Hi,
I have a Digium TE410p T1 card and I've noticed that under asterisk
1.4.17/18 I have problems detecting DTMF in IVRs. I think I've
narrowed the problem down to some sort of interference between the
greeting that is playing and the DTMF tones. DTMF detection seems to
work very reliably when I am
Hi everyone.
I've tried RelaxDTMF and it didn't seem to help.
I can't use Read() or WaitExten() instead of background.
What I was doing was running a test using the following dialplan:
[custom-testdtmf2]
exten => s,1,Answer
exten => s,n,AGI(festival-script.pl|Enter test digits. Then press pound.)
Another note, after looking at the source code, it seems as though
WaitExten() and Read() use ast_waitfordigit() where Background uses
ast_waitstream().
Apparently these 2 functions must behave differently.
-- James
On Wed, Mar 12, 2008 at 9:45 AM, James Lamanna <[EMAIL PROTECTED]> wrote
waitfordigit() where
> Background uses
> ast_waitstream().
>
> Apparently these 2 functions must behave differently.
>
> -- James
>
> On Wed, Mar 12, 2008 at 9:45 AM, James Lamanna
> <[EMAIL PROTECTED]>
> wrote:
> > Hi everyone.
> > I'v
Hi,
Is it possible to get the XML config off of a Linksys SPA-941 or 942 phone?
I've tried http://[ip address]/admin/spacfg.xml however that file
doesn't appear to exist.
Thanks.
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Hi,
I've noticed that sometimes instead of getting a regular ring tone
when calling out on a Zap channel, I get this obnoxious loud noise
which forces me to hang up.
Is this a problem in the Zaptel driver? I seem to recall that ringback
tones are generated by zaptel when dialing out from a SIP phon
the ringback you get over a zap channel (be it
> analogue or digital) is generated by the remote end, /not/ Zaptel.
>
> The ringback you get over a SIP or IAX2 channel is often generated by
> either Asterisk or the SIP/IAX2 device you're calling from.
>
>
> James Lamanna wrote
Hi,
Is there any information that can be gathered from the logs about why
a SIP call was dropped/terminated without either side hanging up?
I've run asterisk pretty verbose and I guess I haven't seen anything
that pops out at me yet.
I'm trying to diagnose why some clients are getting dropped calls
Hi,
While I haven't personally used any of their equipment yet, Brix is
supposed to have good h/w and software for measuring a MOS score:
http://www.brixnet.com/products/BrixCall.shtml
http://www.voiptroubleshooter.com/basics/mosr.html
-- James
> Hello Fellow Users,
>
> I am looking for a way - u
Hi,
Since SetMusicOnHold() is being deprecated, how do we set the channel
musicclass from an AGI script?
Last time I checked you can't call dialplan functions from AGI.
Thanks.
-- James
--
_
-- Bandwidth and Colocation Provided
; On Sun, Oct 5, 2014 at 6:40 PM, James Lamanna wrote:
> > Hi,
> > Since SetMusicOnHold() is being deprecated, how do we set the channel
> > musicclass from an AGI script?
> > Last time I checked you can't call dialplan functions from AGI.
> >
>
> Actually, yo
Hi,
I have an Asterisk server that's been running now for around 2 days.
I've noticed that the resident memory seems to be very high for its current
call load:
PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND
18321 asterisk 20 0 8050m 5.2g 6968 S 13
Its up to 5.8G of resident memory with 28321 calls processed.
The OOM killer is going to kill this soon at this rate (8GB RAM machine).
This seems like a pretty serious problem.
It looks like I'll need to restart asterisk every night
On Fri, Nov 21, 2014 at 10:53 AM, James Lamanna
t; You can disables hypertreading and cut your ram usage to half of that.
>
> I can't see what hardware you are using but I think you need to check that
> the rule above fits your hardware.
>
> b.r.
> Freddi
>
>
>
>
>
>
> On Fri, Nov 21, 2014 at 10:53
ra.
>> So from start memory usage increases until it reaches 17.3 gb and then
>> stabilizes. at that level.
>> You can disables hypertreading and cut your ram usage to half of that.
