Re: [asterisk-users] Voicemail ODBC

2010-06-21 Thread Andraž
Yes, with isql it's working fine, I can see the database and all fields. On Sun, Jun 20, 2010 at 9:11 PM, Tilghman Lesher tles...@digium.com wrote: On Sunday 20 June 2010 13:15:11 Andraž wrote: If I use MySQL with the same fields it's working fine. I think that is something wrong with

[asterisk-users] call dialing

2010-06-21 Thread niksinghania
Hi listusers, I am using call file to dial out the sip on a different machine. The problem is whenever i dial the call lands up on the softphone but i have to pick it up 2 times, for both line 1 and line 2. If i reject it in the 1st time only then both are rejected. channel: SIP/2001

[asterisk-users] features.conf - parkedcalls - transfer

2010-06-21 Thread Aksel Celasun
Hello dear list. I am having issues on parkedcalls. I am using a Cisco SPA525G as a test phone, and I have the transfer button there when I am in a call, But when I want to transfer the current call I am in, I push the transfer button, and onscreen I se Enter Number, and if I enter ex sip

Re: [asterisk-users] Compiling H323

2010-06-21 Thread Tzafrir Cohen
On Sun, Jun 20, 2010 at 07:55:07PM -0400, Michelle Dupuis wrote: And after cleaning everything 323/pwlib/ptlib related, and reinstalling ptlib and h323plus, I can't even get asterisk to compile chan_h323 anymore. Perhaps something old was left over. My .configure run shows: checking

Re: [asterisk-users] How to get asterisk to playback personal greetings using grandstream gxp-2000

2010-06-21 Thread Ishfaq Malik
On 18/06/10 20:22, Eddie Mikell wrote: All: I am using the standard voicemail in asterisk. Everything works well, except, if a users wants to record their own personal greeting, it doesn't playback. I can see the soundfile being created. I suspect it is a setting in the voicemail.conf, or an

Re: [asterisk-users] asterisk appache issue

2010-06-21 Thread pankaj pandey
thanks for reply. how can i give the root permission to apache ? sudo. i also tried sudo . However, without careful configuration you will probably be giving root access to any process that runs as your apache user. I've never done it, but I'm guessing you could create a

Re: [asterisk-users] Voicemail ODBC

2010-06-21 Thread Tilghman Lesher
On Monday 21 June 2010 01:16:30 Andraž wrote: Yes, with isql it's working fine, I can see the database and all fields. On Sun, Jun 20, 2010 at 9:11 PM, Tilghman Lesher tles...@digium.com wrote: On Sunday 20 June 2010 13:15:11 Andraž wrote: If I use MySQL with the same fields it's working

Re: [asterisk-users] Voicemail ODBC

2010-06-21 Thread Andraž
Now it's workin fine. It was problem with drivers, because it doesn't support all kind of fields. I just changed from varblog to picture data type and now it's working fine. Tnx for help. On Mon, Jun 21, 2010 at 2:08 PM, Tilghman Lesher tles...@digium.com wrote: On Monday 21 June 2010 01:16:30

[asterisk-users] ISP down internal phones become unavailable

2010-06-21 Thread Ryan Wagoner
I saw the following lines in the log this morning. From my router logs I see that the connection went down as my ISP was doing maintenance for a few minutes last night. I can understand the external registrations timing out, but why do the phones become unreachable. They are on the internal lan

[asterisk-users] [AGI] What scripting language for embedded hardware?

2010-06-21 Thread Gilles
Hello I'm learning how to work with Asterisk on an embedded system (MMU-less Blackfin processor, 64MB RAM and 256MB NAND), and was wondering what people use as scripting language to handle calls through the dialplan and AGI, considering the hardware limitations? Ideally, I'd rather use a rich

[asterisk-users] using call file

2010-06-21 Thread nikhil singhania
HI list-users, Greetings!! I have been using call file, i playback my file using * application:playback* and once the playback is over the call is disconnected. Is there any way it can wait and also record the dtmf inputs once the playback is over. Thanks in advace Nikhil Kumar summer

Re: [asterisk-users] using call file

2010-06-21 Thread Danny Nicholas
Use a context instead of the playback command. Like this [playit] exten = s,1,NoOp(Answer) exten = s,n,SetMusicOnHold(default) exten = s,n,Waitexten(5,m) exten = s,n,Verbose(play ${ARG1}) exten = s,n,Playback(${ARG1}) So you replace playback(file) with playit(file). _ From:

Re: [asterisk-users] [AGI] What scripting language for embedded hardware?

