[asterisk-users] AUTO: Kevin Larsen is out of the office (returning Mon 01/08/2018)

I am out of the office from Thu 01/04/2018 until Mon 01/08/2018. I am out of the office and will have limited contact. For all emergencies/issues, please contact the helpdesk at helpd...@pioneerballoon.com or 316-688-8777. Note: This is an automated response to your message "[asterisk-users]

Re: [asterisk-users] Rewrite Outgoing Number

asterisk-users-boun...@lists.digium.com wrote on 12/14/2017 09:52:32 AM: > From: "basti" > To: asterisk-users@lists.digium.com > Date: 12/14/2017 09:52 AM > Subject: Re: [asterisk-users] Rewrite Outgoing Number > Sent by: asterisk-users-boun...@lists.digium.com > >

Re: [asterisk-users] Rewrite Outgoing Number

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 asterisk-users-boun...@lists.digium.com wrote on 12/14/2017 09:36:06 AM: > From: "basti" <mailingl...@unix-solution.de> > To: asterisk-users@lists.digium.com > Date: 12/14/2017 09:36 AM > Subject:

[asterisk-users] AUTO: Kevin Larsen is out of the office (returning 07/31/2017)

I am out of the office until 07/31/2017. I am out of the office and will have limited contact. For all emergencies/issues, please contact the helpdesk at helpd...@pioneerballoon.com or 316-688-8777. Note: This is an automated response to your message "[asterisk-users] [asterisk13] Multiple

Re: [asterisk-users] log incoming calls without answering

> I've already proposed your solution (is the most reasonable) but they > have more than 60 analogs lines (no faxes) and some of them terminate in > appliances like alarms, etc, so the solution must not touch in any way > the connection between the line and his termination: doing a analog to >

Re: [asterisk-users] log incoming calls without answering

> From: Fabio Moretti > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Date: 04/20/2017 03:26 PM > Subject: [asterisk-users] log incoming calls without answering > Sent by: asterisk-users-boun...@lists.digium.com > > Hi, >

Re: [asterisk-users] Cisco IP 8841 asterisk integration

> True agree, problem is somehow the people purchased am > supporting to overcome that. Trying level best... around 20 > phones has been purchased Ah, yes, the "we purchased these without consulting you, but it is up to you to make them work" school of thought. It often goes with,

Re: [asterisk-users] FAX CNG detected but no fax extension

> Hello, > I have a question regarding incoming fax to local file (on the > Asterisk server). > While the fax is received properly (I have the tiff file generated > as expected) I get the warning 'FAX CNG detected but no fax > extension' on the consol. > > If the fax is received ok then what

Re: [asterisk-users] Problem "re-parking" calls

> All; > I have a problem with regards to “re-parking” calls and I was > hoping someone could shed some light on the topic. Consider this scenario: > > (1) An inbound call comes in and the attendant answers it > (2) The attendant places the call on hold and the caller is sent to >

Re: [asterisk-users] Asterisk inside network. What phone works well?

> I have Asterisk running well inside our network. I did some > experiments exposing it to internet but had some issues: > 1. NAT issues (voice one way, etc). From what I understand double- > NAT users will always have something like this > 2. Immediately I see people trying to hack into. I did

[asterisk-users] AUTO: Kevin Larsen is out of the office (returning 09/06/2016)

I am out of the office until 09/06/2016. I am out of the office and will have limited contact. For all emergencies/issues, please contact the helpdesk at helpd...@pioneerballoon.com or 316-688-8777. Note: This is an automated response to your message "Re: [asterisk-users] Need ISDN call

Re: [asterisk-users] Getting better Caller ID

> Hello, > > We use Asterisk and as per book we use MAC addresses as user names. > So, when call coming in from outside (SIP trunk) - caller id is good. > > But when users calling each other on extensions - they see MAC > addresses. How would I make it so we see actual names instead of MAC >

Re: [asterisk-users] open source pbx free

> Anyone have any experience running an open source pbx and call > center solution?Need to start a call center of 10 users and i need help > > I have already installer a server with Ubuntu Server 14.04 , E1 installed > > Please advice me how to process from here > > Regards > >

Re: [asterisk-users] Is possible to use FXO Digium card like a Fax modem?

