similar to: Outgoing call with bad/choppy sound

Hi I have a server with the last version of asterisk branches, zaptel branches, 2 Digium Card with TDM800P 16 HPEC channels (Echo Cancelation), 40 Grandstream BT200 and 10 Grandstream GXP2000. zapata.conf echocancel=64 rxgain=0 txgain=0 when i place a call o receive a call, I finish a sentence i hear a ssssssss, AND when the both side talks at the same time i have choppy audio. Any

Hi, I can't get dial-out working. I'm trying to call 523936. Is there something wrong with my setup here? Could someone please give me a few pointers? Regards, Norman Zhang [fwd-out] exten => _8.,1,SetCallerID(${FWDUSERID}) exten => _8.,2,SetCIDName(${FWDUSERNAME}) exten => _8.,3,Dial(SIP/${EXTEN}@fwd,70) exten => _8.,4,Macro(fastbusy) exten => _8.,5,Hangup *CLI>

Actually I am using Asterisk 1.2.7.1, zaptel 1.2.5, libpri 1.2.2 I ve tried many values for rx/txgain togeher with echocancel and relaxdtmf. The detection is not working with call file, manager originate and not with the dial command to the mobile. I have no ideas left. I got it sometimes to work if I use a specific channel (i.e. Dial(ZAP/14/...) But with the same vaules on a second call there

Hi all, i have configured incoming voip traffic as follows: [voipin] exten => _X.,1,SetCallerID(033283077734) exten => _X.,2,Dial,Zap/g4/${EXTEN} exten => _X.,3,Hangup If the call going out the pri dials with an additional '0' before the dialed number. This is for caller number AND called number. But i can't see anything that says set a '0' more in front of the

I have recently started experimenting with Asterisk. I am running the system the other side of the a NAT router and trying to connect to FWD. I have opened UDP ports and have configured sip.conf to handle NAT. The problem: I can call from the FWD phone and the extension on Asterisk rings and there is two way sound so no problem. Now if in the extension.conf file I have, exten =>

Is this replacing Astricon this year? If so it looks like a pretty poor showing in comparison to Astricon Dallas last year. Cheers, Dean ________________________________ From: Carl Ford [mailto:carlf at vonmag.com] Sent: Wednesday, 18 July 2007 9:09 AM To: Dean Collins Subject: Announcing Digium/Asterisk World's Conference Program

Same thing. It will make sense to try Register => <FWDnum>@fwd.pulver.com:<FWDpass>@fwdnat.pulver.com:5082 but in that case Asterisk sends REGISTER sip:fwdnat.pulver.com SIP/2.0 which is not right. It should be sip:fwd.pulver.com but sent thru fwdnat.pulver.com:5082 BR Borut -----Original Message----- Subject: Re: [Asterisk-Users] Using Asterisk with FWD through NAT From:

I have a small HUGE problem with *. I have installed * but I have 2 problems. 1 - Making call to FWD. 2 - Receiving call from FWD More info of the problem at the end. Here is the sip.conf file. ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = sip ;default Default for incoming calls register =>

Hi, My configuration: Sipura 2000 Debian/Sarge Asterisk 1.0.1 built by msp@toshiba on a i686 running Linux I am calling 612@fwd.pulver.com which is Daytimephoneline of pulver.com and for the first second the connection seems to be ok and I hear: <thu .. rsda ..> and nothing more, which suppose to mean <Thursday, ...>. The echo line from 613@fwd.pulver.com does work the same. I never

Using DIAX softphone which seems to be working OK can get to VM/echotest etc in the demo context Am trying to setup FWD but get the following problems Can hear it ringing when dialing FWD no 612 for time. Connects but no sound from remote end. Does anyone have any suggestions. Softphone on 192.168.0.2 asterisk on 192.168.0.3 Netgear RP114 doing NAT to the internet port 5060 being forwarded to

Hi, I've posted a simular message little over a week ago so sorry for reposting. I need to register to freeworld dial up from behind a nat. Using the xten software sip client works fine but with asterisk I don't know how to do it. Last time I posted I got different responses. Some saying I can't register with an outbound proxy from asterisk others said they have done it. If it is

Can multiple FWD accounts be registered? I have the following output in my sip.conf file: register=74928:xxx@fwd.pulver.com/74928 register=75160:xxx@fwd.pulver.com/75160 register=74573:xxx@fwd.pulver.com/74573 [fwd-74928] type=friend secret=xxx username=74928 host=fwd.pulver.com [fwd-75160] type=friend secret=xxx username=75160 host=fwd.pulver.com [fwd-74573] type=friend secret=xxx

Hey all, I'm trying to get my Asterisk server up and running on fwd.pulver.com just to get the hang of it until I get my FXO card in a couple of days. It seems to connect but that's about it. If I try to dial into it from another fwd # it says user is not online. In sip.conf I have the following added: register => xxxxxx:xxxxxx@fwd.pulver.com/489125 [fwd.pulver.com] type=friend

I'm hoping someone can help me out with this. I am basically just trying to separate my outbound and inbound calls into separate contexts instead of having everything in a single context. Any help would be appreciated. Perhaps I've missed something really obvious.... Here is the network layout: <remote> <--TDM400P--> <nattedbox> <--IAX--> <liveipbox>

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

Too bad it is August 6th *P.S. Remember, as a member of the Digium Family we have secured a special discount of 50% off of the conference fee for you if you register by July 29, 2007. To take advantage of this limited time offer, please register here <https://secure.pulver.com/cgi-bin/von?mode=gpur&conf=dawfal07&type=g&pricode=billm>!* Digium, Inc wrote: > > If you

I have tried everything and still can't place / receive calls from the fwd network. At one point today I was able to call my test machine on the fwd network, I'd answer the call on the test machine (which stated Call Connected), but then the computer I was calling from, through the Asterisk server would give me a 403 Error. I am using sjphone software. I am able to call various

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 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.

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