Dallas Carter
2008-Nov-29 19:33 UTC
[asterisk-users] asterisk-users Digest, Vol 52, Issue 81
I was cleaning and working on laptops most of the day. Check my logs, I did
plenty of work.
-----Original Message-----
From: "asterisk-users-request at lists.digium.com"
<asterisk-users-request at lists.digium.com>
To: "asterisk-users at lists.digium.com" <asterisk-users at
lists.digium.com>
Sent: 11/29/2008 1:13 PM
Subject: asterisk-users Digest, Vol 52, Issue 81
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Today's Topics:
1. Re: Any 1.6 SendFAX example ? (Anthony Messina)
2. Re: force channel hangup (Anthony Messina)
3. Re: force channel hangup (Danny Nicholas)
4. Re: force channel hangup (Alex Balashov)
5. Re: force channel hangup (Danny Nicholas)
6. Re: force channel hangup (Tzafrir Cohen)
7. Re: force channel hangup (Tzafrir Cohen)
8. received wrong state events for originate command (Sun xiaoshuang)
9. Trixbox 2.6.1.13 OpenR2 (Yuri)
10. Trixbox 2.6.1.13 OpenR2 (Yuri)
11. libspandsp.so.0: cannot open shared object file: No such file
or directory (Doug)
12. Re: Trixbox 2.6.1.13 OpenR2 (Peter Lindquist)
13. Re: libspandsp.so.0: cannot open shared object file: No such
file or directory (Alex Balashov)
14. Re: Anonymous callerid (Max Alex)
15. GSM gateways - which one ? (Julian Lyndon-Smith)
16. Re: GSM gateways - which one ? (Julian Lyndon-Smith)
17. Re: GSM gateways - which one ? (Michael Graves)
18. Re: Anonymous callerid (Tilghman Lesher)
----------------------------------------------------------------------
Message: 1
Date: Fri, 28 Nov 2008 15:46:53 -0600
From: Anthony Messina <amessina at messinet.com>
Subject: Re: [asterisk-users] Any 1.6 SendFAX example ?
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID: <200811281546.54070.amessina at messinet.com>
Content-Type: text/plain; charset="utf-8"
On Thursday 27 November 2008 05:03:00 Olivier wrote:> Hi,
>
> Do you have any example showing how to use SendFAX ?
> I can see several examples of ReceiveFAX but not a single one showing
> SendFAX.
i'm working on a script to incorporate e-mail <-> fax gatewaying with
asterisk
using programs that are already available in linux.
there are simple examples here:
http://messinet.com/viewvc/asterisk-fax-gw/trunk/
--
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E
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Message: 2
Date: Fri, 28 Nov 2008 15:48:07 -0600
From: Anthony Messina <amessina at messinet.com>
Subject: Re: [asterisk-users] force channel hangup
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID: <200811281548.07486.amessina at messinet.com>
Content-Type: text/plain; charset="iso-8859-1"
On Thursday 27 November 2008 20:25:49 Kelvin Chan wrote:> Hi guys,
>
> I have 1 zap channel in my house shared among couple people. If someone
> dials 911, I want that zap channel to be disconnected right away to make
> way for the 911 call.
>
> I dug through voip-info.org and didn't find much.
> Any hints?
>
i use this:
http://messinet.com/index.php?page_name=Asterisk&wikipage=Asteriske911
--
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E
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Message: 3
Date: Fri, 28 Nov 2008 16:18:57 -0600
From: "Danny Nicholas" <danny at debsinc.com>
Subject: Re: [asterisk-users] force channel hangup
To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
<asterisk-users at lists.digium.com>
Message-ID: <B1937F6ACE7545BA8EC28D22755A7A8D at db0005>
Content-Type: text/plain; charset="us-ascii"
Why wouldn't this work?
exten => _911,1,Hangup(Zap/1)
exten => _911,2,Dial(Zap/1/ww911,60)
exten => _911,3,Hangup()
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Anthony
Messina
Sent: Friday, November 28, 2008 3:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] force channel hangup
On Thursday 27 November 2008 20:25:49 Kelvin Chan wrote:> Hi guys,
>
> I have 1 zap channel in my house shared among couple people. If someone
> dials 911, I want that zap channel to be disconnected right away to make
> way for the 911 call.
