similar to: Instant hangup on busy Zap channel.

Displaying 20 results from an estimated 7000 matches similar to: "Instant hangup on busy Zap channel."

2004 Jan 21
3
Making a call with sample.call
Hi there, I'm having some trouble with getting Asterisk to make a call, I think it should be quite easy, but anyway... Using the following file contents: ## Channel: Zap/3/<TEL NUMBER HERE> MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: phones Extension: 502 Priority: 1 ## Extension 502 is simply one that plays a sound back. When I dump this file into
2007 Jul 03
1
Configuring BLF or Asterisk presence/Hints feature
Hi all, I am working on asterisk 1.2.18 zaptel 1.2.17 Polycom 650 polycom 430 SIP version 2.0.3.0131 for IP 650 SIP version for IP430 2.0.3.0127 freepbx 2.2.1 I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me. Regards FArooq ********************************************** 1 ********************************************** in my
2003 Nov 25
5
Distinctive ring confusion
I am somewhat unsure as to the definition of "Distinctive Ring". What I am trying to achieve is to have Zap connected phones (TDM400P) ring with different cadences depending on whether the call is incoming on the PSTN context or an IAX2 context. Googling, I find this from Mark: I've added distinctive ring support to Asterisk now (also I've added answer confirmation which is
2007 Oct 25
2
Unable to dial out over Zap - span 1 got hangup, cause 44
Hi I posted earlier about having issues connecting to Telewest's ISDN, only to find out later Telewest had forgotten to turn it on - hopefully I'm not having a similar silly problem. My PRI span is now up and operational, but when I try to send a call out over it, I just get congestion tones. Occasionally, I get about one second of ring tones, only for it to cut out and play congestion.
2006 Oct 10
1
Hangup or busy when the peer answer outgoing calls
Hi all.. I have a problem with my asterisk installation. I'm using a Wilcard X100P clone in Spain. Incoming calls work fine, but when I make a outgoing call, a hear the ringing, and the peer phone ring, when the peer answer, asterisk hangup the call, or say busy. This is my conf: zaptel.conf: --------- loadzone = es defaultzone=es fxsks=1 zapata.conf ---------- [channels]
2009 Jul 06
3
Smart-UPS RT 3000 Baud Rate
I have been running a Smart-UPS RT 3000 VA using nut successfully over the 940-0095B cable for a couple of years. I have just received a second unit, but it is slightly different in 3 ways that I have found so far. 1. It's a Smart-UPS RT 3000 XL 2. The 940-0095B cable now uses an RJ45 connector at the UPS end. 3. It now requires a baud rate of 9600 instead of the 2400 used previously.
2008 Oct 14
7
Panasonic x Asterisk if I can emulate Panasonic fast!
Im a 3-days-asterisk-newbie. In 3 weeks, I must have a PBX installed in a new office of ours: Panasonic or Asterisk. Asterisk would be, if I can emulate some Panasonic functions on Asterisk fast, to convince the executives. What I have done until now: Bought 1 Linksys pap2 (2 FXS), 1 Linksys SPA3102 (1 FXS + 1 FXO) for making asterisk tests. Configured Asterisk/Fedora 9 so I can make SIP->PSTN
2007 Jul 05
1
Need Help in Asterisk BLF/Presence/Hints
Hi all, I am working on asterisk 1.2.18 zaptel 1.2.17 Polycom 650 polycom 430 SIP version 2.0.3.0131 for IP 650 SIP version for IP430 2.0.3.0127 freepbx 2.2.1 I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me. Regards FArooq ********************************************** 1 ********************************************** in my
2008 Nov 19
3
TDM400 (?) zap hangup
And if that ain't confusing I don't know what would be. I bought a TDM400 with two modules (FXO, FXS) a couple or so years ago and ended up never using it. Passed it along to a friend who is having some problems with it. (He isn't on this list.) We've both tried searches using Google but haven't been able to find anything that helps. So this is more a question of
2003 Aug 06
1
BRI newbie queries.
Knowing very little about Basic Rate ISDN and having spent the last couple of hours educating myself, I thought I would seek some more informed comment. Please go easy if this is blindingly obvious :) I have a ZyXEL Prestige 100 ISDN Router, a stand alone relic from when we used to access the Net via ISDN. It has an ISDN BRI input, a 10BaseT ethernet connector, an RS232 connector for
2004 Sep 24
2
kernel: Power alarm on module 1, resetting!
