similar to: moh backround?

Hello. Is it possible to change the defined moh sound file within an extension? I have: exten => 18,1,Answer exten => 18,n,Wait(3) exten => 18,n,SetMusicOnHold(durchwahl) exten => 18,n,Dial(SIP/118,15,m) exten => 18,n,Hangup Now i have the situation someone calls and my phone is ringing while moh (durchwahl) is playing. When i pickup the call and press the hold button during

All, I am using Asterisk 1.4.4 and it is not playing any MOH. I think the underlying problem is the following error: [Jun 11 20:25:37] WARNING[3207]: res_musiconhold.c:424 spawn_mp3: Found no files in '/var/lib/asterisk/moh/asterisk' [Jun 11 20:25:37] WARNING[3207]: res_musiconhold.c:506 monmp3thread: Unable to spawn mp3player Now it does not matter what I change in the

Hello. Someone knows what cause 127 mean. The phone that i'm calling rings once and than the connection interrupts: P[ 5] --> l3id:10040 P[ 5] --> cause:127 P[ 5] --> out_cause:127 P[ 5] --> state:ALERTING P[ 5] --> Channel: mISDN/5-1 hanguped new state:CLEANING P[ 5] $$$ CLEANUP CALLED pid:3 best regards -- Thomas Stein knowledgeTools? ....damit Sie sehen, was Sie

Hi. Asterisk 1.4.12.1. I get a lot of message like this. Someone knows what this message mean? Do i have to worry about it? [Nov 7 15:24:25] NOTICE[31145]: cdr.c:434 ast_cdr_free: CDR on channel 'Local/152 at local-f137,1' not posted [Nov 7 15:24:25] NOTICE[31145]: cdr.c:434 ast_cdr_free: CDR on channel 'Agent/152' not posted [Nov 7 15:24:25] NOTICE[31247]: cdr.c:434

Hello. I have a problem recieving fax without a callerid. Somehow the script i'm using fails and i don't know how to fix it. Does anyone have an idea how to solve this? Here an example of a working fax transmission: >fax2mail v2.0 > Triggered on Tuesday, May 29 2007, at 10:38 AM > $1 = CallerID number of fax sender = 02365207150 > $2 = CallerID name of fax sender = >

Hello. Just installed the Beronet BN4S0 card. But i can't connect to my ISDN Line. misdnportinfo gives (what does ":Layer 4 protocol 0x04000001 is detected, but not allowed for TE lib" mean?): best regards and thanks t. asterix asterisk # misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) -> Protocol: DSS1 (Euro ISDN) -> Layer 4 protocol 0x04000001

Hello. I have problems to integrate an isdn phone. I don't know why but the isdn phone rings only once and than it looses its connection to his base station. I can make a call from the isdn phone to an VoIP Phone inside my network but when i pick up the phone the isdn phone also crashes. misdn.conf: [ntport1] ports=5 context=isdn-telefon msns=* extensions.conf: exten =>

Hello. I have a strange problem. Its not possible to pickup a call that was placed with a Siemens SL75 Wlan. When this phone calls an internal number and i try to pickup (*8) the call from my phone i get nothing. It seems i have the call for one second or so but after that the call is being cancelled. No problems with other phones (polycom, grandstream). Attached the complete sip debug log

Hello. Does someone know what "order field" means in followme.conf? The Doku says: number=> <number to call[&2nd #[&3rd #]]> [, <timeout value in seconds> [, <order in follow-me>] ] So an example would be: number=> 123&124&125,10,? It would be nice if someone could enlighten me. cheers t.

Hi, i have a problem with Guild wars, it runs only in the backround. I ran it through terminal and got this: Code: fixme:win:EnumDisplayDevicesW ((null),0,0x32eb54,0x00000000), stub! fixme:win:EnumDisplayDevicesW ((null),0,0x32e6c0,0x00000000), stub! fixme:devenum:DEVENUM_ICreateDevEnum_CreateClassEnumerator Category {cc7bfb41-f175-11d1-a392-00e0291f3959} not found

Hi, I recently upgraded the firmware on my Blackberry 8700 to 4.2, this seems to give it the ability to play wav files. I wondered if anybody out there had managed to get their BB to play the wav files as attached to the Asterisk voicemail emails? Mine seems to ignore the attachment. I am using BES 4.1 for sending these emails out via Exchange 2003 if that makes a difference. thanks Mike

Has anyone tried to do a script to pickup an ITSP voicemail. Lesnet provides an option for an overflow mailbox in the event a caller can get to my * box. I'd like my * to poll it and dump any messages found into my general mailbox Any ideas Similarly, a telco mailbox. It at least has the advantage of having stutter dial tone as a trigger Any hints or suggestions welcome D Dave Bour

Hello, Looking the asterisk 1.8 API documentation (http://www.asterisk.org/astdocs/api/index.html), I see a lot of new fields for sip register uris: register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry] But the *peer* is not explained anywhere. What it is for ? Regards, Guillaume Bour. -- Guillaume Bour<gbour at proformatique.com>

I've had slew of problems with my Bell Canada Single Number Reach (SNR) dropping in the past couple of months. Another outage Monday for several hours has me wondering if there's a way to 1. Make a call out of my system via a PSTN back to my SNR line, say every 30 minutes (this I'm sure is easy enough via the call file...however...) 2. Track the outgoing call and match to an

Dear All: I have several boxes that up and running just great, then we changed internet equipment due to a lightning strike, now all my inbound IAX connections (iax2 show peers) have unknown status. If I log into the remote boxes, it says "Request sent." The authentications haven't changed at all, and all the iax.conf settings are correct. It looks like a firewall issue, but

On Fri, Oct 10, 2014 at 07:32:06PM -0600, Eric Blake wrote: > On 10/10/2014 11:37 AM, Jd wrote: > > Hi > > Looking in to implementing (CBT like) delta backup for KVM. > > Not quite sure what you mean by CBT. > > > > > The following looks promising..(last paragraph) > > http://wiki.qemu.org/Features/Snapshots2 > > > > Libvirt

Hello, I have been using Voicestick inbound (no outbound) successfully for the last few months. Noticed in my logfile that sip registration failed on 6/27/07 at 3am EDT and no successful registration since. Calls to my number eventually timeout as I don't have voicemail setup - as the first step in trouble shooting I tried to enable voicemail on the voicestick website but this fails also

Are there ATAs that allow different phone numbers from one network connection? Such as supporting multiple IP addresses so that each RJ11 has a different extension or some other way?

Stein Hustad System & Database Administrator Norsk Agip A/S ICT Department Tel. NO-51 57 48 77 Fax. NO-51 80 05 65 email: stein.hustad@norskagip.agip.it email: firmapost@norskagip.agip.it Postal: Norsk Agip ,P.O.Box 101 Forus,4064 Stavanger, Norway

Hi, I am looking for hardware sip phone with very good sound quality. Can anyone recommend ? I use to have Grandstream Budge-Tone 100 but I feel that the sound is not very satisfactory and volume too soft Regards ASLAY