Re: [asterisk-users] X-Lite Softphone keeps de-registering?

Doug wrote: > The client is travelling much of the time. > > Is there some way that he can use Port 80 so > that the firewalls that he is behind won't > block the connection? > > Any other hints or suggestions are very > welcome! I suppose the usual answers apply. You can have Asterisk bind to

Re: [asterisk-users] "Real" API for Perl?

Ken D'Ambrosio wrote: > Hi, all. I've used the perl/AGI interface, and... well, I found it kind > of hokey. Granted, this was in 1.2 days -- perhaps things have changed. > Regardless, I guess I have two questions: > 1) Has the Perl/AGI "binding" improved since then? > 2) Is there any chance of

Re: [asterisk-users] BRI card with PCI-E interface

Hi Alberto, 2008/2/1, Alberto Pastore <[EMAIL PROTECTED]>: > > Olivier ha scritto: > > Hi, > > > > Does such card exist ? > > It seems all existing models are designed for PCI buses. > > > > Regards > > > > > > > > > > __

Re: [asterisk-users] Echo() app doesn't work

On Fri, Feb 01, 2008 at 05:01:56PM -0800, Yassen Damyanov wrote: > Hello list, > > New to asterisk and to the list (although experienced in Unix/Linux > administration). > > Short problem description: > -- > I cannot get the Echo() application to run on any 32bit platform

Re: [asterisk-users] Polycom - Buddy Watch not a choice when adding Speed Dial

On Feb 1, 2008 3:53 PM, Thermal Wetland <[EMAIL PROTECTED]> wrote: > Hello, > > On our Polycom phones we can not activate the Buddy Watch feature. > > When you add or edit a contact, the list ends at "Auto Divert".I know > it is the end of the list b/c the down arrow on the right side of the s

[asterisk-users] Polycom - Buddy Watch not a choice when adding Speed Dial

Hello, On our Polycom phones we can not activate the Buddy Watch feature. When you add or edit a contact, the list ends at "Auto Divert".I know it is the end of the list b/c the down arrow on the right side of the screen disappears when I get to Auto Divert. When I add 1 manually to the spee

[asterisk-users] Echo() app doesn't work

Hello list, New to asterisk and to the list (although experienced in Unix/Linux administration). Short problem description: -- I cannot get the Echo() application to run on any 32bit platform I can get my hands on. In contrast, the only 64-bit (amd64 aka x86_64) setup that

[asterisk-users] IAX Registraion Refresh

I have Asterisk 1.4 registering via IAX to another Asterisk machine. How can I change the default registration timeout of 60s? I need my Asterisk box to register every HOUR Anyone? Editting source isn't an option. Doug. _

Re: [asterisk-users] Asterisk 1.4.17 and Teliax DTMF

Carlos Chavez wrote: > I am having a problem with DTMF when sending calls through Teliax > (SIP). In the peer for teliax I defined dtmfmode=rfc2833 and for the > most part it is working. The problem always happens when a user is > trying to call a conference system. They simply cannot get

[asterisk-users] "Real" API for Perl?

Hi, all. I've used the perl/AGI interface, and... well, I found it kind of hokey. Granted, this was in 1.2 days -- perhaps things have changed. Regardless, I guess I have two questions: 1) Has the Perl/AGI "binding" improved since then? 2) Is there any chance of a "real" API for Perl? Thanks mu

[asterisk-users] X-Lite Softphone keeps de-registering?

The client is travelling much of the time. Is there some way that he can use Port 80 so that the firewalls that he is behind won't block the connection? Any other hints or suggestions are very welcome! ___ -- Bandwidth and Colocation Provided by http:

[asterisk-users] Trying to make SIP calls through Asterisk with anonymous connection

I am trying to setup SIP to SIP calling between Asterisk managed networks. I want to make it so that people can call SIP:[EMAIL PROTECTED] and they connect to my Asterisk and get my external IVR then they can dial my extension or navigate extensions just like they would if they had called using a P

Re: [asterisk-users] It's about time! -- Digium PCI-Express Cards

On Friday 01 February 2008 15:31, Matt wrote: > It's about time Digium got on the ball and made PCI-e cards. What are > people's experiences with this card? Anyone know if there are plans for a > PCI-e analog card for FXO use? I have been using 220B's for about 6 months. I have about 20 of them

Re: [asterisk-users] Enterprise or Fedora?

