similar to: Broadvoice - bad quality, dtfm mode

Hello, I have two broadvoice accounts. Lately, very often my broadvoice accounts are in unregistered state. When I log into asterisk I see: voip*CLI> sip show registry Host Username Refresh State sip.broadvoice.com:5060 yyy@s 120 Request Sent sip.broadvoice.com:5060 xxx@s 120 Request Sent xxx and yyy are my ohone numbers, when I do tcpdump on my router, I see only packets comming out my

-------- 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>

can you please share the cdw part # for the $ 10 service contract ? W. Kevin Hunt CCIE #11841 MCSE, Linux+ SME www.huntbrothers.com > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Christopher Jacob > Sent: Saturday, September 25, 2004 9:51 PM > To:

This is BAAAAAAAD! Now even SIP get's "tainted"... http://slashdot.org/articles/04/09/16/1317247.shtml?tid=95 _________________________________________________________________ Surf the net and talk on the phone with Xtra JetStream @ http://xtra.co.nz/jetstream

Chuck, The first thing I would do is to upgrade the load to version 6 or higher. I'm running the latest...version 7.2. (I'm very happy with it) Are you using TFTP to load the configuration or manually configuring the 7960? I know it's a pain to setup TFTP just for a quick test. However, it's well worth it. If you have a CCO account you can find the latest load and config files

Hallo Did you allow udp outgoing on 4569 as well.. i found udp bit different than tcp when comming to firewalls liaan ----- Original Message ----- From: "Bartosz Wegrzyn - asterisk" <junk@lexon.ws> To: <timebandit001@gmail.com>; "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Monday, February 21, 2005 12:29

Hello! I have asterisk updated from CVS on 31/8/2004 with sample configuration. I have just changed the sip.conf to register asterisk with sip proxy in out intranet. Then I can successfully make call to asterisk and go to demo IVR, but no response to dtmfs. I try to make call from several sip phones: Cisco7960, Ata186, Snom200. All of them send telephone-event in INVITE, but asterisk answers

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

My DTFM is not working in current CVS-stable *-1.0.6 and *-1.0.7 but it works in version 1.0.5 (was working with 1.0.3). I'm using SPA-3000 and dtmfmode=inband -- #Joseph

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Hi list, I'm having trouble with DTFM tones detection. Usually, some tones are being received duplicated in Asterisk, some not. As you can imagine, that's a very big problem involving IVR menu options, Meetme conferences protected with passwords, and so on. We are currently using DAHDi 2.2.0.2, module wct4xxp, which is managing a Digium TE220B card, with a hardware echo canceller

Hi, Matt: What?s your server specifications that did you use? Best Regards, Cleviton. -----Mensagem original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]Em nome de Matt Florell Enviada em: sexta-feira, 23 de junho de 2006 11:38 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [Asterisk-Users] Meetme max users

Hello,Maybe I don't understand what DTMF in ASCII means but I can't make my record stop using this syntax in a PHP agi script :fwrite(STDOUT, "RECORD FILE /var/lib/asterisk/ENR/jeanpaul wav '#' 15000 BEEP s=3000\n");The php syntax isn't a problem because I really start recording, I have a beep, the record can't long more than 15sec and after 3sec of silence my

We have some guys working in the US who can't always dial back to our company in Europe easily (lots of clients require authorization to make international calls), so I set up the following: - ipkall.com number links to a FWD number - office Asterisk box registers with FWD Then I programmed Asterisk to accept office extension number using DTFM tones. This works OK. Then I programmed

Hi, Due to the broadvoice often down times I want to switch to new provider. Can anyone recommend a provider that really works and supports asterisk. Thanks

hello, I got those extensions: exten => 555,1,MeetMeCount(500|count) exten => 555,2,Gotoif,$[${count} = 1]?6 exten => 555,3,Meetme,500|pMs|1234 exten => 555,4,Playback,goodbye exten => 555,5,Hangup exten => 555,6,Goto(from-internal-custom,556,1) exten => 555,7,hangup exten => 556,1,System(/bin/cp /etc/asterisk/1-test /var/spool/asterisk/outgoing/) exten =>

I just had to deal with this yesterday. I called Cisco and they gave me a part number for the support contract. I looked around and it was $90... I posted back to this list and was happy when someone gave me the correct part number, which at CDW was $10... Not too bad. Although I can't believe Cisco waists time with $10 service contracts. At that point just make the damn thing free. Anyway,

Hi everyone, THis is my second thread regarding the issue.(before I was having problems with accessing my email, which slow down my responses, sorry for that) My setup looks like this Firewall | | Asterisk---Asterisk (two asterisk servers with the same setup for high avail) | | phones Ports 5060, 10000-20000, 4569, 5036 are forwared to 192.168.1.251 which is virtual ip address on one of the

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, I have made a GUI in PerlTk and I would use R within perl. I actually try to install RSPerl but I have some problems during the compilation In order to call R from perl I use actually an other system based on the OPEN command in perl. It works fine but when I tried to call the fonction plot() of R from perl the plot do not appears and a file named plot.ps is generated. Unfortunalelly, my