search for: retrydi

Displaying 15 results from an estimated 15 matches for "retrydi".

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2009 Jun 26
0
Problem with RetryDial
I issue this command: RetryDial(another-time,10,4,SIP/GXP280_18,10,ghM(cfmc_dial_private^RetryAndQ ueue^SIP/GXP280_18)) Asterisk rings the phone for 10 seconds. Asterisk then waits 10 seconds. Asterisk rings again for 10 seconds. I would expect this to happen a total of 4 times. The problem is that after the second ring for 1...
2005 Feb 04
9
callback on busy
Hello everybody, I would like to implement "callback" function. When I call a person and his extension is busy I can press, for example, 5 and get a callback when his phone is not busy anymore. When I create a call file and copy it to spool call folder asterisk makes a call. One problem is that when extension is still busy my phone rings and I get busy tone of the person who I am
2016 Aug 22
2
Dial and start music on hold after timeout
Thank you for the idea. The problem with RetryDial, is that it will cancel the first call, play the announce and then dial the SIP peer once again, so the telephone will display a missed call. I would prefer to do everything in a single call. Le 22/08/2016 ? 17:57, John Kiniston a ?crit : > You could try using RetryDial() instead of Dial,...
2008 Apr 07
2
DTMF between Asterisk servers.
...onfused on DTMF. A sip peer is registered on two Asterisk servers. No dtmfmode is set for them, the sip peer is 999 on Asterisk 1 and 999 on Asterisk 2. They both register on each other. A call comes in on Asterisk server 1, provider 1, dtmf=inband. Then the call is transferred to Asterisk 2: RetryDial(/var/lib/asterisk/sounds/connecting,15,10,SIP/12351 at 65.xx.xx.10,,t T,) Where 12351 accepts the call on Asterisk 2, and in some cases, that call is transferred out to a PSTN number, or wherever, but not within Asterisk anymore via provider2, dtmf=rfc2833. When the call comes in, I'd li...
2005 Feb 08
11
More complicated huntgroups / delayed ringing
...issed calls" in my > display. > Is there a way to make real huntgroups where I can say "first this and > this, after 15 secs that also"? > > Regards, > Stefan > I've worked on this off and on. I've been thinking about writing a patch to 'Dial/RetryDial' (why isn't retry just a flag for the Dial app?) that would do exactly that. In my office, my boss wants the functionality that you describe. I've attempted this in many different ways so far - sans patching the code. The only thing I've found that is somewhat functional is t...
2016 Aug 23
2
Dial and start music on hold after timeout
...l Gottlieb <isrlgb at gmail.com> wrote: > You could m and make a moh file that has ringing the first 30 sec and then > the anouncment > > ?????? 22 ????? 2016 7:19 PM,? "Jean Aunis" <jean.aunis at prescom.fr> ???: > > Thank you for the idea. The problem with RetryDial, is that it will cancel >> the first call, play the announce and then dial the SIP peer once again, so >> the telephone will display a missed call. I would prefer to do everything >> in a single call. >> >> Le 22/08/2016 ? 17:57, John Kiniston a ?crit : >> >...
2006 Apr 29
0
canreinvite, bandwidth, dial option
...o *d*: This flag trumps the 'H' flag and intercepts any dtmf while waiting for the call to be answered and returns that value on the spot. This allows you to dial a 1-digit exit extension while waiting for the call to be answered - see also RetryDial <http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+RetryDial> o *D(*/digits/*)*: After the called party answers, send /digits/ as a DTMF stream, then connect the call to the originating channel. o *L(*x[:y][:z]*)*: Limit the c...
