similar to: DISREGARD!! Broadvoice outgoing problems

Hello All, I'm just getting into *, and trying to use a Broadvoice account. It works inbound, but Outbound fails no matter what sip.conf parameters I try. From the recent posts here I think it could be: A bad CVS release - I will try to download and build from a new one Broadvoice not challenging and/or Asterisk not responding with an Authorization: in the INVITE header. I am

I have been looking around for Outlook Integration for Asterisk. Saw the Asterisk TAPI wiki page and also ran across this: http://www.fonality.com/pop.cgi?page=pop_pbxtray.tt (PBXtray) It looks like Fonality has managed to make an app that does screen pops and allows click to dial. Has anyone else been able to get this all to work successfully? Looks pretty slick.

you can not ping that address because ICMP is turned off. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Deon Rodden Sent: Monday, July 26, 2004 2:22 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Broadvoice problems again Attn: James Greetings, C:\>ping 147.135.8.129 Pinging

I had my asterisk configuration working very well with broadvoice, but it stopped working this afternoon. I plugged the Cisco 7960 phone I used for my origional signup (just a few days before they offered generic BYOD) and it works fine. I did notice it seems to do all of its comunication through proxy.broadvoice.com (I used tcpdump). I have never contacted broadvoice about using asterisk

I have a couple of DID's through VP Connect and have been having sound quality issues on incoming calls. During the call, the calling parties voice sometimes sound like it is crackling, in other words it is not very crisp. I would liken it to listening to a radio with a blown speaker. This sound defect comes and goes throughout the call. The other person is always audible but it just isn't

I've installed dovecot 1:1.1.11-0ubuntu4.1 on my jaunty system. Access from the same host works well, but when I access remotely (from various MUAs and networks) I get *very* frequent disconnects. Authentication is fine, and I can usually open a mail folder, but larger operations such as copying 20 messages into an imap folder always get disconnected. I can reconnect, but I often get

I am trying to do the following: 1. Call comes in to my * box over IAX (VP Connect DID) 2. Check to see if call should be forwarded to my cell 3. Forward the call to my cell phone and take * out of the media path. I am able to do all of the above except * is not able to natively bridge the call. I am using sixtel and for the call forward portion, but the calls don't connect before sixtel

I am running * 1.0.5 and have been having lots of problems with outgoing calls and their sound quality. I am using ULAW for the codec and sixtel for termination. Basically the problem is that portions of the call seem to be lost and replaced with silence. Sometimes I can't hear the person talking othertimes they can't hear me. This situation comes and goes throughout the call. Bandwidth

Asterisk v1.0 is running on RH 9. I installed festival RPM (festival-1.4.2-16.i386.rpm) and edited the festival.scm file to add: (define (tts_textasterisk string mode) "(tts_textasterisk STRING MODE) Apply tts to STRING. This function is specifically designed for use in server mode so a single function call may synthesize the string. This function name may be added to the server safe

Hi, I got a broadvoice "business" account under the byod(bring your own device) program. I have applied the patches and created new asterisk debian packages. I have the account working on inbound and outbound. The problem area is outbound caller ID. I have 3 other accounts with IAX2 providers and have no problem setting the caller ID on outbound calls. I called them and they

Is there any way to change the outgoing caller id on BroadVoice I have tried SetCallerID(Name <Number>) but that does not work Thanx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050126/bbd95fcf/attachment.htm

Here is the call flow: [ivr-incoming] exten => s,1,LookupCIDName exten => s,2,DigitTimeout(2) exten => s,3,ResponseTimeout(10) exten => s,4,Wait(1) exten => s,5,Background(custom/ivr-incoming) exten => 1,1,Background(pls-wait-connect-call) exten => 1,2,Dial(${RINGPHONENUMBERS},20,r) exten => 1,3,Voicemail,u${VMBOX} exten => 1,4,Hangup Running * 1.0.5. The calling party

Hi, I can't make outgoing calls via Broadvoice. I have tried each and every configuration that was posted to list previously. I am able to receive incoming calls fine. I get the following in asterisk console: ===================================================== asterisk*CLI> show version Asterisk CVS-HEAD-03/10/05-22:51:28 built by vicky@asterisk on a i686 running Linux

>>> TL;DR: please try the patch out and report if it causes "Did not receive >>> identification string" log messages. I believe it does not. Aw crap. My homegrown anti-dos tool for ssh looks for either DNRIS or if logging is verbose enough a connection that didn't result in a login. I give the attacker a few tries and whitelist any successful candidate so I

(Please keep me CC'd as I'm not subscribed to the list) I'd like to request that folks running RELENG_8 (and RELENG_9, though I do not use it) please check the behaviour of their terminal after each of following commands are run (check terminal after each command): top -a (press "q" after 1 screen refresh) top -b If you find that your input characters in your shell

Hello, I have a problem with connecting a Digium X100P card to a Brazilian analog line. Can somebody help me out with this problem? My /etc/zaptel.conf is loadzone=br defaultzone=br fxsks=1 My /etc/asterisk/indications.conf [general] country=br [br] description = Brazil ringcadance = 1000,4000 dial = 425 busy = 425/250,0/250 ring = 425/1000,0/4000 congestion =

Post Office wrote: > Sorry. Your message could not be delivered to: > > Jonathan DEL CAMPO /jdc/ .Y (Mailbox or Conference is full.) Would one of the list admins PLEASE remove this guy from the list? I've been getting NDRs from him for as long as I can remember... -- Best regards, Charles

I can't seem to dial out with Voicepulse Open Access service using *. Incoming works fine. Another user posted a few weeks back that they were having problems and there are some threads at dslreports.com about this as well. Maybe someone here can figure out what the issue is from the sip debug info below. I am at a loss. The audible error message from Allison is 0984 (from VP server) Here is

So disregard this question. I followed Mason Hale''s suggestion that I pass in some worker_args and all is well in the scheduler.yml file. Brian _____ From: Brian Lucas [mailto:blucasCO at gmail.com] Sent: Thursday, February 15, 2007 2:49 PM To: ''backgroundrb-devel at rubyforge.org'' Subject: new syntax for autostart and repeat Hello: I''m a

disregard my map.resources :foo_items, :as => :foo question I was pulling some rookie moves. Thanks Mike -- Mike Mondragon Work> http://sas.quat.ch/ Blog> http://blog.mondragon.cc/