Re: [Asterisk-Users] features.conf *1 Call Recording

John Novack wrote: Can't this guy read? I'll bet he runs Micro$oftware, and has fallen prey to one of its thousands of exploits-du-jour. Set a filter and move on. . . B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. __

Re: [Asterisk-Users] features.conf *1 Call Recording

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Re: [Asterisk-Users] ATXFER

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Re[2]: [Asterisk-Users] features.conf *1 Call Recording

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RE: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

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Re[2]: [Asterisk-Users] features.conf *1 Call Recording

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RE: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

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Re[2]: [Asterisk-Users] features.conf *1 Call Recording

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RE: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

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Re[2]: [Asterisk-Users] features.conf *1 Call Recording

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RE: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

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Re[2]: [Asterisk-Users] features.conf *1 Call Recording

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RE: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

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Re[2]: [Asterisk-Users] features.conf *1 Call Recording

CLI*> show application dial -Original Message- From: "Dave Morrow" <[EMAIL PROTECTED]> To: "Dave Morrow" <[EMAIL PROTECTED]>, "Asterisk Users Mailing List - Non-Commercial Discussion" Cc: Sent: Fri, 12 May 2006 14:12:21 -0400 Delivered: Fri, 12 May 2006 15:16:15 Subject:

RE: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

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RE: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

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RE: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

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RE: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

Eric, but I to continue using version 1.0.9 of asterisk, would have some solution? Regards   Josué  2006/5/12, Eric ManxPower Wieling <[EMAIL PROTECTED]>: If you want dynamic features you must upgrade to Asterisk 1.2.xJosué Conti wrote:> Eric, thank you very much. But It could help in this case me?

Re: [Asterisk-Users] ATXFER

If you want dynamic features you must upgrade to Asterisk 1.2.x Josué Conti wrote: Eric, thank you very much. But It could help in this case me? Regards Josué 2006/5/12, Eric ManxPower Wieling <[EMAIL PROTECTED] >: Josué Conti wrote: > Dinesh,

Re: [Asterisk-Users] Having Rinback tone generation issues with 1.2.7.1

"r" almost never fixes ringback issues. "r" will provide ringback when the caller should hear something ELSE, such as onhook audio "the cellular user you are calling is not in the calling area" Confirm you have /etc/asterisk/indications.conf Not having it can cause ringback issues in some si

Re: [Asterisk-Users] VoiceMail application: "j" option not working as I supposed

Try: exten => _XX,2,VoiceMail([EMAIL PROTECTED],j) and exten => _XX,110,VoiceMail([EMAIL PROTECTED],j) Álvaro Palma wrote: I've the following dialplan. exten => _XX,hint,SIP/${EXTEN} exten => _XX,1,Dial(SIP/${EXTEN},10,j) exten => _XX,2,VoiceMail([EMAIL PROTECTED],u|j) exten => _XX,3,Hangu

Re: [Asterisk-Users] fc5 and link to sources?

On Friday 12 May 2006 18:30, Rich Adamson wrote: > Carlos Alperin wrote: > > Rich, > > > > Check what is the content of /lib/modules/2.6.15-1.2054_FC5/build? > > I see: > [EMAIL PROTECTED] build]# ls > arch crypto initMAINTAINERSREADME usr > blockDocumentation ipc

Re: [Asterisk-Users] fc5 and link to sources?

Carlos Alperin wrote: Rich, Check what is the content of /lib/modules/2.6.15-1.2054_FC5/build? I see: [EMAIL PROTECTED] build]# ls arch crypto initMAINTAINERSREADME usr blockDocumentation ipc Makefile REPORTING-BUGS configs driversKbuild Ma

[Asterisk-Users] VoiceMail application: "j" option not working as I supposed

I've the following dialplan. exten => _XX,hint,SIP/${EXTEN} exten => _XX,1,Dial(SIP/${EXTEN},10,j) exten => _XX,2,VoiceMail([EMAIL PROTECTED],u|j) exten => _XX,3,Hangup() exten => _XX,102,Goto(110) exten => _XX,103,Playback(pbx-invalid) exten => _XX,104,Hangup() exten => _XX,110,VoiceMail([EMAIL

RE: [Asterisk-Users] fc5 and link to sources?

