similar to: RX/TXgain on bristuff/zaptel ?

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

Hello, i'm using a TE410P on some E1/PRI with EuroISDN and experiencing a few audio quality problems with current CVS (both zaptel and asterisk) and the following network ISDN public SIP/zaptel network ---- pri --- ASTERISK GW --- iax --- ASTERISK PBX --- PHONES w/ any codec the rx (public network to local

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 looks like someone decided to bundle a patch that hasn't been merged yet. Good for testing, not so good for initial impressions. In /etc/asterisk/asterisk.conf add or uncomment this: [options] ;silence_suppression=yes And see if that helps. You need a timing source for it to work, which is why it is disabled by default, but the logging might be a bit chatty in any case. Dan

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

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

Hi, I have an old Fileserver which is working correct: This is the smb.conf: [global] security = ads realm = EXAMPLE.COM workgroup = example winbind refresh tickets = Yes winbind use default domain = yes template shell = /bin/bash idmap config * : range = 1000000 - 1999999 idmap config ZFD : backend = rid idmap config ZFD : range = 0 - 200000 hide dotfiles = yes server string =

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

Hi, Oh sorry, of course: The running os is debian 9.9 and I'm running the sernet-samba-ctdb in version 4.9.11-15 This is my configuration: [global] ??? winbind refresh tickets = Yes ??? winbind use default domain = yes ??? template shell = /bin/bash ??? idmap config * : range = 1000000 - 1999999 ??? idmap config ZFD : backend = rid ??? idmap config ZFD : range = 0 - 200000 ??? hide dot

Hai, Great to hear i could help one with a gluster problem :-) And ofcourse your allowed to keep us up2date. So yes, plese, by doing that and sharing the configs it might help other people. Greetz, Louis > -----Oorspronkelijk bericht----- > Van: samba [mailto:samba-bounces at lists.samba.org] Namens > Benedikt Kale? via samba > Verzonden: woensdag 28 augustus 2019 17:37

Hi, I don't have the user root. No changes :( Sometimes a user gets permissions, sometimes not. This net conf is now running: [global] ??? winbind refresh tickets = Yes ??? winbind use default domain = yes ??? template shell = /bin/bash ??? idmap config * : range = 1000000 - 1999999 ??? idmap config EXAMPLE : backend = rid ??? idmap config EXAMPLE : range = 500 - 200000 ??? hide dot files

Hi I have a requirement to bridge Digital ISDN call through an asterisk box but no matter what I setup in the dial plan the second leg of the zap bridge is always set to Transfer Capability of SPEECH, I wondered if any one has come across this and managed to fix it? Thanks in advance for your help Robb

Hi, sorry to bother you: I have three AD in the domain. They all deliver different IDs: root at addc2:~# id testuser uid=3000155(EXAMPLE\testuser) gid=100(users) Gruppen=100(users),3000155(EXAMPLE\testuser),3000036(EXAMPLE\TEAM1),3000014(EXAMPLE\gesch?ftsstelle),3000001(BUILTIN\users) root at addc3:~$ id testuser uid=3000133(EXAMPLE\testuser) gid=100(users)

Hi, I am a beginner in asterisk. Implementing it in my dept in India using TDM400b card with asterisk, zaptel, libpri version latest of CVS HEAD Callerid on my system is coming tough. Asterisk doesnot finishes the callerid spill and Cancells it. After going through code in Callerid.c and chan_zap.c I found that my line is providing caller id of length 8867. Flow enters in zt_call and

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

Hi, I installed Digiums Free Fax for Asterisk and found out, that it automatically retries failed faxes, is there a way to stop that? Thanks Markus

Hi again, regarding my post "plenty of vacuuuming process" a "gluster volume heal" seems to improve the situation. But I still have a strange problem: Sometimes a user don't have permissions to? a restricted folder when h connects to a share or logs in at a windows client. In some times all permissions are granted. If the user creates a file, the user and group is

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, Is there a variable that contains the Sip Peer name? I was using ${CALLERID(num)} for outgoing calls, but when a call is being transferred, that variable contains something else. I need a variable that is always set to the SIP Peer's name. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, yes, I did. I get the same results with "getent passwd testuser" on each node. /etc/ctdb/nodes and /etc/ctdb/public_addresses is exactly the same on each node On each node sernet-samba/stretch,now 99:4.9.12-15 amd64 is installed Yes, I read the documentation. It is strange, that another cluster in another office configured that way is working perfect ;( The load is not as high