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...anaging the list at asterisk-users-admin@lists.digium.com When replying, please edit your Subject line so it is more specific than "Re: Contents of Asterisk-Users digest..." Today's Topics: 1. VoiceMailBox wav file format in EMAIL. (James Gardiner) 2. TDM400P Issues (Jeremy Bogan) 3. Re: TDM400P Issues (Vic Cross) 4. Re: TDM400P Issues (Jeremy Bogan) 5. Re: TDM400P Issues (Christian Hoffmeyer) 6. Re: TDM400P Issues (Jeremy Bogan) 7. Re: ZAPRTC question(s) (Tony Mountifield) 8. Re: TDM400P Issues (Jeremy Bogan) 9. Re: TDM400P Issues (Jeremy Bogan) 10....

Does anyone know a place in Australia that sells a power supply suitable for the IAXy? I haven't had any luck tracking one down. -- jeremy bogan [ jeremy@segpub.com.au ] segment publishing - design.develop.host

i have created some ''public'' controllers like so: ./script/generate controller example and an ''admin'' controller like this: ./script/generate controller admin/example now i want to have caching in the ''public'' controllers so i add something like this: caches_action :show, :whatever but i want the ''admin'' controller to do

...annels configured. ZT_CHANCONFIG failed on channel 2: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? Anyone have any ideas? I tried doing a search but couldn't really find anything. Thanks! -- jeremy bogan [ jeremy@segpub.com.au ] segment publishing - design.develop.host

...06/04-08:50:34 built by root@pbx on a i686 running Linux User2 is running: Asterisk CVS-HEAD-08/13/04-08:18:35 built by root@musique on a i686 running Linux It's strange that everything was working and then all of a sudden they started registering on non-standard ports. Thanks. -- jeremy bogan [ jeremy@segpub.com.au ] segment publishing - design.develop.host

hello all, I am using the msql cdr module to store cdr in db, I realised that it does't capture the start and end time af a particular call record. Therefore I dive into the source code to add the start and end time into the query (add something like cdr->start, cdr->end), but end up getting segfault. the original version of cdr_mysql.so works fine but I need the start time and end

...fine for the last few weeks, until I just stopped and started *. I tried doing a search in the archives and Google, but couldn't come up with anything. Has anyone seen this error b4? I even tried disabling the chan_modem.so driver from modules.conf but it still errors. Thanks! -- jeremy bogan [ jeremy@segpub.com.au ] segment publishing - design.develop.host

I have problem to change from handsfree mode to handset mode. When I switch from handset to handsfree while waiting for connection I press the green speakerphone button once. It is all well. Once it is connected I don't want to give the called party too much echo and I want to switch it back to handset. If I press the green button again I lose the call. Anyone knows whether it is possible

What happens when you do "stop now" like the error states? Sean -----Original Message----- From: Ryan Parlee [mailto:listbox@jesca.com] Sent: Saturday, April 03, 2004 9:56 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Unabled to exit console No matter what I try, Asterisk won't let me out of the console. If I CTRL+C, of course, the process will terminate. I

Hi all, Does anyone know of any providers that can offer local numbers based in the UK via IAX or SIP? We're looking at getting a number based in the UK. Thanks! -- jeremy bogan [ jeremy@segpub.com.au ] segment publishing - design.develop.host

> I don't use it; just figured I'd try to help Thanks Doug... So then for the list - I have chan_console working now.... But I am trying Console/dsp and Console/ALSA and both give an error about not found. What have I missed ? Thanks Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is it at all possible to change the event that observe_field/ prototype watch for? In particular, is there any way to use an event other than ''change'' for a textarea? (Without hacking prototype, that is ;) Thanks. d. -- Daniel Bogan waferbaby : <http://waferbaby.com/> waferboy : <http://waferboy.com/> I make the Internets.

I have looked though the archives on this issue and I can not find a final answar on it. So I will bring it up again. I have an x100p card.....if Asterisk picks up a call, and it see if there are fax tones on it, and if so transfer it somewhere. What is a user has picked it up. Can it still transfer it, or would the user have to? Thanks again, Michael

...ion=yes channel=2 Running ztcfg -vv shows me both cards: Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. Anyone else experienced this problem? Thanks! -- jeremy bogan [ jeremy@segpub.com.au ] segment publishing - design.develop.host

Hello I am trying to configure Asterisk on RedHat Linux 9 with one X100P one port card and one USB one port FXS card. I can modprobe wcusb but ztcfg always return ZT_CHANCONFIG failed on channel 2: No such device or address (6) error message. Also unable to config outgoing call using SIP SoftPhone. Any working examples of configuration files is highly appreciated. I mentoned followin lines

Does anyone know what VoIP phones are available in australia for 'decent' prices (ie, decent for me is under $250 - $300 max) So far I've found the following: D-Link DPH-100S $620 Various Cisco phones at > $500 each ZyXEL Prestige 2000W $380 Haven't been able to find much else around. I would like to get at least one grandstream but the delivery charges will probably add up

Hey guys, Cal someone help me. I'm register voiceplus DID i try to config fllow example but not work. When i test call to number and debug iax2 in my asterisk not found packet. My iax.conf -------- register => in-xxx:yyy@gw5.voicepulse.com [voicepulse] context = voicepulse-incoming secret=yyy auth=md5 type=friend host=gw5.voicepulse.com ------ extention.conf ---- [voicepulse-incoming]

Hi there, I've got an installation where there's 12 POTS line incoming into *, and am trying to get some insight as to which VoIP hard phone would be most suitable for this scenario. Most of the VoIP phones I've looked at only have 4-6 line presentations; is anyone aware of one that has more? I tried to get some info about Snom's Keypad 220 since it has loads of programmable

Does anybody developed Predictive Dialer using Asterisk/Digium PBX? Another question: does anybody developed an Dialer using the X100P board? Julio -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030718/be051d49/attachment.htm

...on) /usr/include/asm/smp.h:66: confused by earlier errors, bailing out make: *** [zaptel.o] Error 1 The server is running Gentoo: Linux pbx 2.4.25_pre7-gss-r2 #2 SMP Tue Apr 13 20:29:53 EST 2004 i686 Intel(R) Celeron(R) CPU 2.40GHz GenuineIntel GNU/Linux Has anyone seen this before? -- jeremy bogan [ jeremy@segpub.com.au ] segment publishing - design.develop.host