similar to: Number Portability Details

Hi folks, I'm running Asterisk 1.4.18 on Mac OS X. I'm using Broadvoice to connect external numbers with this system. So, what I'm dealing with here are inbound calls. The problem is that I'm not able to get DTMF in. I am behind NAT, but I've made sure that the IT folks here have opened and port-forwarded all the necessary ports. This is borne out by the fact that I

This configuration solved my problem. I could have sworn I tried this before. I guess not. I did not need to apply the patch. Also, I am using a regular Registration setup in my sip.conf not broadvoice's funky one... The only thing I can surmise is that order of the variables matters. This is what worked for me: [PPPPPPPPPP] type=peer user=phone host=sip.broadvoice.com

As a general rule of thumb it would be good to make the distinction between 'Credit Card' and 'Debit Card' too. If possible, never ever use a debit card for online purchases. It taps directly into your account and removes REAL money. Credit cards are 'virtual' money in that they are credit and your cash in account remains untouched. That way if your credit card account

I'd be grateful if someone could point me in the right direction. I have a Broadvoice trunk attached to Asterisk which I use for frequent calls to the UK using the following in extensions.conf exten => _0[1-68].,1,Ringing exten => _0[1-68].,2,Dial(SIP/BV/01144${EXTEN:1}) exten => _0[1-68].,3,Hangup The caller hears immediate ringing, though it seems that Broadvoice

Like everyone else who used asterisk with broadvoice, my outgoing calls died last week. I made the appropriate changes, and now basic incoming and outgoing calls are working. However, I have a few call-forwarding rules that are no longer working. It's certainly no coincidence. I can dial to all these number directly, but the problem only appears when there is an incoming broadvoice call, and I

I seem to be getting a connection from telasip but instead of dialing my default extension, nothing happens. I listen to dead air. I have a fxo card configured and working on both inbound and outbound calls. Telasip is working outbound. I put in the recommended (by telasip) changes to the trunk for incoming, e.g. host=gw4.telasip.com insecure=very qualify=yes type=user context=from-pstn Then

With the recent service outage at Broadvoice, there has been a lot of discussion here, on broadband reports, Voxilla, etc., regarding whether VOIP is mature, or "ready for the masses", etc. One particular point I've seen repeated, and with which I agree: "we're willing to deal with less than five 9s, even one or 2 9s, as long as we have good communication regarding the

I have installed asterisk @ home 2.6. I am using a Telasip VOIP account. When I make outbound or inbound calls the calls seem to connect and then get hung up. I was wondering if there was something that I am misisng. I have tried several different sip.conf configurations. Here is what they are currently. telasip-gw canreinvite=yes context=telasip-in dtmfmode=rfc2833 fromuser=jrasxxx

I've had the same excellent responsiveness from telasip, on the rare occasion that I've had issues. YMMV -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tom Vile Sent: Thursday, April 06, 2006 6:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Telasip

RE Message: 5 Date: Sat, 7 May 2005 23:18:46 -0400 From: Andrew Kohlsmith akohlsmith-asterisk@benshaw.com Subject: Re: [Asterisk-Users] At home Asterisk via Broadvoice? To: asterisk-users@lists.digium.com On May 7, 2005 11:04 pm, John Stegenga wrote: > Broadvoice will give me 2 lines, with 2 phone numbers each - distinctive > ring - for a reasonable fee... Please do a google search for

Hi all, I've just setup a sip line with Telasip and when they route the calls to my asterisk box, they include an extension along with the context that is defined in sip.conf for that DID. At first, I couldn't figure why they were getting 404 error from my asterisk box, but then figured out that they are sending the call to an extension that matches my number with them, in the

II noticed that Caller ID presentation is not making it to my cell phone through outound Telasip calls and I don't know why. It may very well have been this way for awhile (or always, not sure I called my cell phone during telasip testing). Does Telasip expect a different format than SetCIDNum(NXXNXXXXXX) ? It has always worked for the Teliax lines. BUT--- It doesn't have a problem

Hello All, I am trying broadvoice's europe plus calling plan for unlimited to Poland. My first attempts though, were not that good. I could hear the other side, but they could not clearly hear me. Is this because broadvoice's connection just is not up to par? Has anyone else been using this plan to europe and had success? Regards, Greg

Has anyone been watching and logging when broadvoice becomes unstable? Is it only peak hours, or is it random? If its somewhat consistant, I'd like to enforce some time of day routing in my dialplan. Otherwise I may just close the account altogether. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Watch the output of 'pri debug span 1' on the Asterisk server while placing the call - bug #4468 (http://bugs.digium.com/view.php?id=4468) might be relevant. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Andy Goss > Sent: Monday, October 10, 2005 5:58 AM > To: Asterisk Users

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 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 got hit with a funny one today; my configuration, which had been running fine for many weeks, suddenly stopped registering with Broadvoice's SIP. I switched among BV SIP hosts, verified my account was OK, even tried upgrading Asterisk. The weird thing was that I could associate OK with my WiFi SIP phone. So it looked like an Asterisk- specific SIP interoperability

I have a SIP trunk to Broadvoice. My Asterisk box (1.4.13) is on public addresses behind a firewall. Originally it was behind a Linksys WRT54G running sveasoft. Sveasoft really can't NOT do NAT even when you turn it off. My Asterisk box is defined as the DMZ box to Sveasoft and it seemed like it was leaving all packets alone. Now I switch to a Centos-based firewall configured with

I was having problems getting a Linksys PAP2T-NA to work with Pitney Bowes mailing station so it could use its modem to dial home and download postage/software updates. After scowering the web, I couldn't seem to find a definite how to article on what settings were needed. I finally came up some settings by combining the information from various places around the 'net. I have typed out