[Asterisk-Users] CallerID retain on internal transfer

I was just looking through the Wiki for some info on how to retain the original caller's callerid when make transfers to internal extensions, and I came across the parameter below: useincomingcalleridonzaptransfer=yes There is nothing in the zapata.conf file from vers. 1.2.7 so I am wondering if

RE: [Asterisk-Users] CallerID retain on internal transfer

I've been wondering about how to do what you're describing as well! Anyone know how it can be done? Regards, Bevan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Sent: Monday, 8 May 2006 4:03 PM To: asterisk-users@lists.digium.com Subject:

[Asterisk-Users] asterisk management interface

Hi, I have to make a web-based management interface of configuring asterisk i wanted to know if it is as simple as reading the .conf files and searching for the required section in the file and adding users etc. or there are other steps involved too?? As I have seen many such built codes

RE: [Asterisk-Users] asterisk management interface

Why make a brand new? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of moona ather Sent: Sunday, May 07, 2006 11:36 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] asterisk management interface Hi, I have to make a web-based

RE: [Asterisk-Users] asterisk management interface

As I know only php and no other langugae like perl or any other... most of the links to such applications i have seen on voip.org site made in php are removed or are inactive. Can you tell me of any such application that i can use or make my own using that made only in php and serving my

[Asterisk-Users] iax2: dropping too many packets

i all, I need help on a trunk between two Asterisk servers. They sitting at different branches, so they connected via a leased line WAN. Calls originate from users at Branch A then go IAX to users at Branch B. at branch A the call sounds absolutly fine, but at branch B there are a few

RE: [Asterisk-Users] asterisk management interface

http://www.freepbx.org Why would you need to create your own? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of moona ather Sent: Monday, May 08, 2006 12:01 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] asterisk

[Asterisk-Users] gxp-2000 Asterisk PSTN

Hi, I have Grandstream GXP-2000 connected to Asterisk, and Asterisk has trunk to VSP for PSTN calls. When ever I place local PSTN call, the landline doesn't hang up right away (40 sec), when I hang up the GXP-2000. The GXP-2000 seems to have problems making international calls as well. Where it

Re: [Asterisk-Users] asterisk management interface

You can't possibly read conf files and present them to the user in a web-based form because of the complexity of the data in those file. Take the extensions configuration file for an example (as this is probably the most complex one). Settings in this file might specify a simple mapping

RE: [Asterisk-Users] asterisk hardware

Hi, I have ~25 AtCom AT320 phones (PA1888S based) and they was not a good experience for me: I ran the risk to abort my Asterisk project thanks to them! I tried both with SIP and IAX2 firmware of every version but voice quality was often unacceptable. In addition, they lose often registration and

AW: AW: [Asterisk-Users] DTMF detection when outgoing call tomobilephones

I have made a step forward regarding my problem. It seems to be a routing problem of the telco provider. The provider is using different routings depending on the called number. We have successfully tested a different routing than the standard one. So thank you for now for the support! Cheers.

[Asterisk-Users] about billing realtime (maybe OT)

I've followed with interest the discussion about realtime billing, anyway, even if it could be a fascinating subject as a developer, I've always felt that from a project management point of view the problem is simply non-existent, because the money lost with wide-grained control is unimportant

[Asterisk-Users] Asterisk 1.2.x with app_rxfax

Does anyone has a working one? mine always receives the fax, but then cannot set back the format and goes away :-( PLease help! -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and

[Asterisk-Users] Junghanns GSM card

Hi,Does anyone have some experience with junghanns GSM cards? I want to know if I can use this cards to send SMS directly from Asterisk box.Cheers Andrew ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To

Re: [Asterisk-Users] Junghanns GSM card

http://www.voip-info.org/wiki/view/VOIP+GSM+Gateways Andrew Nowrot on 5/8/2006 17:08 wrote: snip Does anyone have some experience with junghanns GSM cards? I want to know if I can use this cards to send SMS directly from Asterisk box. snip -- Jim Kou IT Engineer Malico Inc.

