similar to: Asterisk to sip client behind Firewall/NAT-cancall but cannot receive calls ?

Check your FW-1 tracker and see if any sip packets are dropped during call initiation. I had this problem and it went away when I upgraded the BT's firmware to the latest (16). Beware, though, that people on the list claim that this firmware breaks functionality of the message button and autoanswer. I haven't checked this yet, cause I can't afford to go back a version. I prefer a

When I made a call using an older version I saw, using checkpoint's user monitor that the call was indeed using RTP (somewhere between 10000 and 20000, dynamically set for each call). After I upgraded the firmware, the entire conversation stays on the sip port. > -----Original Message----- > From: Jon Lawrence [mailto:jon@lawrence.org.uk] > Sent: Tuesday, December 14, 2004 10:30 PM

Hi, I have following setup: BT100 ---- Firewall/nat 1 (www.ipcop.org) ---- Internet ----Firewall/nat2 (Vigor) ---- Asterisk . I'd like to use BT100 as local extension to Asterisk. I've done simple setup and BT100 can call Asterisk and place outgoing calls. However I cannot set him to qualify, cause it is claimed as unreachable. I have port redirection at Firewall 1 (to 5060 and rtp

None of the unlimited packages you get plan on you using them up. It's like a couple of years ago when the long distance carriers gave us the first phone call of the holiday for free, and people let the phone off hook for the entire week end. It's illegal for sure, and for a couple of reasons. a. you're not allow to resell anything someone else sold you and not pay taxes on it. b.

People on the list tend to think you can't make many cards work on a regular desktop. If you're willing to wait a couple of week I might have an answer for you. _____ From: Robert Augustyn [mailto:augustynr@yahoo.com] Sent: Saturday, December 11, 2004 7:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Will Adtran TSU 600 work with *?

Hi, I'd like to organize my Asterisk to properly handle incoming SIP/IAX/H323 callerids so they can be called back if needed. I have three incoming contexts for sip, iax and h323 calls. To each incoming call I'd like to prepend certain number that will be catched with pattern matching on output calls. For instance for iax I have: [from-iax] exten => s,1,NoOp(IAX call from outside

I have several Linux machines some running on really old hardware and some on brand new, some run old distros (RedHat 6) and some new (FC3 or CentOS). All of them experienced power failure more then once, none of them has failed to load after a reboot. BUT, Asterisk is running your PBX. Your PBX isn't your proxy server, it isn't your web server, mail server, firewall, or whatever

I'm going to install asterisk with four digium cards. Can anyone mention a brand that carries boards with 4 compatible pci slots? Thanks Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200

I think it should work at least for TUN virtual interface as TUn works at IP level. This is a sample configuration. firewall1 lan = 172.16.1.11/19 (ALWAYS ACTIVE) - "Physical Network Interface" - system config as ifcfg-... 172.16.1.10/19 (VIP Keepalived Make active) - Active/Passive configuration with firewall2 firewall1 vpndr1

How do you suggest doing that? How can I convert wav files to gsm files? thanks Shoval Tomer, MCSE IT Manager Softov Advanced System Ltd. Email: shoval@softov.co.il Mobile: 972-55-229220 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031102/f84d7805/attachment.htm

We've bought the G729 codec for lowering SIP bandwidth usage (we're using grandstream phones) and we're quite happy with it up until I tried using IAXPhone 0.2.0 build 116 with my asterisk 1.0.0 installations. Weirdly enough, calls from IAXphone to the GS phone work just fine. So are calls from both phones to voicemail. And from both phones to analog phones connected to FXS ports.

Is it possible to use * as a VOIP gateway? Can I connect asterisk to one of the trunks on my current PBX and on the other side of the world connect another * to the trunk of another regular PBX - is it possible to transfer calls from here to there? I guess I'll need one port FXO card for each asterisk, but I can't figure how to configure the thing. I know I'll need to

This is a vpn for Disater Recovery sites, so it is not necessary to have a seamless failover, strictly speaking. Encryption instead is mandatory. Testing we found that on Keepalived failover remote Tinc take few seconds to reset the connection and correctly re-connect to the new active firewall (probably new firewall resetting the connection + PingTimeout + some seconds to reconnect). This is

This is what I want to avoid :D I want an active Tinc virtual interface active with ip identical of the other firewall, without ip conflict on the same network. Do you know if Tun type virtual interface on one host can have same ip address of another host in the same network without ip conflict ? ie if a tun virtual interface can work active without transmitting on real network ? or if such a

Say I get an ADIT 600 with two FXS8A and one FXO8A from ebay. a. Is it good for Asterisk? b. How do I connect the extensions and lines to it? Do I need a special jack? Can I get that jack in every corner? c. where can I find help for configuring it? d. what kind of backup does it have? Does it need to be reconfigured after a power outage? Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd.

Hi Saverio, I found conflict: 172.16.1.10 00:50:56:1b:ba:5e VMware, Inc. 172.16.1.10 00:50:56:2b:12:e6 VMware, Inc. (DUP: 2) 172.16.1.10 00:50:56:2b:12:e6 VMware, Inc. (DUP: 3) 172.16.1.10 00:50:56:2b:12:e6 VMware, Inc. (DUP: 4) 172.16.1.10 00:50:56:2b:12:e6 VMware, Inc. (DUP: 5) So my assumptions were wrong ! :D Probably Virtual

Shoval, Interesting Mention. I agree, most people don't have CO exp. And I wish daily I had enough. Understand that what I mean by my e-mail is consumer side FXS ports, in broader terms, I mean, customer picks up a phone line, it signals a channel bank which signals *. 24 of those channels. Not channels equipped to Send Signal to the CO that a loop has been made.. meaning FXO. 24

Hi, I believe the issues raised by this message are the same as mine, more on a commercial sense than for self use, but mostly the same. I've seen posts where real-life installations are mentioned, but not a reference to how Asterisk is working on production (and productive) environments. Any experiences would be very welcome I believe, not only on pure technical, but wider, sense. Thanks

I see my own number(or remote called num) instead of caller id on incoming calls on my BT-102. but on Xlite everyything is OK. I'm using * latest CVS. - shabanip

As far as I know it's not legal to join bellster in Israel. It means that you're reselling the minutes you buy from the telco company. It also means that you need a permit from the Israeli ministry of communications cause you're acting as an international call provider. Can't be done here. -----Original Message----- From: Geoffrey S. Mendelson [mailto:gsm@mendelson.com] Sent: