t; exten => h,n,NoOP(${DIALSTATUS})
>
> The endpoint may register from multiple device, so I always have to dial
> it all contacts. Did anyone else face such problem?
>
You need to examine if the returned dial string is empty in your dialplan.
PJSIP_DIAL_CONTACTS retu
git_full:
> Unexpected control subclass '-1'
> -- User entered nothing.
>
You didn't specify the Asterisk version. You can ignore this message.
Current versions simply suppress this message for -1 in that routine.
Richard
--
___
ent in queue.conf is has also changed since
chan_agent no longer exists
in Asterisk 12+.
See
https://www.asterisk-blog.com/2016/02/10/converting-from-chan_agent-to-app_agent_pool/
Richard
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keeping PJSIP stack ?
>
Do you have the following options enabled in pjsip.conf?
;trust_id_inbound=no; Accept identification information received from
this
; endpoint (default: "no")
;trust_id_outbound=no ; Send private identification details to the
endpoint
On Wed, Jun 8, 2016 at 11:57 AM, Michael Maier <m1278...@allmail.net> wrote:
> On 06/06/2016 at 04:40 PM Richard Mudgett wrote:
> > On Sun, Jun 5, 2016 at 3:48 AM, Michael Maier <m1278...@allmail.net>
> wrote:
> >
> >> Hello!
> >>
> >>
s time - there wasn't any call
> or anything other to process.
>
> I've got the complete wireshark trace of the situation described above.
>
Those key exist messages are due to a race condition. From what I've
seen
global SIP nat setting and the per peer/user nat setting for the indicated
peer/users.
The warning messages are indicating a potential security vulnerability in
your
configuration for each peer/user and are describing what can happen and
what you
need to do if those peer/users are exposed to the outs
On Tue, May 3, 2016 at 8:59 PM, Richard Mudgett <rmudg...@digium.com> wrote:
>
>
> On Tue, May 3, 2016 at 6:15 PM, Derek Bolichowski <de...@empire-team.com>
> wrote:
>
>> I posted this over in asterisk-dev, realized I probably should have put
>> it her
this.
>
This issue has been around a long time and was just recently fixed and I
think
it was just released in the latest v11 version.
See https://issues.asterisk.org/jira/browse/ASTERISK-16115
Richard
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ced by ConfBridge
>
Administrator TOOTAI: You must have DAHDI running when using meetme because
DAHDI does the audio mixing for the conference.
Meetme is deprecated and replaced by ConfBridge on all currently supported
Asterisk
versions.
Richard
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to do with SIP. You don't need them
for SIP.
Richard
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otocol
> as SIP port are blocked in my country.
>
>
> please help if it's possible. thanks in advance
>
Please do not hijack threads.
Richard
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At least in version 12.2.0, the code in cdr.c appears to create CDR
records for each pair of users in a conference. This is quadratic
and would seem to be an issue with large conferences.
I got two Asterisk crashes when a lot of people tried to dial into a
conference. They appear quite related
assigned to
lines two, three, and four ( ACCOUNTS 2,3,4).
How do I do this?
Thanks,
--
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rsch...@gmail.com
rsch...@optonline.net
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The patch is actively
being
reviewed/updated to get it merged into the codebase.
[1] https://issues.asterisk.org/jira/browse/ASTERISK-25791
[2] https://gerrit.asterisk.org/#/c/2293/
Richard
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On Tue, Feb 23, 2016 at 3:01 PM, Jefferson B. Limeira <
j...@internexxus.com.br> wrote:
> Ops! Sorry Richard, more information:
>
> # asterisk -V
> Asterisk 11.17.1
> # asterisk -rx 'pri show version'
> libpri version: 1.4.15
>
> I found some information: my a
libpri call
structure
to be left associated with the B channel. It could be glare, call aborted
early,
hangup never completed, channel got RESTARTed, etc.
You will need to read [1] and use the CLI "pri set deb
asterisk and start recording !?
>
> Looking for some ideas and hints.
>
Asterisk will not allow direct media for a call when it has an interest in
the media stream.
In other words, if you enable a feature-code (such as the Dial xXtThHkK
option flags), call
recording, etc.
nd I failed. What is needed on the
> Asterisk side to reply 484 to INVITE? Phones are talking to chan_pjsip
> on Asterisk-13.7.1.
