We are having a weird problem where calls get cut off in the middle. I'm not a
SIP expert but could the INVITE with an empty SDP be the problem?
|Time | 209.220.119.18|
| | | 208.88.61.150 |
|9687.369 | INVITE
PHP has had memory leak issues in the past, though modern versions are
apparently much better.
The thing is, when you write a daemon you must EXPECT it to exit at some point,
maybe the socket went away or the system restarted or you are out of memory,
whatever. You need to make sure you
In Asterisk 1.8+ you should be able to do something like
Set(__SIP_CODEC_OUTBOUND=${ SIP_CODEC_INBOUND})
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian
Sent: Friday, February 15, 2013 8:27 PM
To:
I would like to configure Asterisk send back only a Trying or Progress message
to the SIP client and not any early audio for ringback. I've confirmed
Asterisk is sending RTP when the call is ringing by using rtp debug on Asterisk.
Does anyone have any ideas on how to accomplish this?
I've
I think the default DTMF tone duration is 100ms, if you are dialing 10 digits,
that ends up being 1 second delay just to dial the DTMF, not including
inter-digit delays. Try setting toneduration=50 in chan_dahdi.conf and see
what happens. If you make it too low your telco will miss some
The easiest thing to is renumber one of the networks so they are not using the
same address block.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank
Sent: Thursday, February 07, 2013 12:27 PM
To:
Another option would be a VPN between the phone and the LAN the Asterisk box is
on. VPN software may handle IP address changes better than the Softphone.
This way the IP of the softphone doesn't change.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Looks to me like ${prefix} contains nothing but two quotes.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
Sent: Friday, January 25, 2013 11:09 AM
To: Asterisk Users Mailing List -
What version does the error occur on? I suspect more recent versions of
Asterisk removes extraneous quotes.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, January 25, 2013 11:20
:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Quoting error with gotoiftime
On Fri, Jan 25, 2013 at 9:20 AM, Eric Wieling ewiel...@nyigc.com wrote:
Looks to me like ${prefix} contains nothing but two quotes.
Which is as it should
Using qualify=10 ?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher
Harrington
Sent: Tuesday, January 22, 2013 5:11 PM
To: Danny Nicholas
Cc: Asterisk Users Mailing List - Non-Commercial
I should have actually suggested qualifyfreq=10.
Qualify has nothing whatsoever to do with registration.
-Original Message-
From: Danny Nicholas [mailto:da...@debsinc.com]
Sent: Tuesday, January 22, 2013 5:15 PM
To: Eric Wieling; ch...@acsdi.com; 'Asterisk Users Mailing List -
Non
Yes. That is why we don't use this setting.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
Sent: Tuesday, January 22, 2013 6:44 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Stop Asterisk. Start Asterisk as asterisk -cvvvd then in the CLI issue the
command agi set debug on.
Starting Asterisk as this command will make it run in the foreground and show
you STDERR. You do not normally get to see STDERR when running AGIs in
Asterisk.Since Asterisk is running in
I am also experiencing this issue. Asterisk is in fact running, you can verify
by running asterisk -rvvv (-r connects to an EXISTING asterisk process) or
using ps.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
From voip-info.org:
If srvlookup is turned on, Asterisk supports DNS SRV lookups partially.
Currently, Asterisk only reads the first SRV entry without bothering with
priorities and weights.
Is this still the case with Asterisk 1.8?
--
No. However you can do this: exten = _520xx/_0666XX,1,hangup
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of isr...@gmail.com
Sent: Monday, January 14, 2013 11:57 AM
To: Asterisk Users Mailing List -
I only see that message when I have sip debug enabled. It appears harmless.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Emre Özcan
(Alfacom)
Sent: Friday, January 11, 2013 10:34 AM
To:
In Asterisk extensions.conf and extensions.ael inside $[] = and == are the
same comparison operator. I can't quote where I saw this, but it has been
documented somewhere. The == was added to make things more programmer
friendly.
