similar to: VoiceMail Password problems

Hi, I am setting up an asterisk server for testing purposes and cannot get voicemail to work at all. My host OS is Linux From Scratch 6.3 and the asterisk software versions I built are zaptel-1.4.5.1 and asterisk-1.4.12. I am using the Ekiga softphone on my Ubuntu desktop machine. My asterisk server and client phone are on different computers but are on the same LAN, i.e. no NAT. I have an

Hi I have the following configuration at home one ZAPTEL interface connecting to an FXO card and two SIP UAs connecting to asterisk locally. I have configured extensions.conf such that dialing 9 on the SIP phones allows me to dial an outbound number via the FXO interface . Works fine. What's not working is that pressing 9 should causes either GS BT-100 phone to reacquire a dialtone

Hi all, I am new to asterisk and am looking for a voip provider that supports asterisk. I am aware that their are several vendors to choose from. Any opinions on the best one? thanks Burak Balasaygun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060128/0a97a302/attachment.htm

I am having probelms connecting to voicepulse this morning. Is anybody else having issues.. burak

Hi, I just setup my * with digium. I started testing voicemail first between atas, and i am not sure why it is not prompting me any when the call is not answered or if busy. i only get continuous ringback and the following message: asterisk*CLI> -- Executing Dial("SIP/6882332-1697", "SIP/5104112978|20|tr") in new stack -- Called 5104112978 --

I've google for hours trying to find a discussion of a similar problem as the one I'm having, so forgive me if this has come up before. If it has, please point me in the right direction! The problem occurs when a caller (A) is transferred by an intermediary party (B) to voicemail (Voicemail or VoicemailMain), either directly or by being taken to voicemail when the callee (C) doesn't

Hi, I have Asterisk running with the VoiceMail. Using the latest CVS. I have my extensions.conf setup so that if a SIP caller dials *99 the VoicemailMain() as follows: exten => *99,1,Wait(1) exten => *99,2,VoicemailMain() A couple days ago I installed the MySQL/Voicemail support described at http://www.voip-info.org/wiki-Asterisk+voicemail+database Now for some reason

Hi, I am using the Grandstream HandyTone 486 as a SIP Adapter with a regular phone. Asterisk configuration is listed below. When I attempt to place a H.323 call, I receive the following errors: - Executing Dial("SIP/2000-3029", "OH323/##########@xxx.xxx.xxx.xx:1720|20") in new stack Aug 13 09:13:03 WARNING[20497]: channel.c:1806 ast_request: No translator path exists

Hi I have just been struggling for four days to get SIP working and now as I created a voicemail box, Asterisk has become very unstable and it can't bridge SIP phone to SIP provider calls anymore. Calling internally from one SIP phone to another works fine. Calling internally from a SIP phone to an analog phone on a Zap channel and vice versa works fine. Incoming PSTN calls delivered to

Martijn, I don't use any metadata when encoding and decoding. When I call *FLAC__StreamDecoderStateString[FLAC__stream_decoder_get_state(m_decoder)] * * * it returns FLAC__STREAM_DECODER_SEARCH_FOR_METADATA enum value. Is it an error ? 2013/7/2 Burak Or?un ?zkablan <borcunozkablan at gmail.com> > Hi again, > > I can not solve problem. I want to mention my source code, so

Hello, Asterisk 1.4.26.2, FreeBSD 8.0-RELEASE We have a very simple setup, using SIP softphones and a simple diaplan as follows in the examples below. When I dial the 700 extension it asks me for the extension and password, and it always says "login incorrect". The mail system send the email ok and Ekiga shows that I have vaoicemail, so the only thing that is failing is the actual

Hi again, I can not solve problem. I want to mention my source code, so you may answer easily. This is decoder init stream function. *FLAC__stream_decoder_init_stream(m_decoder, decoderReadCallback, NULL, NULL, NULL, NULL, decoderWriteCallback, NULL, decoderErrorCallback, input_pile_array);* * * Then, callbacks * * /// \brief read callback function of decoder FLAC__StreamDecoderReadStatus

Hi, I am getting the following error when I attempt to listen to voice messages by dialing 9999 (I can hear nothing): --Executing VoiceMailMain ("SIP/2092-6918", "2092") in new stack --Playing 'vm-password' (language 'en') WARNING: app_voicemail.c:4922 vm_authentication: Unable to read password. I read in previous posts that this may be to do with the dtmf

Hi list, i've problems with my * server and the 4 phones which are linked to it. i've 2 grandstream bt100 with the firmware upgraded to 101, a wi-fi phone (i don't know its brand) and another ip phone i don't know its brand. with this sip.conf : [general] port = 5060 bindaddr = 192.168.100.229 context = default ;x changed from default to sip localnet = 192.168.100.0/24

If my extension is 22, and voice mail access number is 909, then with exten => 909,1,voicemailmain(s22) I can access voice mail 22, without number and password prompt. But, I want that every extension can access its voice mail without number and password. So, when I put exent => 909,1,voicemailmain(${calleridnum}) voicemail want only password. I want to eliminate password too, so when I

I'm trying to get VoiceMailMain() to work properly, but it refuses. : ( I am using IMAP_STORAGE, which is functioning fine now... My voicemail.conf user line: 6000 => 1234,Brendan's Mailbox,,,imapuser=brendanmartens at crosscomm.net| imappassword=password 6000 => d,Brendan Martens My voicemail extension in extensions.conf: exten => 700,1,VoiceMailMain() And the output on

While setting my first couple IP phones, I set their voicemail buttons to an extension that runs VoicemailMain. exten => 8500,1,Wait(1) ; voicemail exten => 8500,2,VoicemailMain ; exten => 8500,3,Hangup ; I would like to be able to pass the mailbox number allowing each phone to go in directly but I'd rather tno have

I'll top-post this one because it wasn't sent to the mailinglist but to me. Please reply to list next time. I assume you mean the main.c files in the encode and decode directory under examples. I can't really determine the root cause of your problem with this information, but I think you're trying to feed the decoder blocks that are incomplete. The LOST SYNC error is usually

I've setup the voicemail that auths against the mysql db. Now, everything works ok, except voicemail() calls fail with Sep 26 18:09:34 WARNING[157070336]: app_voicemail.c:1517 leave_voicemail: No entry in voicemail config file for '' all my users are in 'sip' voicemail context, but adding context to it: voicemail(@sip) doesn't help.. while if I put a vmbox # to it, it

Sorry, I am newbie. Sample codes are from https://github.com/oneman/libflac/tree/master/examples/cpp. I used FLAC__stream_decoder_process_single function but it still gives exception. Maybe I could not control read callback, you're right. I will check it and write result in this thread. Thanks for help. 2013/7/1 Martijn van Beurden <mvanb1 at gmail.com> > I'll top-post this