"shadowym" <[EMAIL PROTECTED]> writes: > I cannot think of a single reason to use Fedora for a production anything > when there are alternatives like CentOS. Fedora is bleeding edge stuff and > constantly changing. The advantage of Fedora is that it is very actively maintained -- and asterisk is

Re: [asterisk-users] Asterisk 1.4.17 and Teliax DTMF

At 10:57 AM 2/1/2008, you wrote: > Any tweaks recommended for DTMF and Teliax? try: dtmfmode=auto That's what mine is now after rfc2833 stopped working. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk

Re: [asterisk-users] QueueMember event/LastCall Variable - Format?

Jared Smith wrote: > On Fri, 2008-02-01 at 15:32 -0500, Lee Jenkins wrote: >> What format is the LastCall variable of QueueMember event? I'm looking at: >> 1201897536 for instance. > > Unix epoch format, or number of seconds since Jan 1, 1970 UTC, as I > recall. > Thanks. -- Warm Regards, Le

Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's

- Original Message > From: Mindaugas Kezys <[EMAIL PROTECTED]> > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Sent: Friday, 1 February, 2008 4:04:30 PM > Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's > >For such cases we usually suggest to p

Re: [asterisk-users] SIP Softphones and Citrix ?

I'm working for zoiper.com and i'm willing to help out with ours when needed. Zoa d4rk f1br wrote: > Anyone aware of any SIP softphones that might virtualize well with > Citrix presentation server? I suspect I know the answer already as I > have been researching softphones that work with Ci

Re: [asterisk-users] It's about time! -- Digium PCI-Express Cards

Jason.. great! How long have those analog cards been out? I don't see them on my suppliers list. > > Digium already makes PCI Express analog cards - AEX800 and AEX2400. > > -- > Jason Parker > Digium > > ___ > -- Bandwidth and Colocation Provided by

Re: [asterisk-users] It's about time! -- Digium PCI-Express Cards

Matt wrote: > Just noticed this today: > > Digium TE220B Dual Span T1/E1 PCI Express Card with Octasic DSP-based > Echo Cancellation Module > > > It's about time Digium got on the ball and made PCI-e cards. What are > people's experi

Re: [asterisk-users] QueueMember event/LastCall Variable - Format?

On Fri, 2008-02-01 at 15:32 -0500, Lee Jenkins wrote: > What format is the LastCall variable of QueueMember event? I'm looking at: > 1201897536 for instance. Unix epoch format, or number of seconds since Jan 1, 1970 UTC, as I recall. -- Jared Smith Community Relations Manager Digium, Inc. __

Re: [asterisk-users] SIP Softphones and Citrix ?

There is an option you might consider (if you are starting from scratch). Don't use citrix. Write a web app. Then embed a softphone in that web app. Tim. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users maili

[asterisk-users] QueueMember event/LastCall Variable - Format?

Hi all, What format is the LastCall variable of QueueMember event? I'm looking at: 1201897536 for instance. Thanks ! -- Warm Regards, Lee "Everything I needed to learn in life, I learned selling encyclopedias door to door." ___ -- Bandwidth and

[asterisk-users] It's about time! -- Digium PCI-Express Cards

Just noticed this today: Digium TE220B Dual Span T1/E1 PCI Express Card with Octasic DSP-based Echo Cancellation Module It's about time Digium got on the ball and made PCI-e cards. What are people's experiences with this card? Anyone

Re: [asterisk-users] Enterprise or Fedora?

