:4803836...@zzz.zzz.zzz.zzz;tag=as09ca5622
Call-ID: 176a274d5342aac505d0125979d19...@zzz.zzz.zzz.zzz:5060
Max-Forwards: 70
CSeq: 103 ACK
Contact: sip:WWW.WWW.WWW.WWW:5060
Content-Length: 0
-
--- (9 headers
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
was able to modify the mac address.cfg file to point only the
conference phone to the different firmware so that I could still keep the
other phones on the known working firmware.
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
Do you have sendrpid and trustrpid set to yes for those IAX2 connections?Sent from Lotus TravelerChet W. Stevens --- [asterisk-users] Get CONNECTEDLINE info from other Asterisk system via IAX2 --- From:Chet W. StevensToasterisk-users@lists.digium.comDate:Sat, Jan 5, 2013 7:55
If you want something a little more enterprise ready and tested than a
RaspberryPi, you might take a look at Valcom's products.
http://www.valcom.com
We use them for our paging and have been fairly happy with them. Only had
one small issue that a firmware upgrade took care of.
Kevin Larsen
for the lines in the log files with COMPLETEAGENT, COMPLETECALLER,
and TRANSFER. You can trace any call by the second column as that is the
unique identifier for a specific call.
Queuemetrics (I do not have any association with them other than being a
happy customer)
http://www.queuemetrics.com
Kevin
of up time without an asterisk restart.
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 for a live introductory
numbers I wish to block are placed in my Asterisk Database. If they
exist there, they get answered and they system logs the attempt and plays
back an error message. Otherwise, it simply returns from the subroutine
and continues on the call path like normal.
Kevin Larsen - Systems Analyst - Pioneer
the caller id, and dumps in the span of milliseconds in the
case of a SIP or PRI trunk. Analog line would take just a bit more time.
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
From: Salaheddine Elharit salah.elharit...@gmail.com
To: Asterisk Users Mailing List - Non
Possibly switch to using subroutines instead of Macros. Macros are being
deprecated in place of subroutines.
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
From: Olivier oza_4...@yahoo.fr
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
with that, you settings will work fine.
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
From: asterisk users ast4...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com,
Date: 01/22/2013 05:22 PM
Subject
time to turn it from attended to blind transfer
on my phones).
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
From: Steven Howes steve-li...@geekinter.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com,
Date: 02/04/2013
.
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
From: Frank fr...@efirehouse.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com,
Date: 02/04/2013 09:47 AM
Subject:Re: [asterisk-users] CallerID external call after
that will register to the public network should have it
set to no.
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
From: Frank fr...@efirehouse.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com,
Date: 02/07/2013 08:39 AM
and
should give you the audio.
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
From: Frank fr...@efirehouse.com
To: ch...@acsdi.com, Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com,
Date: 02/07/2013 12:06 PM
Subject:Re
, you might want to
just check this out here:
http://www.raspberry-asterisk.org/
It is probably easier and better than rolling your own all the way
through.
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
From: David Wessell da...@ringfree.biz
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com,
Date: 02/28/2013 04:34 PM
Subject:[asterisk-users] Dynamic Agents in a queue
Sent by:asterisk-users-boun...@lists.digium.com
Hi,
We have
From: Chris Bagnall aster...@lists.minotaur.cc
To: asterisk-users@lists.digium.com,
Date: 03/07/2013 06:43 AM
Subject:Re: [asterisk-users] asterisk with 1000 extensions
Sent by:asterisk-users-boun...@lists.digium.com
On 7/3/13 6:50 am, Bharat Lalcheta wrote:
You can
From: Hans Witvliet aster...@a-domani.nl
To: asterisk-users@lists.digium.com,
Date: 03/11/2013 03:00 PM
Subject:Re: [asterisk-users] digium card and virualbox
Sent by:asterisk-users-boun...@lists.digium.com
I am not mixing. I need this for LAB testing.
How? This PCI
http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
From: Daniel - Asterisk earohua...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
on the wan will be forced to talk directly to the Asterisk server for
everything. You might also want to look at the nonat option of
directmedia.
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
From: David Wessell da...@ringfree.biz
To: Asterisk Users Mailing List - Non
of gotchas
that can happen based on your dial options, so check out this page:
http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite
canreinvite was renamed to directmedia in Asterisk 1.6.2, but the page is
still pretty good with regards to the options that are available.
Kevin Larsen - Systems
;;tt-monkeys to the opposite
channel
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
From: Carlos Alvarez car...@televolve.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com,
Date: 05/02/2013 04:53 PM
Subject
Add MOH_Class onto the example and the idle channel will hear music on
hold until the playback is complete on the other channel.
