[Asterisk-Users] Detect phone pickup, caller ID AGI

it.) -- Kevin DeGraaf ___ 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] AGI - calling functions, CHANNEL STATUS broken?

: Status of 231: -1 Any advice? -- Kevin DeGraaf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] AGI - calling functions, CHANNEL STATUS broken?

still would like to know whether it's possible to call Asterisk functions from within AGI scripts... -- Kevin DeGraaf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update

Re: [asterisk-users] AGI - calling functions, CHANNEL STATUS broken?

number. Oh, well. :-) Thanks. -- Kevin DeGraaf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] AGI - calling functions [SOLVED]

Solved: use Asterisk::AGI; my $AGI = new Asterisk::AGI; my $peerst = $AGI-get_variable(SIPPEER(123|status)); my $peercc = $AGI-get_variable(SIPPEER(123|curcalls)); This works fine in 1.4.18. Thanks. -- Kevin DeGraaf ___ -- Bandwidth

Re: [asterisk-users] RedFone foneBRIDGE2 2e1 - anyone used it or another TDMoE bridge?

cancelling module and lots of zapata.conf tuning...) -- Kevin DeGraaf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo

[asterisk-users] TE121, echo issues, NMIs

, but apparently that's not the case. -- Kevin DeGraaf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk

[asterisk-users] Tracking T1/PRI channel status - inbound vs. outbound

to do this? Thanks. -- Kevin DeGraaf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Multiple lines on Linksys/Sipura phones

why I'd need to purchase it. I have tested a SPA942 with Asterisk, and even without the upgrade, I can easily send/receive/hold four separate calls at a time, using the four line keys. What am I missing? Thanks. -- Kevin DeGraaf ___ --Bandwidth

[asterisk-users] Pre-routing manipulation of calls

pre-incoming context to do the manipulation and then jump to the real incoming context containing the specific extension stanzas? 3. Or, is there another method that would be more elegant? Thanks. -- Kevin DeGraaf ___ -- Bandwidth and Colocation

Re: [asterisk-users] Pre-routing manipulation of calls

database. Is there some advantage (e.g. scalability) to using MySQL? Thanks. [0] http://tinyurl.com/3jn62a -- Kevin DeGraaf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE

[asterisk-users] TE121B server recommendation

like a solid recommendation before I suggest another purchase. Thanks. -- Kevin DeGraaf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

[asterisk-users] foneBRIDGE2 vs. foneBRIDGE2-EC

the other. Thanks. -- Kevin DeGraaf ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Polycom IP4000 - Device does not match ACL

tried using x.x.x.55 as both the proxy value only, the server value only, and (in the given example) both. I also added the following to sip.conf, to no avail: deny=0.0.0.0/0.0.0.0 permit=x.x.x.0/255.255.255.0 Any ideas about what I've missed would be appreciated. Thanks. -- Kevin DeGraaf