Matthew J. Roth wrote: > love U.all wrote: >> i wanna build a production Asterisk box ,will RedHat Linux Enterprise >> Server be more stable than Fedora core Linux or it makes no >> significant difference > I started out running Fedora, but I have migrated away from it for a few > reasons. F

Re: [asterisk-users] LDAP support

There a realtime LDAP driver now in 1.6beta2 On 23/01/2008, Cavalera Claudio Luigi <[EMAIL PROTECTED]> wrote: > Hello, > I've found this information about asterisk and LDAP: > http://www.voip-info.org/wiki/index.php?page=Asterisk+LDAP > which can be out of date. > > I'm trying this http://www.mezz

[asterisk-users] Bypassing a Auth on Invite or Forbiden?

Hello, I have 2 asterisk servers that are not working well together. One is acting like a registrar (PBX01) for all my PAP2's and other SIP/IAX devices. And the other is acting like my sip gateway (PBX02) to various providers. They are both on a private network and should be trusting e

Re: [asterisk-users] Enterprise or Fedora?

I cannot think of a single reason to use Fedora for a production anything when there are alternatives like CentOS. Fedora is bleeding edge stuff and constantly changing. -Original Message- From: Matthew J. Roth [mailto:[EMAIL PROTECTED] Sent: Friday, February 01, 2008 7:39 AM To: Asteri

[asterisk-users] Asterisk 1.4.17 and Teliax DTMF

I am having a problem with DTMF when sending calls through Teliax (SIP). In the peer for teliax I defined dtmfmode=rfc2833 and for the most part it is working. The problem always happens when a user is trying to call a conference system. They simply cannot get into the conference because

Re: [asterisk-users] Enterprise or Fedora?

On Fri, Feb 01, 2008 at 09:58:28AM -0600, Milton Calnek wrote: > > > Matthew J. Roth wrote: > > love U.all wrote: > >> i wanna build a production Asterisk box ,will RedHat Linux Enterprise > >> Server be more stable than Fedora core Linux or it makes no > >> significant difference > > I starte

Re: [asterisk-users] Enterprise or Fedora?

On Fri, 1 Feb 2008, Milton Calnek wrote: > > > Matthew J. Roth wrote: >> love U.all wrote: >>> i wanna build a production Asterisk box ,will RedHat Linux Enterprise >>> Server be more stable than Fedora core Linux or it makes no >>> significant difference >> I started out running Fedora, but I ha

Re: [asterisk-users] CallerID shows wrong values in manager interface

Trust me, I don't WANT you to look at my code, it's butt-ugly! lol, lust kidding... -- but at http://www.astsee.com/ you can download the source code to my AstSee project -- it may provide some insight into what needs to be (or CAN be) gleaned from asterisk. I struggled with all this a year

Re: [asterisk-users] Remote Call Center Agents and Asterisk?

Hello Steve, You are right on track and this is also what we have done with pretty good results. Of course now with Flex/Air there are a number of ways to enhance the service for the Customer/Agent Ed Mail: edpimentl[at]gmail.com Voip: edpimentl [SKype | GoogleTalk ] http://agileoss.com (We

Re: [asterisk-users] Astersik Transcoder support

http://www.digium.com/en/products/voice/tc400b.php Simon Elliston Ball [EMAIL PROTECTED] On 1 Feb 2008, at 17:29, Charles Feng wrote: > Hello All: > > Does the Asterisk support to insert an off the board transcoder > for a call? > > Thanks, > > Charles > > Never miss a thing. Make Y

[asterisk-users] Astersik Transcoder support

Hello All: Does the Asterisk support to insert an off the board transcoder for a call? Thanks, Charles Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools

Re: [asterisk-users] Remote Call Center Agents and Asterisk?

On Feb 1, 2008 10:33 AM, d4rk f1br <[EMAIL PROTECTED]> wrote: > Anyone using Asterisk in a Call Center environment? And more importantly is > anyone supporting home based remote call center agents with an Asterisk > backend? > > My experience with Asterisk is limited, however I have set it up and

Re: [asterisk-users] Remote Call Center Agents and Asterisk?

