similar to: Losing voice on Digium demo server - how to spot problem ?

Hi, if I change Iaxphone settings to qualify=yes it says it's unreachable. Can iax2 clients be monitored with qualify option ? Is this problem related to iaxphone ? Anyone sucessfully using iax qualify feature ? Regards, Rob.

Hi, I'd like to test my links with remote locations. I wonder if there are any echo asterisk server that could be called for quality estimation .... Regards, Robert.

Hello, I am using the vfs_gpfs samba module to map ACLs through samba. It works fine on files, but directory ACLs are ignored. Ex: getfacl /sb/share/myplace/ file: sb/share/myplace/ owner: root group: root user::rwx user:afrankel:rwx group::--- mask::rwx other::--- When I try to access this folder in Windows, I get permission denied. The same permissions on a files, I can open it / modify it

This is my first post to the mailing list and I guess it's a pretty stupid question but I can't figure it out. I hope this is the right forum for these kind of questions. Before I started using R I was using STATA to run a Wilcoxon signed-rank test on two variables. See data below:

Hi, I've spotted weird crash of Asterisk cvs Stable. I have defined queue in queues.conf : [prodaja] music = default announce = queue-markq strategy = ringall context = from-pstn timeout = 15 retry = 5 maxlen = 0 announce-holdtime = no announce-frequency = 30 announce-holdtime = yes monitor-format = gsm|wav|wav49 monitor-join = yes eventwhencalled = yes member => Agent/1000

Hi, I have strange problem. I have 1 SIP client (bt100) and 1 Iax2 client (IAXPhone): - when I call from Iax to SIP sound works - when I call from Sip to Iax sound doesn't work, I get : Jan 25 13:52:22 NOTICE[31334]: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/200/1 of format gsm since our native format has changed to ulaw Why is Asterisk not satisfied with gsm

[I post the same message on Ubuntu forum] Hi, I need help to fix a problem with Samba as PDC and Folder redirection on a Roaming Profile. I know it's not the best place to post this, but I don't know any better place. So if you have suggestion, tell me. Here my problem : I'm in a testing environement with a Samba server setup as a PDC with some share (netlogon, profiles) to support

I sometimes receive e-mails I post to this list, like I did the bug I reported about X problems, but other times not, like my first post about that before I reported the bug. That one appeared in gmane, but Evolution never picked it up. And today I've not seen either of my posts but I know it got there because I did see a reply to my post about the flash plugin and it had a snippet of my post

hi: I setup a small samba 4.0.5 AD DC server. my client is windows 7 and linux. and I use windows 7 with remote managment tools to manage rfc2307 account seetings of samba4 DC. I hope my users can use the same account to use windows and linux. samba4 DC provsion command as below: samba-tool domain provision --use-rfc2307 --function-level=2008_R2 --interactive and smb.conf global

First of all i would like to thanks Timo for the excellent job done in dovecot. We are using it in you prodution server with about 5 millions of emails and it is pretty good, only a performance problens but i guess it is realated to ocfs2 that we are using. So lets begin: We are using dovecot 2.0.6 with maildir in an ocfs2 partiton. It is pretty slow access in peak time, but it is related to

Announcing a new Windows-based IAX/IAX2 client. Please download it and give it a try. Let me know about any bugs, and any missing features. I have yet to come up with a catchy name for it, so at this point it calls itself IAX Phone. (Suggestions? Non-derogatory suggestions, preferably). Download: http://www.sokol-associates.com/Downloads/IaxPhone.zip Reference & Support Page:

Hi, we're testing Asterisk 1 RC 2 behind ordinary router and NAT. Since we're sharing network with web server it seems like voip packets are not coming through fast enough (Digium demo dies after few seconds...). It's the same if I make direct calls (passing Asterisk) so we conclude it's network problem - it also work normally outside our router... I wonder what solutions can we

Hi, I'm trying to implement dynamic routing of incoming calls to local extension if previous outgoing call was unanswered. But after I do Goto to s-NOANSWER, variable ${EXTEN} changes to 's-NOANSWER'. I guess this is normal, but I don't understand why ? How to workaround on this one ? Thanks in advance, regards, Rob. [outbound-capi-ISDN] exten => _0.,1,NoOp(Calling ISDN

Sirs, I have recently been interested in your Maxstat. I have computed with my own programme the ranks (by using the Kaplan-Meier method and the log-rank test) with the formula (Observed-Expected)/(SQR Var). The results are similar but not exact to the M value obtained with the Maxstat. I would like to know whether you are using some correction or adjustment in computing the different ranks. Thank

Hi, I have following problem. Asterisk is connected to ISDN router on BRI interface. ISDN PBX is connected to another channel of BRI interface. Now I'd like to route all incoming calls first to Asterisk and then if caller wants to talk to extension on ISDN PBX then I'd like to route call to another capi channel but free the current one. Is this possible at all or do I need to take 2 capi

Hi, I wonder which PRI interface card is most stable and supported for EuroISDN and Asterisk ? Are they stable enough ? Any tips ? Thanks in advance, regards, Rob.

Hi, has anyone connected Asterisk to Cirpack softswitch sucessfully ? Any howto or more info about needed Asterisk SW and setup ? Thanks in advance, regards, Rob.

Hi, I'd like to use DISA properly for my case - I'd like to handle it right, if user when in DISA doesn't dial any number - how does Asterisk return from DISA cmd ? I'd like to dial some default number if user doesn't dial anything or give him some message - but I don't know what gets executed after DISA if nothing is dialed .... I'm reading this on wiki, but

Hi, I've updated my Asterisk 3 times with : cvs checkout -r v1-0 zaptel asterisk asterisk-addons and then do cd asterisk make clean && make && make install make samples make progdocs and then when I run Asterisk I get : Asterisk CVS-v1-0-02/11/05-01:46:25, Copyright (C) 1999-2004 Digium. Is this a bug in CVS handling or am I doing something wrong ? How to check which

We've bought the G729 codec for lowering SIP bandwidth usage (we're using grandstream phones) and we're quite happy with it up until I tried using IAXPhone 0.2.0 build 116 with my asterisk 1.0.0 installations. Weirdly enough, calls from IAXphone to the GS phone work just fine. So are calls from both phones to voicemail. And from both phones to analog phones connected to FXS ports.