>
Look into the Incomplete application.
https://wiki.asterisk.org/wiki/display/AST/Asteris
On Wed, Feb 17, 2016 at 5:56 PM, Ernie Dunbar <maill...@lightspeed.ca>
wrote:
> On 2016-02-17 15:32, Richard Mudgett wrote:
>
>> On Wed, Feb 17, 2016 at 5:15 PM, Ernie Dunbar <maill...@lightspeed.ca>
>> wrote:
>>
>> Hi everyone.
>>>
&g
>
This is defined by spandsp itself in one of its headers. Have you
installed the spandsp dev package?
Richard
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ing in the documentation for the phone or FreePBX
related to this issue.
Anyone? This is frustrating and I will be grateful for any help.
Thank you!
Richard
--
Richard C. Schroeder
rsch...@gmail.com
rsch...@optonline.net
516-859-1129 - C
rkedcalls context into
your ramais context.
Richard
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eed to look at your config files for the specific
channel driver. Alternatively, you should be able to clear a configured
group by setting it to the empty string.
Richard
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doesn’t appear to have any
> effect. I’ve done some searching and not come up with anything. I don’t
> believe it’s a FreePBX-specific issue, but can’t say for sure. Any
> guidance would be appreciated.
>
rtp_timeout is a per-endpo
on the built-in nokia 95 SIP client.
>>
>
> I haven't heard of this or seen it in testing, I don't think an issue
> exists for it.
>
On the subject of nonce length there is this issue about chan_sip's nonce
length
being too short:
https://issues.asterisk.org/jira/browse/ASTERISK-2506
are large, so here's the backtrace.txt
>
You are linking to a *static* version of the PJPROJECT library.
That is guaranteed to cause crashes. You must remove the static
build of PJPROJECT from your system.
Richard
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On Wed, Sep 23, 2015 at 5:53 PM, Ryan, Travis <ry...@oscarwinski.com> wrote:
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Richard Mudgett
> *Sent:* Wednesday, September 23, 2015 6:52 PM
> *To:* Ast
to Git. We'll try to get it sorted out for the next release.
In addition for the git patches you will need to use -p1 instead of -p0.
Richard
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A Siren codec is not currently available and the one for 12 will not
work. I have no timeframe for when this might change.
So the only option is to build one from the Polycom sources? I'm
already doing this for Siren14 (I forget why).
--
Alas, until we get off our butts, yes. Sorry about that.
Really, we're putting as much effort into fixing things and issues
that affect a lot of people. While siren7/siren14/silk are nice, there
aren't as many people using them as other affected things at this
moment.
Is there something
What is the proper version of the Siren7 codec to use for Asterisk 13.5.0?
Since there's nothing later, does the version for 12.0 work?
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.
- Build hundreds static agent-id in agents.conf
- Dynamic agent-id in mysql table (Not associated with agent.conf). Is
this possible?
Either way is fine as unused static agents don't use much memory.
Richard
= Local/800@agents,0,Name 1,Agent:1001
Replace the above line for agent 1001 with the following:
member = Local/1001@agents,0,Name,1,Agent:1001
For other agents follow the similar pattern:
member = Local/agent-id@agents,0,Name,1,Agent:agent-id
Richard
://69.59.234.67;tag=as69898393'
ubuntu*CLI
Use the AMI Originate action or a call file. You can specify a caller
id there. You cannot specify one from the command line.
Richard
Hi Richard
What should I use for extension? Since I am not bridging an extension with
outbound, but making
specify one from the command line.
Richard
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handle_response_invite: Received response: Forbidden from
'Anonymous
sip:did@69.59.234.67http://69.59.234.67http://69.59.234.67
;tag=as69898393'
ubuntu*CLI
Use the AMI Originate action or a call file. You can specify a caller
id there. You cannot specify one from the command line.
Richard
I'm planning on upgrading to Asterisk 13.4 soon and am looking for the
corresponding Siren7 codec. Where do I find it?
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by subversion. For the git patch you would need to use
-p1 for the subversion
patch you would need to use -p0. The patch program gave you this hint when
it failed to apply
the patch: Perhaps you used the wrong -p or --strip option?.
