similar to: disable comfort noise

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 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 All, I have a sip trunk up and running with a CUCM Express, passing calls fine except for a comfort noise error I'm getting on Asterisk: NOTICE[7520]: rtp.c:788 in process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: x.x.x.x I know Asterisk does not support comfort noise. I have "no comfort noise" on all

hi, when VAD is enabled, can i make the decoder simply produce comfort noise in the event that no voice was detected? i'm working on a p2p voice app. when no voice is detected, i'm thinking that i can make the transmiting endpoint send a signal to notify the remote endpoint that VAD is in effect, instead of having to send the whole packet that doesn't have voice anyway. on the

Hello! I have a sip device that is sending in the SDP: rtpmap:98 g729a It does not seem like Asterisk is negotiating the codec properly, because while the call rings, the rtp lines fail. However, on other sip devices that have "rtpmap:18 g729" in their SDP, things work fine with Digium's commercial g729 license. How do I get "98 g729a" recognized by Asterisk? Thanks,

I'm trying to use Asterisk as a PSTN simulator to run performance tests for echo cancellation algorithms. I'm using the following configuration: SIP <-----> Asterisk 1 <----> Asterisk 2 <----> Echo() Asterisk 1 and Asterisk 2 are connected using E1. Echo() is the dialplan application. The problem is the high delay using this configuration: 20 ms only in Asterisk 2.

I have an Asterisk 1.4 box which is sometimes getting the message below. Here is the weird part, the CNG is coming from ANOTHER ASTERISK SERVER. 209.220.119.19 is an Asterisk 11 box. [Oct 8 11:59:27] NOTICE[20798]: rtp.c:849 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 209.220.119.19

Hi guys, I'm using Zyxel Prestige 2602R, as router/SIP-ua with my architecture SER+Asterisk. Normally, everything is fine. In these days I'm experiencing some problems: some guests said me that, if he call everything is right, but if is called, he cannot hear the caller. Immediately, I though into an RTP-Proxy problem, but is not. Then I saw that message appear on the Asterisk CLI, during

Hi all, Do you have an idea on the best way on how to implement a system with multiple Asterisk servers with BLF working in such a way that a peer on one server can subscribe to another peer on the other server in a seamless manner? Has anyone set-up a system like this before? Thanks! Regards, Ronald -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi everyone, I've got my * system up and running and I'm really pleased. I've gone with G.711 (alaw) and I've stumbled across a problem; when people place calls internally some people think they have been cut off if the line is quiet for a few seconds. Is there a way of getting comfort noise on the call? I'm using the STABLE release and cisco 7960 phones under FC-1 Cheers

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, I'm having some questions related to echo cancellation configuration on a Digium board enabled systems (B410P, TE420, TE420B, ....) for cases when a hardware ech canceller is present or not. I read in TEXXX manual that when setting echocancel=yes in chan_dahdi.conf on a VPMOCT64-equiped system, 128ms hardware echo cancellation was enabled. 1. I'm correct thinking that it is then

Hello, Which one of these ensures that SIP packets are sent and received in a secure format so that users using public wifi don't allow MITM type of attacks or others can't read the plaintext SIP packet info. VPN is not an option. Looking for 2nd most secure to VPN. P.S. Are both options part of the configs of Asterisk or need modules to be selected and installed before doing the

We are using res_fax_digium with a Sangoma PRI card on asterisk 1.8.13 The docs at http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf indicate v34 is supported, but when I enable it I get the message "res_fax_digium.c:1624 dgm_fax_new: V.34 not supported, will be ignored." Is v34 only supported with SpanDSP? Also, the res_fax.conf.sample does not indicate v34 as a valid

Hello list, I need to know about Asterisk's friendly nature with RTCP. I've phones which support RTCP and they connect to the outer world via multiple carriers. In one of my recent packet traces I've observed that the caller initiated a call with rtcp string in SDP while for the same call dialling our from Asterisk to the carrier has no RTCP string in SDP ! Can anyone please tell why

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

Bad day today. Why this new JIRA system not working. I have created issue and submit and i got blank page.. Please someone help me to create BUG!!!!!!!!!!! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110608/e99afa31/attachment.htm>