RE: [Asterisk-Users] asterisk management interface

[EMAIL PROTECTED] wrote: http://www.freepbx.org Why would you need to create your own? Many reasons: 1. not relying on already busy open source developers 2. creating something that you can possibly offer as your own commercial offering 3. have it designed exactly they way you want it from

Re: [Asterisk-Users] H323 to SIP

Farhad Ibragimov wrote: I don’t have practice to work with Asterisk but I see that is a great soft. If you have any idea or some config files can you help me Asterisk is perfectly documented everywhere on the net. Maybe the first place to visit in order to have working asterisk is

Re: [Asterisk-Users] Junghanns GSM card

Andrew Nowrot a écrit : Hi, Does anyone have some experience with junghanns GSM cards? I want to know if I can use this cards to send SMS directly from Asterisk box. They look terrible to me. From the picture on their website it looks like you need one antenna per GSM channel (most

[Asterisk-Users] Session Border Controller (SBC)

Hello to All I have been trialling a SBC for my VoIP solution but the problem I have is that the SBC needs to have an entry in the sip.conf inorder for the asterisk to accept registration requests from the SBC. This is ok if I only have one company registering as the entry in sip.conf for the

Re: [Asterisk-Users] Junghanns GSM card

Does anyone have some experience with junghanns GSM cards? I want to know if I can use this cards to send SMS directly from Asterisk box. They look terrible to me. From the picture on their website it looks like you need one antenna per GSM channel (most gateways use 1 antenna per 4

[Asterisk-Users] [nativecode=Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (111)]

Hello, I have an error when installing AMP, when I do ./install_amp --debug, it show me : Connecting to database..FAILED [DEBUG] [nativecode=Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (111)] ** mysql://user:[EMAIL PROTECTED]/asteriskamp Try running ./install_amp

[Asterisk-Users] app_wakeme.c (Wake-up Call Manager) v0.1.0 released

Greetings, I'm releasing my first attempt at Asterisk development in hopes of gathering a bit of feedback. It's just a wake-up call manager but, seeing as the only one I was able to find was a php-agi script, perhaps an actual app will be useful to someone. Anyway, here's the details.

[Asterisk-Users] Voicemail bomb

I submitted a bug to the tracker (bug)regarding the 256 character limit when copying a voicemail to a list of mail boxes. The bug was closed with this note: Fixed in 1.2 branch, merged to trunk. Could someone explain to me what that means... in English? I searched the release notes of the

[Asterisk-Users] (no subject)

Good day, Hi! i've finish up setting * for my company and they are working reallly great, but i notice when i try to call to mobile phone, i can see the zap channels is bridging successfully but i hear nothing except for a long dialtone like tone, but calling to a regular pots line is working

RE: [Asterisk-Users] app_wakeme.c (Wake-up Call Manager) v0.1.0 released

Any help? # gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC -c -o app_wakeme.o app_wakeme.c app_wakeme.c: In function `app_exec': app_wakeme.c:101:

Re: [Asterisk-Users] Voicemail bomb

seems to me that the bug is fixed. That description means that the bug is fixed in the current 1.2 branch that will become, probably with other modifications, soon or later the next 1.2.x release, and in the trunk branch, that will become the first 1.4.x release. To have the fixed source code

[Asterisk-Users] Hi...Please help me

Hi Friends, Thank you for your quick response. I have successfully implemented Intercom (Dialling within my office LAN) using Asterisk. To implement this, I am using X-Lite Softphone. Now, I want to make calls to US using VoIP Asterisk. I have registered with Vebtel (VoIP IP Telephony Service

RE: [Asterisk-Users] Hi...Please help me

Chandra, In all honesty if they are proprietary and you want to use them you will need a FXO card.  Alternatively there are a few good termination providers out there that are inexpensive. The top 3 most inexpensive that come to mind are: Plainvoip  -  

[Asterisk-Users] Quad ISDN card

Hi all, Somebody know if the AVM C4 Quad ISDN card is supported by the current asterisk version? regards rene ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] [nativecode=Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (111)]

Hi,it cannot find mysql.sock. Is it your mysql running?2006/5/8, ali asma [EMAIL PROTECTED]: Hello, I have an error when installing AMP, when I do ./install_amp --debug, it show me : Connecting to database..FAILED [DEBUG] [nativecode=Can't connect to local MySQL server through socket

