ki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+to+work+through+NAT
--
Michael L. Young
(elguero)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community
- On Feb 5, 2021, at 11:18 AM, Michael L. Young wrote:
> - On Feb 4, 2021, at 4:26 PM, Social Boh wrote:
>> The problem is with this CentOS 7 glibc version:
>> 2.17-317.el7
>> After the library update and system reboog,
>> gotoif Asterisk applicati
- On Feb 4, 2021, at 4:26 PM, Social Boh wrote:
> The problem is with this CentOS 7 glibc version:
> 2.17-317.el7
> After the library update and system reboog,
> gotoif Asterisk application, stop to working
> Any hint to solve?
Until it is resolved, you can do a 'yum history' and
owse/ASTERISK-26143 |
https://issues.asterisk.org/jira/browse/ASTERISK-26143 ]
Not sure if this is the answer to your problem but thought that I would throw
that out there.
Michael L. Young
(elguero)
--
_
-- Bandwidth and
- Original Message -
> From: "sean darcy"
> To: "Asterisk Users Mailing List, Non-Commercial Discussion"
>
> Sent: Tuesday, January 21, 2020 9:22:28 PM
> Subject: [asterisk-users] permission woes on systemd
[..]
> So why would starting asterisk as user asterisk work, but fail using
>
it was never
> actually used.
>
> The system is a 2.4Gh quad-core Xenon with 4G of RAM, so it should have plenty
> of power for what I'm asking it to do. The system is configured via RT using
> a local Mysql database.
>
Which distro are you running? How are you starti
- On May 4, 2016, at 8:49 AM, Mamadou NGOM n...@numericap.com wrote:
> Hello everybody,
> When I call my extension the agi script don't work well. when I look at the
> cli,
> that is what I have:
> AGI Tx >> agi_request: **.php
> AGI Tx >> agi_channel: SIP/myprovider-0007
> AGI Tx >
- Original Message -
> From: cov...@ccs.covici.com
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Thursday, May 29, 2014 6:42:05 PM
> Subject: Re: [asterisk-users] Asterisk 11.10.0 Now Available
>
> > * ASTERISK-23754 - [patch] Use var/lib directory for log fi
- Original Message -
> From: "Michael L. Young"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Friday, May 16, 2014 4:55:30 PM
> Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails
>
> - Original Message
- Original Message -
> From: "Michelle Dupuis"
> To: "Asterisk Users List"
> Sent: Friday, May 16, 2014 4:29:05 PM
> Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails
>
> From: asterisk-users-boun...@lists.digium.com
> on behalf of M
- Original Message -
> From: "Michelle Dupuis"
> To: "Asterisk Users List"
> Sent: Friday, May 16, 2014 3:39:35 PM
> Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails
>
> You're right - but I tried username too and it fails. I can't
> understand why AMI authenticates and AJAM
- Original Message -
> From: "Michelle Dupuis"
> To: "Asterisk Users List"
> Sent: Friday, May 16, 2014 2:43:30 PM
> Subject: [asterisk-users] Login by AMI ok, by AJAM fails
> --
> root@apbx:/tmp# curl
> http://localhost:5039/asterisk/rawman?action=login&user=te
- Original Message -
> From: "Michelle Dupuis"
> To: "Asterisk Users List"
> Sent: Thursday, March 27, 2014 12:55:21 AM
> Subject: [asterisk-users] Security log format / content
> I've noticed that the Asterisk (v11) security log captures attempts
> do dial without first authenticating
- Original Message -
> From: "Andres"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Thursday, January 16, 2014 4:17:53 PM
> Subject: Re: [asterisk-users] Asterisk ignoring nat settings
>
> > I am curious why you would say that "nat=yes" might work over
> > "n
- Original Message -
> From: "Andres"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Wednesday, January 15, 2014 7:51:28 PM
> Subject: Re: [asterisk-users] Asterisk ignoring nat settings
> Why don't you try with nat=yes. It should be equivalent to what you
> From: "Tony Mountifield"
> To: asterisk-users@lists.digium.com
> Sent: Friday, November 8, 2013 10:39:25 AM
> Subject: [asterisk-users] 11.5.0 - SIP registration not retrying after
> failures
>
> I had a SIP problem on an 11.5.0 system that I look after. It
> registers
> with a SIP trun
- Original Message -
> From: "Noah Engelberth"
> I have an Asterisk 11.5 system, using SIP Realtime and operating as a
> ITSP. One of my customer’s endpoints is a NetVanta 7100 PBX system
> that has a SIP trunk connection to my Asterisk box. The NV 7100 has
> a public IP on it that does
- Original Message -
> From: "Carlos Chavez"
> To: asterisk-users@lists.digium.com
> Sent: Thursday, August 1, 2013 8:41:19 PM
> Subject: [asterisk-users] External sip phones register with the servers IP...
