similar to: IVR question

Hi, I have encountered a DTMF issue. My scenario: Access carrier-----sip----> Asterisk-1.4.25.1-----sip---->CiscoGW-----ISDN----->TDM Switch the access carrier sends to Asterisk out of band (rfc2833) dtmf, Asterisk forwards it with SIP INFO method to Cisco gateway, but on TDM switch every digit is duplicated. Is it possible that the carrier sends inband along with rfc2833? Kind

Hi, How can I strip + from the front of the caller ID? I have tried this: exten => s/_+X.,1,Set(CALLERID(name)=${CALLERID(name):1}) But it is not working. Szasz Szabolcs -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100114/a3f18781/attachment.htm

Hi I registered my asterisk box to my SIP provider as an UA. For every call I receive on this trunk, I get the message "That is not a valid conference number". I'm using Asterisk version 1.4.22, I had install the dahdi-linux and dahdi-tools and the conference is working between the phones registered to Asterisk PBX. What's wrong? Thanks. Szasz Szabolcs -------------- next part

Hi, How can I disable comfort noise on Asterisk? Szabolcs Szasz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100129/3b27a653/attachment.htm

Hello Team, IVR selection of QUEUEMETRICS As we know queuemetrics had an IVR selection functionality where it can get the IVR keypress of a caller. We saw this link http://forum.queuemetrics.com/index.php?action=printpage;topic=503.0 and upon checking, its only determined the Queue, I want to get is the per IVR of a caller. Can you help me guys regarding this? I want to implement this with

Hi All, I am using Asterisk 15 server and wanted to configure IVR call simulation. My configuration scenario is 1. A subscriber will register to Asterisk server and start a call. 2. The IVR audio will come from the Asterisk sever to sbscriber. 3. Once the subscriber pressed the botton, the call will connect to a number based on DTMF digit pressed by subscriber. Then call will continue for 30

Hi, We are trying to implement a complex business logic in Asterisk. Executing "Wait_For_Digit" command after playing IVR. We want to stop the IVR once we receive the digit. It is not recognizing the Digit until it completes the IVR. How can we stop the IVR once we receive the digit? Thanks BB -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I am using asterisk 1.4.25.1. when I am sending dtmfs the first digit is not detected. Do you know a workaround for this? Besst regards, Szabolcs Szasz -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110520/b3e29d9f/attachment.htm>

Hi, I need some help, getting to work asterisk MWI. I set up Asterisk as voicemail server for Openser as this tutorial shows : http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+With+OpenSER+1.3 . My voicemail system is working but, I can't get to work the message waiting indicator. It doesn't seems to send the Asterisk any NOTIFY message to the Openser box. How can I

Hi! What is the preferred way to make database lookups from within the dialplan? I only know the MYSQL function from asterisk-addons. Are the other methods too? (e.g. for postgresql, unixodbc) thanks klaus

Ladies and Gentlemen, We already have an Asterisk Call center suite installed at our contact center. Now we wish to commence IVR services. We are offering Health Information Services. Can someone help us to develop this Addon / Solution? Best regards. -- ?????? ????? ????? ???? ??????? Nazir Ahmed Vaid Cell:+92300-8277778 eHealth Services (Pvt) Ltd. http://www.ehealth-services.com NexSource

I wonder if somebody could help me. If I try clean install of dovecot-common dovecot-imapd dovecot-pop3d from deb http://xi.rename-it.nl/debian stable-auto/dovecot-2.0 main I got: Starting IMAP/POP3 mail server: dovecotdoveconf: Fatal: Error in configuration file /etc/dovecot/dovecot.conf: service(managesieve-login): executable is empty failed! Setting up dovecot-imapd

After over a year of alphas, betas, and release candidates I'm happy to announce that Version 1.0 of SafiServer and SafiWorkshop has just been released under the open source license GPL (ver 3). You can download installers from our site www.safisystems.com and the source code can be downloaded from Sourceforge (more details available on our site). If you're not familiar with our system

* Sanjay Patel <spatel at rotateright.com> [2019-10-08 08:07:10 -0400]: > On Tue, Oct 8, 2019 at 7:08 AM Szabolcs Nagy <nsz at port70.net> wrote: > > why is that ok? > > > > Because there are no FP exceptions/signals for this IR opcode: > http://llvm.org/docs/LangRef.html#floating-point-environment so llvm cannot support an iso c frontend on an ieee754 target?

Hello, We have fairly large computer park ~100 computers and we want to switch to samba. I installed a new Ubuntu 18.04 LTS fully updated installed samba, after AD join the domain part works i can see users computers and the server, the samba machine is a domain controller the problem is the DNS with this much machines i wanted to use the BIND back-end. The existing environment has 5 vlans

On Tue, 22 Jan 2019 13:18:40 +0200 Hajdu Szabolcs via samba <samba at lists.samba.org> wrote: > options { >     directory "/var/cache/bind"; > >      forwarders { >          208.67.222.222; 208.67.220.220; >      }; > >     dnssec-validation no; > >     auth-nxdomain no;    # conform to RFC1035 >     listen-on-v6 { any; }; >    

Hi all, I have a lot of call in wich I found that my Asterisk doesn't answer the BYE message, then the BYEs are retransmitted, but the call ends, when the Asterisk sends a BYE. Time AS.TE.RI.SK CA.RR.IE.R1 0 INVITE SDP ( g729 g711A g711U telephone-event) SIP From: sip:1265666072 at 81.209.186.14 <sip%3A1265666072 at 81.209.186.14>To:sip:1234567890 at CA.RR.IE.R1 (5060)

Hi listers, We want to implement one call center with asterisk. The idea is it should be scalable, with openser as an dispatcher and bunch of asterisk servers to do ACD, Queues, Agents things... Easy to say :( Look closely to the current asterisk, we do see some problem: - SIP registrations was stored in astdb. - And queue members also was stored in astdb. - ... asterisk was built as

Hi, Is it possible to use realtime for Music On Hold? Is it also possible to store the music/audio files on the database, same way a voicemail can be stored on the database? Thank You Regards, Nhadie

Hello list, I was wondering if it were possible for asterisk to do a voice recognition type IVR? Like you know how most banks and stuff do, where they ask you to say your selection or key it in? If this is possible, how can I set this up? I'm using FreePBX 2.11 on Linux, CentOS 6.7 32-bit, asterisk version 1.8.32.3. Thanks.