<div dir="ltr"><pre><i>I'm trying to build an application that provides statistics of calls
</i>><i> and call recording. Someone told me this could be done out of band
</i>><i> with a SPAN (?) port that would replicate SIP and media packets to a
</i>><i> separate NIC without having to actually pass the real-calls thru
</i>><i> asterisk. It was explained that this SPAN port would in the SBC
</i>><i> would replicate data received.
</i>><i>
</i>><i>
</i>><i> If this is done, is there a way I can utilize asterisk to interpret
</i>><i> these packets without actually having any control of the calls? If so
</i>><i> how? Sorry, I'm new to telco, so hopefully this post makes sense to
</i>><i> someone.</i></pre>
</div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Apr 30, 2013 at 10:30 PM, <span dir="ltr"><<a href="mailto:asterisk-users-request@lists.digium.com" target="_blank">asterisk-users-request@lists.digium.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Send asterisk-users mailing list submissions to<br>
<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>
<br>
To subscribe or unsubscribe via the World Wide Web, visit<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
or, via email, send a message with subject or body 'help' to<br>
<a href="mailto:asterisk-users-request@lists.digium.com">asterisk-users-request@lists.digium.com</a><br>
<br>
You can reach the person managing the list at<br>
<a href="mailto:asterisk-users-owner@lists.digium.com">asterisk-users-owner@lists.digium.com</a><br>
<br>
When replying, please edit your Subject line so it is more specific<br>
than "Re: Contents of asterisk-users digest..."<br>
<br>
<br>
Today's Topics:<br>
<br>
1. Re: Asterisk 11.3.0 - Mask for new file not correct (David M. Lee)<br>
2. Gateway? (James Wystead)<br>
3. Re: Gateway? (jg)<br>
4. Re: Asterisk 11.3.0 - Mask for new file not correct (Ludovic Bou?)<br>
5. Re: Gateway? (A J Stiles)<br>
6. Re: Can't register to Asterisk 1.6 with old Aastra phones<br>
(Bob Kyeyune)<br>
7. Re: Gateway? (Eric Wieling)<br>
8. hello! (Rahul Pachauri)<br>
9. Asterisk QSIG doesnt send the calling name to Nortel CS1000<br>
(Danilo Dionisi)<br>
<br>
<br>
----------------------------------------------------------------------<br>
<br>
Message: 1<br>
Date: Mon, 29 Apr 2013 12:51:42 -0500<br>
From: "David M. Lee" <<a href="mailto:dlee@digium.com">dlee@digium.com</a>><br>
Subject: Re: [asterisk-users] Asterisk 11.3.0 - Mask for new file not<br>
correct<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
Message-ID: <<a href="mailto:38652031-864B-45A9-A779-B90862F1AE97@digium.com">38652031-864B-45A9-A779-B90862F1AE97@digium.com</a>><br>
Content-Type: text/plain; charset=iso-8859-1<br>
<br>
<br>
On Apr 29, 2013, at 10:51 AM, Ludovic Bou? wrote:<br>
<br>
> The fact is we want to use the RECORDED_FILE function from Application_Record module and create a file with 666 permissions. But when I check the created file, rights are not what I expected.<br>
><br>
> [root@STD1-SRVASTSVI-03 pseudos]$ ll<br>
> -rw-r--r-- 1 asterisk asterisk 51244 mars 29 16:04 Pseudo_2_1111.wav<br>
><br>
> I checked the doc on <a href="https://wiki.asterisk.org/wiki/display/AST/Application_Record" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Application_Record</a> but I didn't find anything about umask permissions. I checked Doxygen, I can see file creation permissions is set to 666<br>
> #define AST_FILE_MODE 0666<br>
> <a href="http://doxygen.asterisk.org/trunk/asterisk_8h.html#a6293b2dae52a2b470494df672a26c42" target="_blank">http://doxygen.asterisk.org/trunk/asterisk_8h.html#a6293b2dae52a2b470494df672a26c42</a><br>
><br>
> What can I do to fix that or debug?<br>
<br>
