[asterisk-users] CSTA support for asterisk

Hi, Can anybody tell me that is their CSTA support for asterisk sanchal ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] french promt

On Thu, 2006-07-27 at 11:23 +0200, Olivier Saulnier wrote: Follow this link: http://svn.digium.com/view/asterisk/sounds/fr/trunk/?rev=34575 You don't need to get them out of subversion. They are available in tarballs on the ftp. ftp://ftp.digium.com/pub/telephony/sounds When installing

Re: [asterisk-users] Getting no Audio with G729

On Jul 27, 2006, at 2:06 PM, Wasif wrote: Hello, Recently I purchased g729 codec and installed in Tribox 1.1 (upgraded 1.1.1)/ Asterisk. I have pointed a DID from my carrier via SIP through g729 to asterisk. Problem is I am not getting any audio even though I am getting DTMF in asterisk.

Re: [asterisk-users] bugs.digium.com

On Thu, 27 Jul 2006, Russell Bryant wrote: Fine... so how do you file an enhancement request then? If there's no way to file an enhancement request, then this is the most appropriate place to file this. The bug tracker is really is not a good place for feature requests. You have to

[asterisk-users] asterisk with CSTA using VAIL SIP TIM

Hi, Can anybody tell me how to configure asterisk for VAIL SIP TIM so that CSTA is utilized... sanchal ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] IP Authorization

Sorry to dredge up an old topic, but could someone help me with this? I need to accept and forward a call from a range of ip addresses without any other authentication. (from-internal) Does anyone have a small snipped of extensions.conf and sip.conf that I can use to implement this? Thanks

[asterisk-users] Multiple Outbound SIP Trunks

I have 3 sip trunks registered with an outside provider, however asterisk always seems to work when going out the third trunk. Any way to round-robin this so that we can make more than one outbound call at a time? Thanks in advance, Aaron ___

RE: [asterisk-users] SNOM 360

On Freitag, 28. Juli 2006 12:37 Dovid Bender wrote: Does anyone know how to set up QoS on the SNOM 360 ? Thanks. What _EXACTLY_ are you trying to accomplish? There is no simply QoS switch on a Snom 360 that will manage things for you. AFAIK all you can do is tell the phone (or * or whatever)

[asterisk-users] Fritz!Box Fon ATA

Hi, I have bought a Fritz!Box Fon ATA in eBay. I’m trying to find information about configuration this box in Asterisk. Its possible use this box like a normal ATA (sipura 3000…) receiving and making ISDN calls from Asterisk? Somebody has information in English about this box? Some example

Re: [asterisk-users] long distance ethernet Asterisk

Light the fiber to get to that 2000ft mark, then use directional antennas to cover the last 2000 ft via wireless. If necessary, you could even use unlicensed 900 mhz gear that runs 802.11g speeds (search for the Ubiquiti SR9), http://www.wlanparts.com/c=*/product/SR9 has it for $149 when in

[asterisk-users] FreePBX Inbound Route

Hi, I have SIP trunk. And I also have a lot of SIP clients. If I want to call from SIP trunk to the Asterisk SIP client, I need to create Inbound route for each endpoint. Maybe is possible to create an endpoint group, because I have a lot of SIP endpoints, and it takes a lot of time to create

[asterisk-users] Rate engine AGI?

Is there an AGI out there which we can call from extensions.conf which will lookup a rate in a MySQL db based on the number the callerer dialed? We don't want anything with tons of features as we are doing all our coding, we just want something that will give us the rate and maybe permission to

RE: [asterisk-users] Polycom_acd_functions SIP trouble

At first, but if you checkout the version 1221, someone has fixed it. svn checkout http://svn.digium.com/svn/zaptel/trunk/ zaptel-trunk -r 1221 Regards, Dean. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Miller Sent: 28 July 2006 04:27 To:

Re: [asterisk-users] Fritz!Box Fon ATA

Hi Manuel, I do not know the ATA, but we have a Fritz!Box fon, wich maybe is equal in setting it up. If you have any problems understanding the german setup, you can contact me, so I can help you in translating the needed Words :-) Normally you only have to do this on the Webinterface:

Re: [asterisk-users] SNOM 360

On 7/28/06, Koopmann, Jan-Peter [EMAIL PROTECTED] wrote: On Freitag, 28. Juli 2006 12:37 Dovid Bender wrote: Does anyone know how to set up QoS on the SNOM 360 ? Thanks. What _EXACTLY_ are you trying to accomplish? There is no simply QoS switch on a Snom 360 that will manage things for you.