>>
>> I can't see what hardware you are using but I think you need to check
>> that the
(Starting a new email topic for this specific issue)
Hi,
What is the maximum size of the frame.c cache in Asterisk 11 and why does
it constantly increase?
This is what I'm up to already:
$ asterisk -rx "memory show summary"
3667584471 bytes (3667366799 cache) in4846685 allocations in file
fr
On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan wrote:
> On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna wrote:
> > Also, how big does the cache in frame.c grow to?
> > I've recompiled with MALLOC_DEBUG on that server:
> >
> > asterisk -rx "memory show summa
On Tue, Nov 25, 2014 at 10:21 AM, James Lamanna wrote:
>
> On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan
> wrote:
>
>> On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna
>> wrote:
>> > Also, how big does the cache in frame.c grow to?
>> > I'
On Wed, Nov 26, 2014 at 3:20 PM, James Lamanna wrote:
>
> On Tue, Nov 25, 2014 at 10:21 AM, James Lamanna
> wrote:
>
>>
>> On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan
>> wrote:
>>
>>> On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna
>>>
Hi,
A couple of our users are reporting that intermittently, their
voicemails are unable to be heard because there is a
milliwatt-sounding tone recorded over the top of it.
Has anyone else encountered this issue?
I have put a recording of the voicemail up online for people to listen
to to see what
Using a frequency analyzer, the tone is composed of 1Khz multiples at
(1, 2, 3, and 4Khz).
Any ideas?
On Mon, Jun 30, 2008 at 2:46 PM, James Lamanna <[EMAIL PROTECTED]> wrote:
> Hi,
> A couple of our users are reporting that intermittently, their
> voicemails are unable to be hear
Here are more specifics that I forgot to include:
- Asterisk version is 1.4.18.
- Call was coming in over a PRI provided by PacWest through a Digium TE410P.
- ulaw codec.
Thanks.
-- James
-- Forwarded message --
From: James Lamanna <[EMAIL PROTECTED]>
Date: Tue, Jul 1, 2
Hi,
We have a few Aastra 480ci phones and we've noticed that in order to
get the phone to receive a call, qualify must be = no.
Apparently the Aastras do not respond to the qualify message (or
respond in a way Asterisk doesn't understand) and Asterisk thinks the
phone is unreachable.
However, this
Hi,
Is there a way to do a blind transfer within an asterisk dialplan (like '##')?
The reason I need this (I think) rather than a regular Goto() is that
I'm trying to do one-touch parking.
I can park a call using one-touch parking and then pick it up again,
however if I try to re-park the call, it
Hi,
Does anyone have some good examples of a Kamalio or OpenSips
configuration that integrates with Asterisk?
Essentially I want to use the SIP router as the UA, but still run all
the calls through Asterisk (for dialplan, etc..)
I've looked for examples on the project web sites, but I haven't foun
Hi,
We've implemented a 'page-all' function for some of our customers, and
we've noticed that
on occasion the page-all will cause asterisk to crash (safe_asterisk
then restarts it again).
The particular customer has about 20 phones, and also has 5 Linksys
932 to monitor the state of these extension
Hi,
I have a strange problem. At a site where there are 20+ phones, there
is one phone that cannot make outbound (to PSTN) calls.
Each call is dropped after 20s with "no response to our critical packet".
Calls to voicemail and internal extensions work fine.
I understand that everything points to a
hanks.
(Oh and please CC me, I'm reading in digest mode..)
-- James
On Fri, May 22, 2009 at 10:36 AM, James Lamanna wrote:
> Hi,
> I have a strange problem. At a site where there are 20+ phones, there
> is one phone that cannot make outbound (to PSTN) calls.
> Each call is droppe
Hi,
Can someone tell me what causes asterisk to "Auto-congest" a phone on
a SIP channel?
Is it just a lag issue to the phone or is there something else going on?
Thanks.
-- James
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Hi,
I'm having an issue where some phones behind a sonicwall are auto-congesting.
The status on "sip show peer" shows ping times anywhere from 80ms all
the way up to 1100ms.
PCs behind the same firewall have a ping time of about 30ms to the PBX itself.
Does anyone know if the sonicwall is insertin
ot; are disabled.