[Asterisk-Users] Re: Voicemail bomb

Or, if you go to http://svn.digium.com, you can drill into the code, fix your bug and look at the diff. Yours is at http://svn.digium.com/view/asterisk/branches/1.2/apps/app_voicemail.c?r1=19394r2=19891sortby=date -- -- Steven http://www.glimasoutheast.org "picciuX" [EMAIL PROTECTED]

Re: [Asterisk-Users] [nativecode=Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (111)]

yes, I have mysql 4.1.18 runs Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Cliquez ici.___ --Bandwidth and

Re: [Asterisk-Users] Quad ISDN card

On Mon, 8 May 2006, René Enskat [Teamware GmbH] wrote: Hi all, Somebody know if the AVM C4 Quad ISDN card is supported by the current asterisk version? I did not test this card, but since it has a CAPI 2.0 interface, it should work with chan-capi.org and Asterisk/OpenPBX. If you need more

[Asterisk-Users] Dialstatus results

Hi all, i just have a question: could i Known the state of a SIP phone without make it a Dial ? Thanks Giordano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options

Re: [Asterisk-Users] [nativecode=Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (111)]

so, check if mysql.sock is in /tmp/mysql.sock insted of /var/lib/mysql/mysql.sock.If so, change the value into AMP.2006/5/8, ali asma [EMAIL PROTECTED] :yes, I have mysql 4.1.18 runs Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez

[Asterisk-Users] Re: Odd internal vs. External dialplan issue

OK, I thinks I have narrowed it down. Our old Legacy PBX is choking on the callerID name. I have a separate issue, where I am not getting the CallerID name from our Telco yet, so incoming Telco calls forward fine to the legacy PBX. Asterisk to Legacy PBX calls transmit the CallerID name and our

Re: [Asterisk-Users] app_wakeme.c (Wake-up Call Manager) v0.1.0 released

Any help? # gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE [snip] make[1]: Leaving directory `/usr/src/asterisk-1.2.7.1/apps' make: *** [subdirs] Error 1 1.2.7.1 is your problem. I've updated the site with

Re: [Asterisk-Users] asterisk hardware

On Sunday 07 May 2006 03:21, Wilson Pickett wrote: I have had three of them for neary two years. Here's an executive review: Which model/vendor? -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE

[Asterisk-Users] canreinvite=no and codecs.

Hi, folks. If i use canreivite=no option in my sip.conf for users is this mean that i need to load 729 and 723 codecs for thos UA that want to transmit it? Or this is just traffic redirection feature? How this option reflect on server load? --

Re: [Asterisk-Users] [nativecode=Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (111)]

it exists in both directories Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Cliquez ici.___ --Bandwidth and

Re: [Asterisk-Users] Junghanns GSM card

Hi,Thanks for all repliesCan anyone tell me if it is possible to send the SMS through this card directly from extensions.conf with some application that takes the text string and converts it to SMS and which colaborates with junghanns card. CheersAndrew

[Asterisk-Users] MySQL replication for voicemail

Hi - We've got a number of offices, and they're all using ODBC message storage using MySQL. I've been trying to get MySQL replication set up so messages left in a voicemail box at one office will get copied to the corresponding voicemail box at all the offices. We're also using MySQL

Re: [Asterisk-Users] Asterisk -- NAT -- Internet -- NAT -- Sipura-3K (No Asterisk)

Joseph: I think that was under the sip tab you have to configure the sip proxy, just put the address of the firewall where you are doing port forwarding to the asterisk box. Also in the nat field at the sipura device put yes. And finally you have to open and forward the UDP ports of the

RE: [Asterisk-Users] app_wakeme.c (Wake-up Call Manager) v0.1.0released

-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Iedema 1.2.7.1 is your problem. I've updated the site with a 1.2 compatible version as well as squashed a stupid bug in v0.1.0. At the moment I don't have access to a 1.2 test box so let

