I have some questions about the T.38 faxing capability. I have been able to
successfully setup the inbound receive fax. However, I am having problems
tracking down the format of the outbound extensions.conf SendFAX command. I
have looked at the code and it looks like it only takes a single param
David Backeberg wrote:
> On Fri, Mar 13, 2009 at 2:30 PM, Marshall Henderson
> wrote:
>
>> On Fri, Mar 13, 2009 at 11:07 AM, David Backeberg
>> wrote:
>>
>>> Again, you'll find people arguing that their voip solution has as low
>>> of a failure rate as a hardware solution. I'm jealous. M
On Fri, Mar 13, 2009 at 2:30 PM, Marshall Henderson
wrote:
> On Fri, Mar 13, 2009 at 11:07 AM, David Backeberg
> wrote:
>> Again, you'll find people arguing that their voip solution has as low
>> of a failure rate as a hardware solution. I'm jealous. My voip fax
>> solution does not yet have tha
On 14/03/2009 10:29 a.m., Doug wrote:
> At 16:10 3/10/2009, Matt Riddell wrote:
> >On 7/03/2009 4:58 a.m., Klaus Darilion wrote:
> >> Hi!
> >>
> >> What are the typical ways to work around the 64 groups limit?
> >
> >What we actually do is store a pickup group with a caller id.
> >
I made another post, it is working, I have queue_log to mysql db and I have
a trigger that made the insert fail.
Sorry for the post!.
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent
Forget about this.
Is still working.
From: Sebastian [mailto:s...@adinet.com.uy]
Sent: viernes, 13 de marzo de 2009 10:05 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: TRANSFER EVENT ON QUEUE_LOG
Hi,
Anyone knows if TRANSFER event on queue_log is still
Sebastian wrote:
> Hi,
>
>
>
> Anyone knows if TRANSFER event on queue_log is still working on 1.6.0.6.
>
> I make an attended transfer (asterisk feature), and I cant see the event.
>
>
>
> Any idea? Should I submit a bug report?
If you do, be sure to headline it in all caps.
--
Alex Ba
Hi,
Anyone knows if TRANSFER event on queue_log is still working on 1.6.0.6.
I make an attended transfer (asterisk feature), and I cant see the event.
Any idea? Should I submit a bug report?
___
-- Bandwidth and Colocation Provided by http
On Friday 13 March 2009 17:41:42 JR Richardson wrote:
> I'm upgrading some PBX's from 1.2 to 1.4 and having a bit of trouble with
> converting the Realtime application to the REALTIME function. I have the
> method down and understand simplistically what is going on, at least enough
> to get my old
Hi All,
I'm upgrading some PBX's from 1.2 to 1.4 and having a bit of trouble with
converting the Realtime application to the REALTIME function. I have the
method down and understand simplistically what is going on, at least enough
to get my old 1.2 apps to run in 1.4 functions. I do not under
I have some questions about the T.38 faxing capability. I have been able to
successfully setup the inbound receive fax. However, I am having problems
tracking down the format of the outbound extensions.conf SendFAX command. I
have looked at the code and it looks like it only takes a single param
2009/3/13 Andrew Thomas
> You now need to compile and install SpanDSP-0.0.6pre3 at least (AGX has
> been changed).
Yes, you're right but I thought that to compile with AGX Asterisk Addon with
either 0.0.4 or 0.0.5 or 0.0.6 spandsp version, you should just just have to
edit the previously mentio
At 16:10 3/10/2009, Matt Riddell wrote:
>On 7/03/2009 4:58 a.m., Klaus Darilion wrote:
>> Hi!
>>
>> What are the typical ways to work around the 64 groups limit?
>
>What we actually do is store a pickup group with a caller id.
>
>So the AsteriskDB has ${DB/pickup/${CALLERID(num)}} and we se
On Fri, 13 Mar 2009, David Backeberg wrote:
[various snippage]
> Again, you'll find people arguing that their voip solution has as low
> of a failure rate as a hardware solution. I'm jealous. My voip fax
> solution does not yet have that low of a failure rate, but I'm
> hopefully getting closer
> Ringing() followed by
> Wait(1)
I made it
exten => echo,1,Ringing()
exten => echo,2,Wait(1)
exten => echo,3,Playback(abandon-all-hope)
exten => echo,4,Echo()
to no avail.
