similar to: Multi-line, multi-registration phones

Okay, so like this? PHONE1+AD0-SIP/2000 PHONE2+AD0-SIP/3000 PHONE3+AD0-SIP/4000 ALL+AD0AJAB7-PHONE1+AH0AJgAkAHs-PHONE2+AH0AJgAkAHs-PHONE3+AH0- Then you would have Exten +AD0APg- s,1,Dial(+ACQAew-ALL+AH0-,20) Is that right? I have read about the Macros but don't understand their use. Could someone provide an example? Sorry about the newby questions... This will hopefully be my

Here is an example of a couple of macros that help me where I have a SOHO with a home phone line and a work phone line. If I pick up line 2 my work line I would prefer the call I make to go out my office phone line same with if I pick up line 1 my home phone line I would prefer it go out my home line but want it to roll if needed. So with this little macro it is possible for that to happen.

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 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

I am trying to get asterisk to email me my voicemail as attachments. What am I missing? Where do I tell it to go for SMTP services? Voicemail.conf: ; ; Voicemail Configuration ; [general] format=wav49|gsm|wav serveremail=pbx.agtcorp.local attach=yes maxmessage=180 skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 append=yes [default] 100 => 1234,Sean Garland,sean@siskiyoutech.com

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: >

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

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 =

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

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]:

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

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

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

Dear Colleagues, We are using the phylog.gls.fit() function from the R package "PHYLOGR" (Diaz-Uriarte R, Garland T: PHYLOGR: Functions for phylogenetically based statistical analyses. 2007. Available at [http://cran.r-project.org/web/packages/PHYLOGR/index.html]) to correct for lack of independence between data points. (In our particular case, the lack of independence is due to

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>>

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

Where do you set the outgoing mail server for use with asterisks mail system? I have entered the info in the voicemail.conf file correctly, but I am still unable to get the voicemail messages via email. I ran a tcpdump on the system while calling in and leaving a voicemail and I don't even see the system try and contact a mail server. HELP!!! Thank you all in advance. Sean Garland

I have the standard parking.conf but extension 700 doesn't show up in my dialplan.... Why? I can dial 701 which tells me that I don't have any calls parked there. 700 just gives me invalid extension noise.... Should I have extension 700 defined elsewhere? Thanks parking.conf [general] parkext =a 700 ; What ext. to dial to park parkpos => 701-705

Hello, 1. Is it possible to install several SIP softphones on the same PC, have them registered to the same Asterisk server and attribute to each softphone a specific extension, ringtones or call forwarding rules ? 2. Is possible to do the same with SIP hardphones ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: