[Asterisk-Users] RE: Media Negotiation Failed
Sebastian Nocetti
sebastian at interband.com.ar
Wed Nov 12 10:55:02 MST 2003
Codecs are g711ulaw, on both Cisco5300... Dial Peer config is showed
below
Los codecs que uso son G711ulaq, en los dos Cisco5300, te muestro los
dialpeers...
GW that not work - GW que no funciona
translation-rule 1017
Rule 0 8002666333 1000
dial-peer voice 1016 voip
destination-pattern 8002666333
translate-outgoing called 1017
session protocol sipv2
session target ipv4:64.76.xx.xx ---> IP DE ASTERISK.
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
GW that work - GW que funciona
translation-rule 7
Rule 0 ^3104 1000
Rule 1 ^3105 1000
dial-peer voice 7 voip
destination-pattern 310[4-5]
translate-outgoing called 7
session protocol sipv2
session target ipv4:64.76.xx.xx ----> IP DE ASTERISK.
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
-----Mensaje original-----
De: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] En nombre de
asterisk-users-request at lists.digium.com
Enviado el: Miércoles, 12 de Noviembre de 2003 02:12 p.m.
Para: asterisk-users at lists.digium.com
Asunto: Asterisk-Users digest, Vol 1 #1869 - 11 msgs
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Today's Topics:
1. Re: DIAX 0.93 with some sound improvements and not only... (Ariel
Batista)
2. Re: OT : For the SQL gurus.. (Roy Sigurd Karlsbakk)
3. Re: Media Negotiation Failed (CW_ASN - Gus)
4. Re: DIAX 0.93 with some sound improvements and not only... (Dan)
5. Re: OT : For the SQL gurus.. (Tilghman Lesher)
6. Re: DIAX 0.93 with some sound improvements and not only...
(reseaux)
7. Re: OT : For the SQL gurus.. (WipeOut)
8. Re: OT : For the SQL gurus.. (WipeOut)
9. TAPI development (Michael Devenijn)
10. Re: OT : For the SQL gurus.. (Ernest W. Lessenger)
11. Dial Plan Sequencing (Stephen R. Besch)
--__--__--
Message: 1
Date: Wed, 12 Nov 2003 10:50:05 -0500
From: "Ariel Batista" <abatista at avionica.com>
To: <asterisk-users at lists.digium.com>
Subject: Re: [Asterisk-Users] DIAX 0.93 with some sound improvements and
not only...
Reply-To: asterisk-users at lists.digium.com
---------- Original Message ----------------------------------
From: "Dan" <dtoma at fx.ro>
>Hi all,
>
>DIAX 0.9.3 is available for download from the same place:
>http://www.laser.com/dante or
>http://www.geocities.com/tdanro
Thank you for the update! I have the following problems with it! When
exiting the program we get a General Protech error. Also when calling
Zap ports it keeps ringing. From DIAX to Sip it works fine! It
actually sound better then before! But I can not call it from SIP get
Audio missmatch. I can call it from normal Zap ports!
Hope this helps! Keep up the work!
--__--__--
Message: 2
Date: Wed, 12 Nov 2003 17:01:10 +0100 (CET)
From: Roy Sigurd Karlsbakk <roy at karlsbakk.net>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] OT : For the SQL gurus..
Reply-To: asterisk-users at lists.digium.com
> >Thanks everyone for your help on this..
> >
> >For those who are interested I have done some speed tests on these
> >two queries (below) on my server and the results are..
> >
> >Test script of 1000 quieries..
> >Query1 ("code" field not indexed) = 47.183s
> >Query1 ("code" field indexed) = 45.731s
> >Query2 ("code" field not indexed) = 109.321s
> >Query2 ("code" field indexed) = 2.302s
Tried fulltext indexing?
--__--__--
Message: 3
From: "CW_ASN - Gus" <cw_asn at fibertel.com.ar>
To: <asterisk-users at lists.digium.com>
Subject: Re: [Asterisk-Users] Media Negotiation Failed
Date: Wed, 12 Nov 2003 13:01:29 -0300
Reply-To: asterisk-users at lists.digium.com
This is a multi-part message in MIME format.