This looks like a client issue, though all of my clients fail. Which clients
are the most standards conforming?
Also, ma
Hi,
I know i doesn't make practical difference, but often it is the far
end that is atually buggy, not out end.
A lot of the work in spandsp to increase success rate is to do with
workarounds for issues in the remote machine,
Steve
On 3/13/09, Marshall Henderson wrote:
> On Fri, Mar 13, 2009
Marshall Henderson wrote:
> Hello everyone-
>
> I recently read the thread entitled "Faxing Success Rate on PRI" which
> dealt with Asterisk/HylaFax/IAXmodem. I'm successfully using this
> 'recipe' in a few instances on systems with only a couple of analog
> lines all the way up to a full PRI wo
Correct you are. Playback just plays a file back to the caller, Ringing
sends a ringing to over the channel (to the user).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lincoln
King-Cliby
Sent: Friday, March
My best guess at the root cause of the problem after looking at the packet
capture was that the phone was not happy seeing the call "connected" before any
of the intermediate states (trying, ringing, etc.) and Ringing() generated the
session progress (e.g. in addition to the in-band ringback it
On Fri, Mar 13, 2009 at 11:07 AM, David Backeberg wrote:
>
> On Fri, Mar 13, 2009 at 11:38 AM, Marshall Henderson
> wrote:
> > I recently read the thread entitled "Faxing Success Rate on PRI" which dealt
> > with Asterisk/HylaFax/IAXmodem. I'm successfully using this 'recipe' in a
> > few instanc
Not a better hack but perhaps more palatable to the listener
Playback(please-wait)
Wait(1)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lincoln
King-Cliby
Sent: Friday, March 13, 2009 1:16 PM
To: 'Asterisk Us
I had this issue with SIP Phones (Cisco 7961) to local voicemail; the issue was
resolved by adding a
Ringing() followed by
Wait(1)
before the VoicemailMain() in the dial plan... it seems like there should be a
better way, and I feel it's rather crude to force the user to listen to a
second
I'm trying to record all calls including calls that are part of a SLA.
Using both monitor and mixmonitor the recording appears to happen (that is,
asterisk logging shows it happening) however the file is never written to. It
doesn't seem to be possible to use the recording that is part of the m
--- On Fri, 3/13/09, Pascal Bruno wrote:
> I have the same situation. My scenario is weird:
Well, I've experienced the same symptoms but in a whole different context. It's
happening in my LAN (no firewalls, no NAT) and only with specific IP phones +
"early dial" + pedantic=yes.
http://bugs
David Ruggles wrote:
> It was with the patch applied, but after I restarted asterisk.
>
> Thanks,
>
Fix committed to Asterisk 1.4 in revision 181990.
Mark Michelson
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-u
I have the same situation. My scenario is weird:
I have a DID with IPkall that points to my asterisk server, and I have it
play a message with Playback() after about 20 seconds call drops and give
me the same message you get: "no reply to our critical packet"
BUT
I have a DID from Vitelity, an
On Friday, 13.03.2009 17:50:57 Danny Nicholas wrote:
> Next Step would be to check/update the firmware on your phones or router.
I don’t think the router is to blame, it does deliver all the packets. And
there are no hardware phones, only numerous software SIP clients.
Which (GNU/Linux) softwar
Tony Mountifield wrote:
> I have been asked by a potential customer whether we can connect an
> Asterisk box to an Ericsson MD110 that has an E1 port with ISDN-USR.
> They are unable or unwilling to upgrade their E1 port to QSIG.
>
> Has anyone here had experience of successfully making such a con
2009/3/12 Giorgio Incantalupo :
> Hi Gavin,
>
> if you can make and receive calls it works...do not worry if your line
> is shown as DOWN, some telco turns it off but it works without problem.
> Remember to ask your telco for the right signalling and set it the right
> way (PTP or PMP).
Thanks. It
2009/3/12 Paulo Santos :
> Gavin Henry wrote:
>> Hi All,
>>
>> We've got msidn configured:
>>
>> Port 1: TE-mode BRI S/T interface line (for phone lines)
>> -> Protocol: DSS1 (Euro ISDN)
>> -> childcnt: 2
>>
>
> I don't know if it depends on the card, but in my case I need to set the
>
On Fri, Mar 13, 2009 at 11:38 AM, Marshall Henderson
wrote:
> I recently read the thread entitled "Faxing Success Rate on PRI" which dealt
> with Asterisk/HylaFax/IAXmodem. I'm successfully using this 'recipe' in a
> few instances on systems with only a couple of analog lines all the way up
> to a
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roman Odaisky
Sent: Friday, March 13, 2009 10:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No reply to our crit
Hi,
thanks for the quick reply.