> There are also cheap USB fax modems that you can attach to an FXO > port and that works fine. All you have to do then is configure > asterisk to detect incoming faxes and route them to that port > (faxdetect=yes?). > > This worked great for me when I had all my incoming calls coming > over

Re: [asterisk-users] what to do when a sip password includes a semicolon

asterisk-users-boun...@lists.digium.com wrote on 03/11/2016 01:43:47 PM: > From: Saint Michael > To: Asterisk Users Mailing List - Non-Commercial Discussion > , > Date: 03/11/2016 01:44 PM > Subject: [asterisk-users] what to do when a sip

Re: [asterisk-users] 2 devices same *actual* extension - can it be done

> Can someone tell me if this is possible? > > I currently have a VOIP phone registered on an Asterisk PBX at a > remote location (working fine). > I want to install an Asterisk PBX at the local location. I will be > porting the current POSTS lines to SIP trunking. > So now I want the remote

Re: [asterisk-users] Passing Caller ID through Digium Gateway

> Hi All, > > I've setup a Digium G100 VoIP gateway to replace an internal PCI VoIP > card in our Asterisk PBX. When using the VoIP card the callerid entries > listed in sip.conf were displayed when calling someone over the PSTN. > Now, however, though the gateway it just displays the

Re: [asterisk-users] Queue logfile txt format in mySQL needed

> From: Thomas > To: asterisk-users@lists.digium.com, > Date: 01/21/2016 04:17 AM > Subject: [asterisk-users] Queue logfile txt format in mySQL needed > Sent by: asterisk-users-boun...@lists.digium.com > > Hello, > > Iam using queues and agents, thats OK. > > I have

Re: [asterisk-users] Forwarding call if extension busy

asterisk-users-boun...@lists.digium.com wrote on 01/04/2016 08:55:40 AM: > My question: > > - two extensions: and > - an active call on > - incoming calls to should be forwarded to (call advice!) and > > I know how can I forward an incoming call to more than an

Re: [asterisk-users] Forwarding call if extension busy

> Kevin Larsen <kevin.lar...@pioneerballoon.com> schrieb: > > > I am not sure if I completely understand what you are trying to do, but it > > sounds like you want to query the DEVICE_STATE function. > > IT WORKS > > Thank you very much! > Glad I

[asterisk-users] AUTO: Kevin Larsen is out of the office (returning 10/24/2015)

I am out of the office until 10/24/2015. I am working in Mexico with limited availability. If the matter is urgent, please contact the Pioneer Helpdesk. Note: This is an automated response to your message "Re: [asterisk-users] Live Recording on the NAS?" sent on 10/15/2015 1:55:13 PM. This

Re: [asterisk-users] Asterisk => Mediant 1000 (AudioCodes) => PSTN (E1)

> > Does anyone have any information for me? > > > Welinghton. > > > > Citando Welinghton Magno Guimaraes : > Hello! > > I am setting up an Asterisk server with a Mediant 1000 (Audiocodes) > to make external links. Does anyone have any manual or

Re: [asterisk-users] Share agents state?

