[asterisk-users] help - UNSUBSCRIBE
Mike Hammett
asterisk-users at ics-il.net
Thu Mar 29 10:16:35 MST 2007
Good job on reading the line at the top of the digest on how to unsubscribe.
--Mike
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
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Sent: Thursday, March 29, 2007 11:44 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] help - UNSUBSCRIBE
Please remove this email from your mailing list.
UNSUBSCRIBE
Thank you.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
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Sent: Thursday, March 29, 2007 9:14 AM
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Subject: asterisk-users Digest, Vol 32, Issue 118
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Today's Topics:
1. Re: Re: Re: Inbound Voice Quality - Speed Change (Tzafrir Cohen)
2. Re: error in FreePBX (Steve Murphy)
3. SV: [asterisk-users] Set(CALLERID(all) not working with
'unknown'call? (jan.sarin at securia.se)
4. Re: Transfering not working - how to debug? (Rizwan Hisham)
5. Off Topic: Open Source USB Softphone (Luis Claudio Santos)
6. Where are Spandsp changelogs or bugs available ? (Olivier)
7. L options in Dial() dont seem to work.... (Mark Reardon)
8. maximum simultaneous calls (Mark Quitoriano)
9. Re: L options in Dial() dont seem to work....
(Eric "ManxPower" Wieling)
10. Asterisk does not reINVITE after 302Redirect &
401Unauthorized (Mushtaq_Ahmed at 3com.com)
11. Re: L options in Dial() dont seem to work.... (Steve Murphy)
12. Is it possible to install CCM on a Linux platform ? (Olivier)
13. Re: L options in Dial() dont seem to work.... (Mark Reardon)
14. Scratchy Audio with Asterisk 1.2.4 over IAX on FreeBSD?
(Benoit Panizzon)
15. Re: Cisco 30VIP Phone (Jason Parker)
16. SIP & NAT (Mike Hammett)
17. Re: maximum simultaneous calls (Matthew J. Roth)
18. RE: SIP & NAT (Alexander Lopez)
19. Re: Multi-line phones - Asterisk uses wrong callerid (Drew Gibson)
----------------------------------------------------------------------
Message: 1
Date: Thu, 29 Mar 2007 15:40:20 +0200
From: Tzafrir Cohen <tzafrir.cohen at xorcom.com>
Subject: Re: [asterisk-users] Re: Re: Inbound Voice Quality - Speed
Change
To: asterisk-users at lists.digium.com
Message-ID: <20070329134020.GC2726 at xorcom.com>
Content-Type: text/plain; charset=us-ascii
On Thu, Mar 29, 2007 at 08:28:53AM -0400, Jim Duda wrote:
> The zttest program results in > 99%.
So you have a working timing source. No need to waste your time here.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir at jabber.org
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
------------------------------
Message: 2
Date: Thu, 29 Mar 2007 07:59:34 -0600
From: Steve Murphy <murf at digium.com>
Subject: Re: [asterisk-users] error in FreePBX
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <1175176774.31166.22.camel at digium2>
Content-Type: text/plain; charset="utf-8"
On Thu, 2007-03-29 at 13:26 +0200, Remco Barendse wrote:
> On Thu, 29 Mar 2007, Carlos JerC3nimo wrote:
>
> > Ive installed asterisk and freepbx. Through the interface ive
> > configured 2 extensions, 6000 and 6001.
> > My problem is that when i try to call from extension 6000 to 6001, i
> > hear this msg "Im-sorry&an-error-has-occured" and the call is
> > terminated.
> > As expected if i call to another number i get an error.
> > i thought the problem might been related with the NAT but if checked
> > and changed some NAT configuration parameters, it didnt worked aswell.
> > As this ever happened to anyone before? Any hints are very appreciated.
> >
> > Thank you very much
>
> I have the same problem, it seems to occur when an extension is busy here.
>
> All my extensions are on local lan with phones having ip addresses in a
> private range without NAT or anything so that is not the problem.
>
> Sounds like an error in the dial pan FreePBX generated.
My suggestion: try a FreePBX mailing list first; the problem *is* more
likely to be in their stuff.
murf
--
Steve Murphy
Software Developer
Digium
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Message: 3
Date: Thu, 29 Mar 2007 16:04:43 +0200
From: <jan.sarin at securia.se>
Subject: SV: [asterisk-users] Set(CALLERID(all) not working with
'unknown'call?
