asterisk users - Apr 2006 - Problem: ringtones stop unexpectedly when multiple channels are dialed

Hello Everyone.  I usually find my own solutions for problems but this time,
after several months, I've given up.

 

My asterisk is set up so that incoming calls from my voip provider ring on
both my sip extension and my cellphone at the same time.  When the system
receives an incoming call, ringtones indicating that the call is being
connected play normally for the first 5 seconds to the caller, but they
suddenly stop as the call to my cellphone starts to make progress.  This
causes some people to hang up, despite the fact that the call is still being
connected.  Callers who stay on the line are able to talk to me on either
the sip extension or the cellphone once I pick up either one.

 

I have tried a lot of workarounds like including a priority to answer the
incoming call, invoke the playtones command before the dial command, but
this doesn't seem to work either.  Can anyone replicate the problem?  Have I
ran into a bug?  I have pasted as much info as I deemed relevant; please let
me know if I'm missing something.  Thanks.

 

Carlos

 

 

This is how I set up my extensions.conf to dial two channels (my sip
extension and my cellphone) when asterisk receives an incomming call.  

 

EXTENSIONS.CONF:

 

[incoming]

exten => 3058472194,1,Dial(SIP/1035&SIP/17864883123@richmedium,50)

exten => 3058472194,2,Wait(2)

exten => 3058472194,3,voicemail(u1000)

exten => 3058472194,103,voicemail(b1000)

 

 

 

 

CONSOLE OUTPUT FOR THE INCOMING CALL:

 

asterisk*CLI>

    -- Executing Dial("SIP/3058472194-ff33",
"SIP/1035&SIP/17864883123@broadvoice|50") in new stack

    -- Called 1035

    -- Called 17864883123@richmedium

    -- SIP/1035-21d1 is ringing

    -- SIP/richmedium-625f is ringing

    -- SIP/richmedium-625f is making progress passing it to
SIP/3058472194-ff33             <-------- (Ringtones stop at this point)

    -- SIP/richmedium-625f answered SIP/3058472194-ff33             

    -- Attempting native bridge of SIP/3058472194-ff33 and
SIP/richmedium-625f

  == Spawn extension (internal, 3058472194, 1) exited non-zero on
'SIP/3058472194-ff33'

    -- Executing Hangup("SIP/3058472194-ff33", "") in new
stack

  == Spawn extension (internal, h, 1) exited non-zero on
'SIP/3058472194-ff33'

 

 

SIP.CONF:

 

register =>
3058472194@sip.broadvoice.com:shhhhhh:3058472194@sip.broadvoice.com

 

[3058472194bv]

type=peer

user=phone

context=incoming

host=sip.broadvoice.com

fromdomain=sip.broadvoice.com

fromuser=3058472194

secret=shhhh

username=3058472194

insecure=very

authname=3058472194

nat=no

dtmfmode=rfc2833

 

[richmedium]

type=friend

username=car3423

secret=shhhh

host=64.135.90.5

dtmfmode=rfc2833

disallow=all

;allow=g729

;allow=g726

allow=ulaw

;allow=ilbc

;allow=gsm

context=disconnected

insecure=very

 

 

 

MY SYSTEM:

 

[root@asterisk ~]# uname -a

Linux 2.6.9-22.0.2.EL #1 Tue Jan 17 06:51:40 CST 2006 i686 i686 i386
GNU/Linux

 

 

 

 

ASTERISK VERSION:

 

Asterisk 1.2.4

 

 

 

CONSOLE DEBUG OUTPUT:

 

Too big for this posting, had to remove it, but can paste on a followup.

 

 

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Try adding 'r' to the dial options. According to "show application
dial":

    r    - Indicate ringing to the calling party. Pass no audio to the calling
           party until the called channel has answered.


exten => 3058472194,1,Dial(SIP/1035&SIP/17864883123@richmedium,50, r)

Julian.

On 4/1/06, Carlos A. Alfaro <carlos@brightspeak.com>
wrote:
>
>
>
> Hello Everyone.  I usually find my own solutions for problems but this
time,
> after several months, I've given up.
>
>
>
> My asterisk is set up so that incoming calls from my voip provider ring on
> both my sip extension and my cellphone at the same time.  When the system
> receives an incoming call, ringtones indicating that the call is being
> connected play normally for the first 5 seconds to the caller, but they
> suddenly stop as the call to my cellphone starts to make progress.  This
> causes some people to hang up, despite the fact that the call is still
being
> connected.  Callers who stay on the line are able to talk to me on either
> the sip extension or the cellphone once I pick up either one.
>
>
>
> I have tried a lot of workarounds like including a priority to answer the
> incoming call, invoke the playtones command before the dial command, but
> this doesn't seem to work either.  Can anyone replicate the problem? 
Have I
> ran into a bug?  I have pasted as much info as I deemed relevant; please
let
> me know if I'm missing something.  Thanks.