-Original Message-
From:
No. You may only have one registration per peer or friend. It cannot be
changed.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of XBrian
Sent: Saturday, January 05, 2013 5:17 PM
To:
Trust me, Verizon doesn't really provide support.What they will do is tell
you something different (often conflicting stuff) when you send in a ticket.
One time they tell us the From must be in e.164 format, other times they say it
does not.We asked for an updated Interop guide weeks
I believe Asterisk 11 is the first version which allows you to enable and
disable faxdetect on the fly.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Friday, January 04, 2013 2:42 PM
To:
It doesn't matter. They still require IPSEC VPN.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael L. Young
Sent: Thursday, January 03, 2013 10:32 AM
To: Asterisk Users Mailing List - Non-Commercial
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher
Harrington
Sent: Wednesday, January 02, 2013 10:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Top
I recommend using WaitForRing instead of Wait.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, January 02, 2013 3:33 PM
To: rwhee...@artifact-software.com; 'Asterisk Users
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Thursday, December 27, 2012 3:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] $100 Bounty: Level 3/Asterisk/Adtran T.38
Pass-Through
I was not aware you needed
In a condition (i.e. not using Set) when you put quotes on one side of the =
sign, then you need to put it on the other side of the = sign as well.
ExecIf(${ARG4} != ]?
I don't know if this is your specific problem, but if you don't fix it, it will
come back and bite you later.
I suspect
I'm trying to get T.38 passthrough working on Asterisk 1.8.18.1. It isn't
working. Calls go through and are answered, but the fax machines are unable
to communicate. I checked the value of CHANNEL(t38passthrough) and it seems
to always be 0. One side is Level 3 T.38 TN and the other side
Setting directmedia=no does not help. The calls still go through and the fax
still fails after switching to T.38.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Thursday, December 27
We are offering $100 (paid via paypal or check) to the first person who assists
us in successfully sending and receiving faxes in the setup described below.
Offer expires Dec 31. We are a direct customer of Level 3, there is no other
carrier involved.
What we want to work:
Level 3 T.38
Eric Wieling ewiel...@nyigc.com
We are offering $100 (paid via paypal or check) to the first person who
assists us in successfully sending and receiving faxes in the setup described
below. Offer expires Dec 31. We are a direct customer of Level 3, there is no
other carrier involved
, December 27, 2012 1:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38
Pass-through
directrtpsetup=yes in sip.conf?
On Thu, Dec 27, 2012 at 12:09 PM, Eric Wieling ewiel...@nyigc.com wrote:
We have set
It does not appear to make any difference. Calls are still failing.
-Original Message-
From: Christopher Harrington [mailto:ch...@acsdi.com]
Sent: Thursday, December 27, 2012 1:20 PM
To: Eric Wieling
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
PM
To: Eric Wieling
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38
Pass-through
Last thing to check, just for sanity's sake:
t38pt_udptl=yes in sip.conf? It defaults to off.
On Thu, Dec 27, 2012 at 12:32 PM
I was not aware you needed SpanDSP for T.38 passthrough.. How will that work
with the UDPTL packets not going through Asterisk.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday,
/Asterisk/Adtran T.38
Pass-through
On 28/12/2012 1:55 AM, Eric Wieling wrote:
We are offering $100 (paid via paypal or check) to the first person who
assists us in successfully sending and receiving faxes in the setup described
below. Offer expires Dec 31. We are a direct customer of Level 3
Cause 20 means your SIP device is not registered or you do not have an IP
specified for it in your peer.
sip show peers will show that.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Scott Huang
Sent:
You need to look at the device which the analog lines plug into. There is
nothing to change in Asterisk for this issue.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Tuesday, December 11, 2012
When will packages.asterisk.org be updated with the RPM's?
Thanks
EKG
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk
Development Team
Sent: Thursday, December 06, 2012 2:47 PM
To: Asterisk Users
The Wiki is (always) out of date. You might consider taking a look at
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#DialplanBasics_id262049
which is likely less out of data.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Will there be an update to the RPM repo on packages.asterisk.org?
