[Asterisk-Users] Asterisk Won't Process Call

Michael Stearne mstearne at entermix.com
Sun Sep 18 08:52:55 MST 2005


We have a basic application that runs a SIP channel to pick up a call
and process it.  We are using Broadvoice and it's been working great. 
We recently rebooted the machine and now when a call comes in Asterisk
picks up the call but does not process it.  Asterisk seems to send the
call back to Broadvoice.  Nothing at all has been changed in the
configuration to warrant this.  Below is the output of sip debug.  Any
help would be a life saver!

<-- SIP read from 147.135.20.128:5060: 
INVITE sip:6092991xxx at 209.3.28.xx:5060 SIP/2.0
Call-ID: 1ff023a-69 at 147.135.20.128
CSeq: 1 INVITE
From: "Brooklyn NY"<sip:3472674xxx at 147.135.20.128;user=phone>;tag=ikmn
To: "Michael Stearne"<sip:s at 209.3.28.xx;user=phone>
Via: SIP/2.0/UDP 147.135.20.128:5060
Contact: <sip:3472674xxx at 147.135.20.128:5060>
Supported: 100rel
RPID-Privacy: party=calling;id-type=subscriber;privacy=off
Remote-Party-ID:
<sip:3472674xxx at 147.135.20.128>;screen=yes;party=calling;privacy=off
Content-Length:  273
Content-Type: application/sdp

v=0
o=2475103479 10 10 IN IP4 147.135.20.247
s=-
c=IN IP4 147.135.20.250
t=0 0
m=audio 10690 RTP/AVP 0 8 2 18 96 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 iLBC/8000
a=rtpmap:101 telephone-event/8000

--- (12 headers 12 lines)---
Using INVITE request as basis request - 1ff023a-69 at 147.135.20.128
Sending to 147.135.20.128 : 5060 (non-NAT)
Found peer 'sip2.broadvoice.com'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 2
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 101
Peer audio RTP is at port 147.135.20.250:10690
Found description format PCMU
Found description format PCMA
Found description format G726-32
Found description format G729
Found description format iLBC
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x51c
(ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc
(ulaw|alaw)Non-codec capabilities: us - 0x1 (telephone-event), peer -
0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 6092991xxx in from-broadvoice
Reliably Transmitting (no NAT) to 147.135.20.128:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 147.135.20.128:5060
From: "Brooklyn NY"<sip:3472674xxx at 147.135.20.128;user=phone>;tag=ikmn
To: "Michael Stearne"<sip:s at 209.3.28.xx;user=phone>;tag=as38d08027
Call-ID: 1ff023a-69 at 147.135.20.128
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:6092991xxx at 209.3.28.35>
Content-Length: 0


---

<-- SIP read from 147.135.20.128:5060: 
ACK sip:s at 209.3.28.xx:5060 SIP/2.0
Call-ID: 1ff023a-69 at 147.135.20.128
CSeq: 1 ACK
From: "Brooklyn NY"<sip:3472674xxx at 147.135.20.128;user=phone>;tag=ikmn
To: "Michael Stearne"<sip:s at 209.3.28.xx;user=phone>;tag=as38d08027
Via: SIP/2.0/UDP 147.135.20.128:5060;received=209.3.28.xx
Content-Length:    0


--- (7 headers 0 lines)---
Destroying call '1ff023a-69 at 147.135.20.128'



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