[asterisk-users] PAP2T always busy on incoming calls with zaptel

Hi, I'm starting to use the new PAP2T instead of the old PAP2-NA for my new installations. I'm having a weird problem: when a call is comming from a zaptel channel (from a bri with bristuff driver) the PAP2T say BUSY to the SIP channel. I have disabled all the features like DND and call

Re: [asterisk-users] Sip phone settings set when user registers

2006/7/27, Nik Engel [EMAIL PROTECTED]: User logs into any phone and the settings of the phone are always thesame. Meaning individual keyassignement is always the same.Hi,Do you mean :1. Without user logins, phones are unusable ? Or do you plan to offer default services (local calls for instance)

Re: [asterisk-users] RE: alcatel ip touch 4068 ... sip?

2006/7/28, Leo Ann Boon [EMAIL PROTECTED]: (AstATN) wrote:Hi Cesc,Alcatel does not offer any SIP Terminal, it phone's are H.323 it's handlefor their own features usages. ( like ADSI type )Common misconception. Their phones are not H.323 despite claims in theirdocumentation. The server has to do

[asterisk-users] asterisk+ooh323.. one way audio issue

Hi guys, I tried to make call from SIP channel to H323 using asterisk+ooh323. The SIP client is x-lite.The problem is that there is one way audio. I hear everything from h323 endpoint, and I see the messages also: Got RTP packet from 66.135.35.xx:5002 (type 3, seq 36250, ts 74400, len

[asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger

Hi, We're seeing a problem on Asterisk 1.2.10 where when we get in in the morning it's continually rotating the logs over and over again, generating 100's of thousands of log rotated 0 byte files:- /var/logs/asterisk # find . -type f -maxdepth 1 | wc -l 176930 /var/log/asterisk # find . -type f

[asterisk-users] Sending email after voicemail

Hi, I'm having trouble getting asterisk to send a voicemail message via email. I can do a mail [EMAIL PROTECTED] from the linux command line and I receive the email fine, and if I look in the exim4 logs it looks ok, has from user, to user and completed but no email is received. Any thoughts?

Re: [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger

check your cron jobs. mayby there is asterisk -rx logger rotate executing too often ? Hi, We're seeing a problem on Asterisk 1.2.10 where when we get in in the morning it's continually rotating the logs over and over again, generating 100's of thousands of log rotated 0 byte files:-

Re: [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger

Filip Drągowski wrote: check your cron jobs. mayby there is asterisk -rx logger rotate executing too often ? Nope - nothing in crontab. Hi, We're seeing a problem on Asterisk 1.2.10 where when we get in in the morning it's continually rotating the logs over and over again, generating

[asterisk-users] CDR IP Authorization

Dear This function retrieves the ip address of the caller ,I want to import the value of (recvip) in the mysql cdr ,how can I do that exten = s,1,NoOp(${SIPCHANINFO(recvip)}) Regards * No employee or agent is authorized to

Re: [asterisk-users] SIP client with video???

Hi, i have xlite too and it works without any problems. ps: what about ekiga? (www.ekiga.org) rich --- Joao Pereira [EMAIL PROTECTED] wrote: Hello to all can someone recommend me a nice SIP client with video for windows?? I tried X-Lite 3.0 but it's a lousy piece of software.

