similar to: meetme application

I finally got 2.4 recompiled with RTC as a module: Module Size Used by Not tainted autofs 13684 0 (autoclean) (unused) acenic 241092 0 (unused) iptable_filter 2412 0 (autoclean) (unused) ip_tables 15864 1 [iptable_filter] e100 62340 1 rtc 9084 0 (autoclean) Here is my

Hi there, I would like to communicate H323 IP phones with SIP phones. My H323 phones are registered to a gnugk GK, and the SIP phones are registered to a asterisk SIP proxy. I could not create a dialplan that works. Inside my extensions.conf file I created the following two entrances: exten => 4,1,Dial(SIP/4) exten => 5,1,Dial(SIP/5) This allows SIP phones call each other.

zaprtc 'load' is failing on my machine (the make was fine, same output as other posts to this list) [xxx@xxx zaptelrtc]# make load sync modprobe zaptel insmod ./zaprtc.o ./zaprtc.o: init_module: Input/output error Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or

Hi there, Here at my company we are willing to use the asterisk IVR system. The problem we are having rigth now is that all our GWs use G729. I've read that in order to asterisk be able to make transcoding from the GSM audio files to G.729, it is necesary to purchase a license from digium. Is this correct? I've seen that licenses are purchased on a per-channel basis. Could

-This is my 'sip.conf' file: ;************************************************************* ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls tos=184 maxexpirey=3600 ; Max length of incoming registration we allow

Hullo. Having found http://bugs.digium.com/bug_view_page.php?bug_id=0000875 I grabbed the original 0.0.1 and Dan's patch, and whilst it didn't apply, I was able to patch the zaprtc.c manually - the Makefile has changed a lot, and I wasn't able to understand the changes. (this is all on a machine that's never had any * on it before, and has a 2.6.5 kernel with a matching

Hi there, since I found a couple of reports with complaints concerning zaprtc I thought that one or the other user might be glad to know that it works indeed. All that was necessary was to copy all *.h files from /zaptel into /zaptelrtc and then do "make" followed by "make load". Of course you'll need to make sure to not have rtc compiled into your kernel, or loaded

Any thoughts on the following? I am running asterisk from CVS (downloaded yesterday's version, just to be sure) on a test system with no digium cards in it, so I have installed ztdummy (see logs and screenshots below) as a timing source. When I call the music on hold extension from a Sipura Sip connected analog phone, I hear nothing and start getting Warning[98310]: chan_sip.c:674

Hi, we are setting a 120-channel conference server and would like to learn if someone already did this (hardware, problems, etc...) Best regards, PauloHM

Yes, I am a newbie too. I am having a problem with meetme. From what I have seen it will work without a Digium card but with audio problems. My goal is just to see how it works not the quality of the audio. When I dial into the conference room the following message is played: "That is not a valid conference number." On the console I get: "unable to open pseudo channel". As

I am configuring an asterisk server and I want to test the incoming configuration with my FXS handsets. I have the FXS lines able to call eachother and they can connect out the FXO lines. I changed the context for the FXS lines to "incoming" so that they would be able to test the setup for incoming calls. For the incoming context I have: [incoming] exten => s,1,Wait(1) exten

Hello, i would like to use Meetme and i need zaptelrtc for that, since i dont have any USB devices or a card from digium. I compiled it on Llinux 2.6.8 and all i got was a zaprtc.o which obviously wont work with a 2.6 kernel: ~/zaptelrtc# make load sync modprobe zaptel insmod ./zaprtc.o insmod: error inserting './zaprtc.o': -1 Invalid module format make: *** [load] Error 1 Is there a

Hi, I am new in asterisk. Is it possible to use it with common dialup modem to connect ptsn to the server? Thanks Regards, Soragan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040123/8e77b5cf/attachment.htm

Hi. I've searched the archives and found nothing regarding collaborating Asterisk with a Panasonic PBX (TD1232 to be exact) Here's my question: Can I use a Wildcard X100P to connect an outgoing line jack (on the Panasonic) to Asterisk, so I can route calls from the PBX to Asterisk, and calls from Asterisk to the PBX? On the hardware page for the X100P card is says it's great for

Before you get too far.... The internet line jacks dont allow outbound calling. they cannot be used as trunk lines to the PSTN. the outbound code has not been written yet. I had to go buy FXO card from digium (that works much better than the Linejack) to get outbound calling to work Dave >>> fplandae@hotmail.com 4/11/2003 5:35:24 PM >>> Hi, I am a newbie. I have been

It seems that the Apollo MD3200 winmodem sold my newegg no longer uses the MD3200 chipset. At least the $13 version I ordered arrived with a different chipset. I made them aware of the issue and hopefully they will be changing their description soon. Anyway, I'm looking for a new source now, anyone know where to get these cards cheap? I know there is ebay, but I'm thus far getting sellers

hi, i've a question. is it possible to buy digium x100p card from italy in some store (also online) without ordering it from USA? on more, did anyone buy a modem with intel chipset 537 or md3200 and where (in italy)? Thanks __________________________________________________________________ Tiscali ADSL Senza Canone, paga solo quello che consumi! Non perdere la promozione valida fino al 27

I've set-up a mirror at ftp.gbnet.net/pub/digium It's (sort of) available via www.gbnet.net/public Mirror's the whole of ftp.asterisk.org/pub Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19

Hi, ztdummy says the following: VoiceBOX:/usr/src/zaptel# modprobe ztdummy /lib/modules/2.4.18/misc/ztdummy.o: unresolved symbol zt_unregister /lib/modules/2.4.18/misc/ztdummy.o: unresolved symbol zt_transmit /lib/modules/2.4.18/misc/ztdummy.o: unresolved symbol zt_receive /lib/modules/2.4.18/misc/ztdummy.o: unresolved symbol zt_register /lib/modules/2.4.18/misc/ztdummy.o: insmod

Hi to everybody, seems that I cannot load the zaptel modules: ztdummy says the following: [root@Hayabusa misc]# modprobe ztdummy /lib/modules/2.4.22/misc/ztdummy.o: unresolved symbol zt_unregister /lib/modules/2.4.22/misc/ztdummy.o: unresolved symbol zt_transmit /lib/modules/2.4.22/misc/ztdummy.o: unresolved symbol zt_receive /lib/modules/2.4.22/misc/ztdummy.o: unresolved symbol zt_register