[asterisk-users] SIP Incoming / Inbound not working for Broadvoice (Asterisk PBX 1.6.1.6)
Qurba Joog
qurbajoog at gmail.com
Fri Dec 25 00:23:59 CST 2009
Hello,
Please forgive me if I'm repeating this post. I have searched and looked for
similar problem with a solution but have not see a similar one.
My outgoing SIP and other channels work fine but the incoming/inbound SIP
call goes straight to Broadvoice voicemail. I see that Broadvoice is
registered when I look at the SIP registry. I have turned on SIP Debug and
it is below.
Anyone know why even when SIP has registered I do not see incoming calls?
Thanks,
------extensions.conf----------------
[global]
[general]
bindport=5060
bindaddr = 0.0.0.0
deny=0.0.0.0/0.0.0.0
externhost=xyz.dyndns.org
localnet = 192.168.1.0/255.255.255.0
disallow=all
allow=ulaw
allow=gsm
delayreject=yes
nochecksums=no
allowguest=no
delayreject=yes
pedantic=no
register => 703XXXYYYY at sip.broadvoice.com:s
ecurepassword:703XXXYYYY at sip.broadvoice.com/5000
[5000]
type=friend
context=internal-phones
secret=xxx
qualify=yes
host=dynamic ; behind nat
dtmfmode=rfc2833
[5002]
type=friend
context=internal-phones
secret=test
qualify=yes
host=dynamic ; behind nat
nat=yes
dtmfmode=rfc2833
[enter_broadvoice]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=703XXXYYYY
secret=securepassword
username=703XXXYYYY
insecure=very
;insecure=port,invite
context=incoming
authname=703XXXYYYY
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
--------extensions.conf--------
[globals]
[general]
autofallthrough=yes
[incoming_calls]
exten => 1703XXXYYYY,1,Dial(SIP/5000)
[internal-phones]
include => outgoing
exten => 5000,1,Dial(SIP/5000,20)
exten => 5002,1,Dial(SIP/5002,20)
[outgoing]
exten => _X.,1,NoOp()
exten => _X.,n,Dial(SIP/enter_broadvoice/${EXTEN})
--------SIP Registry------------------
-*CLI> sip show registry
Host dnsmgr Username Refresh
State Reg.Time
sip.broadvoice.com:5060 N 703XXXYYYY at s 23
Registered Fri, 25 Dec 2009 01:14:03
--------SIP Debug------------------
-*CLI>
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 147.135.32.226:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 68.100.66.3:5060;branch=z9hG4bK4948ded8;rport
Max-Forwards: 70
From: <sip:703XXXYYYY at sip.broadvoice.com<sip%3A703XXXYYYY at sip.broadvoice.com>
>;tag=as376e46ae
To: <sip:703XXXYYYY at sip.broadvoice.com <sip%3A703XXXYYYY at sip.broadvoice.com>
>
Call-ID: 65a8d48738a00d121fc9050e4771d0b7 at 192.168.1.51
CSeq: 147 REGISTER
User-Agent: Asterisk PBX 1.6.1.6
Expires: 120
Contact: <sip:5000 at 68.100.65.3 <sip%3A5000 at 68.100.65.3>>
Content-Length: 0
---
Really destroying SIP dialog '65a8d48738a00d121fc9050e4771d0b7 at 192.168.1.51'
Method: REGISTER
Suuban*CLI>
<--- SIP read from UDP://147.135.32.225:5060 --->
SIP/2.0 200 OK
Call-ID: 65a8d48738a00d121fc9050e4771d0b7 at 192.168.1.51
CSeq: 147 REGISTER
From: <sip:703XXXYYYY at sip.broadvoice.com<sip%3A703XXXYYYY at sip.broadvoice.com>
>;tag=as376e46ae
To: <sip:703XXXYYYY at sip.broadvoice.com <sip%3A703XXXYYYY at sip.broadvoice.com>
>
Via: SIP/2.0/UDP 68.100.66.3:5060;branch=z9hG4bK4948ded8
Contact: <sip:5000 at 68.100.66.3 <sip%3A5000 at 68.100.66.3>>
Expires: 30
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Scheduling destruction of SIP dialog '
65a8d48738a00d121fc9050e4771d0b7 at 192.168.1.51' in 32000 ms (Method:
REGISTER)
[Dec 25 00:42:31] NOTICE[2898]: chan_sip.c:16740 handle_response_register:
Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling
reregistration in 23 s)
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 147.135.32.226:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 68.100.66.3:5060;branch=z9hG4bK35a02655;rport
Max-Forwards: 70
From: <sip:703XXXYYYY at sip.broadvoice.com<sip%3A703XXXYYYY at sip.broadvoice.com>
>;tag=as54c6327a
To: <sip:703XXXYYYY at sip.broadvoice.com <sip%3A703XXXYYYY at sip.broadvoice.com>
>
Call-ID: 65a8d48738a00d121fc9050e4771d0b7 at 192.168.1.51
CSeq: 148 REGISTER
User-Agent: Asterisk PBX 1.6.1.6
Expires: 120
Contact: <sip:5000 at 68.100.66.3 <sip%3A5000 at 68.100.66.3>>
Content-Length: 0
---
Really destroying SIP dialog '65a8d48738a00d121fc9050e4771d0b7 at 192.168.1.51'
Method: REGISTER
Suuban*CLI>
<--- SIP read from UDP://147.135.32.226:5060 --->
SIP/2.0 200 OK
Call-ID: 65a8d48738a00d121fc9050e4771d0b7 at 192.168.1.51
CSeq: 148 REGISTER
From: <sip:703XXXYYYY at sip.broadvoice.com<sip%3A703XXXYYYY at sip.broadvoice.com>
>;tag=as54c6327a
To: <sip:703XXXYYYY at sip.broadvoice.com <sip%3A703XXXYYYY at sip.broadvoice.com>
>
Via: SIP/2.0/UDP 68.100.66.3:5060;branch=z9hG4bK35a02655
Contact: <sip:5000 at 68.100.66.3 <sip%3A5000 at 68.100.66.3>>
Expires: 30
Content-Length: 0
----------------------------------------------
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091225/3b986046/attachment.htm
More information about the asterisk-users
mailing list