ki/display/AST/Asterisk+19+Configuration_codec_opus
https://www.asterisk.org/configuring-opus-encoder-asterisk/
https://www.asterisk.org/asterisk-opus-packet-loss-fec/
- Kevin
On Wed, Jul 20, 2022 at 10:48 AM Brant Merryman
wrote:
> Hi. I am using Asterisk 16.27.0 in FreePBX 15.0.23.11. I
So this turned out more complicated than I originally thought!
My expectation:
Verbosity gets logged using an "at least" check against the current
system's verbose level, which if passed subsequently gets checked against
the logging channel's verbose level. Thus only verbose messages with a
level
On Fri, Sep 10, 2021 at 12:44 PM Jerry Geis wrote:
> HI All,
>
> I am trying to get SIPml5 working with 18.6.0.
> My http.conf file:
> enabled=yes
> bindaddr=myip
> bindport=8088
> serverName=MyName
> tlsenabled=true
> tlsbindaddr=myip
> tlscertfile=/etc/letsencrypt/live/mpname/fullchain.pem
>
>
ols were not enabled on your system so the
backtrace doesn't have any extractable information. Please see the wiki [3]
on how to get a useful backtrace.
Before that though I recommend upgrading to the latest version of Asterisk
[1]. Or if you're set on using a certified version [3
t_attr_g729.c -> res_format_attr_g729.o
>
>
> Is this to be expected or should I make a bug report?
>
>
When you pulled the lasted code this change would have forced a
re-configure. If you haven't already try doing a full clean and rebuild,
and see if you still have the error:
On Tue, Feb 25, 2020 at 4:02 PM Patrick Wakano wrote:
> Hi Kevin!
> Thanks very much for your reply! Much appreciated!
>
You're welcome!
> So I just have a remaining question from this, if the with-ssl is not
> mandatory to have the encryption support, what is it actu
the pjsip startup errors when the --with-ssl is
> used? I could not find a clear explanation for this problem and how to fix
> it
>
There appears to be a bug here. I configured, built, and ran with the same
options mentioned (--with-ssl, etc...) and r
rg/wiki/display/AST/Asterisk+16+Function_CHANNEL
--
Kevin Harwell
Senior Software Developer
Sangoma Technologies
Check us out at: https://sangoma.com & https://asterisk.org
___
asterisk-app-dev mailing list
asterisk-app-...@lists.digium.com
http
terisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/m
gt;
> type = bridge
>
> language = en
>
> internal_sample_rate = 0
>
> mixing_interval = 20
>
> record_file_append = no
>
> max_members = 10
>
> video_mode = follow_talker
>
>
>
> [4]
>
> type = user
>
> admin = no
>
> marked = no
;
So you are probably seeing it work or not in Chrome vs Firefox due to
browser, and codec support of such occurrences.
--
Kevin Harwell
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: https://digium.com &
>
> Is the wiki web page mistaken or is this an actual http.conf setting that
> is undocumented?
>
The page is mistaken. It should not be there. the 'tlscafile' option is not
supported by the Asterisk http server. I've removed it from the wiki.
Thanks for catching that!
;ve played with both embeded branchc and [1] but met no success yet
>
> Best regards
>
> [1] https://github.com/rkday/sipp-samples/blob/master/uac-auth.xml
>
>
--
Kevin Harwell
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW -
Greetings,
I am getting the following error (below) continually in my asterisk log,
related to qualify_frequency I believe. I am trying to use sip trunking with
the company flowroute.
3 questions if I may:
1) Is using qualify_frequency with a sip trunk a common or recommended
practice? I fig
something like this.
Outbound is the easy part. How are you handling inbound SMS->SIP ?
Regards,
Kevin Long
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Greetings !
My goal is to get Twilio trunking working, and with TLS/SRTP.
