similar to: SpanDSP - Training failed (convergence failed) error

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

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

I am trying to connect audio through HDMI on a config. aplay - l gives: **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: VT1708B Analog [VT1708B Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 0: NVidia [HDA NVidia], device 3: NVIDIA HDMI [NVIDIA HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 So I change my

I am getting the following information and asterisk 1.0.7 is not answering the call. Any ideas? jerry ------------------ Sip read: INVITE sip:2828;phone-context=cdp.udp@qg.com:5060;maddr=161.49.198.102;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From: <sip:3173241052;phone-context=+1@qg.com;user=phone>;tag=c22da8c0-13c4-429efa71-657b8da-2ed To:

All, I am connecting to a CS 1000 nortel PBX. I can call out, I have limited success with call in. I get debug traffic that a call is coming in but I get the message "Unable to create/find channel". I was expecting that incoming calls over the trunk would be handled from my sip definition and goto the nortel context. It is not. Below is the actual incoming call debug information. I am

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, I'm pulling my hair down and getting bold :-) ..... I have Asterisk between Panasonic KXTD816 and Euroisdn in Italy (beronet octobri and bristuff Asterisk).... I'm trying to do just plain transfer of call from pbx to ISDN through Asterisk... It seems like PBX hangsup, when call is progressing with no apparent reason. I'd kindly ask for any advice or some working example for

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

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",

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",

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

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

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",

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

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

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: --

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 downloaded spandsp0.0.6pre17 I download http://sf.net/projects/agx-ast-addons for app_txfax and found trunk/app_fax to be newer so I used that. spandsp compiled fine. app_fax compiled when loading I get: [Apr 5 08:55:54] ^[[1;31;40mWARNING^[[0;37;40m[7505]: ^[[1;37;40mloader.c^[[0;37;40m:^[[1;37;40m433^[[0;37;40m ^[[1;37;40mload_dynamic_module^[[0;37;40m: Error loading module

Hi, I'm running into issues receiving faxes which, from what I have read, may be caused by frame slips. While I can find many posts saying to investigate it, I can't find any that describe *how* to debug the problem. Tried searching this list as well to no avail. Any pointers would be greatly appreciated. FYI, I'm running wbel, AMP 1.04, spandsp 2pre4. Faxing to a pstn on a

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