>
> We have just updated our office server to Asterisk 11.4.0 from 1.8.15
> and
>
- Original Message -
> From: "Richard Mudgett"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Thursday, May 2, 2013 8:24:49 PM
> Subject: Re: [asterisk-users] Playing a sound file during a call
>
> > On Thu, May 2, 2013 at 3:37 PM, Carlos Alvarez
> >
> > wrot
- Original Message -
> From: "Leandro Dardini"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Tuesday, March 26, 2013 5:28:22 AM
> Subject: [asterisk-users] rtcachefriends and rtautoclear on change password
>
> Hello friends,
> I am using from a long time rtca
- Original Message -
> From: "Jaap Winius"
> To: asterisk-users@lists.digium.com
> Sent: Thursday, March 21, 2013 5:27:37 PM
> Subject: Re: [asterisk-users] Asterisk 1.8 and dual stack support
>
> That's what I thought would happen. When I set bindaddr=:: and use
> 'netstat -lpn |grep 506
- Original Message -
> From: "Jaap Winius"
> To: asterisk-users@lists.digium.com
> Sent: Thursday, March 21, 2013 12:47:57 PM
> Subject: [asterisk-users] Asterisk 1.8 and dual stack support
>
> Hi folks,
>
> Following an upgrade to Debian wheezy, I'm now running Asterisk
> 1.8.13.1.
> As
- Original Message -
> From: "Bob Pierce"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Cc: g...@westmancom.com
> Sent: Monday, February 4, 2013 6:14:26 PM
> Subject: [asterisk-users] Asterisk 1.8 Streaming MOH timing interface
> We are running Asterisk 1.8.5.0 wi
- Original Message -
> From: "Logan Bibby"
> Does anyone have a good contact for their sales? I've attempted
> calling their Enterprise sales a few times and was just spun around
> in circles. Having a sales rep I can just call would be awesome.
Logan,
We have an account manager that
- Original Message -
> From: "Matthew J. Roth"
> At least Verizon maintains a consistent customer experience. ; )
>
> Overall, we've found the service to be reliable and stable, but when
> there are problems or changes needed you're dealing with Verizon and
> the
> w...h...e...e...l...s
- Original Message -
> From: "Carlos Alvarez"
> Sounds like the same huge effort it takes to work with
> Qwest/Centurylink, and in the long run we found it simply isn't
> worth it. The few benefits of working with an RBOC are countered by
> the many drawbacks of working with an RBOC.
>
- Original Message -
> From: "Matthew J. Roth"
> Your email documents the same experience we had years ago. It was
> strange reading it and I was shocked that nothing has changed in that
> much time. Asterisk will work with Verizon's IP trunking product,
> but
> they're trying to make y
> From: "Carlos Alvarez"
> It may be too late for this, but in working with another RBOC who
> didn't want to deal with Asterisk, I just asked what they do
> support, and modified the headers sent by Asterisk to claim that it
> was one of the devices on that list. Done.
Like everyone else, I was
- Original Message -
> From: "Steven Howes"
>
> I *think* Verizon require IPSEC for the signalling, so it may be
> worth reading up on configuring IPSEC in Linux (or acquiring
> additional hardware) whilst you're looking at the Asterisk part.
> This could have just been for a specific prod
to sucessfully work with Verizon?
Thanks,
--
Michael L. Young
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http
- Original Message -
> From: "Howard Leadmon"
> To: asterisk-users@lists.digium.com
> Sent: Saturday, November 24, 2012 3:19:10 PM
> Subject: [asterisk-users] SIP Debugging Information..
