Magnus,
Can it be the same as I experienced
https://issues.asterisk.org/view.php?id=18542 ? Do not be confused by
the ticket subject, it reflects the symptoms as they looked originally
You can try the patch if applicable and if you know how to compile
Addons in 1.8 separately or if you have
x display going from “Dialing” to “Sending” to
“Sending OK”.
I am sorry to say that I am not smart enough to know what
trace I should start looking at, any knows?
From
William,
Have you tried outgoing calls? What happens there?
Have you restarted the Asterisk after you fixed the typo?
-Vladimir
On 2/10/2011 10:44 PM, William Stillwell wrote:
Yeah, that was a typo, but I fixed, still no dice.
The incoming jabber call doesn’t fire the gtalk
the Asterisk.
-Vladimir
On 2/10/2011 11:40 PM, Vladimir Mikhelson wrote:
William,
Have you tried outgoing calls? What happens there?
Have you restarted the Asterisk after you fixed the typo?
-Vladimir
On 2/10/2011 10:44 PM, William Stillwell wrote:
Yeah, that was a typo, but I fixed, still
:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
*Vladimir Mikhelson
*Sent:* Friday, February 11, 2011 12:51 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Gtalk/Jabber Issue
William
-users-boun...@lists.digium.com] *On Behalf Of
*Vladimir Mikhelson
*Sent:* Friday, February 11, 2011 12:51 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Gtalk/Jabber Issue
William,
I have just noticed that you have several configuration
?
I get just 1 incoming packet, and it just sits there, until it rings
to voicemail.
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
*Vladimir Mikhelson
*Sent:* Friday, February 11, 2011 1:47 AM
*To:* Asterisk
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
*Vladimir Mikhelson
*Sent:* Friday, February 11, 2011 1:59 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Gtalk/Jabber Issue
William,
Another thing. Have you tried
Unknown Caller most likely refers to the CID Name, CID Number should
be provided as your Google Voice number.
On 2/20/2011 5:53 PM, William Stillwell wrote:
And confirmed, just upgraded to 1.8.x.x branch, outbound/inbound working
fine.
Now, no outbound callerid? Shows 'unknown caller' on
...@lists.digium.com] On Behalf Of Vladimir Mikhelson
Sent: Sunday, February 20, 2011 10:16 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Gtalk/Jabber Issue
Unknown Caller most likely refers to the CID Name, CID Number should
be provided as your Google Voice number.
I
Chris,
Can you please provide more details.
What do you exactly mean by broken? Do your call recipients get a
random CID?
Have you tried to call from the GMail WEB interface? Are you getting
the same result?
-Vladimir
On 2/24/2011 8:51 AM, Chris Gentle wrote:
Anybody else noticed that
Chris,
Let me summarize:
1. GV Outbound CID shows Unknown, Unavailable, Out of area
(depending on a recipient's carrier) starting some time around
02/15/2011 if a call is placed via Google Chat/Google Talk/Google
Mail/Asterisk GTalk channel. See
issue on the Google Voice end to me.
-Vladimir
On 2/24/2011 10:40 PM, Vladimir Mikhelson wrote:
Chris,
Let me summarize:
1. GV Outbound CID shows Unknown, Unavailable, Out of area
(depending on a recipient's carrier) starting some time around
02/15/2011 if a call
Dean,
Thank you for great news. Let us see how the second SIP GV incarnation
survives.
-Vladimir
On 3/7/2011 6:51 PM, Dean Collins wrote:
http://www.disruptivetelephony.com/2011/03/google-voice-now-offers-sip-addresses-for-calling-directly-over-ip.html
Nice ;)
--
Pay attention, you have permit=172.16.16.0/24 http://172.16.16.0/24
whereas suggestion was permit=0.0.0.0/0.0.0.0 http://0.0.0.0/0.0.0.0
On 3/10/2011 1:48 AM, RR wrote:
On Thu, Mar 10, 2011 at 2:12 AM, Faisal Hanif fai...@vopium.com
mailto:fai...@vopium.com wrote:
You can add following
Alternatively, you can use OOH323 which is available with yum.