[asterisk-users] Transfer call in SIP

Hello, I am running TrixBox. if already in a call session from ZAPTEL to SIP, the user want to transfer the call to a different extension. The user have to dial *extension ? Any configuration is needed to be done in trixbox? Thanks Victor ___

Re: [asterisk-users] Manager interface

On 27 Jul 2006, at 22:42, Tielin Xu wrote: There are many ways to do the screen pop, I'd like to do this way: 1. Build the manager interface as an event server, which collect agent connet events. 2. Build a Java applet with the constant connection to the event server, each agent starts the

Re: [asterisk-users] long distance ethernet Asterisk

As far as g.SHDSL is concerned, I think you are limited to 4mbit/s We have an internet connection at work is delivered over g.SHDSL, (at 1Mb/s) to a cisco 828 and if I remember right, the ping time is of the order of 10ms. Certainly no problem for SIP. Tim Panton www.mexuar.com

Re: [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger

on one cosnole a do asterisk -r on other i do asterisk -rx "logger rotate" and the result is     -- Remote UNIX connection Asterisk Event Logger restarted Asterisk Queue Logger restarted     -- Remote UNIX connection disconnected how often new log files are created ? = how many log files are

RE: [asterisk-users] Ringing timer

Ok.Thanks a lot! I will try! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Canreinvite

How can I check if SIP re-invite is really working ? I'm trying it with two grandstream gxp2000. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

[asterisk-users] Re: Fritz!Box Fon ATA

phone to another. That doesn't seem easy if phones are installed in different locations. Regards -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060728/b8a1c9 87/attachment-0001.htm

Re: [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger

Filip Drągowski wrote: on one cosnole a do asterisk -r on other i do asterisk -rx logger rotate and the result is -- Remote UNIX connection Asterisk Event Logger restarted Asterisk Queue Logger restarted -- Remote UNIX connection disconnected how often new log files are created ? =

[asterisk-users] registration process

Hi all, I wonder if there are 2 UAs having the same sip account and password. If they both register to the same server in same time. Both of them can register successfully and make calls. Am I right? How can I prevent the above case, say only one UA can register to the server? Please advice.

Re: [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger

asterisk does daily log rotate all along ? i didn't know that it is posiible i create file in /etc/logrotate.d/asterisk (copy of postgrsql and renamed it) /var/log/asterisk/full {     daily     rotate 10     copytruncate     delaycompress     compress     notifempty     create 640 root root }

[asterisk-users] Voicmail Question

Hi list, is it possible to pick up a call from VoiceMail system? Didn't find nothing on voip-info.org Thanks for your answers KAI ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update

RE: [asterisk-users] MWI from Octel to Asterisk

When we were first looking at Asterisk, I explored some options of integrating our existing Octel voicemail systems with it. The only possible way I could come up with (understanding that I am by no means an Octel expert) was DTMF inband integration. The most difficult part seemed to be

[asterisk-users] stream file outputs only silence, even with asterisk example gsm file

Hi all! I am trying to hook up a text to speech system to Asterisk via AGI. The AGI script generates a sound file and tells Asterisk to play that file via STREAM FILE. I am creating the sound file in alaw format. My problem is that I do not get any sound output on my softphone (IAX soft phone)

[asterisk-users] cmd DIAL - Who picked up the call?

Hi, if I do Dial(SIP/peer1/numberZap/g1/Number) can I somehow figure out who exactly picked up the call? In the cdrs dstchannel I can see the channel but not the extension dialed. E.G. a Dial(Zap/10/43) will result in a CDR Zap/10-1 which does not help me unfortunatly. Any ideas? Kind

[asterisk-users] Weird E1 problem

Hi all, I have a weird problem. I have a Digium te400p with 4 E1s coming in to it. When one of the E1 lines is plugged into any of the four connections on the digium card I get YELLOW / RED alerts when I cat /proc/zaptel/SOCKET. But I can move the bad E1 line into any socket on the digium card

Re: [asterisk-users] Re: Fritz!Box Fon ATA

phone config should be portable from one phone to another. That doesn't seem easy if phones are installed in different locations. Regards -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060728/b8a1c9 87

Re: [asterisk-users] IAX2 Connection fails over time...

