similar to: call problem...

I'm using FreePBX 2.5.2.2 with Asterisk 1.6.1.4. If I make a call and then decide to blind transfer them using ## my side of the call is not hung up. Instead it sends me to voicemail. If somebody calls me and then I blind transfer them with ## I am hung up on as expected. I called from 8678 to 28688. I then transferred the call to 8532. Asterisk acts like it wants to hang up, but then

Hi people, I have the following dialplan, but it doesn't have the behavior that I think it should have. [default] exten => 2001,1,Answer exten => 2001,n,Dial(local/3005) exten => 2001,n,Hangup exten => 3005,1,Set(__RINGTIMER=10) exten => 3005,n,Macro(exten-vm,novm,3005) exten => 3005,n,Hangup When I execute the Originate (AMI) with the argument Channel=local/2001, It rings

I have Asterisk 1.2.29, Zaptel 1.2.24 and Freepbx Setup for a queue up to 15 agents through a PRI line, it was working fine for more than 1 year, suddenly, when there is a load on the queue, the asterisk service disconnects and the calls are dropped. And the service starts again after few seconds, and so on. I am not using fax. I checked PRI by zttool and there are no alarms. The cdr logs

I am using asterisk version 1.2.6 with Zaptel version 1.2.5. I have a POTs line coming into a Digium TDM01B. It appears to not be getting CID at all. If I hook up a POTS phone to the line CID comes through fine. Inbound and outbound calls work fine but there is just no CID on inbound for this channel.The incoming route for the channel is Zaptel Channel 0. No DID or CID settings applied. My IP

Skipped content of type multipart/alternative-------------- next part -------------- asterisk1*CLI> soft hangup Zap/1-1 Requested Hangup on channel 'Zap/1-1' == Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on 'Zap/1-1' in macro 'exten-vm' == Spawn extension (ext-local, 220, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' --

Hello, Sometimes the call gets answered for 1 second, but actually the phone has not rang, it?s just the CDR, and asterisk hangup automatically, I cought the log of 1 call like this, I hope you can help me with this. My setup is : <vendor> ----SIP--? <Asterisk> ?----IAX2---? <Asterisk with Dhadi channels> Here: -- Executing [966505103150 at from-internal:1]

I have an Asterisk server it runs great with SIP phones, soft SIP phones (twinkle) and a soft SIP phone app on my Android phone but I am having problems getting two ATA boxes working. I have a Linksys PAP2T, it is unlocked and I have used them before with no problems. I was able to receive calls with from any local SIP phone or from my Link2VoIP connection via the Internet but it could not call

I am seeing the following in my log file (standard trixbox install). One seems to be complaining about an error in the dialplan but it won't tell me what file or what line. The other (maybe related) is complaining about a channel lock. How to do go about trying to figure out what the problem is and how to solve it? ---------------Logfile-------------------------------------------- Nov 14

Hi ALL Have installed asterisk@home 1.0 On FWD DID's, appears that 2 calls are generated to the inbound extention. I have confirmed this on a number of friends boxes also. Does anyone have a fix for this ? I set the DID simply to a custom context and it did the same... Anyone have a way to fix this ? Here is the output...... -- Accepting AUTHENTICATED call from 65.39.205.121, requested

HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound comes through. I have verified that the sounds are in the correct location and that asterisk:asterisk has

Hello List, I am having a strange issue with a trixbox system we installed for a client, and I would appreciate any help on this one. The issue is that occasionally when they go to answer an inbound call from the Sangoma A200 - there is no one there, and they are presented with dial tone. The calling party is hung up. A bit of background: The client actually has two systems install (one at

Hello all, I have been posting some questions about this problems that I cannot yet solve, but I think I have a better diagostic, so maybe someone can give me a clue why it is happenning. I have Asterisk + AMP configured as a PBX with a Customer Center Queue with 4 agents that login/logout dinamically. If there are no agents, queue timesout and gets derived to another queue that somebody

Hi, I set up a ring group. I would like for people who select a certain voice menu option to ring a list of extensions (I have just one extension in there at the moment) and if it doesn't answer to go to an extension's voice mail. I am using a version of asterisk from CVS, last updated a couple of weeks ago. This line in extensions_addtional.conf sends the call to ringgroup 3 if

hi,all when i send a call to other server use SIP trunk, i got this below, <--- SIP read from 222.46.18.52:5060 ---> SIP/2.0 403 Forbidden what's rong with is? > Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others. > Created by Mark Spencer <markster at digium.com> > Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for

I just had a user report that they called out to someone on a cell phone this morning, and after a short conversation, the call was dropped/lost. The person on the cell phone says that this is very rare, and would not suspect the dropped/lost call to be on their side. I have looked at the asterisk/full log as thoroughly as I can, and have pasted the lines which seem relevant to that call below.

Hi, I use: Trixbox-2.2.4 FreePBX-2.3.1.0 Asterisk-1.2.17 BRIstuffed-0.3.0-PRE-1y-e Zaptel-1.2.19 ..with two ISDN cards, often but occasionally the dial out failed but is possible to receive external call. My zapata.conf conf is: [trunkgroups] [channels] language=it context=from-pstn signalling=bri_cpe_ptmp rxwink=300 pridialplan=unknown prilocaldialplan=local switchtype=euroisdn

Hi, We have a PRI Trunk (physical E1) and we are getting some rather weird and very isolocated problems. On outbound calls to specific numbers, it would seem to me that DTMF from the remote side is affecting the local asterisk system. Basically what happens: - We make a OUTBOUND call via the PSTN (PRI Trunk) to a remote System - Remote Answers, and converse - Remote sends DTMF on their site to

Hello everybody, and first of all sorry for my poor English. I'm having trouble with Quadbri installed on Asterisk 1.2.17-BRIstuffed-0.3.0-PRE-1y-e. Everything is working fine, except calling to switched off or "out of coverage" cell phones. In this case I have to wait 40 seconds until Asterisk tell me that "all circuits are busy now" instead of receive cell phone

i have a prablom here, i want to send a call from A to B use sip trunk , the call can sended B,but not work to PSTN. the message from B server. help pls,what's rong? > > <--- SIP read from 192.168.0.176:5060 ---> > INVITE sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151>SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport

Dear All, I have an asterisk server version 1.2.12.1 along with trixbox and I am having this nasty problem, I have a TE200P and have an E1 pri attached to it and zttool says it's OK, I have configured the whole 31 channels into one group as follow: Zapata-auto.conf: callerid=asreceived signalling=pri_cpe switchtype=euroisdn context=from-zaptel group=0 channel=>1-15,17-31