Re: [asterisk-users] TCP Trigger on incoming call request

2011-05-09 Thread Daniel Isenmann
Thank for the hint. I will have a look into it. Daniel -Urspr√ľngliche Nachricht- Von: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von isr...@gmail.com Gesendet: Freitag, 6. Mai 2011 15:22 An: Asterisk Users Mailing List -

[asterisk-users] how to play music when dial fail or time out

2011-05-09 Thread John Wu
Hi all, I need to support this feature. When caller dial if the dial fail or no answer from the called number then play a music. So how to achieve that? Thanks! -- _ -- Bandwidth and Colocation Provided by

[asterisk-users] Slightly OT: Android phone as sip-gw?

2011-05-09 Thread Jay R. Worthington
Hi, i have some spare (read: Boss get's a new one every few month ;)) Android Phones laying around. Does someone know a way of using them as a mobile gateway for asterisk? I could not find any SIP-Gateway in the Market, and i don't think it's possible to use the GSM Audio directly with something

[asterisk-users] OUTBOUND CALLER ID

2011-05-09 Thread mahesh katta
Hi, THIS IS IN DUBAI. I am having PRI line with 100 DID's (00-99) and when we call to any landline or mobile number then it shows us our board number or pilot number (i.e 4663000 means 00).. As i give all the extensions a particular DID, so people from outside world can call them. The problem is

Re: [asterisk-users] OUTBOUND CALLER ID

2011-05-09 Thread A J Stiles
On Monday 09 May 2011, mahesh katta wrote: Hi, THIS IS IN DUBAI. I am having PRI line with 100 DID's (00-99) and when we call to any landline or mobile number then it shows us our board number or pilot number (i.e 4663000 means 00).. In the context through which outgoing calls are placed,

Re: [asterisk-users] Asterisk 1.8 latest branch safe for production ?

2011-05-09 Thread Ishfaq Malik
On Fri, 2011-05-06 at 20:21 +0100, --[ UxBoD ]-- wrote: Are you not seeing issues with *8 call pick up then ? Nope, I double checked it after seeing someone saying they had issues with it and it is fine on the installation I have. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161

[asterisk-users] RTP Path and t or T option

2011-05-09 Thread Dovid Bender
Hi, I have been away from the list for a bit so please forgive me if this has already been covered. From what I understand if I have the t or T option in my dial string then the RTP must go through Asterisk since we need to know if any DTMF was pressed. If both users and the server are not

Re: [asterisk-users] Slightly OT: Android phone as sip-gw?

2011-05-09 Thread Dovid Bender
Jay, AFAIK you can only use chan_datacard with a specific USB modem (I can not recall the name at the moment). I have tried it and it works well for both Audio and SMS messages. You may want to try chan_mobile which works over blue tooth. Regards, Dovid - Original Message -

Re: [asterisk-users] how to play music when dial fail or time out

2011-05-09 Thread Dovid Bender
John, You want to do it only after it fails ? If so you can do something like. Exten = _X., 1, Dial(SIP/${EXTEN}@PEER) Exten = _X., 2, GotoIf($[${DIALSTATUS} = ANSWER]?10) Exten = _X., 4, MusicOnHold() Exten = _X., 10, Hangup - Original Message - From: John Wu To: Asterisk

Re: [asterisk-users] Blacklist with *30

2011-05-09 Thread Dovid Bender
Alejandro, What GUI are you using ? I don't think Asterisk comes with *30 to ban calls. Regards, Dovid - Original Message - From: Alejandro Cabrera Obed aco1...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May

Re: [asterisk-users] OUTBOUND CALLER ID

2011-05-09 Thread mahesh katta
Sir , this is not working On Mon, May 9, 2011 at 1:52 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote: On Monday 09 May 2011, mahesh katta wrote: Hi, THIS IS IN DUBAI. I am having PRI line with 100 DID's (00-99) and when we call to any landline or mobile number then it shows us

Re: [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?

2011-05-09 Thread Sebastian Arcus
Oh yeah - love your idea :-) So just to clarify - I take it the Cisco phones (at least the 7940) are supposed to be run with a tftp server available at all time - not only during the initial configuration? Just making sure I didn't miss something obvious in the documentation. Can somebody

Re: [asterisk-users] Slightly OT: Android phone as sip-gw?