Stuart Sheldon wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hey all, I have a x86 Pentium D asterisk system with two Digium 400's in it. I am establishing a IAX2 Connection to another Asterisk system running on a Solaris server. When a call is placed between the two systems,

[asterisk-users] Zaptel trunk failed to compile

Morning everybody, I try to install an asterisk test server with trunk branch and get this error when compiling zaptel. Asterisk core compile fine as well as SVN 1.2 branch. It's a Debian SARGE running on 2.4.27 kernel. zttranscode.c: In function `zt_tc_mmap': zttranscode.c:378: warning:

Re: [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger

I use logrotate too, because I didn't know of the functionality in Asterisk. Logrotate works fine for me though. Kenny, you should give it a try! K On 7/28/06, Filip Drągowski [EMAIL PROTECTED] wrote: asterisk does daily log rotate all along ? i didn't know that it is posiiblei create file in

[asterisk-users] Re: bugs.digium.com

This is not a bug. It is just the way it works. The sip debug output is verbose output in asterisk console terminology. Also, the verbose setting in logger.conf has no effect for the console in logger.conf. Printing verbose output is only controlled by the set verbose CLI command. I do not

[asterisk-users] Re: gxp-2000 configure line appearances

Thanks this was exactly what I was looking for. Thanks Richard ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

[asterisk-users] Grand stream 2000 will not dial *xx

I just bought a grand stream 2000. It appears that it will not dial any number with a leading * (*70,*71) So I can not dial any of my Apps in * Can anyone point me in the right direction? Thanks, Rich ___ --Bandwidth and Colocation provided by

[asterisk-users] Flash operator panel

Can anybody steer me in the right direction? I have installed the fop and have it working okay, first problem is agent logins not changing the state color when an agent logs in. I configured it on two boxes one works the other doesn't, same configs alll the way. The other is more of me not

RE: [asterisk-users] Grand stream 2000 will not dial *xx

I just bought a grand stream 2000. It appears that it will not dial any number with a leading * (*70,*71) So I can not dial any of my Apps in * What firmware version are you running? We have plenty of GXP2000s in clients' premises with plenty of numbers beginning * without any problems.

Re: [asterisk-users] SNOM 360

I am trying to have thier PC run thru the port on the phone and the phone give prioroty to itself and the rest to the PC. When my client does a big download the phone call gets real bad. The docs from SNOM on TOS (or DIFFSERV) is poor and I dont understand it well enough. Anyone have configs

Re: [asterisk-users] SNOM 360

Also SNOM says by Vlan to set the vlan and then the value for the qos. When you set Vlan to 0 it is supposed to be no Vlan. However once I set it the vlan on the SNOM to 0 and I reboot the phone is no long accessable from the network and I have to reset it. Dovid - Original Message -

Re: [asterisk-users] cmd DIAL - Who picked up the call?

What about DIAL ( |M(macro-name)) and set the userfield in cdr during execution, ... http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Hi, if I do Dial(SIP/peer1/numberZap/g1/Number) can I somehow figure out who exactly picked up the call? In the cdrs dstchannel I can see the channel but

Re: [asterisk-users] IAX2 Connection fails over time...

On Friday 28 July 2006 07:51, Rich Adamson wrote: Any ideas on what we should check? Try changing the codec for the iax link to g726 and report back. Offhand, what are you suspecting? GSM's a pretty light codec in terms of CPU, and he's not lacking in the CPU department either (if this is a

[asterisk-users] sendtext or sip message - where in RFC

I was looking in apps/sendtext.c hoping to find a reference to the RFC number and section etc where this is talked about. Can someone point me where that information is for a SIP message? THanks, Jerry ___ --Bandwidth and Colocation provided by

[asterisk-users] Install asterisk-bristuff for Debian Linux

Following a discussion on this list about a week ago I downloaded and installed Debian Linux. Now I want to install asterisk-bristuff. How do I do that? Better yet, what do I put in /etc/apt/sources.list so I can do apt-get install asterisk-bristuff -- Thanks for your help, Cosmin Prund

Re: [asterisk-users] FreePBX Inbound Route

You could setup a ring group that included all extensions in your inbound route, the default for freepbx is to have an anydid/anycid route so any calls coming in will be sent to whereever you say (see the inbound routes link in freepbx). You will need to install the ring groups module from the

Re: [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger

Koen Van Impe wrote: I use logrotate too, because I didn't know of the functionality in Asterisk. Logrotate works fine for me though. Ok, I believe I see the problem here! I was told (apparently erroneously) that asterisk does rotation itself because they didn't rotate before and now they do.