To: <asterisk-users at lists.digium.com>
Message-ID:
<0FF4F1903968F943B5EA2521CD5296C16EE1C9 at exchange.securia.local>
Content-Type: text/plain; charset="iso-8859-1"
Hi Chris,
Yes the call was from PSTN and your solution worked great! I've read about
SetCallerPres earlier but I didn't connect the dots this time.
Thanks alot! :)
Regards,
Jan
-----Ursprungligt meddelande-----
Fren: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] Fvr Christoph F|rstaller
Skickat: den 29 mars 2007 15:29
Till: Asterisk Users Mailing List - Non-Commercial Discussion
Dmne: Re: [asterisk-users] Set(CALLERID(all) not working with 'unknown'call?
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi Jan,
Is this call from PSTN? Probably the Nr is prohibited in PSTN, then asterisk
doesn't set the CALLERID. Try this:
exten => _3072,1,Answer
exten => _3072,n,SetCallerPres(allowed)
exten => _3072,n,Set(CALLERID(all)=DIRECT <0850553072>)
Look here:
http://www.voip-info.org/wiki-Asterisk%20cmd%20SetCallerPres
chris...
jan.sarin at securia.se schrieb:
> Hi,
>
> This is really strange (but probably simple solution).
>
> The CALLERID(all) setting doesn't seem to work when the incomming
> callerid is 'unknown'.
>
> Dialplan looks like this:
> exten => _3072,1,Answer
> exten => _3072,n,Set(CALLERID(all)=DIRECT <0850553072>) exten =>
> _3072,n,Dial(SIP/2001&SIP/2002&SIP/2003&SIP/2004&SIP/2201&SIP/2202&SIP
> /2
> 203&SIP/2205,30,r)
> exten => _3072,n,Wait(1)
> exten => _3072,n,Goto(custom-incoming-3070,1,1)
> exten => _3072,n,Hangup()
>
> Now, it works if the incomming caller id is NOT 'unknown'. Does anyone
> understand why? We're running Asterisk 1.2.7.
>
> Thanks!
>
> Regards,
> Jan
> _______________________________________________
> --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
- --
Dipl.-Ing. Kurt Krenn - IT-Beratung
Franz-Josef-Strasse 33/4/43, 5020 Salzburg
Tel: +43 662 879512 Fax: +43 662 875960
IP-Tel: +43 780 kkrenn (557366)
Email: c.fuerstaller at kurtkrenn.com
sip: c.fuerstaller at kurtkrenn.com
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_______________________________________________
--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
------------------------------
Message: 4
Date: Thu, 29 Mar 2007 19:27:45 +0500
From: "Rizwan Hisham" <rizwanhasham at gmail.com>
Subject: Re: [asterisk-users] Transfering not working - how to debug?
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<4809880c0703290727t6bd4227dmdc05753388b426b0 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Both end devices should be using same codecs. set dtmf = rfc2833 and set
canreinvite = no in sip.conf for both endpoints. This should solve the
problem. you should also check which codecs support rfc2833 for dtmf and
use that codec.
On 3/29/07, Gordon Henderson <gordon+asterisk at drogon.net> wrote:
>
> On Wed, 28 Mar 2007, Alan Chandler wrote:
>
> > I cannot seem to get any transfers to work at all. The console show I
> > have #1 amd #2 set up for Blind and Attended Transfer, but when I hit
> > these buttons on my handset nothing happens (other than I hear the dtmf
> > tones on the other end of the line).
> >
> > roo*CLI> show features
> > Builtin Feature Default Current
> > --------------- ------- -------
> > Pickup *8 *8
> > Blind Transfer # #1
> > Attended Transfer #2
> > One Touch Monitor *1
> > Disconnect Call * *0
> >
> >
> > I am using the tT options in my dial calls (via a macro)
> >
> > [macro-extension]
> > exten => s,1,Dial(${ARG1},20,tT)
>
> I had to fiddle with other things to make this work (needed for the
> Siemens CP4600 SIP/DECT phone)
>
> I found that the default timeouts were a bit tight for my likings (and the
> people who I was testing this with!)
>
> So in features.conf I have:
>
> transferdigittimeout = 8 ; Number of seconds to wait between digits
> when transfering a call
> featuredigittimeout = 999 ; Max time (ms) between digits for
> ; feature activation. Default is 500
>
> [featuremap]
> blindxfer => #1 ; Blind transfer
> atxfer => ## ; Attended transfer
> disconnect => #0 ; Disconnect
>
> If it's still not working, are you sure the DTMF is being picked
> up/transmitted correctly? If it's in-band, is it a codec other than G711?