2010-06-21 Thread Gordon Henderson
On Mon, 21 Jun 2010, Gilles wrote: Hello I'm learning how to work with Asterisk on an embedded system (MMU-less Blackfin processor, 64MB RAM and 256MB NAND), and was wondering what people use as scripting language to handle calls through the dialplan and AGI, considering the hardware

Re: [asterisk-users] [AGI] What scripting language for embedded hardware?

2010-06-21 Thread Steve Edwards
On Mon, 21 Jun 2010, Gilles wrote: I'm learning how to work with Asterisk on an embedded system (MMU-less Blackfin processor, 64MB RAM and 256MB NAND), and was wondering what people use as scripting language to handle calls through the dialplan and AGI, considering the hardware

Re: [asterisk-users] [AGI] What scripting language for embedded hardware?

2010-06-21 Thread Gilles
On Mon, 21 Jun 2010 14:06:22 +0100 (BST), Gordon Henderson gordon+aster...@drogon.net wrote: You could always type asterisk blackfin into google and see what it suggests. Here, I'll save you the effort: Thanks but I already know this (uCasterisk is deprecated). And can't stand Perl ;-) --

Re: [asterisk-users] [AGI] What scripting language for embedded hardware?

2010-06-21 Thread Danny Nicholas
Even though I'm a PERL Weenie, I'll second this suggestion because you have to have gcc present for PERL or Micro PERL. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Monday, June 21, 2010

Re: [asterisk-users] [AGI] What scripting language for embedded hardware?

2010-06-21 Thread Motiejus Jakštys
If you can install python or PHP in that machine (in means of storage), you are free to run it there. 64 RAM is really enough to run python, so you have to just try if it suits in the application. If it takes too slow to initialize - try to find some embedded versions. openwrt, for instance, has

Re: [asterisk-users] Compiling H323

2010-06-21 Thread Michelle Dupuis
Yuk! I did manage to get it compiling again, but same error. I found an environment variable which makes the loader tell you what it's doing, and when I load chan_h323.so I see that it is running the init code when it segfaults. So I'm ok up to that point... What a mess the entire H.323 is!

Re: [asterisk-users] [AGI] What scripting language for embedded hardware?

2010-06-21 Thread Gilles
On Mon, 21 Jun 2010 17:25:09 +0300, Motiejus Jakštys desired@gmail.com wrote: If you can install python or PHP in that machine (in means of storage), you are free to run it there. 64 RAM is really enough to run python, so you have to just try if it suits in the application. If it takes too

Re: [asterisk-users] [AGI] What scripting language for embedded hardware?

2010-06-21 Thread Philipp von Klitzing
Hey Gilles, for whatever reason your messages appear twice twice on this list. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every

[asterisk-users] Dialplan Gurus? Can Asterisk 1.4x CHANNEL function be used to retrieve info about OTHER channels?

2010-06-21 Thread bruce bruce
Hi Everyone, I want to know if a specific codec type is used at least one. For example, I want to know if out of the 100 calls on the system if there is a 1 channel that is running G.729 codec right now. If using dial-plan and I dial in, I can use this to obtain information about CURRENT channel.

