similar to: asterisk conference error/bug?

Hellos, I am having issues with my meetme conferencing. When I dial the conferencing number, It hangs after a few seconds.I have read somewhere that I need to enable ztdummy, which I have done but still no changes. Here is my log ~=~=~=~=~=~=~=~=~=~= PuTTY log 2009.08.12 18:44:43 =~=~=~=~=~=~=~=~=~=~=~= -- Executing [1;36;40mMacro[0;37;40m("[1;35;40mSIP/1215-fc5b[0;37;40m",

Hello, To do some testing with Asterisk installed the latest Asterisk @ Home in a Vmware system. All worked fine, I can access the web interface (AMP). I have setup the extention and X-Lite softphone according to the description in the Wike (http://www.voip-info.org/wiki-Asterisk+phone+xten+xlite). I can dial 200 (the softphone extention) and 1234 and they connect (the softphone shows this, as

After doing an update from SUSE 9.2 to 9.3 and Checking out the latest from CVS, Asterisk crashes on startup with an apparent MySQL (res_config_register) error: # asterisk -vvvgc > asterisk_startup_error1.log asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_config_mysql.so: un defined symbol: ast_cust_config_register The log is shown below. I've seen the posts

On 1.6.0.18-rc3 using app_fax.so, spandsp-0.0.5, anytime I use SendFAX asterisk restarts: [Nov 15 19:00:36] VERBOSE[17013] logger.c: -- Executing [s at fax-tx-test:1] ESC[1;36;40mNoOpESC[0;37;40m("ESC[1;35;40mSIP/nhi-rive rside-sip-00000000ESC[0;37;40m", "ESC[1;35;40mContext fax-tx-testESC[0;37;40m") in new stack [Nov 15 19:00:36] VERBOSE[17013] logger.c: --

I'm using Asterisk 1.0.4 with AMP and Broad Voice. I have that with only 5 XTen Lite phones. I'm able to call / etc with internal phones just fine. I can call outside Vonage Numbers, and other BroadVoice Numbers. I have vonage where I live (626) and can call that fine. However, other 626 numbers I get similar errors as below. However, everytime, I try to call cell phones, and or

hello, I have been using asterisk now for about 2 years now on a RH8.0 it is our main call gateway. I have on the box 3 T1 TDM cards connected to 2 Rhino channel banks (FXS) and 1 CAC Access bank I (FXO) with so many softphones and ATA 186s. It has been working good till today some few hours ago. i just discovered that there were no dialtone on the phones. Asterisk did not spit out any error, it

Hi, I am currently trying out the asterisk@home (version 1) release of Asterisk, and I want to configure it as follows: Calls from regular telephony network (PSTN) come in through my VoIP provider over SIP and outgoing calls to the PSTN should be routed through the ViOP provider onto the PSTN network. I thus have no direct PSTN connection, but only a SIP connection. Incomming calls

I am having a problem with SpanDSP. What happens is when I send a fax to SpanDSP the fax message seems to fail in the training phase. I think it's a timing error, however I have no idea about how to rectify the problem. I have included a copy of the log below. I am using a Digium TDM-400P card with 2 x FXO ports and 2 x FXS ports. The fax is connected to one of the FXS ports (Zap3). The

I am learning agi scripting using php. I m using phpagi 2.x on asterisk 1.2. I hve written a simple script that reads out the callerid using flite. My problem is that I seems the script is not getting the callerID. Bellow is the script _________________ #!/usr/bin/php -q <?php /** * @package phpAGI_examples * @version 2.0 */ set_time_limit(30);

Greetings list, We are having a puzzle with * (asteriskathome 0.5) and SIP phones (SPA2000 ATA's). If callwaiting is enabled, everything (including call waiting) is normal. If callwaiting is turned off, the phone will not accept incoming calls and the call goes straight to whatever is programmed for the busy voicemail response. It doesn't matter whether reinvite is on or off, or

Hi I'm running safe_asterisk, but get core-files in /tmp - how do I debug them ? I know gdb asterisk core.12370 and bt full But it didn't show anything usefull for me. Can anyone help me ? (Running asterisk 1.0.2 with ast_data /Hans-Henrik ----------------- Last from bt full: priority=200, callerid=0x81b8e90 "Dial", action=1134845864) at pbx.c:1384 e =

Hi! We have a SIP server with a TE410P card with asterisk version Asterisk CVS-D2005.02.12.14.37.11-04/13/05-16:14:03. Sometime the calls get disconnected with now reason and the users get a busy signal. The log file show this for one of the calls that got disconnected: Aug 31 22:51:53 VERBOSE[3911]: -- Accepting call from '46362302' to '36917474' on channel 0/5, span 1 Aug

Hi, I am try to configure Asterisk as PBX system with two interfaces as shown below. One interface pointing to the local subnet with a SIP phone and another interface pointing to the external ISP SIP Sever. SJPhone(X.X.141.32)<--------->(Y.Y.47.149)local-intf-|Asterisk|external- intf(Z.Z.247.106)<-------->(w.w.158.26)ISP-SIP-Server----OutsideWorld I am able to setup a call from the

When trying to make outbound calls I am getting the Warning: File app_dial.c line 313 (wait_for_answer) Unable to forward voice. When making the call it attempts to dial (pounds are actually numbers but replaced to not show numbers we are dialing): Executing Dial("Sip/donas-bd7b", Zap/g1/1##########") in new stack Called g1/1########## Channel 1, span 1 got hangup **Above

Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work inbound or out bound. I always get 'all circuits busy'. May 12 22:27:05 VERBOSE[2442]: -- Executing [1;36;40mDial[0;37;40m("[1;35;40mSIP/101-ad89[0;37;40m",

Hellos, I am looking for the sounds used in this ivr example http://www.voip-info.org/wiki/view/Asterisk+Tips+follow+me. The one with 6900. Any assistance is welcome. -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php

Hellos I am using freepbx and asterisk. I am writing an AGI script to edit the values in findmefollow table. The script will enable users to delete and add follow me numbers from their phones. I want it to enable users enable/disable follow me. I can't seem to find a value in the database that deals with enabling/disabling followme. Please help -- Best Regards, James Mutuku Ndeti Agile

I am running asterisk and I want to achieve the following scenario My goal in the end is to achieve the scenario (example using extension A and Extension B) 1. Extension A has a line apperance of 4(4 calls can ring on it). 2. Extension B calls extension A(which is busy on one of the lines). 3. Extension A sees the second light blinking and hears the beeps (currently working). 4. Extension B is

Hello, I have 2 sites. One(Site 1) has an asterisk PBx and the Other(site 2) has 2 remote soft phones. The latency btw both sites is btw 500ms-700ms. I know this is a shot in the dark...but are there ways of improving the voice quality for the remote calls(btw site 1 and site 2), Other than increasing bandwidth? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994

Hellos, My voip carrier has assigned me a IP based account...where they only give me the IP to call through. I have setup the dial plan exten => _7XXX.,1,Answer() exten => _7XXX.,2,vmauthenticate(${CALLERID(number)}) exten => _7XXX.,3,Dial(SIP/${EXTEN:1}@Y.Y.Y.Y) exten => _7XXX.,4,Hungup() Where Y.Y.Y.Y is the assigned IP. After Dialing I asterisk logs the error SIP/Y.Y.Y.Y-35dc