> Is it possible to share all agents state? if an agent is on the > phone on a queue on one of the Asterisk servers, other servers will > need to about it and therefore, will be able to operate adequately? > For instance, an agent is a member of two queues (app_queue > realtime) and those

Re: [asterisk-users] How to integrate Asterisk with XMPP

> > How to integrate Asterisk with XMPP ? > What you are asking for isn't a simple question to answer. What exactly do you want to accomplish by integrating XMPP? Shared states among multiple extensions? Passing messages between extensions? Depending on what you want and what infrastructure

Re: [asterisk-users] Receiving faxes with spandsp question

I’m trying to add fax functionality to my asterisk installation. Right now I’m focusing on receiving faxes. This is not explained in a book, but I assume that I can use same context, add “fax” extension and if someone calls to send fax - it will autodetect. Right? Per book, I made

Re: [asterisk-users] asterisk email to fax

Since the O.P. said he's using it for his home office, I think he'll be able to control user expectations :-) I provide tech support to my parents on all their computers. The amount of annoyance I have dealt with in the last few months over the fact that a recipe program and various card

Re: [asterisk-users] FXS Solutions for modems and other non jitter tolerant devices

The legal and medical communities still seem to prefer faxing, in the ( mistaken? ) belief that it is more secure. In fact the medical community is fearful of the legal beagles. These groups are really slow to change. At least in the USA The couple of times I have received medical faxes

Re: [asterisk-users] small homebrew pbx

I don't know this 'translates' to Italy, but this is what I would advise somebody in the US to consider, assuming you have a reliable Internet connection. 0) I hope you mean you want to run Asterisk at home instead of 'Asterisk at Home.' A@H was an ancient distribution from around

Re: [asterisk-users] Am I cracked?

Very strange... I ran the Asterisk CLI for other tasks, and suddenly I got this message: == Using SIP RTP CoS mark 5 -- Executing [000972592603325@default:1] Verbose(SIP/192.168. 20.120-002a, 2,PROXY Call from 0123456 to 000972592603325) innew stack == PROXY Call from 0123456

Re: [asterisk-users] Am I cracked?

OK, I set alwaysauthreject = yes and I discovered a allowguest, which I set to no, too. The PBX is behind a Firewall and I just allow UDP 5060 and 1-10100. Now I log the SIP-pakets coming from Internet, too... Hopefully I solved my problem... Make sure you have solved the problem. You

Re: [asterisk-users] Am I cracked?

Make sure you have solved the problem. You don't want to get hit with a phone bill for calls from your location to Israel. Basically, they are hoping that you are running the equivalent of a mail server open relay. They are trying to use you to dial out to another number. You don't

Re: [asterisk-users] RES: RES: How to invoke a binary file from the dial plan?

I love this question, simply because it allows me to talk about one of the neatest features I programmed into my system that barely anyone knows exists. Plus it lines up pretty much exactly with what you are trying to do. We have our gate control system tied into our Asterisk phone

Re: [asterisk-users] Forward loop protection...

Deciding on the mailbox to use is problematic! The dialed-party may be away for an extended period and wants voice mail handled by the forwarded-to party. And then you have the users who would work around this by sharing their voicemail passwords. Not quite as bad as sharing your computer

Re: [asterisk-users] RES: RES: How to invoke a binary file from the dial plan?

Hi Kevin. Thank you very much for the hint! It worked very well! Your example ' exten = 1234,1,System(echo This is a test / var/log/asterisk/test.txt) ' executes when the SIP client (my softphone Jitsi) sends a SIP INVITE to asterisk. So, the softphone tries to establish a

Re: [asterisk-users] RES: RES: RES: How to invoke a binary file from the dial plan?

Hi Kevin. Thank you again for help me! In my case, in the final application for smartphones or in a softphone for PCs, there will be a button on the GUI and the user will have just to touch it, and the door or gate will open. I mean, during an ongoing call, the callee will see a

Re: [asterisk-users] Forward loop protection...

Ia had a server overload today because someone did a call forward to their own extension. To do a call forward I write a key called CFWD with the extensión number and number to dial . The main script tests if the key/value exists and dials the number stored in the database. What

Re: [asterisk-users] How to invoke a binary file from the dial plan?

Hi everyone. I'm new with Asterisk and I have to create a dial plan that will invoke a binary code. That is, asterisk will execute a program in the same machine. How to do it? Let me explain what I have to do: In the project that I am currently working, there is smartphones, SIP

Re: [asterisk-users] RES: How to invoke a binary file from the dial plan?