>
> I dug through voip-info.org and didn't find much.
> Any hints?
>
i use this:
http://messinet.com/index.php?page_name=Asterisk&wikipage=Asteriske911
--
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E
------------------------------
Message: 4
Date: Fri, 28 Nov 2008 17:24:36 -0500
From: Alex Balashov <abalashov at evaristesys.com>
Subject: Re: [asterisk-users] force channel hangup
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <49306FA4.8010206 at evaristesys.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Because hangup (and other behavioural) directives can only be addressed
to a particular instance of a channel use, i.e.
Technology/channel-uniqueID. The latter are not addressable from the
dial plan except implicitly.
Danny Nicholas wrote:
> Why wouldn't this work?
> exten => _911,1,Hangup(Zap/1)
> exten => _911,2,Dial(Zap/1/ww911,60)
> exten => _911,3,Hangup()
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Anthony
> Messina
> Sent: Friday, November 28, 2008 3:48 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] force channel hangup
>
> On Thursday 27 November 2008 20:25:49 Kelvin Chan wrote:
>> Hi guys,
>>
>> I have 1 zap channel in my house shared among couple people. If someone
>> dials 911, I want that zap channel to be disconnected right away to
make
>> way for the 911 call.
>>
>> I dug through voip-info.org and didn't find much.
>> Any hints?
>>
>
> i use this:
> http://messinet.com/index.php?page_name=Asterisk&wikipage=Asteriske911
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
------------------------------
Message: 5
Date: Fri, 28 Nov 2008 16:42:01 -0600
From: "Danny Nicholas" <danny at debsinc.com>
Subject: Re: [asterisk-users] force channel hangup
To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
<asterisk-users at lists.digium.com>
Message-ID: <05592046AC8B4CDB81A58F7447BB04B3 at db0005>
Content-Type: text/plain; charset="us-ascii"
Right you are, Alex. How about (CLI) Zap restart? I was thinking zap
destroy channel 1, but that just kills the channel until you do a zap
restart. That being said, this is an option
exten => _911,1,System('/usr/sbin/asterisk -rx "zap
restart"')
exten => _911,2,System('/usr/sbin/asterisk -rx "zap
restart"')
Second instance is to start the line that was in use during first restart
exten => _911,3,Dial(Zap/1/ww911,60)
exten => _911,3,Hangup()
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Alex Balashov
Sent: Friday, November 28, 2008 4:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] force channel hangup
Because hangup (and other behavioural) directives can only be addressed
to a particular instance of a channel use, i.e.
Technology/channel-uniqueID. The latter are not addressable from the
dial plan except implicitly.
Danny Nicholas wrote:
> Why wouldn't this work?
> exten => _911,1,Hangup(Zap/1)
> exten => _911,2,Dial(Zap/1/ww911,60)
> exten => _911,3,Hangup()
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Anthony
> Messina
> Sent: Friday, November 28, 2008 3:48 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] force channel hangup
>
> On Thursday 27 November 2008 20:25:49 Kelvin Chan wrote:
>> Hi guys,
>>
>> I have 1 zap channel in my house shared among couple people. If someone
>> dials 911, I want that zap channel to be disconnected right away to
make
>> way for the 911 call.
>>
>> I dug through voip-info.org and didn't find much.
>> Any hints?
>>
>
> i use this:
> http://messinet.com/index.php?page_name=Asterisk&wikipage=Asteriske911
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
------------------------------
Message: 6
Date: Sat, 29 Nov 2008 01:10:23 +0200
From: Tzafrir Cohen <tzafrir.cohen at xorcom.com>
Subject: Re: [asterisk-users] force channel hangup
To: asterisk-users at lists.digium.com
Message-ID: <20081128231023.GJ23105 at xorcom.com>
Content-Type: text/plain; charset=us-ascii
On Fri, Nov 28, 2008 at 04:42:01PM -0600, Danny Nicholas
wrote:> Right you are, Alex. How about (CLI) Zap restart? I was thinking zap
> destroy channel 1, but that just kills the channel until you do a zap
> restart. That being said, this is an option
>
> exten => _911,1,System('/usr/sbin/asterisk -rx "zap
restart"')
> exten => _911,2,System('/usr/sbin/asterisk -rx "zap
restart"')
This will disconnect all existing Zap calls.
