similar to: Echo on remote end when using NuFone

Ernest, I hadn't thought of doing that, though having that added protection would be nice. However, what I'm trying to do it have an incoming call at my home number follow me to my cell phone for selected numbers -- Since I already have three way calling, I'd like get Asterisk to essentially three way my cell phone into the call (or my office number, etc.) I understand the

Can anyone tell me the features of the Cisco 7905 with SIP? I mean things like number of lines, speakerphone, transfer buttons, etc. I've seen the Cisco material, but all it told me was how nifty it is and how wonderful the XML interface will be ;) Thanks, --Ernest

I notice the zaptel Makefile option the mark2 option & KFLAGS+=-DAGGRESSIVE_SUPPRESSOR is now gone. Does simply adding these options still compile in a certain echo can - or is there an other method of activating a particular can. I have not had to update my machine that is connected to pstn for a while & I don't want to jump into echotraining without a way to quickly enable what has

Hi all! I updated my snom200 to 2.02t and now MOH from * don?t works anymore... only the MOH from snom server and if i clear the MOH server field in the phone i have no MOH at all..( with the transfer button, moh plays using a extension). Someone with that problem? I downgrade to 2.01s but nothing changes. Miklos -------------- next part -------------- An HTML attachment was scrubbed... URL:

We are trying to implement "area-code dialing" in our asterisk PBX. Basically: we will have a number of customers, who may be in different area codes, that want to direct-dial each other's extensions. We want this to work like a "real" centrex, in that seven-digit numbers should try (1) "local" VoIP extensions, and then (2) "local" PSTN numbers.

I am having some oddness with the 11/11/2003 CVS of *. Specifically, outgoing audio to NuFone doesn't seem to be transmitted (I can hear the other side just fine). My firewall is set to allow all outgoing traffic, and the IAX2 connection is definitely established correctly. Also, I can watch UDP traffic going by on the firewall so I know that * is transmitting. This happens with X-Ten on

Hi all, While pondering my choices for local dial tone service via a bunch of POTS lines or a T1, I began to wonder if perhaps there is another way. Are there VoIP dialtone providers? That is, could I use only my internet connection for voice calls and not have a separate T1/POTS bank for that? I guess I am imagining a company that gateways between the PTSN and the internet backbone.

Is anyone on the list using an Audiocodes gateway with asterisk and SIP? I'm looking at that platform, but I have a couple of issues: 1) Echo cancellation. The echo that I'm hearing with an X100P is unacceptable. Does the Audiocodes do better? 2) Line signalling. I'm using Kewlstart with the X100P, but it looks like the audiocodes uses loopstart only. How does this work with

I'm using the SNOM 200 latest SIP beta (so that I can have the GSM codec, etc). Everything seems to be working fine, but the music on hold doesn't play when I use the HOLD button on the snom. Any suggestions? Thanks, --Ernest

After speaking with several people, and even participating in a forum of several other people with echo issues, I thought I'd share what we've done (well actually what our chief R&D engineer, Brett Bourn has done...) First let me say that normal cheapy PC hardware couldn't be made to function with out echo. We tried on both the single port Digium T1 card and the 4 port Digium T1

In response to the postings by Andrew Kohlsmith and Ernest W. Lessenger: Andrew, I modified the exten line in extensions.conf as you suggested. Unfortunately, It still does not work... Ernest, I spent approx. 4 hours reading list archives (and anything else Google served up) on how to configure iax.conf and extensions.conf to work with Voicepulse. Then, I sent an email to voicepulse

Anybody here having any problems with nufone? Calls are not going thru, when trying to call their customer service number it doesn't go thru. When trying to resolve www.nufone.net I get (sourec: http://www.dnsstuff.com/tools/lookup.ch?name=nufone.net&type=A ): How I am searching: Searching for nufone.net A record at j.root-servers.net [192.58.128.30]: Got referral to M.GTLD-SERVERS.net.

Hi, I've been using nufone DIDs for months with no problem. Now there are codec problems that prevent any kind of calls working. For example, May 5 13:04:12 WARNING[928]: channel.c:2115 ast_channel_make_compatible: No path to translate from IAX2/NuFone@NuFone/25(256) to SIP/wengo-out-968a(4) May 5 13:04:12 WARNING[928]: app_dial.c:1006 dial_exec: Had to drop call because I couldn't

Hello, I've signed up for a NuFone account, and added the following instructions to my config files per NufFones directinos: iax.conf [NuFone] type=peer host=switch-1.nufone.net secret=password extensions.conf (under the [default] context) exten => _1NXXNXXXXXX,1,Dial,IAX2/f00b3r@NuFone/${EXTEN} I then get this message in the CLI: -- Executing Dial("SIP/jake-fe5d",

Hello, I am trying to configure the my asterisk box here with the following **iax.conf*** [NuFone] type=peer host=switch-1.nufone.net secret=xxxxxx ***extensions.conf:*** exten => _1NXXNXXXXXX,1,Dial,IAX2/xxxxxxx@NuFone/${EXTEN} exten => _011N.,1,Dial,IAX2/xxxxxx@NuFone/${EXTEN} I have a couple of Xlite softphones and 2 analogue phones connected to a mediatrix 1102 connected to our lan.

Good evening, I just signed up with Nufone and I am able to receive calls with no problem via my 800 number. Outgoing calls are not going through though. My extensions.conf is as follows: [nufone-out] exten => _91NXXNXXXXXX,1,SetCallerID(mynumber) exten => _91NXXNXXXXXX,2,Dial(IAX2/user:pass@switch-2.nufone.net/${EXTEN:1}) exten => _91NXXNXXXXXX,3,Congestion Whenever I try to

Is nufone having problems taking gsm calls today i had some issues dialing overseas to call my folks. here's snip of what the console displayed -- Executing Dial("SIP/2204-a279", "IAX2/fgravato@nufone/011351217907000|100|T") in new stack Jan 18 10:30:02 WARNING[1200884528]: chan_iax2.c:5036 iax2_request: Unable to create translator path for UNKN to GSM on IAX2[NuFone]/1

The echo canceller algorithms aren't doing anything. We get extreme echo during the conversation, it appears even before the call connects, the echo is there... This only happens with SIP to/from WCFXO (analog POTS). Looking at the Zaptel configuration: /etc/asterisk/zapata.conf: echocancel=yes echocancelwhenbridged=yes rxgain=0.8 txgain=0.8 (although none of the above options

Hello: . I'm having troubles registering on nufone's IAX service . I'm really new to Asterisk Nufone provide me some config examples ... I can dialout but I can't register my * Box, eg. whe I do "iax show registry" I got only a "Request Sent" and later I have a "Timeout" My box is on a Public IP and no firewall. I'm using *0.5 on a

I have ASTCC installed, and compare it with NuFone, however, I find that the billing of NuFone is always a few secondes more (6 to 24 seconds) Does anybody has an explanation / solution for it? bye Ronald