[Asterisk-Users] Help with Asterisk@home and Broadvoice incoming calls..

Sun Jul 24 12:34:01 MST 2005

   Hello everyone,

 Well here is my initial posting to the list, and I will admit Asterisk is new
to me. I just got everything running here a couple days ago, so still learning
the ropes for sure.

 OK, here is my problem.   Currently I have it setup talking to a couple Cisco
IP phones, and some Xten softphones, this works great.   I also got an account
with FreeWorld Dialup using IAX2 and that works super both inbound and
outbound at this time.   I decided to sign up with BroadVoice as they had good
pricing, seems like well supported in the Asterisk community.  

 So when I setup with BroadVoice I got the outgoing calls to them working just
fine, I set it up so I can dial 8, and then any number I desire to reach and
the call goes through.   Now as simple as I thought this would be, if I try
and get an incoming call, it just doesn't work, I think it rolls right into
the BroadVoice Vmail they provide, as nothing rings here, so figure something
is messed up in the call pathway.

 I have spend hours looking at the debug output, and though some of it makes
good sense, I am just to green to really dig into the guts of this sucker yet,
hopefully that will change for me soon.  So I hope someone here on the list
can help me figure out what the heck is wrong with this, and get my incoming
calls from BroadVoice and get this sucker working.

 I am not sure what all information is needed, but I'll post some bits of
output below (with numbers changed), so maybe it will give someone a chance to
help me with this.

In my sip.conf I have:

register=2405243333 at sip.broadvoice.com:123abc:2405243333 at sip.broadvoice.com/20
1

[sip.broadvoice.com]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=2405243333
secret=123abc
username=2405243333
insecure=very
context=frombroadvoice
authname=2405243333
dtmfmode=inband
dtmf=inband 

In my extensions.conf I have:

;setup SIP extension for BroadVoice
[globals]
BVNUMBER=2405243333 ; your calling number 
BVRINGS=201 ; the phone to ring
BVVMBOX=201 ; the VM box for this user

[outrt-003-BroadVoice]
include => outrt-003-BroadVoice-custom
exten => _8.,1,Dial(SIP/${EXTEN:1}@sip.broadvoice.com,30)
;exten => _8.,1,Dial(SIP/${EXTEN:1}@2405243333,30)
exten => _8.,2,Congestion()
exten => _8.,102,Busy()

[frombroadvoice] 
exten => ${BVNUMBER},1,Macro(exten-vm,${BVRINGS}@default,${BVRINGS})
exten => ${VM_PREFIX}${BVVMBOX},1,Macro(vm,${BVVMBOX})

If I look at my normal log output when trying to call in, I see:

Jul 24 15:23:12 DEBUG[1078]: Setting NAT on RTP to 0
Jul 24 15:23:12 DEBUG[1078]: Check for res for 2405243333
Jul 24 15:23:12 DEBUG[1078]: 2405243333 is not a local user
Jul 24 15:23:12 DEBUG[1078]: 2405243333 is not a local user
Jul 24 15:23:12 DEBUG[1078]: Stopping retransmission on
'SD28c9b01-2d5e97b21c9e4e488ce05aeda05558a8-js11002' of Response 623264158:
Found

Now I figured I would turn on 'sip debug' to which I see a lot more, here is
some of that output:

Jul 24 15:24:33 VERBOSE[1078]: 

Sip read: 
INVITE sip:201 at 207.114.0.111 SIP/2.0
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK2qo3b3307oi0j7onc400.1sr
From: "Fork
MD"<sip:4105156666 at 147.135.0.129;user=phone>;tag=SD2o51f01-520831772-112223307
3802
To: "Howard Leadmon"<sip:2405243333 at sip.broadvoice.com;user=phone>
Call-ID: SD2o51f01-62b2c5360881c20fae16de626946fdf0-js11002
CSeq: 623304774 INVITE
Contact: <sip:4105156666 at 147.135.0.128:5060;ep=147.135.0.129;transport=udp>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY
Supported: 100rel
Accept: application/sdp,application/dtmf
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 276

v=0
o=BroadWorks 24463992 1 IN IP4 147.135.0.128
s=-
c=IN IP4 147.135.0.128
t=0 0
m=audio 14942 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