I am using it for couple years with no major problems. Developer is
very responsive. Just finished 1.8 related adjustments to OOH323,
should be available in 1.8.4.
-Vladimir
On 3/10/2011 2:29 PM, Jose P. Espinal wrote:
Bruce B
mailto:bruceb...@gmail.com wrote:
Can you please provide link to the RPM?
Thanks
On Thu, Mar 10, 2011 at 9:58 PM, Vladimir Mikhelson
v...@mikhelson.com mailto:v...@mikhelson.com wrote:
Alternatively, you can use OOH323 which is available with yum
mailto:bruceb...@gmail.com wrote:
Can you please provide link to the RPM?
Thanks
On Thu, Mar 10, 2011 at 9:58 PM, Vladimir Mikhelson
v...@mikhelson.com mailto:v...@mikhelson.com wrote:
Alternatively, you can use OOH323 which is available
Hi,
I know it sounds weird, and this is part of the reason I have not
reported that sooner.As I upgraded from 1.6.2.x to 1.8.x several
months ago I am experiencing this problem. If a call is initiated from
a DAHDI extension after no DAHDI extensions were used for some time,
arbitrary DTMF
Paul,
I have kind of a related question.
asterisk-1.8.4-summary.txt does not always properly link specific
patches to issues. For example, revision 307509 is associated with issue
18542, and it is not reflected in the summary. There may be more like this.
I tried to report this inconsistency
On 5/11/2011 10:15 AM, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of isr...@gmail.com
Sent: Wednesday, May 11, 2011 10:07 AM
To: Asterisk Users Mailing List - Non-Commercial
BTW, is GTalk/Jabber a part of RPM now?
-Vladimir
On 5/13/2011 5:43 PM, Jason Parker wrote:
On 05/12/2011 02:46 PM, Jason Parker wrote:
I'll make it a point to respond to this email when the new builds are
available.
These builds are now available.
--
Elliot,
You need to execute sendDTMF(1)
Articles are available with detailed setup description.
-Vladimir
On 6/14/2011 1:26 AM, Elliot Murdock wrote:
Hello,
To help clarify, Jabber is receiving the incoming packets, but
Asterisk does not seem to be associating it with the gtalk
Similar if not the same behavior still observed as of 1.8.5.0 with FreePBX.
See https://issues.asterisk.org/jira/browse/ASTERISK-17498
-Vladimir
On 7/18/2011 8:07 AM, Steve Davies wrote:
On 18 July 2011 14:05, Lee Archer lee.arc...@thebigword.com wrote:
Seems to be an already reported
Do you have any network devices or VPN tunnels in between the Asterisk
and Avaya?
The reason I am asking it looks like a potential networking issue.
Has this setup ever worked before?
-Vladimir
On 7/27/2011 1:32 PM, troxlinux wrote:
Hi list , I am connecting one avaya with asterisk by
Fixed endless ringing on outgoing gtalk calls for me. Asterisk 1.8.5.0.
-Vladimir
On 8/20/2011 11:46 AM, Paul Belanger wrote:
Afternoon,
Anybody able to confirm the patch in ASTERISK-18301[1] fixes outbound
google voice calls?
I've seen a few reports around the web and like to get more
Alex,
Please post the bug report on the bug tracker. Then your fix has a
chance to be incorporated in a future release.
Thank you,
Vladimir
On 9/12/2011 1:28 PM, Alex Villacís Lasso wrote:
El 01/09/11 14:11, Richard Mudgett escribió:
In our office, we were running an Asterisk 1.6.2.14
Remco,
In my experience something could have gone wrong with the parameters you pass
to WCTDM driver.
You may want to check your /etc/modprobe.d/dahdi.conf
Try loading the driver manually as follows:
1. Stop Asterisk
2. modprobe -r wctdm
3. modprobe wctdm
4. dahdi_cfg -vvv
Good luck,
SipGate.Com used to offer free DIDs in Illinois, but it looks like they
ran out of numbers...
-Vladimir
On 9/23/2011 2:45 PM, Joseph wrote:
Are there any free DID in Illinois 708-839 or area?
