[asterisk-users] call dialing

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 CallerId:

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

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 20

Re: [asterisk-users] Compiling H323

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

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 o

Re: [asterisk-users] asterisk appache issue

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 cre

Re: [asterisk-users] Voicemail ODBC

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 wrote: > > On Sunday 20 June 2010 13:15:11 Andraž wrote: > > > If I use MySQL with the same fields it's working fine. I thin

Re: [asterisk-users] Voicemail ODBC

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 wrote: > On Monday 21 June 2010 01:16:30 Andraž wrote: > >

[asterisk-users] ISP down internal phones become unavailable

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 wit

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

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 lan

[asterisk-users] using call file

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 intern:si

Re: [asterisk-users] using call file

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). _ F

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

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 l

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

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 limitati

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

On Mon, 21 Jun 2010 14:06:22 +0100 (BST), Gordon Henderson 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?

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 8

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

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 on

Re: [asterisk-users] Compiling H323

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?

On Mon, 21 Jun 2010 17:25:09 +0300, Motiejus Jakštys 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 slow to initialize -

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

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 T

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

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

On Mon, Jun 21, 2010 at 2:27 AM, Aksel Celasun 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”, and if I enter

[asterisk-users] Switchboad like application

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 http://www

[asterisk-users] What is the voicemail "u option"

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 ht

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

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 Behal

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

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

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 fr

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

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.c

[asterisk-users] Create Conference and exit myself

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

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 the

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

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 c

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

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. S

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

On Mon, Jun 21, 2010 at 12:10 PM, Ken D'Ambrosio 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. > The split

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

>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 13:1

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

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 [m

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

On Mon, Jun 21, 2010 at 10:19 AM, Warren Selby wrote: > On Mon, Jun 21, 2010 at 12:10 PM, Ken D'Ambrosio 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. The release notes ta

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

o("SIP/611-b7b9ae38", "record-enable|611|OUT|") in new stack [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Executing [...@macro-record-enable:1] GotoIf("SIP/611-b7b9ae38", "1?check") in new stack [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Goto (m

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

Set("SIP/611-b7b9ae38", "_NODEST=") > in new stack > [Jun 21 08:53:28] VERBOSE[21559] logger.c:     -- Executing > [5053203...@from-internal-ntc-custom:3] Macro("SIP/611-b7b9ae38", > "record-enable|611|OUT|") in new stack > [Jun 21 08:53:28] VERBOSE

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

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 u

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

On Mon, Jun 21, 2010 at 3:04 PM, Necati Demir 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?

On Mon, Jun 21, 2010 at 03:12:40PM -0400, David Backeberg wrote: > On Mon, Jun 21, 2010 at 3:04 PM, Necati Demir 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 system. It's no

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

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 Colo

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

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 ad

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

On Mon, Jun 21, 2010 at 7:32 AM, Ryan Wagoner 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 out, but why do t

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

>>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 conta

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

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 su

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

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

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" 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 functionality:

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

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" wrote: Hello everyone. I am

[asterisk-users] AMD

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 s

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

On Mon, Jun 21, 2010 at 12:25 PM, RSCL Mumbai 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?

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

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

On Thu, Jun 3, 2010 at 6:16 AM, Gilles 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 > > http://searchunifiedcommun

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

On Mon, 21 Jun 2010 16:47:08 -0700, CunningPike 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 are still committed

Re: [asterisk-users] AMD

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 [mailto:asterisk-use

[asterisk-users] Update to chan_ooh323 wrapper

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

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 '/sys/bus/xpds/devices/00:0:0/timing_