Re: [asterisk-users] Grand stream 2000 will not dial *xx

Turn off the call features in the phone, by default the *70 codes are enable in the phone so that the phone can do call waiting and such. If you want asterisk to do this you need to disable the feature codes in the phone. On 7/28/06, Chris Bagnall [EMAIL PROTECTED] wrote: I just bought a grand

[asterisk-users] Change the from@ using the voicemail.conf

Hi, I'm trying to setup the voicemail.conf to email messages, but my mail server fails because the from user is [EMAIL PROTECTED] Does anybody know away to change the user part from root? I'm using exim4 to send the emails. Thanks, Dean. ___

[asterisk-users] Asterisk VOIP / Mikrotik

have a 10 mb ethernet connection from my ISP into ether1 on a PC - Mikrotik 2.9.23 installed. ether2 is the rest of my network behind the router. How do I prioritize packets such that VOIP calls ALWAYS get a "clean channel" through to my Asterisk server, which resides behind that router

Re: [asterisk-users] Install asterisk-bristuff for Debian Linux

Google is Your friend http://peen.net/2006/04/15/asterisk-1271-and-zaptel-125-for-debian-sarge/ Following a discussion on this list about a week ago I downloaded and installed Debian Linux. Now I want to install asterisk-bristuff. How do I do that? Better yet, what do I put in

Re: [asterisk-users] long distance ethernet Asterisk

It has been several years since I had to address similar situations, but I used TUT Systems devices back then. worked great. There are several DSL variants which should work ok. On Jul 27, 2006, at 6:02 PM, Manrique Feoli wrote: another thought, if you are in a bowl, all you need to find

RE: [asterisk-users] Change the from@ using the voicemail.conf

Hi Dean, In the voicemail.conf, in the [general] section near the top, I've got ; Who the e-mail notification should appear to come from [EMAIL PROTECTED] My e-mails now come from [EMAIL PROTECTED], making to easy to set up a filter in my e-mail client to move voicemail messages into a specific

[asterisk-users] Which card do you recommend for heavy load application?

I'm thinking to implement an application that may need 120 channels (4 E1 spans) being recorded in WAV49 format simultaneously, with echo cancellation, etc. What card would you recommend for this kind of load? (independently of the underlying hardware, assume the best possible). I've tested a

Re: [asterisk-users] sendtext or sip message - where in RFC

I was looking in apps/sendtext.c hoping to find a reference to the RFC number and section etc where this is talked about. Because sendtext.c is not SIP specific you will not find a reference to SIP related information there. chan_sip.c has a reference to RFC 3428

RE: [asterisk-users] long distance ethernet Asterisk

Brian, While I can't say we've used this specific product, I can say that anything we have used from RAD has been outstanding and highly reliable. http://www.rad-direct.com/App-Ethernet-extender-copper.htm?menuId2=Applicati onMenumenuId=Extenders2 For a season pass or two I'll come help you

Re: [asterisk-users] Install asterisk-bristuff for Debian Linux

Thanks! It works! (at first) I installed my deb from the given repository and I think it all went find. Asterisk starts up and I can get to the console. But... where are the drivers? updatedb / locate sees no zaptel drivers, and I've got none of the zapp tools on the system. Is that a

[asterisk-users] One extension to ring on multiple outside lines

I have a need to have a single extension actually ring on 2 phone lines which are not extensions (they are analog phone lines). Does anyone know a suitable extensions.conf config for this? David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED]

Re: [asterisk-users] One extension to ring on multiple outside lines

- Original Message - From: Dave Morrow [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Fri, 28 Jul 2006 11:34:37 -0300 Subject: [asterisk-users] One extension to ring on multiple outside lines I have a need to have a

Re: [asterisk-users] Voicmail Question

- Original Message - From: Kai Ober [mailto:[EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [mailto:[EMAIL PROTECTED] Sent: Fri, 28 Jul 2006 07:45:22 -0300 Subject: [asterisk-users] Voicmail Question Hi list, is it possible to pick up a call from

RE: [asterisk-users] Grand stream 2000 will not dial *xx

Here is the software version: Program-- 1.1.0.16Bootloader-- 1.1.0.1 When I pick up the line and dial *70 it just disappears and never dials. If I enable early dial it does dial *70 but then it breaks my outbound routes. Thanks Rich I just bought a grand stream 2000. It appears that

Re: [asterisk-users] registration process

- Original Message - From: unplug [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Fri, 28 Jul 2006 07:35:34 -0300 Subject: [asterisk-users] registration process Hi all, I wonder if there are 2 UAs having the same

Re: [asterisk-users] Canreinvite

- Original Message - From: Giordano Grandis [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Fri, 28 Jul 2006 07:01:08 -0300 Subject: [asterisk-users] Canreinvite How can I check if SIP re-invite is really working ?