2011-05-09 Thread randulo
On Mon, May 9, 2011 at 9:47 AM, Jay R. Worthington jayrworthing...@gmail.com wrote: gateway for asterisk? I could not find any SIP-Gateway in the Market, and i Portech has made GSM and CDMA gateways for years - nothing that works with your old Android phones, though.

Re: [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?

2011-05-09 Thread Doug Lytle
Sebastian Arcus wrote: Cisco phones (at least the 7940) are supposed to be run with a tftp server available at all time That is my experience. But, if you're running tftp under Linux, then it's probably spawned by xinetd and won't be running unless the service is requested. Doug -- Ben

Re: [asterisk-users] Asterisk 1.8 latest branch safe for production ?

2011-05-09 Thread Alec Davis
Are you not seeing issues with *8 call pick up then ? -- Thanks, Phil https://reviewboard.asterisk.org/r/1185/ helps with *8 pickup issues, particulary when you have pickupsounds enabled. Alec -- _ -- Bandwidth and

Re: [asterisk-users] Asterisk 1.8 latest branch safe for production ?

2011-05-09 Thread Satish Patel
Thanks to all for reply, I have already put 1.8 in production. Actually we are using basic function so I hope we are good and fingurs cross. -- Sent from my iPhone On May 9, 2011, at 7:18 AM, Alec Davis siva...@paradise.net.nz wrote: Are you not seeing issues with *8 call pick up then ?

Re: [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?

2011-05-09 Thread Sebastian Arcus
On 05/09/2011 12:02 PM, Doug Lytle wrote: Sebastian Arcus wrote: Cisco phones (at least the 7940) are supposed to be run with a tftp server available at all time That is my experience. But, if you're running tftp under Linux, then it's probably spawned by xinetd and won't be running unless

[asterisk-users] OT - Which Android handset with Wifi-only ?

2011-05-09 Thread Olivier
Hi, I would be curious to play with an Android phone with Wifi-only capability. My plan is to install Bria on it and see if it could be used within a couple of WiFi access points, as a high-end wireless phone. Which handset would you recommend ? Regards --

Re: [asterisk-users] Slightly OT: Android phone as sip-gw?

2011-05-09 Thread aster...@ck-lee.com
Chan_datacard can selected model of huawei usb modem for voice and sms. Chan_mobile on the other hand use bluetooth connection for voice. I have not tried sms. For mobile phone, it seeMs nokia is quite good. --- Sent with System SEVEN - the new generation of mobile messaging -original

Re: [asterisk-users] Slightly OT: Android phone as sip-gw?

2011-05-09 Thread Sebastian Arcus
On 05/09/2011 08:47 AM, Jay R. Worthington wrote: Hi, i have some spare (read: Boss get's a new one every few month ;)) Android Phones laying around. Does someone know a way of using them as a mobile gateway for asterisk? I could not find any SIP-Gateway in the Market, and i don't think it's

Re: [asterisk-users] OT - Which Android handset with Wifi-only ?

2011-05-09 Thread mgraves
Lots of Android handsets support wifi, like my G2, aka HTC DesireZ. Michael Graves mgraves mstvp.com o(713) 861-4005 c(713) 201-1262 sip:mjgra...@mstvp.onsip.com skype mjgraves Original Message Subject: [asterisk-users] OT - Which Android handset with Wifi-only ? From:

[asterisk-users] 40sec between dial execution and sending SIP request

2011-05-09 Thread Pezhman Lali
Dear I have a small pbx with asterisk 1.6.2.16. I have a funny problem, there is exactly 40sec between dial execution and sending first invite packet on sip. do you have any idea where the problem is ? Best regards -- Pezhman Lali --

Re: [asterisk-users] Blacklist with *30

2011-05-09 Thread Alejandro Cabrera Obed
Dear Dovis, I'm using Elastix and the dialplan comes with this line: *30,1,Goto(app-blacklist-add,s,1) Any idea ??? Thanks a lot. 2011/5/9 Dovid Bender asteriskus...@dovid.net: Alejandro, What GUI are you using ? I don't think Asterisk comes with *30 to ban calls. Regards, Dovid -

Re: [asterisk-users] OT - Which Android handset with Wifi-only ?

