I suppose D/[0-9#*](N) is the thing it does not like. You should omit
D/ in the beginning and try then.
e
Peter Zeltins wrote:
I'm trying to link up Cisco MGCP-enabled router (residential gateway) with
Asterisk, and it looks like some sort of protocol mismatch, could it be MGCP
0.1 vs 1.0?
AudioCodes has one 24 port fxs sip interface, i have one 8 port fxs with
SIP up and running with *
Michael Kane wrote:
The Cisco 242x (20 or 21), has a 24 port analog interface that supports 16
FXS and 8 FXO. I've delpoyed hundreds of these IAD's signaling with MGCP.
Not sure if it supports
Peter Zeltins wrote:
Did you check debug on cisco - I had this problem with AUEP packet on
ata-186 , cisco generate 510s error when you ask for unsupported
packages ...
try to edit chan_mgcp.c to set MGCP Version to 0.1
I'm trying to link up Cisco MGCP-enabled router (residential gateway)
What'd this device set ya back? Have a url?
-d
At 11:45 PM 6/30/2003 -0700, you wrote:
AudioCodes has one 24 port fxs sip interface, i have one 8 port fxs with
SIP up and running with *
Michael Kane wrote:
The Cisco 242x (20 or 21), has a 24 port analog interface that supports 16
FXS and 8
and another thing: AFAIK aaln/0@ is not permitted by MGCP standards.
Endpoint numbers MUST start from 1.
ekke
Peter Zeltins wrote:
I'm trying to link up Cisco MGCP-enabled router (residential gateway) with
Asterisk, and it looks like some sort of protocol mismatch, could it be MGCP
0.1 vs
I thought that SIP was a voice ONLY specification and that the reasoning behind the
development of SIP was to do purely voice to avoid what has happened in H.323, the
H.323 protocol specification grew to try and be everything to everyone (Voice, Video
and Data Sharing) and so became very
See README.variables or similar in the root of the asterisk source. This
should answer your questions on how to quote the various parameters needed
for sql and other functions.
Regards,
Adam
Please help.
this is generally a problem with arguments that contain ,
or (, i
think.
Hi Adam,
i think the real problem is the ,.
This will be allways replaced through |.
Regrads,
Thomas.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Adam
Goryachev
Gesendet: Dienstag, 1. Juli 2003 10:12
An: [EMAIL PROTECTED]
Betreff: RE:
I had a call today where there were several remote participants using a
speakerphone. They sounded quiet to me. Every time I spoke I got noise at
my end but the respondents never complained of any problems hearing me.
Iain
___
Asterisk-Users
Hello Torbjorn...
Jag borjar med att forwards dej nagra svar som jag fick av mailinglist.
Detta aer kostnader foer en PBX med dator och fyra analog phones + PBX +
external lines and it summons to 305$ + 100$ + PC phones kostnad ...
Alle 18:51, domenica 29 giugno 2003, John Todd ha scritto:
Hello,
We are using asterisk 0.4.0 on debian sid with Cisco 7960 and ATA186
phones.
All sip entries have:
callgroup=1
pickupgroup=1
However I am unable to remotely pickup a ringing phone using *8#. I get
fast busy tone. Is there some flag to add in extensions.conf ?
Thanks in advance,
the exten is *8 only.
didn't know if 0.4.0 has already that features on sip .
It's better to use latest cvs version.
Matteo.
Il mar, 2003-07-01 alle 11:52, Louis-David Mitterrand ha scritto:
Hello,
We are using asterisk 0.4.0 on debian sid with Cisco 7960 and ATA186
phones.
All sip
Hello,
I'm trying to do something really easy : transfer a PSTN call to a H323
client. This works great. Now I'm trying to use the SetMusicOnHold
function. I din't find any doc about it, I've just seen some mails in
the list archive, but it still doesn't work.
That's my extension.conf :
Well, I suposse is a very basic question but,,,for
what is used: callgroup=1 and pickupgroup=1 ?
thanks!
c.mayor
--- Louis-David Mitterrand [EMAIL PROTECTED]
escribió:
Hello,
We are using asterisk 0.4.0 on debian sid with Cisco
7960 and ATA186
phones.
All sip entries have:
Just want to know if this feature was implemented.
Also, how do I do a supervised transfer with sip phones?
Tan
- Original Message -
From: Mark Spencer [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 13, 2003 5:26 PM
Subject: Re: [Asterisk-Users] Sip call pickup ?