> (which might give you problems)
>
> Gordon
> _______________________________________________
> --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
>
--
Regards
Rizwan Hisham
Software Engineer
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Message: 5
Date: Thu, 29 Mar 2007 11:33:07 -0300
From: "Luis Claudio Santos" <listas.lcs at gmail.com>
Subject: [asterisk-users] Off Topic: Open Source USB Softphone
To: asterisk-users at lists.digium.com
Message-ID:
<8e4668af0703290733n6c8aeeu408c7f2b19226224 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
I need a softphone - for usb phone devices - that I can alter (insert logo,
menu, etc).
Does somebody know such one?
[]s
--
Abragos
Luis Claudio
Mobile + 55 21 9215 2888
Mobile +55 15 9141 8402
Office +55 15 2102 5859
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Message: 6
Date: Thu, 29 Mar 2007 16:49:42 +0200
From: Olivier <oza-4h07 at myamail.com>
Subject: [asterisk-users] Where are Spandsp changelogs or bugs
available ?
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<442fbb120703290749i197bcdcbhcdb328724fa9bbb8 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi,
Maybe this question has already been answered but I could find its answer
anywhere.
>From here http://www.soft-switch.org/downloads/spandsp/, I can see a new
0.0.3pre28 version of spandsp have been added in march.
When you open this archive file, you can see a ChangeLog file but its
content mentions last change date from may 2006 :
"06.05.23 - 0.0.3 - Steve Underwood <steveu at coppice.org> - T.38 now
implemented, though it needs further polishing. - G.726 and G.722 now
implemented.
04.08.29 - 0.0.2 - Steve Underwood <steveu at coppice.org> - T.4 no longer uses
libtiff for compresion and decompression on the line side (it is still used
to handle the TIFF files). Spandsp no longer depends on accessing the
"internals" of libtiff. New 1D and 2D compression and decompression code now
handles the line side. This should be more robust than using libtiff, and
handles the fudging of bad scan lines rather better. - T.30 line turn-around
timing corrected. "
Where can you find changelogs ?
Is there any public or private bug list somewhere ?
I'm simply not using the right tools ?
Focusing on Debian, I could find this :
http://packages.qa.debian.org/a/asterisk-spandsp-plugins.html
But I can't make any relation between above mentioned package and spandsp
0.0.3preXX versions.
Your comments ?
Thanks in advance
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Message: 7
Date: Thu, 29 Mar 2007 15:51:50 +0100
From: "Mark Reardon" <asterisk.mark at gmail.com>
Subject: [asterisk-users] L options in Dial() dont seem to work....
To: asterisk-users at lists.digium.com
Message-ID:
<6f6ce67a0703290751s5e21dfcfr9b9e5d0b898c965e at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hello Asterisk users,
Can someone thwack me with a clue stick please?
I am following the Asterisk TFOT book Dial() example trying to get the limit
and announcements to work as per below.
These settings seem to have no effect.
There are no warning messages after 4 minutes or every 30 secs thereafter
and the call lasts longer than 5 minutes.
gunner*CLI> show dialplan
[ Context 'outgoing' created by 'pbx_config' ]
'123' => 1. Answer() [pbx_config]
2. AGI(/usr/local/share/examples/asterisk/agi/agi-
test.agi) [pbx_config]
3. Hangup()
[pbx_config]
'_X.' => 1. Dial(SIP/sipprovider/${EXTEN}||L[300000:240000:30000])
[pbx_config]
2. Hangup()
[pbx_config]
I am using 1.2.17.
/usr/local/sbin/asterisk -vvvvvv -g -dddddd -c
Does not show anything to even indicate * is trying anything unusual with
regards to limits or warnings. It just seems to ignore the dialplan options
altogether.
Cheers
Mark
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Message: 8
Date: Thu, 29 Mar 2007 23:05:57 +0800
From: "Mark Quitoriano" <markquitoriano at gmail.com>
Subject: [asterisk-users] maximum simultaneous calls
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<6b542ec90703290805i7abd8156ic38985e27cf2165f at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi,
what could be the maximum simultaneous calls can asterisk do? i read about
the asterisk business edition review[1] and it can only handle 120
simultaneous calls? i'm using 1.2.x branch of asterisk and i use more or
less 90 simultaneous calls.