--
_
-- Bandwidth and
Great explanation. Makes complete sense to me.
Any workaround you can think of?
-Vladimir
On 9/26/2011 12:26 AM, isr...@gmail.com wrote:
It doesn't work at all with the dahdi timers
The reason it works it works till the first reload is because you are
preloading it before dahdi so it
AJ,
Banging my head other a similar problem here in US.
What I know so far the callerid function produces the following bitmap flag:
1. CID Private Name
2. CID Private Number
3. CID Unknown Name
4. CID Unknown Number
5. CID Message Waiting
6. CID No Message Waiting
For example, Flag=3
@bakko
I do not know anything about 10.0 but 1.6.2 problem most likely can be
fixed by a simple patch which is not being committed for unknown reason
since late August 2011.
https://issues.asterisk.org/jira/browse/ASTERISK-18301?focusedCommentId=183734#comment-183734
-Vladimir
On 10/18/2011
It has been almost a year since I suggested to consider including these
into the RPM build. There was no friction ever since, and I am building
from sources too...
It seems the RPM maintainers think that Google Voice connectivity is an
experimental feature and thus it should not be included in
On 11/27/2011 10:23 PM, Gaurav P wrote:
Do you build from source and copy res_jabber.so and chan_gtalk.so to
the rpm installed directories? Or have you just given up on the
packages and instead build from source?
On Sun, Nov 27, 2011 at 8:35 PM, Vladimir Mikhelson
v...@mikhelson.com
Ken,
Thank you for posting the details. The method worked perfectly.
I was about to give up on connecting via SSH to manually provisioned
Cisco phones.
Thank you,
Vladimir
On 1/15/2012 8:52 PM, Ken Alker wrote:
Flavio,
Thank you for pointing this out! I was using the reference
It's funny. The link
Links | https://issues.asterisk.org/jira/browse/ASTERISK-19202
Produces:
Permission Violation
It seems that you have tried to perform an operation which you are not
permitted to perform.
If you think this message is wrong, please consult your administrators
I have Avaya IPOffice 403 talking to my Asterisk 1.8.x with virtually no
issues using OOH323.
I am having some minor issues with the name portion of the caller ID
sent to Avaya. That may be relalted to a way FreePBX created the dial
plan. Maybe not. Never had time to systematically look into
You do need to wait until FreePBX updates the Asterisk.
Use yum to install the modules you need.
-Vladimir
On 2/21/2012 7:04 PM, Stephen Brown wrote:
DAHDI it is are there any known workarounds? I use the FreePBX
distro and they are a bit behind, so no telling when they will update.
Sean,
I do not have experience with the Amazon service. Cannot advise how to
implement it in their environment.
You need to have a route from your public IP(s) to your Asterisk
instance for all incoming connections on RTP ports.
Absence of this routing explains why SIP connection to your home
Kaya,
I do not have a definitive answer for you, but here are several things I
noticed.
1. fxo*ls*=1 -- I would definitely try /fxoks/ instead
fxs*ls*=2 -- I am not sure about your provider but I would try
/fxsks/ instead
2. [May 13 13:15:31] WARNING[3624]
Michelle,
I forwarded your message to the OOH323 maintainer / developer. Here is
his reply.
Vladimir, there is 1.6 asterisk which is unsupported already, you can
recommend upgrade to 1.8 or higher version. I can produce patch for 1.6
but upgrade is better way
Thank you,
Vladimir
On
*From:* asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir
Mikhelson [v...@mikhelson.com]
*Sent:* Wednesday, June 06, 2012 11:18 AM
*To:* Asterisk Users List
*Subject:* Re: [asterisk-users] OOh323 log fills with : In ooEndCall
call
and I will
send a private link to the patch for reworked ooh323 of required 1.6.
I can't publish this link due to Digium's development policy.
On 6/6/2012 10:51 AM, Vladimir Mikhelson wrote:
Michelle,
I have re-sent your message to the developer. I will let you know
when I get a reply.
I do
Andrew,
Did you try username=cli...@theirdomain.tld?