Ok. Thanks for the hint. But, what exactly is a System() dialplan application? Is it a kind of command that i can call in dial plan? I will look for System() related to dial plans. From the Asterisk CLI type: core show application System It will print out the syntax for the command. One

Re: [asterisk-users] Forward loop protection...

The loop checking is a bit more challenging than that. If Bob forwards to Fred and Fred forwards to Sue, all is well when Bob and Fred head out for a beer. A little later, we’re in deep doo-do0 when Sue forwards to Bob. Could this possibly mean that any person who has CF set should never

Re: [asterisk-users] Signaling incoming call

Hi Kevin! Thanks! It works! I can set the name of the line with CALLERID(name) and see the caller number, too. And, it the number is in the address book, I see the name, too. Perfect! Glad it worked for you. I usually leave the number untouched, but will manipulate the name to suite

Re: [asterisk-users] Signaling incoming call

Hi Steve! Thank you very much! It seems to run! I wrote that: exten = _0049351333,n,Set(__ALERT_INFO=Bellcore-r3) exten = _0049351333,n,SIPAddHeader(Alert-Info: http://www.notused.com \;info=alert-external\;x-line-id=0) and the phone rings with another melody. Very curious

Re: [asterisk-users] Peer is UNREACHABLE

What kind of phone are we talking about, both yours that works and your wife's that does not? Right! Can you ping the unreachable phone and does it respond to a ping? I can ping both phones from the VM Many phones will have a network test function built in to them to help you

Re: [asterisk-users] Peer is UNREACHABLE

I have a problem and I hope someone can help me... I configured an Asterisk on a VM to serve more accounts and act as a proxy to other SIP-providers. The first account running on my phone works without any problem. A second account, running on the phone of my wife, is always UNREACHABLE.

Re: [asterisk-users] Peer is UNREACHABLE

Darryl Moore dar...@moores.ca schrieb: I'd start by turning on sip debugging in asterisk sip set debug ip [your_phone_ip] Really destroying SIP dialog '490d1996593c8e11217828b71aae5c4d@172. 16.34.133' Method: OPTIONS Reliably Transmitting (no NAT) to 192.168.200.11:5060: OPTIONS

Re: [asterisk-users] Peer is UNREACHABLE

No, I'm not sure. And no, I can't make any call, right now... At least, not connected to my Asterisk... If I connect it to the other VM with AsteriskNOW I can call my Twinkle, but NOT my phone connected on my Asterisk, using the proxy. I can see that in the log: [May 28 22:49:51]

Re: [asterisk-users] Asterisk as Proxy and more device for a number

I'm very new in Asterisk and VoIP, and of course I have a problem... :) Well, my problem is, that Deutsche Telekom wants me to change my ISDN to VoIP... :( I must do that, since I have no alternative. Well, I have now two VoIP-phones (Thomson ST2022 and KE1020A). I can configure my two

Re: [asterisk-users] Asterisk as Proxy and more device for a number

Maybe I got it... I installed an asterisk on a VM with Ubuntu 10.04 and I got it connecting to another Test-VM with AsteriskNOW and with an italian VoIP-provider. The very difficult was to understand, that my phone just can manage ONE profile at time, so I had to configure Asterisk to

Re: [asterisk-users] Phone provisioning template Snoms

I am looking for a phone provisioning template for Snom phones, Yealinks and Polycoms. I am always doing deployments of many phones and usually configure each phone one by one for each installation. Any help will be highly appreciated There’s some excellent documentation about

Re: [asterisk-users] Multicast to polycom from asterisk

I am using 11.17.0 - and MulticastRTP. Doesnt seem to work with polycom phones as other devices receive my multicast just fine. Is there something special to do to get multicast working with polycom phones? (other than enable multicast on the actual phone). Didn't see if anyone had

