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Anyone tried the digium TC400B transcoding card? What are your opinions? Thnx

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NVWhisper. Justin ------------------------------ Date: Thu, 08 Mar 2007 16:25:28 -0500 From: Wai Wu <wkwu@calltrol.com> Subject: [asterisk-users] Coaching in asterisk Is there a way to setup a conference where party A can coach another Party B, at the same time, all other parties cannot hear party A? In order words, partis A and B can hear every one, and party A can only be heard by party B. Th...

Hi all, Just ran into some issue with the originate AMI command. It seems that there is a limit of around 120 calls I can place with the originate command simutanously. By that I mean sending Asterisk a lot of originate command very fast. Anyone know if there is a limitation? Thnx. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi everyone, I am running into wall today with simultaneous call limits. I have two Asterisk machines (fast 3GHz C2D with 2GB of ram). I tried to create a lot of sip calls from one machine to the other by issuing AMI Originate commands to one machine. The machine that makes calls plays a message (demo-intruct) upon the other machine answer. The machine receives the calls just waits for 40 seconds

Hi all, I need the zap channels going, but got the following error. What do I need to change in my configuration? Thnx. chan_zap.c: In function `zap_send_keypad_facility_exec': chan_zap.c:2309: warning: implicit declaration of function `pri_keypad_facility' chan_zap.c: In function `pri_dchannel': chan_zap.c:9292: structure has no member named `call' make[1]: *** [chan_zap.o]

Hi all, Any one know how to increase the Linux limit? I am hiting a wall on 200 calls playing files at the same time. From Asterisk console, I am getting messages like Sip_request_call: Unable to build sip pvt data for "asterisk1/700" Too many open files Is this a limit of my Linux box? I only have 512MB of ram. Will increase it to 2G help or I have to change some configuration in

Hello list I am having an issue with Chanspy/SIP that I?m hoping someone has come across and resolved in the past. I am sending calls that come in TDM through T1 ZAP channels and go out to a SIP trunk. If I spy on the SIP channel, I can hear the person on the SIP side of the call just fine, but the person on the ZAP channel fades in and out. If I spy on the ZAP channel, and can hear

Anyone know the gsm encoding mip requirement from g711? Or number of channels can be transcoded from g711 to gsm at a time. Thnx

I am use Fedora 3, and run into a 1.4 compile issue. When 'make install' I got this message. [root@asterix asterisk-1.4.1]# make install make: expand.c:489: allocated_variable_append: Assertion `current_variable_set_list->next != 0' failed. make: *** [utils] Aborted [root@asterix asterisk-1.4.1]#

I didn't know you are courageous. I upgraded to 1.4 last night. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Stephen Bosch Sent: Friday, March 09, 2007 11:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RE: Coaching in asterisk Wai Wu wrote: > Ouch, I

BTW. We only use Asterisk for a few functions. Everything else is done on an extenal application controlling Asterisk through AMI. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Wai Wu Sent: Friday, March 09, 2007 12:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE:

Hi all, Anyone know how to dynamically create meetme conference rooms on the flight? I remembered a while ago there was a switch that tell meetme to create the conference room is the room is not defined in the meetme.conf. It doen't seem to be working for me anymore. Thnx

Have you tried to load the driver with ec disable? Last time (long time ago) when I was working on a quad card, we weren't able to get ec to work with hardware ec on. ________________________________ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Brian Alexander Sent: Thursday, September 27, 2007 10:59 AM To: Asterisk Users

Hi list, My long duration calls are being timeout by my SIP VoIP provider for failure of receiving re-INVITE within their timeout limit. Is there a way to config Asterisk to automatically send a re-INVITE message every 10 to 15 minutes? I looked into the sip.conf file and couldn't find such a parameter. Thnx.

I have a X100P and a TE110P in my Asterisk box. I can either get the X100P or the TE110P to work, but never both. Here's my zaptel.conf span=1,0,0,d4,ami e&m=1-24 fxsls=25 When I load wcte11xp and wcfxo, I will get this error. [root at asterix etc]# modprobe wcte11xp ZT_CHANCONFIG failed on channel 25: No such device or address (6) /lib/modules/2.4.20-8/misc/wcte11xp.o: post-install

Can someone share pointers to Asterisk's Dialogic support? Which boards are supported, driver status, and etc. Thnx