-Vladimir
On 6/15/2012 9:42 AM, Andrew McRory wrote:
asterisk-1.8.13.0
iksemel-1.4
I have a client who setup a gvoice account using their domain in the
login name:
username=client@theirdom...@gmail.com
This appears to have
Amit,
Make sure you have an option to return Digium TDM410P if it does not
work for you.
In my experience Digium TDM410P produce substantial background noise on
certain Dell computers. Generic TDM400 do not have this issue.
On top of that FXO channels exhibit intermittent problems with
are
simple and solvable and not related to the card itself.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir
Mikhelson
Sent: Saturday, June 16, 2012 5:37 PM
To: Asterisk Users Mailing List - Non
On 6/16/2012 5:38 PM, Shaun Ruffell wrote:
On Sat, Jun 16, 2012 at 04:37:06PM -0500, Vladimir Mikhelson wrote:
On top of that FXO channels exhibit intermittent problems with incoming
caller ID, FXS -- with DTMF detection. These two problems manifest
themselves with both Digium and generic
are
simple and solvable and not related to the card itself.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir
Mikhelson
Sent: Saturday, June 16, 2012 5:37 PM
To: Asterisk Users Mailing List - Non
On 6/16/2012 5:38 PM, Shaun Ruffell wrote:
On Sat, Jun 16, 2012 at 04:37:06PM -0500, Vladimir Mikhelson wrote:
On top of that FXO channels exhibit intermittent problems with incoming
caller ID, FXS -- with DTMF detection. These two problems manifest
themselves with both Digium and generic
-users-boun...@lists.digium.com] On Behalf Of Vladimir
Mikhelson
Sent: Saturday, June 16, 2012 7:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Help choosing the right card
Shaun, I respect your opinion, and the swap theory is one of the valid
Wieling wrote:
You have verified this by using the Asterisk's DTMF debug option?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir
Mikhelson
Sent: Saturday, June 16, 2012 9:37 PM
To: Asterisk Users
Steve,
The systems I tested on are all old Dell Dimension systems with plain
old PATA. I disabled all power saving features in the BIOS.
-Vladimir
On 6/17/2012 12:57 PM, Steve Edwards wrote:
On Sun, 17 Jun 2012, Shaun Ruffell wrote:
What I feel is the important clue in this case is the
wrote:
Vladimir Mikhelson v...@mikhelson.com writes:
But interestingly enough, yesterday morning I had zero (0) bytes in the
swap file and still experienced missing DTMF detection on an outgoing
call.
Executables do not get written to swap, their pages just get discarded
under pressure
On 6/17/2012 12:06 PM, Shaun Ruffell wrote:
On Sun, Jun 17, 2012 at 04:39:55PM +0200, Benny Amorsen wrote:
Vladimir Mikhelson v...@mikhelson.com writes:
But interestingly enough, yesterday morning I had zero (0) bytes in the
swap file and still experienced missing DTMF detection
On 6/17/2012 5:56 PM, Shaun Ruffell wrote:
On Sun, Jun 17, 2012 at 03:43:35PM -0500, Vladimir Mikhelson wrote:
On 6/17/2012 12:06 PM, Shaun Ruffell wrote:
I just updated the patch since the memory locks weren't carried
through after the fork call. When I apply the patch on the current
head
On 6/17/2012 6:21 PM, Shaun Ruffell wrote:
On Sun, Jun 17, 2012 at 06:14:07PM -0500, Vladimir Mikhelson wrote:
Shaun, would it be possible to lock specific modules in RAM vs. the
who;e Asterisk application?
It is possible but not without more work. Asterisk would need to
parse the output
Have you tried 1.8.15?
SIP TLS with self-signed certificate seems to be working fine here. The
OS is CentOS 5.8 and there are no chained certificates in my environment.
-Vladimir
On 8/5/2012 1:23 PM, Daniel Pocock wrote:
Package: asterisk
Version: 1:1.8.13.0~dfsg-1+b1
Severity: important
Carlos,
I am waiting for my Grandstreams to arrive too.
Similar reasons. Great feature set, reasonable price.
