similar to: AGI - Choppy Sound

Hello, I'm trying to understand the AMI and use the PHPAGI library to extend and to easy my work ... I'm still trying to figure out how things works, but I haven't found enough resources to make me sure about certain things ... So, I'd like to know if somebody can give me a hand .... ;) My calls are answered almost instantly by an IVR and puts the call on a queue to wait for

Hi my friends ... I need to implement my ivr based on some decisions that would be stored in a database ... like ... to play a sound file if a boolean field in a PostgreSQL database is true ... something like that ... But how do I do that ? Is it through AGI ? If so, can you at least point me through the right direction, because I don't know much about AGI ... Thanks again folks ... Marcelo

Hello,I was using the ResetCDR() right after my IVR answered the calls to set them unanswered until or if somebody pick up the callI just upgraded to version 1.4.19.1 and now ... when the ResetCDR() function is called it seems to just erase the recordIs there another to set the status of the call unanswered ? is it a bug ?Thanks, -------------- next part -------------- An HTML attachment was

I have installed asterisk on numerous servers. Every install was done on Fedora and (White box Linux). I now have zap channels in one of the boxes (T-1). No matter what type of channel I call on (sip or zap) I get some really strange artifacts in the sound, almost like a skip in the playback. It seems to always be in about the same place in the recording. Usually in the beginning of playback. For

Hey List, I'm trying to work out a resolution for domain based SIP authentication. We are working on a virtual PBX type product and want to allow username overlaps on separate domains so that joe.blow at domain1.com and joe.blow at domain2.com are entirely separate of each other. I've found this possible solution (have yet to try to accomplish this):

I am experiencing choppy sound when I bridge from a SIP peer to an IAX peer. I am running Asterisk 1.4.13 on a 2.6.22.9 kernel (Fedora). I am experiencing choppy sound from the SIP peer to the IAX peer but not vice-versa. I know that this is not a bandwidth issue because I don't have choppy sound (with the same codec) when bridging IAX->IAX peers or SIP->SIP peers. My timing source is

Okay everyone - I'm moving away from using sipura 841 phones. I'm starting to test with Polycom IP 501 phones. We plan to upgrade our server to a dual t1 but, for now we have a digium tdm400P with 4 analog lines coming into it. So my question is will upgrading the IP phones with my existing digium tdm400 card be enough to satisfy my users ? or is it really a combo deal needing to

Hi, I am using phpagi for agi scripting. and want to update callerid number but didn't get any success. please help me how to update PHPAGI is new for me. Below is the code which I write. #!/usr/bin/php -q <?php set_time_limit(30); //require(.phpagi.php.); include("phpagi.php"); $agi = new AGI(); //answer the call $agi-> answer();

Hi Here is part of my extensions.conf exten => 8661231234,1,agi,dtmf.php When I dial this number, this is what I see in my asterisk console: -- Accepting AUTHENTICATED call from 198.22.67.70: > requested format = gsm, > requested prefs = (), > actual format = gsm, > host prefs = (gsm|ilbc|speex), > priority = mine -- Executing

Hi there, We're having some complaints of choppy audio from our SIP customers. Asterisk is showing no errors, but I'm getting a lot of these in my syslog: Feb 17 13:34:31 ntop[2863]: **WARNING** packet truncated (14654->8232) The first number varies, but the last number is always 8232. I've read that this is a common MTU size, but none of our interfaces have an MTU of 8232.

Hi all, I have my asterisk setup as following: IP 2 x E1 x-lite <-------> Asterisk -------> PSTN When I place a call from x-lite to PSTN, the quality of the sound in the direction x-lite -> PSTN is very bad. That is, the voice of the x-lite user, heard by the PSTN user is choppy and makes communication not very pleasant. The sound is choppy as if bits of data

Hi there I europe alaw is usual. I have a SIP Phone which perferes ulaw. When my * box has to transcode alaw to ulaw the sound get's one way choppy. (alaw => ulaw is choppy, ulaw => alaw is fine). I managed to fix the issue by forcing my SIP phone to use alaw only, but is this a know issue with asterisk 1.2.13? -Benoit-

Hi Guys I'm conecting 2 astersk servers using this arquitecture (Ext softphone)<==sip==>(asterisk 1)<====iax2 trunk====>(asterisk 2) <===alaw==>(pstn) If i call from the Ext to the asterisk 2 the sound is perfect, but if i call from Ext to the pstn, i can hear perfect but they tell me that sound really choppy, i tried using several codecs (same problem) but i

I just signed up with livevoip for 800 DID and have very choppy audio. From PSTN to my asterisk is ok but asterisk to PSTN is terrible. I am using IAX and was assigned to server iax01.nyc.*. I do not believe it is a bandwidth problem on my end and I have no problems using iax with gafachi. I opened a ticket with livevoip but no response yet. Would I be better off using sip with them? Is there

I have an * setup with sip phones and sip fxo gateway. When a sip phone places a sip/fxo call on hold, MOH is very choppy. It looks like RTP has a real problem with timing if it is not receiving RTP packets. If the outside call that is placed on hold is not generating any audio, the sip/fxo gateway does not send * RTP packets. Is this valid? Is this a problem with the sip/fxo gateway or a problem

Hi I am currently developing a IVR application using PHP/AGI. I am using the PHPAGI class at http://phpagi.sourceforge.net/ to handle the commuication with my *. The application basically asks a caller to enter in some information which is then processed and a answer is read back out to them. I want the application to loop back to the beginning after giving the answer so they can try another

I have had no success in getting the voicemail working on Asterisk 1.2.11 on CENTOS4(2.6 kernel) guest on vmware server 1.0.1. I tried with or without ztdummy device, renice -20 on asterisk process and even real-time priority on the host Windows XP box for the vmware process. I am running on an AMD Athlon 64 X2 4600+. The behaviour is when the voicemail answer, the voice sound ok but when

Hi! I have Asterisk 1.6.1 installed on OpenSuSE 11.0 running with choppy sound. One specialist on the forums asked me if I have DAHDI configured, he assumed that this could be cause of choppy sound problem. > dahdi_test Unable to open dahdi interface: No such file or directory Do I need to configure DAHDI even if I do not have any Zaptel devices? Is there any guide for configuring

I am trying to use swift in PHP/AGI. function swift($text, $escape_digits='', $frequency=8000, $voice=NULL, $fnameIn='') During swift speaking some text I want the caller to be able to press 1, 2 or 3 to do thing 1, thing 2 or thing 3. How are these digit defines and then caught? Thanks, Michael

Hello All, We recently changed our asterisk system to begin using G.729a as the primary codec. We have a Cisco 1700-series router which connects to the PSTN via FXO ports, along with Cisco 7940 SIP phones. Everything is working great, except... When an inbound caller calls into our system, they hear an IVR. When the caller dials an ext (SIP phone), the ringing progress tone is