Richard
This is an interpolated frame from func_jitterbuffer. It's part of
packet loss concealment. What scenario exposed this?
We were testing for clipping by doing Set(VOLUME(RX)=100) but we were
connecting to a ConfBridge that had a jitterbuffer. This occurred when
the phone (SIP) hung up.
--
I'm getting a SIGSEGV at ast_slinear_saturated_multiply at the line:
351 res = (int) *input * *value;
It's called from ast_frame_adjust_volume.
The frame looks like:
(gdb) print *f
$6 = {frametype = AST_FRAME_VOICE, subclass = {integer = 100021, format = {
id =
: If immediate=yes the dialplan execution will always start at
extension
; 's' priority 1 regardless of the dialed number!
;
;immediate=yes
Richard
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CALLERID is a read only variable.
That's not correct. I set it all over the place in my dialplan.
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release? I am running 11.17.1.
No. It means that you have not loaded func_channel.so.
Richard
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Question: is there some built-in way to know if macro
feature1-ClientA is defined? Something liken
ExecIfMacro(feature1-ClientA)?macro(feature1-ClientA):Goto(...).
A macro is a context, so DIALPLAN_EXISTS should work if you specify an
extension and priority that's in the macro
reviews. Those
have been put on their own list since Gerrit requries
code reviews for every branch.
asterisk-code-rev...@lists.digium.com
Richard
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need to set a number to go with a name. Depending upon the
channel driver protocol, a name with no number may not be enough to send
out on its own.
Richard
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them.
Richard
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On Tue, Mar 24, 2015 at 4:59 PM, Jeff LaCoursiere j...@jeff.net wrote:
On 03/24/2015 04:28 PM, Richard Mudgett wrote:
On Tue, Mar 24, 2015 at 4:17 PM, Jeff LaCoursiere j...@jeff.net wrote:
Hello,
I am wondering if asterisk does anything at all to RTP packets passed
from channel
://wiki.asterisk.org/wiki/display/AST/Variable+Inheritance
Richard
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is easy by using the dial
option U(...), but if I dial two extensions at once, when the first
answers, the other stops ringing.
Any idea to make the first continue to ring until the other accept the
call?
Sounds like you want to use the FollowMe dialplan application.
Richard
What are the cons, if any, of enabling a jitterbuffer?Â
Memory and latency.
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in the audio
stream.
Remove the tTkK flags in the Dial command.
Richard
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://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
for more information.
Richard
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functionality.
You usually need to install all of the bridging technologies.
Richard
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On Tue, Dec 9, 2014 at 2:58 PM, Patrick Beaumont
p.beaum...@hatsoffsoftware.co.uk wrote:
Thanks Richard. This is exactly the answer I was looking for.
I'm now assuming that Asterisk 11 was using it's equivalent
bridge_simple but I was getting confused because the only bridge module I
saw
dialplan then you use the
POST /channels/{channelId}/continue
ARI command.
Richard
[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Channels+REST+API#Asterisk12ChannelsRESTAPI-continueInDialplan
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in the [logfiles ] line. It must be
[logfiles].
Richard
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In case it wasn't obvious in the DAHDI release announcement.
Richard
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/Collecting+Debug+Information
Richard
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On Fri, Aug 22, 2014 at 5:08 PM, Richard Mudgett rmudg...@digium.com
wrote:
On Fri, Aug 22, 2014 at 4:55 PM, Mitch Claborn mitch...@claborn.net
wrote:
Asterisk 12.5
I have a reproducible segfault using the MeetMe application. How do I
gather the necessary information (backtrace, core
On Fri, Aug 22, 2014 at 5:09 PM, Richard Mudgett rmudg...@digium.com
wrote:
On Fri, Aug 22, 2014 at 5:08 PM, Richard Mudgett rmudg...@digium.com
wrote:
On Fri, Aug 22, 2014 at 4:55 PM, Mitch Claborn mitch...@claborn.net
wrote:
Asterisk 12.5
I have a reproducible segfault using
and the agent
will be
logged out. The agent will then have to log back in after the current call
completes.
Why are you attempting to request an agent that has a device state
(Agent:agent_id) of busy anyway? That agent could be on another call or in
a
between call wrap-up time.