The AST_FILE_MODE works by the same rules as mode parameter in open(2): "The effective permissions are modified by the process's umask in the usual way: The permissions of the created file are (mode & ~umask)."[1]<br>
<br>
My guess is that the umask of your asterisk process is 022, which is very typical. You'll have to play around with your umask settings and file permissions to get things the way you want them.<br>
<br>
[1]: <a href="http://linux.die.net/man/2/open" target="_blank">http://linux.die.net/man/2/open</a><br>
<br>
> Ludovic BOU?<br>
<br>
--<br>
David M. Lee<br>
Digium, Inc. | Software Developer<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
Check us out at: <a href="http://www.digium.com" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a><br>
<br>
<br>
<br>
<br>
------------------------------<br>
<br>
Message: 2<br>
Date: Mon, 29 Apr 2013 15:56:24 -0400<br>
From: James Wystead <<a href="mailto:szilverthorne@gmail.com">szilverthorne@gmail.com</a>><br>
Subject: [asterisk-users] Gateway?<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
Message-ID:<br>
<CAMoLvkyLF_U5N_8aAOvz40JqQuFOpU=<a href="mailto:A%2BgZaW0%2BziTGaRZXE2g@mail.gmail.com">A+gZaW0+ziTGaRZXE2g@mail.gmail.com</a>><br>
Content-Type: text/plain; charset="iso-8859-1"<br>
<br>
This is going to sound like a dumb-ass question:<br>
<br>
The device that allows you to bridge Asterisk (or any other PBX) into the<br>
pstn.. What is that called? So, I guess, not a SIP trunk, but the device<br>
that actually IS the SIP trunk.<br>
<br>
Am I making sense?<br>
<br>
Thanks<br>
-------------- next part --------------<br>
An HTML attachment was scrubbed...<br>
URL: <<a href="http://lists.digium.com/pipermail/asterisk-users/attachments/20130429/5eba090f/attachment-0001.htm" target="_blank">http://lists.digium.com/pipermail/asterisk-users/attachments/20130429/5eba090f/attachment-0001.htm</a>><br>
<br>
------------------------------<br>
<br>
Message: 3<br>
Date: Mon, 29 Apr 2013 22:35:08 +0200<br>
From: jg <<a href="mailto:webaccounts@jgoettgens.de">webaccounts@jgoettgens.de</a>><br>
Subject: Re: [asterisk-users] Gateway?<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
Message-ID: <<a href="mailto:517ED97C.8090606@jgoettgens.de">517ED97C.8090606@jgoettgens.de</a>><br>
Content-Type: text/plain; charset=UTF-8; format=flowed<br>
<br>
Here are your answers:<br>
<br>
1st question: Anything that makes sense.<br>
2nd question: Maybe<br>
<br>
Please, explain your setup.<br>
<br>
jg<br>
<br>
<br>
<br>
------------------------------<br>
<br>
Message: 4<br>
Date: Tue, 30 Apr 2013 10:35:58 +0200 (CEST)<br>
From: Ludovic Bou? <<a href="mailto:lboue@afone.com">lboue@afone.com</a>><br>
Subject: Re: [asterisk-users] Asterisk 11.3.0 - Mask for new file not<br>
correct<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
Message-ID: <<a href="mailto:1379623362.9592526.1367310958910.JavaMail.root@afone.com">1379623362.9592526.1367310958910.JavaMail.root@afone.com</a>><br>
Content-Type: text/plain; charset=utf-8<br>
<br>
----- Mail original -----<br>
De: "David M. Lee" <<a href="mailto:dlee@digium.com">dlee@digium.com</a>><br>
?: "Asterisk Users Mailing List - Non-Commercial Discussion" <<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
Envoy?: Lundi 29 Avril 2013 19:51:42<br>
Objet: Re: [asterisk-users] Asterisk 11.3.0 - Mask for new file not correct<br>
<br>
<br>
On Apr 29, 2013, at 10:51 AM, Ludovic Bou? wrote:<br>
<br>
> The fact is we want to use the RECORDED_FILE function from Application_Record module and create a file with 666 permissions. But when I check the created file, rights are not what I expected.<br>
><br>
> [root@STD1-SRVASTSVI-03 pseudos]$ ll<br>
> -rw-r--r-- 1 asterisk asterisk 51244 mars 29 16:04 Pseudo_2_1111.wav<br>
><br>
