similar to: new voicemail messages

Displaying 20 results from an estimated 7000 matches similar to: "new voicemail messages"

2003 Jun 30
0
stuck channel
I'm getting this intermittent problem, sometimes a zap channel gets stuck after a call. Below is a snapshot of the channel. Any ideas what can be happening? Name: Zap/1-1 Type: Zap UniqueID: 1056988772.10 Caller ID: (N/A) DNID Digits: (N/A) State: Up (6) Rings: 0 NativeFormat: 68 WriteFormat: 4 ReadFormat: 4 1st File Descriptor:
2004 Dec 11
0
Cisco 7960 and Asterisk...not working....
Sorry if this comes in twice. Wasn't subscribed first time :-( Anyone help me here...... It worked once :-( I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22 I have the 7690 with a SIP iamge (Whatever latest is ) I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server.
2004 Dec 29
0
Channel Zap/4-1 in prering state
Does anyone kmow what these errors mean or how they can be fixed. I'm using asterisk on a Fedora Core 2 box with a TDM400P with 2 fxo and 2 fxs ports. Dec 29 17:17:52 WARNING[6019]: chan_zap.c:5469 ss_thread: Channel Zap/4-1 in prering state, but I have nothing to do. Terminating simple switch, should be restarted by the actual ring. -- Hungup 'Zap/4-1' == Starting post
2004 Dec 21
2
CallerID returned with error on channel 'Zap/4-1'
I am using version: CVS-v1-0-12/13/04-18:46:23 with a TDM400p (2 fxo, 2 fxs ports) and I keep getting errors along with phantom calls: Dec 21 16:02:07 NOTICE[5872]: chan_zap.c:5363 ss_thread: Got event 17 (Polarity Reversal)... Dec 21 16:02:14 WARNING[5872]: chan_zap.c:5434 ss_thread: CallerID returned with error on channel 'Zap/4-1' my analog phone reads caller ID info fine when
2004 Jul 05
2
fax detection and X100P
Hi everybody I am having problem detecting fax with my X100P. I have RedHat 8 as OS and an X100P and a TDM400P. The X100P being plugged into PSTN. I have successfully installed tiff-v3.5.7 and spandsp-0.0.1 and also patched Asterisk wthout problem. Here is my zapata.conf file context=cda signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5
2004 Dec 13
1
Repost: Cisco 7960 and Asterisk...not working....
Anyone help me here? I am a newbie so be gentle ;-)...... It worked once and then I played with the configs. I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22 I have the 7690 with a SIP iamge (Whatever latest is ) I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server. Here
2005 Jan 27
1
Stumped by BroadVoice SIP
Hello guys. I am a fairly new user to Asterisk, and I'm just having a tough time. My goal is to set up a VOIP PBX. I have signed up with a BroadVoice number, and I have three systems with SIP phones. The PBX and the SIP phones are all behind a Cisco PIX running NAT. I am using Asterisk CVS version from yesterday. I also tried 1.0.3 with little luck. The SIP phones are two X-Lites on
2003 Jul 22
3
busydetect and random hangups
Hi, I'm having random hangup problems with zap channels. If I turn busydetect off in Zapata.conf, * fails completely to detect a user hangup in the middle of a script. On the other hand, if I turn it on, everything works much better, but long calls tend to be hung up without a motive. Any other parameter that I can try? Any #define that I can tweak and recompile? Will callprogress
2005 Mar 18
0
T100P: Can't Make/Receive Zap Calls (Long Newbie Blah)
All, Alright, I've looked around the internet, the voip-info.org wiki, and browsed the contents of this mailing list. While I've found a couple of scenarios that are close to this one, I haven't found one that uses my particular card (T100P). Without further delay -- I have successfully configured internal SIP services between a Snom 200 and a Windows X-Lite client and have
2004 Jul 18
6
PSTN Gateway X101P
I am trying to setup a simple pstn gateway using Asterisk and a X100p card. I have got everything installed using Redhat 9 and am able to load Asterisk. I also configured sip and I am able to connect to the asterisk gateway with Xlite on the windows side. I am able to dial 1000 and get the welcome message. What I am NOT able to do is dial a seven digit local or 10 digit long distance number and
2005 Sep 19
0
Voicemail() application returning -1 on a hangup
Hi, I am trying to insert a system() call right after the call to Voicemail() in order to notify people that they have received new voicemail. (In case you are interested, I was following the tips here, but I have had to change a few things: http://www.voip-info.org/wiki-Asterisk+tips+callback ). My setup works when someone leaving a message hits # to exit the voicemail after speaking, or
2003 Oct 17
2
Beta testers for visual configuration tool f or asterisk
Count me in too. -----Original Message----- From: sip [mailto:sip@intology.com] Sent: Friday, October 17, 2003 1:56 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Beta testers for visual configuration tool for asterisk count me in ----- Original Message ----- From: "Paulo Mannheimer" <paulohm@instant.com.br> To: <asterisk-users@lists.digium.com>
2009 Sep 15
0
1.6.2.0-rc1 intermittent voicemail problem ?