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charset="iso-8859-1"
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MensajeFijate en los 'voice codecs' de los dial-peers.
----- Original Message -----=20
From: Sebastian Nocetti=20
To: asterisk-users at lists.digium.com=20
Sent: Wednesday, November 12, 2003 12:41 PM
Subject: [Asterisk-Users] Media Negotiation Failed
Hi, I have this scenario
Cisco 5300 (public ip. 200.47.xx.xx) <---> Asterisk (public ip: =
64.76.xx.xx) <--> Cisco 3600 (public ip: 64.76.xx.xx , same network than
=
* )
When a calls comes in Cisco 5300, this send this calls with SIP to *,
= asterisk plays a welcome message and resend call to Cisco 3600 that
have = 4 analog lines connected... but after cisco play welcome message
and = when send SIP to 3600, I have this error:
v=3D0
o=3Droot 20045 20045 IN IP4 64.76.xx.xx -> asterisk ip address
s=3Dsession
c=3DIN IP4 64.76.xx.xx -> asterisk ip address.
t=3D0 0
m=3Daudio 15372 RTP/AVP 0 101
a=3Drtpmap:0 PCMU/8000
a=3Drtpmap:101 telephone-event/8000
a=3Dfmtp:101 0-16
(no NAT) to 64.76.xx.xx:5060 -> 3600 ip address
Sip read: LI>
SIP/2.0 400 Bad Request - 'Media Negotiation Failed'
Via: SIP/2.0/UDP 64.76.xx.xx:5060;branch=3Dz9hG4bK31ba01da -> asterisk
= ip address
From: "1143724956" <sip:1143724956 at 64.76.xx.xx>;tag=3Das33c45436 -> *
= ip address
To: <sip:1152672000 at 64.76.xx.xx> -> 3600 ip address
Call-ID: 28b30df021508ba32b21208459e10765 at 64.76.126.30
Warning: 304 64.76.xx.xx:0 "Media Type(s) Unavailable" -> 3600 ip =
address
CSeq: 102 INVITE
then I have too another GW 5300, with same IOS and same config.. and =
with it, all work OK!!!... I don't understand what is the problem!!...
IT WORKS OK!!!..
Cisco 5300 (public ip. 64.76.xx.xx) <---> Asterisk (public ip: =
64.76.xx.xx) <--> Cisco 3600 (public ip: 64.76.xx.xx , same network than
=
* )
Some clue?....
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charset="iso-8859-1"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>Mensaje</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1"> <META content=3D"MSHTML 6.00.2734.1600"
name=3DGENERATOR> <STYLE></STYLE> </HEAD> <BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Fijate en los 'voice codecs' de los=20
dial-peers.</FONT></DIV> <BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style=3D"FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV=20
style=3D"BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: =
black"><B>From:</B>=20
<A title=3Dsebastian at interband.com.ar=20
href=3D"mailto:sebastian at interband.com.ar">Sebastian Nocetti</A> =
</DIV>
<DIV style=3D"FONT: 10pt arial"><B>To:</B> <A=20
title=3Dasterisk-users at lists.digium.com=20
=
href=3D"mailto:asterisk-users at lists.digium.com">asterisk-users at lists.dig
i=
um.com</A>=20
</DIV>
<DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Wednesday, November 12, =
2003 12:41=20
PM</DIV>
<DIV style=3D"FONT: 10pt arial"><B>Subject:</B> [Asterisk-Users] Media
=
Negotiation Failed</DIV>
<DIV><BR></DIV>
<DIV><SPAN class=3D725243315-12112003><FONT face=3DArial size=3D2>Hi,
= I have this=20
scenario</FONT></SPAN></DIV>
<DIV><SPAN class=3D725243315-12112003><FONT face=3DArial=20
size=3D2></FONT></SPAN> </DIV>
