[asterisk-users] define extension to send calls to gatekeeper

hello every one, i have an asterisk system and want to act as gateway and send calls to cisco gatekeeper. this is my h323.conf file: [general] port=1720 binaddr=192.168.0.YY context=from-trunk faststart=yes h245tunneling=yes gatekeeper=192.168.0.XX //cisco address progress_setup=8 progress_alert=

Re: [asterisk-users] asterisk fax in debian

Enable logging and see what happens when you start and stop the iaxmodem. Obviously, it doesn't register, but the messages might be helpful. Since iaxmodem is somewhat older, you might have to disallow call tokens in iax.conf. jg -- _

Re: [asterisk-users] Issue dialing out

>> Do a "pri set debug" (or whatever it is called in 1.4 (zap?)) Zap/Zap >> bridging should work, it did on my PRIs and still does with DAHDI. Only >> thing I can think of is the TON/NPI might be a problem (but doubt it >> since SIP/Zap works). >> >> > > Thanks so much for your suggestions. > > I'm

Re: [asterisk-users] asterisk fax in debian

I am having an issue with iax2 for the fax. when i am issuing iax2 show peers, i am getting *CLI> iax2 show peers Name/UsernameHost Mask Port Status Description iaxmodem/iaxmod (null) (D) 255.255.255.255 0 Unmonitored 1 iax2 peers [0 onl

Re: [asterisk-users] Issue dialing out

On Sat, Jun 15, 2013 at 4:03 PM, Daniel Tryba wrote: > On Sat, Jun 15, 2013 at 03:02:41PM -0400, Andre Goree wrote: >> Setting the CID did not work, unfortunately :( > [...] >> I'm going to try another number that we have through them in hopes >> that it'll complete and I'll let you know if that w

Re: [asterisk-users] Issue dialing out

On Sat, Jun 15, 2013 at 03:02:41PM -0400, Andre Goree wrote: > Setting the CID did not work, unfortunately :( [...] > I'm going to try another number that we have through them in hopes > that it'll complete and I'll let you know if that works. Do you have > any other suggestions on what you think

Re: [asterisk-users] I need a second opinion on a new phone system deployment

On Sat, Jun 15, 2013 at 10:28:50AM -0600, Nunya Biznatch wrote: > Answer - There's a couple reasons I'm thinking this way, which may be > misguided so thanks for making me think about it. First is redundancy. > Offloading the PRIs and analog phones from the primary PBX means if > there's an issue,

Re: [asterisk-users] SIGTRAN Integration

What about projects like YATE, DiaStar, and mobicents (even though I have no idea how to approach that project in terms of downloading etc..). Are there any mature SS7/SIGTRAN stacks? Kind Regards, Nick. -- _ -- Bandwidth and Co

Re: [asterisk-users] Issue dialing out

>> They are either incompetant or lying to you. Call appears to succeed (it >> is answered) and gets disconnected after 23s. You are not generating the >> message, so the calls gets back to you telco. >> >> Most likely someone is filtering on callerID (which is a good thing IMHO). >> Set the caller

Re: [asterisk-users] Issue dialing out

On Sat, Jun 15, 2013 at 2:18 PM, Daniel Tryba wrote: >> Jun 15 13:06:05 VERBOSE[30232]: -- Executing Dial("Zap/1-1", >> "zap/g1/1XX|20|tT") in new stack >> Jun 15 13:06:05 VERBOSE[30232]: -- Called g1/1XX >> Jun 15 13:06:08 VERBOSE[30232]: -- Zap/2-1 answered Zap/1-1 >> Jun 15 13:0

Re: [asterisk-users] Issue dialing out

> Jun 15 13:06:05 VERBOSE[30232]: -- Executing Dial("Zap/1-1", > "zap/g1/1XX|20|tT") in new stack > Jun 15 13:06:05 VERBOSE[30232]: -- Called g1/1XX > Jun 15 13:06:08 VERBOSE[30232]: -- Zap/2-1 answered Zap/1-1 > Jun 15 13:06:08 VERBOSE[30232]: -- Attempting native bridge of Zap/1-1

Re: [asterisk-users] I need a second opinion on a new phone system deployment

On 15/6/13 7:00 pm, Carlos Alvarez wrote: Interesting product that I was very interested in, but the licensing has one huge glaring problem. Be sure to read the FAQ carefully. If your hardware fails and you replace almost anything in the machine, you have to pay for the product again. Not to

Re: [asterisk-users] I need a second opinion on a new phone system deployment

Interesting product that I was very interested in, but the licensing has one huge glaring problem. Be sure to read the FAQ carefully. If your hardware fails and you replace almost anything in the machine, you have to pay for the product again. On Sat, Jun 15, 2013 at 10:42 AM, Michelle Dupuis

Re: [asterisk-users] I need a second opinion on a new phone system deployment

... For redundant/failover of Asterisk checkout HAAST at www.generationd.com The HAAST product sits between Linux and Asterisk, monitors for failures etc, and then fails over to another Asterisk box. It effectively creates a low-cost cluster, moving IP's etc to ac

[asterisk-users] Issue dialing out

Hello all. I'm having trouble resolving an issue with our Asterisk system that seems to have popped up recently (no one knows for sure when the issue started). I'm still somewhat of a Asterisk newb and have been tasked with administrating the system as the previous administrator has left the compa

Re: [asterisk-users] I need a second opinion on a new phone system deployment

Thanks to everyone for the responses. I really appreciate it. I'll answer all questions and suggestions in this one email. (at the bottom) On 6/14/2013 9:43 AM, Nunya Biznatch wrote: Howdy All, They say opinions are like belly buttons, everybody has one. (that's the "clean" version of the

Re: [asterisk-users] Asterisk sends the INTERNAL IP address of my equipments to my SIP friends?!?

Andreas Sikkema wrote: > > On 6/13/13 16:20 , Matthew J. Roth wrote: > > It's hard to be certain without seeing a full SIP trace, but I think the > > INVITE > > with the internal IP is actually a re-INVITE that Asterisk is sending to > > establish a native bridge between the SIP friend and the SIP

Re: [asterisk-users] Asterisk sends the INTERNAL IP address of my equipments to my SIP friends?!?

On 6/13/13 16:20 , Matthew J. Roth wrote: > It's hard to be certain without seeing a full SIP trace, but I think the > INVITE > with the internal IP is actually a re-INVITE that Asterisk is sending to > establish a native bridge between the SIP friend and the SIP gateway to PSTN > converter. It