similar to: Voicemail/Password Issue

Anyone try the new Cisco firmware for the 79xx sip phones? In my test it seems to work fine for a little and than soon the phone looses its time. At first the status shows clear, and then it appears to get confused about the ntp time source and the time goes away on it. No features, just bug fixes. Perhaps I have something wrong with my ntp settings, but the other firmwares work just fine. --

Is there a way to get the OnHold music to restart without restarting asterisk? -- respectfully, Joseph - (606) 477-2355 x140 ------=============

Does anyone have any experience with an Asterisk compatible softphone application which meets the following criteria: 1) Is able to use touch screen rather than mouse for on-screen functions. 2) Has an API which can be used to export Caller ID info to another App on the same compuer. Thanks Bill

Was in the process of upgrading a 7960 to SIP and in advertently applied a skinny image (P003G302.bin), now no matter what i put in OS79XX.TXT and OS7960.TXT it simply wont upgrade. The phone is pulling down OS79XX.TXT from my TFTP server but then goes on to repetedly ask for SEP<MACaddr>.cnf. Help!! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, No one seems to have any issue with the following posting. Can any one suggest how to install/configure channel drivers to work. Basically I am trying to send the SIP calls to GNUGK but Asterisk reports the error "No channel driver found". >>> I was trying to compile the oh323 channel driver but unable to compile the openh323_1_13_5 (which is the only required version as

06.09.2016 16:42, George Joseph ?????: > > > On Tue, Sep 6, 2016 at 7:32 AM, Dmitriy Serov <serov.d.p at gmail.com > <mailto:serov.d.p at gmail.com>> wrote: > > Hello. > > Several months server working on asterisk 13.7 and pjproject 2.5 > (installed separately). Once a day the server crashes or hangs and > is familiar sores that written

Does anyone have a more clear beep tone for the voicemail? The default one seems to cut off and give the feeling that there is a problem with the vm :) If no one has this available, I may try to create a new one. Thanks. -- respectfully, Joseph - ------=============

What third party programs are available for parsing the queue_log file and CDR file? I know about XC-AST, but management would prefer a php based solution. What have other admins done to retrieve detailed call information about the queue system? Anyone develop their own that they don't mind sharing? --johann

Does anyone know of a device that will take an SS7 link and convert it to a PRI? -- respectfully, Joseph - (606) 477-2355 x140 ------=============

In asterisk 1.6 the voicemail prefix u b don't work, I have: exten => 1,3,Voicemail(u11) and it keeps telling me: No entry in Voicemail config file for "u11" exten => 1,3,Voicemail(11) works, Isn't prefix "u" suppose to play: "The person at extension ... 11 ... is unavailable," ? -- Joseph

The chan_sccp page at http://www.voip-info.org/tiki-index.php?page=chan_sccp2 has been updated. See the bottom of the page. Thanks. Comments welcome. -- respectfully, Joseph =============== ---------------------= ********** =

I made a little php script that creates a 79xx config if you give it the mac address, ext, etc. Is this something that would be of interest to anyone? Likely it could be improved on. And there may be some variations that I have not thot of. -- respectfully, Joseph =============== ---------------------= ********** =

Is there any way to play background music on a sip phone while the phone is not in use like many legacy pbx's offer? Could you take 7960 and use the 6th line in a similar fashion to the all setup maybe? Thoughts ideas? -- respectfully, Joseph - (606) 477-2355 x140 ------=============

Hello. Several months server working on asterisk 13.7 and pjproject 2.5 (installed separately). Once a day the server crashes or hangs and is familiar sores that written watchdogs. Yesterday I decided to upgrade to 13.11 and bundled pjproject (2.5.5). Solved all the problems with compilation I started asterisk several times and each time after 5-7 seconds was seg fault. So I didn't get

Can not check voice mail-box externally. I'm trying to log-in externally (from PSTN line) to check my "voice-mail" so I created context to authenticate log-in ... exten => 7,4,Authenticate(01894546) exten => 7,5,DISA(4789|disa-access) Authentication works OK, I get inside dial none enter my mailbox extension but it doesn't accept my mailbox password even though it is the

What codec should be used to connect a * box to a cisco router which has a t1 with 24 trunks coming in? My router voip dial plan looks like this: dial-peer voice 2 voip destination-pattern [1,2,,3,5,8].. session protocol sipv2 session target ipv4:10.x.x.x dtmf-relay cisco-rtp codec g711ulaw no vad ! The problem I have is when more than one call is on it, sometimes the quality gets very

Has anyone tried to get dialogic ss7 trunking to work with Asterisk? I did some googling but nothing helpful turned up... Could a person get an intel dialogic and get * to see it like a zap channel? And use it for incoming and outgoing trunk access? -- respectfully, Joseph ------=============

I've tried to simplified the dial plan and use "n" instead of numbers but I've noticed it is not executing my voicemail if I substitute number with "n" In the example below when the call is not answered, it does not go to voicemail; call just hangup. exten => 1,1,Playback(transfer) exten => 1,n,Dial(${sales_support}&IAX2/iaxy-322,20,jrw) exten =>

Asterisk to Asterisk voicemail not working (accessing voicemail from another asterisk). PSTN to Asterisk is working, but not between two asterisk :-( I've tried setting my asterisk dtmf to rfc2833, inband it is not working. The other Asterisk Linksys is set dtmf = auto -- Joseph

I just upgraded to asterisk-1.2.14 and using default "streamplayer" though, I don't think is has anything to do with the voice messaging system, does it? When I enter the mailbox to listen to the recored message I press "1" and when the message starts playing all it plays is: "First messge received" and silence. The error message I get: Mar 25 11:39:02