Re: [asterisk-users] features.conf - parkedcalls - transfer

2010-06-21 Thread Warren Selby
On Mon, Jun 21, 2010 at 2:27 AM, Aksel Celasun ak...@abacus-it.no wrote: I am using a Cisco SPA525G as a test phone, and I have the transfer button there when I am in a call, But when I want to transfer the current call I am in, I push the transfer button, and onscreen I se “Enter Number”,

[asterisk-users] Switchboad like application

2010-06-21 Thread voip crazy
Hello all, Anybody could point me any clue about an Open Source or licensed switchboard for my users? ARI or FOP is not enought for my users. Thanks in advance. VoipCrazy -- _ -- Bandwidth and Colocation Provided by

[asterisk-users] What is the voicemail u option

2010-06-21 Thread Eddie Mikell
All: Still trying to get Grandstream to play personal greetings recorded by user - no luck. Someone mentioned the u option. What is that? Something in voicemail.conf? Eddie -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] What is the voicemail u option

2010-06-21 Thread Danny Nicholas
RTFM - B = busy, U = unavailable, S = Silent. User can record custom busy and unavailable messages; if you use S you can just playback any kind of message you want. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf

Re: [asterisk-users] What is the voicemail u option

2010-06-21 Thread Doug Lytle
Eddie Mikell wrote: user - no luck. Someone mentioned the u option. What is that? core show application voicemail Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. --

[asterisk-users] DAHDI: Inbound BRI call, DDI not presented

2010-06-21 Thread Scott Stingel
Hello- I have a system with one D410P and one B200P (both OpenVox). All is well with the D410P, inbound and outbound, and I can initiate calls on the B200P BRI span, but there may be something wrong with my inbound BRI setup: there is no indication of an inbound call when I dial in to it

Re: [asterisk-users] [AGI] What scripting language for embedded hardware?

2010-06-21 Thread Edwin Quijada
Uhmmm.. remember for each channel you run perl or php interpreter so with that amount of memory maybe this can be a problem. For that kind of project I'd use C or java as fastagi protocol From: desired@gmail.com Date: Mon, 21 Jun 2010 17:25:09 +0300 To: asterisk-users@lists.digium.com

[asterisk-users] Create Conference and exit myself

2010-06-21 Thread RSCL Mumbai
Hi, I am using Trixbox trixbox CE 2.6.2.3 (Stable) using Asterisk 1.4.22-4 I am looking for the following functionality: `` I receive a call from Mr. A. I put Mr. A on hold. I dial Mr. B I connect Mr. A's

[asterisk-users] How to find a single call in logs

2010-06-21 Thread Douglas Mortensen
Hello everyone. I am wondering whether there is a certain technique I should use to identify all log lines in the asterisk/full logfile that are related to a single call. If a user reports that something strange happened with a certain call, I'd like to be able to easily go back and look at

[asterisk-users] Polycom firmware: split vs. combined

2010-06-21 Thread Ken D'Ambrosio
http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html Howdy, all. What's the difference between split and combined firmware, which can be seen at the above link? I've googled to no avail, I'm afraid. Thanks! -Ken -- This message has been scanned for viruses and dangerous

Re: [asterisk-users] features.conf - parkedcalls - transfer

2010-06-21 Thread Ira
At 12:27 AM 6/21/2010, you wrote: Almost 10 seconds, before the transfer to sip 200 is made, can I reduce that timer? And I can’t see any button on the Cisco phone which will function like a “direct transfer now”, do I have to wait…? On my Aastra phones, I press Transfer 101 Transfer. So

Re: [asterisk-users] Polycom firmware: split vs. combined

2010-06-21 Thread Warren Selby
On Mon, Jun 21, 2010 at 12:10 PM, Ken D'Ambrosio k...@jots.org wrote: http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.htmlHowdy, all. What's the difference between split and combined firmware, which can be seen at the above link? I've googled to no avail, I'm afraid. The

Re: [asterisk-users] Polycom firmware: split vs. combined

2010-06-21 Thread Jimmy Godbout
From polycom web site: PLEASE NOTE: Combined download should be used where phones may be running pre-4.0 BootROM. Split download file is recommended, but requires that all phones are running BootROM 4.0 or newer. -Original Message- From: k...@jots.org Sent: Mon, 21 Jun 2010

Re: [asterisk-users] Polycom firmware: split vs. combined

2010-06-21 Thread John Balogh
Read: http://downloads.polycom.com/voice/voip/relnotes/spip_ssip_3_1_2_relnote s.pdf starting page 11/88: 1.4 Distribution Files 1.4.1 Release using individual (split) files 1.4.2 Release using Combined Image HTH, JDB -Original Message- From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] Polycom firmware: split vs. combined

2010-06-21 Thread Jonathan Thurman
On Mon, Jun 21, 2010 at 10:19 AM, Warren Selby wcse...@selbytech.com wrote: On Mon, Jun 21, 2010 at 12:10 PM, Ken D'Ambrosio k...@jots.org wrote: Howdy, all.  What's the difference between split and combined firmware, which can be seen at the above link?  I've googled to no avail, I'm afraid.