Hello all,
I got the following error compiling h323 support in the latest cvs. Below
the error is a diff to the file that I got to make it work. I took an
example out of sip as far as the syntax for ast_rtp_new. Not sure if it is
correct or not, but it seems to work. Please correct me if I am
Seems like windows messenger is using it for video comm. and file/session
sharing too. And of course for messaging.
- Original Message -
From: WipeOut . [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, July 01, 2003 9:26 AM
Subject: RE: [Asterisk-Users] * Video changes
I thought
Hi folks,
There was a bug with the GotoIfTime
built-in command, under certain circumstances a variable contained garbage,
screwing up correct time identification.
Im submitting now a patch to Mark so this can be
fixed.
PauloHM
To all who need more queue functionality,
We are contracting Digium to enhance the queue app for our call center needs.
Please read the following email conversation and give your ideas. Unless a glaring
omission is found in my specification we will have them start tomorrow (Wednesday). I
There are my logs :
Asterisk Ready.
*CLI -- Starting simple switch on 'Zap/1-1'
-- Executing SetMusicOnHold(Zap/1-1, default) in new stack
-- Executing ResponseTimeout(Zap/1-1, 20) in new stack
-- Set Response Timeout to 20
-- Executing Dial(Zap/1-1, OH323/192.168.1.215) in
You need to look at show application meetme in the asterisk CLI
but for it to work you need to have some kind of zaptel hardware or
emulate it with zttdummy (but for that you need to have usb-uhci like USB
controller)
and then
exten = 1000,1,Meetme,1000
Martin
On Tue, 1 Jul 2003, Serge
To pick up a call that rings someone elses phone that is in the callgroup
as your pickupgroup.
Martin
On Tue, 1 Jul 2003, carlos del mayor wrote:
Well, I suposse is a very basic question but,,,for
what is used: callgroup=1 and pickupgroup=1 ?
thanks!
c.mayor
--- Louis-David Mitterrand
Hi,
if you dont have usb-uhci you can also use your realtime clock
to generate zaptel timing. Make sure you dont have rtc support
compiled into your kernel and grab zaprtc from:
http://www.junghanns.net/asterisk
regards
kapejod
--
Klaus-Peter Junghanns
CEO,CTO
Junghanns.NET GmbH
Breite
Mark,
My experience with Siemens is that they are very expensive. Both units
have a name in the industry. Your solution is different. Your
competition will beat the hell out of that fact your different, don't have
backing, blah, blah, blah. Now you need to sell the concept that there are
Fabrice Tereszkiewicz wrote:
There are my logs :
Asterisk Ready.
*CLI -- Starting simple switch on 'Zap/1-1'
-- Executing SetMusicOnHold(Zap/1-1, default) in new stack
-- Executing ResponseTimeout(Zap/1-1, 20) in new stack
-- Set Response Timeout to 20
-- Executing
Will try to change to this:
Agent picks up phone and dials extension to 'login app':
exten = 750,1,QueueLogin(QueueName, wrap-up-time)
This would allow for quick agents to log into a queue for faster
processing and allow slower processing for slow agents. An agent would
simply log out
Have you considered how statistics for things like abandon rate will be
kept? How about service level stats, i.e. how many calls were abandoned
in the first 10s, 20s, 30s, etc. Also, most call center managers want
information on average hold time and longest call on hold for several
time
Hi,
I'm trying to setup Asterisk to allow users to dial out to the PSTN using a
remote box supporting h.323. I'm using chan_h323.so, and I'm able to make
outbound calls to a client like netmeeting with a line like this:
exten = 242,1,Dial(h323/xxx.xxx.xxx.xxx)
And I'm able to receive incoming
Same prob here. 15 SIP phones only get eco when going to the PSTN...
if you find something let me know
Dave
[EMAIL PROTECTED] 7/1/2003 8:53:13 AM
Hello,
I can't have asterisk working without echo when I place a call from IP
phone (SIP or H323) to a PSTN Phone. The called number as no
On Tuesday 01 July 2003 09:08 am, Paulo Mannheimer wrote:
Hi folks,
There was a bug with the GotoIfTime built-in command, under certain
circumstances a variable contained garbage, screwing up correct
time identification.
I'm submitting now a patch to Mark so this can be fixed.
What exactly
On 2003-07-01 at 16:00, Tamas Levente ([EMAIL PROTECTED]) wrote:
Seems like windows messenger is using it for video comm
Likewise, I notice that Apple's new iChat AV requires that port 5060 is
open, so it looks like they too are using SIP.
