search for: asric

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2006 Mar 01
9
MOH native files
Where can I find alaw, ulaw, gsm, g729 formats for native music on hold? I have some mp3 files and I have tried to transcode them to above, but it seams that SOX can't do that. Please, tell me where to download some MOH files (in above formats) or how to transcode mp3? Thank you for your time! -- Tomislav Parcina tparcina#lama.hr
2006 Jun 13
4
how to hang the zap channel
hello, I got those extensions: exten => 555,1,MeetMeCount(500|count) exten => 555,2,Gotoif,$[${count} = 1]?6 exten => 555,3,Meetme,500|pMs|1234 exten => 555,4,Playback,goodbye exten => 555,5,Hangup exten => 555,6,Goto(from-internal-custom,556,1) exten => 555,7,hangup exten => 556,1,System(/bin/cp /etc/asterisk/1-test /var/spool/asterisk/outgoing/) exten =>
2006 Jan 16
1
Zapata.conf and Realtime
Sorry if this is a double-posting. I tried sending the following message Friday afternoon, but it still hasn't made it to the list. Based on the comments in the extconfig.conf file, zapata.conf *should* support being loaded realtime. Has anyone succeeded in doing so, and what does the schema, etc look like? Thanks! Steve
2006 Feb 24
1
ImportVar Syntax
I am trying to use ImportVar to get some information out of a SIP/ZAP channel. I cannot seem to find an example of the syntax, or what variables I can access. Basically, I would like to output which person is being called. i.e: SIP/25 calls SIP/21. 25 executes a macro, and the result is SIP/21. The info that I want is stored in the channel's "Direct Bridge" variable. I have
2006 Mar 24
1
[1.2.5] DTMF not being set correctly (RESEND)
I apologize if this gets posted twice. Tried once about 5 or so hours ago, and still have not seen the message on the list.... -------------------------------- I am having trouble getting DTMF mode to be set to inband on incoming calls. I have the following set, and for some reason the connection is still negotiated with rfc2833. [outbound] type=friend secret=XXXXXXX username=XXXXXXX
2006 Apr 13
0
SIP register question
I am trying to link an asterisk box up to a SIP server on the same subnet. The SIP server does not have a password (and is locked down by IP number 'allow'). How do I specify this on the register line? Based on the documentation, the line looks like this: register => user[:secret[:authuser]]@host[:port][/extension] It looks like user@host is the minimum required. Is there anyway
2006 May 16
1
GXP-2000 w/ 1.1.0.11 firmware
I had provisioning via tftp working on this phone. I have verified that after the firmware upgrade, it contacts the tftp server and downloads the cfgMACADDR file, and the ring/etc files successfully. Unfortunately, changes made to the config file don't make it to the phone (SIP account info/server info, etc). The script that I am using to generate the binary files is loosely based on
2006 May 23
1
Getting the Server IP
Hello all! Can anyone think of an *easy* way to get the IP number of the server running asterisk from within the dialplan? Thank you in advance! Steve
2007 Mar 05
4
Polycom Questions
Any Polycom gurus out there? If so, I have a few config file questions. First off, does anyone have the daylight savings time rules written for this Sunday's big change? Secondly, if there any way in the config file to tell the phone not to display the number of missed calls? I don't mind it keeping the missed calls list, I just don't want that running count. Lastly, I am trying