For example http://packages.asterisk.org/centos/5/asterisk-1.8/x86_64/RPMS/
Latest is showing 1.8.15.1.
Thanks
EKG
--
_
-- Bandwidth and Colocation Provided
No, you don't need your script for this.
Prevent attacks by using fail2ban to block brute force attacks using iptables,
securing your server at the OS level, and NEVER EVER EVER let leave the web GUI
for FreePBX open to the internet. I'm sure others have more suggestions.
Over the years 100%
We are getting this message on an Asterisk 1.4.44 box.
[2012-11-19 08:49:27] WARNING[11785] app_voicemail.c: List of extensions is too
long (1323). Truncating.
I know Asterisk removed many of limitations in string lengths in in 1.6+. Does
anyone know if this also applies to app_voicemail?
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Monday, November 19, 2012 12:01 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Max app_voicemail line length
We are getting this message on an Asterisk 1.4.44 box.
[2012-11-19 08:49:27] WARNING[11785
Polycom phones after firmware 2.x register to BOTH the primary and backup
servers.
On Thu, Nov 15, 2012 at 8:59 AM, Chris Nighswonger
cnighswon...@foundations.edu wrote:
Would the simplest approach to failover be to just configure my
primary asterisk server as the
module unload res_musiconhold.so
and
module load res_musiconhold.so
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus
Sent: Tuesday, November 13, 2012 1:00 PM
To: Asterisk Users Mailing List -
Hi all,
I'm trying to troubleshoot an issue with my SIP service. All outgoing
calls work normally. The following is a SIP debug log from Asterisk. The
test setup is as follows:
One Yealink SIP-T22P phone (10.0.0.107), extension 10, configured to talk
to my local FreePBX/Asterisk 11.0 server
(first use in this function)
This is identical to the error reported in this patch fix:
https://issues.asterisk.org/jira/browse/DAHTOOL-60?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel
How would I apply the patch included in the above url?
[eric@pepper ~/src/asterisk-complete
Thanks Adolphe
Seems I have that already:
[eric@pepper ~] $ dpkg -l linux-headers-`uname -r`
||/ NameVersionArchitecture
Description
No, it isn't.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher
Harrington
Sent: Thursday, October 25, 2012 12:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
I'm setting up a test server with a Digium TE122 and am getting the following
error on the console, spewing as fast as it can. Does anyone have any idea
what this error might be?
[Oct 19 11:24:53] NOTICE[2076]: chan_dahdi.c:3108 my_handle_dchan_exception:
PRI got event: Event 59 (59) on
I seem to recall seeing somewhere recently where there was a bugfix for
ulaw/alaw conversion which would cause poor audio.
Have you tried updating your Asterisk to the latest of whatever major version
you are running?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Have you tried Dial instead of Transfer?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deepesh D
Sent: Thursday, October 11, 2012 2:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Try: exten = _00336123412xx/_44XX.,1,Set(RINGTIME=90,g)
Notice the _ on your callerid pattern
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Geoffrey Yeoh
Sent: Thursday, October 11, 2012 1:15 PM
To:
Has anyone seen issues with recent Sangoma T-1 cards and Sangoma Analog cards
on multiple different servers?
On T-1: we get NO traffic, no interrupts, and no increase in number of packets
and the PRI does not come up.
On Analog: The ports do NOT go red when you unplug the phone line from FXO
Once an option is set in the chan_dahdi.conf file it applies to every channel
= line listed after the setting, until the option is changed. This is all you
really know.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
A port is not a door if there is nothing listening on the port.
Open ports are not a security issue. Stuff running on open ports are.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
Sent:
' rejected because extension not found.
I am running an ancient Asterisk 1.4.26.2 (yes I know what I should be doing).
Why might the deny not be working?
