similar to: BUG: FOP reports incorrect (duplicate) IP address until restarted

Hi, I have an odd problem when doing a blind transfer. The transfer is intiated and the transferred caller hears nothing until the timeout. I have tried setting the 'r' and the 'm' variables in the dial command. Nothing happens when I use the 'r' variable when I use the 'm' variable I briefly hear music on hold and then it stops until the timeout for no answer

Hi, I am not sure if this is a bug in FOP (Flash Operator Panel), a configuration error on my part or a bug in Asterisk. I am using Asterisk 1.2.5 and Zaptel 1.2.4 on a Centos 4.2 server with Linux version 2.6.9-22-EL-i686. Kernel updates are excluded and the server has been updated using 'yum -y update'. Okay here is the scenario: I am using AgentCallBackLogin as an extension in my

A call comes in from the PSTN, Asterisk answers it, it goes to the directory, and then to the extension the caller designates and the user at that extension answers. The user at the extension then wants to transfer the call to another extension; on the Cisco 7940 they push the ?more? soft key, then the ?Transfer? soft key, then enter the extension number they want to transfer to, and hit the

Hi What causes deadlock? Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x82acb10', 10 retries! Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x8298160', 10 retries! Here is the portion of the log: Apr 5 14:02:42 NOTICE[23363] chan_zap.c: Got event 18 (Ring Begin)... Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing

Sorry if this comes in twice. Wasn't subscribed first time :-( Anyone help me here...... It worked once :-( I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22 I have the 7690 with a SIP iamge (Whatever latest is ) I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server.

Does anyone kmow what these errors mean or how they can be fixed. I'm using asterisk on a Fedora Core 2 box with a TDM400P with 2 fxo and 2 fxs ports. Dec 29 17:17:52 WARNING[6019]: chan_zap.c:5469 ss_thread: Channel Zap/4-1 in prering state, but I have nothing to do. Terminating simple switch, should be restarted by the actual ring. -- Hungup 'Zap/4-1' == Starting post

I am using version: CVS-v1-0-12/13/04-18:46:23 with a TDM400p (2 fxo, 2 fxs ports) and I keep getting errors along with phantom calls: Dec 21 16:02:07 NOTICE[5872]: chan_zap.c:5363 ss_thread: Got event 17 (Polarity Reversal)... Dec 21 16:02:14 WARNING[5872]: chan_zap.c:5434 ss_thread: CallerID returned with error on channel 'Zap/4-1' my analog phone reads caller ID info fine when

Hi all, I'm playing with Asterisk and I've already configured all needed .conf files. It works quite well, but now I need your help to tune the system: when I place a call from a softphone to the PSTN, I can't hear directly Telco's tones and I can't use its services, e.g. a mobile's answering machine. I don't know if I have to modify the dialplan or if it depends on my

Anyone help me here? I am a newbie so be gentle ;-)...... It worked once and then I played with the configs. I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22 I have the 7690 with a SIP iamge (Whatever latest is ) I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server. Here

My phone still says Not-Registered. I have a Polycom SoundPoint 600 SIP phone. Here is my sip.conf file: ; ; SIP Configuration ; [general] context=default ; Default context for incoming calls port=5060 ;added bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ;

Hello, I'm trying to do call forwarding based on this: http://www.voip-info.org/wiki/view/Asterisk+call+forwarding In the extensions.conf file I have several context defined (local, longdistance, mobile, international and so on). Each user can be associated with different context (so can make only i.e. local calls). How to set calls forwarding only to numbers that are available in

Hi everyone! I'm quite a newbie at this Asterisk stuff so please bear with me. We've recently decided to start training in Asterisk via AsteriskNow! Asterisk version is 1.4.18.1 through AsteriskNow! 1.02 The box we have is paired with a Digium TE110P and we've managed to get it to the point where incoming calls via a single DID (from NTT Japan) can be received and answered

I am trying to setup a simple pstn gateway using Asterisk and a X100p card. I have got everything installed using Redhat 9 and am able to load Asterisk. I also configured sip and I am able to connect to the asterisk gateway with Xlite on the windows side. I am able to dial 1000 and get the welcome message. What I am NOT able to do is dial a seven digit local or 10 digit long distance number and

Hello guys. I am a fairly new user to Asterisk, and I'm just having a tough time. My goal is to set up a VOIP PBX. I have signed up with a BroadVoice number, and I have three systems with SIP phones. The PBX and the SIP phones are all behind a Cisco PIX running NAT. I am using Asterisk CVS version from yesterday. I also tried 1.0.3 with little luck. The SIP phones are two X-Lites on

I'm trying to use the FollowMe app with Asterisk 1.4.17. I've followed the WIKI page on setting it up but I can't seem to get it to work. Here is my Asterisk version: pbx1*CLI> core show version Asterisk 1.4.17 built by root @ pbx1 on a i686 running Linux on 2008-01-10 12:08:48 UTC Here is a log of when the FollowMe is being called: NOTE: I've tried to use the AstDB as

Hi, What does the following error mean: Apr 5 12:39:40 NOTICE[22755] app.c: Huh....? no dial for indications? Here is the 'full' log around the error: Apr 5 12:38:24 VERBOSE[22755] logger.c: -- outgoing agentcall, to agent '3002', on 'Local/510@default-6b6c,1' Apr 5 12:38:24 VERBOSE[22755] logger.c: -- Called Agent/3002 Apr 5 12:38:24 VERBOSE[22755]

I am trying to receive both CID and DNIS from the telco through a non-pri T1. Currently I have the T1 setup and operational both outbound and inbound calls are completed as should be expected. The calls came in and were placed in the context specified in zapata.conf on exten => s,1. I have requested that the telco provide callerid (they call it ANI) along with 10 digit dnis for my 800

Hi list, i've problems with my * server and the 4 phones which are linked to it. i've 2 grandstream bt100 with the firmware upgraded to 101, a wi-fi phone (i don't know its brand) and another ip phone i don't know its brand. with this sip.conf : [general] port = 5060 bindaddr = 192.168.100.229 context = default ;x changed from default to sip localnet = 192.168.100.0/24

Hi all, Running bristuffed 1.2.4 system with solely Cisco 7940/7960 phones with SIP. I'm using also op_panel 0.25 (snapshot). I'm using * queues. I want to properly implement DND via *78 and *79. I'm using op_panel's documentation RECIPE 1 solution with astdb and dnd variables and this is fine for FOP. The DND works in normal cases, since I catch it with my Macro dialsip, HOWEVER

ONLY ON MONDAY! Well it used to work - calls between my aaln's that is. I moved from debain to redhat (same conf. files for asterisk) and this is what I get.. looks like several errors. errors I never got before. Also asterisk isn't observing the digits as I dial them like it used to however it still trys to route the call when I'm finished dialing. Anyone with a though on this?