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I guess what I meant to ask was for a way to do it from within extensions.conf. Using either the Dial command or if there is another method to do the three way calling. >Press flash on your phone (asterisk will intercept that) and then when you >have a dialtone press *0 then asterisk will send the flash to PSTN line. > >regards >Martin > >On Thu, 28 Aug 2003, Carlton J.

...s,1,Dial,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 lik...

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

...sounds better. 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? >...

Thanks for all your help Martin, Guys, This is a good find and hopefully could help someone else. I've been having a problem with forwarding voicemail from one mailbox to another. I ran down the sendmail and soundcard path and came up goose eggs. With intuitive guidance from Martin Pycko (Digium), I switched from Redhat 9 Kernel linux-2.4.20-8 to Redhat 8 Kernel linux-2.4.18-14 and it seemed to solve the problem I was having. There is still a little weirdness going on but the voicemail forward command is working. During a -vvvvdgc session, I get: Urgent handler -- Playing ...

I've tried the precompiled version of gastman, but it does'n work properly under windows, so I would like to try to compile it under windows, so maybe it begins to work. I've tried to compile gastman with mingw, but the db3.1 libraries request the cygwin include files and libraries. Then I've copied the cygwin's include files and libraries to mingw's directory, but the make

...1. Re: CE certification for Europe (d hinton) > 2. Re: MP3player problem (Michael Bielicki) > 3. Re: FAX over IAX (Steve Underwood) > 4. Re: CE certification for Europe (Mark Spencer) > 5. Re: CE certification for Europe (Steve Underwood) > 6. Re: MP3player problem (Martin Pycko) > 7. Re: MP3player problem (Martin Pycko) > 8. Re: FAX over IAX (Martin Pycko) > 9. Re: "Broken Dial Tone" for voice mail? (Steven Critchfield) > 10. Re: CE certification for Europe (d hinton) > 11. RE: FAX over IAX (Brian Jones) > >--__--__-- > >Messag...

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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!

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 ? Ma...

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 funct...

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 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

Hi I want to set up * as a conference bridge. I would like to be able to conference is SIP calls (up to 12) I am looking through all available documentation for * to get info on how it is done. No luck so far. Can somebody direct me to the info in this subject? Thank you Serge _________________________________________________________________ Protect your PC - get McAfee.com VirusScan Online

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

<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; --

Does anyone know what this means? It is in DMESG, and we have people complaining about echo. Disabled echo canceller because of tone (rx) John

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, After upgrading the kernel on an Asterisk box, asterisk segfaults on startup. It seems like it's the g729 codec that causes this: #0 0x4015acad in memset () from /lib/libc.so.6 #1 0x4022686a in load_module () at codec_g729b.c:416 #2 0x08054794 in ast_load_resource (resource_name=0x80d1068 "codec_g729b.so") at loader.c:298 #3

...particularly the 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 > > Ma...

Is it possible to support 32 or 64 E1 in a linux box with Wildcard E400P board ? I'd like to make large scale PPS system. Thanks in advance. See Young Oh. DIGITALWAVE,INC. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20030304/c1f8d74f/attachment.html>