Re: [Asterisk-Users] app_wakeme.c (Wake-up Call Manager) v0.1.0released

app_wakeme.c:465: warning: type defaults to `int' in declaration of `STD_MOD1' STD_MOD1 isn't even found in the version for 1.2. You're still using the incorrect version. Download the 1.2 version and let me know... https://tk-labor.dyndns.org:8443/pub/files/app_wakeme/app_wakeme_v0.1.1_1.2.c

[Asterisk-Users] UpState NY SIP provider

Hi, Anyone has good/bad experience with SIP providers in upstate NY? Any recommendations of such provider who works great with Asterisk? Thanks, Andre Courchesne ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing

RE: [Asterisk-Users] Call Center Phone with Auto Answer

Firstly the auto-answer on both the 301 and 501 phone is set to on, but it doesnt seem to have an effect. Ill have to look into this _ALERT_INFO variable. Not much experience with it here. Could you give me a dial plan example that would work? Here is what we have now.

Re: [Asterisk-Users] Quad ISDN card

René Enskat [Teamware GmbH] wrote: Somebody know if the AVM C4 Quad ISDN card is supported by the current asterisk version? It is supported. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --Bandwidth and Colocation provided by

Re: [Asterisk-Users] asterisk management interface

start shameless plug I invite you to look at our interface PhoneCALL. I designed it from the ground up, all 100% php. If anything, just to learn how it's done. http://www.vecsector.com/phonecall end shameless plug Thanks! --Dustin moona ather wrote: As I know only php and no

[Asterisk-Users] AstLinux 0.4 Released - with build system

Hello Everyone, After many months (and seemingly no updates), I am happy to announce AstLinux 0.4. AstLinux 0.4 was built with the new AstLinux build system. The AstLinux build system is based off of buildroot2 and includes everything you need to build AstLinux (and more). It is very

Re: [Asterisk-Users] UpState NY SIP provider

I'm using www.widevoip.com with asterisk, it works great. They provide us a VPN connection and a SIP user. Regards. Lucas Alvarez Andre Courchesne - Consultant wrote: Hi, Anyone has good/bad experience with SIP providers in upstate NY? Any recommendations of such provider who works great

Re: [Asterisk-Users] MySQL replication for voicemail

I have voicemessage ODBC storage working and MySQL replicating, but I didn't setup the replication after-the-fact, therefore didn't need the 'LOAD DATA FROM MASTER'. You may want to try a different method (more manual) of synching your machines. Does your voicemessages table contain specially

RE: [Asterisk-Users] app_wakeme.c (Wake-up Call Manager)v0.1.0released

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Iedema ... You're still using the incorrect version. Download the 1.2 version and let me know... OK, solved. Thank you. DV ___ --Bandwidth and Colocation provided by

Re: [Asterisk-Users] CallerID retain on internal transfer

Joe wrote: I was just looking through the Wiki for some info on how to retain the original caller's callerid when make transfers to internal extensions, and I came across the parameter below: useincomingcalleridonzaptransfer=yes There is nothing in the zapata.conf file from vers. 1.2.7 so I am

[Asterisk-Users] duration / billsec problem

hi guys, I am experiencing some problems with cdr. I have a provider with whom I am connected with SIP. I use asterisk (of course), they use cisco. Sometimes when I make a call, my asterisk is logging the call like this: Disposition: Answered Duration: 13 Billsec: 0 When in fact the call lasted

Re: [Asterisk-Users] app_wakeme.c (Wake-up Call Manager)v0.1.0released

OK, solved. Thank you. Congrats :) Let me know how it works out for you. --Michael I. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

RE: [Asterisk-Users] asterisk management interface

Those are reasons for WANTING to create your own, he specifically said I HAVE to make my own and I wanted to know why he HAS TO create his own when there are fantastics products already available. There is a huge difference in saying I would like to create my own and I have to create my own. I

[Asterisk-Users] Dialing status detection

Hi, Anyone has hints to share about dialing result detection. By that I mean the ability to detect what answered: - Human - Answering machine - Fax - Disconnected number. Any hints or pointers appreciated. Andre Courchesne ___

Re: [Asterisk-Users] FOP flash panel: how to reload config files when running

I just issue a 'service op_panel reload' at the command line and it reloads the configuration without stopping the service :)On 5/7/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:On Sun, May 07, 2006 at 08:44:41AM -0400, Doug Lytle wrote: Wilson Pickett wrote: No, you have to kill the op_server app