Re: [asterisk-users] Multicast to polycom from asterisk

I hesitate to promote the name here since this is non-commercial discussion... but Polycom... Polycom phones... If mentioning Polycom is OK, I think mentioning a possible commercial solution is OK. In that case, the product in question is the Algo 8180 SIP Audio Alerter. I will

Re: [asterisk-users] Determining if a queue member is paused in Dialplan logic. [1.8]

asterisk-users-boun...@lists.digium.com wrote on 03/25/2015 01:38:26 PM: I'm looking at enabling autopause on one of my queues where my queue members are bad about leaving their desks without pausing. The problem I see is that when the queue pauses an Member it doesn't jump into the dialplan

Re: [asterisk-users] [OT] switches

so how does a client pc find the server if there's no NAT? by IP address?? That makes no sense, to me, if the switch isn't assigning addresses. Switches have a MAC table that keeps track of which MAC addresses are on which ports. That's how they decide where to route packets.

Re: [asterisk-users] System() command refuses to execute bash script

asterisk-users-boun...@lists.digium.com wrote on 03/02/2015 08:27:07 AM: From: Stefan Viljoen viljo...@verishare.co.za To: asterisk-users@lists.digium.com, Date: 03/02/2015 08:27 AM Subject: [asterisk-users] System() command refuses to execute bash script How can I use System to run a

Re: [asterisk-users] BlindXfer Sensitivity

Hi Guys We have a client running on a polycom vvx400 IP phone on our asterisk 1.8.18 system The issue we have is the switchboard lady uses ## to transfer calls but sometimes it just does not work and just plays the DTMF tone to the calling party. Is there any way to adjust the

Re: [asterisk-users] Is Asterisk a Linux only system?

I know that it runs on other systems but do other ports get the same attention? I have been running it on a NetBSD server for about a year now and while it mostly works it just crashes from time to time with no explanation or core dump. I have improved the situation by expanding my

Re: [asterisk-users] JITTERBUFFER function

WTF is a jitterbuffer? http://lmgtfy.com/?q=jitterbuffer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

Re: [asterisk-users] Dialing from phonebook, and hiding the dialed number from the user.

Hi, does anyone have a recommendation for a SIP phone, which allows dialing from a phonebook, and hiding the dialed number from the end users? Also from the call history of course. It seems Mitel can do this, and I have a use case where this is a requirement. I don't know about a phone

Re: [asterisk-users] Polycom SoundStation 6000 Dropping Registration

asterisk-users-boun...@lists.digium.com wrote on 01/23/2015 10:24:24 AM: Hello, I'm having a problem with a few Polycom SoundStation 6000s. Everything works fine, but they drop registration to asterisk after about maybe 30 minutes – the phone does not re-try to register and if you try to

Re: [asterisk-users] About voip gateway

I want to create a voip service, I do not know much about it, but the first thing I want to know if more than one client can make a call at the same time through internet to the PSTN, and what gateway should I use for this. I think the first recommendation any of us will have is to

Re: [asterisk-users] OT - Is T.38 possible on SPA8800 FXO port ?

I know all this. My question came from the fact that as strange as it may seem, SPA3102 and similar products do not offer the SIP features depending on terminating/originating port. More precisely, when a SIP fax call comes in through an FXS port, it triggers T.38 while it doesn't trigger

Re: [asterisk-users] E1 - Cisco - Asterisk and vice verso

Hi, Change my Dynastar E1 gateway to Cisco with E1 module, but can't make easiest dialplan. All my routing i made on asterisk, so i need that cisco all calls from E1 send via sip to Asterisk and all calls came from Asterisk by sip send to E1. From E1 to Asterisk already work, but

Re: [asterisk-users] AstriDevCon 2014:Agenda item Deprecate AMI/AGI(Ben Klang)

no file to forward would cause a crash, but other than that, I haven't seen any problems in normal day to day usage. I always thought that the general consensus was that the 11.x series was quite a bit more stable than the older versions. Kevin Larsen