RE: [asterisk-users] One extension to ring on multiple outside lines

Yes, to some extent it is what I want, but I want it to dial outside lines (ie. 800-555-1212 and 800-666-3434) insteand of a Zap channel. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615

Re: [asterisk-users] Grand stream 2000 will not dial *xx

does it reach the asterisk console? and have you turned off the dial features in the phone?On 7/28/06, Cavanna, Richard [EMAIL PROTECTED] wrote:Here is the software version:Program-- 1.1.0.16Bootloader-- 1.1.0.1When I pick up the line and dial *70 it just disappears and never dials.If I enable

[asterisk-users] Re: Grand stream 2000 will not dial *xx

Tom, Disabling the features worked. Thanks. Richard ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Transfer call in SIP

- Original Message - From: Victor Moreno [mailto:[EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Fri, 28 Jul 2006 06:57:48 -0300 Subject: [asterisk-users] Transfer call in SIP Hello, I am running TrixBox. if already in a call session from ZAPTEL to SIP, the user want to

Re: [asterisk-users] Install asterisk-bristuff for Debian Linux

I installed the following packages as well: ii libzap-dev 1.0.1-1 Zapata telephony interface library (developm ii libzap11.0.1-1 Zapata telephony interface library (runtime) ii zaptel

[asterisk-users] SMS functionality of bristuff (0.3.0-PRE-1r) with a Junghanns duo GSM PCI card

Hi all, I was quite excited to unearth the gsm send sms channel destination message command in chan_zap.c, but now I've hit a dead end in my efforts to deal with _received_ messages. When my card receives an SM, it prints a... -- SMS received on span 1. PDU: number ...message to the console,

Re: [asterisk-users] CDR IP Authorization

- Original Message - From: Khaled Chehab [mailto:[EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [mailto:[EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Fri, 28 Jul 2006 06:34:05 -0300 Subject: [asterisk-users] CDR IP Authorization Dear This function

[asterisk-users] Extending call parking to display park extension on the handset display

I am moving this thread from -dev to -users. The thread is below. Synopisis, I am relatively new to asterisk and even though I've looked through the docs I can not find a way to accomplish what I am trying to do. I am trying to, upon park, send a message to a SIP phone which will display the

RE: [asterisk-users] Change the from@ using the voicemail.conf

Bad form replying to myself, I know, but it looks like my outlook stripped the carriage return. Should be ; Who the e-mail notification should appear to come from [EMAIL PROTECTED] With the comment on the line above the serveremail line Cheers Mat -Original Message- From: [EMAIL

Re: [asterisk-users] PAP2T always busy on incoming calls with zaptel

- Original Message - From: Olivier MONNET [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Fri, 28 Jul 2006 05:11:35 -0300 Subject: [asterisk-users] PAP2T always busy on incoming calls with zaptel Hi, I'm starting to

Re: [asterisk-users] Multiple Outbound SIP Trunks

- Original Message - From: Aaron Anderson [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Fri, 28 Jul 2006 03:52:59 -0300 Subject: [asterisk-users] Multiple Outbound SIP Trunks I have 3 sip trunks registered with an

Re: [asterisk-users] One extension to ring on multiple outside lines

Dave Morrow wrote: Yes, to some extent it is what I want, but I want it to dial outside lines (ie. 800-555-1212 and 800-666-3434) insteand of a Zap channel. Then exten = 145,1,Dial(Zap/g1/18005551212IAX2/[EMAIL PROTECTED]/18006663434) Where g1 is defined in zapata.conf to go to a PRI or

RE: [asterisk-users] One extension to ring on multiple outside lines

- Original Message - From: Dave Morrow [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Fri, 28 Jul 2006 11:48:48 -0300 Subject: RE: [asterisk-users] One extension to ring on multiple outside lines Yes, to some extent

