similar to: Problem with hangup

Hello, we have set up e164.lu as a test zone, as the delegation for 2.5.3.e164.arpa hasn't been completed yet. For all those who want to call the numbers currently availble directly via SIP, please use the zone name in your enum.conf. If you decide to use the zone, please tell me at mstorck@luxadmin.org, so as soon as the 2.5.3.e164.arpa zone is ready, I will mail you, so you may disable

Hello, I'm looking for a similar feature... Dial a number via ZAP/g1 after the line gets answered wait 10 seconds send DTMF Regards, Marc -- Network Manager Marc Storck LuxAdmin.Org mstorck@luxadmin.org Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352

Hello, one of the numbers where historically configured to act the following way: 123456: Ring All Desks 123456-1: Ring Desk 1 123456-2: Ring Desk 2 ... (I think you get the idea) Configuring asterisk to do the same isn't that hard, but I now have one problem, with users calling that number from PSTN. Those particular users go off-hook and start dialing the number. The ZAP Channel claims

Hello, I have a weird problem. My * server, a Pentium Celeron 1200 with 512 MB Ram and a Digium E100P card, is performing very well for IAX2, SIP and ZAP communication. There is no delay in transcoding, no packet loss etc etc. Now I added DUNDi, and I added +/- 8 peers in the dundi-test context and 1 peer in the GPA-bound e164 context. My server shows all but 1 peer as OK. DUNDi Ping times

Hello, more and more operators in Europe offer music instead of ring tunes. E.g. instead of the 400 Hz or whatever tunes, the caller will hear J-Lo, or Mozart.... Currently I will have to answer the line to do that. Is there a way to do this with asterisk? Regards, Marc -- CTO Marc Storck MS Networks SA mstorck@luxadmin.org Internet Service

Hello, we're in process of testing an interconnection with a trans-european carrier. But the carrier wants the SIP signalling to server 1 and the RTP stream to server 2. How do I configure asterisk to work with that type of installation. It seems they are using NexTone as SIP Signaling and RTP servers. Can someone help me??? Regards, Marc -- CTO Marc Storck

There was a post in the 'wiki' for an application to provide an outcall when there is a voicemail is left on asterisk. I am having a problem that this application will only work if the caller presses the pound sign at the end of recording. As most people just hang up, this application isn't working. Can any offer suggestions to accomplish this out call?

I don't think it's the snom, (the break key is set to "off") the "#" key is not being interpereted by the PBX as an attempt to initiate a transfer. Is this an error in my extensions.conf? Brian > >Message: 4 >Date: Wed, 15 Dec 2004 19:39:39 -0500 >From: Info <info@idatasys.com> >Subject: Re: [Asterisk-Users] Help with transferring a second call

Hi, I am using smbmount (3.0.0) on a debian GNU/Linux computer to mount a Windows 98 folder. I edit C++ files from the Linux computer and compile them on the Windows computer with VC++ 6.0. What I don't understand is the different behavior between when a file is edited with an external editor on windows (let's say wordpad) and when edited with an external editor via Samba (let's say

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",

> 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've been using my voicepulse connect number for over a month now, but today it simply won't connect. My partner and I each have a number, both are mapped in my iax.conf and extensions.conf files. This has been working fine. Today, either number gives this message: Feb 16 21:53:14 NOTICE[4330]: chan_iax2.c:5757 socket_read: Rejected connect attempt from 66.234.228.170, request

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

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.

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 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

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

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