BTW: As of Asterisk 1.4.22 / 1.6.0 'dahdi restart' actually works as
promised and you don't need to run it twice.
> Second instance is to start the line that was in use during first restart
> exten => _911,3,Dial(Zap/1/ww911,60)
> exten => _911,3,Hangup()
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen at xorcom.com
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
------------------------------
Message: 7
Date: Sat, 29 Nov 2008 01:13:45 +0200
From: Tzafrir Cohen <tzafrir.cohen at xorcom.com>
Subject: Re: [asterisk-users] force channel hangup
To: asterisk-users at lists.digium.com
Message-ID: <20081128231345.GK23105 at xorcom.com>
Content-Type: text/plain; charset=us-ascii
On Fri, Nov 28, 2008 at 05:24:36PM -0500, Alex Balashov
wrote:> Because hangup (and other behavioural) directives can only be addressed
> to a particular instance of a channel use, i.e.
> Technology/channel-uniqueID. The latter are not addressable from the
> dial plan except implicitly.
For a Zap channel the unique ID will mostly be '1' . In some cases it
will be '2'. So:
exten => _911,1,Hangup(Zap/1-1)
exten => _911,n,Hangup(Zap/1-2)
exten => _911,n,Dial(Zap/1/ww911,60)
exten => _911,n,Hangup()
I wonder, though, how long does it take for the hangup to take effect. A
hangup requests the channel to hang up. This is done later in the
channel context. I wonder if it is normally done quickly enough.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen at xorcom.com
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
------------------------------
Message: 8
Date: Sat, 29 Nov 2008 11:02:43 +0800
From: "Sun xiaoshuang" <xiaoshuangsun at gmail.com>
Subject: [asterisk-users] received wrong state events for originate
command
To: asterisk-users at lists.digium.com
Message-ID:
<96fb6d240811281902w3a7fdf2ckfee4a46ed3847af5 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hey all,
Something is wrong when i use originate command to call my phone
(Asterisk1.4.22 + xp100 card).
Actually, i have two problems.
The first one: If i fire a outgoing call using originate command directly,
after my pc startup, i will receive below error message:
[Nov 26 07:58:53] NOTICE[6559]: channel.c:2898 __ast_request_and_dial:
Unable to request channel Zap/1/13xxxxxxxxx
but i can call the FXO using my phone, everything seems perfect! After the
incomming call, i fire outgoing call using originate again, it works now, my
phone can ring, i also can pick up it(I seems originate did not create a new
Zap channel,just used an exsiting channel?).
But the second problem produced, i received the Dialing, UP, Newexten events
before my phone ringing. It is supposed that i send an originate command
(like Dial application), the last state should be Dialing... until i pick up
my phone or timeout.
These problems only for Zap channel, if i fire a outgoing call to SIP
channel, it works well.
What wrong with me ?
Here is my php script:
$socket =
fsockopen("127.0.0.1","5038",$errno,$errstr,$timeout);
fputs($socket,"Action: Login\r\n");
fputs($socket,"Username: tester\r\n");
fputs($socket,"Secret: test\r\n\r\n");
fputs($socket,"Action: Originate\r\n");
fputs($socket,"Channel: Zap/1/13XXXXXXXX\r\n");
fputs($socket,"Context: callme\r\n");
fputs($socket,"Exten: s\r\n");
fputs($socket,"Priority: 1\r\n\r\n");
fclose($socket);
Best regards,
Xiaoshuang
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Message: 9
Date: Sat, 29 Nov 2008 02:18:50 -0200
From: Yuri <yuri.asterisk at gmail.com>
Subject: [asterisk-users] Trixbox 2.6.1.13 OpenR2
To: asterisk-users at lists.digium.com
Message-ID:
<52068a5e0811282018x6053d1bcv2c9df7d0969939ad at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
*Good morning! *
*I verified that the trixbox version Trixbox 2.6.1.13 has support for
OpenR2, I verified in the repository that has to libraries of the project
openR2, but I don't manage to do to work in the trixbox, when I type the
command (it colors show channeltypes)ele no demostra the support to MFC+R2,
they could help finding out which package is necessary of the trixbox and
which the necessary configurations that should make!
I have been installing the trixbox version 2.6.1.13 and a Digium 110p, they
put in the trixbox only get to do to work in ISDN!