2011-05-09 Thread randulo
On Mon, May 9, 2011 at 2:20 PM, mgra...@mstvp.com wrote: Lots of Android handsets support wifi, like my G2, aka HTC DesireZ. Wouldn't ANY modern one have wifi? That would be odd if it didn't, would it not? :r -- _ --

Re: [asterisk-users] Blacklist with *30

2011-05-09 Thread Dovid Bender
Try the Elastix forums. - Original Message - From: Alejandro Cabrera Obed aco1...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, May 09, 2011 15:35 Subject: Re: [asterisk-users] Blacklist with *30 Dear Dovis, I'm

[asterisk-users] Ustream feed as MOH

2011-05-09 Thread Dovid Bender
Hi, Has anyone ever tried getting the Audio of ustream (ustream.tv) in to Asterisk for MOH ? Regards, Dovid -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] Background music during a call

2011-05-09 Thread Ioan Indreias
I have tested the following dialplan and it could be used as a starting point. What you have to resolve is how to generate different MeetMe conference room - in the example we have only one room = 1234 If you prefix the dialled extension with 1 = you will have a lovely chat. With 2 - cursing

[asterisk-users] Trying out a new version with sangoma card

2011-05-09 Thread Nicolas Ross
Hi ! We curently have a centos 5 / asterisk 1.4 server that we have some DTMF problems with. It has a Sangoma A104d card and only port one is used to connect to the PSTN. Port 2 is conencted via a cross-over cable to a RAS for modem access and port 3 is connected for data communication via

Re: [asterisk-users] Cannot install dahdi-linux on (old) PAE kernel.

2011-05-09 Thread stephen.hindmarch
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jason Parker Sent: 06 May 2011 20:01 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cannot install dahdi-linux

Re: [asterisk-users] Background music during a call

2011-05-09 Thread Ioan Indreias
Updated dialplan: fix a typo when using MOH variable and now you have truly dynamic conference rooms. Have fun, Ioan. + exten = _[12]XXX,1,Set(__MM=${EPOCH}) exten = _1XXX,n,Dial(SIP/${EXTEN:1},,G(chat-room,love,1)) exten =

Re: [asterisk-users] Trying out a new version with sangoma card

2011-05-09 Thread Jim Dickenson
Make sure the firmware on the card is latest. I had a problem, not like your, and flashing the card to the latest firmware resolved it. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On May 9, 2011, at 6:11 AM, Nicolas Ross wrote: Hi ! We curently have a centos 5 /

Re: [asterisk-users] how to play music when dial fail or time out

2011-05-09 Thread Enrico Cicconi
Hi, have you tried to manage all with dialplane ? just an example: [incoming] **exten = s,1,Dial (SIP/your_called_party,20) exten = s,n, Playback(music_message) . In the first step the call is redirect to the configured called party and if without answer (busy, not logged, not answered)

Re: [asterisk-users] Trying out a new version with sangoma card

2011-05-09 Thread Nicolas Ross
Make sure the firmware on the card is latest. I had a problem, not like your, and flashing the card to the latest firmware resolved it. I did the upgrade, I will make another test when appropriate. I will also upgrade my curent card, I am curent at version 25, wich dates 2007, it might solve

[asterisk-users] asterisk syntax highlighting for gedit

2011-05-09 Thread Naomi Rosenberg
Hi, Apologies if this is a duplicate - been having mail server issues and I don't think I managed to send it when I tried this morning. It seems there is no .conf syntax highlighting script available for gedit. I'm thinking of putting one together myself, but don't want to reinvent the wheel.

Re: [asterisk-users] asterisk syntax highlighting for gedit

2011-05-09 Thread Alex Balashov
On 05/09/2011 10:32 AM, Naomi Rosenberg wrote: So I'm just enquiring if anyone knows of one that already exists that i've missed. Not to inflame editor-related passions, but vim does quite a good job. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200

Re: [asterisk-users] asterisk syntax highlighting for gedit

2011-05-09 Thread Ishfaq Malik
On Mon, 2011-05-09 at 15:32 +0100, Naomi Rosenberg wrote: Hi, Apologies if this is a duplicate - been having mail server issues and I don't think I managed to send it when I tried this morning. It seems there is no .conf syntax highlighting script available for gedit. I'm thinking of

Re: [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?