-- James
>On Wed, Oct 22, 2008 at 11:35 AM, James Lamanna <[EMAIL PROTECTED]> wrote:
>> Hi,
>> I'm having an issue where some phones behind a sonicwall are auto-congesting.
>> The status on "sip show peer" shows ping times anywhere from
Hi,
I've noticed that if I have a multi-line linksys (942 or 962) phone
with the same sip registration mapped to each line key, that if all
the lines are full the phone will accept another call. I would expect
the phone to respond with "busy" so the call would to directly to
voicemail.
Ha
phones with different numbers
of lines, I really do not want to do this manually for every extension
I have.
-- James
> 2008/12/5 James Lamanna <[EMAIL PROTECTED]>
>
>> Hi,
>>
>> I've noticed that if I have a multi-line linksys (942 or 962) phone
>> with th
Hi,
A couple of our customers are having issues with doing attended transfers.
What happens is Caller A receives a call, they transfer to Caller B,
tell Caller B who is calling, etc.. and then
hit the "Transfer" key again to transfer the call.
Caller A's side hangs up as expected, but the call is n
Hi,
I've been trying to use the FaxGateway application to send T.38 out
over Zaptel using asterisk but I don't seem to be having any luck.
I'm executing it in the dialplan like: FaxGateway(Zap/g0/[number])
Has anyone had any luck using this thing and can enlighten me on how
it's supposed to be use
Hi,
Here's part of the log that I see.
In this case I'm testing on a box that unfortunately doesn't have a
PRI connection.
I've so far tested with just voice calls so far, but as you can see,
FaxGateway can't even dial out to the SIP trunk properly.
Here's also what the dialplan looks like:
exten
Hi,
I'm trying to get T.38 passthrough to work to T38Modem and Hylafax so
I can terminate T.38 faxes from an ATA.
However I haven't had much luck. I've tried two supposedly T.38
capable ATAs, the Grandstream 206 and the 502.
The 286 seems to connect to T38Modem, but Hylafax doesn't get any fax data
Hi,
Using rfc2833, I constantly have the problem that if a user presses
digits reasonably fast,
I will see a bunch of DTMF end emulation of 'X' queued on SIP/.
This queuing screws up the DTMF because the digits never get sent!
Is there a way to prevent asterisk from queuing DTMF? Or at leas
Hi,
One of our customers has an issue with the callee not being able to hear them.
It seems to happen very frequently on one number in particular where
there are about 3 IVR menus to dial through
before getting to a live person. However, this does not happen on every call.
Running tcpdump on the RT
dge until the real one
> can take effect.
I'm not sure how that would help in this case.
The call is answered by the remote end and then
the caller can hear the audio of the IVR menus.
Or am I missing something here?
-- James
>
> -Original Message-
> From: asterisk-users-
Hi,
I had a Trixbox 1.4.18 that I "yum update"d to 1.4.22.1.
Now, I seem to have a huge problem with phones not staying registered
(registrations worked perfectly at 1.4.18).
I phone will register the first time I plug it in, and then once you
make a call and hangup (or sometimes even during the ca
xpires=60
> session-minse=120
> session-refresher=uac
> register => 104:xx...@yy.com/104
> defaultip=192.168.23.114
> mailbox=104
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf
e helpful.
-- James
>
> -Original Message-
> From: James Lamanna [mailto:jlama...@gmail.com]
> Sent: Wednesday, January 28, 2009 1:59 PM
> To: asterisk-users@lists.digium.com
> Cc: da...@debsinc.com
> Subject: Re: asterisk-users Digest, Vol 54, Issue 94
>
>> Da
[moving to asterisk-users by request]
On Tue, Jan 27, 2009 at 12:56 AM, John Todd wrote:
>
> On Jan 26, 2009, at 7:38 PM, James Lamanna wrote:
>
>>> On Jan 26, 2009, at 8:53 PM, James Lamanna wrote:
>>>
>>>> Hi,
>>>> Is it just me, or does DT
Hi,
I'm trying to run Asterisk 1.4.23.1 on a small ARM linux board (TS-7800).
Everything compiles fine, but on startup Asterisk always crashes while
loading chan_sip.