Thank you very much*
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Message: 10
Date: Sat, 29 Nov 2008 02:21:16 -0200
From: Yuri <yuri.asterisk at gmail.com>
Subject: [asterisk-users] Trixbox 2.6.1.13 OpenR2
To: asterisk-users at lists.digium.com
Message-ID:
<52068a5e0811282021g699b486ape62497ca948e1db2 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Good morning!
I verified that the trixbox version Trixbox 2.6.1.13 has support for OpenR2,
I verified in the repository that has to libraries of the project openR2,
but I don't manage to do to work in the trixbox, when I type the command
(show channeltypes) he doesn't demonstrate the support to MFC+R2, they could
help finding out which package is necessary of the trixbox and which the
necessary configurations that should make!
I have been installing the trixbox version 2.6.1.13 and a Digium 110p, they
put in the trixbox only get to do to work in ISDN!
Thank you very much
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Message: 11
Date: Fri, 28 Nov 2008 23:43:22 -0600
From: Doug <Doug at NaTel.net>
Subject: [asterisk-users] libspandsp.so.0: cannot open shared object
file: No such file or directory
To: asterisk-users at lists.digium.com
Message-ID: <984185085-267834602 at mail1.natel.net>
Content-Type: text/plain; charset="us-ascii"; format=flowed
libspandsp.so.0: cannot open shared object file: No such file or directory
Created the symlink:
/usr/local/lib# ls -lt lib*
lrwxrwxrwx 1 root staff 19 2008-11-28 22:42 libspandsp.so.0 ->
libspandsp.so.1.0.0
-rw-r--r-- 1 root staff 1849266 2008-11-13 13:26 libspandsp.a
-rwxr-xr-x 1 root staff 865 2008-11-13 13:26 libspandsp.la
lrwxrwxrwx 1 root staff 19 2008-11-13 13:26 libspandsp.so ->
libspandsp.so.1.0.0
lrwxrwxrwx 1 root staff 19 2008-11-13 13:26 libspandsp.so.1 ->
libspandsp.so.1.0.0
-rwxr-xr-x 1 root staff 1433877 2008-11-13 13:26 libspandsp.so.1.0.0
Edited /etc/ld.so.conf:
# Begin ------ /etc/ld.so.conf
include /etc/ld.so.conf.d/*.conf
/usr/local/lib
# End: ------- /etc/ld.so.conf
Googled the heck out of it:
<http://www.google.com/search?q=libspandsp.so.0:+cannot+open+shared+object>
Still can't find the answer. Any ideas?
------------------------------
Message: 12
Date: Sat, 29 Nov 2008 12:12:56 +0600
From: "Peter Lindquist" <peter.lindquist.th at gmail.com>
Subject: Re: [asterisk-users] Trixbox 2.6.1.13 OpenR2
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID: <b7e687c20811282212ta58e5q23e47122ee8455f7 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
On Sat, Nov 29, 2008 at 10:18 AM, Yuri <yuri.asterisk at gmail.com> wrote:
> *Good morning! *
>
> *I verified that the trixbox version Trixbox 2.6.1.13 has support for
> OpenR2, I verified in the repository that has to libraries of the project
> openR2, but I don't manage to do to work in the trixbox, when I type
the
> command (it colors show channeltypes)ele no demostra the support to MFC+R2,
> they could help finding out which package is necessary of the trixbox and
> which the necessary configurations that should make!
> I have been installing the trixbox version 2.6.1.13 and a Digium 110p,
> they put in the trixbox only get to do to work in ISDN!
>
> Thank you very much*
>
>
> Hi Yuri,
I also read that 2.6.1.13 would have OpenR2 support built in but found that
this was not entirely true. The library package is in the repository, but
support for OpenR2 is not in the provided Asterisk package. I ended up
downloading the source and recompiling from the OpenR2 site.
//Peter
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Message: 13
Date: Sat, 29 Nov 2008 01:14:58 -0500
From: Alex Balashov <abalashov at evaristesys.com>
Subject: Re: [asterisk-users] libspandsp.so.0: cannot open shared
object file: No such file or directory
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <4930DDE2.7040704 at evaristesys.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Paste 'ldd /usr/sbin/asterisk'.