2011-05-09 Thread Warren Selby
On Mon, May 9, 2011 at 7:07 AM, Sebastian Arcus s...@open-t.co.uk wrote: On 05/09/2011 12:02 PM, Doug Lytle wrote: Sebastian Arcus wrote: Cisco phones (at least the 7940) are supposed to be run with a tftp server available at all time That is my experience. But, if you're running tftp

Re: [asterisk-users] 40sec between dial execution and sending SIP request

2011-05-09 Thread Warren Selby
On Mon, May 9, 2011 at 7:26 AM, Pezhman Lali l...@lopl.net wrote: Dear I have a small pbx with asterisk 1.6.2.16. I have a funny problem, there is exactly 40sec between dial execution and sending first invite packet on sip. do you have any idea where the problem is ? Check the dial timeout

Re: [asterisk-users] OUTBOUND CALLER ID

2011-05-09 Thread Carlos Rojas
Hello Do you set your callerid in the context outgoing? [outgoing] exten = _X.,1,Set(CALLERID(num)=4663000) exten = _X.,n,Dial(.. On Mon, May 9, 2011 at 4:45 AM, mahesh katta maheshka...@flexydial.comwrote: Sir , this is not working On Mon, May 9, 2011 at 1:52 PM, A J Stiles

Re: [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?

2011-05-09 Thread Cassius Smith
On 5/9/11 6:02 AM, Doug Lytle supp...@drdos.info wrote: Sebastian Arcus wrote: Cisco phones (at least the 7940) are supposed to be run with a tftp server available at all time That is my experience. But, if you're running tftp under Linux, then it's probably spawned by xinetd and won't be

Re: [asterisk-users] asterisk syntax highlighting for gedit

2011-05-09 Thread Naomi Rosenberg
no worries, i'm not as passionate as some! It just happens gedit is the one I've gravitated towards, despite what I'm sure are good reasons to use something more hardcore. And I also fancy the project of writing the highlighting script, it would be a nice little job for me and I'm sure there

Re: [asterisk-users] Occasional call from asterisk

2011-05-09 Thread Bruce B
Thanks for the input. Long ago the CDR showed asterisk as the CLID but it doesn't anymore so I am puzzled now how to even stop taking calls because my CLID is now blank and I can't refuse any call with no CLID. *WARNING[11002] chan_dahdi.c: CallerID returned with error on channel 'DAHDI/2-1'*

[asterisk-users] Free Alarms sound

2011-05-09 Thread amit salunkhe
Dear All Can anyone let me know where i can free sound file whcih i can use for system monitoring alrams. Regards Amit-- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a

Re: [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?

2011-05-09 Thread Sebastian Arcus
On 05/09/2011 03:40 PM, Warren Selby wrote: /snip Thanks for the reply. No, I run tftpd directly from rc.local script (on Slackware). That's fine - I just wanted to know I wasn't doing something wrong. If everybody else is in the same boat - I'll just be along for the ride

Re: [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?

2011-05-09 Thread Warren Selby
On Mon, May 9, 2011 at 10:48 AM, Sebastian Arcus s...@open-t.co.uk wrote: That's strange. Mine get stuck on the booting phase, looking for the tftp server, if they can't find it there. Even if I change the dhcpd option not to pass out any tftp server. Any ideas what did you configure

Re: [asterisk-users] Asterisk 1.8 latest branch safe for production ?

2011-05-09 Thread --[ UxBoD ]--
- Original Message - On Fri, 2011-05-06 at 20:21 +0100, --[ UxBoD ]-- wrote: Are you not seeing issues with *8 call pick up then ? Nope, I double checked it after seeing someone saying they had issues with it and it is fine on the installation I have. Which release are you

Re: [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?

2011-05-09 Thread Sebastian Arcus
On 05/09/2011 04:50 PM, Warren Selby wrote: On Mon, May 9, 2011 at 10:48 AM, Sebastian Arcus s...@open-t.co.uk mailto:s...@open-t.co.uk wrote: That's strange. Mine get stuck on the booting phase, looking for the tftp server, if they can't find it there. Even if I change the dhcpd

Re: [asterisk-users] Asterisk 1.8 latest branch safe for production ?

2011-05-09 Thread satish patel
Which release are you running as this is still open https://issues.asterisk.org/view.php?id=18654 -- Thanks, Phil I am using current SVN branch 1.8 and We aren't using above call pickup features. _ -- Bandwidth and

Re: [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?

2011-05-09 Thread A J Stiles
On Monday 09 May 2011, Cassius Smith wrote: On 5/9/11 6:02 AM, Doug Lytle supp...@drdos.info wrote: Sebastian Arcus wrote: Cisco phones (at least the 7940) are supposed to be run with a tftp server available at all time That is my experience. But, if you're running tftp under Linux, then

[asterisk-users] conf syntax highlighting for gedit

2011-05-09 Thread Naomi Rosenberg
Hi, It seems there is no .conf syntax highlighting script available for gedit. I'm thinking of putting one together myself, but don't want to reinvent the wheel. So I'm just enquiring if anyone knows of one that already exists that i've missed. Thanks Naomi Rosenberg

Re: [asterisk-users] OT - Which Android handset with Wifi-only ?

