similar to: Incoming callers aren't hearing ring

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 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, blind transfer and answering machine(during the pressing of 999). During attended transfer and blind transfer , sometimes we cannot hear the sound of 'pbx-transfer'. The same problem occurs during answering machine. I cannot understand why this is happening? log is :

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

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

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 haven't used Asterisk in many years, but in searching for a good podcasting solution that will allow me to record three or four participants to individual tracks (which would allow me to go in and do noise removal on each participant individually, giving a higher quality), I came up with the idea to use Asterisk. Now I've installed it and got it all set up and did a test

Hello, As a podcaster I use Asterisk extensively and often have several people in a conference room. We'll record the calls via a SIP phone connected to a sound mixer. Is there an easy way to bump up the audio bitrate for all callers connected to the Asterisk server and improve the general sound quality? The server is not used much outside of recording the podcast. We're not opposed to

Does anyone know of an efficient (or at least polynomial in the worst case) way to map a local DS node to all complete and/or global nodes it may correspond to in any of the function's callers? Thanks, --Patrick -- If I'm not here, I've gone out to find myself. If I get back before I return, please keep me here.

Most of the time, when someone calls in from the outside on a POTS line, and possibly over IAX as well, they don't hear any ringing sound while the internal SIP phones ring. If you call from an inside SIP phone, even forcing it into the incoming context, you hear the ringing. The outside calls can answer and talk fine; just no ring indication. Is there a setting that controls this?

is there something that needs to be set up to make the 'called' and 'callers' buttons work on this phone? all i get is the backlight to switch on and off. Jason Kawakami

Hi Guys, Does anyone have knowledge about max number of conference rooms, and max number of conference callers in one room? Thank you so much. jintwo __________________________________ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web http://birthday.yahoo.com/netrospective/

Hi, I'm wondering firstly if somebody already did this and what the best way might be to go about it. Basically I want to have a context in * which could just be a regular extension that you can transfer an incoming caller to. This extension could then do several things, the one bit I need advise on is sending a sms to the callers cell/mobile phone with driving directions. 1) Firstly it

Hello all, We have a periodic issue with app_queue where we run into problems that generally require a restart of asterisk to address. We were looking into doing an "unload" and "load" of app_queue as a possible solution that would keep our already connected calls up, but the "unload" always seems to crash asterisk if we have callers in our queues. I was looking

I'm running chan_capi on a number of systems in France, France Telecom offer the possibility of having the caller's name, but say we must configure for EuroISDN+. Google doesn't show much and the best I could see was in Dutch. Any Europeans solved this one? Rgds -- Dave Cotton <dcotton@linuxautrement.com>

Here's how it's currently working: 1) Call comes in 2) Operator parks call (700) 3) Operator picks up call on another phone (701) 4) Operator tries to transfer to a different phone (we use #0) but the transfer doesn't work. We can transfer initial callers all we want and it works fine. Once a call is parked, however, we can no longer transfer the caller. Any ideas? Thanks, Jay