RE: [Asterisk-Users] Call Center Phone with Auto Answer

Does the phone ring, just not auto-answer? Thanks, Steve Totaro From: Kevin Savoy [mailto:[EMAIL PROTECTED] Sent: Monday, May 08, 2006 10:17 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Call Center Phone with Auto

Re: [Asterisk-Users] asterisk management interface

yet another shameless plug Sorry, but this made me think of something else. PhoneCALL is 100% capable of being rebranded and resold as your own if you get a developer license from us. Furthermore, we can also function as the programming resource for the business, so all you have to worry

[Asterisk-Users] Expansion module

Hello All, Does anyone know of an expansion module (keypad extension for attendant) that works with the gxp-2000? - Jason ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options

[Asterisk-Users] How do I monitor a Zap channel ...

How do I monitor a Zap channel as soon as the telephone is off the hook, till it is on the hook again? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

RE: [Asterisk-Users] Call Center Phone with Auto Answer

Correct. We have to hit the answer button. In a call center environment such as ours we dont want to give the agents the option of not answering the call when they are logged in. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, May 08,

RE: [Asterisk-Users] Expansion module

Its not available just quite yet, should be out in a few weeks time. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL

Re: [Asterisk-Users] Re: Voicemail bomb

Thanks for the input guys. Nice to understand what's going on. Steven, how do you know that particular diff is the fix for that particular bug (not that I doubt you, just curious how to reference the code to the bug). Thanks again, Garth On 5/8/06, Steven [EMAIL PROTECTED] wrote: Or, if you

RE: [Asterisk-Users] Call Center Phone with Auto Answer

Why not just use AgentLogin and let them listen to music until a call comes in? Thanks, Steve Totaro From: Kevin Savoy [mailto:[EMAIL PROTECTED] Sent: Monday, May 08, 2006 11:27 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE:

Re: [Asterisk-Users] How do I monitor a Zap channel ...

I'm using ChanSpy. Set up an extension with ChanSpy, dial the given extension and don't hang up (put it on speakerphone). When there's no one on the given zap channel you'll here silence. As soon as someone's on the channel you'll be listening to them. If you don't like ChanSpy there's

RE: [Asterisk-Users] Call Center Phone with Auto Answer

Because we will have many of these phones in remote locations and we dont want to be chewing up bandwidth with agents not on calls. Am I making the right assumption here that phones that are idle will not be taking up bandwidth where ones with MOH playing would be? From: [EMAIL

Re: [Asterisk-Users] MySQL replication for voicemail

Hi Gary - I have voicemessage ODBC storage working and MySQL replicating, but I didn't setup the replication after-the-fact, therefore didn't need the 'LOAD DATA FROM MASTER'. You may want to try a different method (more manual) of synching your machines. Yes, I've been reading about some

RE: [Asterisk-Users] Call Center Phone with Auto Answer

That is correct. Just use IAX trunking and speex. You will be fine. Thanks, Steve Totaro From: Kevin Savoy [mailto:[EMAIL PROTECTED] Sent: Monday, May 08, 2006 12:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Call

RE: [Asterisk-Users] asterisk management interface

[EMAIL PROTECTED] wrote: Those are reasons for WANTING to create your own, he specifically said I HAVE to make my own and I wanted to know why he HAS TO create his own when there are fantastics products already available. There is a huge difference in saying I would like to create my own and

Re: [Asterisk-Users] Junghanns GSM card

Can anyone tell me if it is possible to send the SMS through this card directly from extensions.conf with some application that takes the text string and converts it to SMS and which colaborates with junghanns card. i think yes, and sure for voismart -- WoodOO-[P]an[G]alaktikan[A]gent-People

[Asterisk-Users] I: Dialstatus results

Hi all, i just have a question: could i Known the state of a SIP phone without make it a Dial ? Thanks Giordano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update

[Asterisk-Users] Running down an echo problem on outgoing calls

HiI am having some echo on outgoing calls on a pstn line (made from a sip phone)and am looking into some fixes involving rxgain and txgain, butbefore I go down that road, I want to make sure I have everythingsetup correctly.My zapata.conf has cat /etc/asterisk/zapata.conf [trunkgroups]; define

[Asterisk-Users] Re: Re: Voicemail bomb

I made some assumptions. I drilled into apps/voicemail.c and looked for reference to 256 character limit. The note on that change is "Bug 6947 - Allow vm broadcasts to more than 256 characters worth of mailboxes" But it would be nice if the bug system referred to theSVN change or at least

[Asterisk-Users] Most comprehensive management?