Rich, Check what is the content of /lib/modules/2.6.15-1.2054-FC5/build? If it is empty, then you need to do yum install kernel-devel again. Also you can check running uname -a to see if you have the same release that the one that you're checking. Regards, Carlos Alperin -Original Message

[Asterisk-Users] Re: MeetME Conferencing

In article <[EMAIL PROTECTED]>, Mike Clark <[EMAIL PROTECTED]> wrote: > Damon Estep wrote: > > > Can anyone point me to a sample or information on using MeetMe like this? > > > > Conference room is set up with 2 PINs, one for the moderator and one > > for the participants. > > > > Participants ge

[Asterisk-Users] Re: Call parking from legacy PBX over PRI??

I did a test with ParkAndAnnounce and the call back is not going to fly here. Does anyone have a version that talks back during the transfer like Park() does? I piggybacked off of another feature request in the bug system that is very similar. http://bugs.digium.com/view.php?id=6953 I hope doing

RE: [Asterisk-Users] DUNDi and Voicemail

Thanks Aaron. That'd probably work. However, we also have an asterisk box dedicated to ACD, and we face the same problem with that. Phones don't register with it directly, but it still needs to know their location. Ideally we need one solution to address both the voicemail and acd servers. Doug

[Asterisk-Users] Re: Call parking from legacy PBX over PRI??

Here is a CLI of the problem: Here is a CLI of the problem: == Timeout for Zap/47-1 parked on 5401. Returning to park-dial,Zap/47,1 -- Executing Dial("Zap/47-1", "Zap/47||t") in new stack == Everyone is busy/congested at this time (1:0/0/1) -- Hungup 'Zap/47-1' Zap/47-1 could be any of

[Asterisk-Users] hi guys, a new newbie here needing help :D

I just installed rpm binaries in a new mandriva and I see a frew error messages with asterisk -vvvcfg, btw I would also like a little guidance to just set up a couple sip phones to start playing with soft phone communication with 3 pcs on the network thanks :) ng '/etc/asterisk/agents.conf':

[Asterisk-Users] fc5 and link to sources?

Just installed fc5, installed correct kernel source, and trying to compile zaptel-1.2. Changed the link in /lib/modules/2.6.15-1.2054_FC5 to point to /usr/src/redhat/SOURCES. Like: lrwxrwxrwx 1 root root 23 May 12 15:21 build -> /usr/src/redhat/SOURCES A 'make install' still complains with: mak

Re: [Asterisk-Users] DUNDi and Voicemail

You were doing so good too. The voicemail application has a function to run an external app to notify about voicemail. We have scripts on the main servers that recieve notification from a voicemail server script that particular phones have a certain number of messages. That script then runs

Re: [Asterisk-Users] How to determine if a device is in a call

Thanks to everyone who responded. I was able to modify the freepbx paging code to use something like the suggested macro and it worked well. For those who may be interested, the following Page macro works for Linksys SPA942 phones: [macro-page]; ; ; Paging macro: ; ; Check to see if SIP device

[Asterisk-Users] Cell phone dialed digits too short to be recognized by asterisk

I'm having a big problem where digits dialed from certain cell phones are too short to be recognized by my asterisk server. I'm running AAH 2.8. Some cell phones don't allow the caller to hold down the digits and have the tones play as long as they hold them down for. They just play a short ton

[Asterisk-Users] DUNDi and Voicemail

Ugh. We thought we'd fixed some problems by using regexten and DUNDi. Guess not. We have a configuration with three Asterisk boxes. Phones register with a single, primary asterisk box under normal conditions. For voicemail deposit, retrieval, we trunk the calls over to our asterisk voicemail ser

