similar to: strange voicemail issue

Hello I just settup a realtime mysql table for sip_peers. All peers (friends) is autenticateing but when i want to initiate a call between them i got the following error. Someone have some ideea? Thank you. ---<Cut Here>--- pbx*CLI>console dial 1014 == Console is full duplex -- Executing [1014@default:1] Dial("OSS/dsp", "SIP/1014|40|t") in new stack

Is there a memory leak in asterisk 1.4? The other day with asterisk 1.4.0 I noticed that top was reporting a RES of 106 meg for the asterisk process. Restarting the process brought it down to more like 4 meg, but it grew over time to be 20+. So yesterday morning I upgraded to 1.4.4 in case this is something that had been addressed. Again I started with a RES of like 4meg or so, but this

Hi all, We have an asterisk installation here that uses realtime for voicemails through ODBC. It works very well except that every now and then (ie four or five days or so) it breaks. I have included a log from the CLI of the most recent break, it looks like this: ---------------- Start of output -- Executing Dial("SIP/sip.ict.ru.ac.za-b7721690",

Hello, I have an E1 connected to an * server, which takes incoming calls and verifies the existance of the called number in our internal E164 tree. Now there is a number that exists on one of the servers, but the phone has registered itself, so the dial plan executes an hangup. This hangup however is not transmitted to the E1, the calling party hears no dial tone, but also no hangup or

> Well that didn't work....I now get this error > > > Jan 12 16:56:21 NOTICE[4989]: app_dial.c:746 dial_exec: Unable to > create > channel of type 'SIP' > == Everyone is busy/congested at this time > -- Executing VoiceMail("IAX2/iaxfwd@65.39.205.121:4569/5", "b") in > new > stackJan 12 16:56:21 WARNING[4989]:

I (sometimes) get this error message: WARNING[17191]: app_voicemail.c:2411 leave_voicemail: No entry in voicemail config file for 'irstname.lastname' I can see the value of the argument is "firstname.lastname" when this line executes in the std-exten macro: exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable... But the error message drops the first character. It

Hello steve. Here is a patch I wrote for app_voicemail.c which does exactly as you describe. When the outgoing message is playing, if the listener hits the "*" key, they're prompted for a mailbox and password, whereupon they can check their voicemail as if they were using the internal phone. I found no other way of doing this. If you patch your app_voicemail.c, I have V1.44 from

I've been continuing to experiment with MySQL. I'm having absolutely no luck getting asterisk to read voicemail configuration data and mailbox configuration data from mysql tables instead of from voicemail.conf. The default Asterisk setup that reads from voicemail.conf and extensions.conf works fine. I'm using Asterisk CVS-v1-0-12/12/04-15:58:29 on a Whitebox Enterprise Linux box.

I have been testing with asterisk 1.4.11 and have found a segmentation fault while using voicemail. It happens when I try to forward a voicemail. As soon as I press the option the server crashes. I ran asterisk up inside gdb and got the following stack trace ==================================================================== Program received signal SIGSEGV, Segmentation fault. 0x00140adf in

Hi All; Already in the voicemail.conf file, I added the extension 500 and kindly find below my voicemail configuration: [Internal] 0 => 1234,Gama Operator,Operator at gama.com 500 => 1234,Operator,Operator at gama.com 501 => 1234,Employer Name,employer_email at gama.com 502 => 1234,Employer Name,employer_email at gama.com Asterisk version is 1.8 and currently I am getting this

Hi all, I am trying to write a program that sends SIP requests to asterisk. My aim is to make asterisk record as many voicemails it can at a time. The design of the program is like this: There are two processes: One main process and a child process (No flames pls. I have very little idea about pthreads and dl modules) The main program asks the user to input the number of test instances. When

hi all, I can not get voicemail working in realtime with asterisk-1.2.6. extconfig.conf is correct voicemail => odbc,asterisk,voicemail_users i am getting the fallowing error Executing Answer("SIP/xx.xx.xx.xxx-0a02e1c0", "") in new stack -- Executing Set("SIP/xx.xx.xxx-0a02e1c0", "foo=102") in new stack -- Executing

Hi I do have asterisk running in real time I do want to add voicemail to real time. I did follow : http://www.voip-info.org/wiki-Asterisk+RealTime+Voicemail However when I do try to make a voicemail I do get : [Nov 20 12:17:04] NOTICE[16501]: chan_sip.c:5506 process_sdp: No compatible codecs, not accepting this offer! -- Executing [999alijawad at a2billing:1]

Hello, We recently upgraded our Asterisk box from 1.4 to 1.6.1. In both versions of voicemail you can press 3 for advanced options, 5 to leave a message and enter an extension to leave a voicemail. This feature worked fine under 1.4. Now under 1.6.1 all the prompts are the same but when you enter the extension it reads back the extension (or says the recorded name if present) then goes straight

Hi. I am looking for the proper syntax for the Cut application. I am working on a "Feature Code" extension that drops a caller directly into a voicemail box. Here is what I have: exten => _55.,1,Answer() exten => _55.,2,Cut(VMEXT=EXTEN|55|2) exten => _55.,3,Voicemail(u${VMEXT}) exten => _55.,4,Hangup() When I dial 551100, the system tries to process this but I get

Hi All, The situation: When I dial into VoiceMailMain(@context), put in my VM # 1001 and Password 1001, no problem, but at the voicemail main audio prompt (Alison), when I ?press 3 for advanced options? then ?press 5 to leave a message? I put in a mailbox number 1002 within the same [context], but VoiceMailMain looks for the mailbox in the [default] context and will not recognize the mailbox I?m

I tried to share my spool directory so I could get monitored calls, and now this error comes up when I try to leave a message in any of my voicemail boxes. Mar 24 12:48:35 WARNING[344081]: app_voicemail.c:1488 leave_voicemail: Error opening text file for o utput -- Recording the message Mar 24 12:48:35 WARNING[344081]: file.c:906 ast_writefile: Unable to open file /var/spool/asterisk/v

Hi list, in asterisk 1.4 and maybe earlier it was possible to use voicemail system with mailboxes starting with some special characters like *. The line in voicemail.conf was like this: *123 => , AB,,,tz=cet|attach=no| Calling exten => s,n,Voicemail(*123,su) is working in asterisk 1.4. In Asterisk 1.8 the above scenario is not working any more. The Voicemail application reports an

Hi, I input "0203#" after "mailbox?" voice prompt from Voicemail cmd on extensions.conf such as exten => 0021,1,Ringing exten => 0021,2,Wait(1) exten => 0021,3,Voicemail exten => 0021,4,Hangup *CLI> -- Executing [0021 at sip:1] Ringing("SIP/0103-09a308b0", "") in new stack -- Executing [0021 at sip:2]

Dear all, I've configured Asterisk Voicemail, but after some tests I realised that when some call is sent to the voicemail of someone which username begins with "j" letter, Asterisk gives me the error: WARNING[5865]: app_voicemail.c:2412 leave_voicemail: No entry in voicemail config file for 'ohn' (for a called user named john, for example) Is this some kind of