Richard
On Tue, Aug 12, 2014 at 11:24 AM, Deepak Rawat deepaksingh.ra...@gmail.com
wrote:
Thank you for the response Richard and Matthew! It's good to hear that you
are working on fixing the 5s delay. I was really puzzled by it and found
the idle time by trial and error. Is there any documentation
-zero on
'SIP/0015652CABE8_1-006c'
I checked func_db.c and the code is in place for DB_DELETE.
DB_DELETE is a function not a dialplan application.
Use:
same = n,Set(DB_DELETE(office/${DBKey})=)
or
same = n,Set(DELETED_VALUE=${DB_DELETE(office/${DBKey})})
Richard
know how can I solve this problem?
You might want to try bri_net_ptmp since bri_net is for point-to-point
operation
and you are not getting any response from the BT. The BT may be expecting
to operate in point-to-multi-point mode. Otherwise, you may have a cabling
problem.
Richard
On Fri, Aug 1, 2014 at 1:26 PM, ker...@tekno-soft.it wrote:
On Fri, 1 Aug 2014 12:39:18 -0500, Richard Mudgett wrote:
On Fri, Aug 1, 2014 at 12:03 PM, Roberto Fichera wrote:
snip
Does anyone know how can I solve this problem?
You might want to try bri_net_ptmp since bri_net is for point
On Tue, Jul 22, 2014 at 12:45 PM, Eric Wieling ewiel...@nyigc.com wrote:
Making LinkedID available in the dialplan would also be useful.
LinkedID is already available in the dialplan: CHANNEL(linkedid)
Richard
by chan_dahdi when libpri support is available and when
libpri version has the reverse charging feature.
Richard
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I'm interested in finding out what the source ip is of an invite in the
dialplan (Asterisk 11).
${CHANNEL(recvip)}
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not exist due to a configuration error, the
default parking lot is used.
Richard
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/display/AST/Variable+Inheritance
Richard
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On Mon, Jun 16, 2014 at 4:05 AM, babak bk1...@yahoo.com wrote:
Hi
I have done everything richard told to enable ECT .
below is my trace, anyone can help ?
-- DAHDI/i1/09123278669-4 answered DAHDI/i1/88050048-3
-- Native bridging DAHDI/i1/88050048-3 and DAHDI/i1/09123278669-4
PRI
I'm having the error as shown belowÂ
Connecting to 'wss://54.xxx.xxx.xxx:8080/ws' SIPml-api.js?svn=224:1
==stack event = starting SIPml-api.js?svn=224:1
__tsip_transport_ws_onerror SIPml-api.js?svn=224:1
__tsip_transport_ws_onclose SIPml-api.js?svn=224:1
==stack event = failed_to_start
Richard
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dialing the last digit. There is some
attempt to do this using line supervision by polarity reversal. However,
this is not normally supported by the telco.
Richard
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application so it doesn't know what you want to do.
Read the core show application dial documentation.
Richard
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On Fri, May 23, 2014 at 7:05 PM, Armen K armen...@hotmail.com wrote:
Hi everyone,
I was referred to this mailing list by Richard Mudgett regarding the
following thread on Issue Tracker as it's a feature request and not a bug:
https://issues.asterisk.org/jira/browse/PRI-170
We've got
/Pattern+Matching
for the rules of pattern matching.
Richard
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are generally intended for human interaction with Asterisk.
Richard
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On Tue, May 6, 2014 at 1:01 PM, Richard Kenner ken...@gnat.com wrote:
That is definitely a leak and the fix looks good.
Thanks.
That leak is most likely the one biting you.
It definitely is.
Committed the fix for this leak on Asterisk v12 branch in -r413454.
There is another leak
Committed the fix for this leak on Asterisk v12 branch in -r413452.
This leak also applied to Asterisk v11.
Thanks.
Is this for both the one in the talking callback or the one in
handle_cli_confbridge_kick or both (the fix is similar in both)?
--
On Wed, May 7, 2014 at 4:43 PM, Richard Kenner ken...@gnat.com wrote:
Committed the fix for this leak on Asterisk v12 branch in -r413452.
This leak also applied to Asterisk v11.
Thanks.