RE: [Asterisk-Users] Dialling a DUNDi Route

> -Original Message- > From: Leif Madsen [mailto:[EMAIL PROTECTED] > Sent: Friday, May 12, 2006 12:27 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Dialling a DUNDi Route > > > On 5/12/06, Florian Overkamp <[EMAIL PROTECTED]> wrote: > > D

RE: [Asterisk-Users] Paging and Auto Answer on Grandstream GXP2000

You are correct, That is why the PAGE() Application was made. It creates a MeetMe room, calls the Technologies on the list and transfers the calls to the temp MeetMe.       From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora Sent: Friday, May

Re: [Asterisk-Users] Paging and Auto Answer on Grandstream GXP2000

I must be missing something.  Seems to me that only one phone would connect.  This is just a plain dial command that rings all those extensions and when one answers, the rest stop ringing.   Right?  On 5/12/06, Tom Vile <[EMAIL PROTECTED]> wrote: make the dial command like so:exten => PAGE203,n,Dia

Re: [Asterisk-Users] Dialling a DUNDi Route

On 5/12/06, Florian Overkamp <[EMAIL PROTECTED]> wrote: Douglas Garstang wrote: > We are using a backend MySQL database for call flow, not user agent > registration info. Just how, exactly, is a backend database going to > replicate registration data between Asterisk servers? Realtime has > been

Re: [Asterisk-Users] Dialling a DUNDi Route

On 5/12/06, Florian Overkamp <[EMAIL PROTECTED]> wrote: Douglas Garstang wrote: > We are using a backend MySQL database for call flow, not user agent > registration info. Just how, exactly, is a backend database going to > replicate registration data between Asterisk servers? Realtime has > been

[Asterisk-Users] Plain Text Passwords for IAX and SIP

Can someone tell me if passwords are sent in plain text when using IAX? I have been told already that SIP automatically encrypts the password? Anyone know of some good Asterisk security links, docs, articles? Thanks! ___ --Bandwidth and Colocation pro

[Asterisk-Users] voicemailmain()

Hi,  in the menu of voicemailmain, appear a lot of options, there is a way to leave only some of them?   Also I want to know if there is a option that erase all message in a user box.     Best REgards   Ever Zalazar ___ --Bandwidth and Colocation pro

Re: [Asterisk-Users] Paging and Auto Answer on Grandstream GXP2000

make the dial command like so: exten => PAGE203,n,Dial(SIP/203&SIP/204&SIP/205SIP/206,5) On 5/12/06, Forrest Beck <[EMAIL PROTECTED]> wrote: Thanks. I like that method. Do you think if I add all my extensions (say 40 of them) to new Dial commands after "exten => PAGE203,n,Dial(SIP/203,5)" Li

RE: [Asterisk-Users] features.conf *1 Call Recording

I found the issue.   It was my Dial command!   In my dialplan I had Dial(SIP/100|20|Ttr,,wW) as this was something I gleaned from a sample config for call forwarding.  I removed the |20|Ttr and now the call recording works!  Anyone know what the |20|Ttr did anyhow?   David Morrow Technical Sy

[Asterisk-Users] Dial Command Reference for SIP channel

Hi all.  I was reading a sample config someone had posted relating to call forwarding, and in it, they use a Dial command with components that I cannot find any reference to.   Can someone point me to a reference which could explain the difference between Dial(SIP/100|20|Ttr,,wW) and Dial(S

RE: [Asterisk-Users] Paging and Auto Answer on Grandstream GXP2000

Thanks. I like that method. Do you think if I add all my extensions (say 40 of them) to new Dial commands after "exten => PAGE203,n,Dial(SIP/203,5)" Like this: exten => PAGE203,n,Dial(SIP/203,5) exten => PAGE203,n,Dial(SIP/204,5) exten => PAGE203,n,Dial(SIP/205,5) exten => PAGE203,n,Dial(SIP/20

RE: [Asterisk-Users] Voicemail WAV to PDA Problems

> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Kerry Garrison > Sent: Friday, 12 May 2006 12:07 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] Voicemail WAV to PDA Problems > > Ou