> I checked the doc on <a href="https://wiki.asterisk.org/wiki/display/AST/Application_Record" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Application_Record</a> but I didn't find anything about umask permissions. I checked Doxygen, I can see file creation permissions is set to 666<br>
> #define AST_FILE_MODE 0666<br>
> <a href="http://doxygen.asterisk.org/trunk/asterisk_8h.html#a6293b2dae52a2b470494df672a26c42" target="_blank">http://doxygen.asterisk.org/trunk/asterisk_8h.html#a6293b2dae52a2b470494df672a26c42</a><br>
><br>
> What can I do to fix that or debug?<br>
<br>
The AST_FILE_MODE works by the same rules as mode parameter in open(2): "The effective permissions are modified by the process's umask in the usual way: The permissions of the created file are (mode & ~umask)."[1]<br>
<br>
My guess is that the umask of your asterisk process is 022, which is very typical. You'll have to play around with your umask settings and file permissions to get things the way you want them.<br>
<br>
[1]: <a href="http://linux.die.net/man/2/open" target="_blank">http://linux.die.net/man/2/open</a><br>
<br>
<br>
You were right, it was necessary to change asterisk process umask. I put the following in /etc/init.d/asterisk init script and it works:<br>
# umask 002 to create files with 0664 and folders with 0775<br>
umask 002<br>
<br>
Thanks a lot<br>
<br>
<br>
<br>
------------------------------<br>
<br>
Message: 5<br>
Date: Tue, 30 Apr 2013 10:57:32 +0100<br>
From: A J Stiles <<a href="mailto:asterisk_list@earthshod.co.uk">asterisk_list@earthshod.co.uk</a>><br>
Subject: Re: [asterisk-users] Gateway?<br>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"<br>
<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
Message-ID: <<a href="mailto:201304301057.32260.asterisk_list@earthshod.co.uk">201304301057.32260.asterisk_list@earthshod.co.uk</a>><br>
Content-Type: Text/Plain; charset="iso-8859-6"<br>
<br>
On Monday 29 April 2013, James Wystead wrote:<br>
> This is going to sound like a dumb-ass question:<br>
><br>
> The device that allows you to bridge Asterisk (or any other PBX) into the<br>
> pstn.. What is that called?<br>
<br>
Usually it is an expansion card that plugs into a PCI or PCI express slot on<br>
the motherboard; so most people would just call it an analogue telephony card<br>
(such as a TDM410P, for instance) or an ISDN card (such as a TE410P). One<br>
that connects to the mobile networks would be called a GSM card.<br>
<br>
Analogue telephony cards are further subdivided into two flavours; FXO (which<br>
connects to an exchange line) and FXS (which connects to a telephone, and<br>
provides the necessary line bias and ringing voltages). Usually a single card<br>
will provide for multiple lines, by fitting either FXO or FXS modules as<br>
required.<br>
<br>
--<br>
AJS<br>
<br>
Answers come *after* questions.<br>
<br>
<br>
<br>
------------------------------<br>
<br>
Message: 6<br>
Date: Tue, 30 Apr 2013 14:21:49 +0300<br>
From: Bob Kyeyune <<a href="mailto:bkyeyune@gmail.com">bkyeyune@gmail.com</a>><br>
Subject: Re: [asterisk-users] Can't register to Asterisk 1.6 with old<br>
Aastra phones<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
Message-ID:<br>
<CAPd1dq_ktetBpLHhXDm=0wr4QTRj2MHmp4-9ufCjG=<a href="mailto:QmvzWuug@mail.gmail.com">QmvzWuug@mail.gmail.com</a>><br>
Content-Type: text/plain; charset="iso-8859-1"<br>
<br>
am also stuck with Alcatel lucent IP Touch 4018<br>
any one connected them to Asterisk<br>
<br>
thanks<br>
<br>
Regards.<br>
Kyeyune Bob<br>
Network & IT Engineer<br>
+256 774 702 258<br>
<a href="mailto:bob.kyeyune@onesolutions.ug">bob.kyeyune@onesolutions.ug</a><br>
<br>
Integrated IT services from<br>
Plot 57B Luthuli Avenue Bugolobi, Kampala<br>
<br>
<br>
<br>
<br>
<br>
<br>
On Sun, Apr 28, 2013 at 11:56 PM, Carlos Alvarez <<a href="mailto:carlos@televolve.com">carlos@televolve.com</a>>wrote:<br>
