Hi List;
Can someone advise me why in the below context, it
does not run the Background step? Once I dial 1000,
then it hangup and give congestion signal? If I
comment the ResponseTimeOut, then it run the
Background but it does not wait till caller enter the
digits, once the sound file finish, then it hangup
(congestion signal), also in all the situation, it
does not go for the t extension, why? Is it because I
am originating the call from local extension (an
handset connected to FXS port) and the call should be
originated from FXO or IP Trunk, or what is the
problem exactly?
[Test_Bilal]
include => KuwaitInternal
include => EgyptInternal
exten => 1000,1,Goto(s,1)
exten => s,1,Answer()
exten => s,2,ResponseTimeout(5)
exten => s,3,Background(WelcomeMessage)
exten => 0,1,Dial(SIP/EgyptOperatorSIP,10)
exten => 0,2,Background(WelcomeMessage)
exten => 0,2,Playback(vm-nobodyavail)
exten => 0,3,Hangup()
exten => 0,102,Playback(tt-allbusy)
exten => 0,103,Hangup()
exten => i,1,Playback(pbx-invalid)
exten => i,2,Goto(EgyptIncomingPSTN,s,1)
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup()
Any help??
Regards
Bilal
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bilal ghayyad wrote:> Hi List; > > Can someone advise me why in the below context, it >You never told us what version you are running. If it's version 1.2, make sure you have set priorityjumping=no in your extensions.conf or use the waitexten application. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."
On Sunday 14 October 2007 17:35:04 bilal ghayyad wrote:> Can someone advise me why in the below context, it > does not run the Background step? Once I dial 1000, > then it hangup and give congestion signal? If I > comment the ResponseTimeOut, then it run the > Background but it does not wait till caller enter the > digits, once the sound file finish, then it hangup > (congestion signal), also in all the situation, it > does not go for the t extension, why? Is it because I > am originating the call from local extension (an > handset connected to FXS port) and the call should be > originated from FXO or IP Trunk, or what is the > problem exactly? > > [Test_Bilal] > > include => KuwaitInternal > include => EgyptInternal > exten => 1000,1,Goto(s,1) > exten => s,1,Answer() > exten => s,2,ResponseTimeout(5) > exten => s,3,Background(WelcomeMessage) > exten => 0,1,Dial(SIP/EgyptOperatorSIP,10) > exten => 0,2,Background(WelcomeMessage) > exten => 0,2,Playback(vm-nobodyavail) > exten => 0,3,Hangup() > exten => 0,102,Playback(tt-allbusy) > exten => 0,103,Hangup() > exten => i,1,Playback(pbx-invalid) > exten => i,2,Goto(EgyptIncomingPSTN,s,1) > exten => t,1,Playback(vm-goodbye) > exten => t,2,Hangup()Go read the top of configs/extensions.conf.sample, specifically the part about the "autofallthrough" parameter. -- Tilghman
bilal ghayyad wrote:> Can someone advise me why in the below context, it > does not run the Background step?> [Test_Bilal] > > include => KuwaitInternal > include => EgyptInternal > exten => 1000,1,Goto(s,1) > exten => s,1,Answer() > exten => s,2,ResponseTimeout(5) > exten => s,3,Background(WelcomeMessage) > exten => 0,1,Dial(SIP/EgyptOperatorSIP,10) > exten => 0,2,Background(WelcomeMessage) > exten => 0,2,Playback(vm-nobodyavail) > exten => 0,3,Hangup()1, 2, 2, 3. - That is not supposed to work.> exten => 0,102,Playback(tt-allbusy) > exten => 0,103,Hangup()Don't use priority jumping. There are many examples how to do it better. btw: Kuwait, Egypt - Are you going to become a VoIP provider ;) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Gesch?ftsf?hrer: Stefan Wintermeyer Handelsregister: Neuwied B 14998
bilal ghayyad wrote:> Dear Phellepe;? It's a bit uncommon to change other people's names.> It was 1.4 and I set priorityjumping and set > autofallthrough and look like fine, need to test more.Ok. So you seem to have made your decision. Although I don't understand why there's no need to do testing as you said your dialplan doesn't work.> But there are two issues related to your reply: > > 1) Why I do not have to use priority jumpingYou are free to use it, but it is deprecated, meaning future versions of Asterisk will not support it. Apart from that it looks ugly and makes your dialplan hard to maintain.> and what > is the alterantive better solution (where I can see > such these examples)?[macro-stdexten] in extensions.conf for example.> But by the way, I need a service provider, so what do > you know about dundi and how I can send calls for > them? Are they really a group that have routes?DUNDi is a protocol http://en.wikipedia.org/wiki/DUNDI http://www.voip-info.org/wiki-DUNDi Start a new thread please and ask the list, not me. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Gesch?ftsf?hrer: Stefan Wintermeyer Handelsregister: Neuwied B 14998