similar to: echo cancel from telco

I'm using bristuffed Asterisk with ISDN/ZAPHFC I have VERY ugly (outgoing) sound through ISDN/HFC if echocancel=yes in zapata.conf, but without echocancel I have bad (incoming) echo Through PSTN/FXO sound is ok with or without echocancel. I tried other echo cancellers (in zconfig.h) two times: ECHO_CAN_KB1 (this was default) ECHO_CAN_MARK2 ECHO_CAN_MG2 after any change I compiled (make

My TE411p does not seem to like to have two PRIs from different telcos (span 1 and span 2). I can get one working, but not the other. However, if I use vpmsupport=0 when loading the wct4xxp module, they both work. But here is the problem, vpmsupport=0 disables the on board echo cancellation. Any ideas? BTW, here is zaptel.conf span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs bchan=1-23 dchan=24

Hello Everyone, Unlike going through DIDx, DIDLogic etc.., we have an option of getting DIDs directly from local telco Bell Canada. Currently our SIP Trunk provider assigned a DID to us, and as you know, they just redirect requests it to our PBX. However, when dealing directly with a telco, what equipment will we need? Basically giving us the same capability as a DID provider. If someone can

Hello, I have a crossover PRI(Asterisk server to PBX) and a regular telco PRI T1 line and currently have echocancel=yes and echocancelwhenbridged=yes on those spans in zapata.conf. I was discussing CPU load with another Asterisk user and he mentioned that PRIs don't need echo cancelation and that turning it off will reduced CPU load on the server. I checked many sample configs and the

I have a channelized T1 coming in from our telco, terminated onto a TE405. There are three channelbanks serving internal analog extensions, and about 10 Cisco 7960s. I have no reports of echo on the analog extensions (as expected). The 7960 users complain of occasional echo (seems like 1 in 5 calls). Only the SIP user hears the echo, not the caller. I have echocancel=yes, echotraining=yes,

I use a TDM400 card to interface with my telco. I used asterisk voice mail. However, if I'm on the telco line while another call comes in, obviously it cannot go to asterisk voice mail but instead bounces to the Telco voice mail. Is there any means by which I can get asterisk detect the Message Waiting Indicator (MWI) from the telco? Is there some application or variable which can be

> > hi, > > How good or bad is the EC in Asterisk? > > Can anyone prove that it works at all and what it's limitations are? > > I ask cause I have some problems with this myself which variate from > call to call, and I see from others that Echo Cancel is a quite common > topic. > > Jan > > I have installed systems that had absolutely no echo

Greetings, Our telco called last week, saying that a lot of channels on our PRIs are blocked. And with blocked they have the following description in the Siemens exchanges: BBAC BLOCKED BACKWARD This status is set when the partner exchange has a blocking set and the signaling of the trunk (non-CCS7) is able to report this blocking in the backward direction. This status can

Hi guys, I spent a couple of hours in Google, but the problem appears to be uncommon, so I'd like to ask about it here. The problem is exactly the opposite to "Asterisk does not detect FXO hangup". In my case it's the Telco who does not appear to be detecting Asterisk's hangups. Telco is Telus in Vancouver, Canada. The setup is very simple - Telco -> FXO/TDM400p

Hello Please help me, i need transfer a call in asterisk to other telco number and free the channel. Can i do with any q931 function?. Thanks a lot Aris... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090528/2bcd93ae/attachment.htm

Ok, so I'm setting up my small office. I have my asterisk machine setup and I have 3 sip phones connected as my stations and a 4 port FXO card ready to go(planning on only using 2 lines currently). What should I now order from my telco(sbc in this case) Everytime I call, they want to sell me this expensive $50 package that bundles everything and that's for a single line. Is there a

Hello, We have an asterisk 1.2.13 box that use a Digium TE205P T1 PRI connect to the telco, users mainly use snom 320/300 SIP phones. When dialing to an external phone number with custom ringback music, users reported that they could not hear the music but can only hear the standard ring tone generated by the system. Is there any kind of settings need to allow the ringback music pass to the

as my first project with *, i would like to replace our old neax2400(sds) with an * server. i've got an X100p and a TDM400 on hand already. for the CO lines, the X100p works ok with fxsks signaling though there are still strange things happening every now and again but more testing is on the way. my real big problem is the DID lines which our telcos call DDI lines; (incoming calls only) i

I've upgraded to asterisk-1.6.0-beta2. I'm trying to get the new Telco MWI detection function working. It doesn't appear to be working. I have this in zapata.conf ; PSTN connected here ;immediate=no ;busydetect=yes ;busycount=8 ;musiconhold=default mwimonitor=yes ;mwilevel=512 mwimonitornotify=/usr/local/sbin/zapnotify.sh faxdetect=incoming signalling=fxs_ks context=incoming

I've been having trouble with making outbound calls to my TELCO from a TDM400 card (FXS KS signalling) after upgrading from 1.6-beta9 to 1.6.0. The problem is completely intermittent. When it fails, I get this message: [Nov 9 19:12:26] WARNING[18916] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) At some point, it starts working, but I don't know what

Hello all. I have an NEC PBX connected via a TE210p E1 line to an asterisk 1.6 box. NEC -> E1 -> TE210P:1 -> * -> TE210P:2 -> E1 -> Telco Incomming calls from the telco to the asterisk box to the NEC work fine with indials and everything. Works sweet. Outbound from the NEC to the Asterisk box fail. Giving an long dial tone that then times out. Ie, pick up NEC handset, dial

I am trying to obtain optimum gain settings for a bank of analog lines connected to a channel bank. My telco has provided a 'Type 102' test line to use for incoming level calibration. This is functionally equivalent to app Milliwatt(), but provides tone from the CO inwards. Question is, how should one use this a 0dbm test source with ztmonitor? Am I correct in understanding that a 0dbm

>Hi Jerry > >As others have eluded to, the 'PTT' feature can mean different things to different >people depending on their background.> > >Is it fair to say that you're looking for a one-touch button which initiates a call to >the other end and causes the other end to automatically answer in speakerphone >mode? >If that would foot the bill then have a

Hi! My telco is Deutsche Telekom and they got about 30 SIP servers right now. Currently I've set up a template for incoming calls in sip.conf and added each SIP server by it's IP address like this: [DTAG-in-1](DTAG-in-template) host=217.0.16.103 ... [DTAG-in-30](DTAG-in-template) host=217.0.20.99 I've done that to improve security and to be able to assign all calls coming in

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.