2011-05-09 Thread Olivier
2011/5/9 randulo rand...@randulo.com On Mon, May 9, 2011 at 2:20 PM, mgra...@mstvp.com wrote: Lots of Android handsets support wifi, like my G2, aka HTC DesireZ. Wouldn't ANY modern one have wifi? That would be odd if it didn't, would it not? Yes, of course, all dual-mode phones support

Re: [asterisk-users] OT - Which Android handset with Wifi-only ?

2011-05-09 Thread Steve Underwood
On 05/10/2011 12:55 AM, Olivier wrote: 2011/5/9 randulo rand...@randulo.com mailto:rand...@randulo.com On Mon, May 9, 2011 at 2:20 PM, mgra...@mstvp.com mailto:mgra...@mstvp.com wrote: Lots of Android handsets support wifi, like my G2, aka HTC DesireZ. Wouldn't ANY modern one

[asterisk-users] iax2 issue in asterisk

2011-05-09 Thread satish patel
Hey guys! I have issue between iax vs iax2 following is my setup asterisk-1.2 --IAXAsterisk-1.8 I am able to call from 1.8 to 1.2 over iax but no from asterisk 1.2 to 1.8 --

[asterisk-users] high PDD

2011-05-09 Thread Abid Saleem
Hi, I am using Asterisk 1.4.17 for my C4 routing but I am experiencing a high pdd of around 20 seconds. Could you please help me to reduce it and what could be the reason. Thanks Abid Saleem --

Re: [asterisk-users] iax2 issue in asterisk

2011-05-09 Thread Gordon Henderson
On Mon, 9 May 2011, satish patel wrote: Hey guys! I have issue between iax vs iax2 following is my setup asterisk-1.2 --IAXAsterisk-1.8 I am able to call from 1.8 to 1.2 over iax but no from asterisk 1.2 to 1.8 Might you be missing requirecalltoken=no

Re: [asterisk-users] iax2 issue in asterisk

2011-05-09 Thread satish patel
Awesome! root@:~# cat /etc/asterisk/iax.conf | grep requirecalltoken ; By setting 'requirecalltoken=no', call token validation becomes optional for ; that peer/user. By setting 'requirecalltoken=auto', call token validation ; can require it from this peer. So, requirecalltoken is

Re: [asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS

2011-05-09 Thread Paul Hayes
Hi, It looks to me that the 401 unauth packets aren't getting back to the phones. Which suggests a network/router/nat issue rather than anything wrong with the asterisk or phone configuration. Cheers, Paul. On 8 May 2011, at 01:59, GNUbie gnu...@gmail.com wrote: Hello all, I have

[asterisk-users] Rates Importer Tool

2011-05-09 Thread A E [Gmail]
Hi All, new to the list. Wondering if anyone has / knows of, a good rate importer tool that can be used to standardize and normalize the ratesheets / rate decks etc. obtained from various carriers so they can be analysed and imported into a DB or be saved as a CSV or something? Thanks so much in

Re: [asterisk-users] Rates Importer Tool

2011-05-09 Thread Jason Aarons (AM)
I know most billing software sell this as a monthly service. You get cd-rom every month where they have collected the published tarrif tables filed with the FCC. You load it on the software to analyze call costs. I'm guessing this is a lot of labor hours/manual work thus they charge for

Re: [asterisk-users] Rates Importer Tool

2011-05-09 Thread A E [Gmail]
On Mon, May 9, 2011 at 3:05 PM, Jason Aarons (AM) jason.aar...@dimensiondata.com wrote: I know most billing software sell this as a monthly service. You get cd-rom every month where they have collected the published tarrif tables filed with the FCC. You load it on the software to analyze

[asterisk-users] Need help defining a stackexchange (i.e. stackoverflow) for telephony

2011-05-09 Thread Simon P. Ditner
For those of that are fans of stackoverflow.com, and stackexchange.com, there's an effort to define a telephony stackexchange site. It's still in the definition phase. What it needs to move forwards is more votes on on/off topic questions, and perhaps some better questions to vote for or

