i have a sip provider and i am sending the my call on his server but getting the following error. can any body help me on urgent basis. it wud b gr8 help to me. after this error i also gave the sip debug response after the error.
– Executing [088888787678687@CS:1] Dial(“OSS/dsp”, “SIP/088888787678687@provider|100|Ttr”) in new stack
*CLI> – Called 088888787678687@provider
– Call on SIP/provider-081e4460 left from hold
<< Console Has Been Retrieved from Hold >>
– SIP/provider-081e4460 is making progress passing it to OSS/dsp
– Got SIP response 500 “Calling is failed (unspecified)” back from 83.98.222.4
– SIP/provider-081e4460 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel ‘OSS/dsp’ status is ‘CONGESTION’
[Mar 17 18:48:15] WARNING[7382]: chan_oss.c:682 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
SIP DEBUG RESPONSE*****
*CLI> dial 011442476702601@CS
[Mar 17 18:18:51] WARNING[24414]: chan_oss.c:682 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
*CLI> – Executing [011442476702601@CS:1] Dial(“OSS/dsp”, “SIP/011442476702601@provider|100|Ttr”) in new stack
*CLI> Audio is at 82.150.141.193 port 11472
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
*CLI> Reliably Transmitting (no NAT) to 83.98.222.4:5060:
INVITE sip:011442476702601@sip1.belcentrale.nl SIP/2.0
Via: SIP/2.0/UDP 82.150.141.193:5060;branch=z9hG4bK39d7eee3;rport
From: “asterisk” sip:asterisk@82.150.141.193;tag=as038a9fa1
To: sip:011442476702601@sip1.belcentrale.nl
Contact: sip:asterisk@82.150.141.193
Call-ID: 059aa03d32888a841e70150021a93f6b@82.150.141.193
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 17 Mar 2007 17:18:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 291
v=0
o=root 24414 24414 IN IP4 82.150.141.193
s=session
c=IN IP4 82.150.141.193
t=0 0
m=audio 11472 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
*CLI> – Called 011442476702601@provider
*CLI>
<— SIP read from 83.98.222.4:5060 —>
SIP/2.0 100 trying – your call is important to us
Via: SIP/2.0/UDP 82.150.141.193:5060;branch=z9hG4bK39d7eee3;rport=5060
From: “asterisk” sip:asterisk@82.150.141.193;tag=as038a9fa1
To: sip:011442476702601@sip1.belcentrale.nl
Call-ID: 059aa03d32888a841e70150021a93f6b@82.150.141.193
CSeq: 102 INVITE
Server: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0
<------------->
— (8 headers 0 lines) —
<— SIP read from 83.98.222.4:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 82.150.141.193:5060;branch=z9hG4bK39d7eee3;rport=5060
Record-Route: sip:83.98.222.4;ftag=as038a9fa1;lr
From: asterisk sip:asterisk@82.150.141.193;tag=as038a9fa1
To: sip:011442476702601@sip1.belcentrale.nl;tag=122eb31f2bb5d31c45b31cb551caa4b4
Call-ID: 059aa03d32888a841e70150021a93f6b@82.150.141.193
CSeq: 102 INVITE
Server: Sippy
Content-Length: 264
Content-Type: application/sdp
v=0
o=root 15047281 52828 IN IP4 83.98.222.4
s=session
t=0 0
m=audio 43540 RTP/AVP 8 3 0 101
c=IN IP4 83.98.222.4
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
<------------->
— (10 headers 12 lines) —
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 83.98.222.4:43540
Found description format PCMA for ID 8
Found description format GSM for ID 3
Found description format PCMU for ID 0
Found description format telephone-event for ID 101
Capabilities: us - 0x8010e (gsm|ulaw|alaw|g729|h263), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 83.98.222.4:43540
[Mar 17 18:18:51] WARNING[24435]: channel.c:2816 set_format: Unable to find a codec translation path from alaw to g729
[Mar 17 18:18:51] WARNING[24435]: channel.c:2816 set_format: Unable to find a codec translation path from alaw to g729
– Call on SIP/provider-081e6a38 left from hold
<< Console Has Been Retrieved from Hold >>
– SIP/provider-081e6a38 is making progress passing it to OSS/dsp
<— SIP read from 83.98.222.4:5060 —>
SIP/2.0 500 Calling is failed (unspecified)
Via: SIP/2.0/UDP 82.150.141.193:5060;branch=z9hG4bK39d7eee3;rport=5060
Record-Route: sip:83.98.222.4;ftag=as038a9fa1;lr
From: asterisk sip:asterisk@82.150.141.193;tag=as038a9fa1
To: sip:011442476702601@sip1.belcentrale.nl;tag=122eb31f2bb5d31c45b31cb551caa4b4
Call-ID: 059aa03d32888a841e70150021a93f6b@82.150.141.193
CSeq: 102 INVITE
Server: Sippy
<------------->
— (8 headers 0 lines) —
– Got SIP response 500 “Calling is failed (unspecified)” back from 83.98.222.4
Transmitting (no NAT) to 83.98.222.4:5060:
ACK sip:011442476702601@sip1.belcentrale.nl SIP/2.0
Via: SIP/2.0/UDP 82.150.141.193:5060;branch=z9hG4bK39d7eee3;rport
From: “asterisk” sip:asterisk@82.150.141.193;tag=as038a9fa1
To: sip:011442476702601@sip1.belcentrale.nl;tag=122eb31f2bb5d31c45b31cb551caa4b4
Contact: sip:asterisk@82.150.141.193
Call-ID: 059aa03d32888a841e70150021a93f6b@82.150.141.193
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
-- SIP/provider-081e6a38 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel ‘OSS/dsp’ status is ‘CONGESTION’
[Mar 17 18:18:59] WARNING[24432]: chan_oss.c:682 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
Really destroying SIP dialog ‘059aa03d32888a841e70150021a93f6b@82.150.141.193’ Method: INVITE
<< Hangup on console >>
Reliably Transmitting (no NAT) to 83.98.222.4:5060:
OPTIONS sip:sip1.belcentrale.nl SIP/2.0
Via: SIP/2.0/UDP 82.150.141.193:5060;branch=z9hG4bK080e1dea;rport
From: “asterisk” sip:asterisk@82.150.141.193;tag=as5236b9bf
To: sip:sip1.belcentrale.nl
Contact: sip:asterisk@82.150.141.193
Call-ID: 1ca121473046c5307527594c511f556b@82.150.141.193
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 17 Mar 2007 17:19:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<— SIP read from 83.98.222.4:5060 —>
SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP 82.150.141.193:5060;branch=z9hG4bK080e1dea;rport=5060
From: asterisk sip:asterisk@82.150.141.193;tag=as5236b9bf
To: sip:sip1.belcentrale.nl
Call-ID: 1ca121473046c5307527594c511f556b@82.150.141.193
CSeq: 102 OPTIONS
Server: Sippy
<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘1ca121473046c5307527594c511f556b@82.150.141.193’ Method: OPTIONS