Hi
I have what seems like a similar problem. In the last few days, I stopped
receiving calls from
my Broadvoice number into Asterisk. The account activity page at Broadvoice does
not show the
calls as either missed or incoming. Unfortunately, Broadvoice's line is
"we don't support
Asterisk". :-/ Can you see the calls you're attempting to place in your
Broadvoice activity
page?
Klaus
--- Michael Stearne <mstearne@entermix.com> wrote:
> 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@209.3.28.xx:5060 SIP/2.0
> Call-ID: 1ff023a-69@147.135.20.128
> CSeq: 1 INVITE
> From: "Brooklyn
NY"<sip:3472674xxx@147.135.20.128;user=phone>;tag=ikmn
> To: "Michael Stearne"<sip:s@209.3.28.xx;user=phone>
> Via: SIP/2.0/UDP 147.135.20.128:5060
> Contact: <sip:3472674xxx@147.135.20.128:5060>
> Supported: 100rel
> RPID-Privacy: party=calling;id-type=subscriber;privacy=off
> Remote-Party-ID:
> <sip:3472674xxx@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@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@147.135.20.128;user=phone>;tag=ikmn
> To: "Michael
Stearne"<sip:s@209.3.28.xx;user=phone>;tag=as38d08027
> Call-ID: 1ff023a-69@147.135.20.128
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
> Contact: <sip:6092991xxx@209.3.28.35>
> Content-Length: 0
>
>
> ---
>
> <-- SIP read from 147.135.20.128:5060:
> ACK sip:s@209.3.28.xx:5060 SIP/2.0
> Call-ID: 1ff023a-69@147.135.20.128
> CSeq: 1 ACK
> From: "Brooklyn
NY"<sip:3472674xxx@147.135.20.128;user=phone>;tag=ikmn
> To: "Michael
Stearne"<sip:s@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@147.135.20.128'
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