- Original Message -
From: Peter Svensson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, June 06, 2005 9:30 AM
Subject: Re: [Asterisk-Users] Disa - how it returns on user not dialing
anynumbers ?
On Mon,
hello, All AreskiCC users:
I faced some problems in using AreskiCC. one is when I reload the
asterisk server, the system display some errors such as execution 30 ..
second one is there is no data display for admin added before. Does
anyone know how to solve the problems, Please tell me! thanks in
Hi
To debug SIP, either you can use asterisk debugging,
from console give command sip debug on
Or the best applicatio is Ethereal from www.ethereal.com
Thanks Regards
Ritesh Jalan
Senior Engineer - Test Audit
Net4India Ltd.
703 Bikaji Cama Bhawan
11 Bikaji Cama Place
New Delhi - 110029
Ph:
I am not sure of what Digium hardware you are looking to use, but I am using
IBM xSeries 300 (Celeron 1. GHz, 768 Meg RAM, 20 Gig IDE drive) for a quad
T1 (405P). Thus far, I have not hit 96 active users, but have also not run
into any issues. FC1 is the OS and * 1.0.7 stable.
-Original
Hi Michiel,
-Original Message-
I been searching on the wiki and google for ENUM in NL.
All I could find were some docs from the Dutch Financial
Department about taskforces and plans. But it all links to
dead pages and no-longer-connected phone numbers.
Is there anyone who knows
Can someone please tell me WTF this has to do with *???
-Ronan
On Sun, 2005-06-05 at 15:43 -0400, C F wrote:
On 6/3/05, Karl J. Vesterling [EMAIL PROTECTED] wrote:
Hmmm... Seems we have a clear case here of if you can't defend the
position with lies and retoric attack the credibility
When dialling a number, my 7960 always says External call. Is there
any chance to tell it the difference or make it display the dialled
number (especially nice when using speeddials)?
CallInfoMessage issue, it should be fixed by Jan in the CVS, I will
check it out
Is there any way
I am using the TE110P and the TDM400P 2x2. I have problems with both on the
Dell Servers the Poweredge 800 and the SC420. I was looking for solutions
for future customer deployments with either a TE110P or TE405P or TE410P. It
is great to know that the IBM x series work well.
Syed Akbar
Alico
On Jun 6, 2005, at 4:31 PM, Chris Mason (Lists) wrote:
Digium did acquiesce and allow me to relicense the codec today,
essentially
they asked me how any times I would like to be able to re-license,
I hope you answered, As many times as is necessary to ensure I'm
continually able to
Hi folks,
We have been diax with a usb phone on windows...
normally we keep diax on a usb memory stick
this produces the problem we need to use a hub of 2 ports.
Has anyone found a USB phone handset which Actually say even 64Megs of
ram ??
would be much more convenient.
Gary
.
Dear all,
just upgraded to cvs head june 6th, using 1.0.7 sip.conf but can't accept
any calls from SIP proxy. Anyone encountered the same problem?
[general]
context=sip-in
recordhistory=yes ; Record SIP history by default
port=5070 ; UDP Port to bind to
How do I run an external script on completing a call?
Like if I want to send email to the caller.
Thansk
Eric
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I wonder how to allow more then one codec in AMP ([EMAIL PROTECTED]) GUI?
For example I want to configure like this
allow=gsm
allow=g729
...
I can add these by editing sip_additional.conf, but i want
to add codecs using AMP, any suggestions?
Thanks
Erdem HAKI [EMAIL
Hi Florian,
On 09:05, Tue 07 Jun 05, Florian Overkamp wrote:
Hi Michiel,
-Original Message-
I been searching on the wiki and google for ENUM in NL.
All I could find were some docs from the Dutch Financial
Department about taskforces and plans. But it all links to
dead pages
Thank you for ypur answer.
The problem is the way I pass the argument to the custom application
I think, but I could be wrong, that only
[custom-did-route]
exten = _0101234XXX,1,Macro(exten-vm,${EXTEN:[EMAIL PROTECTED],${EXTEN:7})
could go correctly... if only the DID colud reach this entry !
In
Hi all,
i'm trying to install Asterisk with Junghanns QuadBri ISDN card: I have
followed all the instructions, and when I run |ztcfg -tvv I get
|
| Zaptel Configuration
==|
SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
Channel map:
Channel 01: Individual Clear
I am using Asterisk CVS-HEAD-06/02/05-19:37:27. It seems that every
call I made was duplicate.
Jun 8 00:11:30 DEBUG[21733]: chan_h323.c:411 oh323_call: Placing
outgoing call to 87874586, 101
-- Called 87874586
Jun 8 00:11:31 DEBUG[21733]: rtp.c:472 ast_rtp_read: RTP NAT: Using
address
On Tue, June 7, 2005 10:14, Michiel van Baak said:
Hi Florian,
On 09:05, Tue 07 Jun 05, Florian Overkamp wrote:
Hi Michiel,
-Original Message-
I been searching on the wiki and google for ENUM in NL.