<DIV><SPAN class=3D725243315-12112003><FONT face=3DArial =
size=3D2>Cisco 5300 (public=20
ip. 200.47.xx.xx) <---> Asterisk (public ip: 64.76.xx.xx) =
<-->=20
Cisco 3600 (public ip: 64.76.xx.xx , same network than * =
)</FONT></SPAN></DIV>
<DIV><SPAN class=3D725243315-12112003><FONT face=3DArial=20
size=3D2></FONT></SPAN> </DIV>
<DIV><SPAN class=3D725243315-12112003><FONT face=3DArial size=3D2>When
= a calls comes=20
in Cisco 5300, this send this calls with SIP to *, asterisk plays a =
welcome=20
message and resend call to Cisco 3600 that have 4 analog lines =
connected...=20
but after cisco play welcome message and when send SIP to 3600, I
= have=20
this error:</FONT></SPAN></DIV>
<DIV><SPAN class=3D725243315-12112003><FONT face=3DArial=20
size=3D2></FONT></SPAN> </DIV>
<DIV><SPAN class=3D725243315-12112003><FONT face=3DArial =
size=3D2>v=3D0<BR>o=3Droot=20
20045 20045 IN IP4 64.76.xx.xx -> asterisk ip =
address<BR>s=3Dsession<BR>c=3DIN=20
IP4 64.76.xx.xx -> asterisk ip address.<BR>t=3D0 0<BR>m=3Daudio =
15372 RTP/AVP 0=20
101<BR>a=3Drtpmap:0 PCMU/8000<BR>a=3Drtpmap:101 =
telephone-event/8000<BR>a=3Dfmtp:101=20
0-16<BR> (no NAT) to 64.76.xx.xx:5060 -> 3600 ip =
address<BR>Sip read:=20
LI><BR>SIP/2.0 400 Bad Request - 'Media Negotiation Failed'<BR>Via:
=
SIP/2.0/UDP 64.76.xx.xx:5060;branch=3Dz9hG4bK31ba01da -> asterisk =
ip=20
address<BR>From: "1143724956"=20
<sip:1143724956 at 64.76.xx.xx>;tag=3Das33c45436 -> * ip =
address<BR>To:=20
<sip:1152672000 at 64.76.xx.xx> -> 3600 ip =
address<BR>Call-ID: <A=20
=
href=3D"mailto:28b30df021508ba32b21208459e10765 at 64.76.126.30">28b30df021
5=
08ba32b21208459e10765 at 64.76.126.30</A><BR>Warning:=20
304 64.76.xx.xx:0 "Media Type(s) Unavailable" -> 3600 ip =
address<BR>CSeq:=20
102 INVITE</FONT></SPAN></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><SPAN class=3D725243315-12112003><FONT face=3DArial size=3D2>then
= I have too=20
another GW 5300, with same IOS and same config.. and with it, =
all work=20
OK!!!... I don't understand what is the =
problem!!...</FONT></SPAN></DIV>
<DIV><SPAN class=3D725243315-12112003><FONT face=3DArial=20
size=3D2></FONT></SPAN> </DIV>
<DIV><SPAN class=3D725243315-12112003>
<DIV><SPAN class=3D725243315-12112003><FONT face=3DArial=20
size=3D2></FONT></SPAN> </DIV>
<DIV><SPAN class=3D725243315-12112003><FONT face=3DArial size=3D2>IT =
WORKS=20
OK!!!..</FONT></SPAN></DIV>
<DIV><SPAN class=3D725243315-12112003><FONT face=3DArial=20
size=3D2></FONT></SPAN> </DIV>
<DIV><SPAN class=3D725243315-12112003><FONT face=3DArial =
size=3D2>Cisco 5300 (public=20
ip. 64.76.xx.xx) <---> Asterisk (public ip: 64.76.xx.xx) =
<-->=20
Cisco 3600 (public ip: 64.76.xx.xx , same network than * =
)</FONT></SPAN></DIV>
<DIV><SPAN class=3D725243315-12112003><FONT face=3DArial=20
size=3D2></FONT></SPAN> </DIV>
<DIV><SPAN class=3D725243315-12112003><FONT face=3DArial=20
size=3D2></FONT></SPAN> </DIV>
<DIV><SPAN class=3D725243315-12112003><FONT face=3DArial size=3D2>Some
=
clue?....</FONT></SPAN></DIV></SPAN></DIV></BLOCKQUOTE></BODY></HTML>
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--__--__--
Message: 4
From: "Dan" <dtoma at fx.ro>
To: <asterisk-users at lists.digium.com>
Subject: Re: [Asterisk-Users] DIAX 0.93 with some sound improvements and
not only...
