similar to: Voicemail problem

I have 2 sip phones and an analog phone attached to a Digium USB fxs device. I would like the analog phone to ring when transfers are made to it, but I don't want it to ring when a call comes in from outside, although I would like the person at that phone to be able to pick up the phone and answer the incoming call. Is that possible? Thanks Sean Garland, MCP+I, A+ Siskiyou Technology

It appears that my replies aren't getting to the list. Just testing to see what is going on... Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040105/cc18a485/attachment.htm

I am looking for common practice ideas on how to handle multiple line phones. Is it common with asterisk to have the lines appear as programmable buttons? Or to just have itcm like buttons and use the dial 9 approach? What I am specifically interested in, is to have my line one appear on the first button (sip polycom phones) line two appear on the second button, and use the third as an intercom

I have a very annoying problem that we hear on our end, but the other party doesn't hear. There are random beeps and echo type noises that occur. They are present during voicemails, and present on my end during calls. Is anyone experiencing the same deal? I have asked this a number of ways on the list, and never get a response... Thank you. Sean Garland Mount Shasta, CA

Hi Guys, My name is Marcias, and I am setting up for the first time an Asterisk PBX, I am learning as I go along. I have been able to download and install Asterisk, Libpri and I have been able to get Asterisk up and running. I have several questions: 1 .I can call the Asterisk server from my Xten phone and it picks up. I have 3 computers (one of them being the asterisk server) I can seem to call

Is there a way to append a 91 in front of all incoming caller id numbers? What I am interested in is this - when a call comes in and the caller ID comes across, it is in the format 9165551212. That would be fine, but if I want to call them right back, and I choose that call and 'dial' on my phone, it fails because it hasn't dialed a 9 first. I assume also that if there was a 9, it

Hi When i make a call using IAX2, the log of the remote asterisk say Nov 17 20:20:12 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Nov 17 20:20:22 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Nov 17 20:20:26 NOTICE[28686]: File rtp.c, Line 263

I'm getting some warnings: NOTICE[xxx]: File rtp.c, Line 264 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible Asterisk Version: CVS-01/06/04-13:50:26 Cisco ATA 186 version: v3.0.0 atasip (Build 031210A) Is this something I should be concerned about? Anyone know how to "turn off" the RFC3389 support on the ata 186? Thanks!

nslookup google.com ;; Got recursion not available from 192.168.1.236, trying next server Server: 192.168.1.235 Address: 192.168.1.235#53 Non-authoritative answer: Name: google.com Address: 172.217.4.78 Interesting little bit about the "Recursion not available" On Thu, Jun 30, 2016 at 9:52 AM, lingpanda101 at gmail.com < lingpanda101 at gmail.com> wrote: >

Hosts: 127.0.0.1 localhost localhost.localdomain localhost4 localhost4.localdomain4 ::1 localhost localhost.localdomain localhost6 localhost6.localdomain6 192.168.1.235 bus-ny-dc-01.domain.domain.com bus-ny-dc-01 Resolv: # Generated by NetworkManager search domain.domain.com nameserver 192.168.1.236 nameserver 192.168.1.235 Smb.conf: # Global parameters [global] workgroup =

from both DC: dig google.com dig @8.8.8.8 google.com First dig will use resolvers declared into /etc/resolv.conf. Second dig forces usage of 8.8.8.8. Both commands should reply the same things, on all DC. 2016-06-30 15:58 GMT+02:00 lingpanda101 at gmail.com <lingpanda101 at gmail.com>: > On 6/30/2016 9:57 AM, Garland McAlexander wrote: > >> nslookup google.com

Yes, it's set up with 8.8.8.8 I'm able to ping it without issues, jut not able to resolve anything externally. On Thu, Jun 30, 2016 at 9:09 AM, mathias dufresne <infractory at gmail.com> wrote: > To get recursion working with internal DNS you only need to set up: > dns forwarder = <IP of your main DNS> > > Is it configured? > > If yes and packets can go

It's samba internal DNS. Only one DNS zone, and it's domain.domain.tld. It'll function perfectly, and then cease to function at a random time. On Thu, Jun 30, 2016 at 5:31 AM, Mueller <mueller at tropenklinik.de> wrote: > What kind of DNS, bind or internal? > With bind an samba 4.3.4 I have an issue and I have to restart bind an > avahi: > s4slave named-sdb[8750]:

Hi All, I've got a newly created Samba4 domain. I'm running into a strange issue where my internal DNS on my first domain controller is "breaking" causing it to not resolve any external hosts. It'll still resolve internal hosts without issue. This is only on the first DC, the second DC is running perfectly fine and can access external hosts without issue. There is

On Wed, Jul 13, 2016 at 02:35:08AM -0400, Garland McAlexander wrote: > CIFS is single threaded, if you plan on using it with a large amount of Oh, "my" cifs as a protocol is multi-threaded. It's part of the Samba implementation that is single threaded. However, for large file I/O Samba is multi-threaded. Volker

I am getting (with older image): RFC3389 support incomplete. Turn off on client if possible How do I turn that off for the DG104s? Or if I can't how do I tweak asterisk? I see posts about ATA-186's having an audiomode, but the closet I came to was inbanddtmf. I tried =0 and =1, no effect. Thanks! -- Zot O'Connor <zot@zotconsulting.com> White Knight Hackers, Inc.

I have a ATA 186 with SIP firmware 3.1 when I changed the configurations to use the g.726 codec I received many erros and the calls doesn't work. I changed the fields: - LBRCodec: 6 <- the code for g.726 - TXCodec: 6 - RxCodec: 6 The errors: Jul 9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to calculate samples for format G726 Jul 9 13:15:37 NOTICE[1192491824]:

I still have problems with beeps and clicks on all my calls. I have polycom sip phones. I also can hear the beeps and clicks on some of my messages, which would lead me to believe that it is more of a decoding problem on the zaptel card. Any ideas? Thanks Sean Garland

Hi, forgot that... CIFS and NFS. May be we split both to different servers. One CIFS, one NFS. Regards . Götz Am 13.07.16 um 08:26 schrieb Garland McAlexander: > What shares will you be using? (NFS, CIFS, etc) > > On Wed, Jul 13, 2016 at 2:11 AM, Götz Reinicke - IT Koordinator > <goetz.reinicke at filmakademie.de > <mailto:goetz.reinicke at filmakademie.de>>

Hi, I copied a setup from an older 1.8.5 installation to an 11.15 installation, and I'm having problems getting call files to work. Here is the extension setup I'm using: [outbound-swift] exten => _[a-zA-Z].,1,Answer exten => _[a-zA-Z].,n,Playback(AAA/check_ip_failure) ;exten => _[a-zA-Z].,1,Swift("${EXTEN}") exten => _[a-zA-Z].,n,Goto(1) [mis-phone] exten =>