similar to: Incomming call issue

Hi, I'm running two boxes side by side, identical specs and setup but with differing dialplans. Both are on ast/zap/libpri versions 1.0.9. Both boxes share the same folder for voicemail, exported via NFS from another file server. Everything was working fine for an extended period of time, until just recently when someone rebooted Box A. Now when I dial an extension associated with a SIP

I have a customer running Asterisk 1.4.22 and FreePBX 2.5.0 that is having problems with Voicemail. They can listen to their voicemail but on the weekend it stopped delivering messages via email. The only thing I can notice is that the permissions for the files on teh voicemail directories are created with no permissions at all. Here is the listing on one of the mailboxes: 4 -------rw- 1

Hi, I'm having trouble getting asterisk to report MWI to a Cisco CCME. I record a message in mailbox 29, but the subsequent MWI notifications I see continue to report no messages waiting. Are they reporting for the wrong mailbox? Is there some other option I have to set or change? I'm running asterisk-1.4.22 Since the mailbox is in [home] in voicemail.conf, I've tried things like

Hey all, I recently installed fedora core 5 and then installed samba. I however encountered a problem. The smbd thread would not start. Can anytone shead some light on why this might have happened. In the original installation of the OS I installed samba. However I encountered that same problem. The smbd would not start. I then tried installing samba from the source and still the same

Hi folks, I want to fix hole in my asterisk set up. I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN, Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go 'other' places. This senario works fine. Now the issue is someone else running a vocal or another SIP proxy can redirect his calls to my * as well. Those calls two will come through general

Hi, Would anyone have a clue on this issue? I'm running asterisk 1.4.13. Trying to get a WAV voicemail file attachment sent to my email address. Voicemail is working fine. Email notification of a new message works fine. However, when I set up voicemail.conf to have an attachment of the file sent to email I get an error message like this on the console: [Feb 17 13:33:37]

Hi, We can't read the messages in our mailbox always getting -- <SIP/tootaiAUDIO-00000001> Playing '/var/spool/asterisk/voicemail/default/100/Old/msg0002.slin' (language 'fr') [Oct 11 13:24:50] WARNING[26778]: app_voicemail.c:7802 play_message: Playback of message /var/spool/asterisk/voicemail/default/100/Old/msg0002 failed As you see Asterisk try to read

Hello, I'd like to take a look at how the taskPR package performs on our computational cluster, and this morning I've spend some time trying to do a test install of this package on one of our linux machines. Unfortunately I've been unsuccessful in completing the install, and I wondered if someone could please advise me. As I've noted I working on a linux machine, and it's

I have gotten some cisco VIP 12 and VIP 30 IP phones that I would like to use with asterisk, I have set them up using chan_skinny. The phones work well, except the only problem is that it is like the cisco phones are muted. When I talk on the cisco phones I can hear my self through the ear peice, but the person who I am calling can not hear me at all. I have tried various cisco phones from various

Thank you for your time. I've put my replies to your questions in-line, below. On Wednesday, June 07, 2017 10:19:41 AM Antony Stone wrote: > On Tuesday 06 June 2017 17:54:59 Mike Diehl wrote: > > > Hi all, > > > > I'm upgrading to Asterisk 13.14.0 x86_64. During my beta testing, I've > > discovered that my server crashes as soon as I leave a

Hi everyone, I've set up a little Asterisk system with a Digium TDM400P and everything works splendidly except for the messages callers leave. Every message that a caller leaves is very faint. I've already set volgain=6.0 in voicemail.conf, and that seems better, but to be at a good volume I estimate I may need to go up to 40.0. Is that reasonable? One interesting artifact is that at

PCIe endpoint framework provides APIs to implement PCIe endpoint function. This framework allows defining various PCIe endpoint function behaviors in software. This patch extend the framework for virtio pci device. The virtio is defined to communicate guest on virtual machine and host side. Advantage of the virtio is the efficiency of data transfer and the conciseness of implementation device

Hi All, I have an issue with the ODBC voicemail storage option with asterisk. All appears to work fine, however, I get several sql execute warnings. I was wondering if anyone out there could help me get to the bottom of what is causing this and how I could possibly go about rectifying it. The warning message we are getting is as follows: WARNING[30115]: app_voicemail.c:1280 delete_file: SQL

I am seeing lots of these in my logs and there are often a hundered or so imap/dovecat process running. I am running RC Core3. Can anyone shead some light on how to correct this ? Jun 16 08:38:24 jidmail dovecot(pam_unix)[27653]: check pass; user unknown Jun 16 08:38:24 jidmail dovecot(pam_unix)[27653]: authentication failure; logname= uid=0 euid=0 tty= ruser= rhost= Jun 16 08:38:24 jidmail

Hey guys, can you shead some light on this? I will copy my mgcp.conf and post below, but here is the problem. I can't get call waiting to work with my MGCP device. I already have one call going, and I can hear the second call come in, I flash over to it, but all I get is a dial tone, * puts the 1st call on mute/hold, but I never get the second, and it terminates. I flash back over and pick

Scenario, I got some very good help earlier from Joseph getting me up and started but I have a couple of small problems still. Setup: FC2 Asterisk 1.0 Zaptel 1.0, TMP400P, FXS Port 1 FXO Port 4 Analog dialout line and Analog handset plugged in. Problems: 1. Incoming calls work and the phone rings and can be answered no problems, (although I wouldn't mind being able to adjust the ring but

Hi all, I'm upgrading to Asterisk 13.14.0 x86_64. During my beta testing, I've discovered that my server crashes as soon as I leave a voicemail message. I'm using odbc voicemail storage as well as mysql dynamic configuration. I'm using unixODBC 2.3.2-r2 with myodbc 5.2.7-r1 I suspect that the odbc drivers are the problem. Is ther an alternative drive that I should be using?

Hi My plan is joining XP Pro client to authenticate via Samba 3.0 domain or workgroup. The problem is the XP client at first login interface didnt show logon location like to "This Computer" and "AJK-ITS" workgroup, so it cannot authenticate to Samba Server. Any help would be appreciated. I have configured those machines like these: on XP Pro client I've set:

Hi Steve, I am having this problem in which RxFax is still holding the file after receiving a complete fax. Somehow the zap channel is still active but on the fax client it was sent successfully. If you call the line it is still busy. Changed from phase 3 to 4 >>> MCF: 8c HDLC underflow in state 8 Changed from phase 4 to 3 Slow carrier up <<< DCN: fb DCN with final frame tag

Hi all, I have hit a wall configuring a TDM400, I have set these up before without issue but today I just can't seem to figure out what I am doing wrong. On an incoming call the following is produced in the Asterisk console with verbose 4 -- Starting simple switch on 'Zap/2-1' Mar 22 16:12:34 NOTICE[2051]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... Mar 22