Re: [asterisk-users] Asterisk- VoIP Phones ring from Ext. 101 but that Ext. does not exist in extensions.conf

Hello, a user outside the office regularly gets a call from ext. 101 but that extension does not exist in my extensions.conf. when the user pickup the phone no one answers. Any Idea how to fix this issue? that user uses Polycom SP 450, First thing to look at is at the time the user

Re: [asterisk-users] Record ANSWERED call

The problem is it records all incoming calls include those with the disposition of NO ANSWER, FAILED, BUSY, UNKNOWN.. For example the NO ANSWER call will leave a 44byte wav file in my ${RECDIR} How can I record only the calls with the disposition of ANSWERED? May be I should run a

Re: [asterisk-users] Call Flow Documentation Tools

asterisk-users-boun...@lists.digium.com wrote on 09/12/2014 09:07:36 AM: I have been researching software for documenting pbx call flow paths and I was just wondering if anyone out there is using anything they have found particularly useful or cool. I am looking for something preferably

Re: [asterisk-users] Special functionality for Secretary/Boss

asterisk-users-boun...@lists.digium.com wrote on 09/04/2014 11:57:40 AM: We are currently migrating from a Nortel pbx to Asterisk and we have been able to convert most of the functions that people are used to but there is one I have no clear idea how to do. The scenario is: Boss

Re: [asterisk-users] XMPP + Asterisk integration - a practical and simple example

http://www.mundoopensource.com.br/en_page_xmpp_asterisk_pratical_example/ Wish I had seen this when I was setting it up on my systems. Played around quite awhile using something other than OpenFire and couldn't get it working no matter what I did. Switched to OpenFire and while it wasn't

Re: [asterisk-users] OT: Question on Caller ID (Spoofing calls with Asterisk)

I got a call from an overseas call center telling me about the problems with the Windows machine I was using. They wanted to remote in and fix things for me ... (Ignore the fact I use a MacBook Pro or an ASUS laptop with Debian). What I found curious was the caller's name was Asterisk, and

Re: [asterisk-users] FYI: Block Comments

The configuration parser can do a lot of things. Out of curiosity amongst those reading this - how many of you know about templates? I use templates and wish the realtime parser would understand them as well.-- _ --

Re: [asterisk-users] Asterisk 12 - queue variables not passed to local channel

Asterisk 12.5 I'm using AMI to initiate a call me now feature from the web site. The AMI looks like: Action: Originate Channel: Local/s@callmenow Context: dial-to-customer Exten: s Priority: 1 Async: true Variable: CHANNEL_TO_CUSTOMER=SIP/voipms/111222 Timeout: 99 Dial

Re: [asterisk-users] Better info on call failure

asterisk-users-boun...@lists.digium.com wrote on 08/13/2014 08:31:01 AM: From: Nick Olsen n...@flhsi.com To: asterisk-users@lists.digium.com, Date: 08/13/2014 08:31 AM Subject: [asterisk-users] Better info on call failure Sent by: asterisk-users-boun...@lists.digium.com Hey everyone,

Re: [asterisk-users] Sending and receiving fax with Digium FFA

Hello. I've been trying to setup Free Fax for Asterisk on a Debian machine with Asterisk 1.8. I have managed to register and installed the Digium modules. Sending and receiving through it have resulted in failure. The output of fax show capabilities is: Registered FAX Technology

Re: [asterisk-users] The plain old PBX functionality

I am not sure why a previous response refers to this module as 'toxic'. It is a free to use module which allows a host of Digium phone features to be quickly implemented with Asterisk, like security-enhanced auto provisioning. Without creating a large off-topic response, there is a segment

Re: [asterisk-users] The plain old PBX functionality

back in the old analog telephony days there was digital PBX-es and digital system phonesets. This phonesets have had many individual illuminatable buttons connected with extensions. The PBX can show on the buttons if some extension is ringing (blinks) or busy (constant light), and the user

Re: [asterisk-users] Loud Ringers and paging systems...

if you use a papt2 or so spa2101 then you could have alert info set to different lengths or styles of ringers i use that in a dorm with phones and have the phones ring short rings at night so it wont wake up the students I do not use either of those devices, but after posting this

Re: [asterisk-users] Loud Ringers and paging systems...

