similar to: one way choppy sound problem !

Hello list, I've been experiencing choppy sound as well. The version on Asterisk I was using originally was dated 10/24/03 (I think), the problem appeared after I updated from that version. My setup is a little different though. I'm having choppy sound only on some incoming calls -- from PSTN->PBX (between spans on a TE410) and PSTN->SIP. We use Cisco 7940 handsets and we also

Hi, I can not send voicemails as an attachement. When setting the "attach=yes" option in voicemail.conf the mails get rejected from the mail server: ----- Transcript of session follows ----- 451 4.4.1 timeout writing message to higgs.elka.pw.edu.pl.: Connection timed out with higgs.elka.pw.edu.pl. 451 4.4.1 timeout writing message to elektron.elka.pw.edu.pl.: Connection timed out

Hi all, I was wondering whether any of you have experience/info on Cable and/or ADSL modems that would come together with a SIP phone adaptor. What I am interested in is something that would plug directly into you ISP's cable (be it ethernet or adsl/phoneline), would combine a modem/router/nat such that on the other you could simply plug in your RJ-45 cable for your PC and a RJ-11 cable for

Hi, Since I've upgraded my * to 0.7.1 I see no new cdr's in my MySQL. Does this new version of * only work through ODBC ? Do I have connect to MySQL through ODBC now ? Regards, Dave

Anybody out there use Ser along with *? Any advantages disadvantages? Is this even a good idea?

Hello, I can not seem to be able to get StripMSD and Prefix to work for me in extensions.conf. This is an example of what I have: exten => _050.,1,StripMSD,1 exten => _50.,Prefix,01051 exten => _001051.,1,Dial(${TRUNK2}/${EXTEN}) exten => _001051.,2,Busy exten => _001051.,102,Busy What I want to achieve is to call 001051501657887 on TRUNK2 when dialing 0501657887. dialing

Can anyone _recommend_ a downloadable OSS softphone that _works_ under Linux and is compatible with Asterisk. So far I have tried kphone and linphone and had problems with both, and I am still waiting to hear back from the X-Lite beta folks. -- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just

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

I am experiencing a situation where I am getting intermittent choppy audio. Here is the network layout: Termination provider -> IAX2 over the Internet -> 20Mb fiber connection -> router -> Asterisk My ATA connection goes into the router between the fiber and the Asterisk server on another interface here is the layout from me to Asterisk: Sipura ATA (SPA1001 running

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

Hi ! I am trying to configure two E100P cards, but I am a bit confused with zapta.conf in what I am trying to achieve. The * will be connected to a pstn switch with two E1 PRI lines. The E1 lines will be used for incoming calls as well as outgoing calls. My problem now is what to put in zapta.conf, I would like to group all channels from both cards together (if that's possible). Does this

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

Hi, I have * working with X-Lite and Sipura adapters, but I have one person who is linux based, and is trying to use Linphone and Kphone. His end works, but I get very bad echo on my end. Have any of you folks been able to get linux based soft phones working well with *? I'd appreciate links to howtos/docs if you have them, and/or samples of working configs for * and the linux

I've beaten myself bloody dealing with this one... No luck so far. In summary, incoming calls from Gizmo establish, but neither get nor send sound. Outbound calls to Gizmo work fine (well a bit choppy but work) My thought is that the SIP connection is being made fine, but the RTP is getting stopped / blocked / misdone somewhere. Here is the thing: Asterisk 2.5 on Linux (No hardware

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

Hi After a day of running asterisk, I got choppy sound when fw ip->pstn When I restart asterisk the sound is fine, Anyone had same problem? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091009/be8bbf2f/attachment.htm

Im new to Astererisk. I compiled the latest CVS and setup the server. It looks like things are working. I'm running kphone, x-lite and sjphone to test things out. The kphone (local to the asterisk server) can call and receive calls from any of the 2 windows machines. The first windows phone I start I can send/receve calls the second one I cannot. I. No matter which one I start first only

Hi all, How scalable is asterisk ? I am considering using asterisk as a VoIP platform/gateway between Internet and PSTN (switches) to offer services to home customers. What goes along with it is eventually a lot of users - upto thousands probably. Is load balancing possible with multiple asterisk boxes ? Does anyone have any sort of info/experience with such projects ? How would asterisk cope

Hi all, I am planning to use VoIP gateways to connect remote offices to Asterisk. Not having much experience with these and Asterisk I would appreciate any info/experience that you might share with me as to their interoperability with Asterisk. I am interested with in rather bigger gateways (order of E1's) from: AudioCodes - Mediant Mediatrix - 1531 Quintum tenor Multupath D3000 Has anyone

Hi, I have problems with Asterisk recognizing dtmf tones sent by Mediatrix 1104 FXS. I can not enter mailbox number (voicemail) or pin code (meet-me). Asterisk shows 'username not entered' when dialing in voicemail. Both asterisk and Mediatrix have set inband dtmf. Can anyone help me out ? Best regards, Dave