similar to: Calling to Broadvoice via Linux MASQ (NAT)

Hi, I am pretty new to all of this but was able to set up an asterisk server and have been able to succesfully connect to asterisk with x-lite as sip client. I have also connected asterisk to FWD (using iax2) and to voipjet (also using iax2). Now I am trying to connect asterisk to Stanaphone. It has to register as a SIP client but I am not being succesful at all. My asterisk server sits behind a

Outward dialing is a no brainer. VoipJet is the best outbound call provider I have come across. Period. It always works for me and the call quality is always very very good. So far that seems to be the norm for them. I am still working on getting my inward DIDs solidified so no opinion there... W -----Original Message----- From: asterisk-users-bounces@lists.digium.com

I can receive incoming calls. I can see via sip debug that I am communicating with BV just fine. When I call a BV number it goes through just fine. When I call any other # I get: "We're sorry your call can not be completed at this time. Please hang up and try your call again later" Broadvoice tech support does not see any errors and they see that I am registered just fine. I have

This may be a newbie SIP/NAT question. If so I am sorry. But any help would be appreciated. My Asterisk server is behind an ipchains box and I am trying to connect to Broadvoice. All works fine without the NAT. I have a global nat=yes prior to my register, but the sip debug allows shows "no nat)". Is this "label" issue, and am I barking up the wrong tree? Sip.conf....

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

After months of BroadVoice ignoring my trouble tickets for dropped calls, delayed termination, etc., I'm throwing in the towel. While they have credited $19.95 to my account, they refuse to credit anything more, despite ALL of the problems I've had. I feel the least they could do is credit the remaining $8.61 to my account, yet they won't. I haven't really been following up on

Hello, on my old system I''m using ipchains. Can anyone help me with converting rule /sbin/ipchains -A forward -j MASQ -s source_addr -d destination_addr 443 -p tcp to shorewall. I know that I can write eth0 source_addr to /etc/shorewall/masq file but I can''t found where I can specify the destination address. The reason for this is to allow one user (computer) access only to

Wondering about being able to see MASQ activities with IPTABLES. With IPCHAINS I used -M -L to make this possible. Nothing like that with iptables, at least as far as I can see. Any thoughts on that fellows.. --- Ted Gervais Coldbrook Nova Scotia Canada B4R1A7

After two weeks of reading about asterisk and joining this mailing list, I finally decided jumping on the asterisk bandwagon... Asterisk rocks!!! I have a www.Stanaphone.com SIP (free) for incoming line and a www.VOIPJET.com IAX line for outbound. I also have a www.Vonage.com line (gives me 500 outbound minutes) and a Cingular cell phone (gives me 800 minutes) and I also use Skype fairly

Does anyone have any useful contact information for Broadvoice? I've been unable to call Argentina from my Broadvoice line for almost a month now. Their techs keep telling me they're "looking into it" but I get no real information. I've had several other tickets open for almost 8 months now. I called their PR company this morning and apparently they dropped Broadvoice

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

Now that I have most everything actually working I've noticed that about every 3-4 days on average..... and at worse... Once a day my asterisk box seems to lose it's registered state with our sip provider and no longer will take any incoming calls. The caller simply hears a fast busy (reorder) If I do a reload at the command prompt all is well for another few days..... What I'm

Hello All, Right now I have several providers. Voipjet, Teliax, and more recently Broadvoice. Broadvoice gives me unlimited to europe, but what I want to do is determine the best way to setup a dialplan so for example, certain countries will go through the cheapest route. I am really only interested in Poland, Russia and Turkey. Poland is free on broadvoice, but not for cellular, which I may

> 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. Yes, this is exactly what is happening. Thanks a lot. I am thinking about adding a special case for the IBM 800 number since it is the only one my company is complaining about. Currently I have this in my dialplan:

Hi, i have a following problem. Looks like this ROUTER --------------DEF.GWa aaa.aaa.aaa.aaa | DEF.GWb bbb.bbb.bbb.bbb | ---------------- -------------------

Thanks for the reply. Forgive me for being na?ve, however have jumped in to this asterisk project at work due to some circumstances beyond my control and I don't know a lot about carriers and how this all works. I am figuring it out, but it's a lot of trial by fire. As far as I know, we only use 1 carrier for our system. We have a PRI from NuVox and we use 7 channels for our asterisk

Dear Tinc Experts, I have been struggling for some time now, with Tinc pre3, and firewall rulesets and routing. I did once manage to get Tinc to work okay in a test-bed environment. I then tried to set it up for a 'real-life' setup and cannot get it to work properly. My real-life setup looks like this: Network A: 192.168.1.0 / 255.255.255.0 192.168.1.7 tap1 device gateway >

Hi, i'm having problems with broadvoice incoming calls. I can perfectly place calls but my Asterisk Box is having problems when registering with the SIP Proxy. Sometimes it register and the call gets into asterisk, but without sound (seems to be NAT problems) and sometimes its not possible for asterisk to receive the calls. Everything was working great exactly for a month, but a week ago it

I have searched the list and cannot find a sip 400 solution posted that solves my problem. If anyone has any thoughts or suggestions on the following I would greatly appreciate it. I didn't have this error before Broadvoice made their changes this weekend. Now when I make a call it connects but, I cannot hear anything on the other end... The full message I have is: 8 headers, 0 lines

I'm experiencing a "Failed to authenticate on INVITE" error (see output below) whenever I try to MAKE a call through the Broadvoice account. I noticed some others had the same problem but it went away when they rebuilt Asteris w/ a new version. N such luck for me! I'd be grateful for any assitance. Here's what I've done so far: 1) I downloaded the latest stable