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]
D
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
>
> ok thanks for the answer, i wi
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"
> To: asterisk-users@lists.digium.com
> Date: 12/14/2017 09:36 AM
> Subject: Re: [asterisk-users] Rewrite Out
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 tra
> 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
> digi
> 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,
>
> I've some analogic lines and I'm asked if it's possi
> 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, "
> 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 'f
> 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
> extension
> 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 co
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 genera
> 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
> add
> 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
>
> Yves
> 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 a
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 password includes a
semicolon
> Sent by: asterisk-users
> 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 lin
> 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 default
> 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 interesting information for
> Kevin Larsen 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 was a
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 ext
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 is
>
> 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 instructions
> on how to proceed?
>
> Aster
60 is ringing
[Sep 21 11:32:18] VERBOSE[31614][C-424d] pbx.c: [Sep 21 11:32:18] ==
Spawn extension (cs, redacted, 5) exited non-zero on
'SIP/003567101-705c'
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208--
__
> 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 queues
>
> 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 y
> 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 ma
> 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
> 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 fa
> 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
> > 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
> 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 probl
> 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 fro
> 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
> 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
> 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 es
> 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
>> 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 shoul
> 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
> 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 comm
> 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,
> 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
> 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 melo
> 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 22:49:5
> Darryl Moore 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 sip:00493
> > 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
> 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
UNREACHA
> 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
> 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
> > 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 pro
> > 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 Aler
> 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 a
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 dialpl
> 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.
http://en.
asterisk-users-boun...@lists.digium.com wrote on 03/02/2015 08:27:07 AM:
> From: "Stefan Viljoen"
> To: ,
> 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 bash script?
Just to rule out some weird permission
> 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
> 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 in
> 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:
http://www
> 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
asterisk-users-boun...@lists.digium.com wrote on 01/23/2015 10:38:27 AM:
> Would you be willing to send the configuration from asterisk for this?
It is pretty bog standard, but sure:
[111]
callerid="Conf Rm" <111>
secret=x
type=friend
host=dynamic
call-limit=5
context=xx
qualify=
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 t
> 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 rese
> 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 tr
> > 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 wo
id a store and forward of a fax
that if I tried to forward the fax and it had 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 th
come along.
All I can say is that having experience with a few versions of Asterisk,
it seems to get better and more stable as new versions come along. Perhaps
a bit of faith that they are not trying to kill off their product simply
by having
> I'm not going to say you will never experience a crash in a supported
> version of Asterisk. The issue tracker would quickly make me out to be
> a liar.
>
> On the other hand, if you are using a supported version of Asterisk,
> then you can report the issue [1], get a backtrace [2], and we'll do
> 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
> 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
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 preferabl
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:
>
>
>
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
com
> 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,
> 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.--
_
-- Band
> 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: 9
asterisk-users-boun...@lists.digium.com wrote on 08/13/2014 08:31:01 AM:
> From: "Nick Olsen"
> To: ,
> 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,
>
> Currently, I've got a PBX that is e
> 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
> 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 segm
> 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
>
> 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
> 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 ye
don't want to triple 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 introdu
> 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
> 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 i
asterisk-users-boun...@lists.digium.com wrote on 07/16/2014 01:46:09 PM:
> From: "Haley,Scott A"
> To: "asterisk-users@lists.digium.com" ,
> Date: 07/16/2014 01:46 PM
> Subject: [asterisk-users] Simultaneous Ring
> Sent by: asterisk-users-boun...@lists.digium.com
>
> I have a need to issue a d
asterisk-users-boun...@lists.digium.com wrote on 07/09/2014 10:19:11 AM:
> From: Olivier
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> ,
> Date: 07/09/2014 10:19 AM
> Subject: [asterisk-users] How to monitor non-SNMP SIP devices ?
> Sent by: asterisk-users-boun...@lists.digium
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
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> ,
> Date: 06/24/2014 05:49 PM
> Subject: Re: [asteri
asterisk-users-boun...@lists.digium.com wrote on 06/24/2014 05:36:16 PM:
> From: motty cruz
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> ,
> Date: 06/24/2014 05:36 PM
> Subject: Re: [asterisk-users] share mailbox Asterisk 1.8.22
> Sent by: asterisk-users-boun...@lists.digium.
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
> From: "Claude Hayn"
> To: ,
> 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 connected to our provider. PBX 2 is a
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
> 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 Dav
sn't already on a call, but would remain 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
> 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 li
> From: Josh Metzger
> 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 currently have a
> From: Josh Metzger
> 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 simple: "Page(Mu
> From: Matthew Jordan
> 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...--
__
> 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-
> 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 Definiti
> 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
asterisk-users-boun...@lists.digium.com wrote on 04/16/2014 05:56:32 AM:
> From: "Peter Reid"
> To: ,
> 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,
>
> Just new to Asterisk and am completely st
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