Date: Wed, 12 Nov 2003 18:06:38 +0200
Organization: Personal account
Reply-To: asterisk-users at lists.digium.com
Hi,
----- Original Message -----
From: "Ariel Batista" <abatista at avionica.com>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, November 12, 2003 5:50 PM
Subject: Re: [Asterisk-Users] DIAX 0.93 with some sound improvements and
not only...
.....
> Thank you for the update! I have the following problems with it! When
exiting the program we get a General Protech error.
This is a known bug (see the help file).... Hope to be solved when the
IAX2 version will be available
> Also when calling Zap ports it keeps ringing.
Try to put a line in extensions.conf before the dial one
xxx,1,Answer
xxx,2,Dial(....
> It actually sound better then before!
The noise (the microphone one especially when used on a notebook) must
be drastically reduced now.
> But I can not call it from SIP get Audio missmatch.
What type of SIP phone?... I have test it with CIsco 7960 and it works
as expected.. Where did you gtet this message (on SIP phone or on DIAX)?
Best regards,
Dan
--__--__--
Message: 5
From: Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] OT : For the SQL gurus..
Date: Wed, 12 Nov 2003 10:19:34 -0600
Reply-To: asterisk-users at lists.digium.com
On Wednesday 12 November 2003 10:01, Roy Sigurd Karlsbakk wrote:
> > >Thanks everyone for your help on this..
> > >
> > >For those who are interested I have done some speed tests on these
> > >two queries (below) on my server and the results are..
> > >
> > >Test script of 1000 quieries..
> > >Query1 ("code" field not indexed) = 47.183s
> > >Query1 ("code" field indexed) = 45.731s
> > >Query2 ("code" field not indexed) = 109.321s
> > >Query2 ("code" field indexed) = 2.302s
>
> Tried fulltext indexing?
Fulltext indexing won't get you anything, considering that these queries
aren't searching for non-0-based-offsets in substrings.
-Tilghman
--__--__--
Message: 6
From: reseaux <reseauxit at yahoo.it>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] DIAX 0.93 with some sound improvements and
not only...
Date: Wed, 12 Nov 2003 17:27:28 +0000
Reply-To: asterisk-users at lists.digium.com
Hi Gavin
i have the same error when i try to run DIAX with Wine.
thanks
Dimitri
On Wednesday 12 November 2003 15:23, Gavin Hamill wrote:
> On Wed, 2003-11-12 at 15:07, Dan wrote:
> > DIAX 0.9.3 is available for download from the same place:
>
> Hi Dan :)
>
> Do you know if anyone has successfully run DIAX on Linux with Wine?
>
> After installing the VB6 runtime DLL, I ran diax.exe and got
>
> fixme:ole:CoRegisterMessageFilter stub
> fixme:ole:OLEPictureImpl_Construct Unsupported type 3
> fixme:ole:OLEPictureImpl_SaveAsFile (0x404068d0)->(0x40406bc8, 0,
> (nil)), hacked stub. fixme:ole:VarParseNumFromStr
> (L"2",flags=80000000,....), partial stub! fixme:ole:VarParseNumFromStr
> numparse: cDig=30, InFlags=954 fixme:ole:VarNumFromParseNum
> (..,dwVtBits=20,....), partial stub! fixme:ole:VarParseNumFromStr
> (L"-99",flags=80000000,....), partial stub!
> fixme:ole:VarParseNumFromStr numparse: cDig=30, InFlags=954
> fixme:ole:VarNumFromParseNum (..,dwVtBits=20,....), partial stub!
> fixme:ole:OLEPictureImpl_FindConnectionPoint tried to find connection
> point on {33ad4ed2-6699-11cf-b70c-00aa0060d393}?
>
> and then a 'Runtime Error '6': Overflow' dialog with 'OK' ..