I see this concerning message in my log:
[Feb 7 16:50:26] ERROR[20596] res_sorcery_config.c: Could not create an object
of type 'endpoint' with id ’twilio' from configuration file ‘pjsip.conf’
Thus, ‘pjsip show e
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
Hello,
All my asterisk systems use only IPv4 currently. I have one phone which is on
T-Mobile network, and this network is only IPv6 now.
The phone can register fine, because T-Mobile does NAT64 and it connects fine
to my IPv4 asterisk server.
But in the SDP for a call setup, this phon
> 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
e. Which channel type (chan_sip,
local channel, chan_pjsip) is involved, and how you are enabling the jitter
buffer (dialplan function vs configuration) would be good to know as well.
[1] https://issues.asterisk.org
[2] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
Thanks!
--
Kevi
!
Kevin Long
--
_
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New to Asterisk? Start here:
https://wiki.asterisk.org
on the phone, the call
fails.
Perhaps this is just not documented, or may not be implemented yet. Anyone
have a thought?
Thank you,.
Kevin Long
--
_
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is complete, it then connects to
the IP instead of the hostname, and the mismatch occurs ?
Any help appreciated,
Thanks,
-Kevin
--
_
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Check out the new
> 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, "
s removed as
the bulk of its code was moved into the res_pjsip core.
--
Kevin Harwell
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
_
Is there are way to specify the display name of a resource in a resource
list? I have setup a resource list in Asterisk 13 for 1234. All is working
on the device, but I want to show "Joe User" instead of "1234". Any
thoughts?
--
> 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
I see that you can configure RLS in pjsip.conf, but does this work with
realtime? The wiki refers to pjsip.conf for configuration, but since many
of the other items can be in the the DB, I was wondering if RLS can as
well.
--
_
other
applications, and am curious if anyone has a working example or if this is
even possible?
Thank you,
Kevin Long
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suite);
> On May 30, 2016, at 11:49 AM, Kevin Long wrote:
>
>
>
> Hi folks,
>
>
> At least several endpoints (soft phone and desk phones) are supporting
> various 256 bit ciphers for SRTP these days. I *believe* libsrtp has been
> updated to allow this
with the know-how be willing/able to submit a patch ?
Thank you,
Kevin Long
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New to Asterisk? Join us for a live introductory webinar every Thurs
> 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
Personally I am about to try asterisk on proxmox using containers since they
run code "native". I've had timing issues on conference calls (stutter) with
VMware esxi . Not sure about KVM I hope it's also better than esxi too.
Sent from my iPhone
> On Apr 6, 2016, at 9:13 AM, Markos Vakondios w
> 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
second factor of authentication
besides the SIP secret , since in my current setup, despite using a TLS/SSL
cert for the server, the server only verifies the client by the SIP secret.
Regards,
Kevin Long
smime.p7s
Description: S/MIME cryptographic signature
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
Thanks John,
For anyone reading this using FreePBX - simply switching the default conference
app from MeetMe to ConfBridge seems to be a drastic improvement, have not
stress tested but running a conf now with no stutter on Confbrdige app.
Cheers,
Kevin Long
> On Mar 9, 2016, at 12:17
wondering if anyone has anything I could try to fix or mitigate the problem in
ESXi environment .
We have freepbx (asterisk 11 chan_sip) and test environments asterisk 13.7/8
pjsip .
Thank you again,
Kevin Long
smime.p7s
Description: S/MIME cryptographic signature
provisioning system assumes that both devices will use the same SIP
extension for auth however.
Normally we would use separate extensions and a follow-me , but if there is any
way to use the same extension, I need to figure it out.
Thank you,
Kevin Long
smime.p7s
Description: S/MIME
oseph wrote:
>
>
>
> On Fri, Mar 4, 2016 at 1:16 AM, Kevin Long wrote:
> Hi George the patch was from here , you wrote it I believe . I pulled
> asterisk 13 from git, apply this patch which fixed RTP issue , but I think
> tla transport issue came back for me .