___
There was one option I tried to add, and failed at :(
While a call is in the queue, every XXX seconds play a message.
All agents are still helping other callers, please continue to hold,
or press *1 to leave a message.
Also, tell the person their position in the queue. You are currently
caller
Sure, here it goes.
As you may notice, a local instance of the variable ast_include is used
in function pbx_builtin_gotoiftime
As the local variable is not initialized to zero, its minmask
bitfields contain garbage, thus sometimes yielding true for unallowed
times.
BTW, nice work on the
exten = _91XX,1,Dial(H323/${EXTEN:[EMAIL PROTECTED])
${EXTEN:1} will grab all the digits you sent in 91XX and the :1, in
${EXTEN:1}, tells it to drop the first digit.
Michael
I'm trying to setup Asterisk to allow users to dial out to the PSTN using a
remote box supporting
Hello -
I'm looking for a cross-section of your experiences for large-scale
voicemail system deployment using Asterisk, where large scale means
500 active users or more with PRI channel access. I have a situation
that may call for an extremely large stand-alone voicemail
installation, and I
Will try to change to this:
Agent picks up phone and dials extension to 'login app':
exten = 750,1,QueueLogin(QueueName, wrap-up-time)
This would allow for quick agents to log into a queue for faster
processing and allow slower processing for slow agents. An agent would
simply log out if
We had planned to use the CDR_MySQL database to retrieve near-real-time
data from the pbx. I believe abandoned calls can be traced down in the
CDR database. Reporting will be done with Crystal Reports. I am creating
a VB app for the management interface for Asterisk that I will be
Hey Has anybody dealt with any of these products?
They Seem to be at a good price point. but I've never used one so I was
hopeing that sombody here Might have some insight. Their configurations are
as follows.
ACOM200B: 2 Port FXS
ACOM200D: 2 Port FXO
ACOM200E: 1 FXO and 1 FXS
This looks to me
On Tuesday 01 July 2003 12:13 pm, Paulo Mannheimer wrote:
Sure, here it goes.
As you may notice, a local instance of the variable ast_include is
used in function pbx_builtin_gotoiftime
As the local variable is not initialized to zero, its minmask
bitfields contain garbage, thus sometimes
Excellent idea! I will see if we can do it. Thanks.
Jim Friedeck
-
TC wrote:
Will try to change to this:
Agent picks up phone and dials extension to 'login app':
exten = 750,1,QueueLogin(QueueName, wrap-up-time)
This would allow for
Does anyone know of an Asterisk function that will report how many calls
are on hold in a queue?
I gave a small patch to mark that is in cvs that returns a count value when
the manager events
for join Q leave Q are fired
___
Asterisk-Users mailing list
Could you provide details of which sip phones you are using. For instance,
the SNOM 200 has echo problems on firmware ver 1.16b. Upgrading to 1.16k
resolves most of the echo issue.
Tan (telappliant.com)
- Original Message -
From: Dave Packham [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Justin Eckhouse wrote:
exten = 244,1,Dial(h323/xxx.xxx.xxx.xxx/PSTN-NUMBER-HERE)
This is bad... if you use this kind of exten line PSTN-NUMBER-HERE will
be the H.323ID Asterisk will use to make the call.
exten = 244,1,Dial(h323/[EMAIL PROTECTED]) is the proper
format.
Jeremy McNamara
I think the key point to the queue application was hit upon earlier:
statistics. About one-half of the problem with queues is getting the
functionality into Asterisk; the rest is reports.
What you measure, you manage. While I have not worked with getting
statistics out of the queue
Hello All,
Couple of quick (hopefully) questions.
1. I noticed in the latest h.323 cvs log that callerid is now supported.
Is there any special configuration needed to get this to work. I have tried
callerid= in h323.conf to no avail. Calls from a h.323 device show
callerid as the user
Hi,
Anyone succeed using InternetPhoneWizard as a console device in Asterisk?
I can only place some SIP calls from the iConnect Phone application, but
will be great to be able to use it as a Console phone directly on the
Asterisk box.
Thanks,
Dan
___
Steven,
I tried the following:
[Conferences]
exten = ,1,MeetMeCount(|var)
exten = ,2,SayNumber(${var})
exten = ,3,Meetme()
but I get the following error:
NOTICE[48152]: File pbx.c, Line 900 (pbx_substitute_variables_temp): Wrong
use of LEN(VARIABLE)
It still let's me join
I'm uncertain why you're not able to get SIP working for your user
agents (SIP clients.) With Cisco equipment, as an example, it works
quite well and almost every 79xx or ATA-186 I have is behind a NAT,
and this configuration is duplicated across a dozen or more systems
now running behind
Should wrap up time be something associated with a queue, or with an
agent?
