[Asterisk-Users] Trouble completing a call.

Thu Nov 10 16:52:47 MST 2005

About 80% of the time I get a reorder trying to call out from my
Asterisk box (FreeBSD 5.4-RELEASE / Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h)
talking to a cisco 3640 (IOS 3600 Software C3640-IS-M, Version
12.3(16), RELEASE SOFTWARE (fc4)).  

I have 23 DS0 that rarely have more than 2 calls going out (perhaps
more if this can be fixed) and the same coming in.  

The following is a SIP debug output from a failed call.

Any thing I am missing on this?

Thanks much...
Tim
---
asterisk2*CLI>
    -- Executing SetCIDNum("SIP/1015-ab1c", "14153492100") in new stack
    -- Executing Dial("SIP/1015-ab1c", "SIP/17033911880 at 192.168.1.1") in
new stack
We're at 192.168.1.2 port 17990
Answering/Requesting with root capability 0x4 (ulaw)
Answering with capability 0x2 (gsm)
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:17033911880 at 192.168.1.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK757f3e87
From: "Tim Pozar" <sip:14153492100 at 192.168.1.2>;tag=as1bd4c20b
To: <sip:17033911880 at 192.168.1.1>
Contact: <sip:14153492100 at 192.168.1.2>
Call-ID: 57af3d0431cefe950cac01bf70a6a918 at 192.168.1.2
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Thu, 10 Nov 2005 23:44:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 58967 58967 IN IP4 192.168.1.2
s=session
c=IN IP4 192.168.1.2
t=0 0
m=audio 17990 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 (no NAT) to 192.168.1.1:5060
    -- Called 17033911880 at 192.168.1.1
asterisk2*CLI>

Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK757f3e87
From: "Tim Pozar" <sip:14153492100 at 192.168.1.2>;tag=as1bd4c20b
To: <sip:17033911880 at 192.168.1.1>;tag=B9E2BB70-13D5
Date: Thu, 10 Nov 2005 23:44:09 GMT
Call-ID: 57af3d0431cefe950cac01bf70a6a918 at 192.168.1.2
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0

10 headers, 0 lines
asterisk2*CLI>

Sip read:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK757f3e87
From: "Tim Pozar" <sip:14153492100 at 192.168.1.2>;tag=as1bd4c20b
To: <sip:17033911880 at 192.168.1.1>;tag=B9E2BB70-13D5
Date: Thu, 10 Nov 2005 23:44:09 GMT
Call-ID: 57af3d0431cefe950cac01bf70a6a918 at 192.168.1.2
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Contact: <sip:17033911880 at 192.168.1.1:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 232

v=0
o=CiscoSystemsSIP-GW-UserAgent 5794 8607 IN IP4 192.168.1.1
s=SIP Call
c=IN IP4 192.168.1.1
t=0 0
m=audio 17658 RTP/AVP 0 101
c=IN IP4 192.168.1.1
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

13 headers, 10 lines
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.1:17658
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4
(ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1
(g723)
    -- SIP/192.168.1.1-a06f is making progress passing it to SIP/1015-ab1c
Reliably Transmitting:
CANCEL sip:17033911880 at 192.168.1.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK757f3e87
From: "Tim Pozar" <sip:14153492100 at 192.168.1.2>;tag=as1bd4c20b
To: <sip:17033911880 at 192.168.1.1>
Contact: <sip:14153492100 at 192.168.1.2>
Call-ID: 57af3d0431cefe950cac01bf70a6a918 at 192.168.1.2
CSeq: 102 CANCEL
User-Agent: Asterisk
Content-Length: 0

 (no NAT) to 192.168.1.1:5060
Scheduling destruction of call
'57af3d0431cefe950cac01bf70a6a918 at 192.168.1.2' in 15000 ms
  == Spawn extension (default, 917033911880, 2) exited non-zero on
'SIP/1015-ab1c'
asterisk2*CLI>

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK757f3e87
From: "Tim Pozar" <sip:14153492100 at 192.168.1.2>;tag=as1bd4c20b
To: <sip:17033911880 at 192.168.1.1>
Date: Thu, 10 Nov 2005 23:44:12 GMT
Call-ID: 57af3d0431cefe950cac01bf70a6a918 at 192.168.1.2
Content-Length: 0
CSeq: 102 CANCEL

8 headers, 0 lines
asterisk2*CLI>

Sip read:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK757f3e87
From: "Tim Pozar" <sip:14153492100 at 192.168.1.2>;tag=as1bd4c20b
To: <sip:17033911880 at 192.168.1.1>;tag=B9E2BB70-13D5
Date: Thu, 10 Nov 2005 23:44:12 GMT
Call-ID: 57af3d0431cefe950cac01bf70a6a918 at 192.168.1.2
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0

10 headers, 0 lines
Transmitting:
ACK sip:17033911880 at 192.168.1.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK757f3e87
From: "Tim Pozar" <sip:14153492100 at 192.168.1.2>;tag=as1bd4c20b
To: <sip:17033911880 at 192.168.1.1>;tag=B9E2BB70-13D5
Contact: <sip:14153492100 at 192.168.1.2>
Call-ID: 57af3d0431cefe950cac01bf70a6a918 at 192.168.1.2
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0

 (no NAT) to 192.168.1.1:5060
Destroying call '57af3d0431cefe950cac01bf70a6a918 at 192.168.1.2'
asterisk2*CLI> 