If chan_sip is removed, it starts up fine, but I really need SIP to work.
Any ideas?
Thanks.
-- James
__
Hi,
I have a couple of users who are having a peculiar problem.
On some outbound numbers where there is a deep IVR tree (3+
selections), and then a live person picks up,
the live person will be unable to hear them on the phone, but they can
hear the live person.
I've done packet traces and it appea
Hi,
I'm trying to use iaxmodem against a very old version of asterisk
(1.0.7 - its a debian sarge embedded system),
yet when asterisk gets a call from iaxmodem, it says that the "format
for the call is unknown".
Does anyone know if there is a minimum version of asterisk that is
compatible with iaxm
On Sat, Feb 7, 2009 at 1:19 PM, Lee Howard wrote:
> James Lamanna wrote:
>>
>> Hi,
>> I'm trying to use iaxmodem against a very old version of asterisk
>> (1.0.7 - its a debian sarge embedded system),
>> yet when asterisk gets a call from iaxmodem, it sa
bridge of IAX2/iaxmodem0-11201 and IAX2/iaxmodem1-1796
On Sat, Feb 7, 2009 at 1:44 PM, James Lamanna wrote:
> On Sat, Feb 7, 2009 at 1:19 PM, Lee Howard wrote:
>> James Lamanna wrote:
>>>
>>> Hi,
>>> I'm trying to use iaxmodem against a very old version
Hi,
Is it possible yet to restart a single Dahdi span in any version of
Asterisk? (instead of all of them)
Thanks.
-- James
--
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New to Asterisk? Join us for
Hi,
I have a bunch of different customers on an Asterisk Box (the PBX).
This Asterisk Box is behind another Asterisk box that provides a PSTN
connection.
Up to this point I've been using IAX between the 2 Asterisk boxes, but
I would like to use SIP instead.
After doing some testing I have the follo
Hi,
I work for a VoIP provider in Southern California. We are looking
for someone very knowledgeable in Asterisk/VoIP to help work on the following:
- Maintenance of current Asterisk servers, updating Asterisk, monitoring
load, and other sysadmin tasks
- Devise and implement scalability strategie
On Mon, Jul 2, 2012 at 12:13 AM, Olle E. Johansson wrote:
> No.
>
> This is probably because you are using phone numbers as names of devices with
> type=friend in sip.conf.
> That's generally a bad idea.
>
> The SIP channel matches an incoming call this way:
>
> 1. Take the From: user name and m
Hi,
I'm testing out a server with asterisk 1.8.15.0 on it.
I'm experiencing static occurring on almost 90% of calls on this particular
server.
All test phones are using SIP, and calls to/from PSTN servers are delivered
using IAX2.
I have other production servers running 1.4.x that do not have this
On Thu, Nov 8, 2012 at 8:47 AM, Richard Mudgett wrote:
> > I'm testing out a server with asterisk 1.8.15.0 on it.
> > I'm experiencing static occurring on almost 90% of calls on this
> > particular server.
> > All test phones are using SIP, and calls to/from PSTN servers are
> > delivered using I
Hi,
I have a PSTN Asterisk box that's connected to other dialplan PBXes through
IAX2.
Recently this box was upgraded to 1.4.44 with the latest DAHDI version.
I've noticed that if one of the dialplan PBXes calls Congestion(), the PRI
will return ISDN code 34 (as its supposed to do).
However, the is
/display/AST/Asterisk+Versions
>>
>> You might get more help or better behavior by updating to a newer more
>> current version of Asterisk, such as 1.8 which will be receiving bug fixes
>> into October 2014.
>>
>>
>> On Wed, Dec 19, 2012 at 3:47 PM, James Lama
Hi,
I've noticed on asterisk 1.8.18 I'm hearing the blip of '#' DTMF to end the
recording on the recording itself.
Is there an easy way to truncate the last 200ms of the recording or so to
eliminate this?
The DTMF is coming in through rfc2833 and not inband.
Thanks.
-- James
--
__
On Wed, Feb 20, 2013 at 10:49 AM, James Lamanna wrote:
> Hi,
> I've noticed on asterisk 1.8.18 I'm hearing the blip of '#' DTMF to end
> the recording on the recording itself.