<br>
> We have a new customer with a lot of old phones like the 9133i. They<br>
> won't register, and we see some very strange behavior with them. If<br>
> the SIP peer exists, they simply fail silently, with no error in the<br>
> CLI or the messages log. Nothing works, but no errors.<br>
><br>
> If the peer does not exist, it's clear that it's registering improperly:<br>
><br>
> [2013-04-28 13:34:31] NOTICE[3058] chan_sip.c: Registration from<br>
> 'abc123 <sip:abc123@>' failed for '68.2.x.x' - No matching peer found<br>
><br>
> Typically of course we'd expect to see: <sip:abc123@server><br>
><br>
> We're running the latest available firmware, but it's from 2009. Any<br>
> ideas on this before we just trash all the older phones?<br>
><br>
> --<br>
> Carlos Alvarez<br>
> TelEvolve<br>
> 602-889-3003<br>
><br>
> --<br>
> _____________________________________________________________________<br>
> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
><br>
> asterisk-users mailing list<br>
> To UNSUBSCRIBE or update options visit:<br>
> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
><br>
-------------- next part --------------<br>
An HTML attachment was scrubbed...<br>
URL: <<a href="http://lists.digium.com/pipermail/asterisk-users/attachments/20130430/f7f1bc6b/attachment-0001.htm" target="_blank">http://lists.digium.com/pipermail/asterisk-users/attachments/20130430/f7f1bc6b/attachment-0001.htm</a>><br>
<br>
------------------------------<br>
<br>
Message: 7<br>
Date: Tue, 30 Apr 2013 09:11:48 -0400<br>
From: Eric Wieling <<a href="mailto:EWieling@nyigc.com">EWieling@nyigc.com</a>><br>
Subject: Re: [asterisk-users] Gateway?<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
Message-ID:<br>
<616B4ECE1290D441AD56124FEBB03D081713F8F0C2@mailserver2007.nyigc.globe><br>
<br>
Content-Type: text/plain; charset="us-ascii"<br>
<br>
On Monday 29 April 2013, James Wystead wrote:<br>
> This is going to sound like a dumb-ass question:<br>
><br>
> The device that allows you to bridge Asterisk (or any other PBX) into<br>
> the pstn.. What is that called?<br>
<br>
For 1 - 2 ports they are usually called an ATA (Analog Terminal Adapter). For more than 2 ports they are usually called Media Gateways.<br>
<br>
<br>
<br>
------------------------------<br>
<br>
Message: 8<br>
Date: Tue, 30 Apr 2013 21:42:43 +0800 (SGT)<br>
From: Rahul Pachauri <<a href="mailto:rahul.pachauri@ymail.com">rahul.pachauri@ymail.com</a>><br>
Subject: [asterisk-users] hello!<br>
To: hr ccsgroups <<a href="mailto:hr.ccsgroups@gmail.com">hr.ccsgroups@gmail.com</a>>, coolguyrocks<br>
<<a href="mailto:coolguyrocks@in.com">coolguyrocks@in.com</a>>, simbus hr <<a href="mailto:simbus.hr@simbustech.com">simbus.hr@simbustech.com</a>>, gowdanar<br>
<gowdanar@ChetanaSforum.com>, asterisk users<br>
<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>>, hr <<a href="mailto:hr@metaoption.com">hr@metaoption.com</a>>, rcnoida<br>
<<a href="mailto:rcnoida@ignou.ac.in">rcnoida@ignou.ac.in</a>><br>
Message-ID:<br>
<<a href="mailto:1367329363.26314.YahooMailNeo@web194905.mail.sg3.yahoo.com">1367329363.26314.YahooMailNeo@web194905.mail.sg3.yahoo.com</a>><br>
Content-Type: text/plain; charset="utf-8"<br>
<br>
<a href="http://seed4life.org/wp-content/themes/twentytwelve/basesball.php?hfazq792vlxjd" target="_blank">http://seed4life.org/wp-content/themes/twentytwelve/basesball.php?hfazq792vlxjd</a><br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
________________<br>
You're going to find that many of the truths we cling to depend entirely upon one's point of view. -- Obi-Wan Kenobi<br>
-------------- next part --------------<br>
An HTML attachment was scrubbed...<br>