[asterisk-users] How to tell if a dropped call is my fault

2010-06-21 Thread Douglas Mortensen
, recordingcheck|20100621-085328|1277132008.6419) in new stack [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck [Jun 21 08:53:29] VERBOSE[21559] logger.c: recordingcheck|20100621-085328|1277132008.6419: Outbound recording not enabled

Re: [asterisk-users] How to tell if a dropped call is my fault

2010-06-21 Thread dotnetdub
:28] VERBOSE[21559] logger.c:     -- Goto (macro-record-enable,s,4) [Jun 21 08:53:28] DEBUG[21559] app_macro.c: Executed application: GotoIf [Jun 21 08:53:28] VERBOSE[21559] logger.c:     -- Executing [...@macro-record-enable:4] AGI(SIP/611-b7b9ae38, recordingcheck|20100621-085328

[asterisk-users] when to use e1/t1 card?

2010-06-21 Thread Necati Demir
This is a really rookie question: when should i use TE110P ISDN PRI Card? -- Necati DEMİR --- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join

Re: [asterisk-users] when to use e1/t1 card?

2010-06-21 Thread David Backeberg
On Mon, Jun 21, 2010 at 3:04 PM, Necati Demir nde...@demir.web.tr wrote: This is a really rookie question: when should i use TE110P ISDN PRI Card? -- Necati DEMİR When you have a single PRI / BRI line you wish to terminate into an asterisk system. --

Re: [asterisk-users] when to use e1/t1 card?

2010-06-21 Thread Tzafrir Cohen
On Mon, Jun 21, 2010 at 03:12:40PM -0400, David Backeberg wrote: On Mon, Jun 21, 2010 at 3:04 PM, Necati Demir nde...@demir.web.tr wrote: This is a really rookie question: when should i use TE110P ISDN PRI Card? When you have a single PRI / BRI line you wish to terminate into an asterisk

Re: [asterisk-users] when to use e1/t1 card?

2010-06-21 Thread Miguel Molina
El 21/06/10 14:04, Necati Demir escribió: This is a really rookie question: when should i use TE110P ISDN PRI Card? -- Necati DEMİR --- When you need to... -- _ -- Bandwidth and

Re: [asterisk-users] when to use e1/t1 card?

2010-06-21 Thread Steve Edwards
On Mon, 21 Jun 2010, Necati Demir wrote: This is a really rookie question: when should i use TE110P ISDN PRI Card? From an economic standpoint? When you have more than x POTS lines where x depends on where you are in the world. Generally speaking, somewhere around 8 to 12. There are many

Re: [asterisk-users] ISP down internal phones become unavailable

2010-06-21 Thread Lacy Moore
On Mon, Jun 21, 2010 at 7:32 AM, Ryan Wagoner rswago...@gmail.com wrote: I saw the following lines in the log this morning. From my router logs I see that the connection went down as my ISP was doing maintenance for a few minutes last night. I can understand the external registrations timing

Re: [asterisk-users] features.conf - parkedcalls - transfer

2010-06-21 Thread Aksel Celasun
And I can't see any button on the Cisco phone which will function like a direct transfer now, do I have to wait...? Thank you for your reply. In my Dialplan menu on the SPA525g, I have a field where the input are, and I must say, I don't know if this is the right one, but the field contains

Re: [asterisk-users] features.conf - parkedcalls - transfer

2010-06-21 Thread Aksel Celasun
Hello, and thank you for your response. When I push transfer, the buttons with the function transfer disappears, and then I enter the sip number, Wait 10 seconds and then it transfers with the MOH in the background, when the connection/channel is made, Then transfer button is revealed again

[asterisk-users] How do I access the Dialstatus numeric code received?

