[Asterisk-Users] caller id over IAX
I have calls come in on a PSTN line and I can see the called ID info just fine. Sometimes I route those calls to a remote office over IAX, but there the callerd ID information is somehow lost.. it appears on the phones there as UNKNOWN NUMBER asreceived Is it possible to have the called ID info from the incoming call passed through over the IAX connection? Should I post some config files? Any help welcomed. Other than this problem our setup works great. DB ___ 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] transfer question
I have two Asterisk boxes, Box1 has TDM22B and Box2 has a TDM11B. From Box1 I can pick up Zap/1 and dial 300 causing Zap/1 on Box2 to ring. From Box2 I can pick up Zap/1 and dial 100 causing Zap/1 on Box1 to ring. PSTN calls come into Box1. They ring thru to Box2 on Zap/1 via IAX, then if no answer they ring Zap/1 on Box1. When such a call comes in and is answered on Box1, I can transfer to Box2 by dialing #300. But when such a call is answered on Box2, attempts to transfer to Box1 by dialing #100 fail. If I can dial 100 from Box2, then why can't I transfer using the very same Dial command? exten = 100,1,Dial(IAX2/name:[EMAIL PROTECTED]/100) When I watch the attempt on the CLI of Box1, it tells me that there is no such extension 1 in context ' '. I think I have a conceptual problem.. can anyone help? DB ___ 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] Very odd musiconhold
I've followed the docs on how to configure musiconhold. I then added an extension to test it like this: exten = 6601,1,WaitMusicOnHold(30) And when I dial that extension I do hear something for 30 seconds - but it is not what I'd call music... it sounds more like the sound effects from a scifi flick - now and then it sounds like jail cell doors closing and echoing. Kinda cool but not what I expected and it would probably confuse or scare customers. I see no errors displayed on Asterisk's CLI nor anything that I could find in logs. I tried downloading another mp3 and got the same odd sounds. Any ideas ?? Very odd DB ___ 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] Problems starting Asterisk with TDM22B
Im getting the following error when I try to start Asterisk WARNING[16384]: chan_zap.c:765 zt_open: Unable to specify channel 1: No such device or address Nov 29 21:40:43 ERROR[16384]: chan_zap.c:6195 mkintf: Unable to open channel 1: No such device or address I'm using a TDM22B card. = # Zaptel Configuration File # loadzone=us defaultzone=us fxoks=1-2 fxsks=3-4 == Do I need to supply more config files or does some one recognize this error? Please help. DB ___ 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] Problems starting Asterisk with TDM22B
Im getting the following error when I try to start Asterisk WARNING[16384]: chan_zap.c:765 zt_open: Unable to specify channel 1: No such device or address Nov 29 21:40:43 ERROR[16384]: chan_zap.c:6195 mkintf: Unable to open channel 1: No such device or address I'm using a TDM22B card. = # Zaptel Configuration File # loadzone=us defaultzone=us fxoks=1-2 fxsks=3-4 == Do I need to supply more config files or does some one recognize this error? Please help. DB ___ Asterisk-Users mailing list Asterisk-Users at 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 1. Try to run: # ztcfg -v What happened!? 2. Did you plug the TDM at the power supply!? Denis. Hi - yes the power is connected and: [EMAIL PROTECTED]:/etc/asterisk# ztcfg -v Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. I do notice a several second delay when I do 'modprobe wctdm', but no errors are displayed. Any more advice? I'm kinda stuck here. DB ___ 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] Remote answer not detected
Here's my a section of my simple extensions.conf exten = s,1,Answer exten = s,2,playback(thx4call) exten = s,3,Dial(Zap/1|15) ; Calls channel 1 exten = s,4,playback(trying_bert) exten = s,5,Dial(Zap/4/2326932|15) exten = s,6,Voicemail,u100 exten = s,7,hangup exten = s,104,Voicemail,b100 exten = s,105,hangup exten = s,106,Voicemail,u100 It works, but when the call is routed out on ZAP/4 (at priority 5), Asterisk seems to not realize the call is answered. After 15 seconds it proceeds to voicemail interrupting the call. Can anyone help? DB ___ 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] Remote answer not detected
Il dom, 2004-11-14 alle 00:13, DB ha scritto: Here's my a section of my simple extensions.conf snip exten = s,5,Dial(Zap/4/2326932|15) exten = s,6,Voicemail,u100 snip It works, but when the call is routed out on ZAP/4 (at priority 5), Asterisk seems to not realize the call is answered. After 15 seconds it proceeds to voicemail interrupting the call. Can anyone help? eh, perhaps with some details about your zap... ie what card? zaptel.conf? zapata.conf? matteo, still without divinatory powers Hi - thanks for the reply - here's that info: card is TDM22B zaptel.conf: == fxoks=1-2 # Make sure that the FXS(green) modules are closest fxsks=3-4 # This is for the FXO module(s) becaus defaultzone=us loadzone=us == zapata.conf: === [trunkgroups] [channels] switchtype=national signalling=fxo_ls rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes callprogress=yes progzone=us signalling=fxo_ks echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs. echocancelwhenbridged=yes echotraining=400 ; Asterisk trains to the beginning of the call, number is in milliseconds callerid=asreceived group=1 context=MD_line1 ; Points to the default context of your extensions.conf channel = 1 signalling=fxo_ks echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs. echocancelwhenbridged=yes echotraining=400 ; Asterisk trains to the beginning of the call, number is in milliseconds callerid=asreceived group=2 context=MD_line2 ; Points to the default context of your extensions.conf channel = 2 signalling=fxs_ks group=3 context=incoming_9141252 channel= 3 ; Again if you only have one FXO module remove the '-4' signalling=fxs_ks group=4 context=incoming_3493729 channel= 4 ; Again if you only have one FXO module remove the '-4' DB ___ 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] Dialplan question - doesn't quite work
Hi - I have a zaptel card with 4 modules - 2 fxs and two fxo. I have two phone lines coming into my house. For now I want an incoming call to ring a phone here, and then if no answer to ring another number (by calling out on the other line) for 15 seconds... then if no answer send to voicemail. It seems to work, except the last part... the outgoing call doesn't time out... if not answered it will ring for eternity. exten = s,1,Answer exten = s,2,playback(thx4call) exten = s,3,Dial(Zap/1|15,t) ; Calls channel 1 exten = s,4,playback(trying_bert) exten = s,5,Dial(Zap/4/12168810880|15,r) exten = s,6,Voicemail,u100 exten = s,7,hangup exten = s,104,Voicemail,b100 exten = s,105,hangup exten = s,106,Voicemail,u100 exten =? s,107,hangup I am a newbie but have done lots of reading and playing around. ANy advice is welcome. DB ___ 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] Dialplan question - doesn't quite work - more info
Below is a section of my extensions.conf. As you can see I am trying to route an incoming call to a remote phone number if it isn't picked up within 15 seconds locally. Now this basically works.. the call is routed to the remote number. But often it will still timeout and go to voicemail.. as if Asterisk thinks the remote call did not connect - and this interrupts the call. Sometimes however the voicemail does not kick in and it works fine. The rxgain and txgain settings may be having some effect but I'm not certain. Does anyone have any advice? exten = s,1,Answer exten = s,2,playback(thx4call) exten = s,3,Dial(Zap/1|15) ; Calls channel 1 exten = s,4,playback(trying_bert) exten = s,5,Dial(Zap/4/2326933|15) ;exten = s,5,Dial(Zap/4/12168810880|15) exten = s,6,Voicemail,u100 exten = s,7,hangup exten = s,104,Voicemail,b100 exten = s,105,hangup exten = s,106,Voicemail,u100 ___ 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
