[Asterisk-Users] Limit incoming calls
We have 2x BRI's connected to Asterisk which give us a total of 4 lines (using the bristuffed package). We would like to limit the number of incoming calls to 2 calls and if a 3rd call comes in, we would like this to go to another extension (voicemail or similar). Is this possible in Asterisk? Thanks -- James Doherty - Systems Engineer Zeald.com - Websites That Work! Web: www.zeald.com | Ph: +64 9 415 7575 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Budgetone ringing volume
We have an office full of budgetones and find the ringing volume to be too loud especially when all the phones are ringing at once on an incoming call. I've tried setting the volume on each phone but the volume setting doesn't survive through reboots or power offs. Has anyone found a way to make the volume setting stick? Thanks -- James Doherty - Systems Engineer Zeald.com - Websites That Work! Web: www.zeald.com | Ph: +64 9 415 7575 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk calls back after phone call
I'm using a Grandstream IP phone to call someone through our asterisk
pbx. The PBX is running "Asterisk 1.0.3-BRIstuffed-0.2.0-RC3" and uses
2x ZAP-HFC cards.
When I call someone, if the call isn't answered and then I hang up, I
get "487" coming up on the grandstream phone. If I pick up the receiver
again and then hang up, the PBX starts calling me back and when I pickup
and listen, there is silence.
Using debug mode doesn't indicate anything suspicious:
-- Executing Dial("SIP/james-d986", "Zap/g1/8171208") in new stack
-- Called g1/8171208
-- Zap/1-1 is making progress passing it to SIP/james-d986
-- Hungup 'Zap/1-1'
== Spawn extension (default, 8171208, 1) exited non-zero on
'SIP/james-d986'
The callback happens after the last line above. I don't believe this was
happening before I upgraded to 1.0.3 (I was previously running a pre-1.0
cvs build from about 2 months before asterisk 1.0 was released).
Has anyone experienced this behaviour before?
--
James Doherty - Systems Engineer
Zeald.com - Websites That Work!
Web: www.zeald.com | Ph: +64 9 415 7575
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HFC cards and Asterisk
Hi all, In our asterisk box we have a Fritz card and an HFC card. So far I've had the Fritz card working well, but its time to connect the HFC card up to the ISDN line and get it working. I've followed the instructions on voip-info.org here: http://www.voip-info.org/wiki-Asterisk+zaphfc+install and here: http://www.voip-info.org/wiki-Asterisk+zaphfc+install But when it comes to running "ztcfg -v", I get this: ZT_CHANCONFIG failed on channel 1: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? My /etc/zaptel.conf looks like this: loadzone=nz defaultzone=nz span=1,1,3,ccs,ami bchan=1-2 dchan=3 and my /etc/asterisk/zapata.conf looks like this: switchtype = euroisdn signalling = bri_cpe_ptmp pridialplan=local prilocaldialplan=local pritrustusercid = yes echocancel=yes immediate=yes group = 1 context=demo channel => 1-2 Any ideas? -- James Doherty Zeald.com Network Operations Ph: +64 9 415 7575, Fax: +64 9 443 9794 Web: http://www.zeald.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can only call asterisk once
I have Asterisk RC2 setup here with a Fritz ISDN card on Debian Woody. I'm using chan_capi-0.3.5 and fcpci-suse8.2-03.11.02. Settings are pretty much the default, except in order to get the fcpci module to compile, I had to follow the instructions here: http://www.voip-info.org/tiki-index.php?page=Asterisk%20AVM%20Fritz%20CAPI%20Driver%20Install but the drivers would only compile if I left the CCFLAGS as they were. Now I've been able to successfully call Asterisk from a POTS phone. We have a block of 10 numbers that are on the ISDN line. Once I call one of those numbers, no more calls will go through after that. I have to restart Asterisk in order for it to answer another call. Has anyone seen this behaviour before? It's certainly not ideal ;) Thanks -- James Doherty Zeald.com Network Operations Ph: +64 9 415 7575, Fax: +64 9 443 9794 Web: http://www.zeald.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
