similar to: Fax support?

I can't seem to make the fax detection work. Here's an excerpt from zapata.conf: signalling=fxs_ks group=0 context => guestaccess channel => 47-48 and from extensions.conf: [guestaccess] include => incomingmain [incomingmain] exten => s,1,Dial,Zap/1&Zap/9&Zap/10&Zap/11|24 exten => s,2,Voicemail,u7000 exten =>

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hey, I bought a license for 729 but I can't use it this is the message. == Registered translator 'g729tolinb' from format 8 to 6, cost 99999 == Registered translator 'lintog729b' from format 6 to 8, cost 18 == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI> WARNING[5126]: File chan_sip.c, Line 1601 (process_sdp): No compatible codecs!

Hi, I have some questions regarding IAX, IAX2 and encrypted authentication. How can I know if IAX or IAX2 is used between two * servers? There is any guide about how to configure encrypted authentication (not in clear text)between two * servers? I "hear" on this list a couple of days ago that port 5036 is the default one for IAX and something else (4XXX) for IAX2. Trying 'iax

I'm just curious if anyone has the X100P & Callerid receiving working outside US. Replies are appreciated. Also if it's not working for you in a certain coutry you can respond too. regards Martin

Martin Pycko <martinp@digium.com> writes: > sure, use the 'n' option of the queue and put voicemail app as the next > priority Will that work? From my read of the code, the timeout parameter is only checked while the call is being sent to an agent's phone (inside the try_calling function). The timeout doesn't seem to be checked while the user is waiting to get to

Hi, Does Asterisk support SS7? Google shows an old new post from Feb. 2002 stating that OpenSS7 would help add SS7 support to Asterisk, but presently OpenSS7 seems to be dead and I can't seem to find anything about it at Asterisk or Digium's sites. What happened? -- Regards, Tais M. Hansen ComX

<P>hi Everyone,</P> <P>We are configuring an ISDN PRI E1 with an E100P card, when you load the drivers, and starts the asterisk, cards also starts fine, givin following output,</P> <P>*CLI> <BR>&nbsp; == D-Channel on span 1 up<BR>&nbsp;&nbsp;&nbsp; -- B-channel 1 successfully restarted on span 1<BR>&nbsp;&nbsp;&nbsp; --

Hello, Is there any configuration in zapata.conf for fax detection (or transmission)? When I try to send a fax trought asterisk, the line 'Fax Handled:' is always set to "no". The scenario is: [ata186]---sip---[asterisk]---e1 E&M---[pstn] Fax sometimes goes without problem and sometimes the fax machine can't send the fax. Thanks Eduardo

My * server is connected with a SIP proxy, which handles different domains. So to dial a number in the SIP proxy domain from a phone on *, I use Dial(SIP/1234@domain1.com). The problem is that the Dial convert domain1.com to IP address, which causes the SIP proxy returns a 404. I'm wondering if host name can be preserved at least in the request-uri (To header can be optional.) Thanks, Howard

Does * support ISDN Primary Rate Interface (PRI) - 2B Transfer Capability for T-1/PRI? In other words the ability to take a call and join it to another call and then drop off letting the CO-switch take over. -Kevin Kevin Fjelsted, President AltiCom CTI, Inc. Track Me Down! One number Access, Press 11# during the voice mail message greeting to have me F-O-U-N-D! Phone: 612.259.0722 Fax:

Hi, When I try to run the command wmix to mix two WAV files recorded by the Monitor application I get the following warning in the console and the macro exit at that point. Running the command from a standard system console it works. More, even from this macro it works and produce a valid mixed file, but still get that error and the macro cannot continue. Why? I have tried even with a simple

Hi all! I'm newbie, so here goes my situation: I have succefully compiled the cvs version as shown in asterisk website in some linux distros: Debian (2.4.22), Conectiva, Fedora Core 1 and in all of them, * starts and consumes all the cpu (on top). Does anybody know this issue? Thanks! Testa

In all the information on Asterisk it takes about TDMoE to link asterisk servers together. Is this IAX??? How would I use TDMoE. Maybe my first question should be, What is it??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030516/cd74bddb/attachment.htm

I have a client that would like to use asterisk to link their multiples locations together. However, if a person in the remote office dials 911, How can the 911 operator determine WHERE the emergency is?? Since all calss would be going out of the PRI in the main location, the police/fire trucks will show up at our COLO!! I know that there are some that are doing this multi site setup, how did

Problem I have is this. outside firewall (extension 2003) can call me inside firewall (extension 2000) and all is fine. If I call from inside firewall (extension 2000) to outside firewall (extension 2003) I hear no ringing and person at other end can pick up and I hear for maybe a half second then I go to voicemail. If I add another extension on the outside then communication between outside

incominglimit is already implemented for SIP. Just specify under the endpoint how many incoming connections are allowed. For example, [cisco] type=friend username=cisco secret=blah nat=yes ; This phone may be natted host=dynamic canreinvite=no ; Cisco poops on reinvite sometimes qualify=200 ; Qualify peer is no more than 200ms away

Hi list, Sometimes I receive a Red Alarm in my E1 trunk (E&M immediate start signaling), and just few seconds after this, all alarms are cleared. This problem ocurrs many times/day, and if are calls in progress, these calls just hang-up. Could it be an asterisk bug? Or may I contact the PSTN provider? Thanks Eduardo

Hello, I am not entirely sure the subject line captures what I am trying to do, but hopefully this description of the problem will help folks to see my challenge and hopefully offer constructive assistance. I have an experimental setup where I measure the decrease in oxygen in small vials as an organism, such as an oyster, consumes the oxygen. Each vial is calibrated before the experiment and

I have been playing around with asterisk for a week or so now and haven't had too much trouble getting things to work but one thing seems to puzzle me. I have been patient hoping that there was a configuration error on the server or that the toll-free gateway was down but nothing has changed. I have the following configuration context for IAXTEL: [iaxtel] exten =>