R: [asterisk-users] Canreinvite

Ok, thanks, also if i do not have rtp debug (i'm using asterisk 1.0.9) Hi -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Joshua Colp Inviato: venerdì 28 luglio 2006 12.54 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re:

Re: [asterisk-users] CSTA support for asterisk

- Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Sent: Fri, 28 Jul 2006 03:03:13 -0300 Subject: [asterisk-users] CSTA support for asterisk Hi, Can anybody tell me that is their CSTA support for asterisk Due to the fact that

Re: [asterisk-users] stream file outputs only silence, even with asterisk example gsm file

- Original Message - From: Guido Sohne [mailto:[EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Fri, 28 Jul 2006 08:19:21 -0300 Subject: [asterisk-users] stream file outputs only silence,even with asterisk example gsm file Hi all! I am trying to hook up a text to

Re: [asterisk-users] One extension to ring on multiple outside lines

Dave Morrow wrote: I have a need to have a single extension actually ring on 2 phone lines which are not extensions (they are analog phone lines). exten = 145,1,Dial(Zap/1Zap/2) That line would dial both Zap/1 and Zap/2 whenever someone called 145. The first one to answer gets the call. Is

Re: [asterisk-users] Message waiting question...

Hi On 7/27/06, Joshua Colp [EMAIL PROTECTED] wrote: I don't believe there's anything configurable but if you open app_voicemail.c there's two declarations, VOICEMAIL_DIR_MODE and VOICEMAIL_FILE_MODE which set the permissions. DIR mode is at 0770 right now and FILE mode is at 0660. Hum..

RE: [asterisk-users] Change the from@ using the voicemail.conf

Thanks, that worked great. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mat Stace Sent: 28 July 2006 14:58 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Change the from@ using the voicemail.conf Hi Dean, In

RE: [asterisk-users] comcast info -- somewhat offtopic

-Original Message- From: Derek Whitten [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 12, 2006 1:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] comcast info -- somewhat offtopic Martin Joseph wrote: On Jul 12, 2006, at 7:45 AM,

[asterisk-users] wav49 for voicemail attachment not playing

I'm trying to use the wav49 attachment, but it will not play on my machine. I'm running windows xp with media player 10, it comes up with codec 'Microsoft GSM 6.10' not available. Microsoft stated that the GSM 6.10 is included in media player 10. Has anybody else had this problem? Could it be the

[asterisk-users] asterisk cdr shows FAILED

Hi, I'm having trouble in that my asterisk cdr is showing a lot of calls failing. The asterisk cdr shows disposition FAILED, and the last app is: DialIAX2/[EMAIL PROTECTED]/12125551234 I removed my username and changed the phone number there. Any idea what causes this, and how I can

[asterisk-users] Source Directory of ASterisk

Hi, I am using TriBox 1.1.1/Asterisk. I want to know where I can find source directory of Asterisk in system so I can install Asterisk audio conversion module (http://redice.krisk.org/res_conv-0.1.tgz) to convert ulaw prompts into g729 prompts. It requires to point Asterisk source Include

Re: [asterisk-users] CSTA support for asterisk

http://www.google.com/search?hl=enq=define%3A+CSTAbtnG=Google+Search On 7/28/06, Joshua Colp [EMAIL PROTECTED] wrote: - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Sent: Fri, 28 Jul 2006 03:03:13 -0300 Subject: [asterisk-users]

Re: [asterisk-users] Flash operator panel

Hi Jordan, You might want to subscribe to FOP mailing list. You can do that from http://www.asternic.org Can anybody steer me in the right direction? I have installed the fop and have it working okay, first problem is agent logins not changing the state color when an agent logs in. I

RE: [asterisk-users] Re: bugs.digium.com

-Original Message- From: Steven [mailto:[EMAIL PROTECTED] Sent: Friday, July 28, 2006 6:44 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: bugs.digium.com This is not a bug. It is just the way it works. The sip debug output is verbose output in asterisk

Re: [asterisk-users] SNOM 360

I would be surprised if the problem is at the phone. I have nearly a hundred 360s, 190s and not one of them suffers from that problem in the default setting. The phone handles it automatically. BUT..if I download from an external site and I pipe the call over the internet without setting any

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