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
From: Carlos Alvarez car...@televolve.com
To: Asterisk Users Mailing List - Non-Commercial
at the start are your
friend. Once you understand all the ins and outs of the migration, you can
start moving to the new instance on a faster pace. It is possible to do it
with virtually no downtime.
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
From: Andre Goree
Are you using cdr_adaptive_odbc.conf to populate it? If so, there is no
Asterisk analog to calldate. You would need an alias set up. Mine looks
like:
alias start = calldate
so that the start of my call is what gets logged to the database as the
calldate.
Kevin Larsen
From: Jairo ja
application. Person A then hits
transfer again to finish a blind transfer. At this point, the musiconhold
that the caller hears cuts out and is not replaced by the m(ringing)
audio. Any thoughts on if it is possible to make this work?
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
From: jon pounder j...@inline.net
To: asterisk-users@lists.digium.com,
Date: 07/18/2013 09:00 AM
Subject:Re: [asterisk-users] LUA
Sent by:asterisk-users-boun...@lists.digium.com
On 07/18/2013 09:56 AM, jacob.e.mi...@l-3com.com wrote:
I am attempting to setup my server
if that parameter is missing, then the code would in fact default to 2400
as a safe value.
Kevin Larsen - Systems Analyst
From: Zoltán Fekete bl...@gyoz.info
To: asterisk-users@lists.digium.com,
Date: 07/21/2013 04:40 PM
Subject:[asterisk-users] Fwd: Re: Asterisk T.38 Pass-Through
From: Steve Davies davies...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com,
Date: 07/29/2013 10:53 AM
Subject:[asterisk-users] Connected Line presentation in 1.8.x
upwards
Sent by:
From: Asmaa Ahmed asabatg...@hotmail.com
To: asterisk-users@lists.digium.com
asterisk-users@lists.digium.com,
Date: 10/09/2013 10:36 AM
Subject:[asterisk-users] Calling a demo menu after voicemail
authintication
Sent by:asterisk-users-boun...@lists.digium.com
Hello,
asterisk-users-boun...@lists.digium.com wrote on 10/28/2013 01:29:13 PM:
From: Eddie Mikell emik...@rimmkaufman.com
To: asterisk-users@lists.digium.com,
Date: 10/28/2013 01:29 PM
Subject: [asterisk-users] Tired of dropouts and garbled phone calls
- where to go next?
Sent by:
asterisk-users-boun...@lists.digium.com wrote on 01/02/2014 10:03:19 AM:
From: motty cruz motty.c...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com,
Date: 01/02/2014 10:02 AM
Subject: [asterisk-users] Asterisk 1.8.22.0 Polycom ip
asterisk-users-boun...@lists.digium.com wrote on 01/16/2014 08:55:31 AM:
From: Gareth Blades mailinglist+aster...@dns99.co.uk
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com,
Date: 01/16/2014 08:55 AM
Subject: Re: [asterisk-users] Weird issue
asterisk-users-boun...@lists.digium.com wrote on 02/18/2014 01:35:13 PM:
From: Nick Cameo sym...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com,
Date: 02/18/2014 01:35 PM
Subject: Re: [asterisk-users] Host = Dynamic in a Register Free
On 13/3/14 6:27 pm, Eric Wieling wrote:
This is an example of why I top post. Who wrote what?
Of course, if you use a mail client that's capable of quoting correctly,
it all works beautifully.
Outlook can quote correctly, but it is an all or nothing setting it would
appear. Lotus
readsql=SELECT name FROM asterisk_sippeers WHERE name = '%477'
Not sure what database you are accessing, but have you tried the
following:
readsql=SELECT name FROM asterisk_sippeers WHERE name LIKE '%477'--
_
-- Bandwidth and
I neither have a 2000 in sip,conf nor I want to have one.
2000 doesn't have an IP and I want to get rid of it, honestly.
I'd really want to know, where this 2000 is burned in
and how to erase it.
sip show peers does'nt show a peer 2000 nor I have a user 2000.
Something that lives at
Sure , here is the reasult.
mysql SELECT name FROM asterisk_sippeers WHERE name LIKE '%477' ;
+-+
| name|
+-+
| Y_MD_vlungu_477 |
+-+
1 row in set (0.00 sec)
What happens when you use that in your func_odbc.conf? Does your
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
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
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
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--
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,
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
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
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--
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...--
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
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
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
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
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-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
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
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
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
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
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 ?
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:
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'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
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
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
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
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
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
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
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,
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
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.--
_
--
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
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
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
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
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
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
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
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
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
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
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
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
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:
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
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
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
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
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.
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
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
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
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
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
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
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
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
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
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
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
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
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