[Asterisk-Users] RE: snmp and asterisk

hi david, can you explain me this please? If Sangoma hardware support snmp ithink it would be a better choice than digium . How can we know the state of the sangoma cards with an snmp agent ? Harry --- David Yat Sin <[EMAIL PROTECTED]> a écrit : > Hi Harry, > The Sangoma Card when used for TDM

[Asterisk-Users] Having Rinback tone generation issues with 1.2.7.1

Today i move our central server to 1.2.7.1 , and im having some issues with SPA Phones and RinbackTone. Without r option, it also happens. Is having anyone this issue? I think it has not been changed anything sustancially to happen this to me. It is happening between extensiones (canreinvite=y

RE: [Asterisk-Users] features.conf *1 Call Recording

I have one Sipura SPA-841 which is configured to use dtmfmode=info and one Cisco 7905 which is using the default signalling (I believe this is rfc2833) I have also set relaxdtmf=yes in sip.conf   I've tried pressing *1 on both phones (they are both on my desk) and both behave the same.   ;;

Re: [Asterisk-Users] features.conf *1 Call Recording

hi Dave i get the following log on *CLI>    -- Attempting native bridge of SIP/200-39f4 and SIP/204-2ce4    -- Playing 'beep' (language 'en')    -- User hit '*1' to record call. filename: wav|auto-1147452537-200-204|m     -- Playing 'beep' (language 'en')    -- User hit '*1' to stop recording call

[Asterisk-Users] Cisco 7970 problems

Has anyone had problems with a Cisco 7970 running sip image SIP70.8.0-2SR1S hanging up zap channels?   Calls to SIP and IAX are fine. Just when the call goes out via the zap channels   I have some Cisco 7960 running SIP and they work fine.   Any ideas?   Thanks-Eric Hall   __

RE: [Asterisk-Users] MeetME Conferencing

Your welcome. It certainly could be done entirely in the dialplan using similar logic, but this required a bit less mental horsepower. If your desire to avoid AGI, is based on performance concerns, note that I have systems (Dell 2850 2xXEON 3.0) that terminate 8 PRIs and have had ALL channels lo

[Asterisk-Users] Help Avaya 4606

Hello all, I have asterisk working well with, Sipura, but I do not manage to form several phones avaya 4606, someone could have formed one avaya with asterisk? is it possible? update the firmware of the phone, but I do not achieve that it registers, I hope that someone could help me greeting

RE: [Asterisk-Users] Voicemail WAV to PDA Problems

Our system is running all of the latest code and freepbx and would send the attachment to my MDA just fine and I was able to play it without any problem. My problem was that the MDA is a worthless turd and a complete joke as a phone. I took it back and switched to the backberry 8700g which has its

[Asterisk-Users] Music on Hold restart at beginning for each call

I am using the m option on the dial command to play a message instead of ringing. The message is something like "please wait while I try to locate your party" so I need it to start at the beginning for each call. I think there might be a way in 1.2.x be we are not ready to upgrade yet so a sol

[Asterisk-Users] Voicemail WAV to PDA Problems

Our asterisk server has been up and running for over a year and it works great. I have emails going to my account as an attachment and I can listen to them on the desktop and it works fine. I just got a T-Mobile MDA that runs Windows Pocket (or whatever they call it) and it can check email.

RE: [Asterisk-Users] features.conf *1 Call Recording

It's quite strange. When I press *1 I do not hear a tone indicated that it's even trying to record.   David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com   Tel: (519) 963-3020 Fax: (519) 451-6615   < Lead, follow or get out of the w

Re: [Asterisk-Users] Hint priority

On 5/12/06, Jerry Jones <[EMAIL PROTECTED]> wrote: I believe the hint priority must be in the same context as the phones extension number, in this [local] Additionally, it may not be the first 'exten =>' line, at least in some versions, so best to put them at the end of the context. PLUS: Avo

