I was cleaning and working on laptops most of the day. Check my logs, I did plenty of work.
-----Original Message----- From: "[EMAIL PROTECTED]" <[EMAIL PROTECTED]> To: "[email protected]" <[email protected]> Sent: 11/29/2008 1:13 PM Subject: asterisk-users Digest, Vol 52, Issue 81 Send asterisk-users mailing list submissions to [email protected] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than "Re: Contents of asterisk-users digest..." 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 <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Any 1.6 SendFAX example ? To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> 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 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: This is a digitally signed message part. Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20081128/b8a75dd1/attachment-0001.pgp ------------------------------ Message: 2 Date: Fri, 28 Nov 2008 15:48:07 -0600 From: Anthony Messina <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] force channel hangup To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> 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 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: This is a digitally signed message part. Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20081128/a7a59093/attachment-0001.pgp ------------------------------ Message: 3 Date: Fri, 28 Nov 2008 16:18:57 -0600 From: "Danny Nicholas" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] force channel hangup To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <[email protected]> Message-ID: <[EMAIL PROTECTED]> 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] force channel hangup To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> 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: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] 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" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] force channel hangup To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <[email protected]> Message-ID: <[EMAIL PROTECTED]> 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] 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 <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] force channel hangup To: [email protected] Message-ID: <[EMAIL PROTECTED]> 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:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ------------------------------ Message: 7 Date: Sat, 29 Nov 2008 01:13:45 +0200 From: Tzafrir Cohen <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] force channel hangup To: [email protected] Message-ID: <[EMAIL PROTECTED]> 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:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ------------------------------ Message: 8 Date: Sat, 29 Nov 2008 11:02:43 +0800 From: "Sun xiaoshuang" <[EMAIL PROTECTED]> Subject: [asterisk-users] received wrong state events for originate command To: [email protected] Message-ID: <[EMAIL PROTECTED]> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081129/d3746b26/attachment-0001.htm ------------------------------ Message: 9 Date: Sat, 29 Nov 2008 02:18:50 -0200 From: Yuri <[EMAIL PROTECTED]> Subject: [asterisk-users] Trixbox 2.6.1.13 OpenR2 To: [email protected] Message-ID: <[EMAIL PROTECTED]> 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* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081129/1ab6aaee/attachment-0001.htm ------------------------------ Message: 10 Date: Sat, 29 Nov 2008 02:21:16 -0200 From: Yuri <[EMAIL PROTECTED]> Subject: [asterisk-users] Trixbox 2.6.1.13 OpenR2 To: [email protected] Message-ID: <[EMAIL PROTECTED]> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081129/9853fc5f/attachment-0001.htm ------------------------------ Message: 11 Date: Fri, 28 Nov 2008 23:43:22 -0600 From: Doug <[EMAIL PROTECTED]> Subject: [asterisk-users] libspandsp.so.0: cannot open shared object file: No such file or directory To: [email protected] Message-ID: <[EMAIL PROTECTED]> 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" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Trixbox 2.6.1.13 OpenR2 To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" On Sat, Nov 29, 2008 at 10:18 AM, Yuri <[EMAIL PROTECTED]> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081129/4d020b5f/attachment-0001.htm ------------------------------ Message: 13 Date: Sat, 29 Nov 2008 01:14:58 -0500 From: Alex Balashov <[EMAIL PROTECTED]> 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 <[email protected]> Message-ID: <[EMAIL PROTECTED]> 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" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Anonymous callerid To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Cc: Asterisk Developers Mailing List <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]>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 > -- > > _______________________________________________ > -- 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081129/977731ca/attachment-0001.htm ------------------------------ Message: 15 Date: Sat, 29 Nov 2008 13:19:19 +0000 From: Julian Lyndon-Smith <[EMAIL PROTECTED]> Subject: [asterisk-users] GSM gateways - which one ? To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> 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. ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email ______________________________________________________________________ ------------------------------ Message: 16 Date: Sat, 29 Nov 2008 14:57:02 +0000 From: Julian Lyndon-Smith <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] GSM gateways - which one ? To: Gordon Henderson <[EMAIL PROTECTED]> Cc: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> 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 ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email ______________________________________________________________________ ------------------------------ Message: 17 Date: Sat, 29 Nov 2008 09:06:41 -0600 From: "Michael Graves" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] GSM gateways - which one ? To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]>, "[EMAIL PROTECTED]" <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> 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 >______________________________________________________________________ > >_______________________________________________ >-- 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 > -- Michael Graves mgraves<at>mstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ------------------------------ Message: 18 Date: Sat, 29 Nov 2008 11:26:30 -0600 From: Tilghman Lesher <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Anonymous callerid To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> 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 ------------------------------ _______________________________________________ --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 End of asterisk-users Digest, Vol 52, Issue 81 ********************************************** _______________________________________________ -- 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