> Is there an easy way to truncate the last 200ms of the recording or so to
> elim
Hi,
I have a problem with forwarding a voicemail and prepending a message to it.
If a user just forwards a voicemail, everything works fine.
However, if a user prepends a message to the voicemail when forwarding, the
voicemail that is forwarded only contains the prepended message and not the
origin
Hi,
I have a 1.8.22 Asterisk (Box A) connected to a 1.4.32 Asterisk box (Box B)
through SIP.
The 1.4.32 box is then connected to the PSTN through PRIs.
I've noticed there are occasions where I am seeing duplicated DTMF.
I've verified from the SIP trace from the phone that there is only a single
'3'
Hi,
I'm looking for some advice on how to solve DTMF issues.
I have 2 boxes, one which is the connection to the PSTN (PSTN) through
PRIs and SIP trunks, and a second (PBX) which has UAs registered to
it.
We have a customer that has an existing pbx that we trunk analog lines
to using a GXW-4008.
The
Hi,
I've been trying to fix NOTIFY replies (specifically keep-alives) in 1.4.42
(I can't upgrade to 1.8.x at the moment for various reasons).
I've noticed for user agents that have a VIA header with a different
port than the port the NOTIFY was sent from,
the NOTIFY reply will always be sent back
On Fri, Dec 30, 2011 at 6:02 AM, Kevin P. Fleming wrote:
> On 12/30/2011 04:07 AM, James Lamanna wrote:
>>
>> Hi,
>> I've been trying to fix NOTIFY replies (specifically keep-alives) in
>> 1.4.42
>> (I can't upgrade to 1.8.x at the moment for various r
On Fri, Dec 30, 2011 at 8:35 AM, James Lamanna wrote:
> On Fri, Dec 30, 2011 at 6:02 AM, Kevin P. Fleming
> wrote:
>> On 12/30/2011 04:07 AM, James Lamanna wrote:
>>>
>>> Hi,
>>> I've been trying to fix NOTIFY replies (specifically keep-alives) in
>
On Fri, Dec 30, 2011 at 11:55 AM, Kevin P. Fleming wrote:
> On 12/30/2011 12:29 PM, James Lamanna wrote:
>>
>> On Fri, Dec 30, 2011 at 8:35 AM, James Lamanna wrote:
>>>
>>> On Fri, Dec 30, 2011 at 6:02 AM, Kevin P. Fleming
>>> wrote:
>>>
See the following SIP trace.
Where in the world does Asterisk get port 1025 to respond to?
This is asterisk 1.6.x.
Thanks.
-- James
<--- SIP read from zzz.zzz.zzz.44:9363 --->
NOTIFY sip:pbx1.mydomain.com SIP/2.0^M
Via: SIP/2.0/UDP 192.168.1.140:9363;branch=z9hG4bK-b9a860d3^M
From: "xxx-xxx-xxx
Hi Jeff,
On Thu, Jan 6, 2011 at 11:28 AM, Jeff LaCoursiere wrote:
>
>
> On Wed, 5 Jan 2011, James Lamanna wrote:
>
>> See the following SIP trace.
>> Where in the world does Asterisk get port 1025 to respond to?
>> This is asterisk 1.6.x.
>>
>
>
HI Ye,
On Mon, Jan 10, 2011 at 10:04 AM, Ye Liu wrote:
> Hi folks,
>
> I'm currently running a modified version of Asterisk 1.6.1.1, I
> observed an unexpected behavior of my system today:
>
> 1. SIP device A successfully registered extension 100;
> 2. SIP device B tried to register extension 100
Hi Jonas,
On Thu, Jan 13, 2011 at 8:19 AM, Jonas Kellens wrote:
> Hello,
>
> can /var/log/messages/queue_log be saved in a MySQL database ??
>
> So it would be easier to work with...
I don't think Asterisk has this support built-in...maybe 1.8 does?
However, what I do to manage queue_log is I ha
Hi Duncan,
On Wed, Jan 12, 2011 at 10:13 AM, Duncan Turnbull wrote:
> Hi Thorsten
>
> Thanks very much, at this point my preference is rfc2833 but I will try some
> other options.