>
>
> I did a little googling, but didn't seem to find anything specific
> to
> answer the question
- Original Message -
> From: "Joseph"
> To: asterisk-users@lists.digium.com
> Sent: Saturday, November 24, 2012 12:54:12 AM
> Subject: [asterisk-users] * Waiting for asterisk to shutdown .
>
> I'm running asterisk on a small box,
> Intel-R-_Atom-TM-_CPU_330_@_1.60GHz
> and whe
- Original Message -
> From: "Felix Vazquez"
> To: asterisk-users@lists.digium.com
> Sent: Friday, November 16, 2012 11:20:46 AM
> Subject: [asterisk-users] Intruder
> I am in the asterisk CLI and can see an unidentified caller trying
> the make calls out of the asterisk system. How do
- Original Message -
> From: "Ishfaq Malik"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Wednesday, November 14, 2012 9:25:37 AM
> Subject: Re: [asterisk-users] Error: astobj2.c: refcount -1 on object
>
> Thanks for the advice but that's not really a quick a
- Original Message -
> From: "Ishfaq Malik"
> To: asterisk-users@lists.digium.com
> Sent: Wednesday, November 14, 2012 4:05:21 AM
> Subject: [asterisk-users] Error: astobj2.c: refcount -1 on object
>
> Hi
>
> I'm using 1.8.7.0. This morning I got an alert telling me
>
> Asterisk on exi
- Original Message -
> From: "Patrick Lists"
> To: asterisk-users@lists.digium.com
> Sent: Tuesday, November 13, 2012 4:35:54 AM
> Subject: Re: [asterisk-users] Asterisk crashing when trying to start a Jabber
> session with ejabberd
>
> On 11/13/2012 12:11 AM, Phil Reynolds wrote:
> [sni
- Original Message -
> From: "sean darcy"
> To: asterisk-users@lists.digium.com
> Sent: Wednesday, November 7, 2012 9:20:58 AM
> Subject: Re: [asterisk-users] 11.0.1: more sip registry woes
>
> On 11/06/2012 09:45 PM, Michael L. Young wrote:
> >
- Original Message -
> From: "sean darcy"
> To: asterisk-users@lists.digium.com
> Sent: Tuesday, November 6, 2012 7:51:04 PM
> Subject: [asterisk-users] 11.0.1: more sip registry woes
>
> Upgrade to 11. This worked on 10.X.X
>
> sip.conf:
>
> register=>:@nyc.teliax.net
>
> telnet nyc.
o be performed by those affected. Especially, as in the case of what Raj
mentioned at the beginning of his prior email, not too many people may even be
affected by this change just like he won't be.
Michael L. Young
(elguero)
PS: If you can't tell, I am really for this change and do
- Original Message -
> From: "Ira"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Wednesday, October 3, 2012 3:21:50 AM
> Subject: Re: [asterisk-users] Case-sensitivity of Dialplan variables.
>
> At 07:59 PM 10/2/2012, you wrote:
> >
> >While true that most us
to working in environments such as Linux
where variables and file names are case sensitive.
If someone is moving from a GUI interface to CLI, then they would/should know
that case sensitivity is important and therefore the change shouldn't
"L" lower case and the second one is "i" upper case.
I didn't quite follow this logic. Your example, in my mind, would actually be
easier to debug with this change.
If you know that variables are case sensitive, you know that you have to check
for a typo in your var
- Original Message -
> From: "Thorsten Göllner"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Monday, June 18, 2012 11:52:15 AM
> Subject: [asterisk-users] Asterisk 1.8.13.0 / problem with cdr logging
> (mysql, odbc)
>
>
> /etc/odbcinst.ini
>
- Original Message -
> From: "Jayesh Labade"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Friday, May 25, 2012 2:09:58 AM
> Subject: Re: [asterisk-users] Asterisk MixMonitor starts recording 44
> bytes file
> Hello Michael,
> Thanks a lot for your immediat
- Original Message -
> From: "Jayesh Labade"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Thursday, May 24, 2012 4:10:29 PM
> Subject: [asterisk-users] Asterisk MixMonitor starts recording 44
> bytes file
> Hello All,
> I have installaed asterisk 10.4 in m
=as66c75bd7
>
> CSeq: 102 INVITE
>
> Call-ID: 4ce934e47314480b45073a7d768cb0c8@192.168.9.250
>
> Server: Epygi Quadro SIP User Agent/v5.1.16 (QUADRO-FXO)
>
> Content-Length: 0
I think the "404 Not Found" being returned
- Original Message -
> From: "David C Klaverstyn"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Sunday, February 12, 2012 7:02:15 PM
> Subject: [asterisk-users] No valid transports available, falling back
> to 'udp'.
