675@213.137.73.178;tag=6761889C-1316
Date: Wed, 12 Mar 2003 17:11:48 GMT
Call-ID: 0bb13c9319498c833440fc9e192cb139@213.16.62.32
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Content-Length: 0


9 headers, 0 lines
Interface is ppp0
IP Address is 213.16.62.32
Sip read: 
SIP/2.0 488 Not Acceptable Media  
Via:  SIP/2.0/UDP 213.16.62.32:5060;branch=0efccd4b
From: "asterisk" <sip:asterisk@213.16.62.32>;tag=545f9b5c
To: sip:14158645225@213.137.73.178;tag=D7559CE0-3D4
Date: Wed, 12 Mar 2003 17:11:22 GMT
Call-ID: 20b05d914ee5138f43c79def2c87d12a@213.16.62.32
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Content-Length: 0

Matteo Brancaleoni

2003-Mar-12 12:12 UTC

head link

R: [Asterisk-Users] SIP/G723/iconnect with todays CVS version isn't working

Same for me , when I call from one sip fxs gw phone to the snom one.
I can hear audio only on from the sip gw and not from the snom.
Thery're using only alaw/ulaw (only accepted in sip.conf).
Yesterday all was working.

asterisk users - Mar 2003 - SIP/G723/iconnect with todays CVS version isn't working

[Asterisk-Users] SIP/G723/iconnect with todays CVS version isn't working

R: [Asterisk-Users] SIP/G723/iconnect with todays CVS version isn't working