Will your approach handle ringing more than one of the three extensions simultaneously? --Don Not if they are in the same paging zone, but neither would using the night ringer function on the pa system, so I consider that acceptable. Not even sure what would be considered correct in

Re: [asterisk-users] Unregistered ports on SPAxxxx

I've got a few devices, SPA112's and SPA8000's, that are giving me problems. Each device has a separate SIP credential for each port, but sometimes, only a few of the ports register. So, the device will be running fine for a while, then suddenly one or more of the ports will become

[asterisk-users] Loud Ringers and paging systems...

my paging hardware just to add one tiny piece of functionality. Kevin Larsen-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] Asterisk 12 and DPMA

I read somewhere that DPMA is not supported for Asterisk 12. Can anyone confirm or deny that? If not supported yet, will it be? If so, when? Per this link: https://wiki.asterisk.org/wiki/display/DIGIUM/Digium+Phone+Module+for+Asterisk+(DPMA)+v+2.0 It would seems that Digium is under the

Re: [asterisk-users] Simultaneous Ring

asterisk-users-boun...@lists.digium.com wrote on 07/16/2014 01:46:09 PM: From: Haley,Scott A scott.ha...@edwardjones.com To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com, Date: 07/16/2014 01:46 PM Subject: [asterisk-users] Simultaneous Ring Sent by:

Re: [asterisk-users] How to monitor non-SNMP SIP devices ?

asterisk-users-boun...@lists.digium.com wrote on 07/09/2014 10:19:11 AM: From: Olivier oza.4...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Date: 07/09/2014 10:19 AM Subject: [asterisk-users] How to monitor non-SNMP SIP devices ?

Re: [asterisk-users] share mailbox Asterisk 1.8.22

I have done this for one of my users in a very similar fashion. When 102 checks the voicemail, do they hear the correct voicemails? Ours clears just fine in this situation. Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 asterisk-users-boun...@lists.digium.com wrote on 06

Re: [asterisk-users] share mailbox Asterisk 1.8.22

asterisk-users-boun...@lists.digium.com wrote on 06/24/2014 05:36:16 PM: From: motty cruz motty.c...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Date: 06/24/2014 05:36 PM Subject: Re: [asterisk-users] share mailbox Asterisk 1.8.22

Re: [asterisk-users] share mailbox Asterisk 1.8.22

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 asterisk-users-boun...@lists.digium.com wrote on 06/24/2014 05:49:39 PM: From: motty cruz motty.c...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Date: 06/24/2014

Re: [asterisk-users] second connected PBX not showing Caller ID

From: Claude Hayn chayn...@gmail.com To: asterisk-users@lists.digium.com, Date: 05/31/2014 04:43 PM Subject: [asterisk-users] second connected PBX not showing Caller ID Sent by: asterisk-users-boun...@lists.digium.com Hello, We have two asterisk PBXs connected. PBX 1 has SIP trunks

Re: [asterisk-users] 'restart when convenient'

asterisk-users-boun...@lists.digium.com wrote on 05/28/2014 10:37:25 AM: pbx1*CLI core restart when convenient Waiting for inactivity to perform restart Ignoring asterisk restart request, already in progress. After doing 'core restart now' and hitting Enter really hard ;) Asterisk did

Re: [asterisk-users] BLF and notifyringing in Asterisk 11

Unfortunately, notifyringing is only set in the [general] section in sip.conf. It does not have a peer level override. It would be nice if it was set on a peer by peer basis - that would be a useful improvement. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive

[asterisk-users] BLF and notifyringing in Asterisk 11

at a confirmed state if a second call came in while already on a call. Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] authoritative sql definitions for Asterisk Realtime Architecture ARA

Here are links to the Asterisk Wiki for CDR and SIP tables. I didn't find extensions listed, but it's pretty simple and I can provide the structure for that if needed, but it would be without a definitive source beyond me having used it for years. :-) I think the problem with those links

Re: [asterisk-users] Multicast RTP

From: Josh Metzger joshdmetz...@gmail.com I'm currently working with Asterisk 11.8.1 trying to get Multicast RTP working (it's not) with some Polycom phones, and I'm really trying to determine if Asterisk or the phones are the issue. I THINK it's Asterisk... In extensions.conf I have a

Re: [asterisk-users] Multicast RTP

From: Josh Metzger joshdmetz...@gmail.com Interesting. I thought the latest Polycom software supported multicast, but that Polycom forum link says otherwise. What DOES work is using the built-in paging feature, so maybe the solution, in this case, is to do it without Asterisk at all. We

Re: [asterisk-users] Putting a notice in the logs from the dialplan

From: Matthew Jordan mjor...@digium.com Ha! Just when you think you've found every corner of Asterisk, you turn around and there's something else. Just goes to show, you learn something new every day. Look on the bright side, you did say it would be easy to write just such a module...--

Re: [asterisk-users] FW: clients unable to auth

asterisk-users-boun...@lists.digium.com wrote on 04/16/2014 05:56:32 AM: From: Peter Reid peter.r...@morodo.co.uk To: asterisk-users@lists.digium.com, Date: 04/16/2014 05:56 AM Subject: [asterisk-users] FW: clients unable to auth Sent by: asterisk-users-boun...@lists.digium.com Hi Guys,

Re: [asterisk-users] DUNDi with SIP Mapping

From the reading and testing I have done it doesn't look like SIP supports a username and password in the Dial string. I currently have the following mapping. priv = dundi-extens,0,SIP,dundi:pass@1.1.1.1/$ {NUMBER},nounsolicited,nocomunsolicit,nopartial On the sending side I see

Re: [asterisk-users] FW: clients unable to auth

Thank you guys – your advice was spot on. I will now reach out earlier and not struggle with issues like this for 2 weeks J You sound like you are just getting started with Asterisk. A couple pieces of advice that helped me when I was starting out: 1. Get a copy of Asterisk: The

Re: [asterisk-users] DUNDi with SIP Mapping

I wanted to move to DUNDi to simplify the setup. It looks like I need to switch to IAX trunks to be able to do this. You are a bit outside of what I have done, but this looks like it might be what you want to do with SIP: http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP--

Re: [asterisk-users] Need to hire recordings for an IVR

I wonder if anybody know how to hire Alice or some professional voice-artist. I need to record 12 messages for a customer. Assuming you mean Allison, her information is here: http://www.digium.com/en/products/ivr/allison-smith--

Re: [asterisk-users] Asterisk 11 under VMware?

From: Johan Wilfer li...@jttech.se Sounds very good. Do you have this experience with WMware in particular or with virtualization in general? We run our Asterisk 11 instance in VMWare as well. They share the hardware with multiple other boxes. We do give Asterisk priority over most other

Re: [asterisk-users] Asterisk CLI Banner

asterisk-users-boun...@lists.digium.com wrote on 03/28/2014 10:51:13 AM: From: Haider Khalil haiderkha...@hotmail.com Thank you Thorsten Göllner. Matthew, What does violating license of Asterisk means ? Does it means I won't be able to use any commercial modules or asterisk

[asterisk-users] XMPP issues in Asterisk 11.6.0 for distributed device states...

be recreated, but that seems extreme as I put more servers into the system. Any thoughts on a better way to handle xmpp and making sure new servers can access the proper nodes? Kevin Larsen - Systems Analyst - Pioneer Balloon Company

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