> 1. Do you have the incoming 1-2 holes in your firewall so the
> remote server can get it's reply back to *?
This was what I checked first. Both firewalls let everything through.
> 2. If #1 is ok, try putting an Answer command in front of your Dial
> Com
Hello everyone-
I recently read the thread entitled "Faxing Success Rate on PRI" which dealt
with Asterisk/HylaFax/IAXmodem. I'm successfully using this 'recipe' in a
few instances on systems with only a couple of analog lines all the way up
to a full PRI worth of Iaxmodems.
However, I'm finding
It was with the patch applied, but after I restarted asterisk.
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 da...@safedatausa.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roman Odaisky
Sent: Friday, March 13, 2009 9:57 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] No reply to our critical packet
Hi,
I've instal
The USA is on DST now, but Europe is not.
If you are in Europe, be aware that the VoIP Users Conference
conference will start one hour early today. In Paris, that translates
to GMT+1 or 5PM, in the UK 4PM.
Grand Central is about to be re-branded as Google Voice.
http://www.google.com/voice
Changes
David Ruggles wrote:
> I'm sorry, but it looks like it's working correctly now. I will update the
> bug if I am able to verify any problems.
>
> Thanks,
Heh, no reason to be sorry for it working :)
When you say it works now, was this with or without the patch applied?
Mark Michelson
___
Hi,
I’ve installed Asterisk for use as a SIP server. I can call people, but one
strange thing happens: if I call someone with a SIP account outside my server
(for example, sip:enum-echo-t...@sip.nemox.net) everything is fine, if I call
any Asterisk extension it also works, but the call gets dis
The ERC NPAs for Toll free are:
Toll-Free Special Use NPAs
* 800-NXX-
* 888-NXX-
* 877-NXX-
* 866-NXX-
* 855-NXX-
Those are the US toll free NPAs.
The "Easily Recognizable Codes" (ERC) NPAs have identical last two digits.
NOT 881 for instance, but like,
I have been asked by a potential customer whether we can connect an
Asterisk box to an Ericsson MD110 that has an E1 port with ISDN-USR.
They are unable or unwilling to upgrade their E1 port to QSIG.
Has anyone here had experience of successfully making such a connection?
I have found a couple of
I'm sorry, but it looks like it's working correctly now. I will update the
bug if I am able to verify any problems.
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 da...@safedatausa.com
-Original Message-
From: asterisk-users-boun...@
Umm, I don't think a called number sends any callerid info as there's
probably not even a protocol for that. Maybe you need to post a sample
CDR. The only thing I could think of is if you are calling an internal
extension and asterisk is posting the callerid you have defined for that
extensio
David Ruggles wrote:
> The patch doesn't work for me. Here's what I did:
>
> Changed to my asterisk-1.4.23.1 directory
> Executed the wget / patch command from the link you provided
>> make
>>> saw that res_features.so was recompiled
> Moved /usr/lib/asterisk/modules/res_features.so to res_feature
Kevin P. Fleming wrote:
> Mark Michelson wrote:
>
>> You can work around the bug, although it's not exactly optimal. What you can
>> do
>> is to modify your dialplan as follows:
>>
>> exten => 301,n,Set(DYNAMIC_FEATURES=monkey)
>
> Couldn't you just set _DYNAMIC_FEATURES here and have it get
>
Did these announcements stop coming on the [asterisk-announce] group? I
only seem to get sporadic announcements there.
Asterisk Development Team wrote:
> The Asterisk Development Team is pleased to announce the first release
> candidate of Asterisk 1.6.0.7, tagged as version 1.6.0.7-rc1. Release
Cary,
You also forgot 880, 881, 882 although I'm not sure I've ever even come
across one of those.
Cary Fitch wrote:
> In my previous reply, I may be wrong, "877" is probably a valid toll free
> NPA, add it in the mix.
>
> Cary Fitch
>
> -Original Message-
> From: asterisk-users-boun...@
Alex, What country is your call center located?
