[Asterisk-Users] Asterisk and Broadvoice, no incoming voice

Jason Schafer js at have.net
Mon Sep 26 09:08:10 MST 2005 

> Does asterisk says something in the verbose console?

I'm not sure what the verbose console is, but I can run sip debug and 
post the output when I make an inbound call.

> please post your sip.conf relevant entries for BroadVoice. 

[general]
port = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
pedantic=no ; added for Broadvoice support 8/3/05 EK
externip=216.xxx.xxx.xxx
localnet=172.xxx.xxx.0/255.255.255.0


I have just
> cancelled with BroadVoice (too much latency for the places i wanted to 
> call), so i never used the incoming number. But im glad to help if i can.

I have outbound setup on VOIPJet, my intent with the Broadvoice is to 
setup a forward on busy with my landline to roll over to the BV number.

Here's the output from sip debug

m=audio 14008 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

13 headers, 12 lines
Using latest request as basis request
Sending to 147.135.0.128 : 5060 (non-NAT)
Found no matching peer or user for '147.135.0.128:5060'
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 147.135.0.128:14008
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 - 0xc (ulaw|alaw), peer - audio=0x51c 
(ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 
0x1 (g723)
Looking for s in from-sip-external
list_route: hop: 
<sip:610253xxxx at 147.135.0.128:5060;ep=147.135.0.129;transport=udp>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0e4102041k9sak0k1.1sr
From: "Schafer Trish 
"<sip:610253xxxx at 147.135.0.129;user=phone>;tag=SD28clb01-1612693231-1127750324179
To: "Jason Schafer"<sip:484549xxxx at sip.broadvoice.com;user=phone>
Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002
CSeq: 160704490 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:s at 216.xxx.xxx.xxx>
Content-Length: 0


  to 147.135.0.128:5060
     -- Executing Wait("SIP/147.135.0.129-095da350", "1") in new stack
     -- Executing Goto("SIP/147.135.0.129-095da350", "from-pstn|s|1") in 
new stack
     -- Goto (from-pstn,s,1)
     -- Executing GotoIf("SIP/147.135.0.129-095da350", 
"1?from-pstn-reghours|s|1:") in new stack
     -- Goto (from-pstn-reghours,s,1)
     -- Executing GotoIf("SIP/147.135.0.129-095da350", 
"0?from-pstn-reghours-nofax|s|1:2") in new stack
     -- Goto (from-pstn-reghours,s,2)
     -- Executing Answer("SIP/147.135.0.129-095da350", "") in new stack
We're at 216.xxx.xxx.xxx port xxxxx
Answering with preferred capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0e4102041k9sak0k1.1sr
From: "Schafer Trish 
"<sip:610253xxxx at 147.135.0.129;user=phone>;tag=SD28clb01-1612693231-1127750324179
To: "Jason 
Schafer"<sip:484549xxxx at sip.broadvoice.com;user=phone>;tag=as2a994d31
Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002
CSeq: 160704490 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:s at 216.xxx.xxx.xxx>
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 1782 1782 IN IP4 216.xxx.xxx.xxx
s=session
c=IN IP4 216.xxx.xxx.xxx
t=0 0
m=audio 14138 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

  to 147.135.0.128:5060
     -- Executing Wait("SIP/147.135.0.129-095da350", "1") in new stack
asterisk1*CLI>

Sip read:
ACK sip:s at 172.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0u4103gtgb94c0080.1sr
From: "Schafer Trish 
"<sip:610253xxxx at 147.135.0.129;user=phone>;tag=SD28clb01-1612693231-1127750324179
To: "Jason 
Schafer"<sip:484549xxxx at sip.broadvoice.com;user=phone>;tag=as2a994d31
Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002
CSeq: 160704490 ACK
Contact: <sip:610253xxxx at 147.135.0.128:5060;transport=udp>
Max-Forwards: 69
Content-Length: 0


