similar to: IAX2 Port strangeness

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

Hi, I just got my TDM400P card (2 modules) and i installed it no probs. The card is detected fine, but for some reason when I add the card to zaptel.conf i get the following error: --snip-- 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? --snip-- My /etc/zaptel.conf looks

Hi All, Something really weird is going on. I just stopped and started my asterisk server, and it just decided to stop starting. I only made one change to the config file for the musiconhold.conf, and now for some strange unknown reason, i'm getting the errors: [chan_modem_i4l.so]Apr 28 15:26:42 WARNING[16384]: loader.c:240 ast_load_resource:

I have two grandstream budtetone-100 and cisco 7960g phones. When I talk via speaker phone on either of the phones I get a lot of echo. Any suggestions? Also how do I turn on the mark echo canceller. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, April 13,

Hi, I restarted my * box before and now my second X100P card has stopped showing up in Asterisk. It shows up in my dmesg in Gentoo, but only 1 channel is being registered: [chan_zap.so] => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, FXS Kewlstart signalling == Registered channel type 'Zap' (Zapata Telephony

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

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 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

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

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Hello, On my windows xp machine, with cygwin, I run rsync to backup important files on another drive (samba share, mapped on h:). It works fine. I currently use the following command line: rsync -rtv --del --delete-excluded --progress --iconv=ISO-8859-1,UTF-8 --partial \ --exclude "/Videos/" \ --exclude "/Ma musique/" \ /cygdrive/d/Documents/ \

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

We have an Asterisk server that speaks IAX2 to Magrathea to get to the PSTN. Our local phones are a mix of Cisco 7940s and Grandstream BT100s all configured for SIP with silence-suppression disabled. Everything is configured to use a-law encoding. The version is: sip*CLI> show version Asterisk CVS-05/06/04-18:45:57 built by root@sip on a i686 running Linux Incoming callers are complaining of

On Fri, Apr 3, 2020 at 4:09 PM Patrick DERWAEL <patrick at derwael.be> wrote: > Exactly the same... ??? > > > [root at plexvm ~]# mount -t cifs //192.168.1.200/mp3 /home/plex/Musique > --verbose -o username=plex,password=plex,domain=DERWAEL,ro,auto,vers=3.0 > mount.cifs kernel mount options: > >