>
&g
rge Joseph
> wrote:
>
>
>
>> On Thu, Mar 3, 2016 at 8:25 PM, Kevin Long
>> wrote:
>>
>> Thanks George I appreciate the info . Being able to see what codec is in
>> use for call in progress is very handy sometimes.
>>
>> As far as the
ot working and the internal IP being sent in the
SDP from asterisk - I applied this patch to the codebase and recompiled I am
seeing the TLS “new transport” issue again , I think.
Regards,
Kevin Long
smime.p7s
Description: S/MIME cryptographic
again,
Kevin Long
smime.p7s
Description: S/MIME cryptographic signature
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs
Hi Joshua,
This Asterisk 13 was pulled from git master branch just 2-3 days ago:
GIT-13-d1495b .
I used this very recent source code to overcome a pjsip problem (you can see my
email list post from a few days ago)
Thanks again
smime.p7s
Description: S/MIME cryptographic signature
--
__
Hi Joshua,
Looking at the transmitted SIP packets from Asterisk, it looks like Asterisk
is only sending it’s own internal IP (it is behind a NAT too, with proper port
forwarding) .
I did set in my transport the external_signaling_address and
external_media_address , and I have now put tr
Thank you for the response Joshua .
I had rtp_symmetric=yes before I wrote the email, then I set it to no,
restart asterisk, and tried to make the call from the remote endpoint again but
still tcpdump is showing me the RTP packets are being sent from Asterisk to the
private IP.
tcpdump
ved SIP request (1282 bytes) from TLS:72.52.31.109:55256 --->
INVITE sip:4...@dev1.domain.com SIP/2.0
Via: SIP/2.0/TLS
10.128.30.239:55253;branch=z9hG4bK-524287-1---bf28eb29eb900b43;rport
Max-Forwards: 70
Contact:
To:
From: "Kevin";tag=0af40611
Call-ID: MGE5OWFhMDY5OGFhYzM4ZDIxNjA5OGRj
, firewall, or Asterisk/pjsip that is the culprit .
Regards,
Kevin Long
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Description: S/MIME cryptographic signature
--
_
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New to Asterisk? Join us for a
asterisk is still trying to create *new* TLS outbound
connections to my endpoints, which are failing.
Thank you for your time
Kevin
-
My simple pjsip config file:
[transport-tls]
type=transport
protocol=tls
bind=0.0.0.0:5061
local_net=10.50.55.0/24
external_media_address=x.x.x.x
> 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
k you,
Kevin Long
output from “openssl ciphers” on my Asterisk box:
ECDHE-RSA-AES256-GCM-SHA384
ECDHE-ECDSA-AES256-GCM-SHA384
ECDHE-RSA-AES256-SHA384
ECDHE-ECDSA-AES256-SHA384
ECDHE-RSA-AES256-SHA
ECDHE-ECDSA-AES256-SHA
DHE-DSS-AES256-GCM-SHA384
DHE-RSA-AES256-GCM-SHA384
DHE-RSA-AES256-SHA256
DH
unclear as to
whether I truly need 2 separate public IPs for the turn server to work, which I
have seen mentioned in some of the documents.
Thank you for your time.
Regards,
Kevin Long
smime.p7s
Description: S/MIM
> 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
h the latest release of
13.7.0. Actually the new StatsD Dialplan application currently resides in
master only. A small change to the res_statsd api was made and got tagged
with that issue number for some reason, thus making it look as if the
StatsD application feature was added to 13.
--
Kevin
> 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
assigned the client device.
Does asterisk send RTP traffic to the IP which is in the IP headers of the SIP
REGISTER , or can a client “specify” it’s truly reachable IP ?
I hope this makes sense.
Regards,
Kevin Long
Greetings,
I use TLS and SRTP on all my extensions. I use openssl and distribute my root
certificate to my endpoints. Most of the time my calls work just fine.
Sometimes I receive a repeating error in my log files however, and I don’t know
why this is happening. I’m wondering if this is
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 ongoi
> 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
> 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
> 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.
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