Mark
On Tue, 1 Jul 2003, Jim Friedeck wrote:
Will try to change to this:
Agent picks up phone and dials extension to 'login app':
exten = 750,1,QueueLogin(QueueName, wrap-up-time)
This would allow for
The meetmecount app is supposed to tell you the number of participants in
a certain conf number. However it does not create the var variable.
The error about wrong use of LEN( was do to the fact that your var
variable does not exist and it was a bug. It's fixed now.
Martin
On Tue, 1 Jul 2003
First, why did you set the from address as my email address?
Secondly, did you apply the patch? So far as I know, Mark has not put my
patch in CVS yet.
Steven
On Tue, 2003-07-01 at 14:40, [EMAIL PROTECTED] wrote:
Steven,
I tried the following:
[Conferences]
exten =
What configuration of hardware/software are you running. I just checked
picking up with *8/transfer on zaptel/SIP and it works on our Digium PBX.
I placed a call from SIP to Zap, picked it up with Zap (*8) and transfered
to Zap
and also
place a call from Zap to Zap, picked it up with SIP-Snom200
Hello, NAT/Firewall is truelya problem in the
ITSP arena. Thereisone solutionIknow of that works
wellas an integrated DHCP/NAT/Firewall into a SIP aware
firewall. Check out www.intertex.se and look at the IXX66 products. They even have a
device that integrates DSL/NAT/Firewall. Or, one can
First, why did you set the from address as my email address?
I just replied via my webmail application since I was on a different
computer. It apparently doesn't parse the From field properly.
Secondly, did you apply the patch? So far as I know, Mark has not put my
patch in CVS yet.
I thought
Agent would be more flexible. Actually I think we are logging in the
device, right? I had a discussion with TC on the phone and we were
trying to figure out who was logging in. Will it be an agent or a
device? The ability to enter a destination phone number leads me to
believe the agent is
I think most of that information can be ascertained from the CDR
database through deduction. Ideally it would be available through the
management interface in realtime. Anyone feel like writing it? I don't
have the time to train myself to be a Jedi-Guru Asterisk programmer and
our budget is
Mark,
How hard would it be to write a simple app to cancel wrap-up via an
extension? Like dialing *99 to short- out the remaining wrap-up time?
Jim Friedeck
---
Mark Spencer wrote:
Should wrap up time be something associated with a queue, or with an
agent?
Mark
On
On Tue, 2003-07-01 at 15:39, [EMAIL PROTECTED] wrote:
First, why did you set the from address as my email address?
I just replied via my webmail application since I was on a different
computer. It apparently doesn't parse the From field properly.
You might want to get your administrator,
This might fit in with something I've worked on a bit but haven't had time
to complete yet. Basically an in memory CDR modification. So the CDRs
would get logged to a linked list and then try all available backends. If
a backend returns an error condition then the CDR will be retried again
Could probably make '#' terminate wrapup time immediately or something.
Mark
On Tue, 1 Jul 2003, Jim Friedeck wrote:
Mark,
How hard would it be to write a simple app to cancel wrap-up via an
extension? Like dialing *99 to short- out the remaining wrap-up time?
Jim Friedeck
I am unclear on the definition of backend. Do you mean each event would
be stored in Asterisk until its reporting is guaranteed to the csv and
MySQL systems? I am unfamiliar with the internals of CDR. Is there a
list of required 'backends' that CDR gets written to? If there is then I
can see
Would it be more flexible to approach this differently, with a dtmf to
indicate that the agent is done with wrap up?
So they get off a call and can wrap up the call for as long as
necessary, and then hit * or something that marks them as available
again rather than working against a timer to get a
Could you give some details about setting up a stun server?
I'm doing some tests, and were successful using snom + stund
from vovida . But I got a no-go with budgetones
(that needs stund on a standard port that's 3478).
When my snom contacts the stund server, I get a lot
of info about the
John,
Thanks for the detailed guide.
As you mentioned, the situation where two ATAs behind NAT want to establish
a direct connection is not resolved yet. In that case, the canreinvite would
have to be set to no and some other solution outside of * would have to be
used to traverse the NAT. Have
Sorry, I still don't know what you're talking about.