>
> I don't know if any of these messages are even remotely useful, but
> I've included them for completeness :)
>
> Cheers,
> Gavin.
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
--__--__--
Message: 7
Date: Wed, 12 Nov 2003 16:27:30 +0000
From: WipeOut <wipe_out at onetel.com>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] OT : For the SQL gurus..
Reply-To: asterisk-users at lists.digium.com
Andy Powell wrote:
>>Thanks everyone for your help on this..
>>
>>For those who are interested I have done some speed tests on these two
>>queries (below) on my server and the results are..
>>
>>Test script of 1000 quieries..
>>Query1 ("code" field not indexed) = 47.183s
>>Query1 ("code" field indexed) = 45.731s
>>Query2 ("code" field not indexed) = 109.321s
>>Query2 ("code" field indexed) = 2.302s
>>
>>
>>
>
>OUCH! those times are loooooooooooong!
>
>Andy
>
>
>_
>
I agree the first three are long, but the last one works out to just
over 26000 queries per min.. I didn't think that was bad for a PII 350..
:)
Later..
--__--__--
Message: 8
Date: Wed, 12 Nov 2003 16:29:16 +0000
From: WipeOut <wipe_out at onetel.com>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] OT : For the SQL gurus..
Reply-To: asterisk-users at lists.digium.com
Roy Sigurd Karlsbakk wrote:
>>>Thanks everyone for your help on this..
>>>
>>>For those who are interested I have done some speed tests on these
>>>two queries (below) on my server and the results are..
>>>
>>>Test script of 1000 quieries..
>>>Query1 ("code" field not indexed) = 47.183s
>>>Query1 ("code" field indexed) = 45.731s
>>>Query2 ("code" field not indexed) = 109.321s
>>>Query2 ("code" field indexed) = 2.302s
>>>
>>>
>
>Tried fulltext indexing?
>
>
>
Due to the nature of the search I don't think it would have benefitted
from fulltext indexing..
Later..
--__--__--
Message: 9
Date: Wed, 12 Nov 2003 16:36:09 +0100
From: "Michael Devenijn" <michael.devenijn at dkma.be>
To: <asterisk-users at lists.digium.com>
Subject: [Asterisk-Users] TAPI development
Reply-To: asterisk-users at lists.digium.com
This is a multi-part message in MIME format.
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Has anyone ever worked opn TAPI stuff to make asterisk work with it ?
=20 I'm a Windoze C++ developer dig'n into asterisk (and linux at the
same =
time) since a few months and i'm quite interested in creating a TAPI =
driver for asterisk.=20 =20 so if anybody did any research in that way
please inform me. =20 Also i've you think it's quite impossible to do it
we can discuss our = idea's =20 =20 Michael Devenijn=20 DKMA bvba =20
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charset="iso-8859-1"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"><HTML =
DIR=3Dltr><HEAD><META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html;
= charset=3Diso-8859-1"></HEAD><BODY><DIV><FONT face=3DArial =
color=3D#000000 size=3D2>Has anyone ever =0A= worked opn TAPI stuff to
make asterisk work with it ?</FONT></DIV>=0A= <DIV><FONT face=3DArial
size=3D2></FONT> </DIV>=0A= <DIV><FONT face=3DArial size=3D2>I'm a
Windoze C++ developer dig'n into = asterisk =0A= (and linux at the same
time) since a few months and i'm quite = interested in =0A=
creating a TAPI driver for asterisk. </FONT></DIV>=0A= <DIV><FONT
face=3DArial size=3D2></FONT> </DIV>=0A= <DIV><FONT face=3DArial
size=3D2>so if anybody did any research in that = way please =0A= inform
me.</FONT></DIV>=0A= <DIV><FONT face=3DArial
size=3D2></FONT> </DIV>=0A= <DIV><FONT face=3DArial size=3D2>Also
i've you think it's quite = impossible to do it =0A= we can discuss our
idea's</FONT></DIV>=0A= <DIV><FONT face=3DArial
size=3D2></FONT> </DIV>=0A= <DIV><FONT face=3DArial
size=3D2></FONT> </DIV>=0A= <DIV><FONT face=3DArial
size=3D2>Michael Devenijn </FONT></DIV>=0A= <DIV><FONT face=3DArial
size=3D2>DKMA bvba</FONT></DIV>=0A= <DIV><FONT face=3DArial
size=3D2></FONT> </DIV></BODY></HTML>
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--__--__--
Message: 10
Date: Wed, 12 Nov 2003 08:53:01 -0800
To: asterisk-users at lists.digium.com
From: "Ernest W. Lessenger" <ernest at oacys.com>
Subject: Re: [Asterisk-Users] OT : For the SQL gurus..
