similar to: Asterisk 11 on Centos: Voicemail crashes when recording message

I am totally not understanding this : My IAX.conf : register => BOX-YOCAN:passwd at remote_asterisk_ip On remote Asterisk : *CLI> [Aug 30 20:37:07] -- Registered IAX2 'BOX-YOCAN' (AUTHENTICATED) at ip:4569 So this is normal... Now the following : [remoteasterisk] type=peer host=ip remote asterisk auth=md5 secret=passwd On the remote Asterisk : [BOX-YOCAN] type=user

Situation: * New Install of Asterisk * 7960 w/ SIP 7.4 Image * 7912 w/ SIP040406A * 3 Lines Defined on the 7960 (5104,3100,2100) Questions (configs are below): * Why won't the MWI light on the Cisco? I've tried: * mailbox=2100 * mailbox=2100@default * mailbox=2100@wvlandsales-voicemail * Does anything look goofy overall? :-) Thanks, George Sip.conf [2100]

Good day to all-- First off let me say that I have been very pleased with the mailing list. I have learned a ton of stuff just reading other peoples questions and comments. I really enjoyed the VOIP Conference call on Friday morning. Still working on figuring out the best approach to custom voicemail emails (the reason I joined this group); however, we have more pressing issues. I

Hi all, I have the following 7 x 7 matrix. ?I am trying to figure out how to label the columns to something more descriptive other than [,1], [,2], etc. I have tried the c(x,y,z,) function, but I get a error returned stating that my vectors need to be the same length. Do I need to convert this to something else such as a list and then repack it? Thanks, Kindra ?? ? ? ? Volume Time ? ? ? ?[,1]

Dear All, I have recently setup a completely new AD domain on my Linux server, running Samba 4.8.3. From the server, I can authenticate via kerberos and get users and groups through winbind etc. When I try to join a freshly installed Mac running macOS 10.13.6, I receive the error: "Unable to add server. Authentication server failed to completed the requested operation. (5103)" The Mac

I am using the tree package. One option is the cv.tree, which is supposed to run cross-validations on your tree models. Is there somewhere I can find some documentation on this function? I have the help file that comes with the library, but I need more, especially on what the output is. Thanks, Warren Schlechte HOH Fisheries Science Center 5103 Junction Hwy Mt. Home, TX 78058 Phone

On Sat, Jul 28, 2018 at 11:40:26AM +1200, Andrew Bartlett wrote: > On Sat, 2018-07-28 at 00:10 +0100, Phillip Potter via samba wrote: > > Dear All, > > > > I have recently setup a completely new AD domain on my Linux server, running Samba 4.8.3. From the server, I can authenticate via kerberos and get users and groups through winbind etc. When I try to join a freshly

Hello, did updates from 1.0.15 to 1.2.11 and is in most cases succesful. Sometimes dovecot leaves errormessage in Log: IMAP(username): FETCH [HEADER] for mailbox INBOX UID 5106 got too little data (copying): 516 vs 528 Until now only Inbox was effected and no other folders. I delete cachesfiles in ~mail/.imap to solve this but is there a way to do this automatical (self healing)? Dovecot

Hi all, I inherited a cpu-stats script from the previous *nixadmin at our department. This script relies on a rsh-command to get the vmstats from the remote machines and then using a perl script to push it to a web server. Now I''ve just added a new machine running CentOS 5.2 x64 to the script and I get the error message in the subject line; "poll: protocol failure in circuit

Any ideas why bind is putting the tmp files in the [chroot]/var/named directory and not in /tmp or /var/tmp? [root at devserver21 chroot]# Aug 15 14:08:21 devserver21 named[5101]: loading configuration from '/etc/named.conf' Aug 15 14:08:21 devserver21 named: named reload succeeded Aug 15 14:08:21 devserver21 named[5101]: dumping master file: tmp-XXXXQ5X9mC: open: permission denied Aug 15

Hi folks, I am having a small problem with asterisk-1.6.2.7 + app_fax on OpenBSD 4.7 -release. Everything seems to work fine. I have a macro which answers, receives the fax to a tiff, and then runs a script (mailfax) to convert that to pdf and email it. It all works perfectly except for some errors I am seeing in the console. After it hangs up I get a dozen or so messages in the cli

Hi list,* *I am not able to make video calls between two sip accounts. below is the information. please help me where I am missing the configuration.* Extensions.conf* exten => 111,1,Answer() same => n,Dial(SIP/2206,60,r) same => n,Hangup() *SIP.conf* [2218] type=friend secret=******* callerid="Virendra" <9172341457> host=dynamic ;

I was just on 70minute call (IAX2 -> Internet -> IAX2) and during that time I heard several "pops", or "clicks". Each time it happened, I saw the following message: Aug 19 15:36:36 NOTICE[1173711792]: rtp.c:429 ast_rtp_read: RTP: Received packet with bad UDP checksum Any ideas what causes these, and why they turn in to a "pop", instead of just silence, or a

Hello List. We are having some problems using t.38 together with a Cisco voice router at one of our providers end. We are using the new digium asterisk fax module to generate the fax, and when we use together with our internal Audiocodes Mediant 2000 gateways, we have no issues what so ever, and the faxes go right through. When we send faxes to our other provider, who has cisco hardware

To me it's a error if I can't complete calls using the ATA configured to use the g726 codec. I just tried it usign a sipura SPA-2000 (preferred codec: G726-32) and I received NOTICES and WARNINGS, but I can't complete a call. On a zap channel: -- Executing Dial("SIP/2007-e4d8", "Zap/1/2217008") in new stack -- Called 1/2217008 -- Zap/1-1 answered

Hi All, I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router with a PRI card in it, handing off to a PBX and vise verse. Calls in and out are working fine except for DTMF from Asterisk to the 2600. DTMF from the 2600 to Asterisk is fine. Here are the Asterisk console warnings I get when I send DTMF from Asterisk to the 2600: == Forcing Marker bit, because SSRC has changed Jun

Any one see these? Are they benign, or is some system tuning required to remove them? Can't seem to find a resolution in the archives. If you have a link, it would be appreciated. Jun 2 10:58:58 NOTICE[163044272]: rtp.c:470 ast_rtp_read: Unknown RTP codec 19 received Jun 2 10:58:59 NOTICE[163044272]: rtp.c:470 ast_rtp_read: Unknown RTP codec 72 received Jun 2 10:59:00 NOTICE[163044272]:

After just having update to the latest CVS I am getting the following message when I call VoicemailMain(): -- Executing VoiceMailMain("SIP/2302-1a12", "") in new stack -- Playing 'vm-login' (language 'en') -- Playing 'vm-password' (language 'en') -- Playing 'vm-youhave' (language 'en') -- Playing

On 6/26/07, JR Richardson <jmr.richardson at gmail.com> wrote: > Hi All, > > I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router > with a PRI card in it, handing off to a PBX and vise verse. Calls in > and out are working fine except for DTMF from Asterisk to the 2600. > DTMF from the 2600 to Asterisk is fine. > > Here are the Asterisk console warnings

Hello everyone! I tried to send a fax over SIP with an Asterisk Server in the middle (no Digium Cards, etc. installed, everything is SIP only, the PSTN-Gateway is external). Whenever I start sending a Fax to a PSTN destination, the Call gets answered and asterisk tries to build a native bridging: -- Attempting native bridge of SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 Then the following