On Feb 1, 2008 10:33 AM, d4rk f1br <[EMAIL PROTECTED]> wrote: > Anyone using Asterisk in a Call Center environment? And more importantly is > anyone supporting home based remote call center agents with an Asterisk > backend? > > My experience with Asterisk is limited, however I have set it up and

Re: [asterisk-users] Enterprise or Fedora?

My suggestion - use the distro which you know best. We use Debian (200+ installations). It works stable for us because we know how to achieve it. Others use Fedora/Centos - because they are experts in these systems. Stability and performance of the system does not depend on the distro - only on

Re: [asterisk-users] Problem picking up a call with PickUpChan or PickUp (asterisk-users Digest, Vol 43, Issue 1)

That sounds like a CANCEL message being improperly routed somewhere along the line. Nothing you can do with the config to fix that. N. Paul Madley wrote: > Hi, > > >> Does anyone of you has a working configuration with SNOM phones that are >> able to pickup a call from a flasing LED? >>

Re: [asterisk-users] Unicall

Thank you Moisés, we are indeed going from 1.4.9 to 1.4.17, we will backup channels/chan_unicall.c and the Makefile entries of channels/Makefile and do our upgrade to .17. You are indeed correct on the Unicall, we have astunicall-1.4.9-0.1, I thought it was the same thing. Now I know better :)

Re: [asterisk-users] Enterprise or Fedora?

We are using Fedora because that is what the company we got our system from recommended. If I was doing the system myself I would throw in my vote for CentOS. I am using it for a database server and I have had no problems with it at all. It is about as stable and secure of Linux distro as I have ev

Re: [asterisk-users] pulling my hair out over voicemail

Ok, I have made some progress debugging this. I dont believe it has anything to do with asterisk or my phone. Rather I think it is an issues with STUN and/or my Linksys router at home. The phones I am testing all sit behind a NAT'd firewall, your basic Linksys router for the Home DSL user. Th

Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's

Hello, For such cases we usually suggest to put 2 boxes in your infrastructure: 1. Main billing gateway - where all PBX'es are connected (all client's remote PBX'es and your Local PBX) 2. Local PBX - where user's without PBX'es are connected Then user connects in following way: User -> Local

Re: [asterisk-users] Problem picking up a call with PickUpChan or PickUp (asterisk-users Digest, Vol 43, Issue 1)

Hi, > Does anyone of you has a working configuration with SNOM phones that are > able to pickup a call from a flasing LED? Unfortunately (as far as I'm aware) this is a bug in the 1.4.17 release, and therefore I don't think any config changes will fix it. We've been told to roll back to our pre

Re: [asterisk-users] Enterprise or Fedora?

Matthew J. Roth wrote: > love U.all wrote: >> i wanna build a production Asterisk box ,will RedHat Linux Enterprise >> Server be more stable than Fedora core Linux or it makes no >> significant difference > I started out running Fedora, but I have migrated away from it for a few > reasons. F

Re: [asterisk-users] Meetme voice quality problems

Administrator TOOTAI wrote: > This is not true if you're using B410P cards. We always face timing > problem as we can't -Asterisk stability issues- add X100P or TDM400P > with those cards Daniel, I thought that using an empty TDM400P as a timing source may no longer be the best solution due to

Re: [asterisk-users] Enterprise or Fedora?

love U.all wrote: > i wanna build a production Asterisk box ,will RedHat Linux Enterprise > Server be more stable than Fedora core Linux or it makes no > significant difference I started out running Fedora, but I have migrated away from it for a few reasons. Fedora has a very short life cycle

Re: [asterisk-users] G729 version to be downloaded for my machines

Download for Pentium4 Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad Sent: Wednesday, January 30, 2008 10:55 PM To: asterisk-users@lists.digium.com S

[asterisk-users] Codec preference selection, codec negotiation

Hi List; Is it possible to do configuration at the user context to let him use codec1 if destination support and if not, then use codec 2? For example, to let user1 use codec g729 if he needs to call user2 and user2 support g729, and if user2 does not support g729 then use g711 (alaw or ulaw), is

[asterisk-users] Remote Call Center Agents and Asterisk?

