similar to: What is web login password for Asteirsk@Home

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 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 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 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

I have 2 sip accounts setup - 200 and 202. If I do sip show peers I get: sip show peers Name/username Host Dyn Nat ACL Mask Port Status 202/202 192.168.0.6 D 255.255.255.255 5060 Unmonitored 201/201 (Unspecified) D 255.255.255.255 5060 Unmonitored 200/200 192.168.0.3 D 255.255.255.255 5060

Hello I setup Asterisk@Home with purely VoIP and it worked fine. I then added an X100P card so I could call out / take inbound calls via PSTN and that went fine. But I have just added a TDM400P card (specifically a TDM30B) and now problems. Here is some of the output. Any ideas on what I should be looking at next? When I run genzaptelconf -s -d I get lots of erors on screen - bit I can

Hello I see it is possible to buy Flash Disks up to 4GB now. Has anyone any experience of building an Asterisk system with a flash disk as the only storage device? Any brands you recommend? Is 2 or 4GB enough for an Asterisk installation? Typically how many MB is required for voicemail recording files for say a 10 user system? What about voicemail - I suppose files could be emailed and

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

Hello I have a Junghanns ISDN BRI card for incoming calls and use SIP Polycom IP300 phones. My config files look like this: features.conf pickupextn = *8 zapata.conf context=frompstnisdn group=1 callgroup=1 pickupgroup=1 I also edited sip.conf like this: group=1 callgroup=1 pickupgroup=1 But on internal and incoming calls if I dial *8 from any phone I cannot pickup. Do I need to add

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 have played about with a TDM400 card and plugged in some standard analog phones. I am using the card in FXS mode - for analog extensions. I did notice that one of my phones did not ring and I wondered why. I later read in Paul Mahler's book VoIP Telephony with Asterisk that in his section on the TDM400 on page 127 he says "In the UK, you may need an adapter that provides a

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:

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 have a Grandstream GXP2000 with latest firmware. When I use it holding the handpiece I don't hear any echo - neither does other end. However, if I use it handsfree, the other end notices echo when they speak - ie their voice is echoy. I hear their voice being a bit echoy. Is this purely down to the IP Phone? Is there anything I can do about it? I considered buying a more

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 am using a Junghans quadBRI ISDN card and it is loaded and working. In Asterisk if I connect to ISDN line it is detected and tells me so. In my zapata.conf I have (abbreviated): [channels] switchtype=euroisdn signalling = bri_cpe context=default group=1 channel => 1-2 ;plus group 2 - 4 zaptel.conf: loadzone=uk defaultzone=uk # qozap span definitions # most of the values should

Hello I am using a Junghanns QuadBRI ISDN card - the module name is qozap. If I like at my interrupt assignment, qozap is sharing interrupt 10 with libata and uhci_hcd. I think libata is the IDE hard drive module and uhci_hcd is a USB module. linux:~ # modprobe qozap linux:~ # cat /proc/interrupts CPU0 0: 12634579 XT-PIC timer 1: 10 XT-PIC i8042