>
> The system is generating audible tones (that I can hear), although I think
> the audio is generated by the last
Hi,
I noticed that Dahdi starting producing these error messages:
ERROR[29250] chan_dahdi.c: No more room in scheduler
ERROR[29250] chan_dahdi.c: Asked to delete sched id -1???
during which time I could not send any calls or receive calls on at
least one of my Dahdi spans.
The only way to clear t
Hi,
I've noticed that my MeetMe install seems to think chan_dahdi is missing:
app_meetme.c: No DAHDI channel available for conference, user
introduction disabled (is chan_dahdi loaded?)
However, it definitely is since I have 3 PRIs functioning normally :)
Is there anything I should check before I
Hi,
Has terminating T.38 to PSTN found its way into the asterisk 1.6 mainline yet?
I remember seeing an app_gateway floating around at some point a while
ago, but I never had any luck with it.
Thanks.
-- James
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Hi, I have a PRI problem where it appears that my system is not
responding to SETUP messages on a channel.
It seems to be retransmitting a significant number of RELEASE messages
to clear a call that is most likely
to be long gone.
This causes a huge issue because I get a bunch of hangup cause 102s
On Fri, Feb 12, 2010 at 12:54 PM, James Lamanna wrote:
> Hi, I have a PRI problem where it appears that my system is not
> responding to SETUP messages on a channel.
> It seems to be retransmitting a significant number of RELEASE messages
> to clear a call that is most likely
>
Hi,
I have a case where SIP channels will not be destroyed, resulting in
further calls to ChanIsAvail() to fail.
The process (I believe) to replicate this is the following:
- Make a call to another SIP phone that is an "intercom" call (Auto-Answer)
- For whatever reason, the phone happens to go UN
Hi,
I have a serious problem with Asterisk 1.4.18.
Every so often, usually after Asterisk has been running for a few days
consistently, phones start dropping registrations.
However, when this happens, doing a "sip show peer" on those
extensions shows them as "OK".
Therefore, I have no way to tell t
Oliver wrote:
> How many phones are concerned ?
The box currently has about 380 active phone registrations.
Thanks.
Please CC me directly as well because I'm on digest mode.
-- James
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Hi,
I remember seeing a T38 Gateway application for Asterisk 1.6 floating
around, but I can't seem to find it again.
Does anyone have any pointers to it? I really want to be able to send
an incoming T38 connection directly to the PSTN.
Thanks.
-- James
___
> On Fri, Jun 26, 2009 at 11:10 AM, James Lamanna wrote:
>> Hi,
>> I remember seeing a T38 Gateway application for Asterisk 1.6 floating
>> around, but I can't seem to find it again.
>> Does anyone have any pointers to it? I really want to be able to send
>>
>On Fri, Jun 26, 2009 at 1:31 PM, James Lamanna wrote:
>> The use case is that a customer has a fax machine attached to an ATA.
>> The ATA sends T38 to Asterisk over SIP, then I need to forward that out
>> the PSTN.
> Got it. I'm saying why not skip the ATA and aste
>On Fri, Jun 26, 2009 at 11:10 AM, James Lamanna wrote:
> Hi,
> I remember seeing a T38 Gateway application for Asterisk 1.6 floating
> around, but I can't seem to find it again.
> Does anyone have any pointers to it? I really want to be able to send
> an incoming T38 con
On Thu, Jun 4, 2009 at 11:08 AM, James Lamanna wrote:
> Hi,
> I have a serious problem with Asterisk 1.4.18.
> Every so often, usually after Asterisk has been running for a few days
> consistently, phones start dropping registrations.
> However, when this happens, doing a "sip
isk, presumably because there are stuck calls or something. Very
annoying as well.
-- James
>>>On Thu, Jun 4, 2009 at 11:08 AM, James Lamanna wrote:
>>> Hi,
>>> I have a serious problem with Asterisk 1.4.18.
>>> Every so often, usually after Asterisk has been r
Hi,
Some users have been reporting a peculiar problem.
The are having an issue when they dial out to some multi-level IVRs
where you make 2 or 3 touchtone choices and then are connected to a
live operator.