[1] http://www.voiptalk.org/products/Asterisk+Business+Edition
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Message: 9
Date: Thu, 29 Mar 2007 10:10:33 -0500
From: "Eric \"ManxPower\" Wieling" <eric at fnords.org>
Subject: Re: [asterisk-users] L options in Dial() dont seem to
work....
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <460BD6E9.6080007 at fnords.org>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Mark Reardon wrote:
> Hello Asterisk users,
>
> Can someone thwack me with a clue stick please?
> I am following the Asterisk TFOT book Dial() example trying to get the
> limit
> and announcements to work as per below.
>
> These settings seem to have no effect.
>
> There are no warning messages after 4 minutes or every 30 secs thereafter
> and the call lasts longer than 5 minutes.
>
> gunner*CLI> show dialplan
> [ Context 'outgoing' created by 'pbx_config' ]
> '123' => 1. Answer() [pbx_config]
> 2. AGI(/usr/local/share/examples/asterisk/agi/agi-
> test.agi) [pbx_config]
> 3. Hangup()
> [pbx_config]
> '_X.' => 1. Dial(SIP/sipprovider/${EXTEN}||L[300000:240000:30000])
> [pbx_config]
> 2. Hangup()
> [pbx_config]
>
> I am using 1.2.17.
>
> /usr/local/sbin/asterisk -vvvvvv -g -dddddd -c
>
> Does not show anything to even indicate * is trying anything unusual with
> regards to limits or warnings. It just seems to ignore the dialplan
options
> altogether.
You are using [ ] instead of ( ) after L
------------------------------
Message: 10
Date: Thu, 29 Mar 2007 11:16:34 -0400
From: Mushtaq_Ahmed at 3com.com
Subject: [asterisk-users] Asterisk does not reINVITE after 302Redirect
& 401Unauthorized
To: asterisk-users at lists.digium.com
Message-ID:
<OF85EA8FBC.40575101-ON852572AD.0050A1FF-852572AD.0053EB40 at 3com.com>
Content-Type: text/plain; charset="us-ascii"
Hi,
I'm testing sip trunking on Asterisk (v1.4.0-beta3) with various voip
service providers and stumbled on this
issue. This very well may be a known issue or something misconfigured in
my extensions.conf/sip.conf files.
The service provider requires registration and authentication. The
asterisk is registered for incoming calls which
work fine. Problem is with outbound calls from asterisk which the service
provider authenticates (user/passwd
already configured in config files and tested).
1. Asterisk sends INVITE to primary voip server service provider (SP)
2. SP responds with 302 redirect with secondary server as a contact
3. Asterisk re-INVITES to secondary server
4. Secondary server challenges with a 401 Unauthorized
5. Asterisk does NOT re-invite with the authentication fields even though
they are configured
properly.
If Asterisk INVITE's directly to Secondary server and avoids the 302,
asterisk properly autheniticates after the 401
and call goes thru succesfully (thats how I know the credentials work).
Does anyone know if this is a known limitation (being fixed in the next
beta version) or if this
may be configuration related?
Thanks,
Mushtaq Ahmed
3Com Corporation
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Message: 11
Date: Thu, 29 Mar 2007 09:18:00 -0600
From: Steve Murphy <murf at parsetree.com>
Subject: Re: [asterisk-users] L options in Dial() dont seem to
work....
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <1175181480.31166.31.camel at digium2>
Content-Type: text/plain
On Thu, 2007-03-29 at 15:51 +0100, Mark Reardon wrote:
> Hello Asterisk users,
>
> Can someone thwack me with a clue stick please?
> I am following the Asterisk TFOT book Dial() example trying to get the
> limit and announcements to work as per below.
>
> These settings seem to have no effect.
>
> There are no warning messages after 4 minutes or every 30 secs
> thereafter and the call lasts longer than 5 minutes.
>
> gunner*CLI> show dialplan
> [ Context 'outgoing' created by 'pbx_config' ]
> '123' => 1. Answer()
> [pbx_config]
> 2.
> AGI(/usr/local/share/examples/asterisk/agi/agi-test.agi) [pbx_config]
> 3. Hangup()
> [pbx_config]
> '_X.' => 1. Dial(SIP/sipprovider/${EXTEN}||
> L[300000:240000:30000]) [pbx_config]
There's the prob: It's L(...) not L[...]. Another case of a silently
rejected syntax error.