2010-06-21 Thread CDR
I need to access number received after a I dial a SIP or H323 call? suppose I get one of these: *404 Not found **486 Busy here **408 Request Timeout **480 Temporarily unavailable **480 Temporarily unavailable **403 Forbidden (+) ** 410 Gone **301 Moved Permanently **410 Gone ** 404 Not Found (=)

Re: [asterisk-users] Create Conference and exit myself

2010-06-21 Thread Zeeshan Zakaria
I am sure you'll have to write your own dialplan for it in asterisk. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-21 12:31 PM, RSCL Mumbai rscl.mum...@gmail.com wrote: Hi, I am using Trixbox trixbox CE 2.6.2.3 (Stable) using Asterisk 1.4.22-4 I am looking for the following

Re: [asterisk-users] How to find a single call in logs

2010-06-21 Thread Zeeshan Zakaria
Every call is assigned a unique SIP channel id. I usually look for this id and then grep the log file by this id. It looks something like SIP/201-a08rfr7... if I remember correctly. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-21 1:13 PM, Douglas Mortensen d...@impalanetworks.com wrote:

[asterisk-users] AMD

2010-06-21 Thread Tetra Informatica
Hi I am using the AMD application in a power dialing. All works well when I use an internal extension but when I try to use an external number, the AMD every times returns non human status. Also the AMDCAUSE returns Total-Time-5500. I am using the default parameters in AMD.CONF. Anybody has

Re: [asterisk-users] Create Conference and exit myself

2010-06-21 Thread Paul Belanger
On Mon, Jun 21, 2010 at 12:25 PM, RSCL Mumbai rscl.mum...@gmail.com wrote: What is the simplest way to achieve this ?? Use the transfer button on your phone? -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com

Re: [asterisk-users] How do I access the Dialstatus numeric code received?

2010-06-21 Thread Tilghman Lesher
On Monday 21 June 2010 16:09:22 CDR wrote: I need to access number received after a I dial a SIP or H323 call? suppose I get one of these: *404 Not found **486 Busy here **408 Request Timeout **480 Temporarily unavailable **480 Temporarily unavailable **403 Forbidden (+) ** 410 Gone

Re: [asterisk-users] Is this failed Asterisk setup typical?

2010-06-21 Thread CunningPike
On Thu, Jun 3, 2010 at 6:16 AM, Gilles codecompl...@free.fr wrote: Hello        I just read this article and would like some feedback from experienced Asterisk users: === Failed open source VoIP deployment leads to hosted VoIP strategy By Jessica Scarpati snip

Re: [asterisk-users] Is this failed Asterisk setup typical?

2010-06-21 Thread Gilles
On Mon, 21 Jun 2010 16:47:08 -0700, CunningPike cunningp...@gmail.com wrote: Not in our experience as a 500-phone, 20-site install for a municipal government. We are just migrating from our first generation install to replacement hardware (to new blades from servers that are now 5 years old) and

Re: [asterisk-users] AMD

2010-06-21 Thread John Rose
Sometimes you have to play some audio before calling AMD or other audio functions for whatever reason... Like play 100ms of silence in a .wav file immediately after answer. This causes RTP to be sent out to the carrier. John From: asterisk-users-boun...@lists.digium.com

[asterisk-users] Update to chan_ooh323 wrapper

2010-06-21 Thread Michelle Dupuis
I see that objective systems has updated their ooh323 stack, but it is not compatible with the latest chan_ooh323 wrapper available on their site. Has anyone update the chan_ooh323 wrapper for Asterisk 1.6.2.x ? Michelle -- _

[asterisk-users] Can't make calls out of Astribank - show regdump doesn't show voltage - Getting warning due to old driver

2010-06-21 Thread bruce bruce
Hi Guys, An 8 channel Astribank is connected to Trixbox 2.8 and I ran freepbx-module-zapauto but I get the following when running these commands and can't make calls out: [Trixbox]# dahdi_genconf xpporder /usr/sbin/dahdi_genconf: warning - OLD DRIVER: missing