1.6.2.0-rc1. I am having trouble with voice mail intermittently not working correctly on CHANUNAVAIL. (it may happen for other statuses too, haven't checked). Basically here's what happens: -- Executing [1651xxxxxx at mydids:1] Macro("SIP/ipkall-trunk-14838bc8", "phone,1651xxxxxx") in new stack -- Executing [s at macro-phone:1]
2006 Jun 19
2
Asterisk voicemail problem with isdn avm fritz!card
Hello everyone, I have Asterisk SVN-trunk-r7498 installed on a server (celeron 2.4 Ghz, 256MB) with a TDM40b a TDM04b and an avm fritz!card pci. I experience a problem with voicemail: my messages are good unless the incoming call comes from isdn, which means via the avm fritz!card. In this case (and in this case only) the message is disjointed and I can hear at most 1 second out of a 1 minute
2004 Apr 07
2
error 488 - Not Acceptable Here
I have a setup of 3 Cisco 7940 running Sip image 6.3. All these phone are registered by the below information *CLI> sip show peers Name/username Host Mask Port Status 2002/2002 192.168.22.199 (D) 255.255.255.255 5060 Unmonitored 2001/2001 192.168.22.200 (D) 255.255.255.255 5060 Unmonitored 2000/2000 192.168.22.198 (D)
2004 Sep 15
1
Extension based call forwarding using capiECT
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I try to get callers forwarded to by mobile phone when they dial a certain digit. In my extensions.conf I have defined the following: [279xxxx] exten => s,1,SetLanguage(de) exten => s,2,Wait,5 exten => s,3,BackGround(demo-congrats) exten => s,4,Goto(boksa,#,1) exten => 3,1,VoiceMail,u1 exten => 4,1,VoicemailMain exten =>
2006 May 06
3
Voicemail error
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
2004 Aug 30
4
Newbie - Voicemail Password Help
Hello All. I'm just beginning with Asterisk and I have it all working now. I'm using Asterisk 1.0 RC1. My only question is this; when I check my voice mail the PBX simply says "password". I wanted to make it say "please enter your voice mail password" so I am using Background(pls-enter-vm-password). However now I hear "Please enter your voice mail password
2003 Dec 11
0
FW: Iax, Iax2 and Iaxcomm
Talking to myself ... ;-) Solved this by ... disallow=all allow=gsm ;allow=ulaw ;allow=alaw -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Paulo Mannheimer Sent: quinta-feira, 11 de dezembro de 2003 09:02 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Iax, Iax2 and Iaxcomm Hi, I'm trying
2004 Jun 26
2
Newbie needs help
I've been banging my head on a brick wall for about an hour now trying to understand why the following doesn't work (which is even provided as an example in the distribution!). The goal is to create a voicemail-only extension not associated with a phone. I'd rather not have an extension dedicated to VoicemailMain(), so I would like the user to be able to hit '*' during