Doug wrote:
> libspandsp.so.0: cannot open shared object file: No such file or directory
>
> Created the symlink:
>
> /usr/local/lib# ls -lt lib*
> lrwxrwxrwx 1 root staff 19 2008-11-28 22:42 libspandsp.so.0 ->
> libspandsp.so.1.0.0
> -rw-r--r-- 1 root staff 1849266 2008-11-13 13:26 libspandsp.a
> -rwxr-xr-x 1 root staff 865 2008-11-13 13:26 libspandsp.la
> lrwxrwxrwx 1 root staff 19 2008-11-13 13:26 libspandsp.so ->
> libspandsp.so.1.0.0
> lrwxrwxrwx 1 root staff 19 2008-11-13 13:26 libspandsp.so.1 ->
> libspandsp.so.1.0.0
> -rwxr-xr-x 1 root staff 1433877 2008-11-13 13:26 libspandsp.so.1.0.0
>
>
> Edited /etc/ld.so.conf:
>
> # Begin ------ /etc/ld.so.conf
>
> include /etc/ld.so.conf.d/*.conf
>
> /usr/local/lib
>
> # End: ------- /etc/ld.so.conf
>
>
> Googled the heck out of it:
>
<http://www.google.com/search?q=libspandsp.so.0:+cannot+open+shared+object>
>
> Still can't find the answer. Any ideas?
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
------------------------------
Message: 14
Date: Sat, 29 Nov 2008 12:51:57 +0530
From: "Max Alex" <max.asterisk at gmail.com>
Subject: Re: [asterisk-users] Anonymous callerid
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Cc: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Message-ID:
<c8ba87ad0811282321n1077c629td0f29b46b5929eb1 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi
Thanks for your reply.
Actully we are getting the anonymous callerid from the originated phone
(blocked from phone) so we need to override the callerid and then pass to
network.
we need to send out caller id. That is why we tried to override it.
But we are not able to override it.
Please help for this!
Thanks,
Max Alex
Voip Developer
On Fri, Nov 28, 2008 at 7:47 PM, Philipp Kempgen
<philipp.kempgen at amooma.de>wrote:
> Max Alex schrieb:
>
> > I have one issue regarding override callerid when i have anonymous
call.
> > I have added PAI in sip header and also set sendrpid = yes in sip.conf
> > but the callerid is not overriding while i am sending call to three
digit
> > calling like 911.
>
> The caller ID sent to emergency or law enforcement numbers is
> network-provided not user-provided so you can't override it.
>
> Philipp Kempgen
>
> --
> http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com
> Amooma GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de
> Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
> --
>
> _______________________________________________
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Message: 15
Date: Sat, 29 Nov 2008 13:19:19 +0000
From: Julian Lyndon-Smith <asterisk at dotr.com>
Subject: [asterisk-users] GSM gateways - which one ?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <49314157.4040703 at dotr.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
I've been asked to purchase a gsm gateway for use with our asterisk
server (for our use, not reselling)
I have a spare ISDN port on the server, so I have use either a PRI or
VOIP gsm gateway.
What would people recommend ? Has anyone used the QuesCom 400 ?
I would also love to know a rough idea of cost ;)
Once I've gotten the info, I'll post a message on the biz list for a
quotation.
Thanks
Julian.
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Message: 16
Date: Sat, 29 Nov 2008 14:57:02 +0000
From: Julian Lyndon-Smith <asterisk at dotr.com>
Subject: Re: [asterisk-users] GSM gateways - which one ?
To: Gordon Henderson <gordon at drogon.net>
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <4931583E.6000707 at dotr.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Thanks Gordon,
I have been playing with the Portech, but was wanting a "larger"
solution (20+ channels)
Julian.
Gordon Henderson wrote:> On Sat, 29 Nov 2008, Julian Lyndon-Smith wrote:
>
>> I've been asked to purchase a gsm gateway for use with our asterisk
>> server (for our use, not reselling)
>>
>> I have a spare ISDN port on the server, so I have use either a PRI or
>> VOIP gsm gateway.
>>
>> What would people recommend ? Has anyone used the QuesCom 400 ?
>>
>> I would also love to know a rough idea of cost ;)
>>
>> Once I've gotten the info, I'll post a message on the biz list
for a
>> quotation.