URL: <<a href="http://lists.digium.com/pipermail/asterisk-users/attachments/20130430/fbb4af3e/attachment-0001.htm" target="_blank">http://lists.digium.com/pipermail/asterisk-users/attachments/20130430/fbb4af3e/attachment-0001.htm</a>><br>
<br>
------------------------------<br>
<br>
Message: 9<br>
Date: Tue, 30 Apr 2013 18:30:27 +0200<br>
From: Danilo Dionisi <<a href="mailto:dionisi.danilo@gmail.com">dionisi.danilo@gmail.com</a>><br>
Subject: [asterisk-users] Asterisk QSIG doesnt send the calling name<br>
to Nortel CS1000<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
Message-ID:<br>
<<a href="mailto:CAAbtBUys04Zw8u5jan%2BzcjybOfxrVks_wiGy5C60YTjrZ-JQbg@mail.gmail.com">CAAbtBUys04Zw8u5jan+zcjybOfxrVks_wiGy5C60YTjrZ-JQbg@mail.gmail.com</a>><br>
Content-Type: text/plain; charset="iso-8859-1"<br>
<br>
Hello to all,<br>
<br>
I have a problem with an asterisk qsig.<br>
<br>
I have three machines:<br>
<br>
Nortel CS1000 --- Card Sangoma PRI ---> Asterisk QSIG ---SIP Trunk---><br>
Asterisk<br>
<br>
I use Snom phones on Asterisk.<br>
If I call from Asterisk to Nortel, Nortel reminds me of the name of the person<br>
i'm calling and I visualize on the display of Snom phone, but if I call from<br>
Nortel to Asterisk, the QSIG does not send Nortel on the display of the<br>
name of the person i'm calling ... why?<br>
<br>
example:<br>
Snom phone = "Danilo <1001>"<br>
Nortel phone = "Marco <2002>"<br>
<br>
If I call from Nortel to Asterisk, I have the display of the Snom "Marco <<br>
2002>" and the display of Nortel "Danilo <1001>"; If I call from Nortel to<br>
Asterisk, I have the display of the Snom "Marco <2002>" and the display of<br>
Nortel "<1001>"<br>
<br>
This is my / etc / asterisk / chan_dahdi.conf<br>
<br>
[channels]<br>
cc_offer_timer=20<br>
ccbs_available_timer=4800<br>
ccnr_available_timer=7200<br>
cc_recall_timer=20<br>
cc_agent_policy=native<br>
cc_monitor_policy=native<br>
pridialplan=private<br>
prilocaldialplan=private<br>
<br>
context=default<br>
usecallerid=yes<br>
hidecallerid=no<br>
callwaiting=yes<br>
usecallingpres=yes<br>
callwaitingcallerid=yes<br>
threewaycalling=yes<br>
transfer=yes<br>
canpark=yes<br>
cancallforward=yes<br>
callreturn=yes<br>
echocancel=yes<br>
echocancelwhenbridged=yes<br>
relaxdtmf=yes<br>
rxgain=0.0<br>
txgain=0.0<br>
group=1<br>
callgroup=1<br>
pickupgroup=1<br>
immediate=no<br>
facilityenable=yes<br>
callerid=asreceived<br>
<br>
<br>
<br>
;Sangoma A104 port 1 [slot:4 bus:17 span:1] <wanpipe1><br>
switchtype=qsig<br>
context=from_nortel<br>
group=0<br>
echocancel=yes<br>
faxdetect=incoming<br>
signalling=pri_cpe<br>
channel =>1-15,17-31<br>
-------------- next part --------------<br>
An HTML attachment was scrubbed...<br>
URL: <<a href="http://lists.digium.com/pipermail/asterisk-users/attachments/20130430/b1c54de0/attachment-0001.htm" target="_blank">http://lists.digium.com/pipermail/asterisk-users/attachments/20130430/b1c54de0/attachment-0001.htm</a>><br>
<br>
------------------------------<br>
<br>
_______________________________________________<br>
--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--" target="_blank">http://www.api-digital.com--</a><br>
<br>
AstriCon 2010 - October 26-28 Washington, DC<br>
Register Now: <a href="http://www.astricon.net/" target="_blank">http://www.astricon.net/</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
<br>
End of asterisk-users Digest, Vol 105, Issue 39<br>
***********************************************<br>
</blockquote></div><br><br clear="all"><br>-- <br>BIPIN RAGHUVANSHI<br>OPERATION HEAD<br>ASTERISK (DEVELOPMENT AND RESEARCH) <br><a href="http://WWW.EHORIZONS.IN" target="_blank">WWW.EHORIZONS.IN</a><br><a href="mailto:bipinraghuvanshi@gmail.com" target="_blank">bipinraghuvanshi@gmail.com</a><br>
<a href="mailto:bipin.singh@ehorizons.in" target="_blank">bipin.singh@ehorizons.in</a><br>
</div>