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How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Thanx, Daniel Arohuanca Lagos +51 1 3594122 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080722/f9612f97/attachment.htm

...----- > An HTML attachment was scrubbed... > URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081121/8392150e/attachment-0001.htm > > ------------------------------ > > Message: 14 > Date: Fri, 21 Nov 2008 10:04:57 -0500 > From: "Noah Miller" <noahisaacmiller at gmail.com> > Subject: Re: [asterisk-users] Limit the number of users in a meetme > conference? > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: > <8699dcab0811210704w2...

Hi All - I've been trying to compile Zaptel w/ HPEC, but I've been unsuccessful. The system is CentOS 4.4, zaptel version 1.2.15. I believe I've got all the requisite files, and they're in the right locations in the zaptel tree. When I compile, I get the following warning from make: Warning: could not find /usr/src/zaptel-1.2.15/hpec/.hpec_x86_32.o.cmd for

Hi - Does anybody know if an FXS generates line voltage when Dahdi/Zaptel is disabled? Thanks, Noah

Hi - Just wondering if anyone has gotten a Polycom VVX phone to successfully do an Auto Answer with asterisk. I have an older generation of Polycom phones that do this just fine, but I can't seem to make the VVX phones work. I tried the guide here: http://community.polycom.com/t5/VoIP/FAQ-How-can-I-change-my-Ringtone-or-Ring-in-a-special-manner-for/td-p/5167 And I have this in my diaplan:

Hi I have the following setup to make outgoing calls: X-Lite (build 34025) at home behind NAT -> Internet -> Asterisk at work behind NAT -> Internet -> VoIP provider -> GSM gateway -> cellphone. I just tried calling my own cellphone, but there is no sound either way. Here's what I did on the X-Lite at home in the Topology section: IP address : Discover global address

Hi - We've got a number of offices, and they're all using ODBC message storage using MySQL. I've been trying to get MySQL replication set up so messages left in a voicemail box at one office will get copied to the corresponding voicemail box at all the offices. We're also using MySQL replication for the voicemail user info, and that part works just fine. I'd like to

After an upgrade from asterisk 1.2.13 to 1.2.25 including upgrading libpri, zaptel, the incoming calls to a TDM400P REV I, with 3 FXO modules stop working. The board is working, I tested in another server with the 1.2.13 asterisk version. When a call is incoming, I do a ztmonitor to check the rx and tx values, but nothing appears on screen. Also changed the pci slot where the board is. The

Hi Thiru - > Could u tell me ,how to connect 2 asterisk server using sip as a > clients... > asterisk server are in same network... You can connect them either as "friends" or as "users/peers". I generally recommend the user/peer method for connecting two servers since it clearly delineates which codecs and contexts are allowed. Your sip.conf files will look

Dear all, I have small question in sip.conf I added [service] type=friend ;username= ;secret= qualify=900 host=X.X.X.X dtmfmode = rfc2833 disallow=all ;allow=g729 allow=gsm allow=alaw allow=ulaw and I can proccess incoming call from soft phone only I calling on number that is used in extensions.conf(in example below it is 1) exten => 1,1,Answer; exten => 1,2,Playback(hello-world,skip);

Hello, I would like to double check what Echo Cancellation my Digium Card uses. I thought I bought the little more expensive card that integrates EchoCancellation. How can I check? root at sn1:~# zaptel_hardware pci:0000:0b:08.0 wcte12xp+ d161:8000 Wildcard TE121 root at sn1:~# ztcfg -v Zaptel Version: SVN-branch-1.4-r4309 Echo Canceller: MG2 Is MG2 the correct one that I am supposed

Hi, This is Chandra. I have Polycom IP 501 phone. Its showing wrong time on the display screen. How can I set the "New York" time? What value I have to give to GMT offset value? Look forward to your response. Thank you. Regards, Chandra. --------------------------------- Ahhh...imagining that irresistible "new car" smell? Check outnew cars at Yahoo! Autos.

Does anyone know if there is a known list of telemarketers? Something like http://whocalled.us/ with an easier access? We could all benefit if there was such a thing :-) If there is enough interest, I could put up a database that everyone can benefit from. I just need some suggestions on: (1) Adding new numbers based on community responses (some rule to sanity check) (2) Method that everyone

Hi I'm a little lost on how to set things up with the two FXO cards I have: I want card #2 to dial a number when a call comes in on card #1. Using the following configuration, card #1 picks up the line and remains silent, instead of dialing out through card #2. Anybody knows what's wrong? --------- /etc/zaptel.conf --------- # Zaptel Configuration File # fxsks=1,2 loadzone=fr

Hi, Could anyone tell me how to read the values in the "zonedata.c" file? I am looking at the "zt_tone_ringtone" field mainly. Thank you. Jad Wauthier -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070512/4c0387be/attachment.htm

We are doing this with the latest spandsp, iaxmodem and hylafax. Seems to work very well for us so far. -Jonathan > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Michael Gaudette > Sent: Tuesday, March 21, 2006 3:34 PM > To: 'Asterisk Users Mailing List - Non-Commercial

Hi All - I'm looking for a Wifi SIP phone that can do multiple line appearances. It seems the Spectralink Netlink e340 can do multiple lines, though I can't figure out how many. Does anybody know of any others that can do at least two line appearances? Thanks, Noah

Hi All - I haven't worked much with ZAP handsets before, but I've got a client who is insistent on using a particular phone. My questions: 1. With multiline analog phones, if I've got multiple phones, each connected to a different FXS interface, is there a way to make the line status lights on the other phones show that a particular FXO is in use (like a key system, or like SIP

Hi Steve - Sorry for the dupe, but since this is now way off-thread, I thought I'd create a new one (and correct my spelling mistake). > Just my personal experience, but I do not find IAX to be very reliable. > Is there any particular reason you are not using SIP? I'm curious as to your negative experiences with IAX. I generally use it for multi-office installations, and have had

Hi All - Anyone know if the "callpark" feature is in ABE? Is there a comprehensive list of the differences between ABE and the open source version? I've only seen a bullet-point chart which has no real detail. Thanks, Noah