Is this for both the one in the talking callback or the one in
handle_cli_confbridge_kick or both
Really, I think we're pretty positive there's a ref leak (since
otherwise, the CBAnn channel would be long gone). If you can get a
ref debug log and the standard Asterisk DEBUG log showing the
problem, that would help a lot in finding out what is going on.
I think the bug is in
On Tue, May 6, 2014 at 5:45 AM, Richard Kenner ken...@gnat.com wrote:
Really, I think we're pretty positive there's a ref leak (since
otherwise, the CBAnn channel would be long gone). If you can get a
ref debug log and the standard Asterisk DEBUG log showing the
problem, that would help
That is definitely a leak and the fix looks good.
Thanks.
That leak is most likely the one biting you.
It definitely is.
There is another leak in handle_cli_confbridge_kick() if the
participant to kick is not in the conference.
Confirmed. I missed that one in my code reading. I just
It may show up in 'bridge show all' - but I'd actually expect it not
to show up there either.
Actually, it does. I have a screen full of bridges with 0 channels.
I just tried an experiment where all I have is
exten = 329,1,Answer(1000)
same = n,Confbridge(1234)
with absolutely nothing else
Please go ahead and open an issue and attach the refs log and the full DEBUG
log. That will allow us to understand what's occurring here.
I need to wait until I'm sure this isn't something I caused somehow,
so I need to first understand why I'm seeing this and nobody else is.
--
If the reference count on the bridge is off, you should see the conference
bridge 'hanging around' after the last participant has left.
And how would I be sure this is the case? I did core set debug 1 and
didn't see the debug line about destroying the conference, but it doesn't
show up in
Originate
Richard
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asterisk-users
Really, I think we're pretty positive there's a ref leak (since
otherwise, the CBAnn channel would be long gone). If you can get a
ref debug log and the standard Asterisk DEBUG log showing the
problem, that would help a lot in finding out what is going on.
That can't be done in the 12.2.0
After an upgrade to Asterisk 12, I'm collecting channels. When I enter
and then exit a conference room, I see:
-- CBAnn/207-067f;1 Playing 'confbridge-leave.slin' (language 'en')
-- Channel CBAnn/207-067f;2 joined 'softmix' base-bridge
5edb1920-3774-4ba3-8c4d-23e8fd04519c
--
On Tue, Apr 29, 2014 at 5:10 PM, Richard Kenner ken...@gnat.com wrote:
After an upgrade to Asterisk 12, I'm collecting channels. When I enter
and then exit a conference room, I see:
-- CBAnn/207-067f;1 Playing 'confbridge-leave.slin' (language
'en')
-- Channel CBAnn/207
The announcer channel joins/leaves the conference as it has sounds
to play. If the channel still hangs around after the conference is
destroyed then there is a problem.
There's a problem. ;-)
But thanks for pointing to how that's supposed to be handled.
--
If the channel still hangs around after the conference is destroyed
then there is a problem.
Am I missing something obvious: I'm looking in the confbridge_exec
function. I see a conference = NULL line, but no attempt to free
that structure, which is what I understand will destroy the playback
What distro are you building on?
CentOS 5.10.
Both have the libraries listed in install_prereq.
Indeed it has all but 2 or 3 of those libraries (none related to uuid), but
after running that script, it was still missing what it needed for uuid.
Unfortunately, there's no upgrade path from
e2fsprogs-devel is the package that provides uuid.h on centos 5
I tried that first and it didn't seem to. I'm pretty sure I needed
uuid-dce-devel.
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When I run ./configure, it aborts with:
checking for uuid_generate_random in -luuid... no
checking for uuid_generate_random in -le2fs-uuid... no
checking for uuid_generate_random... no
configure: error: *** uuid support not found (this typically means the uuid
development package is missing)
I think you need the libuuid and libuuid-devel packages.
yum list available was not showing any such package.
I installed a few other packages, including uuid-dce-devel and one of them
did the trick, but the install-prereq script wasn't good enough.
--
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Asterisk and CM this morning, and it works great providing that you allow
for anonymous calls.
-Original Message-
From: Haley,Scott A scott.ha...@edwardjones.com
Sent: Wednesday, April 23, 2014 9:36am
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