Re: [asterisk-users] Blacklist with *30

2011-05-09 Thread Alejandro Cabrera Obed
Dear, finally I implement the functionality code *94 in order to access the blacklist menu from my own extension and put another extension in the black list of Asterisk. But after blacklisting a given extension, when I call from that extension to my own extension the call always rings, it is not

[asterisk-users] Really, really loud ringers

2011-05-09 Thread Justin Sherrill
Anyone have some recommended equipment for alerting people to calls in a noisy environment? I have Polycom IP550 phones set up in some really noisy environments - our mine hoists - and they tend to drown out the ringers. I'm using Clarity WR100s now. They're analog devices, attached to

Re: [asterisk-users] Really, really loud ringers

2011-05-09 Thread Andrew Latham
On Mon, May 9, 2011 at 4:40 PM, Justin Sherrill justin.sherr...@americanrocksalt.com wrote: Anyone have some recommended equipment for alerting people to calls in a noisy environment? I have Polycom IP550 phones set up in some really noisy environments - our mine hoists - and they tend to

[asterisk-users] Voicemail Configuration

2011-05-09 Thread John Marvin
Hi, I can't figure out a way of achieving what I want to do with the voicemail feature. I thought I'd ask here to see if there are any creative solutions that I have not considered. What I want to do is have a message that says Press 1 for Dick, or 2 for Jane. Then, depending on which

[asterisk-users] Call ends when using SendDTMF(*)

2011-05-09 Thread m j
I'm not sure why but my call is being ended when I SendDTMF(*). I'm using agi to originate a call and set the context,extension,priority to test,1,1 respectively. I've got the following in my extensions.conf: [test] exten = 1,1,Answer(); same =n,Wait(5); same =n,Verbose(1, Sending *);

Re: [asterisk-users] Voicemail Configuration

2011-05-09 Thread Roger Burton West
On Mon, May 09, 2011 at 03:00:19PM -0600, John Marvin wrote: However, I want to record what is said during that time and send it to a third voicemail box once the caller hangs up without having pressed 1 or 2. You could use Monitor to record the whole call, then use an AGI to do something with it

Re: [asterisk-users] Voicemail Configuration

2011-05-09 Thread John Marvin
On 5/9/2011 3:08 PM, Roger Burton West wrote: You could use Monitor to record the whole call, then use an AGI to do something with it on hangup if the other conditions haven't been satisfied...? I understand how to do the first part, and I at least understand that I could do something fancy

Re: [asterisk-users] [SOT] Virtualising Asterisk

2011-05-09 Thread Jan Bakuwel
Hi Phil, Happily running with the following here: dom0: Debian Lenny Xen 3.2-1 2.6.26-2-xen-amd64 domU: Asterisk 1.4 Debian Lenny 2.6.26-2-xen-amd64 domU: Asterisk 1.6 Debian Squeeze 2.6.32-5-amd64 (which is a Xen-aware kernel) domU: Asterisk 1.8 Debian Squeeze 2.6.32-5-amd64 (which is a

Re: [asterisk-users] asterisk syntax highlighting for gedit

2011-05-09 Thread Matt Riddell
On 10/05/11 2:32 AM, Naomi Rosenberg wrote: Hi, Apologies if this is a duplicate - been having mail server issues and I don't think I managed to send it when I tried this morning. It seems there is no .conf syntax highlighting script available for gedit. I'm thinking of putting one together

Re: [asterisk-users] Rates Importer Tool

2011-05-09 Thread Markus
Hi, new to the list. Wondering if anyone has / knows of, a good rate importer tool that can be used to standardize and normalize the ratesheets / rate decks etc. obtained from various carriers so they can be analysed and imported into a DB or be saved as a CSV or something? I'm using

Re: [asterisk-users] Rates Importer Tool

2011-05-09 Thread A E [Gmail]
On Mon, May 9, 2011 at 7:58 PM, Markus unive...@truemetal.org wrote: Hi, new to the list. Wondering if anyone has / knows of, a good rate importer tool that can be used to standardize and normalize the ratesheets / rate decks etc. obtained from various carriers so they can be analysed and

Re: [asterisk-users] QueueCallerAbandon is not triggering after 1.8.3.3...

2011-05-09 Thread Louis Carreiro
Has anyone else noticed this? v/r, Me On Fri, May 6, 2011 at 12:11 PM, Louis Carreiro carreir...@gmail.comwrote: Has anyone else noticed that QueueCallerAbandon is not showing up in the AMI after the 1.8.3.3? Am I missing something? I'm getting what seems like everything else but