My primary interest is security. Grandstream claims their intermediate
and higher-end models support TLS and SRTP. I am really tired of trying
to make Cisco phones to communicate
Guys,
Is it possible to leave the Mantis on permanently?
It allows to productively search and work with issues recorded in it.
Search, convenient straight forward layout, patch download URLs,
everything just works there.
JIRA maybe is convenient for the management and developers. I just
On 8/28/2012 4:11 PM, Paul Belanger wrote:
On 12-08-28 10:25 AM, Matthew Jordan wrote:
- Original Message -
From: Vladimir Mikhelson v...@mikhelson.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc: asteriskt...@digium.com
Sent
On 8/28/2012 5:58 PM, Alec Davis wrote:
It allows to productively search and work with issues
recorded in it.
Search, convenient straight forward layout, patch download URLs,
everything just works there.
JIRA maybe is convenient for the management and
developers. I just
guess, as
and
will do a quick test. I'll let you know soon. It took them 9 days to
start looking into the issue.
I will update this thread with progress.
Regards,
Vladimir
On 8/17/2012 11:30 AM, Carlos Alvarez wrote:
On Fri, Aug 17, 2012 at 9:08 AM, Vladimir Mikhelson
v...@mikhelson.com mailto:v
.)
*From*: Vladimir Mikhelson v...@mikhelson.com
*Sent*: Friday, August 31, 2012 9:07 PM
*To*: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] Grandstream VoIP phones
Carlos,
So far
On 9/1/2012 8:27 AM, Patrick Lists wrote:
On 01-09-12 04:14, Vladimir Mikhelson wrote:
[snip]
* Ability to send host name or other CN not equal to the phone IP in
TLS negotiation
Afaik you usually put alternative CNs in SubjectAltName in the
certificate. Have you tried
Raj,
I am just confirming it happens here as well.
CentOS 5.7. Asterisk 1.8.15.1. DAHDI 2.6.1.
Digium, Inc. Wildcard TDM410 4-port analog card (rev 11)
Loadzone = us
The problem started manifesting itself after I switched to 1.8.x from
1.6.2.x
Typical scenario: a caller apparently hangs up,
Hans,
I did not try 10 or 11 as I run 1.8.15. Following are the related conf
files.
*gtalk.conf*
[General]
context = default
allowguest = yes ; Required if you want to accept calls from
people Not on your contact list.
bindaddr=private IP ;; These two settings are very critical
On 9/12/2012 12:05 AM, Raj Mathur (??? ?) wrote:
On Wednesday 12 Sep 2012, Vladimir Mikhelson wrote:
Raj,
I am just confirming it happens here as well.
CentOS 5.7. Asterisk 1.8.15.1. DAHDI 2.6.1.
Digium, Inc. Wildcard TDM410 4-port analog card (rev 11)
Loadzone = us
The problem
On 9/12/2012 5:33 PM, Sebastian Arcus wrote:
On 10/08/12 18:38, Chad Wallace wrote:
On Tue, 31 Jul 2012 09:44:26 +0100
Sebastian Arcuss...@open-t.co.uk wrote:
I have two setups with SIP hardware phones as extensions and POTS
lines as trunks. Internal SIP to SIP calls are crystal clear, but
On 9/12/2012 1:41 AM, Hans Witvliet wrote:
On Wed, 2012-09-12 at 00:01 -0500, Vladimir Mikhelson wrote:
Hans,
I did not try 10 or 11 as I run 1.8.15. Following are the related
conf files.
gtalk.conf
[General]
context = default
allowguest = yes ; Required if you want
On 9/13/2012 5:24 PM, Sebastian Arcus wrote:
On 13/09/12 00:47, Vladimir Mikhelson wrote:
On 9/12/2012 5:33 PM, Sebastian Arcus wrote:
On 10/08/12 18:38, Chad Wallace wrote:
On Tue, 31 Jul 2012 09:44:26 +0100
Sebastian Arcuss...@open-t.co.uk wrote:
I have two setups with SIP hardware
On 9/14/2012 6:04 PM, Alec Davis wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri
Sent: Saturday, 15 September 2012 8:45 a.m.