When the live operator connects the operator cannot hear them or
sometimes it results in dead
Does Asterisk 1.6 fully support RFC4235?
Or is it the same "implementation" as 1.4?
Thanks.
-- James
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.as
Hi,
I'm coming up with ideas about building a cluster of asterisk servers,
and am exploring the virtualization option.
I'm curious to know some real-world data about how many extensions a
VMWare install on good hardware could support.
I've seen stories about how the hypervisor timeslicing can wreak
>On Fri, Aug 7, 2009 at 11:47 AM, James Lamanna wrote:
>> Hi,
>> I'm coming up with ideas about building a cluster of asterisk servers,
>> and am exploring the virtualization option.
>> I'm curious to know some real-world data about how many extensions a
>
.
Please email me directly for more details or any questions about the position.
Thanks.
James Lamanna
Warp2Biz, Inc.
___
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Hi,
I'm trying to configure 2 parallel sip trunks between 2 boxes.
However I seem to have the problem that when making a call from Box 2
to Box 1, it sometimes
says authentication failed because it is using the username of the other trunk.
Here's my configuration:
Box 1:
[dp-dp2]
type=peer
userna
Hi,
I have two asterisk boxes A&B connected together via IAX.
Phones register to Asterisk box A, and Asterisk box B is the PSTN connection.
When dialing a number from a phone registered to A that DAHDI returns as BUSY,
the Busy(20) application returns immediately instead of playing the busy tone.
T
Hi,
Does anyone have any good empirical data suggesting what the maximum
number of PRI calls (incoming and outgoing)
without hardware echo cancellation can be handled on a single box is?
I have a TE410P T1 (1st gen) card and I'm seeing interesting errors of
D-Channels going down and then coming bac
Zeeshan A Zakaria wrote:
>On Wed, Mar 24, 2010 at 5:42 PM, James Lamanna wrote:
[snip]
>>
>> The specs of the machine are, Dual Xeon 2.80Ghz (both single core but w/HT)
>> 4GB memory.
>> Running asterisk 1.4.26.3 (32-bit)
>> with libpri-1.4.7 and zaptel-1.4.12.
Hi,
I have about 10 Cisco 7960s behind a PIX 506E (IOS v6.3) firewall.
After some period of time, asterisk says that some of them are
unreachable, and the phones lose their registration.
The only way to make the phones recover is to clear the NAT
translation tables for the phones on the PIX (clear
the phone is idle, so this way Asterisk will make sure to always have
> communication going trhough that connection so your NAT/firewall won't just
> close it.
Sorry, should have mentioned that all these phones have qualify=yes
and nat=yes in sip.conf.
Thanks.
-- James
> On Sat, Mar 27, 2010
On Sun, Mar 28, 2010 at 8:28 PM, Steve Edwards
wrote:
> On Sun, 28 Mar 2010, Joseph Begumisa wrote:
>
>> Can anyone recommend a 24 fxs port voip gateway that has worked well with
>> asterisk? I have a couple of analog handsets that I want to hookup to my
>> asterisk server? Any tested and tried
Hi,
I'm trying to figure out the cause of a soft lockup I experienced:
Mar 29 09:38:24 pstn1 kernel: BUG: soft lockup - CPU#0 stuck for 10s!
[asterisk:32029]
Mar 29 09:38:24 pstn1 kernel: Pid: 32029, comm: asterisk
Mar 29 09:38:24 pstn1 kernel: EIP: 0060:[] CPU: 0
Mar 29 09:38:24 pstn1
ng. It is actually a Dell SC1425 - Dual, dual-core Xeon Processors.
I'm hopefully going to be able to stress test this machine to see if I
can make it panic again with the PRI card IRQ isolated to CPU0. If so,
I'll see if it does the same thing on the other cores...
-- James
> --
>
On Mon, Mar 29, 2010 at 9:23 PM, James Lamanna wrote:
> On Mon, Mar 29, 2010 at 8:38 PM, Matt Watson wrote:
>> Dell server by any chance?
>> I have a similar problem with a TE220B in a Dell 1950 III server - i've seen
>> several other people having issues with digiu
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