Jul 24 15:24:33 VERBOSE[1078]: 13 headers, 12 lines
Jul 24 15:24:33 VERBOSE[1078]: Using latest request as basis request
Jul 24 15:24:33 VERBOSE[1078]: Sending to 147.135.0.128 : 5060 (non-NAT)
Jul 24 15:24:33 VERBOSE[1078]: Found peer 'sip.broadvoice.com'
Jul 24 15:24:33 DEBUG[1078]: Setting NAT on RTP to 0
Jul 24 15:24:33 VERBOSE[1078]: Found RTP audio format 0
Jul 24 15:24:33 VERBOSE[1078]: Found RTP audio format 8
Jul 24 15:24:33 VERBOSE[1078]: Found RTP audio format 2
Jul 24 15:24:33 VERBOSE[1078]: Found RTP audio format 18
Jul 24 15:24:33 VERBOSE[1078]: Found RTP audio format 96
Jul 24 15:24:33 VERBOSE[1078]: Found RTP audio format 101
Jul 24 15:24:33 VERBOSE[1078]: Peer audio RTP is at port 147.135.0.128:14942
Jul 24 15:24:33 DEBUG[1078]: Peer audio RTP is at port 147.135.0.128:14942
Jul 24 15:24:33 VERBOSE[1078]: Found description format PCMU
Jul 24 15:24:33 VERBOSE[1078]: Found description format PCMA
Jul 24 15:24:33 VERBOSE[1078]: Found description format G726-32
Jul 24 15:24:33 VERBOSE[1078]: Found description format G729
Jul 24 15:24:33 VERBOSE[1078]: Found description format iLBC
Jul 24 15:24:33 VERBOSE[1078]: Found description format telephone-event
Jul 24 15:24:33 VERBOSE[1078]: Capabilities: us - 0xc (ulaw|alaw), peer -
audio=0x51c (ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc
(ulaw|alaw)
Jul 24 15:24:33 VERBOSE[1078]: Non-codec capabilities: us - 0x1 (g723), peer -
0x1 (g723), combined - 0x1 (g723)
Jul 24 15:24:33 DEBUG[1078]: Check for res for 2405243333
Jul 24 15:24:33 DEBUG[1078]: 2405243333 is not a local user
Jul 24 15:24:33 VERBOSE[1078]: Looking for 201 in frombroadvoice
Jul 24 15:24:33 VERBOSE[1078]: Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK2qo3b3307oi0j7onc400.1sr
From: "Fork
MD"<sip:4105156666 at 147.135.0.129;user=phone>;tag=SD2o51f01-520831772-112223307
3802
To: "Howard
Leadmon"<sip:2405243333 at sip.broadvoice.com;user=phone>;tag=as524e3026
Call-ID: SD2o51f01-62b2c5360881c20fae16de626946fdf0-js11002
CSeq: 623304774 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:201 at 207.114.0.111>
Content-Length: 0

 to 147.135.0.128:5060
Jul 24 15:24:33 DEBUG[1078]: 2405243333 is not a local user
Jul 24 15:24:33 VERBOSE[1078]: 

Sip read: 
ACK sip:201 at 207.114.0.111 SIP/2.0
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK2qo3b3307oi0j7onc400.1sr
From: "Fork
MD"<sip:4105156666 at 147.135.0.129;user=phone>;tag=SD2o51f01-520831772-112223307
3802
To: "Howard
Leadmon"<sip:2405243333 at sip.broadvoice.com;user=phone>;tag=as524e3026
Call-ID: SD2o51f01-62b2c5360881c20fae16de626946fdf0-js11002
CSeq: 623304774 ACK

Jul 24 15:24:33 VERBOSE[1078]: 6 headers, 0 lines
Jul 24 15:24:33 DEBUG[1078]: Stopping retransmission on
'SD2o51f01-62b2c5360881c20fae16de626946fdf0-js11002' of Response 623304774:
Found
Jul 24 15:24:33 VERBOSE[1078]: Destroying call
'SD2o51f01-62b2c5360881c20fae16de626946fdf0-js11002'

I worked though most of my other issues, but this one has for sure been
kicking my butt, after spending a LOT of hours trying to track it, I figured
it was time to see if someone with more experience could lend a hand.  Would
be real nice to get incoming calls to this box working, so any help is much
appreciated...

---
Howard Leadmon - http://www.leadmon.net 