> Hi All,
> I just installed Asterisk 10.
- Original Message -
> From: "Remco Barendse"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Friday, September 23, 2011 5:27:27 AM
> Subject: [asterisk-users] TDM400 FXO stopped working
> Hi list
>
> I have 2 servers with a TDM400 card, port 1 populated by an
- Original Message -
> From: "Mike Diehl"
> To: asterisk-users@lists.digium.com
> Sent: Tuesday, August 30, 2011 5:13:22 PM
> Subject: Re: [asterisk-users] Polycoms rebooting themselves
>
> Well, we've taken the time to check out the wiring. It's only 3
> years old and
> looks like the p
- Original Message -
> From: "Patrick Lists"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Friday, July 8, 2011 1:58:36 PM
> Subject: Re: [asterisk-users] Issue 0019268 Patch Asterisk
>
> On 07/08/2011 07:32 PM, Mark Rosedale wrote:
>
> > * channels/sig_
- Original Message -
> From: "Chris Maciejewski"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Thursday, May 19, 2011 9:39:57 AM
> Subject: [asterisk-users] ConfBridge - Failed to find a bridge technology to
> satisfy capabilities
>
> Hi,
>
> I am trying t
- Original Message -
> From: "Olle E. Johansson"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Wednesday, April 27, 2011 3:34:03 PM
> Subject: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
>
> Friends,
>
> We have a discussion on asterisk-dev
--
Michael L. Young
Administrative Claim Service, Inc. | IT Manager
600 Main Street, Suite 5, Winchester, MA 01890
www.acsacc.com
Phone 781-721-1998
- Original Message -
> From: "Andrew Stewart"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of JR Richardson
> Sent: Tuesday, March 30, 2010 6:55 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Dropped Calls
>
> > I've written
saw this problem
while running CentOS 5.1 XEN kernel and if you search their bug tracking
system you will see some reports about this bug. A search on google
revealed some possible solutions.
This was the first thought that came to my mind when I saw this.
Regards,
Mi
.
Michael L. Young
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of François Delawarde
> Sent: Monday, May 14, 2007 10:24 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] zaptel huge irq problem
>
>
> zaptel.conf
> ---
> loadzone=uk
> defaultzone=uk
>
>
> span=1,1,1,ccs,hdb3,crc4,yellow
> span=2,0,1,ccs,hdb3,crc4,yellow
>
> bchan=1-15,32-46
> dchan=16,47
> bchan=17-31,48-62
> ---
> where span 1 is to the provider
only one span should be the primary timing source. The other span should
either be at 0 (not used as a timing source) or set as a secondary timing
source.
Hope this helps.
Michael L. Young
___
--Bandwidth and Colocation provided by Easynews.com --
ast
t helps explain echo and some steps in Asterisk for reducing
echo: http://www.xorcom.com/pdfs/AB007_Echo.pdf
Michael L. Young
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options
Bill,
I had the same issue over the weekend. Yesterday, there was an announcement
on the list, I think it was from Kevin Fleming, that the svn repositories
where out of sync and they had to be redone from scratch. So, I ended up
clobbering my local working copy and did a checkout of the trunk.
I just wanted to see if any one else has seen this or
could help point me in the right direction on this problem.
I have a TE411P card in my * box. I am running FC4 x86_64.
I used to have two TE110 cards in the same box that worked without any
problems. Since changing to the TE411P cards,
We are in the process of installing a PRI line and we
are going to connect it to an Asterisk Box.
Verizon called us today to find out some information. I am
surprised that they have never heard of Asterisk or Digium. But anyways, they needed
some information in order to set up the circuit
62 matches
Mail list logo