>
> Have had good results with Porech ones & Guessing you're in the UK
> from whois on the domain name, so:
>
> ?130:
>
http://www.voipon.co.uk/portech-gsm-gateways-portech-voip-gsm-gateway-c-3_192_193.html
>
> or
> ?125:
>
http://www.discountphonesystems.co.uk/acatalog/Portech_VoIP_GSM_Gateways.html
>
>
> Ethernet+SIP in, GSM out...
>
> (Wait until Monday when the VAT rate drops ... I bet this weekend is
> going to be a pi$$ poor shopping weekend!!!)
>
> (and I don't work for either those companies, just use them for
hardware)
>
> Gordon
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Message: 17
Date: Sat, 29 Nov 2008 09:06:41 -0600
From: "Michael Graves" <mgraves at mstvp.com>
Subject: Re: [asterisk-users] GSM gateways - which one ?
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>, "asterisk at
dotr.com"
<asterisk at dotr.com>
Message-ID: <200811291506.mATF6hKp008160 at mail818.megamailservers.com>
Content-Type: text/plain; charset="iso-8859-1"
Portech makes larger rack mounted modular multi-channel gateways as
well. Not sure about the ISDN interface, but certainly with T-1/E-1
PRI.
Michael
On Sat, 29 Nov 2008 14:57:02 +0000, Julian Lyndon-Smith wrote:
>Thanks Gordon,
>
>I have been playing with the Portech, but was wanting a "larger"
>solution (20+ channels)
>
>Julian.
>
>Gordon Henderson wrote:
>> On Sat, 29 Nov 2008, Julian Lyndon-Smith wrote:
>>
>>> I've been asked to purchase a gsm gateway for use with our
asterisk
>>> server (for our use, not reselling)
>>>
>>> I have a spare ISDN port on the server, so I have use either a PRI
or
>>> VOIP gsm gateway.
>>>
>>> What would people recommend ? Has anyone used the QuesCom 400 ?
>>>
>>> I would also love to know a rough idea of cost ;)
>>>
>>> Once I've gotten the info, I'll post a message on the biz
list for a
>>> quotation.
>>
>> Have had good results with Porech ones & Guessing you're in the
UK
>> from whois on the domain name, so:
>>
>> ?130:
>>
http://www.voipon.co.uk/portech-gsm-gateways-portech-voip-gsm-gateway-c-3_192_193.html
>>
>> or
>> ?125:
>>
http://www.discountphonesystems.co.uk/acatalog/Portech_VoIP_GSM_Gateways.html
>>
>>
>> Ethernet+SIP in, GSM out...
>>
>> (Wait until Monday when the VAT rate drops ... I bet this weekend is
>> going to be a pi$$ poor shopping weekend!!!)
>>
>> (and I don't work for either those companies, just use them for
hardware)
>>
>> Gordon
>
>
>______________________________________________________________________
>This email has been scanned by the MessageLabs Email Security System.
>For more information please visit http://www.messagelabs.com/email
>______________________________________________________________________
>
>_______________________________________________
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>
--
Michael Graves
mgraves<at>mstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:mjgraves at pixelpower.onsip.com
skype mjgraves
fwd 54245
------------------------------
Message: 18
Date: Sat, 29 Nov 2008 11:26:30 -0600
From: Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
Subject: Re: [asterisk-users] Anonymous callerid
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <200811291126.30958.tilghman at mail.jeffandtilghman.com>
Content-Type: text/plain; charset="iso-8859-1"
On Friday 28 November 2008 08:17:24 Philipp Kempgen
wrote:> Max Alex schrieb:
> > I have one issue regarding override callerid when i have anonymous
call.
> > I have added PAI in sip header and also set sendrpid = yes in sip.conf
> > but the callerid is not overriding while i am sending call to three
digit
> > calling like 911.
>
> The caller ID sent to emergency or law enforcement numbers is
> network-provided not user-provided so you can't override it.
If only that were actually true. I have experience with a school which sent
no CallerID at all on 911 on this theory, and as a result, ambulance services
were delayed by a few minutes because the wrong 911 center was contacted
(different county). Luckily, the student (peanut allergy) survived and we
learned this valuable lesson.
--
Tilghman
------------------------------
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End of asterisk-users Digest, Vol 52, Issue 81
**********************************************
Apparently Analagous Threads
Wisdom of the Ancients
