Carlos A. Alfaro
2006-Apr-01 18:46 UTC
[Asterisk-Users] Problem: ringtones stop unexpectedly
I should've mentioned that before. I've tried doing that and it has no
effect. I've tried both upper and lower-case 'r's.
I've also tried a workaround that I thought would work, but it doesn't:
Answering the call and then using the playtones(ringing) command before
connecting to my cellphone.
-----Original Message-----
Date: Sat, 1 Apr 2006 19:59:46 +0100
From: "Julian J. M." <julianjm@gmail.com>
Subject: Re: [Asterisk-Users] Problem: ringtones stop unexpectedly
when multiple channels are dialed
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID:
<a60dba290604011059k242fd5c6lf9398b4228c624e3@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
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.
------------------------------
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
End of Asterisk-Users Digest, Vol 21, Issue 2
*********************************************
Have you tryed phoning a fixed line instead of a cell phone? is this giving the same result? I assume your outgoing call to a the cellphone goes through a Zap channel. Try another one (e.g. Zap channel 2), and let us know the result. Alyed ---------------------------------------- Return-Path: <asterisk-users-bounces@lists.digium.com> Sat Apr 01 18:47:36 2006 Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP; Sat, 1 Apr 2006 18:47:36 -0700 I should've mentioned that before. I've tried doing that and it has no effect. I've tried both upper and lower-case 'r's. I've also tried a workaround that I thought would work, but it doesn't: Answering the call and then using the playtones(ringing) command before connecting to my cellphone. -----Original Message----- Date: Sat, 1 Apr 2006 19:59:46 +0100 From: "Julian J. M." Subject: Re: [Asterisk-Users] Problem: ringtones stop unexpectedly when multiple channels are dialed To: "Asterisk Users Mailing List - Non-Commercial Discussion" Message-ID: Content-Type: text/plain; charset=ISO-8859-1 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 wrote:> > > > Hello Everyone. I usually find my own solutions for problems but thistime,> 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 stillbeing> 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? HaveI> ran into a bug? I have pasted as much info as I deemed relevant; pleaselet> me know if I'm missing something. Thanks.------------------------------ _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest, Vol 21, Issue 2 ********************************************* _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060401/5ecc89a0/attachment.htm
Carlos A. Alfaro
2006-Apr-03 00:46 UTC
[Asterisk-Users] Problem: ringtones stop unexpectedly
Actually the outgoing call is going out through a sip channel, and perhaps I
should say the two calls. I am making two sip calls with one dial command
in the second priority:
[incoming_sip_calls_from_pstn]
exten => 3058472194,1,Dial(SIP/1035,10,r);## To ring on the sip extension
for 10 seconds
exten => 3058472194,2,Dial(SIP/1035&SIP/17864883123@richmedium,50,r);##
To
call sip extension + cellphone
Tried calling land lines as well, but still only one or two ringtones are
played.
I tried boiling down the problem and realized that it only happens when the
Dial command is used a second time. If I ring on two sip channels in
priority 1:
exten => 3058472194,1,Dial(SIP/1035&SIP/17864883123@richmedium,50,r);##
To
call sip extension + cellphone
the 'ringing' tones are played to the calling party for as long as the
call
is not answered, provided I use the r option.
When I try to dial an outgoing number for the first 10 seconds in priority
1, and dial another number in priority 2, playtones stop after the second
number is dialed and the caller will not hear anything from that point on,
until he hangs up or the call is answered.
I feel like I made some progress just by simplifying the problem, but I can
only guess this is a bug, what do you think?
-----Original Message-----
Date: Sat, 1 Apr 2006 21:23:06 -0700
From: Alyed Tzompa <alyed.tzompa@simitel.com>
Subject: Re: [Asterisk-Users] Problem: ringtones stop unexpectedly
To: <asterisk-users@lists.digium.com>,<julianjm@gmail.com>
Message-ID: <e43e3a016dbb402ba9ce4513b9a67588@simitel.com>
Content-Type: text/plain; charset="iso-8859-1"
Have you tryed phoning a fixed line instead of a cell phone?
is this giving the same result?
I assume your outgoing call to a the cellphone goes through a Zap channel.
Try another one (e.g. Zap channel 2), and let us know the result.
Alyed
----------------------------------------
Return-Path: <asterisk-users-bounces@lists.digium.com> Sat Apr 01 18:47:36
2006
Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by
mail11.webcontrolcenter.com with SMTP;
Sat, 1 Apr 2006 18:47:36 -0700
I should've mentioned that before. I've tried doing that and it has no
effect. I've tried both upper and lower-case 'r's.
I've also tried a workaround that I thought would work, but it doesn't:
Answering the call and then using the playtones(ringing) command before
connecting to my cellphone.
-----Original Message-----
Date: Sat, 1 Apr 2006 19:59:46 +0100
From: "Julian J. M."
Subject: Re: [Asterisk-Users] Problem: ringtones stop unexpectedly
when multiple channels are dialed
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Message-ID:
Content-Type: text/plain; charset=ISO-8859-1
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 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.
------------------------------
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
End of Asterisk-Users Digest, Vol 21, Issue 2
*********************************************
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20060401/5ecc89
a0/attachment-0001.htm
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20060403/3e2d3a88/attachment.htm