similar to: Listening for DTMF when dialling (sorry, accidentally sent the previous message too early!)

Hi all, I want to set up an extension which dials a group of phones while at the same time plays a message ("Press 1 to leave a message") and listens for DTMF. I haven't played around yet but the way I read the docs this isn't possible as Thanks, Peter Spikings This message has been comprehensively scanned for viruses, please visit http://virus.e2e-filter.com/ for details.

Hi all, Does anyone know a way of playing a sound while the dial command is running? I want to play a sound every 10 seconds while relevant phone(s) are ringing and have ring tones played to the caller in the gaps. I am aware of the fact that you can play a class of MoH during the dial but I can see two problems with that.... firstly that the caller won't hear the sound from the start and

hi asterisk users How can i make asterisk "ignore" invalid extensions, and go on playing the background soundfile? Normally, asteriks will take the user to the invalid extension if the caller presses anything other than 1 or 2 in the following context:: [example] exten => s,1,Answer() exten => s,2,Background(hello-world) exten => s,n,Goto(s,2) exten =>

Renato Golin via cfe-dev <cfe-dev at lists.llvm.org> writes: > On Thu, 10 Oct 2019 at 22:26, David Greene <dag at cray.com> wrote: >> That would be a shame. Where is test-suite run right now? Are there >> bots? How are regressions reported? > > There is no shame in making the test-suite better. That's not what I meant, sorry. I mean it would be a shame

> -----Original Message----- > From: cfe-dev <cfe-dev-bounces at lists.llvm.org> On Behalf Of Renato Golin > via cfe-dev > Sent: Friday, October 11, 2019 11:24 AM > To: David Greene <dag at cray.com> > Cc: llvm-dev at lists.llvm.org; cfe-dev at lists.llvm.org; Gerolf Hoflehner > <ghoflehner at apple.com>; openmp-dev at lists.llvm.org; lldb-dev at

Hello Alan, I had the same issue and I needed to add this line: ldap server require strong auth = no to smb.conf. Then, just restart/reload samba and it should work. On Fri, Jul 8, 2016 at 8:37 AM, Alan Hughes <alanhughes at e2eservices.co.uk> wrote: > Last night we updated out Samba-4 AD server to version 4.2.14 usng the > SERNEt packages, running on SLES 12. We have a number of

Its should not be needed to drop and recreate. Try the following. Delete the created records in the "wrong" zone(s) Try to delete the "wrong" zone. ( ignore the error) Restart bind Restart samba. Pray and check again. Are the "empty zones gone and are your dns records there again? Greetz, Louis > -----Oorspronkelijk bericht----- > Van: samba

Last night we updated out Samba-4 AD server to version 4.2.14 usng the SERNEt packages, running on SLES 12. We have a number of services (mail services, MANTIS, etc) that access the server via the LDAP interface and in all cases we discovered that none of them where able to establish a successful LDAP connection after the upgrade.   Previously we used plain LDAP to access the server, i.e. we did

Folks   I've managed (due to me being fat-fingered that morning) to get a DNS zone in a Samba-4 DNS setup screwed up.   Basically I was trying to add a new A record to an internal domain "e2eservices.co.uk" using the MS administration tools (not the samba-tool CLI). However instead of adding the entry "styx" to the domain, I accidently added

Hey all, I'm having problems reliably dialling out my FXO card. About 30% of the time I'll get a "your call cannot be completed as dialed". I'm thinking it might be the dialling speed, but I can't find any configs that change that setting. Any suggestions for troubleshooting? Thanks, Brad Waite

Hello, I'm wondering if it is possible to ring X number of extensions simultaneously, and each answered call can be handled with some code. I can do a huntgroup-esque way of dialling, but I want all the dialled numbers to be picked up. I hope this makes sense.. If not please say.. Many thanks! Andy -------------- next part -------------- An HTML attachment was scrubbed... URL:

hello everybody, one of our customers which wants a soft transfer between his old pbx to asterisk and sip. the setup is as follows: telco <---pri---> asterisk <---pri---> legacy pbx everything is fine exept that when dialling from the legacy pbx it takes about 3 seconds before the asterisk start to dial. where does this delay come from? has it to do with

Hi all. I am seeing incoming calls from digital lines (mobiles e.g.) dialling my main number + 3-digit extension just fine ("Accepting voice call from '11234567' to '250' on channel 0/1, span 1"). The problem however is with calls from analog lines: "Accepting voice call from '13331846' to '25' on channel 0/1, span 1" * just sees 2 digits, not

Hello, I'm experimenting with Overlap Dialling in asterisk 1.6. I've enabled this in sip.conf and on the SNOM 300 phone. My problem is that asterisk dials out as soon as it matches an extension without waiting to see if the user is going to type in more digits. Is there a way to set a timeout per channel or globally? I'd like Asterisk to wait for a few seconds once its found a match

hi there was wondering if anybody knows this.. have successfully installed shady dial and the agent is now logging in successfully i've enabled postgres debugging and i found out that no request had been made by the shady dialer to query the database for the numbers... also i am able to login to the queue simply by entering the agent id... it doesnt ask for the password...it simply plays the

Hi :) I have a little 'slow dialling' problem. When I dial, e.g. 200# for the Asterisk 'echo test' demo application from my PBX extension 1010, I see this in the console the instant I press the # key: -- Starting simple switch on 'Zap/65-1' -- Accepting overlap call from '1010' to '200' on channel 0/3, span 3 then exactly 3 seconds elapses, and

I'm trying to use Jeremy's suggestion of dialling using just the peer name instead of user:pass@peer but I'm running into some really funky issues. It does the same thing with both VoicePulse and another * server I have. [voicepulse] type=peer host=gw5.voicepulse.com trunk=yes user=USERNAME pass=PASSWORD and in my dialplan: Dial(IAX2/voicepulse/${EXTEN:2}@VPWS,90,r) The log shows

Hello This is a fun one for the list... Twice now, the Police have contacted us to say they have had a silent call then hangup from our landline number to the 999 service. As a matter of course, they follow up these calls in case someone is in distress. Nobody here was in distress - well, no more than normal! The Police aren't hugely happy when we tell them it must be a mistake. Thing

Using 1.0.9: If I have: exten => s,1,Dial(SIP/5555&SIP/12345@192.168.1.1) How can I return the DIALSTATUS variable for the second SIP channel ONLY if the second SIP channel is busy, regardless of the dialstatus of the first SIP channel? What I want is, if the second SIP channel is busy go to n+1 or n+101 regardless of the status of the first SIP channel. tia

Hi, Red Hat 9.0 Asterisk 1.2.7.1 I'm having a bit of an intermittent problem with my SIP account. Often (but not always) when I start * or RELOAD my dial plan from the CLI I get this message: >Apr 30 11:01:20 WARNING[12785]: chan_sip.c:11822 add_realm_authentication: Format for >authentication entry is user[:secret]@realm at line 31 >Apr 30 11:01:21 WARNING[12785]: acl.c:244