Reply-To: asterisk-users at lists.digium.com
At 11:07 AM 11/10/2003, you wrote:
>Thanks everyone for your help on this..
>
>For those who are interested I have done some speed tests on these two
>queries (below) on my server and the results are..
>
>Test script of 1000 quieries..
>Query1 ("code" field not indexed) = 47.183s
>Query1 ("code" field indexed) = 45.731s
>Query2 ("code" field not indexed) = 109.321s
>Query2 ("code" field indexed) = 2.302s
>
>Query2 has additional overhead in the script as well because it has to
>itterate through the number and build up the query..
>
>Query1 is far simpler to use in a script becasue the query does not
>have to be built up..
Since you only need to do a simple lookup, why not either (a) build your
own db or (b) use berkely DB or some other fast database engine? Since
all
you really need to do is a prefix search on a key:
struct node {
char num;
struct node* p0;
struct node* p1;
struct node* p2;
struct node* p3;
struct node* p4;
struct node* p5;
struct node* p6;
struct node* p7;
struct node* p8;
struct node* p9;
char* desc;
}
That's 48 bytes per record (not counting the description). Memory usage
will depend on how much data you need to store, but lookups would be
O(k),
where k is the length of the key.
--Ernest
--__--__--
Message: 11
Date: Wed, 12 Nov 2003 12:06:03 -0500
From: "Stephen R. Besch" <sbesch at acsu.buffalo.edu>
To: asterisk users list <asterisk-users at lists.digium.com>
Subject: [Asterisk-Users] Dial Plan Sequencing
Reply-To: asterisk-users at lists.digium.com
I have an interesting dilemma with sequencing in the dialplan. Up to
now, I have assumed that the extensions in the dial plan were tested in
the order that they appear in extensions.conf. In other words, I have
the following fragment which was designed to dial toll free on the PSTN
and all other long distance on VoIP:
[longdistance]
include => local
;Handle local, etc first. (or so
I thought!)
exten => _91NXXNXXXXXX,1,Dial(${VPLSTRUNK}/${EXTEN:1}) ;Dial long
distance through VoiP
exten => _91NXXNXXXXXX,2,Congestion
;OOPS! No lines available?
:
:
[local]
:
exten => _91800NXXXXXX,1,Dial(${PSTNTRUNK}/${EXTEN}) ; Long distance
toll free accessed through PSTN trunk interface
exten => _91800NXXXXXX,2,Congestion
exten => _91888NXXXXXX,1,Dial(${PSTNTRUNK}/${EXTEN})
exten => _91888NXXXXXX,2,Congestion
exten => _91877NXXXXXX,1,Dial(${PSTNTRUNK}/${EXTEN})
exten => _91877NXXXXXX,2,Congestion
exten => _91866NXXXXXX,1,Dial(${PSTNTRUNK}/${EXTEN})
exten => _91866NXXXXXX,2,Congestion
; The rest of the local definitions, etc
:
I expected that the "_918" definitions would be tested first, followed
by the "_91N" definitions. Unfortunately, it appears as if the
definitions made using the "include=" operator are always tested last.
This means that the toll free numbers dialed by people in the
longdistance context are always routed over VoIP rather than PSTN
because they match the "_91N" pattern. While I can fix this with a
complicated set of conditionals or dial string patterns, I wonder if
anyone has found a more elegant solution, remembering that I want to
give some extensions access to only the local context, but still provide
toll free service for everyone (i.e, I don't want to move the "_918"
definitions into the longdistance context).
Stephen R. Besch
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