similar to: How to get 1.2.7 asterisk

Hi all, I am handed a project to setup *. The requirement is that it can handle 8 T1s. Half of the calls coming into the system will be routed to SIP extensions (with transcoding). The machine we have in our disposal is a new dual Xeon 3.2gHz server with 2g of ram and an dual 1000mb nic. Voice will be coming in from the PSTN (through 2 quad digium cards) in g711ulaw, and most of the time will

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

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, Is there a way to connect two live calls through the manager api without directing them to a meeting room? Currently, I can connect them by sending them to a meeting room. However, I don't know what the overhead is, and I kind of think that if I can connect them or link them up, the overhead would be minimum. -------------- next part -------------- An HTML attachment was scrubbed...

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

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 all, I am setting up a script to record all the call. There are two app for recording. "Monitor" and "Mixmonitor", one mixing the audio on the fly and one mixing it at the end but also allow a option not to mixing the audio at all. If mixing the audio on the fly is not that taxing on the CPU, I would like to use 'mixmonitor' app. My question is, what is penalty on

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

My TE411p does not seem to like to have two PRIs from different telcos (span 1 and span 2). I can get one working, but not the other. However, if I use vpmsupport=0 when loading the wct4xxp module, they both work. But here is the problem, vpmsupport=0 disables the on board echo cancellation. Any ideas? BTW, here is zaptel.conf span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs bchan=1-23 dchan=24

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

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

Hi, Anyone know how to compile asterisk for a hyperthreaded processor? Thnx

It sounds like Naren and company has their own CRM application. They need a predictive dialer that allows third party app integration. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Matt Florell Sent: Monday, March 13, 2006 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Hi all, I've 2 * tryning to connect each other Server A is already registred on server B But server B never registers in server A I always get this: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGREJ Timestamp: 00018ms SCall: 00004 DCall: 00003 [XXX.XXX.XXX.XX:4569] CAUSE : Registration Refused CAUSE CODE : 29 Any tip? Best regards, Marco Mouta

I'm trying to set the Pickup feature. I'm setting my extensions.conf as: exten => _*.,1,Pickup(SIP/${EXTEN:1}) but if, for example, extension 03 is ringing by a call made from extension 01, and I try to pick it up from extension 02 (by dialing *03 from extension 02), I can see in the Asterisk console (Verbosity set to 10): -- Executing Dial("SIP/01-512c",

Can someone explain me this message: "chan_iax2.c: Ooh, voice format changed to ..." Where can I find a list of numeric codes used to identify voice format? Then, sometime I get an infinite loop of messages like these: DEBUG[15015] chan_iax2.c: Ooh, voice format changed to 1 WARNING[15015] channel.c: Unable to find a codec translation path from g723 to alaw DEBUG[15015]

Wanted any /all used out of service Digium boards Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060410/3892463b/attachment.htm

Hi everybody, A customer requires G726-40 with Asterisk... I know G726-32 is pseudo-standard, but he definitely wants G726-40... Is there any (easy) way, to integrate G726-40 into Asterisk? Has anyone done this before? Any hints? Please help! Due to a misunderstanding, my product manager already offered this to the customer and now i do not know how to do it... Thanks a lot in advance,

Hi everyone, On the Polycom 601 phones we are using, the forward feature works very nicely for agents that are out on trips. I was wondering if there is a way to test to see if they have the forward option enabled. When it is enabled the call comes in and gets -- Got SIP response 302 "Moved Temporarily" response and then it uses the correct outbound macro to forward the call to the

Somebody knock where i change name?s User-Agent Thank?s -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 2703 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060424/3684c52f/attachment.bin