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...Zap/1&Zap/9&Zap/10&Zap/11|24 exten => s,2,Voicemail,u7000 exten => s,3,Hangup exten => fax,1,Dial,Zap/3 When I dial in, Asterisk simply rings Zap channels 1, 9, 10, and 11, whether it's a fax or non-fax call. What am I missing? Thanks, D. --- Martin Pycko <martinp at digium.com> wrote: > From: Martin Pycko <martinp at digium.com> > To: <asterisk-users at lists.digium.com> > Subject: Re: [Asterisk-Users] Fax support? > Date: Mon, 3 Mar 2003 11:41:48 -0600 (CST) > > let's say you have one T1 span configured like this >...

Is there any way to receive and send faxes using a T100 card? If so how is it done? Gene Kochanowsky Solution Sciences, Inc.

...o run IE on the Windows partition (instead of in fake_windows). When I get home to my Linux machine I will try setting the connection to a LAN connection. (Do I just go into Tools | Internet options and select it there?) Thanks again, Martin Polley Technical Communicator http://www.surf-com.com/ martinp@surf-com.com Tel: (+972) (4) 9095-732 Mobile: (053) 864-280 ICQ 15617901 -----Original Message----- From: Thomas Wickline [mailto:twickline2@triad.rr.com] Sent: Tuesday, September 10, 2002 12:18 PM To: Martin Polley Subject: Re: (no subject) Martin Polley wrote: >Hi, > >I am trying...

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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!

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 t...

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

...how do I do this? Which key in which registry file do I need to change? Will this solve the problem? (I tried via Tools | Internet Options (optimistically), but I got a bunch of "not yet implemented" error messages...) TIA, Martin Polley Technical Communicator http://www.surf-com.com/ martinp@surf-com.com Tel: (+972) (4) 9095-732 Mobile: (053) 864-280 ICQ 15617901 -----Original Message----- From: Martin Polley Sent: Tuesday, September 10, 2002 1:54 PM To: twickline2@triad.rr.com Cc: wine-users@winehq.com Subject: RE: IE5.5 (was:(no subject)) Thanks for the reply! I tried the ins...

Any ideas on the following? (CVS 10/01/2003) Only reference I could find was a Zaptel change log update... 2003-09-02 18:23 martinp * wct4xxp.c (1.6): Get rid of the Double missed interrupt message every time you load the driver and an email refering this to serial console usage. Something I should worry about? Oct 3 22:48:01 cti-350 kernel: TE410P: Double/missed interrupt detected Oct 3 22:49:01 cti-350 kernel: TE410P: Doub...

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

...e line in the > general > section which is > > TRUNK=SIP/??????? > > Using this method would be easier. > > How do you tell asterisk how many lines are available at the gateway > > > Dave > ----- Original Message ----- > From: "Martin Pycko" <martinp@digium.com> > To: <asterisk-users@lists.digium.com> > Sent: Thursday, August 07, 2003 12:34 PM > Subject: Re: [Asterisk-Users] Sip Trunk config > > > > exten => _9X.,1,Dial,SIP/${EXTEN:1}@ip_of_the_gateway > > > > regards > > Martin > > &gt...

...en the T1 would need to be unframed. I don't believe this is an option in zaptel! If however, it is putting bits on at a rate of 1.536 Mbit/s and adding 8000 bit/s for framing then you may be able use the suggestion below. Don Pobanz On Thursday, April 03, 2003 3:28 PM, Martin Pycko [SMTP:martinp@digium.com] wrote: > You could configure the channels of port 1 as > clear=1-24 > in zaptel.conf > > This way you'll have one big pipe on /dev/zap/1 accessible for you. > Unfortunatelly you cannot do clear=1-96 and have it all 4 spans > on /dev/zap/1. > > regards >...

I don't know if 911 uses caller ID or BTN (Billing Telephone Number) 900 calls, operator calls, and 800 calls use the BTN not the Caller ID... Anyone???? 3. Re: E911 and asterisk (Martin Pycko) Message: 3 Date: Mon, 21 Jul 2003 12:05:38 -0500 (CDT) From: Martin Pycko <martinp@digium.com> To: <asterisk-users@lists.digium.com> Subject: Re: [Asterisk-Users] E911 and asterisk Reply-To: asterisk-users@lists.digium.com Isn't that enough to set up a proper Caller ID NAME ? Martin On Mon, 21 Jul 2003, Alex Lopez wrote: > I have a client that would like to us...

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

...ter. I'll be trying the other win app thats up-and-coming on the list later. It seems to have broken iptel, but that's not as important to me right now. Perhaps there could be some flag on the register line to turn the externip on or off. -- Andrew Thompson Quoting Martin Pycko <martinp@digium.com>: > It doesn't care about the phones. If you phones are behind nat use nat=yes > for each defined account. > > Martin > > On Tue, 4 Nov 2003, Shoval Tom wrote: > > > Will extern IP work if I had multiple phones connected behind NAT? > > > &g...

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

Hi, I'm having random hangup problems with zap channels. If I turn busydetect off in Zapata.conf, * fails completely to detect a user hangup in the middle of a script. On the other hand, if I turn it on, everything works much better, but long calls tend to be hung up without a motive. Any other parameter that I can try? Any #define that I can tweak and recompile? Will callprogress

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