Re: [asterisk-users] Attended transfer and dialplan

On Fri, May 29, 2009 at 11:35 PM, Olivier wrote: > > > 2009/5/29 Danny Nicholas > >> I’m pretty sure that attended transfer is a “features” function, not a >> dialplan one. >> > > Yes, you're right but do you think there's such a big difference between > both that it shouldn't be easy or even p

[asterisk-users] Achoring MEdia

I have 2 hosts that Asterisk is in between of...and for both I have canreinvite=no - but asterisk still sends re-invite to get out of the media path.   Proxy 1 --> Asterisk--> Proxy 2   I want asterisk to anshor media..   In extenstions.conf I have an entry to send calls for say 5551000 to proxy

Re: [asterisk-users] Digium Fax Driver

Tilghman Lesher wrote: > On Sunday 07 June 2009 19:39:50 Lee Howard wrote: > >> Tilghman Lesher wrote: >> What's the use case for the Digium driver? Am I missing something by not using it? >>> While they accomplish the same goal, the commercial driver is based upon >

Re: [asterisk-users] Digium Fax Driver

On Sunday 07 June 2009 19:39:50 Lee Howard wrote: > Tilghman Lesher wrote: > >> What's the use case for the Digium > >> driver? Am I missing something by not using it? > > > > While they accomplish the same goal, the commercial driver is based upon > > a different codebase, > > Ok. > > > provides s

Re: [asterisk-users] How run AsyncAGI commands in background

On Sun, Jun 7, 2009 at 4:37 PM, Jose Arias wrote: > Hi Moy, > > I'll do it so, but for your answer, it seems you are thinking about it as it > could be a bug. I don't think so. I mean: the redirect action on a channel > in AsyncAGI stops the current agi execution. It's the normal behavior. It's > t

Re: [asterisk-users] CDR question

Jim-- On Thu, Jun 4, 2009 at 1:40 AM, Jim Boykin wrote: > Hi, > > Asterisk does not post CDR when dial status is CHANUNAVAIL. CDR's are, at the current time, and always have been attached to the channel struct; so, if you don't create a channel, then there is nowhere to attach a CDR, and no w

[asterisk-users] remote queue members

Hi all, I'd like to know the best way to deal with queue member that are reached trough a SIP trunk. Let me explain: 1) "Master" asterisk box with my call queue 2) "Slave" asterisk box with a channel bank interface the two boxes are connected trough a SIP trunk, and the dialplan in the 1st box

Re: [asterisk-users] Digium Fax Driver

Tilghman Lesher wrote: >> What's the use case for the Digium >> driver? Am I missing something by not using it? >> > > While they accomplish the same goal, the commercial driver is based upon > a different codebase, Ok. > provides support for patented fax protocols, Really? V.34-fax (33,60

Re: [asterisk-users] Digium Fax Driver

On Sunday 07 June 2009 16:28:53 sean darcy wrote: > Steve Underwood wrote: > > Elliot Murdock wrote: > >> Hello! > >> I have a 64 bit Asterisk system and am wondering how to use Digium's > >> 32 bit fax driver. Is there some kind of emulation that can be used? > >> Thanks! > >> Elliot > > > > Use

Re: [asterisk-users] How run AsyncAGI commands in background

Hi Moy, I'll do it so, but for your answer, it seems you are thinking about it as it could be a bug. I don't think so. I mean: the redirect action on a channel in AsyncAGI stops the current agi execution. It's the normal behavior. It's the way to stop a playfile on a channel if it was previou

Re: [asterisk-users] Digium Fax Driver

Steve Underwood wrote: > Elliot Murdock wrote: >> Hello! >> I have a 64 bit Asterisk system and am wondering how to use Digium's >> 32 bit fax driver. Is there some kind of emulation that can be used? >> Thanks! >> Elliot > Use the FAX support built into Asterisk 1.6 and you won't have that > li

Re: [asterisk-users] Call recording in - out

Hello I did as you told me, but the problem remains. Im using Asterisk 1.2.x and this is my config: queues.conf - [general] persistentmembers = no [queue_1] persistentmembers = no monitor-format=wav monitor-join=yes monitor-type=MixMonitor wrapuptime=3 tim

Re: [asterisk-users] Ekiga, Twinkle and from where to start with open source

César Sequeira schrieb: > I try to connect Qutecom in my Asterisk Server but without success. > > What field I need to complete? > > Username; > Password; > Realm (asterisk IP Address); Default: "asterisk" > Server (asterisk IP Address); > Proxy (asterisk IP address); > > It's correct?

Re: [asterisk-users] Asterisk Clustering

Thanks Noah for your helpful reply. My setup will be 2 Asterisk (Trixbox) servers, Active/Passive, 2 PRIs through 2 Vega 400 gateways. i tried to follow some threats, like as the below: http://www.trixbox.org/forums/trixbox-forums/open-discussion/ha-cluster But i think something wrong in

Re: [asterisk-users] Teliax: where's the space in CALLERID(num) from?

