similar to: asterisk registered as UA

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! I'm setting up my IVR system, how can I register in a mysql database the IVR menus accessed by the clients ? Thanks a lot, Szasz Szabolcs

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, 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

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

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

* 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)

Try again with these settings. auth-nxdomain no;    dnssec-validation no; I suggest review you settings https://wiki.samba.org/index.php/BIND9_DLZ_AppArmor_and_SELinux_Integration Stop and start bind and samba (first bind) Try again. Greetz, Louis > -----Oorspronkelijk bericht----- > Van: samba [mailto:samba-bounces at lists.samba.org] Namens > Hajdu Szabolcs via samba

On Tue, 22 Jan 2019 11:12:37 +0200 Hajdu Szabolcs via samba <samba at lists.samba.org> wrote: > I configured it but no luck apparmor is configured as the link > suggests i tried to rejoin and deleted the local database manually > but then still recreates these five zones with CNF and gives the > error. > > CNF = Collision Something is creating the objects in AD and

I installed DAHDI (2.1.0.3) on a machine with asterisk 1.4.22 and libpri 1.4.7 and I am getting the error: -- Requested transfer capability: 0x00 - SPEECH -- Called 23/317506XXXX -- Channel 0/23, span 1 got hangup, cause 99 -- Hungup 'DAHDI/23-1' on DIALING out. Calling in seems to work just fine. Seems like everything is configured fine... system.conf

Hi, I have set up an asterisk with TLS and SRTP support. The SRTP is working with Phonerlite softphone. I have problem with the SRTP, when I make calls on Audiocodes gateway . I got the folloowing messages on asterisk: [Oct 2 10:59:48] NOTICE[24868]: sdp_crypto.c:232 sdp_crypto_process: Crypto life time unsupported: crypto:1 AES_CM_128_HMAC_SHA1_80

Hello, I just made a asterisk / dahdi fresh install on Debian 8.4, and ended up with the following packages : $ sudo dpkg -l|grep -Ei 'dahdi|asterisk|libpri' ii asterisk 1:11.13.1~dfsg-2+b1 amd64 Open Source Private Branch Exchange (PBX) ii asterisk-config 1:11.13.1~dfsg-2 all Configuration

Hello, I have a Sangoma A200 analog card with 2 FXO ports. It's working well with asterisk 1.4.22 and Zaptel. I decided to upgrade to asterisk 1.6/dahdi. I compiled and installed, dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.10 wanpipe-3.4.1 asterisk 1.6.1.1 My analog card is recognized in dahdi_hardware. However, asterisk cannot compile chan_dahdi.so. I've tried passing

I have a small system, server, client and 2 phones. Phones are polycom 501's. In general all is working fine. I can call the two phones, speak etc... I can have the server call each phone and play a wave file. However, when trying to setup a direct dial number of 1044 that calls another machine running asterisk - something ODD is happening. ; This is not working.... [smvoice-sip] exten

I was running : asterisk 1.2.24 zaptel 1.2.21 libpri 1.2.6 I remove zaptel and compiled asterisk 1.4.22 libpri 1.4.7 dahdi 2.1.0 dahdi_cfg -vvv DAHDI Tools Version - 2.1.0 DAHDI Version: 2.1.0 Echo Canceller(s): Configuration ====================== SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel

Hi, It's not clear to me which asterisk version drops support for zaptel in favor of dahdi. Dahdi and zaptel can "coexist" in some 1.4 versions but it seems that from 1.4.22 onward, chan_zap.so is not built. Documentation within the 1.4.23.1 tarball indicates that one can keep using the zap*.conf files by switching an option in asterisk.conf. Is this true or should I forcefully