similar to: ipphone voicemail problems

Displaying 20 results from an estimated 1000 matches similar to: "ipphone voicemail problems"

2004 May 13
0
ISDN & Voicemail: Strange Behaviour
Hi, whenever I include a "Ringing" Line in some Voicemail Extension I get an error when a call from the outside (via ISDN) comes in, but it works when an internal (SIP-phone) calls the extension. Here is my configuration for testing: ------------extensions.conf------------ [isdnext] ; strep external "101", our number and leave only extension exten =>
2004 May 21
0
unable to use EXEC in AGI
dear list if I use EXEC in an agi script I get the following doing EXEC VoiceMailMain -- AGI Script Executing Application: (VoiceMailMain) Options: ((null)) May 21 04:25:10 WARNING[1209214400]: chan_phone.c:422 phone_read: Error reading: Resource temporarily unavailable May 21 04:25:10 WARNING[1209214400]: res_adsi.c:205 __adsi_transmit_messages: Un able to send CAS May 21 04:25:10
2005 Jul 21
1
IAXY & Voicemailmain problem
I have the original version of the IAXY. I had it laying around collecting dust, now Im actually putting it to use. When I call my voicemail extension (8500), Before I get the voice prompts from the voicemail app, I hear tones that sound like the caller id tones that are heard when montoring a phone call. While watching my Asterisk CLI, I see this error at the sound of each tone: Jul 21 23:06:03
2006 Jun 08
0
ipPhone and ATA with UPNP
Hello, I'm looking for ipPhone and ATA with UPNP and perhaps also STUN auto provisioning via https or . G729 If someone know a good product.. Thanks Laurent -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060608/efca9a57/attachment.htm
2010 Feb 02
0
Issue when reloading
Hello list! I?m having an issue when reloading Asterisk, I?ve had this problem in Asterisk 1.6.1.6 so I upgrade to 1.6.2.1 version, but I still have the same error. For example, I send a "reload" in Asterisk CLI and this is the output: isb152*CLI> reload == Parsing '/etc/asterisk/extconfig.conf': == Found == Parsing '/etc/asterisk/manager.conf': == Found
2008 Jul 23
1
1.4.21.2: Linking res_crypto causes segmentation fault.
Hi, i tried to compile Asterisk 1.4.21.2 on a server which i have been using with many previous Asterisk versions, without any problems. But with 1.4.21.2 it failed: ---------------------------------- [CC] res_adsi.c -> res_adsi.o [LD] res_adsi.o -> res_adsi.so [CC] res_agi.c -> res_agi.o [LD] res_agi.o -> res_agi.so [CC] res_clioriginate.c -> res_clioriginate.o
2006 Feb 28
0
Asterisk hangs up - h323
This is third time today that my Asterisk hangs up. It seams that I have problems with h323. I'm using ooh323 from Asterisk add-ons. I have the following configuration Asterisk 1.2.1 Asterisk-addons 1.2.1 Fedora Core 4 I'm using SIP phones and h323 trunk to my VoIP provider Like I said this is third time today that he hang's up. First time, I came at work and Asterisk was down.
2007 Jan 09
0
Asterisk + 7910 + Skinny Reset
I have a bunch of 7910's that I managed to get registered with Asterisk 1.2.14: managed5*CLI> skinny show devices Name DeviceId IP TypeId R Model NL -------------------- ---------------- --------------- ------ - ------ -- test7 SEP0004C1878F8E 192.168.0.226 6 Y 7910 1 The problem is that the phone resets when I attempt to make a call from it or place a call to it. If I pick up I have
2009 Aug 21
3
Core dump gets created while accessing voicemail
Hi ALL, When i was accessing the voice message it suddenly goes dead and after that i couldn't able to retrieve the voicemessage again from my mailbox . This happens once in a while for any configured mailboxes I am using the following system configuration. asterisk 1.4.22.1 odbc storage of voicemail messages centos 5.2 64bit unixODBC-2.2.11-7.1 mysql-connector-odbc-3.51.12-2.2
2006 Mar 13
2
Unknown signalling method 'pri_cpe'
Anyone have any idea what this is talking about. Here is my zapata.conf [channels] switchtype=5ess signalling=pri_cpe echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived group=1 context=default musiconhold=default faxdetect=incoming channel => 1-23 Here is my zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 # set this to 1-15,17-31 for E1 dchan=24 # set this to 16 for
2004 Jan 14
1
Skinny behind NAT?