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
On 9/14/2012 6:04 PM, Alec Davis wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri
Sent: Saturday, 15 September 2012 8:45 a.m.
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
On 9/14/2012 10:11 PM, Matthew Jordan wrote:
- Original Message -
From: Vladimir Mikhelson v...@mikhelson.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, September 14, 2012 9:24:41 PM
Subject: Re: [asterisk-users] DTMF
On 9/14/2012 11:04 PM, Matthew Jordan wrote:
- Original Message -
From: Vladimir Mikhelson v...@mikhelson.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, September 14, 2012 10:39:30 PM
Subject: Re: [asterisk-users] DTMF
On 9/15/2012 6:16 AM, Alec Davis wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Vladimir Mikhelson
Sent: Saturday, 15 September 2012 5:56 p.m.
To: Asterisk Users Mailing List - Non
Hopefully the initial poster still has the configuration to
produce the files for you.
Are you saying the DTMF logs I attached do not provide enough
evidence to support the theory of the DTMF length being the
cause of this issue?
-Vladimir
Vladimir,
What was the
Matt,
Please see my answers in-line. Thank you for looking into the issue.
-Vladimir
On 9/15/2012 12:36 PM, Matthew Jordan wrote:
- Original Message -
From: Vladimir Mikhelson v...@mikhelson.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
On 9/15/2012 5:16 PM, Alec Davis wrote:
[2012-09-15 22:36:44.489226] DTMF[1706] channel.c: DTMF end
'4' received on
SIP/alec-0009, duration 1660
Alec,
Interestingly in your log DTMF durations are even greater
than in my original sampling. Well,
On 9/15/2012 6:28 PM, Matthew Jordan wrote:
- Original Message -
From: Vladimir Mikhelson v...@mikhelson.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, September 15, 2012 1:11:14 PM
Subject: Re: [asterisk-users] DTMF
On 9/17/2012 4:39 PM, Josh Hopkins wrote:
snip
While asterisknow uses freepbx to control the config files. Where and how
would I go about putting this into freepbx or another loaded config file that
where something like the above would work. Thanks,
/Josh
Josh,
You may want
I am still to see a single bit of help from them.
I will continue updating this thread.
-Vladimir
On 8/31/2012 8:07 PM, Vladimir Mikhelson wrote:
Carlos,
So far the experience with DP715 is extremely negative.
It all starts with the WEB interface which is only served on port 80,
no https
On 10/1/2012 4:15 PM, Mark Michelson wrote:
Hi!
I've been confronted with an interesting issue to resolve. The
issue is located here:
https://issues.asterisk.org/jira/browse/ASTERISK-20163
The issue involves case-sensitivity of channel and global variables
in the dialplan. Current
On 10/2/2012 9:12 PM, Warren Selby wrote:
On Mon, Oct 1, 2012 at 4:15 PM, Mark Michelson mmichel...@digium.com
mailto:mmichel...@digium.com wrote:
Hi!
I've been confronted with an interesting issue to resolve. The
issue is located here:
snip
So respond here and let
Hi,
Did anybody upgrade the kernel to 2.6.18-308.16.1.el5 on CentOS 5.7?
If the answer is Yes did you run into issues with DAHDI 2.6.1?
I am observing the missing kmod-dahdi-linux.i686
2.6.1-1_centos5.2.6.18_308.16.1.el5 in Digium depository.
I do not seem to be able to compile DAHDI 2.6.1
Hi,
I am experiencing a similar issue on my FXO DAHDI lines with proper call
supervision supposedly set up. The problem described below does not
happen 100% of the times, but still several times a day. Scenario seems
to be similar to the one the original poster described, i.e. a caller
hangs up
Roy,
Many will say that it all depends on your provider supporting T.38, and
that you should forget it otherwise.
My practical experience shows otherwise. I am able to receive faxes on
SIP lines pretty reliably with no T.38 support. The biggest issue for
me is CED tones detection. If CED is
-COOP (2667)
On 11/6/2012 6:28 AM, Chris Nighswonger wrote:
On Mon, Nov 5, 2012 at 10:43 PM, Vladimir Mikhelson
v...@mikhelson.com mailto:v...@mikhelson.com wrote:
My practical experience shows otherwise. I am able to receive
faxes on
SIP lines pretty reliably with no T.38 support
Yep, there has been no updates for at least the last 2.5 months.
