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Howard Leadmon

2005-Jul-24 12:34 UTC

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

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@sip.broadvoice.com:123abc:2405243333@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@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@147.135.0.129;user=phone>;tag=SD2o51f01-520831772-112223307
3802
To: "Howard
Leadmon"<sip:2405243333@sip.broadvoice.com;user=phone>
Call-ID: SD2o51f01-62b2c5360881c20fae16de626946fdf0-js11002
CSeq: 623304774 INVITE
Contact:
<sip:4105156666@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@147.135.0.129;user=phone>;tag=SD2o51f01-520831772-112223307
3802
To: "Howard
Leadmon"<sip:2405243333@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@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@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@147.135.0.129;user=phone>;tag=SD2o51f01-520831772-112223307
3802
To: "Howard
Leadmon"<sip:2405243333@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

dbruce

2005-Jul-24 13:08 UTC

head link

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

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@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@leadmon.net>
To: <asterisk-users@lists.digium.com>
Sent: Sunday, July 24, 2005 1:34 PM
Subject: [Asterisk-Users] Help with Asterisk@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@sip.broadvoice.com:123abc:2405243333@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@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@147.135.0.129;user=phone>;tag=SD2o51f01-520831772-1122233
07> 3802
> To: "Howard
Leadmon"<sip:2405243333@sip.broadvoice.com;user=phone>
> Call-ID: SD2o51f01-62b2c5360881c20fae16de626946fdf0-js11002
> CSeq: 623304774 INVITE
> Contact:
<sip:4105156666@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@147.135.0.129;user=phone>;tag=SD2o51f01-520831772-1122233
07> 3802
> To: "Howard
>
Leadmon"<sip:2405243333@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@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@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@147.135.0.129;user=phone>;tag=SD2o51f01-520831772-1122233
07> 3802
> To: "Howard
>
Leadmon"<sip:2405243333@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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> http://lists.digium.com/mailman/listinfo/asterisk-users
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