Re: [Asterisk-Users] Bristuffed Asterisk: Hangup problems

On 5/12/06, stoffell <[EMAIL PROTECTED]> wrote: On 5/11/06, Tim Robinson <[EMAIL PROTECTED]> wrote: > There is a lot of junk in your zapata.conf that you do not need, as it > relates to analogue lines. This might be causing confusion? I have tried a similary config to yours, doesn't helps. I ha

Re: URGENT please [Asterisk-Users] call parked / MOH

Harry,Please note that you have sent this message to the group several times today. If anyone has an answer for you, they will reply. There is no need to continually send it to the group.Alex On 5/12/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > > Hello,> >> > How can I park a call or put on h

[Asterisk-Users] Re: Problem setting locale for voicemail

Ok, this is my voicemail.conf: [general] attach=yes charset=ISO-8859-1 emailbody=${VM_NAME}:\n\nUd ha recibido un nuevo mensaje de voz, de ${VM_DUR} segundos.\n\nTeléfono: ${VM_CALLERID}.\nFecha: ${VM_DATE}.\n\nEl mensaje ha sido adjunto a este correo.\n\nPor favor, no responda este mensaje, e

Re: [Asterisk-Users] S100-FX v2 audio quality

We did communicate this to the manufacturer and they fixed 1 issue with bad power supplies. We tried multiple codecs but it was still unreliable, so we went back to the IAXy and no issues. All calls came in over a PRI. Did not want to waste to much time on these, maybe we will look again in th

Re: [Asterisk-Users] Paging and Auto Answer on Grandstream GXP2000

This is what I use: [ext-paging] exten => PAGE203,1,Set(__SIPADDHEADER=Call-Info: answer-after=0) exten => PAGE203,n,Set(__ALERT_INFO=Ring Answer) exten => PAGE203,n,Set(__SIP_URI_OPTIONS=intercom=true) exten => PAGE203,n,Dial(SIP/203,5) exten => Debug,1,Noop(dialstr is LOCAL/[EMAIL PROTECTED]) e

URGENT please [Asterisk-Users] call parked / MOH

> > Hello, > > > > How can I park a call or put on hold a caller from > > an > > analogue to sip agents ? > > > > PSTN===FXO/asterisk=sip agents > > > > When I press hold key or #800 the channel is > hangup > > ?? > > > > Harry > > Regards __

RE: [Asterisk-Users] Paging and Auto Answer on Grandstream GXP2000

There is some additional functionality coming in future firmware versions. See http://www.voip-info.org/wiki/index.php?page=Asterisk+Paging+and+Intercom On Fri, 2006-05-12 at 14:54, Forrest Beck wrote: > Asterisk 1.2.7.1 and Zaptel 1.2.5 > > -Original Message- > From: [EMAIL PROTECTED] >

Re: [Asterisk-Users] S100-FX v2 audio quality

Tom Vile wrote: Same problem with audio quality. Got rid of them. Also the context line only allowed 12 characters and we need more than that for some installations, I didn't want to have to rename 100 contexts to less than 12 characters. Which audio codecs were you using? I'm using g729 to co

URGENT please [Asterisk-Users] call parked / MOH

> Hello, > > How can I park a call or put on hold a caller from > an > analogue to sip agents ? > > PSTN===FXO/asterisk=sip agents > > When I press hold key or #800 the channel is hangup > ?? > > Harry > Regards

Re: [Asterisk-Users] email -> fax gateway with billing possibilities?

> does anyone have an idea how it could be possible to do email -> fax > gatewaying with asterisk + app_txfax, but still keep track of who > sent the fax? i've thought a little about smtp auth, but it doesn't > look too easy to integrate smoothly with asterisk i don't know what your prob

Re: [Asterisk-Users] Re: Sangoma A200D problem

Last time I had this problem was following a unclean powerdown and the solution was: - Kill Asterisk - Stop wanpipe - cd /etc/wanpipe/wan_ec - In there there should be 2 files: wan_ec_pid wan_ec_socket= - Delete those files - Perform a reboot of your sys

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