> 2. Hangup()
> [pbx_config]
>
> I am using 1.2.17.
>
> /usr/local/sbin/asterisk -vvvvvv -g -dddddd -c
>
> Does not show anything to even indicate * is trying anything unusual
> with regards to limits or warnings. It just seems to ignore the
> dialplan options altogether.
>
> Cheers
>
> Mark
> _______________________________________________
> --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
------------------------------
Message: 12
Date: Thu, 29 Mar 2007 17:18:32 +0200
From: Olivier <oza-4h07 at myamail.com>
Subject: [asterisk-users] Is it possible to install CCM on a Linux
platform ?
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<442fbb120703290818i3fce8d86pd57c29454901aa25 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi,
I know this question doesn't exactly relate to the core of this list but I
thought it does relate to its "hacker spirit".
Is it possible to install a Cisco Call Manager 5.X on a non-Cisco appliance
?
A friend of mine working for a Cisco VAR told me his colleagues couldn't
make it, even for testing purpose.
Do you agree ?
Regards
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Message: 13
Date: Thu, 29 Mar 2007 16:21:06 +0100
From: "Mark Reardon" <asterisk.mark at gmail.com>
Subject: Re: [asterisk-users] L options in Dial() dont seem to
work....
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<6f6ce67a0703290821h73d6971bw76c420aa66281d87 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Ahhh - in the tfot book they use [] not ().
Thanks a million
On 3/29/07, Eric ManxPower Wieling <eric at fnords.org> wrote:
>
> Mark Reardon wrote:
> > Hello Asterisk users,
> >
> > Can someone thwack me with a clue stick please?
> > I am following the Asterisk TFOT book Dial() example trying to get the
> > limit
> > and announcements to work as per below.
> >
> > These settings seem to have no effect.
> >
> > There are no warning messages after 4 minutes or every 30 secs
> thereafter
> > and the call lasts longer than 5 minutes.
> >
> > gunner*CLI> show dialplan
> > [ Context 'outgoing' created by 'pbx_config' ]
> > '123' => 1. Answer()
> [pbx_config]
> > 2. AGI(/usr/local/share/examples/asterisk/agi/agi-
> > test.agi) [pbx_config]
> > 3. Hangup()
> > [pbx_config]
> > '_X.' => 1.
> Dial(SIP/sipprovider/${EXTEN}||L[300000:240000:30000])
> > [pbx_config]
> > 2. Hangup()
> > [pbx_config]
> >
> > I am using 1.2.17.
> >
> > /usr/local/sbin/asterisk -vvvvvv -g -dddddd -c
> >
> > Does not show anything to even indicate * is trying anything unusual
> with
> > regards to limits or warnings. It just seems to ignore the dialplan
> options
> > altogether.
>
> You are using [ ] instead of ( ) after L
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
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Message: 14
Date: Thu, 29 Mar 2007 17:21:37 +0200
From: Benoit Panizzon <benoit.panizzon at imp.ch>
Subject: [asterisk-users] Scratchy Audio with Asterisk 1.2.4 over IAX
on FreeBSD?
To: asterisk-users at lists.digium.com
Message-ID: <200703291721.40835.benoit.panizzon at imp.ch>
Content-Type: text/plain; charset="iso-8859-1"
Hi all
We run an * 1.2.4 under FreeBSD with ztdummy kernel module.
zttest reports 99.9something % of accuracy, so timing should be fine.
SIP connections work fine, but we have a strange problem with IAX2
connections.
When an IAX2 call originates from the FreeBSD Asterisk to another Asterisk,
the sound is scratchy (sounds a bit like a 50Hz ground loop).
It's not a problem of the 'other' asterisk, as the problem could be
reproduced
with * 1.2.5/Linux and * 1.4.2/Linux.
If the IAX2 call originates from another * to the FreeBSD one, the sound is
clear.
Format used is alaw. (ulaw also shows that problem, gsm doesn't work at all,
but that could be a codec problem of the 1.2.4 gsm implementation)
Playing around with the IAX jitterbuffer settings does not affect the
scratching sound in any way.
Any idea what the cause could be?
