similar to: Broadvoice users...

I finally got my incoming and outgoing to work on Broadvoice with a configuration file that is no where close to the one given by them. Here it Is (sip.conf). For others who have a working config could u please share so that I could compare. Thank You [broadvoice] type=friend username=[number] fromuser=[number] secret=[password] host=sip.broadvoice.com fromdomain=sip.broadvoice.com

If you see a wj0011 version of firmware for Zyxel Prestige 2000W floating around (I found it in a German forum), KEEP AWAY. It completely trashed the wireless networking in my phone. -- ========================================== Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 FWD: 512237

P.S. I _had_ read the other posts that suggest changing to unicast sntp mode. This didn't help. I eventually setup a new ntp server on my LAN, and used it as a sync source. Everything seems OK now. Obviously a problem with my other ntp server. Cheers. -------- Original Message -------- Subject: Cisco 7940 - Disappearing Clock Date: Thu, 28 Jul 2005 11:50:35 +1000 From: Rod Bacon

I was wondering if anyone had experienced the following with asterisk stable. After a period of time (can vary), If I stop asterisk and try to run ztcfg -v to reinitialise my quad e1 card, the server will lock up. Sometimes it's a complete lockup, where it won't even return pings, other times it seems to be "partially screwed". -- ==========================================

I am wanting to front-end a legacy PBX with an asterisk box. I have done plenty of asterisk work over the last 6 months to PRI circuits, but not with a PBX being involved. I know I can use asterisk and digium cards in this manner, but do I need separate cards for the PRI -> Asterisk side to the Asterisk -> PBX side, or will a 4-port PRI card do the job? (I already have a spare one of

I'm new to H.323 and I have noticed that there are two separate channel drivers for * available - the inbuilt one, and oh-323. I had trouble compiling oh-323 with the current cvs stable, so I tried the inbiult one (with specifiec recommended versions of openh323 and pwlib). It compiled cleanly but I am told that it is not recommended (unstable?). Can someone with first-hand * H.323

I posted something on this a week ago, at which time I was told that this was an 'old' issue. Since then, I've spent hours looking, but can't find the answer. For some reason, some of my CDRs (both to CSV and MySQL) are being written with the wrong callerid. As best as I can determine, they are being written with the CLID of the _last_ caller to access the specific ZAP channel

I've posted this to SNOM, but was wondering wheter anyone here has issues with SNOM 190 phones not showing the correct DST adjusted time (using the latest firmware). -- ========================================== Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 ? ?Fax: +613 99401650 FWD: 512237 ? ? ? ? ? ? ? ? ?

When I dial a PSTN (via * with Digium Quad-E1) number, I hear two sumultaneous ring tones. One is coming from *, the other I assume is from the ATA. Is there an intelligent way to get around this?

I have googled until blie in the face, WiKi'd until physically exhausted and searched through every Asterisk repository that I can find, all to no avail... No matter which version of SpanDSP I use, with which version of libtiff, Asterisk, ... I simply cannot send faxes. I can receive faxes *perfectly* (sending from standard fax macine on POTS) into tiff file (via Digium analogue card),

Can someone who is knowledgable in the traditional telco space please give me a layman's explanation (or point me to an appropriate url) of LBO as per the zaptel configuration file? # The line build-out (or LBO) is an integer, from the following table: # 0: 0 db (CSU) / 0-133 feet (DSX-1) # 1: 133-266 feet (DSX-1) # 2: 266-399 feet (DSX-1) # 3: 399-533 feet (DSX-1) # 4: 533-655 feet (DSX-1)

The one I like: http://www.rhetorical.com/cgi-bin/demo.cgi is toast. I think they went broke or got aquired by someone. Also, is there a Festival voice that sounds as good as Rhetorical or the AT & T stuff? The default one is barely legible. Since Festival is a little brutal to configure, I'd like to get someone's recommendation then go through the pain of reconfiguring it only once.

HELP! Ok, so I have the following SIP.CONF: [general] context=default port=5060 bindaddr=10.1.1.200 externip = XX.XXX.XX.XX localnet=10.0.0.0/255.0.0.0 disallow=all allow=ulaw allow=g729 allow=g726 allow=alaw register => ##########@sip.broadvoice.com:XXXXXXXXX:##########@sip.broadvoice.com/1234 [sip.broadvoice.com] type=peer host=sip.broadvoice.com fromdomain=sip.broadvoice.com

I did a quick google search of the lists site and couldn't find a definitive answer, so if it's there, I apologize for asking again. Starting about noon yesterday, I am no longer able to send/receive calls via Broadvoice. When calling in, I get a fast busy, and when calling out I get the following error: -- Executing Dial("SIP/112-572a",

Hi, all. The asterisk setup is working fine, receiving calls via broadvoice "initially". ? When call comes in via broadvoice number, asterisk picks it up and routes correctly, as long as the call came in within ~2 min from the previous one. In other words, as long as a call comes in within ~2 min since the previous one, asterisk will answer the call. However, if the call comes in

Is anyone else experiencing 404 errors on outbound dial with Broadvoice? I've followed the instructions posted by Broadvoice to configure sip.conf, and inbound calling works fine. Every time I try to dial out, I get a 404 "Not Found" error. Here are the relevant sections from my configs. sip.conf: context=broadvoice-in register =>

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

So it was all working well and then suddenly I'm unable to get incoming calls from BV. Outgoing is fine. I'm using AAH. I have the following settings; register=9738281625@sip.broadvoice.com:PASSWORD-GOES-HERE:9738281625@sip.broadvoice.com/2208 [broadvoice] username=9738281625 user=phone type=peer secret=PASSWORD-GOES-HERE qualify=1000 port=5060 nat=yes insecure=very

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

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