I see that [EMAIL PROTECTED] and FreePBX are going along similar lines with web based management interfaces.. My Asterisk box has one analog phone, one analog line, 3 SIP phones, IAX inbound numbers, an IAX outbound trunk, IVR menus and voicemail boxes in different contexts for each of the

RE: [Asterisk-Users] MySQL replication for voicemail

Another approach you may want to consider for data redundancy that does not rely on MySQL's finicky replication stuff is DRBD. Think of it as RAID-1 across Ethernet. I have used it in production on some VERY busy ( 1200 qps) MySQL servers for a couple years with no problems. Another nice side

Re: [Asterisk-Users] Running down an echo problem on outgoing calls

snip echocancel=yes echocancelwhenbridged=yes You probably want that off, since when you bridge 2 ZAP channels you should not have any echo. echotraining=yes snip But, when I run asterisk and ask it about the zap channels, I get that echo cancellation is OFF. zap show channel 2Echo

[Asterisk-Users] Message on Hold

Hi, I know that I can have an AutoAttendent menu play when someone is in a queue to say something like Press 1 now to leave a message, or to continue holding stay on the line... However, is there anyway to prevent that from happening until the caller has been on hold for say 5 minutes? In other

Re: [Asterisk-Users] Most comprehensive management?

It will do these things just fine. FYI [EMAIL PROTECTED] is an ISO that will setup linux, FreePBX and various other things for you, you can of course just download FreePBX and install it by yourself. On 5/8/06, WipeOut [EMAIL PROTECTED] wrote: I see that [EMAIL PROTECTED] and FreePBX are going

[Asterisk-Users] Looking for New Service Provider

Must sell DID's in the U.S at a fair price. Must pass Caller ID number info on both 1.area-code calls as well as on ALL 800 calls. I do not expect something for free but, I must pass Caller ID so that the distant end will see the Calling Parties number for verification. email [EMAIL PROTECTED]

Re: [Asterisk-Users] Most comprehensive management?

Another FYI: The latest [EMAIL PROTECTED] release (2.8) includes FreePBX (sounded like you thought they were seperate entities). It should be able to manage anything a roll-your-own server could. On 5/8/06, Tom Vile [EMAIL PROTECTED] wrote: It will do these things just fine.FYI [EMAIL PROTECTED]

[Asterisk-Users] Asterisk/Zaptel 64-bit?

Dear All, I was wondering will there be any problems or changes that I will need to do to compile the current Asterisk(1.2.7)/Zaptel(1.2.5)/Libpri(1.2.2) source from www.asterisk.org into a 64-bit binaries? I am currently using the following hardware for my new server. CPU: Pentium D 930 3.0

[Asterisk-Users] Non-supervised pass-through

I'm trying to get asterisk to pass through a call without requiring supervision on the line. Any thoughts? Thanks, Frank ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options

Re: [Asterisk-Users] Non-supervised pass-through

Frank Garcia wrote: I'm trying to get asterisk to pass through a call without requiring supervision on the line. Any thoughts? This is the default unless you do something to answer the line. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery.