--
Eric Smith
--
_
-- Bandwidth and Colocation Provided by http://www.api
You do not have an exten = 700972595637212 in the context in extensions.conf
that SIP device is set to in sip.conf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Smith
Sent: Monday, October 01, 2012 4
You are set up as a USA PRI, but not dialing a USA TN.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz
Sent: Wednesday, September 26, 2012 11:13 AM
To: 'Asterisk Users Mailing List - Non-Commercial
You are doing it wrong. I know 50 bazillion Asterisk dialplan examples on the
internet do it the same way. It is still wrong.
When you do a Dial on the dialplan you need check the value of DIALSTATUS or
HANGUPCAUSE before dialing again. Both variables will give you some indication
of why
Going to n+101 was deprecated in Asterisk 1.2 or 1.4. Don't use it.. Read the
docs for Authenticate and see what diaplan variables you can check.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk
: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Friday, September 14, 2012 10:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Fax Detect on Demand
2012/9/13 Eric Wieling ewiel
Discussion
Subject: Re: [asterisk-users] Fax Detect on Demand
2012/8/16 Eric Wieling ewiel...@nyigc.com
Using Asterisk 1.8.mumble. We would like to use fax detect on demand.
Both chan_dahdi and chan_sip support setting fax detetect on a static
basis,
For curiosity's sake
Try adding qualify=yes
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Tuesday, September 11, 2012 3:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Your best bet is a carrier class device from someone like Adtran and convert
the PRIs to SIP before passing the calls to Asterisk.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA
Sent:
pbx*CLI core show function CHANNEL
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stefan at WPF
Sent: Friday, August 24, 2012 6:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Adding the IPs of ALL local interfaces to /etc/hosts has helped solve this
issue for me in the past.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Thursday, August 23, 2012 4:25 PM
To:
Using Asterisk 1.8.mumble. We would like to use fax detect on demand.
Both chan_dahdi and chan_sip support setting fax detetect on a static basis,
but no way I've been able to find to enable/disable it on demand in the
dialplan.
In 1.4 we used the NVFaxDetect 3rd party app, but that no longer
Has anyone experimented with increasing the DAHDI chunk size in improve fax
reliability? If so, did it help, hurt, or not make any difference?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
Using n with labels is what most people do. A dialplan isn't javascript, you
don't need two hundred 3 line functions.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini
Sent: Friday, August 03,
...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Wednesday, August 01, 2012 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem
Sounds like DAHDI/4 is a FXO port. FXO ports are considered answered when
dialing is completed
Is there an ETA on when this will show up on packages?
Thanks for the work!
EKG
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
I'm getting the following warning with SOME calls on 1.8.14.1 Is it a cause
for concern? Is there a way to fix it? I can't tell for sure if it is
impacting calls or not.
WARNING[27796]: chan_sip.c:20497 handle_response_invite: just did sched_add
waitid(4077) for sip_reinvite_retry for
The Adtran NetVanta series has a number of good devices which support PRI/SIP
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Saturday, July 28, 2012 6:43 PM
To: Asterisk Users Mailing List -
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming
Sent: Friday, July 27, 2012 10:45 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] still got ReceiveFax() problem, how to
1) Does anyone know of any software to debug the g711cap audio files Asterisk's
res_fax generates? Google has not been very helpful.
2) These files are in WAV format, but my Windows Media Player cannot play them.
The Linux file command reports RIFF (little-endian) data, WAVE audio,
Microsoft
@lists.digium.com
Subject: Re: [asterisk-users] T.38 (PRI) Fax Debugging
On 07/20/2012 09:48 AM, Eric Wieling wrote:
Neither of your questions relate to T.38 :-) Maybe you meant T.30 instead.