I would also like to find out what the plans are.
-Vladimir
On 11/23/2012 5:43 PM, Eric Germann wrote:
Will there be an update to the RPM repo on packages.asterisk.org?
For example
Zohair,
I am not sure about the specifics of 7942 as I use 7906.
Connected line CID shows up on my 7906 with the following sip.conf settings:
* trustrpid=yes
* sendrpid=yes
-Vladimir
On 2/15/2013 11:09 AM, Zohair Raza wrote:
Hi,
Is it working for anyone?
I have tried with
, Vladimir Mikhelson
v...@mikhelson.com mailto:v...@mikhelson.com wrote:
Zohair,
I am not sure about the specifics of 7942 as I use 7906.
Connected line CID shows up on my 7906 with the following sip.conf
settings:
* trustrpid=yes
* sendrpid=yes
-Vladimir
Chris,
Thank you for sharing. It will help one day when 11 will become stable
enough to consider it for a production system.
Interestingly, jabber.conf has the same exact setting and the same exact
value and comment in 1.8.20.1:
priority = 1;; Resource priority
Unless the
Bilal,
Here is the respective section from my working 7906 .conf file:
dateTimeSetting
dateTemplateM/D/Ya/dateTemplate
timeZoneCentral Standard/Daylight Time/timeZone
ntps
ntp
name172.29.100.11/name
ntpModeUnicast/ntpMode
/ntp
On 3/23/2013 10:45 AM, Harley Peters wrote:
I'm running asterisk 1.8.10.1 and can confirm it works the same way.
I had it set to 1 originally and it worked fine at first then suddenly
stopped.
It drove me crazy until I ran across this link:
Oliver,
You may want to look into the latest README for DAHDI 2.7.0
How to get OSLEC from dahdi-linux-extra
HTH,
Vladimir
On 6/6/2013 12:05 PM, Olivier wrote:
Hi,
I'm used to add OSLEC source code into asterisk and use it as default
echo canceller.
Currently, I'm proceeding this way:
Hi All:
Has anybody tackled the latest Google Voice issue where incoming and
outgoing calls for certain Google Voice accounts fail?
I have filed the bug report with details
https://issues.asterisk.org/jira/browse/ASTERISK-22176
For incoming calls Asterisk does not reply to the initial XML
A quick update.
The nick: theory was proven to be wrong. The incoming calls
consistently fail with or without nick: tag.
I am concentrating on the incoming calls for now.
-Vladimir
On 7/21/2013 3:34 PM, Vladimir Mikhelson wrote:
Hi All:
Has anybody tackled the latest Google Voice issue
12:02 PM, Vladimir Mikhelson wrote:
A quick update.
The nick: theory was proven to be wrong. The incoming calls
consistently fail with or without nick: tag.
I am concentrating on the incoming calls for now.
-Vladimir
On 7/21/2013 3:34 PM, Vladimir Mikhelson wrote:
Hi All:
Has anybody
Richard,
And what is condition 33 after all? Maybe it needs to be processed,
not ignored.
Thank you,
Vladimir
On 10/23/2013 7:06 PM, troxlinux wrote:
thnk Richard Mudgett for your quick response , but I have a question I
am using the asterisk 11 with ooh323 by default, I can update it?
these warning on my console
2013/10/23 Matthew Jordan mjor...@digium.com mailto:mjor...@digium.com
On Wed, Oct 23, 2013 at 7:12 PM, Vladimir Mikhelson
v...@mikhelson.com mailto:v...@mikhelson.com wrote:
Richard,
And what is condition 33 after all? Maybe it needs
IPSec VPN is what we use. The biggest issue is QoS.
I do not see any other inexpensive solution.
-Vladimir
On 12/4/2013 12:10 PM, Rodrigo Borges Pereira wrote:
Have seen mostly a lot of hq/branch implementations using IPSEC over
standard Internet connections.. that's what seems most cost
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