similar to: Problem of hearing transfer' s sound

Dear all, We have been using asterisk for 4 years. Now we have got problems which occurs during the attended transfer. During attended transfer, sometimes we cannot hear the sound of 'pbx-transfer'. Sometimes we can hear little portion of 'pbx-transfer's sound. That means sound also become noisy. I cannot understand why this is happening? log is : -- Started music on hold,

Dear all, We have been using asterisk for 4 years. Now we have got problems which occurs during the attended transfer. During attended transfer, sometimes we cannot hear the sound of 'pbx-transfer'. I cannot understand why this is happening? log is : -- Started music on hold, class 'default', on SIP/113.34.235.13-b7a3f110 -- <SIP/0000185148-092db338> Playing

Dear sir, We have been using asterisk for 4 years. Now we have got problems which occurs during the attended transfer. But we are not always getting this problem. Sometimes it happens. But now we cannot understand why this is happening? problem is:"Failed to play transfer sound! " The log of asterisk is as like as followings: [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Invalid SIP

Dear all, Currently I am using asterisk 1.4.23.1. . Over the period of 1 week, the memory in use starts off at 50% and continues to climb until it hits 99%. When memory usage ratio become 50% or more, the quality of calls become extremely noisy. The call quality goes back to being perfect once I reboot the machine, but I was to try and avoid having to reboot the machine every week. the following

i don't khow really much about php i just used the source file from casterclub.com for the shoutcast server and improvise/adjust it from there for the icecast server. apparrently i don't know how to authenticate correctly as this is what is returned as well $client_connections = HTTP/1.0 401 Authentication Required WWW-Authenticate: Basic realm="Icecast2 Server" You need to

Hi all, first of all thanks for your comments and ideas. I wrote because i wanted to know if i'm wrong or not and to let others khow how some companies operate. I work on technology, i work in the world we move, and i usually are in charged of handle situatios like that, and what i can tell all of you is that, if the system is faulty upon recepion, the only one common practice is open an RMA

hello Phi ., what operating system are you developing under ? if you are using linux, i would be happy to send you my simple scripts ( as you requested ). but they rely on lynx and grep . i don't know if windows has equivalents. a:/, On Mon, 22 Dec 2003, Phi Tran wrote: > i don't khow really much about php > i just used the source file from casterclub.com for the shoutcast server

Hi all I don't know if my message are correct in this forums. I create a program in c++ who use statistical function. I want to execute this function in R (in particular for use packages ade4, lattice, bioconductor...) Until now, my program work for simple function ("plot", "rnorm"...) but I can't use library My class are : // in constructor int argc = 1;

Greetings, Sounds like a simple thing to do, but I was not able to do it on these particular phones. Snom wiki was not helpful. My client wants to keep his phones pointed to UTC time, no DST, no change in timezone, i.e. to stay at 0 hours difference. The phones are provisioned from a tftp server. If I remove 'dst' value from the provisioning file, on bootup phones force users to pickup

Greetings all- Long story short - I find myself suddenly running a Asterisk PBX after old PBX suddenly died. Fortunately, I had been "playing" with Asterisk (via Trixbox) on a server in consideration of replacing our aged Merlin Legend - so over the course of last weekend, I brought my testbed PBX up to full operation and now supports about 30 users. All in all, went smoother than

Hi to all! I have this config, PSTN <--> AS5300 <--> ASTERISK I am using the Cisco as5300 to receive incoming calls and routing them to Asterisk for IVR. When I ran asterisk this is what I get when calling the voicemail demo. *CLI> -- Executing Playback("SIP/-081058b8", "transfer|skip") in new stack -- Executing Macro("SIP/-081058b8",

Me again, another problem. As I said before, I have 2 lines going into "incoming" context. When client calls, I press Flash, client hears music on hold (only on voip line as said in previous post), when I get back and press Flash again to get back to my client I cannon hear him, but he hears me without problems. I have just tested in on the LAN, same situations, happens everytime.

Hi Asterisk Gurus! My lab asterisk box has 1 FXO and 1 FXS ports in it. I connect a GSM phone to the FXO port. I connect a regular corded phone to the FXS port. The Dial() application for both incoming and outgoing calls specifies the A(hello-world) flag. From another GSM phone, if I call the extension (corded) phone attached to the box, it plays the hello-world file when I pick it up. But

Dear Folks, I'm not able hearing the voice of client but on other hand client able to hearing my voice.I'm not able to find out the problem where is i'm wrong. I'm getting continues following error: chan_sip.c:10391 check_via: '' is not a valid host Configuration DAHDI Tools Version - 2.9.0.1 DAHDI Version: 2.9.0 Regards akihlesh -------------- next part

Hello all. I am able to programmatically decode speex just fine (playing others' encodings), but my encoding eats the beginning of words. If I encode a word that gradually increases in volume, like "wonderful", I hear "nderful", but if I encode something percussive like "beep" I hear almost all of it. It's as if the modeller does not detect the start of a

Hi Does anyone know if there is a method to calculate a goodness-of-fit statistic for quantile regressions with package quantreg? Tanks [[alternative HTML version deleted]]

Hello, We have a DID partner sending traffic to Asterisk via SIP, but we are not hearing ringtones. When we call the same extension via SIP, we can hear that's it"s ringing (virtually).. Is is something related with call-progress not recognized by DID provider ? Thanks, ________________________ a b d o u l aba at gcomnetworks.com SIP: (131) 229-1002 at sip.freeipcall.com

We just switched from three x100p's to a te410p for handling our incoming/outbound calls. Everything works great, except incoming callers don't hear a ring while they are waiting for one of us to pick up. The phones themselves ring fine, but the caller doesn't hear anything until someone picks up, or it transfers to voicemail. Any clues as to what may be happening?

Hi list, I enabled the transfer function in my dialplan, but I may configure it wrong because sometime when I call a SIP extension number from one FXS port, the SIP side will hear word "transfer", I hear nothing, after that, the call conversation is fine.I'v had this problem for a long time, could not get clue where I configure it wrong. here is my related config part: sip.conf:

hi everyone i'm trying to get the variables from status.xml using php (based on the Shoutcast status file from casterclub.com) here is my file <?php $ip = "xxx.yyy.zzz"; $port = "xxxx"; $password = "xxxx"; <p>////////////////start the parsin action\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\\ //opening socket $fp =