1) Does anyone know of any software to debug the g711cap audio files
Asterisk's res_fax generates
We are using res_fax_digium with a Sangoma PRI card on asterisk 1.8.13
The docs at http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf
indicate v34 is supported, but when I enable it I get the message
res_fax_digium.c:1624 dgm_fax_new: V.34 not supported, will be ignored. Is
v34
, July 18, 2012 11:28 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 1.8.13 / res_fax / res_fax_digium
On 07/18/2012 10:06 AM, Eric Wieling wrote:
We are using res_fax_digium with a Sangoma PRI card on asterisk 1.8.13
The docs at http://docs.digium.com/FAX
Remove the ,i to start with. Do you have the various rpid related options in
sip.conf set?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Wednesday, July 18, 2012 12:08 PM
To: 'Asterisk Users
Why would you NOT want the connectedline info sent immediately?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett
Sent: Wednesday, July 18, 2012 12:24 PM
To: Asterisk Users Mailing List -
Channels can be in more than one group.
Make g0=1-15,17-31,32-46,48-62 and
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mahesh katta
Sent: Tuesday, July 10, 2012 10:04 AM
To: Asterisk Users Mailing List -
Recent Polycom firmware versions (4.x, I think) also have support for user
sort of stuff. See the 4.x Admin Guide.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming
Sent: Tuesday, July 10,
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Tuesday, July 10, 2012 5:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FreePBX: using
Asterisk, and by extension FreePBX, automatically end the voicemail recording
when the caller hangs up. You have some OTHER issue. Perhaps Asterisk is not
detecting the hangup?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
I've never seen this on incoming calls, only outgoing calls.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Colin
Sent: Friday, June 29, 2012 8:11 AM
To: Asterisk Users Mailing List - Non-Commercial
Is there anything specific in the plaintext SIP packets you want to secure?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B
Sent: Friday, June 22, 2012 1:57 PM
To: Asterisk Users Mailing List -
We have been quite disappointed by the Adtran VQM. It often shows calls which
had audio issues as being close to perfect. It also often shows calls which
sound perfect as having significant quality issues.
We don't allow reinvites so this might be part of the issue. I don't have a
lot more
This is a Polycom question, not an Asterisk question.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz
Sent: Tuesday, June 19, 2012 1:38 PM
To: 'Asterisk Users Mailing List - Non-Commercial
In my experience when you have intermittent problems with incoming caller ID,
FXS -- with DTMF detection you have to adjust your rxgain and/or txgain. I am
NOT a fan of Digium cards, but these CallerID and DTMF issues are simple and
solvable and not related to the card itself.
-Original
I was assuming incoming DTMF detection. Try toneduration=250 in chan_dahdi to
increase the duration of transmitted DTMF on your DAHDI channels. If that
fixes it, try lowering it. I find 80 usually works with even the worst IVRs.
-Original Message-
From:
-Commercial Discussion
Subject: Re: [asterisk-users] Help choosing the right card
Eric,
Thank you for the suggestion.
In fact the problem is with FSX channel which fails to catch some DTMF tones
from a phone which places an outgoing call. Shaun's theory was a delay related
to swapping
For some reason 1.4.4.x was not reading chan_dahdi.conf. When I symlinked it
to zapata.conf it worked.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Wednesday, May 30, 2012 2:35 PM
When will this be available at packages.asterisk.org?
Thanks!
EKG
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
What does the output of g729 show licenses show? If it doesn't show licenses
then Asterisk is not licensed for G729 codec.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim King
Sent: Tuesday, June 05,
Last time I checked (a few years ago) Sangoma has half height brackets
available. Contact their support or sales.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ade Vickers
Sent: Friday, June 01, 2012 10:41
-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria Anamarta
Sent: Saturday, June 02, 2012 12:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk pickup call on first ring
Thanks Eric for the prompt reply
This is incorrect. The vast majority of settings in chan_dahdi.conf are
applied when you do a module reload chan_dahdi.so
You cannot change signaling, switchtype, or add or remove channels (I'm sure
there are a few others) on a module reload, but most settings will be applied
on a reload.
If
Has anyone experienced an issue with Sangoma analog cards where the card
suddenly stops working? Trying to dial out shows the channel as busy, even
though there is no active call on that port?
This happened to us often when we used Digium cards (in fact this issue is why
we stopped using
301 - 400 of 3919 matches
Mail list logo