RE: [Asterisk-Users] Call Center Phone with Auto Answer

Ok I can get this to work now the next problem is since the agent stays off-hook when a call is presented to them there is no indication of what call this is. Being an inbound call center we have 100s of clients. 1,000s of toll frees and DNIS. We use the Asterisk callerID function to

RE: [Asterisk-Users] Dialing status detection

Hiya, the disconnected number and fax are fairly easy to detect, as the behavior on these connections is consistent. Answering machines and Human is a bit trickier. You can : 1) do a Background detect for noise to try and pick the hello, then a pause that happens when a human answers. 2)

Re: [Asterisk-Users] Voicemail indication for analog phones

Any CLASS MWI should also work. I do believe there is support at least on the Digium cards, and I think the Vega also. I am just installing a Vega now and will try to verify, but it might be awhile before I can. On May 7, 2006, at 3:46 PM, Garth Summey wrote: I have 20 or so users with

[Asterisk-Users] transfer variables

Can I get the channel name (and variables) of the call the iniated the transfer? It looks like the Transfer() application makes a new local channel and goes back to the context of the SIP client, and in there I need to do things differently based on who started the transfer.

[Asterisk-Users] PSTN Incoming call on real line disrupts VoIP call over DSL circuit

I haven't seen anything this strange, and it's 100% reproducible. I'm hoping that there are some clever ideas out there for what to look for, since I can test to my heart's desire on this one... My Dad lives in Florida, and has a Bellsouth DSL line. Of course, he has a regular POTS line

Re: [Asterisk-Users] PSTN Incoming call on real line disrupts VoIP call over DSL circuit

I would guess either the DSL itself is bad or perhaps the dsl Modem. perhaps calling Bellsouth would be helpful? Does other Internet traffic get interrupted also? On May 8, 2006, at 1:42 PM, Hadar Pedhazur wrote: I haven't seen anything this strange, and it's 100% reproducible. I'm

Re: [Asterisk-Users] PSTN Incoming call on real line disrupts VoIP call over DSL circuit

Jerry Jones wrote: I would guess either the DSL itself is bad or perhaps the dsl Modem. perhaps calling Bellsouth would be helpful? Does other Internet traffic get interrupted also? The rest of the Internet traffic is _not_ interrupted, and his voice continues to be heard, so even that part

Re: [Asterisk-Users] PSTN Incoming call on real line disrupts VoIP call over DSL circuit

Well, I have no idea how DSL lines are connected in the US but what happens to a normal Internet connection when the phone is being picked up ? Test scenario could be that your Dad is listening to an Internet radio station or other audio stream and then being called BTW, how are the real

RE: [Asterisk-Users] Call Center Phone with Auto Answer

Not sure if it is possible or not but you could record the company names, use cepstral, or festival. The part I am unsure of is whether you can use a variable in the announce= field in queues.conf. Thanks, Steve Totaro From: Kevin Savoy [mailto:[EMAIL PROTECTED] Sent:

Re: [Asterisk-Users] MySQL replication for voicemail

Hi Josh - Another approach you may want to consider for data redundancy that does not rely on MySQL's finicky replication stuff is DRBD. Think of it as RAID-1 across Ethernet. I have used it in production on some VERY busy ( 1200 qps) MySQL servers for a couple years with no problems. I would

Re: [Asterisk-Users] Call Center Phone with Auto Answer

Ok I can get this to work now the next problem is since the agent stays off-hook when a call is presented to them there is no indication of what call this is. Being an inbound call center we have 100's of clients. 1,000's of toll frees and DNIS. We use the Asterisk callerID function to assign a

Re: [Asterisk-Users] PSTN Incoming call on real line disrupts VoIP call over DSL circuit

Juergen K. Zick wrote: Well, I have no idea how DSL lines are connected in the US but what happens to a normal Internet connection when the phone is being picked up ? Test scenario could be that your Dad is listening to an Internet radio station or other audio stream and then being called

RE: [Asterisk-Users] Call Center Phone with Auto Answer

This may be the way to go but not the best. Our agents frankly aren't the brightest people and I can see them forgetting it as soon as it is said to them, or they are not paying attention and missing the announcement but it is something to look into. Thanks -Original Message- From:

Re: [Asterisk-Users] PSTN Incoming call on real line disrupts VoIP call over DSL circuit

Hadar Pedhazur wrote: Jerry Jones wrote: I would guess either the DSL itself is bad or perhaps the dsl Modem. perhaps calling Bellsouth would be helpful? Does other Internet traffic get interrupted also? The rest of the Internet traffic is _not_ interrupted, and his voice continues to be

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