Philipp Kempgen wrote: > sean darcy schrieb: >> I'm having trouble setting callerid with teliax. I use a simple dial-out >> subroutine to set the callerid depending on the calling extension, and >> then dial out. Teliax is saying they're not seeing any callerid info. > >> exten => s,n,Set(CALLER

Re: [asterisk-users] Call recording in - out

Hi, I had similar issue which happened when record option was mentioned in both agents.conf and queues.conf. When I commented the recordagentcalls option in agents.conf, it started to work. Mention the monitor option only in the queues.conf file. Do try. Regards, Kurian Thayil. On Sun, 2009-06-

[asterisk-users] Call recording in - out

Hello to all I'm trying to record the calls going to my queues, but asterisk creates 2 files, one with the inbound and another with the outbound sound. I know Sox should mix the 2 files automatically in the end, but this isn't happening. I have sox installed in my server. How can I force Sox to

[asterisk-users] BUSYDETECT_* flags

Hello all, I have asterisk-1.4.25, and having some problems with analog lines becouse my Telco does not have "disconnect supervision". So, i realize there are some switches on main/dsp.c: BUSYDETECT_MARTIN, BUSYDETECT_TONEONLY, and BUSYDETECT_COMPARE_TONE_AND_SILENCE. As far as i could read

[asterisk-users] ANI

When Asterisk sends a call to "a phone company" via a PRI/Dahdi, does it actually send CALLERID(ANI), or only CALLERID(NUM)? Cary Fitch ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIB

Re: [asterisk-users] How run AsyncAGI commands in background

Never mind, it was my mistake. I had some problems with my email client. Regards Jose 2009/6/7 Philipp Kempgen > Moises Silva schrieb: > > On Sat, Jun 6, 2009 at 7:18 PM, Philipp > > Kempgen wrote: > >> Jose Arias schrieb: > >>> Hi, > >>> Asterisk 1.4.18 > >>> AsyncAGI patch from //http://moythr

Re: [asterisk-users] Called party name with Cisco-2,811 gateway

On Sun, Jun 7, 2009 at 4:20 AM, Yehavi Bourvine wrote: > Hello, > >   I am using a Cisco 2,811 gateway to connect Asterisk over PRI to our > Nortel TX-1 PBX. Up to now I had only the calling party names passed both > ways. After upgrading the Cisco to the latest release (12.4.24T) it began > honori

Re: [asterisk-users] chan_dahdi missing in * 1.6.1.1

On Sun, Jun 7, 2009 at 6:19 AM, Steve Repo wrote: > I have a Sangoma A200 analog card with 2 FXO ports. It's working well > with asterisk 1.4.22 and Zaptel. I decided to upgrade to asterisk > 1.6/dahdi. > > I compiled and installed, > > dahdi-linux-2.1.0.4 > dahdi-tools-2.1.0.2 > libpri-1.4.10 > wa

Re: [asterisk-users] Meetme timeout

On Sun, Jun 7, 2009 at 5:19 AM, Kurian Thayil wrote: > extension is hit by channel. I wasn't aware that we can scan a channel > continuously using an AGI. If so, how could we do that? A somewhat ugly solution that gets the job done: 1) crontab, set up a periodic job that does something like 2) ast

[asterisk-users] Callback with a2billing

Hello, Can anyone give me a sample configuration of Callback feature on a2billing. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.dig

Re: [asterisk-users] How run AsyncAGI commands in background

Moises Silva schrieb: > On Sat, Jun 6, 2009 at 7:18 PM, Philipp > Kempgen wrote: >> Jose Arias schrieb: >>> Hi, >>> Asterisk 1.4.18 >>> AsyncAGI patch from //http://moythreads.com/testasync2.diff >>> // >>> Regards >> >> So what? >> > What do you mean with "so

Re: [asterisk-users] Meetme timeout

Hello Danni, As you said, I went through the post and found that is applicable everytime no matter how many members are there in a conference. And I understand that I cannot completely rely on that. I need to do some logical tweaks with some other application like an AGI to crack this issue. I was

[asterisk-users] chan_dahdi missing in * 1.6.1.1

Hello, I have a Sangoma A200 analog card with 2 FXO ports. It's working well with asterisk 1.4.22 and Zaptel. I decided to upgrade to asterisk 1.6/dahdi. I compiled and installed, dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.10 wanpipe-3.4.1 asterisk 1.6.1.1 My analog card is recognized i

[asterisk-users] Called party name with Cisco-2,811 gateway

Hello, I am using a Cisco 2,811 gateway to connect Asterisk over PRI to our Nortel TX-1 PBX. Up to now I had only the calling party names passed both ways. After upgrading the Cisco to the latest release (12.4.24T) it began honoring the "remote-part-ID" field sent by Asterisk and sends the *call

Re: [asterisk-users] Sangoma A200 wt HW echo on 64bit Centos , 4 fxo and crosstalk!

That is correct. That is the first test we did. On 07/06/2009, Moises Silva wrote: > On Sat, Jun 6, 2009 at 3:18 PM, Gavin Henry wrote: >> Every call as soon as the sangoma card is live. >> >> Speak to Konrad on your techdesk for more info. >> >> Thanks. >> > > I'll speak with him on Monday. > >