Thanks,
Al
On Fri, Mar 13, 2009 at 7:36 AM, Asterisk wrote:
> Dear All,
>
> I have a small call center in which I have to define least cost routing for
> outbound calls. For now I have always done this by routing numbers to
> different provider
You've had some good suggestions so far but honestly the brute force
method is not that difficult. I have been in the process of trying to
make my dialplan more concise (fewer statements) but haven't tried to do
anything about this one:
; Toll-Free
exten => _1800NXX,1,Macro(dial-avail-tf,$
I believe it's echo and/or jitter being measured when the call is
connected as I recall it being explained. This issue has existed for a
long time and I'm not sure there's much you can do about it except to
wait for a second before speaking when a call is connected. I think
maybe I have traine
> If there is NAT between the phone and * then that can be responsible.
>
> Also, Eyebeam (et al)'s ICE setting causes this.
STUN server settings also contribute on eyeBeam. You have to turn off ICE
and if you're not using a STUN server check the "Use a specified STUN
server" checkbox while leav
If it's anything like the UK, it won't make a difference... for example:
o2 mobile number ported to orange mobile...
On most providers you still pay the o2 rate.
three mobile ported to o2...
you still pay the three rate (which isn't so good since it's far more
expensive than o2).
Cheers
2009/3/13
2009/3/13 Andrew Thomas
> I think I understand what you mean now. The biggest difference between
> CLI and ANI is that ANI can't be blocked/withheld (like you can with CLI
> by using 141). It also uses different signalling. This is mainly used
> by law enforcement agencies to trace calls etc.
Mark Michelson wrote:
> You can work around the bug, although it's not exactly optimal. What you can
> do
> is to modify your dialplan as follows:
>
> exten => 301,n,Set(DYNAMIC_FEATURES=monkey)
Couldn't you just set _DYNAMIC_FEATURES here and have it get
automatically inherited to the outboun
Dear All,
I have a small call center in which I have to define least cost routing for
outbound calls. For now I have always done this by routing numbers to different
providers according to the number prefix.
However, a new law became effective now which allows people to switch between
provider
I "reverse" the inbound calls so they appear as outbound calls for agents,
all of our calls are managed by the dialer i've written and integrate
directly to our CRM, essentially asterisk is only providing the SIP/IAX
functionality to me everything else is done via php...
so...
inbound call comes i
I think I understand what you mean now. The biggest difference between
CLI and ANI is that ANI can't be blocked/withheld (like you can with CLI
by using 141). It also uses different signalling. This is mainly used
by law enforcement agencies to trace calls etc.
So, you want the number - regardl
On 13 Mar 2009, at 10:43, Julian Lyndon-Smith wrote:
> We already have CLI. I need ANI ;)
Why? Just out of interest.. If people withold CLI its usually for a
reason..
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asteris
David Quinton wrote:
> On Thu, 12 Mar 2009 10:21:06 +, Julian Lyndon-Smith
> wrote:
>
>
>> Has anyone in the UK got ANI to work on an inbound call ?
>>
>> Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30
>>
>
>
> AFAIK (and our E1 doesn't go to * box)
> a) you mean C
Hi Andrew
Andrew Thomas wrote:
> Please explain (in English) what you mean by ANI.
>
http://www.tech-faq.com/ani-automatic-number-identification.shtml
Julian
> Thanks
>
>
> -->> -Original Message-
> -->> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-
Yes, I had already tried that and it didn't work. Asterisk doesn't send any
RTP.
Regards,
Santi
On Fri, Mar 13, 2009 at 11:06 AM, Steve Howes wrote:
>
> On 13 Mar 2009, at 09:51, Santiago Gimeno wrote:
> > I know it looks like a problem with the phones but, is there a way
> > to configure ast
Please explain (in English) what you mean by ANI.
Thanks
-->> -Original Message-
-->> From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
-->> boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith
-->> Sent: 12 March 2009 10:21
-->> To: Asteris
On 13 Mar 2009, at 09:51, Santiago Gimeno wrote:
> I know it looks like a problem with the phones but, is there a way
> to configure asterisk so it sends RTP during silent periods?
Asterisk.conf
transmit_silence_during_record = yes
___
-- Bandwidth
That's at least 2 of us then Paul ;).
