Re: [asterisk-users] app_swift.c:338 engine: Failed to set voice

. even if it's not, you don't have David registered. try making that: voice=Marta (or possibly: voice=Marta-8kHz) then restart asterisk and give it another shot. -- Jeremy Kister http://jeremy.kister.net

[asterisk-users] parking on ast 1.6.2.8

a clear syntax error. What's a solution to letting someone who's retrieved a call from the parking lot re-park the call ? -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] Cell phone redialer?

to the party but with our caller ID. you're looking for DISA. http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA Example 2 should slip right into your extensions.conf. -- Jeremy Kister http://jeremy.kister.net

[asterisk-users] app_swift 1.6.2 DTMF issue

://jeremy.kister.net/code/app_swift-1.6.2.patch -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options

Re: [asterisk-users] app_swift 1.6.2 DTMF issue

On 1/10/2010 5:33 PM, Jeremy Kister wrote: With app_swift 1.6.2 + asterisk 1.6.1.12, I've found that if you enter DTMF during cepstral playback, the first digit of ${SWIFT_DTMF} is [un]set in an odd way. The problem lies within f-subclass inside the else if of line 436. the code seems

Re: [asterisk-users] app_swift 1.6.2 DTMF issue

On 1/10/2010 5:33 PM, Jeremy Kister wrote: With app_swift 1.6.2 + asterisk 1.6.1.12, I've found that if you enter DTMF during cepstral playback, the first digit of ${SWIFT_DTMF} is [un]set in an odd way. I fixed it up by ignoring the f-subclass and starting the dtmf_listener right away

[asterisk-users] asterisk-users archive

http://lists.digium.com/pipermail/asterisk-users/ Trusting user-generated date fields? sweet. :D -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] identifying channel for softhangup

On 12/29/2009 1:01 AM, Jeremy Kister wrote: e.g., in the first call, below, the channel name is SIP/vgw1-0075 -- the second call (on the same FXO port after a soft hangup on the CLI) is SIP/vgw1-0077 How can I extract this information in the dialplan so that I can use

Re: [asterisk-users] identifying channel for softhangup

, and then the call goes out the landline. then if a second 911 call goes out, then it goes out over sip. I have all that working, except the SoftHangup -- because the channel name is not static. So I need to look it up somehow on the fly, or configure the channel name to be static/predictable. -- Jeremy

Re: [asterisk-users] identifying channel for softhangup

On 12/29/2009 3:54 PM, Danny Nicholas wrote: You could do a System(core show channels) and grep out 911 and kill everything else; probably easier as an AGI call that a dialplan function, but both can be done. great idea; thanks! -- Jeremy Kister http://jeremy.kister.net

Re: [asterisk-users] identifying channel for softhangup

(SaferSIPDial,${EMERGENCY_NUM}) -- Jeremy Kister http://jeremy.kister.net./ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo

[asterisk-users] identifying channel for softhangup

' -- Executing [...@extensions:1] Hangup(SIP/141-0076, ) in new stack == Spawn extension (extensions, h, 1) exited non-zero on 'SIP/141-0076' -- Jeremy Kister http://jeremy.kister.net./ ___ -- Bandwidth and Colocation Provided by http://www.api

[asterisk-users] calls ending up in default context

of where i think i'm directing it. Can someone tell me what I have misconfigured? 1760 config: http://kister.net/tmp/vgw1-confg extensions.conf: http://kister.net/tmp/extensions.conf.txt sip.conf: http://kister.net/sip.conf.txt -- Jeremy Kister http://jeremy.kister.net

[asterisk-users] clever ways to share an extension between sip and fxs

/transferring for those who would use it. I know that I'm not looking for Dial(SIP/xSIP/y) - as documented, this handles nothing like what I'm looking for. Ideas? -- Jeremy Kister http://jeremy.kister.net./ ___ -- Bandwidth and Colocation Provided

[asterisk-users] announcement tone to callees of app_page

) parameter to app_page would be perfect. with auto-answer turned on with my cisco 7940 phones, i find it lethal that someone can page my phone and listen to what i'm doing without me realizing it (unless i look at the phone to see it's on speakerphone) -- Jeremy Kister http://jeremy.kister.net

Re: [asterisk-users] app_swift issue

app_swift.c: In function ‘engine’: app_swift.c:402: error: incompatible types in assignment app_swift.c: In function ‘load_module’: app_swift.c:546: error: ‘AST_MODULE’ undeclared (first use in this function) try the patch at http://jeremy.kister.net/code/app_swift-1.6.2.patch -- Jeremy Kister

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