2016 Aug 22
3
Dial and start music on hold after timeout
Sorry, I forgot to write that the SIP peer must keep ringing while the announcement is being played. Le 22/08/2016 ? 17:42, John Kiniston a ?crit : > This seems like the obvious answer but maybe I'm misunderstanding the > question. > > exten => s,1,Dial(SIP/alice,20) > same => n,Playback(myannouncement) > same => n,NoOP(Whatever else you want to do goes
2016 Aug 23
2
Dial and start music on hold after timeout
...il.com> wrote: > >> You could m and make a moh file that has ringing the first 30 sec and >> then the anouncment >> >> ?????? 22 ????? 2016 7:19 PM,? "Jean Aunis" <jean.aunis at prescom.fr> ???: >> >> Thank you for the idea. The problem with RetryDial, is that it will >>> cancel the first call, play the announce and then dial the SIP peer once >>> again, so the telephone will display a missed call. I would prefer to do >>> everything in a single call. >>> >>> Le 22/08/2016 ? 17:57, John Kiniston a ?...
2009 May 21
0
Asterisk 1.4.25 Now Available
...a PRI channel really signals progress. - Closes issue #13034. Reported, patched, and tested by klaus3000. * Avoid destroying the CLI line when moving the cursor backward and trying to autocomplete. - Closes issue #14373. Reported, and patched by eliel. * Fix a crash due to too few arguments to RetryDial. - Closes issue #14852. Reported, and patched by junky. * Fix RFC2833 issues with DTMF getting duplicated and with duration wrapping over. - Closes issue #14815 and #14460. Reported by geoff2010 and moliveras. Patched by dimas. Tested by geoff2010, file, dimas, ZX81, moliveras. Thank you...
2005 Aug 22
0
Dial, RING with a digit interrupt
...rce the call in to off-hook voice announce. If you called an extension that was busy, pressing 6 would make the called party's phone ring you back when they hang up. These are two seemingly simple ideas, but I've not yet come up with a good concept of how. I've looked at the command RetryDial, which in many ways comes close, but misses. Something tells me this is going to be a "playtones" type of situation, but I have no clue if, or how its possible to generate ringtone on the handset, then drop in to the next priority to actually RING the extension, all the while waiting...
2008 Mar 19
0
Deadair in queues.
Hello, Asterisk Server A makes an outbound call, and upon connect: exten =>1,n,RetryDial(/var/lib/asterisk/sounds/connecting,0,3,SIP/${connectto},,tT ) (${connectto} most of the time happens to be 12345 at 66.xx.xx.66 or 54321 {IP masqueraded ofcourse}) ..transfers it to * Server B (i.e 66.xx.xx.66) via SIP. (Background info, Server B registers on Server A as 1000, and Server...
2009 May 21
0
Asterisk 1.4.25 Now Available
...a PRI channel really signals progress. - Closes issue #13034. Reported, patched, and tested by klaus3000. * Avoid destroying the CLI line when moving the cursor backward and trying to autocomplete. - Closes issue #14373. Reported, and patched by eliel. * Fix a crash due to too few arguments to RetryDial. - Closes issue #14852. Reported, and patched by junky. * Fix RFC2833 issues with DTMF getting duplicated and with duration wrapping over. - Closes issue #14815 and #14460. Reported by geoff2010 and moliveras. Patched by dimas. Tested by geoff2010, file, dimas, ZX81, moliveras. Thank you...
2008 Apr 04
5
Ring back when free?
Has anyone here implemented "Ring back when free" in Asterisk? The way it works in the UK is as follows: 1. A calls B. B is engaged (busy). 2. A hears "The number you called is busy. To use ringback, press 5" 3. A presses 5, and hears "Your ringback request has been accepted". 4. A hangs up. 5. Later, B hangs up. The system then calls A (if A is now busy, it
2005 Jun 15
1
app_dial.c:977 dial_exec_full: Unable to create channel of type 'Zap' (cause 0)
...kipping chan_agent.so] [skipping chan_mgcp.so] [skipping chan_local.so] [skipping chan_skinny.so] [skipping chan_oss.so] [skipping chan_modem_i4l.so] [skipping chan_phone.so] [app_dial.so] => (Dialing Application) == Registered application 'Dial' == Registered application 'RetryDial' [app_playback.so] => (Sound File Playback Application) == Registered application 'Playback' [app_voicemail.so] => (Comedian Mail (Voicemail System)) == Registered application 'VoiceMail' == Registered application 'VoiceMailMain' == Registered applicati...