Clients behind NAT can talk to Asterisk without difficulty, and I use
that functionality all the time. If that is not the case for you,
I'm afraid you'll have to be much more specific about your problems
for anyone to help you. Despite
There was a problem with CVS zaptel and asterisk that caused it to fail
to play prompts when devfs was installed. Please update zaptel (libpri if
you use it) and asterisk. Thanks.
Mark
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Get a trace using Ethereal when the phone boots up and look in the warning
field of the sip message, if it lists your firewall type as symetric theres
a good chance your out of luck using that firewall. I'm a bit confused
regarding your port selection, as 3478 is cleared stated as the broadcast
Sorry to answer your question, you need to down load the source from vovida
and compile it. Follow the instrustion in the readme on the main page. Do
not use ports indicated 1 and 1000x. Use 3478 and 3479. Oh for the
alternate stun server (-a option) add 127.0.0.1. It's really straight
Today's frustrated programmer award goes to Linphone, which has the
following debug output:
(linphone:28655): LinphoneCore-WARNING **: this fucking remote sip phone did not answered properly to my sdp offer!
I get this message when I connect to linphone using a softphone, or when
I try to use
Maybe I mis-understood the question or the architecture. I assumed (I
know), the SIP UA sat behind the NAT and Asterisk sat on the public IP
network.(there are inhererent signaling problems in this scenario and will
not work without either the device having the ability to learn the WAN IP
address
On 2003-07-01 at 13:27, Jim Gottlieb (That's Me!) wrote:
-- Starting simple switch on 'Zap/1-1'
== Unknown extension 's' in context 'intrunk' requested
I also see logged:
File chan_zap.c, Line 3833 (ss_thread): Got a non-Feature Group B input on channel 1.
Assuming EM Wink instead
What version of X-Lite are you using. The latest is build v1035. There
where problems in earlier releases with SDP values, that could be the reason
you not seeing invites or media. I had issues only with the media not
setting as X-lite tried to negotiate media with another endpoint and teh SDP
John,
When you say you have SIP clients working behind NAT is this with ports
mapped from a public ip to the phone? I.e. can many phones sit behind 1
public ip and recieve incomming calls, and make outgoing calls?
- Justin
On Tue, 1 Jul 2003, John Todd wrote:
Sorry, I still don't know what
No, it works fine. SIP UA behind the NAT. Asterisk outside the NAT.
nat=1 set on the SIP peer. Works fine. Really. It does.
I use Cisco equipment for my UA's. The catch might be that the Cisco
devices are more clever than their counterparts, and will compare
the Via: header against their
Yes, I have one location where there are a dozen or so behind the
same NAT. Things work fine for inbound and outbound. I'm sure there
is a theoretical limit based on what an 8 or 16 bit integer can hold,
but I'm not worried about hitting that problem any time soon.
JT
John,
When you say
No, I have not tested alternatives. Perhaps Mark can interject a
comment here on if he has REINVITEs working for devices behind
different NATs or if that is on the agenda? I haven't experimented
widely on SIP/NAT interactions since it became stable in the CVS code.
JT
John,
Thanks for the
Your correct, Cisco devices stuff the WAN address in the Via: header which
in turn allows the proxy to correctly register the UA for an incoming call
attempt to that UA. If Mark is mentioning STUN as I said before, the only
devices I'm aware of are the SNOM 100 and Grandstream 101. These devices
At 20:08 2003-07-01 -0400, Michael Kane wrote:
What version of X-Lite are you using. The latest is build v1035. There
where problems in earlier releases with SDP values, that could be the reason
you not seeing invites or media. I had issues only with the media not
setting as X-lite tried to
To find out what version yuor using, dial *999 and a debug/trace window will
appear. In the SIP messages it will indicate the type of UA your using and
the version. example below: try another call attempt with this window
open and capture the call flow and send it to me. See below in bold or
You may be correct about the Via: header, but you're incorrect in the
concept as to how it relates to Asterisk, notably in your reversal of
what side of the transaction is putting data in the Via: header to
make SIP work correctly.
This is cluttering up the list. Talk to me off line if you
Please don't take the discussion of SIP interactions off list. I already
have NATed SIP clients working with *, but * still has problems where
its own external IP is not public and it is trying to use external SIP
services. A full discussion on list could spawn an Asterisk SIP FAQ -
and I think
Do you have any instructions as to how to compile and install your rtc
package? I downloaded it, but it is unclear to me as to whether I need to
compile and instal the Zaptel stuff first or if I can just use the source,
etc.
Any help would be appreciated.
Sean
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