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

Sun Jul 24 13:08:29 MST 2005

Your [frombroadvoice] context is incorrect. You have set a global variable
BVNUMBER and used it as the extension match in the context. The problem is
that the extension match syntax does not support variable substitution
unless you are using a relatively current CVS HEAD. As Asterisk at home is
based on CVS STABLE, you can't use variable substitution.

You will need to replace the ${BVNUMBER} with valid extension match syntax.
You can use the 's' extension or a general match patern '_X." and do the
specific matching within the dialplan to determine is you wish to accept the
call ie: gotoif($["${BVNUMBER}" = "${EXTEN}"]?x) (replacing 'x' with a valid
priority).

Regards,
Derek

----- Original Message -----
From: "Howard Leadmon" <howard at leadmon.net>
To: <asterisk-users at lists.digium.com>
Sent: Sunday, July 24, 2005 1:34 PM
Subject: [Asterisk-Users] Help with Asterisk at home and Broadvoice
incomingcalls..

>
>    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=SD2o51f01-520831772-1122233
07
> 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=SD2o51f01-520831772-1122233
07
> 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=SD2o51f01-520831772-1122233
07
> 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
>
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users