Can skinny work behind NAT? I have a Cisco 7910 using SCCP behind NAT that has one way audio. The called party cannot hear the calling party who's using the 7910. skinny.conf ; ; Skinny Configuration for Asterisk ; [general] port = 2000 ; Port to bind to, default tcp/2000 bindaddr = 0.0.0.0 ; Address to bind to dateFormat = M-D-Y ; M,D,Y in any order (5 chars max)
2005 Jul 18
0
Crash on reload only with autoload=no
Hi, I've been having a little problem with my asterisk servers, I have 4 identical asterisk servers setup (same hardware, same OS, same config). Once in a while (once or twice a day) one of the server crashes on the cron job reload. But I realized this only happens on 3 of the 4 servers. Tried to spot the difference between that one server that wasn't crashing. The difference I found was
2005 Aug 14
1
ogg causing me heart burn
Dear forum, I have a install of asterisk using AMP. I followed the install guide off the AMP site. http://amp.coalescentsystems.ca/docs/AMP_Installation_Guide_v1.4.pdf When I start using amportal start or asterisk -ccccv I received this in my log. The last line is that ogg failed. I have found nothing on the web about this, and I am not even sure where to start troubleshooting. Any help
2005 Jan 23
0
Upgrade to the newest cvs now asterisk will not start
Hello group I just update to the newest CVS now I'm not able to get asterisk to start. No error during the make or make install I did a make clean before the make;make install Any help would be great!!!! Here is the output asterisk -vvvvvgcd Parsing /etc/asterisk/asterisk.conf Parsing /etc/asterisk/extconfig.conf == Binding realtime_ext to mysql/realtime/extensions_table == Binding
2005 Jan 23
0
Upgrade to the newest cvs now asterisk will notstart
Looking at the error I tried moving chan_modem* out of the modules folder and asterisk started and its working again... Not sure what changed in the chan_modem_i4l.so but removing it from the folder fixed my problem. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Eric Hall Sent: Sunday, January 23, 2005
2004 Aug 24
2
Voicemail & "Couldn't read username" error
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
2004 Aug 26
1
Newbie needs help - Dev_Kit_Lite installation problem
Installing DevkitLite hardware (Very similar to John Lange's post on Tue Oct 08 2002) I cannot get anything to work on the phone connected to the s100u. I dont know what to do. Can someone please help me? I used the sample configuration files from digium documentaion that was supposed to be "sane" defaults for the kit. Very similar to John Lange's post on Tue Oct 08 2002 Here
2004 May 05
1
MySQL and VoiceMail again
Hi, At first I would like to express how much I like Asterisk. Amazing product. I compiled Asterisk with mySQL support for CDR and Voicemail. Everything seems to be fine, I can see that Asterisk connects to mysql and logs CDRs. I can also see that the VOicemail app is also logged in, however I can not access any mailboxes. Similar messages to others, app_voicemail.c:3011 vm_execmain: Couldn't
2004 Aug 08
1
No Sound and Jungle:
Hi everyone, I am running asterisk on red hat linux 9 box. The sound card is Intel 82801db AC' 97 audio and the module is i810_audio. It runs well with other applications like xmms and the standard tests deliver a sound . I have also tried to record voice and that works well too. 1-)Now when i run asterisk and i dial out an extension to play any sound there is none. The same thing
2004 Sep 28
1
Newbie 2 PBX VOIP, protocol ?'s using Cisco 827 7910
I am replacing a dead pbx with *. There are four lines I will be using. There is a Cisco 827-4v already in place so I will move the lines from the pbx to it. I am working with Cisco 7910 phones and I understand they use the Skinny/SCCP protocol. I am not sure if I should use chan_skinny or chan_sccp? However my main question is with communication. Do I need to use the same protocol between the