Mit freundlichen Gr|ssen
Benoit Panizzon
--
I m p r o W a r e A G - System Services
______________________________________________________
Zurlindenstrasse 29 Tel +41 61 826 93 00
CH-4133 Pratteln Fax +41 61 826 93 01
Schweiz Web http://www.imp.ch
______________________________________________________
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Message: 15
Date: Thu, 29 Mar 2007 10:29:40 -0500 (CDT)
From: Jason Parker <jparker at digium.com>
Subject: Re: [asterisk-users] Cisco 30VIP Phone
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<21597668.84411175182180937.JavaMail.root at jupiler.digium.com>
Content-Type: text/plain; charset=utf-8
----- "Chris Nighswonger" <cnighswonger at foundations.edu> wrote:
> That is the conclusion I came to and was confirmed today in a very
> brief chat with one of the individuals listed as a developer on the
> chan_skinny module. He said that they could be implemented.
>
> What I would like to know, and do not understand, is the relationship
> between the code in chan_skinny.c which sets up the softkeys which
> are
> implimented and the actual key positions on the phone. With this
> info,
> I can hack the code to impliment other of the keys (ie. speed dial,
> etc.).
>
> Thanks,
> Chris
Search the code for "30VIP", there are only like 2-3 places where it's
referenced.
It should be immediately obvious how it works.
--
Jason Parker
Digium
------------------------------
Message: 16
Date: Thu, 29 Mar 2007 10:51:40 -0500
From: "Mike Hammett" <asterisk-users at ics-il.net>
Subject: [asterisk-users] SIP & NAT
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<asterisk-users at lists.digium.com>
Message-ID: <080201c7721a$24b29cf0$0f01010a at MidgetMan>
Content-Type: text/plain; charset="us-ascii"
I hate SIP. The only reason I'm doing this is that its cheaper than
deploying the server to a colo facility. My provider has given me a
non-standard IP block, so I can't do typical routing.
I have Asterisk server <-> MT w\NAT <-> PPPoE <-> MT <-> Provider.
I setup a dst-nat on 5060 to the Asterisk box.
Audio from Asterisk --> PSTN works great. Audio Asterisk <-- PSTN does
not.
Ideas?
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Message: 17
Date: Thu, 29 Mar 2007 11:52:02 -0400
From: "Matthew J. Roth" <mroth at imminc.com>
Subject: Re: [asterisk-users] maximum simultaneous calls
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <460BE0A2.8070206 at imminc.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Mark Quitoriano wrote:
> what could be the maximum simultaneous calls can asterisk do? i read
> about the asterisk business edition review[1] and it can only handle
> 120 simultaneous calls? i'm using 1.2.x branch of asterisk and i use
> more or less 90 simultaneous calls.
Mark,
We are regularly running 250-300 simultaneous calls in an inbound call
center environment. We had stability issues for a long time, but using
weights on the queues was the needle in the haystack that was causing
our problems. After removing them, our only failure in the past month
has been a single segmentation fault.
We are running Asterisk Business Edition. I can't find a quote on
Digium's website about their licensing limits, but the site you linked
actually quotes up to 240 calls. Our case is not the norm, because we
have a special agreement with Digium allowing us to have larger licenses.
Note that the use of ABE isn't a necessity for handling this many
calls. The stable releases of the open source version perform well, but
they don't have any support attached to them. Handling this many calls
requires eliminating as much overhead on the Asterisk server as
possible, running on high-capacity hardware, and troubleshooting the
problems that inevitably arise.
Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
------------------------------
Message: 18
Date: Thu, 29 Mar 2007 12:09:07 -0400
From: "Alexander Lopez" <Alex.Lopez at OpSys.com>
Subject: RE: [asterisk-users] SIP & NAT
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<E918F2FD95450648B7F8C957D92D52714B7D9A at exmail.corp.opsys.com>
Content-Type: text/plain; charset="us-ascii"
What do you mean by 'non-standard' IP block?
Is the Asterisk machine behind a NAT, or are only your clients?
Did you look at the nat setting sin sip.conf?
Do you have a static public address that can be routed to the Asterisk
box?
________________________________
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Mike
Hammett
Sent: Thursday, March 29, 2007 11:52 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] SIP & NAT
I hate SIP. The only reason I'm doing this is that its cheaper than
deploying the server to a colo facility. My provider has given me a
non-standard IP block, so I can't do typical routing.
I have Asterisk server <-> MT w\NAT <-> PPPoE <-> MT <-> Provider.
I setup a dst-nat on 5060 to the Asterisk box.
