similar to: 1.2.1 "Silence suppression is disabled" whatthehell?

I upgraded from 1.0.9 to 1.2.1. In 1.0.9 everything worked perfect. Now, I call in my IVR, and after navigating in menus when I get dialtone for dialing extension, Sound is choppy and I get bunch of messagess: -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=30) -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=30) -- Silence

I upgraded from 1.0.9 to 1.2.1 My IVR which worked perfectly on 1.0.9, now hangup with no reason (at least I could not find a cause) When this hangup happen, I can read: == Auto fallthrough, channel 'IAX/user-20' status is 'BUSY' This happening also with ZAP channels I'm really disappointed with 1.2.1, what is benefit from upgrade if I must spend couple days to get my system

I'm using bristuffed Asterisk with ISDN/ZAPHFC I have VERY ugly (outgoing) sound through ISDN/HFC if echocancel=yes in zapata.conf, but without echocancel I have bad (incoming) echo Through PSTN/FXO sound is ok with or without echocancel. I tried other echo cancellers (in zconfig.h) two times: ECHO_CAN_KB1 (this was default) ECHO_CAN_MARK2 ECHO_CAN_MG2 after any change I compiled (make

Somebody must have! Please, send me a g723 and/or g729 (for Asterisk) to pisac@hotmail.com (antispam subject: codec) Thanks, thanks, thanks... :-)

I upgraded from 1.0.9, to 1.2.1. I was using this line exten => s,1,gotoif($[${CALLERIDNUM::3} = 066]?mycity,1:other,1) it selecting calls if callerid begins with some number pattern (from some city) But, it's not working anymore in Asterisk 1.2.1 when I test this with noop(${CALLERIDNUM::3}) I get full callerid, not just first 3 numbers like it use to be on 1.0.9 Why?

Do bristuffed zaptel (zaphfc) supporting rxgain/txgain in zapata.conf? I'm changing rxgain in zapata.conf, and reloading zaptel, but sound level on ISDN(HFC) is always the same (loud).

Voice-mail messages aren't sent to e-mail address. I have two Asterisk servers, first one is upgraded from 1.0.RC2 to 1.0.9, and second one is from 1.0.7 to 1.0.9. Both Asterisk have EXACTLY same "voicemail.conf" configuration, but second Asterisk don't sending voice mail messages through e-mail! I'm using almost default "voicemail.conf" with just one mailbox

If my extension is 22, and voice mail access number is 909, then with exten => 909,1,voicemailmain(s22) I can access voice mail 22, without number and password prompt. But, I want that every extension can access its voice mail without number and password. So, when I put exent => 909,1,voicemailmain(${calleridnum}) voicemail want only password. I want to eliminate password too, so when I

Hi all, I'm getting some noise gate like effects on our sip lines & I think I need to disable silence supression, I'm searching docs & not finding where this can be set, does * have a setting to turn this off? basically what's happening is when we stop talking, the other end hears total silence, but when we talk, they can hear the background noise in the office, this sounds odd

Juan, I am running a Calling Card application on a Dell PowerEdge 2850 with Asterisk 1.0.7. Recording conversations I have seen on my server causes the processors to burn more than necessary so I would recommend what William from Signate recommended: " Consider saving recorded calls in a database on a separate server. It will be simpler to build a retrieval interface that does not

Why Asterisk showing (on SCCP and H323 phones) different CID related to type of Incoming channel: If incoming channel is SIP, on phone is displayed CALLERIDNUM If incoming channel is ZAP, on phone is displayes CALLERIDNAME It vas very frustrating! I lost couple hours of my time to find that my dialplan is not faulty, but asterisk is!

Do somebody using CISCO AS5350 with Asterisk? Which protocol do you using: H323, MGCP, SIP? This direction: [12sp->Asterisk->h323->as5350->isdnPSTN] is ok But reverse: [isdnPSTN->as5350->h323->Asterisk->12sp] cannot hear 12sp, but 12sp hear PSTN (codec g711u) -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm using sipgate.de as my sip provider. When I'm using xlite as client on sipgate.de, everything works fine: I call number, hear ringing (real progress tone form called party, not one generated in xlite) and then talking with called person. But, when I'm using Asterisk as sip client on sipgate.de, I don't hear progress tones: I hear only one (locally generated) ring tone, and

When incoming DATA call arrive on ISDN BRI, asterisk (zaphfc) recognise that this is DATA call, but answering anyway (playing IVR messages, etc...) How to stop that? I want that only VOICE calls are answered, and DATA/FAX to be ignored. (I'm using Asterisk 1.2.1 Brisftuffed 0.3.0-PRE-1f, ZapHFC) Log: -- Accepting data call from 'XXXXXXXX' to '3001' on channel 0/2, span 1

Hello, I run Asterisk in a 100% VOIP installation with the Polycom IP-500 phones. Every once and a while I have problems with either dropped calls between Asterisk and my provider, or invalid RTP audio streams with phones behind NAT. I have had a few Asterisk developers look into my installation and even my provider check my setup but still am having problems. They tell me that I need to

Hello, I have several customers describing something like annoying "silence suppression". So I did some tests and I can confirm. After disabling echo cancellation in zapata.conf the "silence suppression" effect is gone, but there is a little echo of course. I do not have this problem on other boxes where I am using oslec as an echo canceller. All my calls are SIP to ZAP (E1).

Does the Asterisk server support VAD (aka Silence Suppression)? I want my SIP phones (7960's) to use VAD when dialing out the x100P interface. I know the phone can do VAD , can the Asterisk server be setup to do it? and if so, where do I set the configuration? Thanks Lee Goodman -------------- next part -------------- An HTML attachment was scrubbed... URL:

Let's say that I have a call coming in to Asterisk through a TDM400P and going out through SIP to someone on the Internet. Is there any configuration option that would allow me to do silence suppression on the RTP stream generated by Asterisk on behalf of the TDM400P connected user? SIP phones allow me to do this easily, but I'd like to be able to conserve upstream bandwidth on calls that

We have an Asterisk server that speaks IAX2 to Magrathea to get to the PSTN. Our local phones are a mix of Cisco 7940s and Grandstream BT100s all configured for SIP with silence-suppression disabled. Everything is configured to use a-law encoding. The version is: sip*CLI> show version Asterisk CVS-05/06/04-18:45:57 built by root@sip on a i686 running Linux Incoming callers are complaining of

Hi Friends, Currently i am using the asterisk 1.4.x version. In that i want to enable to silence suppression in the SIP calls. Please tell me the configuration changes to be done. Thanks in advance, balasam. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090106/dde500d5/attachment.htm