-->> -Original Message-
-->> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
-->> boun...@lists.digium.com] On Behalf Of Paul Hales
-->> Sent: 11 March 2009 00:04
-->> To: Asterisk Users Mailing List - Non-Commercial Discuss
Hello,
I have been experiencing audio problems when accessing the Voicemail
application using DECT phones and the g729 codec. The issue is that whereas
the vm-password is always played correctly by the DECT phone, the rest of
audio files, randomly, are played or not by the DECT phone. Everything w
You now need to compile and install SpanDSP-0.0.6pre3 at least (AGX has been
changed).
After you've done that - try AGX again.
HTH
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: 11 March 2009
On Thu, Mar 12, 2009 at 09:42:28PM +, Tarek Sawah wrote:
>
> Hello,
> a local prison contacted us regarding some calling card solution.
> they need 4 E1s to serve 120 rooms in that prison.
120 concurrent calls? (do you assume that most of those lines will be
busy most of the time?)
Normall
On Thu, Mar 12, 2009 at 4:22 PM, BJ Weschke wrote:
> nik600 wrote:
>> Hi to all.
>>
>> What can i do if a customer needs to log in the CDR all the dialpan
>> actions related to a call?
>> I mean, not only the lastapp e the lastdata but all the dialpan actions!
>>
>> I know that the actual CDR syst
I'm only half joking: what about parsing the "full" log looking for command
inviocations and channel IDs? this would be completely transparent, albeit
insane :)
l.
2009/3/12 nik600
> Hi to all.
>
> What can i do if a customer needs to log in the CDR all the dialpan
> actions related to a call?
You should look at the queue() command invocation.
Thanks
l.
2009/3/12 Darrin Henshaw
> Hello,
>
>
>
> We had an incident recently where a call was in queue for an extended
> period of time. We use queuemetrics for reporting, and it reports that the
> call was waiting for 20 minutes. The di
On Thu, 12 Mar 2009 21:42:28 +, Tarek Sawah
wrote:
>
>Hello,
>a local prison contacted us regarding some calling card solution.
>they need 4 E1s to serve 120 rooms in that prison.
If there's only one person per room, then I'm not sure that they need
*4* E1s if you think about it...
_
If there is NAT between the phone and * then that can be responsible.
Also, Eyebeam (et al)'s ICE setting causes this.
Steve
On 3/13/09, Mike Diehl wrote:
> Hi all,
>
> I've got a problem where many times, there is silence at the first 1-2
> seconds of a call. Then it clears up and it's crysta
On Thu, 12 Mar 2009, Tarek Sawah wrote:
>
> Hello,
> a local prison contacted us regarding some calling card solution.
> they need 4 E1s to serve 120 rooms in that prison.
> we are planning on using 4 servers to serve the calls and one for the database
> servers' specifications are:
> 2.8 Dual Cor
On Thu, 12 Mar 2009 10:21:06 +, Julian Lyndon-Smith
wrote:
>Has anyone in the UK got ANI to work on an inbound call ?
>
>Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30
AFAIK (and our E1 doesn't go to * box)
a) you mean CLI
b) you have to pay BT extra for "Calling Line I
sean darcy writes:
> The regular long distance is set up so users can but don't have to
> dial one. That's pretty easy, just one more exten statement. But it's
> a pain dealing with all the 8xx area codes that are toll free.
We try to "canonicalize" dialled numbers as soon as they enter the
syst
On Thu, Mar 12, 2009 at 09:53:48AM +0100, wrote:
> On 3/11/2009, "Hĺkan Källberg" wrote:
> >On Wed, Mar 11, 2009 at 04:16:43PM +0100, Christian Victor wrote:
> >> 2009/3/11 Hĺkan Källberg
> >> > Does anyone of you have Caller Presentation working in the other
> >> > direction?? My mv370 is work
Check your echo can settings.
On Fri, Mar 13, 2009 at 3:06 AM, Mike Diehl wrote:
> Hi all,
>
> I've got a problem where many times, there is silence at the first 1-2
> seconds of a call. Then it clears up and it's crystal clear. I've not
> put a sniffer on it, yet, but I suspect that the media
Hi all,
I've got a problem where many times, there is silence at the first 1-2
seconds of a call. Then it clears up and it's crystal clear. I've not
put a sniffer on it, yet, but I suspect that the media channel is still
being set up. The server shouldn't be too overloaded. Can anyone give
me
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