Anyone using Asterisk in a Call Center environment? And more importantly is anyone supporting home based remote call center agents with an Asterisk backend? My experience with Asterisk is limited, however I have set it up and installed it previously and had it working for home usage and for simpl

[asterisk-users] SIP Softphones and Citrix ?

Anyone aware of any SIP softphones that might virtualize well with Citrix presentation server? I suspect I know the answer already as I have been researching softphones that work with Cisco CallManager that can be virtualized if you will with Citrix and have come to learn that its not something th

Re: [asterisk-users] Asterisk-Addons install success-Could not find ooh323.conf

[EMAIL PROTECTED] wrote: > Am I doing something wrong? What I should do to get ooh323.conf cp asterisk-ooh323c/h323.conf.sample /etc/asterisk/ooh323.conf -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth an

Re: [asterisk-users] play promt at the same time to calling and callee

2008/2/1, Giedrius Augys <[EMAIL PROTECTED]>: > > Hello, > >I want that, when call is answered , callee and calling would hear > different prompts and after promts the calls would be bridged. I've tried > this situation: > exten => s,1,Set(LIMIT_CONNECT_FILE=hello-world) > exten => > s,2,Dial

Re: [asterisk-users] Unicall

Mark, you are confusing terms here. You DO NOT have Unicall 1.4.9-0.1, libunicall does not even use that convention for its versions. What you have is astunicall-1.4.9-01. AstUnicall is just a package with patches and proper versions of spandsp, libsupertone, libunicall, libmfcr2, zaptel and Aster

Re: [asterisk-users] AgentLogin by console

Thanks Tzafir, but this functionality needs sombody answer the call. I need to do this automatically. On Jan 22, 2008 4:10 PM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > On Tue, Jan 22, 2008 at 04:04:29PM -0200, equis software wrote: > > Hi! > > Is there any way to login an agent by console com

Re: [asterisk-users] BRI card with PCI-E interface

On Fri, 2008-02-01 at 15:01 +0100, Alberto Pastore wrote: > Olivier ha scritto: > > Hi, > > > > Does such card exist ? > > It seems all existing models are designed for PCI buses. > > > > Regards > > > > > > > > > > _

Re: [asterisk-users] realtime warning

yes. On Feb 1, 2008 12:07 PM, Russell Bryant <[EMAIL PROTECTED]> wrote: > > Rilawich Ango wrote: > > Hi, > > The server log shows the following message. > > > > [Jan 29 04:59:02] WARNING[1896] config.c: Realtime mapping for > > 'sippeers' found to engine 'mysql', but the engine is not available >

[asterisk-users] Enterprise or Fedora?

i wanna build a production Asterisk box ,will RedHat Linux Enterprise Server be more stable than Fedora core Linux or it makes no significant difference _ Express yourself instantly with MSN Messenger! Download today it's FREE! http

Re: [asterisk-users] BRI card with PCI-E interface

Olivier ha scritto: > Hi, > > Does such card exist ? > It seems all existing models are designed for PCI buses. > > Regards > > > > > ___ > -- Bandwidth and Colocation Provided

[asterisk-users] BRI card with PCI-E interface

Hi, Does such card exist ? It seems all existing models are designed for PCI buses. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.di

Re: [asterisk-users] Problem with DTMF dialing

On Thursday 31 January 2008 11:52:09 pm Ian wrote: > Sorry for taking so long to reply, > > This email got lost in translation, again. > > Ian > > Ian said the following on 30-Jan-08 03:57 PM > > > Thaks for the speedy reply > > > > Tzafrir Cohen said the following on 30-Jan-08 12:37 PM: > >> On We

