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Hello I have created an iax exten in my iax.conf file: [300] type=friend username=300 secret=*** context=default host=dynamic callerid="some name" <300> auth=md5 Then in my extensions.conf I have: exten => 300,1,Dial(IAX/${EXTEN},20) exten => 300,2,Hangup I can dial from iaxComm (a soft IAX client) and that works fine. But when I try to dial 300 get: WARNING[22077]:

Hello I am running Asterisk on SUSE Linux Professional 9.3 on a VIA Epia M Series motherboard - CPU runs at 1GHz. There is no fan - just a large heatsink. Currently system is running off standard IDE hard drive - because I couldn't get astlinux to run with my Digium TDM04B card (only PCI card in system). Strangely I also have the same system also running SUSE Linux running as a file

Hello I want to setup an Asterisk in several offices with 4 BRI ISDN. I am looking for recommendations on hardware. Criteria would be ease of setup, reliability and cost. The Eicon 4 BRI cards seem fairly pricey. Shame Digium don't do a ISDN BRI card. Angus -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello When I dial out from my Asterisk (using Digium analog TDM04B card over pstn line), calls appear to be from +34<rest of number> I am in UK which is +44 so cannot work out why seeing +34. In my zapata.conf I have: loadzone = uk defaultzone = uk I can't find any country specific stuff in any other conf files. Any ideas how I can correctly set so that calls from my asterisk do

Hello I am using a Snom 190 and the quality seems OK. Trouble is the volume is quite low and full volume on the Snom is still too low. Is there something I can do on the asterisk to increase the volume? Angus

Hello On a Sipura SPA-841 handset (and also at other end) you hear a sea wave like sound - it gets louder then softer and continually repeats. I don't remember hearing this when using other handsets. But what is this effect? How can I reduce it? Angus

Hello I have been asked by a client to process a list of telephone numbers. Asterisk should call each number in turn and if the recipient of the call answers, play a message - eg from a wav. How would I go about doing that? Angus

Does VoicemailMan have to be installed ? Why not available. I have setup a mailbox in voicemail.conf and I can leave a voicemail - just cannot pickup up using *97. My *97 code in extensions.conf: exten => *97,1,Answer exten => *97,2,VoicemailMain(${CALLERIDNUM}@default) exten => *97,3,Hangup asterisk console: Verbosity was 8 and is now 12 -- Executing

Hi My asterisk output is: chan_sip.so => (Session Initiation Protocol (SIP)) Asterisk Ready. -- Registered SIP '201' at 192.168.0.55 port 33906 -- Saved useragent "X-Lite release 1011s stamp 41150" for peer 201 -- Executing [907768385144 at default:1] Dial("SIP/201-083e75c0", "ZAP/g1/907768385144|60") in new stack [Oct 4 11:54:27]

...xgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include AMP configs #include zapata_additional.conf ;Include genzaptelconf configs #include zapata-auto.conf What am I doing wrong? Angus Comber angus@NOSPAMiteloffice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050620/4e977642/attachment.htm

Hello I am struggling to get my TDM04B working. Just to rule out a hardware problem how can I check that the hardware works? How can I then check that the drivers are loaded correctly? Angus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050620/470458c6/attachment.htm

Hello Maybe a silly question, but after some searching couldn't find answer. Is there a number I can dial to pickup and listen to my voicemail messages on my SIP phone? I am used to eg dialling *17 to pickup my voicemail messages on Avaya system? Angus -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello I am (attempting) to run the astlinux version of Asterisk on a VIA embedded platform. I have a TDM04B and pretty sure zaptel.conf and zapata.conf setup OK. They worked fine with same card in traditional PC anyway. I think need the module wcfxs for a Digium TDM04B card. Is this module not part of astlinux? Do I need to download it? Or is it in opt? I see wctdm - but think that is

Hello I have installed a TDM04B and disabled any devices not required in my PC. (TDM04B is analog card with 4 ports to plug into telephone co lines). I am running this version of * Asterisk CVS-HEAD-01/10/05-02:11:15-AstLinux built by root@ls on a i686 running Linux As you see below the wctdm module is loaded: pbx root # lsmod Module Size Used by binfmt_misc 12296 1 - Live

Hello In my extensions.conf file: [frompstnisdn] exten => s,1,Dial(SIP/200&SIP/202,20) exten => s,2,Voicemail(su200) exten => s,3,Hangup I use the s, start, extension to handle incoming calls. In my zapata.conf: context=frompstnisdn This works ok on another asterisk box I setup. But on incoming calls I get: -- Extension '787367' in context 'frompstnisdn'

Hello I am using a VIA Epia ME6000 with a 600MHz Eden Fanless CPU. Is this likely to be enough power for a 8 extension system with 6 external pstn lines? How important is cpu? Is there some measure, eg xMHz CPU per extension or something benchmark? I have installed 512MB memory - again any benchmark for asterisk memory usage? Angus

Hello I am starting to learn AGI. I have setup an extension to play the agi-test.agi perl script and the output I get is this on console: On Polycom 300: -- Executing Answer("SIP/200-72d2", "") in new stack -- Executing AGI("SIP/200-72d2", "agi-test.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-test.agi --

Hello We have setup a doorbell which has an inbuilt analog phone which is connected to our Asterisk via a SPA2000 ATA. The problem we are getting is that when a caller presses the buzzer it is taking two or more minutes to finally call the reception phone. In the SPA2000 I have set dtmfmode to be inband. I notice that with the asterisk you dial a number and then it waits for a timeout

Hello I have downloaded asterisk-addons but when I make install get: cc -fPIC -I../asterisk -D_GNU_SOURCE -DMYSQL_LOGUNIQUEID -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:164:64: macro "AST_LIST_REMOVE" requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: error: