[Asterisk-Users] Help with Asterisk@home
andBroadvoiceincomingcalls..
dbruce
dbruce at bananatel.ca
Sun Jul 24 14:04:12 MST 2005
Ok.. I screwed up...
You have a register statement:
register=2405243333 at sip.broadvoice.com:123abc:2405243333 at sip.broadvoice.com/
201
so, the incoming call from broadvoice will be sent to extension 201 in the
frombroadvoice context.
To ensure what is going on, use this as your context.
exten => _X.,1,Noop(Incoming call for extension ${EXTEN} in context
frombroadvoice)
That will tell you exactly what is being sent into the context.
You are using the latest AAH, so the variable substitutions will work.
I expect that you will end up using the following in frombroadvoice:
exten => ${BVRINGS},1,Macro(exten-vm,${BVRINGS}@default,${BVRINGS})
Sorry for the confusion....
Regards,
Derek
----- Original Message -----
From: "Howard Leadmon" <howard at leadmon.net>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<asterisk-users at lists.digium.com>
Sent: Sunday, July 24, 2005 2:27 PM
Subject: RE: [Asterisk-Users] Help with Asterisk at home
andBroadvoiceincomingcalls..
>
> OK, I think I understood what you were saying, but let me type this in
here as
> like I said I am for sure trying to figure this sucker out still..
>
> I just tried the following:
>
> exten => s,1,Macro(exten-vm,${BVRINGS}@default,${BVRINGS})
>
> I also tried this:
>
> exten => 2405243333,1,Macro(exten-vm,${BVRINGS}@default,${BVRINGS})
>
>
> Of course I told it to reload the configs, and I get the exact same
reaction
> from it when I call the number.
>
> Is there any other type of debugging or output that would be helpful?
>
> Also funny you mention that the variable wouldn't work on the extension,
as
> maybe it's done a little different, but I have this for my FreeWorld IAX
> connection and incoming calls on it work great.
>
> [fromiaxfwd]
> exten => ${FWDNUMBER},1,Macro(exten-vm,${FWDRINGS}@default,${FWDRINGS})
> exten => ${VM_PREFIX}${FWDVMBOX},1,Macro(vm,${FWDVMBOX})
>
> With the various variables for FWD set up top. Not sure about what is in
the
> CVSHEAD, I haven't gotten good enough to try that out yet, but I do have
the
> most current asterisk at home, which is using asterisk 1.0.9 at this time.
>
> Anyway still very confused, and hopefully you will have some ideas.
>
>
> ---
> Howard Leadmon - howard at leadmon.net
> http://www.leadmon.net
>
>
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> > bounces at lists.digium.com] On Behalf Of dbruce
> > Sent: Sunday, July 24, 2005 4:08 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Help with Asterisk at home and
> > Broadvoiceincomingcalls..
> >
> > 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
> > >
> > >
> > >
> > > _______________________________________________
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> >
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>
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