[asterisk-users] Unicall

Hello All, we have Asterisk 1.4.9, Unicall 1.4.9-0.1, and Zaptel 1.4.5.1. If we were to update or recompile Asterisk, would we need to do anything with Unicall or Zaptel? Thanks in advance ___ -- Bandwidth and Colocation Provided by http://www.api-

[asterisk-users] play promt at the same time to calling and callee

Hello, I want that, when call is answered , callee and calling would hear different prompts and after promts the calls would be bridged. I've tried this situation: exten => s,1,Set(LIMIT_CONNECT_FILE=hello-world) exten => s,2,Dial(SIP/trunk-out/37052390920|60|rL(10)A(conf-entering

Re: [asterisk-users] Meetme voice quality problems

Matthew J. Roth a écrit : > [...] > > I settled on using an empty TDM400P as a timing source, because it is a > simple solution that "just works." This may still be your best bet, but > I'll defer judgment on that to the list because Asterisk has evolved > quite a bit since I made that decision

Re: [asterisk-users] meetme music on hold - when only conference member problem

I have, I have ztdummy module loaded in the kernel On Feb 1, 2008 11:59 AM, Gordon Henderson <[EMAIL PROTECTED]> wrote: > On Fri, 1 Feb 2008, Tomasz Zieleniewski wrote: > > > Hi, > > > > I have the following problem that when someone connects to my conference > and > > is the only member > > musi

Re: [asterisk-users] h priority problem

On 2/1/08, Paul Hales <[EMAIL PROTECTED]> wrote: > > I need to carry a variable over into the 'h' priority - so I can go back > and clean up DB entries in a mysql database (time of call and so on) > > I tried using UNIQUEID but it seems that 'h' generates a new one. > > Anyone have any ideas? What

Re: [asterisk-users] meetme music on hold - when only conference member problem

On Fri, 1 Feb 2008, Tomasz Zieleniewski wrote: > Hi, > > I have the following problem that when someone connects to my conference and > is the only member > music on hold is played just for one second or less and then stops: > > [Feb 1 10:38:46] -- Started music on hold, class 'default', on c

Re: [asterisk-users] h priority problem

You can use account code and userfield. You can set userfield to anything you want in the dialplan. On Feb 1, 2008 3:31 PM, Doug Lytle <[EMAIL PROTECTED]> wrote: > Paul Hales wrote: > > > > > > Anyone have any ideas? What can I use to carry a variable over into > > 'h'?? > > > > Lets see what you

Re: [asterisk-users] h priority problem

Paul Hales wrote: > > > Anyone have any ideas? What can I use to carry a variable over into > 'h'?? > Lets see what you have so far. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."

[asterisk-users] call log notice messages

I am getting quite a lot of these notices: Feb 1 10:22:01 NOTICE[2487] cdr.c: CDR on channel 'mISDN/3-2' not posted Feb 1 10:22:01 NOTICE[2487] cdr.c: CDR on channel 'mISDN/3-2' lacks end Feb 1 10:22:06 NOTICE[2471] cdr.c: CDR on channel 'mISDN/3-1' not posted Feb 1 10:22:06 NOTICE[2471] cdr.c

[asterisk-users] meetme music on hold - when only conference member problem

Hi, I have the following problem that when someone connects to my conference and is the only member music on hold is played just for one second or less and then stops: [Feb 1 10:38:46] -- Started music on hold, class 'default', on channel 'SIP/sip.touk.pl-0083dad0' [Feb 1 10:38:46] -- S

Re: [asterisk-users] CallerID shows wrong values in manager interface

Thanks all :) Appreciate it. On Feb 1, 2008 12:04 PM, Ex Vito <[EMAIL PROTECTED]> wrote: > I've struggled with this recently. In short: > > > - Observed behaviour is expected as of asterisk 1.2 and later, > as previously described by Mojo > > - If you want to get the caller id for the c