similar to: Broadvoice

Am I doing something wrong? I can't get this dial command to timeout.... Dial(Zap/g1/xxxxxxx,20) -- Gary White admin@netpathway.com Network Administrator Internet Pathway 105 D East Church Street Voice: 601-776-3355 P. O. Box 777 Fax: 601-776-2314 Quitman, MS 39355

Guys, I have a problem that I can seem to run down. I'm running CVS-HEAD-09/10/04-18:14:15 on a Celeron 500 with an Intel chipset motherboard. The audio from my GS phone to * is sometimes decoded at about twice the normal speed on some outgoing calls. Also.when recordind a message to * from the GS it also does the same. This happens about 1 out of every 5 records or calls. This only

I am after using a web crm system which has a button to then get asterisk to dial the contact. For this I was looking at call files, which appear good for the job, I have one small problem with them though. 1/ file is created 2/ external number is called 3/ the external party answers 4/ the external party now hears ringing as you extension is now being called - bad! What I would like to

Today, We have added INVITE Authentication. This seems to bring a large amount of problems to people in the way since they can't make outbound calls. Here's what needs to be done. You need to add three variables to your peers or friends, username, authuser, and secret. username=<phonenumber> authuser=<phonenumber> secret=<registration password> Dan

Is the Broadvoice service up? I just signed up with them and started receiving calls in no time but could not make calls. And after a few minutes I cannot even place calls. register => [number]:[password]@sip.broadvoice.com [broadvoice] type=peer fromuser=[number] host=proxy.lax.broadvoice.com fromdomain=sip.broadvoice.com context=from-broadvoice dtmfmode=inband any help would be

Guys, This is not really a bug but a question. When starting *, using the -c and entering a running *, using the -r command, you get color text. But, when starting * with no option, as I have it automatically do from the rc.init scripts, entering *, with the -r command, * does not have color text. Why? -- Gary White admin@netpathway.com Network

Greetings, I have a problem making a call from Asterisk to Cisco H323 PSTN gateway using H323 channel. I can call but there are no sound in both way. If I call H323 gateway directly from SJPhone I have no problem with sound. Any advice are welcome. Thanks in advance.

Hi, Is anyone using mozPhone? If so any feedback you can provide? Thanks, Glenn

I have added the three lines to the sip.conf file based on the latest changes from broadvoice. I can receive incoming calls but cannot place any outgoing calls. The error I get is: *CLI> -- Registered to '69.73.19.178', who sees us as IPADDRESS:4569 -- Attempting call on SIP/Broadvoice/5068012 for application Playback(demo-congrats) (Retry 1) Mar 8 08:35:21 NOTICE[29290]:

Hello all, When I call the Broadvoice number all is good. When I try to call out through DISA on my broadvoice line i get the following: Executing Dial("SIP/147.135.0.129-0815bc60", "SIP/16037862111@proxy.bos.broadvoice.com|30") in new stack -- Called 16037862111@proxy.bos.broadvoice.com -- Got SIP response 480 "Temporarily Not Available" back from

I am trying to setup one time caller id block on my system(activated when an incoming call matches *811XXXXXXXXXX), and I have had little to no luck. Could you take a look at my context/macro definition and help me figure out what I am missing? Here is my context for my dialplan: include=default plancomment=user-default

-------- Original Message -------- Subject: Re: [Asterisk-Users] BroadVoice configuration changes for Outbound Date: Sun, 06 Mar 2005 19:11:22 -0500 From: MF Hulber <mark@hulber.com> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com>, dan@mirrorlynx.com References: <200503060703.XAA12457@comand.net>

Hello all, I have been searching thru all docs that I can find on wiki and such but can not get an answer. I am trying to collect a date from user input in the form of digits dialed from the phone to use in an agi script to do a database look up. I have tried to use "Get Data filename, timeout, maxdigits " in the agi script. In * console I get message saying playing filename but it

Hello all, I have seen some chatter again about DTMF. I see most of the talk about DTMF around not being able to get an external IVR to recognize digits, not a big issue for me at this time but sill interesting. My issue though, is with talk off on a zap channel. It seems to be getting worse or maybe my patience is thinning. All my calls go out and come in pstn through an FXO as I do not

I want to buy a new server to run Asterisk and after looking at prices for the Athlon XP 3000+ it costs the same as an Athlon 64 at the same speed rating. I was wondering if Zaptel/Asterisk will compile/work on an Athlon 64? -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V.

Hello all, OS OpenSuSE 10.3 * ver 1.4.26.2 zaptel ver. 1.12 Digium TE122 I have a request for remote users to be able to dial through the system so that the sales managers can barge/chanspy on the sales force. I have the DISA part working with authentication(rather straight forward) but what I can not figure out is how to enable the supervisors to be able to barge on these calls. Is there a

Hi, I have two accounts with broadvoice. Now, I want to be able to distinguish between them. I though that this would be simple by adding "/EXTEN" at the end of the register statement. For example: register => num1:pass@sip.broadvoice.com/1000 Unfortunately, this is not working. When I call into my box I hear busy tone. My config looks like this: [root@voip asterisk]# cat sip.conf

Hi all, I've been trying to use BroadVoice as a SIP service provider. They don't officially support * but are helpful when it comes to answering questions for setup parameters. They claim they have no firewalls or access lists that need to be set up so I can get access to their servers. However, something's still not quite right when I use the parameters. It looks like our Asterisk

Hello All, I have 4 Broadvoice lines. If I call any of the lines it shows that is coming from the first line. exaple register=XXXXXXXXX1@sip.broadvoice.com:passwd:XXXXXXXXX1@sip.broadvoice.com register=XXXXXXXXX2@sip.broadvoice.com:passwd:XXXXXXXXX2@sip.broadvoice.com register=XXXXXXXXX3@sip.broadvoice.com:passwd:XXXXXXXXX3@sip.broadvoice.com

I have configured asterisk with the AMP php configuration utility. I am able to make outgoing calls through broadvoice but incoming calls are sent to BV's Voicemail and never actually enter the IVR. When I show sip debug info through the asterisk prompt it actually reads the incoming call from BV but then issues a busy signal sending the call to BV's voicemail. I also modified