9 headers, 0 lines
     -- Executing SetVar("SIP/147.135.0.129-095da350", "intype=aa_2") in 
new stack
     -- Executing Cut("SIP/147.135.0.129-095da350", "intype=intype|-|1") 
in new stack
     -- Executing GotoIf("SIP/147.135.0.129-095da350", "0?7:9") in new stack
     -- Goto (from-pstn-reghours,s,9)
     -- Executing GotoIf("SIP/147.135.0.129-095da350", "0?10:12") in new 
stack
     -- Goto (from-pstn-reghours,s,12)
     -- Executing GotoIf("SIP/147.135.0.129-095da350", "0?13:15") in new 
stack
     -- Goto (from-pstn-reghours,s,15)
     -- Executing Goto("SIP/147.135.0.129-095da350", "aa_2|s|1") in new 
stack
     -- Goto (aa_2,s,1)
     -- Executing GotoIf("SIP/147.135.0.129-095da350", "0?4") in new stack
     -- Executing Answer("SIP/147.135.0.129-095da350", "") in new stack
     -- Executing Wait("SIP/147.135.0.129-095da350", "1") in new stack
asterisk1*CLI>

Sip read:


0 headers, 0 lines
     -- Executing SetVar("SIP/147.135.0.129-095da350", 
"DIR-CONTEXT=general") in new stack
     -- Executing DigitTimeout("SIP/147.135.0.129-095da350", "3") in new 
stack
     -- Set Digit Timeout to 3
     -- Executing ResponseTimeout("SIP/147.135.0.129-095da350", "7") in 
new stack
     -- Set Response Timeout to 7
     -- Executing BackGround("SIP/147.135.0.129-095da350", 
"custom/aa_2") in new stack
     -- Playing 'custom/aa_2' (language 'en')
asterisk1*CLI>

Sip read:


0 headers, 0 lines
12 headers, 0 lines
Reliably Transmitting:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 216.xxx.xxx.xxx:5060;branch=z9hG4bK0cefd7e0
From: <sip:484549xxxx at sip.broadvoice.com>;tag=as630d5ca8
To: <sip:484549xxxx at sip.broadvoice.com>
Call-ID: 2d3c52c27ad7b993384cb89f5bca4c9b at 127.0.0.1
CSeq: 135 REGISTER
User-Agent: Asterisk PBX
Authorization: Digest username="484549xxxx", realm="BroadWorks", 
algorithm=MD5, uri="sip:sip.broadvoice.com", nonce="1127748659968", 
response="a41b375d644bc72e8ef3a7049435c1e7", opaque=""
Expires: 120
Contact: <sip:s at 216.xxx.xxx.xxx>
Event: registration
Content-Length: 0

  (no NAT) to 147.135.0.128:5060
asterisk1*CLI>

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.xxx.xxx.xxx:5060;branch=z9hG4bK0cefd7e0
From: <sip:484549xxxx at sip.broadvoice.com>;tag=as630d5ca8
To: <sip:484549xxxx at sip.broadvoice.com>
Call-ID: 2d3c52c27ad7b993384cb89f5bca4c9b at 127.0.0.1
CSeq: 135 REGISTER
Contact: <sip:s at 172.xxx.xxx.xxx>;expires=1918


7 headers, 0 lines
Destroying call '2d3c52c27ad7b993384cb89f5bca4c9b at 127.0.0.1'
asterisk1*CLI>

Sip read:
BYE sip:s at 172.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0ee10do30sas9b3s1.1sr
From: "Schafer Trish 
"<sip:610253xxxx at 147.135.0.129;user=phone>;tag=SD28clb01-1612693231-1127750324179
To: "Jason 
Schafer"<sip:484549xxxx at sip.broadvoice.com;user=phone>;tag=as2a994d31
Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002
CSeq: 160704491 BYE
Max-Forwards: 69
Content-Length: 0


8 headers, 0 lines
Sending to 147.135.0.128 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0ee10do30sas9b3s1.1sr
From: "Schafer Trish 
"<sip:610253xxxx at 147.135.0.129;user=phone>;tag=SD28clb01-1612693231-1127750324179
To: "Jason 
Schafer"<sip:484549xxxx at sip.broadvoice.com;user=phone>;tag=as2a994d31
Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002
CSeq: 160704491 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:s at 216.xxx.xxx.xxx>
Content-Length: 0


  to 147.135.0.128:5060
   == Spawn extension (aa_2, s, 7) exited non-zero on 
'SIP/147.135.0.129-095da350'
     -- Executing Hangup("SIP/147.135.0.129-095da350", "") in new stack
   == Spawn extension (aa_2, h, 1) exited non-zero on 
'SIP/147.135.0.129-095da350'
Destroying call 'SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002'
asterisk1*CLI>

Sip read:


0 headers, 0 lines
asterisk1*CLI>

Sip read:


0 headers, 0 lines
asterisk1*CLI>

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