asterisk users - Apr 2005 - BROADVOICE - Incomming calls are dropped after 1-2 min

Hi,

All my calls are dropped after 1-2 min. (sometimes less)
Did anyone experience similar problems.
When the call is dropped asterisk cli says:
spawn extension menu,2,7 which is the last extension that asterisk
executed before the call dropped.
How can I find out why the call was dropped.

Bart,

Bart,

Greetings. I saw your posting on the asterisk users list regarding 
broadvoice.  I am trying to fix my own broadvoice issue with asterisk as 
well, but I have not even been able to get to your issue yet.  I can 
make outgoing calls, however no incoming calls work.  If I may ask, how 
did you come upon your configuration for asterisk and broadvoice?

Thanks
Craig



Bartosz Wegrzyn - asterisk wrote:
>Hi,
>
>All my calls are dropped after 1-2 min. (sometimes less)
>Did anyone experience similar problems.
>When the call is dropped asterisk cli says:
>spawn extension menu,2,7 which is the last extension that asterisk
>executed before the call dropped.
>How can I find out why the call was dropped.
>
>Bart,
>
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@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
>  
>

Ok here are my config files.
Are you behind any firewall? Are you using NAT?

//cat sip.conf
[general]
externip=your domain name
bindaddr = 0.0.0.0
port=5060
localnet=192.168.1.0/255.255.255.0
disallow=all
allow=ulaw
register => phonenum:pass@sip.broadvoice.com
tos=0x18
srvlookup=yes
nat=never
insecure=yes
realm=your domain name

[sip.broadvoice.com]
type=peer
username=phonenum
fromuser=phonenum
authuser=phonenum
secret=secret
host=sip.broadvoice.com
context=sip
fromdomain=sip.broadvoice.com
canreinvite=no
nat=never
dtmfmode=inband

[broadvoice-incoming]
type=peer
host=147.135.8.128
context=from-broadvoice
qualify=yes
canreinvite=no
disallow=all
allow=ulaw
nat=never

[broadvoice-incoming2]
type=peer
host=147.135.0.128
context=from-broadvoice
qualify=yes
canreinvite=no
disallow=all
allow=ulaw
nat=never

[broadvoice-incoming3]
type=peer
host=147.135.4.128
context=from-broadvoice
qualify=yes
canreinvite=no
disallow=all
allow=ulaw
nat=never

//cat extensions.conf (only part for broadvoice)
;incoming calls from broad voice
[from-broadvoice]
exten => s,1,Ringing()
exten => s,2,wait(5)
exten => s,3,Playback(welcome)
exten => h,1,Hangup
exten => i,1,Hangup
exten => t,5,Hangup

[out-broad]
exten => _1NXXNXXXXXX,1,wait();
exten => _1NXXNXXXXXX,2,Dial(SIP/${EXTEN-1}@sip.broadvoice.com)
exten => _1NXXNXXXXXX,3,Congestion
exten => h,1,Hangup
exten => i,1,Playback(invalid)
exten => i,2,Playback(please-try-again)
exten => t,5,Hangup

[out-broad-world]
exten=_01130.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01131.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01132.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01133.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01134.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_011351.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_011352.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_011353.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_011378.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01139.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01141.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_011420.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01143.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01144.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01145.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01146.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01147.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01148.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01149[2-9].,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01154.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01155.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01156.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01160.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01161.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01164.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01165.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01181.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01182.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_011852.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_01186.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_011886.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_011972.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten=_011.,2,congestion() ; No answer, nothing
exten=_011.,102,busy() ; Busy

I hope above helps.
Include whatever you need in your default plan in asterisk.

Bart,


> Bart,
>
> Greetings. I saw your posting on the asterisk users list regarding
> broadvoice.  I am trying to fix my own broadvoice issue with asterisk as
> well, but I have not even been able to get to your issue yet.  I can
> make outgoing calls, however no incoming calls work.  If I may ask, how
> did you come upon your configuration for asterisk and broadvoice?
>
> Thanks
> Craig
>
>
>
> Bartosz Wegrzyn - asterisk wrote:
>
>>Hi,
>>
>>All my calls are dropped after 1-2 min. (sometimes less)
>>Did anyone experience similar problems.
>>When the call is dropped asterisk cli says:
>>spawn extension menu,2,7 which is the last extension that asterisk
>>executed before the call dropped.
>>How can I find out why the call was dropped.
>>
>>Bart,
>>
>>
>>_______________________________________________
>>Asterisk-Users mailing list
>>Asterisk-Users@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
>>
>>
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@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
>

I have a writeup on configuring BroadVoice with Asterisk that many people
have used successfully at http://www.geekgazette.com
-Kerry
 

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Craig Simon
Sent: Monday, April 11, 2005 3:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] BROADVOICE - Incomming calls are dropped
after1-2 min

Bart,

Greetings. I saw your posting on the asterisk users list regarding
broadvoice.  I am trying to fix my own broadvoice issue with asterisk as
well, but I have not even been able to get to your issue yet.  I can make
outgoing calls, however no incoming calls work.  If I may ask, how did you
come upon your configuration for asterisk and broadvoice?

Thanks
Craig



Bartosz Wegrzyn - asterisk wrote:
>Hi,
>
>All my calls are dropped after 1-2 min. (sometimes less) Did anyone 
>experience similar problems.
>When the call is dropped asterisk cli says:
>spawn extension menu,2,7 which is the last extension that asterisk 
>executed before the call dropped.
>How can I find out why the call was dropped.
>
>Bart,
>
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@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
>  
>

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@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