Audio from Asterisk --> PSTN works great. Audio Asterisk <-- PSTN does
not.
Ideas?
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Message: 19
Date: Thu, 29 Mar 2007 12:13:28 -0400
From: Drew Gibson <drew at oanda.com>
Subject: Re: [asterisk-users] Multi-line phones - Asterisk uses wrong
callerid
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <460BE5A8.9080400 at oanda.com>
Content-Type: text/plain; charset=UTF-8; format=flowed
Thanks Andrew, I understand the issue now.
Removing "insecure=very" allows the Grandstream phones to work, they
register separate lines on separate ports (eg Line 1=5060, Line 2=5062,
etc).
Unfortunately I cannot find a port setting for the Aastra 480i, I shall
get on their case.
regards,
Drew
Andrew Joakimsen wrote:
>
;---------------------------------------------------------------------------
---
>
> ; Definitions of locally connected SIP phones
> ;
> ; type = user a device that authenticates to us by "from" field to
> place calls
> ; type = peer a device we place calls to or that calls us and we
> match by host
> ; type = friend two configurations (peer+user) in one
> ;
>
>
> Thus if you have two "peers" using the same IP address AND port it
> will probably match. First try to remove insecure=very from your
> configuration file, that alone might resolve it. If not you need to
> insure that each line gets its own port.
>
> On 3/28/07, Drew Gibson <drew at oanda.com> wrote:
>> I have some phones (and an ATA) that are shared between two users who
>> each have separate voicemail but they are not behaving as desired nor
>> expected.
>>
>> Incoming calls show up on the correct lines.
>> Calls originating from the device are seen, at the terminating device,
>> as coming from the account listed last in sip.conf, regardless of the
>> line selected.
>>
>> This creates three main issues I would like to resolve:-
>> 1. The person called sees the wrong callerid
>> 2. The CDR records the call against the wrong account
>> 3. Picking up voicemail requires multiple extra steps
>>
>> Is there a way around this??
>>
>> Scenario:-
>> Phone 1 has three lines 101, 102, 103
>> Phone 2 has 1 line 202
>>
>> User 1 selects line 101 at Phone 1 and dials 202 (to Phone 2)
>> User 2 at Phone 2 sees call coming from extension 103 instead of 101
>>
>> With 'sip debug' enabled at the console, I see an INVITE issued (on the
>> Phone 1 to Asterisk leg) from the correct extension, 101, to 202 but the
>> call leg from Asterisk to Phone 202 shows an INVITE from 103 to 202.
>> 103 happens to be the last listed in sip.conf and the first listed in
>> 'sip show peers' (I have confirmed that this is dependent on the order
>> in the conf file, not numeric order)
>>
>> sip.conf :-
>> [general]
>> port = 5060
>> bindaddr = 0.0.0.0
>> pedantic = no
>> autocreatepeer = no
>> context = sip
>> registertimeout=20
>> localnet = 10.10.10.0/255.255.255.0
>> srvlookup = yes
>> tos=0xb8
>> rtptimeout=300
>> rtpholdtimeout=1800
>> maxexpirey=3600
>> defaultexpirey=1200
>>
>> [sip-101]
>> ; Aastra 480i phones for general office
>> type=peer
>> insecure=very
>> disallow=all
>> allow=ulaw
>> allow=alaw
>> host=dynamic
>> dtmfmode=auto
>> canreinvite=no
>> context=office-dial
>> qualify=yes
>> username=101
>> secret=xxxxxx
>> mailbox=101
>> callerid="User 1" <101>
>>
>>
>> sip show peers :-
>> 103/103 10.10.10.181 D 5060 OK
>> (157 ms)
>> 102/102 10.10.10.181 D 5060 OK
>> (159 ms)
>> 202/202 10.10.10.184 D 5060 OK
>> (4 ms)
>> 101/101 10.10.10.181 D 5060 OK
>> (160 ms)
>>
>>
>> Asterisk 1.2.15
>> Phones tested:- Aastra 480i, Grandstream GXP2000, Grandstream HT-386 ATA
>>
>> --
>> Drew Gibson
>>
>> Systems Administrator
>> OANDA Corporation
>> 416-593-6767 x322
>> www.oanda.com
>>
>> _______________________________________________
>> --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
>>
> _______________________________________________
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> asterisk-users mailing list
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>
--
Drew Gibson
Systems Administrator
OANDA Corporation
416-593-6767 x322
www.oanda.com
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