I've installed a TDM04B and a TDM40B. I haven't plugged any lines into them yet but I'm starting to see this in my logs... [root@webster asterisk]# grep alarm /var/log/messages Sep 20 09:13:22 webster kernel: Power alarm on module 1, resetting! Sep 22 11:07:07 webster kernel: Power alarm on module 1, resetting! Sep 22 16:10:55 webster kernel: Power alarm on module 1, resetting! Sep
2005 Jul 24
2
Busy Lamp Field SIP Phone
Does anyone have a recommendation for a good SIP phone with a busy lamp field? I need my operator to be able to see extension status for about 20 extensions and transfer via HOLD + extension button. I've got a pair of SNOM 360s with the sidecar, but I'm very disappointed with them. The buttons are cheap and rubbery like a Sipura 841, the handset cord is short and cheap, the audio quality
2005 Jul 25
1
why zap call transfer fails?
Hi, I am configuring Asterisk with TDM400 card with 1 FXS and 1 FXO module. My first goal is to allow phones to be able to call out through the asterisk PBX. After channels and dial plans setup, Zap/1 connect to phone and Zap/4 connect to provider , when I dial the phone, the following message shows in the asterisk console ------------------ Starting simple switch on 'Zap/1-1' --
2006 Jan 20
1
instant fallback to zap in case of sip/iax/xyz-failure
i would like to carry some oversea pstn-destinations via sip to providers like stanaphone, however, in case of a network-failure or if the provider is not available, i want to fallback to the zap-channels so the call is carried out to the pstn directly. the usual approach would be to check the dialstatus(e.g.NOANSWER). however, asterisk tries >60seconds to reach that peer(even when the ip
2005 Jan 17
1
transfers with zap channel
Ok, I've seen discussion before on doing transfers (attended and unattended), there seems to be much confusion over it. As things sit, I've been trying (unsuccessfully) to do transfers with a zap channel (analog phone attached to TDM400). I have no idea what I'm missing. My current understanding is that I need to have transfer=yes in zapata.conf, do a flash hook, dial the 2nd
2007 Apr 12
4
Zap failure: cause 66 - Channel not implemented
Hi, I just compiled and installed Asterisk-1.4.2 along with zaptel-1.4.1 and libpri-1.4.0 on a Debian machine with a TDM400P card. Everything goes ok but when I try to make a call through the ZAP channel get an error message about NO ZAP CHANNEL AVAILABLE. Ztcfg and zttool show the card correctly installed. When I tried to use the debug command ZAP SHOW, it was not present in the CLI. My
2007 Nov 23
1
AMI Newstate Ringing events -- Inconsistent caller id ?
Hello list, I'm observing what I believe to be inconsistent behaviour regarding "Newstate" AMI events for the "Ringing" state. As such I come to you asking for experience or advice: am I wrong or should I file a bug ? I present you a short introduction which I feel is relevant; however, if you want to go straight to my technical question, please scroll
2007 Feb 20
0
Can't get ANSWEREDTIME after hangup using ZAP
Dear all, I tried to make a call with PHP AGI. $rc = execute_agi("EXEC DIAL ZAP/g1/$myphonenumber|60|rhHL(" . ($max_total_seconds * 1000) . ":60000:30000) "); $rc = execute_agi("GET VARIABLE ANSWEREDTIME "); And I can't get the answered time after caller hangup in this method. But if I use a SIP channel as below: $rc = execute_agi("EXEC DIAL
2003 Oct 05
0
Zap Analog Line Hangup Problem
Hi.. I'm having the next problem... with the busy detect = yes... If i have it... The * it hang up the calls when they are active... ( Incoming ant Outgoings calls). If i haven't it .... * Doesn't detect when some one hang up and never close the channels... WHAT I CAN DO ????
2004 Apr 21
0
SIP ACK // CSeq 0 => ZAP Channel hangup
Szenario: UA(Grandstream) => PROXY(SER) => GATEWAY(*) => PSTN After sending the SIP ACK From Gateway (*) ACK sip:123456@127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK86c0bd474ea746b5 From: "Me" <sip:123456@mydomain.de>;tag=0f63d269bc25545d To: <sip:100@mydomain.de>;tag=as05df60b5 Contact: <sip:100@192.168.0.1> Call-ID: