similar to: A-law distortion

I'm not sure where to check next, so I'm reaching out to those that know this stuff better than I. I've got Asterisk up and running, but I've still got an occasional audio issue. Once in a while (maybe 1 out of every 20-30 calls), the audio becomes heavily distorted, but only on the local side. The party on the other end says the audio is fine. We can hear them, although

Hello, i'm using a TE410P on some E1/PRI with EuroISDN and experiencing a few audio quality problems with current CVS (both zaptel and asterisk) and the following network ISDN public SIP/zaptel network ---- pri --- ASTERISK GW --- iax --- ASTERISK PBX --- PHONES w/ any codec the rx (public network to local

Hi, I am trying to get a prototype working based on Asterisk and Quicknet cards. I currently have to systems set up each with a LineJack card. I have the systems working, but can't get the voicemail demo to work properly. Messages to the user telephone set from the voicemail system are clear, but recordings left through the phone are distorted. Sounds like the audio has lots of echo and

Hello, i have succesfullu setup asterisk with Sangoma E1 card, evrything works well but i don't know how to pass indications from telco switch to the user - when users call bad number telco switch shuld talk "unallocated number" but its only send PRI_CAUSE 1. How to pass voice indications thru asterisk to clients? My /etc/zaptel.conf: span=1,0,0,CCS,HDB3,CRC4 dchan=16

Hello, We have an asterisk setup at our client's site using a TE205P. The line to telco is a 23 channels T1 PRI, however the line has random echo problems (about 5-10% of the calls)! Can anybody tell me if echo cancellation is really needed even at a T1 PRI to the telco? Because people keep saying when they deploy voip solution in Hong Kong using T1 PRI, there is no need of echo cancellation.

Hi all, I have a brand new TDMxxx card with 3 FXO modules and one FXS. It has replaced my old 3 X100P cards. The FXO part work as before, after some adjustments in the rxgain/txgain part. The problem I have is with the FXS module. I can place calls to SIP/IAX or PSTN destinations without any problems, but the sound received by the other part is much to strong and a little bit distorted. I have

The value of 14800 is correct. I had issues with my TDM400p with 2x FXO's installed and using the Xlite client. I could not get the echo stable at the initial call. Changing to a hard phone made everything work correctly. I still had problems with the off location I called, but mostly worked great. To see the TX gain I created an extension with the Milliwatt command attached to it and called

I have a TDM400 running telco lines on ZAP2-4 My after hours config is supposed to receive the incoming call then divert it to my home phone by calling out one of the other zap channels available. console output as such... Starting simple switch on 'Zap/3-1' -- Executing Dial("Zap/3-1", "ZAP/G1/0823274210") in new stack Aug 20 19:06:44 NOTICE[646]:

Hello list... 1) Is there an IAX softphone that supports any kind of voicemail indicator? 2) I have 2 TDM400Ps installed in a system. I need the audio on the analog phone (FXS modules) to be amplified somewhere between 10 and 15db so I set rxgain and txgain to 15 for the FXO interfaces and FXS interfaces. When a call comes in on the FXO at this setting, the call sometimes has about 20 seconds of

Firstly, the problem... Ever since I installed and setup asterisk, I have had various problems, initially it was echo caused by the ISDN (isdn4linux) card I was using. So, I upgraded to the X101P from digium. I still had echo, so I figured it was also caused by the ATA186 (cisco) I was using. So, I upgraded again to the TDM40B quad FXS card. This solved pretty much all my problems, except

Hello all! I have tried and tried to resolve this one to no avail. Hopefully one of you can help... The system in question is a Compaq Evo D600 (iirc) business desktop, with a 1.4GHz Pentium 4 and 512mb of RAM, running a stock install of PoundKey 1.2. It has two Digium cards installed: a TDM400P with four FXO modules, and a TE110P hooked to a Carrier Access Adit 600 which serves 8

On Wed, 13 Dec 2006, Wireless wrote: > Hi Gordon > > I too have this problem with one of my two BT lines, very very annoying. I > was using a TDM400P and am now using a Sangoma A200 (lightning got the TDM), > I think the sangoma is very slightly better (less echo) but I might just be > kidding myself. Another think I've found it that using a faster CPU (now on > a 450

Hi, I have been experimenting with Asterisk and the linejack card and have discovered the follow: 1) The linejack in FXS mode works well with asterisk as long as the format in the phone.conf is set to ulaw. With mode set to slinear, gsm recorded messages played by voicemail appear to be slightly choppy (sounds like a click at the end of each frame). 2) Using the configuration above, the

I've got a system where I'm integrating a Nortel Option 11c with a Trixbox 2.0.0 system using a Sangoma A101 T1 card. (Running on a Dell PowerEdge 350) We've got things mostly up and running and all seems well... except... If I call from a SIP extension (X-lite soft phone) dialing 9xxxx where xxxx is an extension on the Opt 11, the call goes through to the Opt 11 but I have terrible

Hi all, I have a brand new TDMxxx card with 3 FXO modules and one FXS. It has replaced my old 3 X100P cards. The FXO part work as before, after some adjustments in the rxgain/txgain part. The problem I have is with the FXS module. I can place calls to SIP/IAX or PSTN destinations without any problems, but the sound received by the other part is much to strong and a little bit distorted. I have

Hi all. I have this configuration: Telco <-----(E1)----->TE410P//Dual Xeon Server 2.4Ghz<-----(Ethernet)----->Switch<----->GS//BT The Server is running RedHat Linux 8.0 with kernel 2.4.18-14-smp and we are having the following 2 issues: 1.- When making calls from the GrandStream to the PSTN the audio is choopy, plus theres is a pulsing sound, but when the GS

BUG/SYMPTOMS: 1.Under certain circumstances, octoBRI (and most likely quadBRI) ISDN cards (Junghanns/CologneChip) severely distort certain ISDN payload. 2.Although these claims relate to the bristuff patch, the problem might not be limited to bristuff and in fact be rather asterisk/zaptel/libpri related. 3.Signal distortion is limited to the use of the CVS version of the Bristuff patch for

I am trying to receive both CID and DNIS from the telco through a non-pri T1. Currently I have the T1 setup and operational both outbound and inbound calls are completed as should be expected. The calls came in and were placed in the context specified in zapata.conf on exten => s,1. I have requested that the telco provide callerid (they call it ANI) along with 10 digit dnis for my 800

Hello List! We have problems setting the right caller id on outgoing calls. The Asterisk Pbx is located in Bucarest(Romania), our Telco provider is rcs-rds.ro. We have the local telefon number 40787 00-99, associated to our PRI E1 Line. Where 00-99 are the DID numbers available. The telco is aspecting a 3 digit long Callerid from us, for example like "710", for the extension 10.

so i have been running v1-0 on all of my test boxes for about a month now testing iax/sip/res_xxx. I decided to put it into production so I updated a box that was running 0.9.? that had been working perfectly for months and low and behold the inbound line from telco now intermittantly doesn't clear and none of the other channels can dial out on that line. I have tested the line in this