All I could find were some docs from the Dutch Financial
Department about
Ok, just a thing...cuold is see a sample peer in tuou extensions.conf
Thanks
Giordano
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Gavin Hamill
Inviato: lunedì 6 giugno 2005 21.50
A: asterisk-users@lists.digium.com
Oggetto: RE: R: R: R:
It's appear that you have defined 4 channels in you zapata.conf, but the
ztcfg -tvv only detect 2B and 1D.
Can you paste you zapata.conf?
Regards
- Original Message -
From: Daniele Gianetti [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, June 07, 2005 10:21 AM
Thanks for answering so quickly, I have tried several zapata.conf and
zaptel.conf with no result: this is the latest try
zaptel.conf
loadzone=it
defaultzone=it
# qozap span definitions
# most of the values should be bogus because we are not really zaptel
span=1,1,3,ccs,ami
span=2,0,3,ccs,ami
Hi
All,
Can
anyone give me some pointers about what is the requirement at both sides?
I
already have OH323 support in Asterisk, but have no clue how to configure the
HiPath.
Thanks,
Ohad
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Hi!
How did you manage to get 4 Fritz cards in the same
box? Could you give
details on what kernel and kmodules you used? Which
asterisk channel
are you using? chan_capi? chan_misdn?
Thanks,
Leandro
Hi,
in order to get 4 Fritz cards running in the same box,
I have patched the AVM's
Hi, I want to install * and te405p on Dell Poweredge 1850. Can
I do that successfully? Any one has successful experience on that
scenario?
Thanks.
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Hello gentlemen, I am new here.
I have a D-Link DPH-80S SIP phone (it's a non-US model), and I am trying
to make it work with Asterisk. I tried versions 1.0.7 and yesterday's
CVS and the behavior is the same.
The phone registers with no problem, and can accept calls.
But when I try to make
Unfortunately I believe there is a lot of truth to it. The speed in
which 911 legislation took effect is no coincidence and you don't see
the big telcos complaining about it. He's right about the price issue
too. Do you see how much big providers charge for VoIP service?
MARK.
Colin
I tried to add the databases from the localhost to the database server
and changed the every /etc/asterisk/*.conf from host=localhost to
host=192.168.10.10
(my dababase server)
When I restart asterisk, I do not get any errors, but after a phone call
I see:
Jun 7 18:11:56 ERROR[7877]:
Have you given the access to your asterisk server in your database server??
Thanks Regards
Ritesh Jalan
Senior Engineer - Test Audit
Net4India Ltd.
703 Bikaji Cama Bhawan
11 Bikaji Cama Place
New Delhi - 110029
Ph: +91-11-26160129 ext. 131
Web site: http://www.net4india.com
Eric, the SIP/RTP protocol does not inherently work well in NAT'd
environments. There are several commerical solutions out there to help
users traverse NAT routers and firewalls successfully, with varying
levels of success. I find that many commercial nat routers for home
users (most
Hi,
What is the easier way to configure 2 Asterisk to
comunicate one with each order?
Atenciosamente,
---Cleyverson Pereira
Costa(27) 99220111 / (27) 33392247MSN: [EMAIL PROTECTED]
___
Asterisk-Users
Quoting Ronald Wiplinger [EMAIL PROTECTED]:
Did you just copy the database files over to the new server?
You may have to grant the user rights again. Also remember that mysql
rights are host sensitive. If all your previous grants were localhost based,
instead of ip address/netmask based, you
Ritesh is likely correct, have a look at:
http://dev.mysql.com/doc/mysql/en/adding-users.html
You will want to make sure that for the mysql user you are using to
connect to the mysql server you have granted permissions in the 'mysql'
database in the 'db' table for some or all foreign 'Host's
No additional hardware would be needed if you connect to a voip provider
for your inbound/outbound calls. If you want to plug phone lines or a
PRI into your asterisk box directly, visit digium.com and have a look at
the various PCI cards they offer.
-mike
infra struct wrote:
I have a
Mirko Marghitola wrote:
Asterisk don't send the 180 Ringing SIP message to the calling phone
when the called party is ringing. How can I force asterisk to send the
ringing messages? The option 'r' in the Dial() command or the Ringing()
command didn't solve the problem.
Mirko
Did the sip
Reading through the code, I don't see a way of exiting the transfer and
regaining the call with the customer, unless the third party hangs up or
maybe doesn't answer and the dialplan doesn't do anything else with the
call (send the call into voicemail).
I suggest you request this feature
On Monday 06 June 2005 17:12, Louis-David Mitterrand wrote:
With kernel 2.6 you only get ISDN tty devices through the capidrv
module, no more analog modem login support with the obsoleted diva2i4l
module.
I've never used it on 2.6, and only with the official Eicon supplied drivers
with the
on Dell Poweredge 1850. Can I do that
successfully? Any one has successful experience on that scenario?
Thanks.
-- next part --
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I've found a woman whom is happy to help make English voice files!
Ironic that she should be in New Zealand.
More when I have the files.
--
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
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I tested IAXy with my asterisk server in US, using both DSL. It was
working
fine.
I gave it to my friend who was traveling to Canada. He is saying that it
is
not working with Rogers Cable. It is getting busy tone after 20-30
seconds.
Is it possibly port blocking? or any other
Ritesh Jalan wrote:
Have you given the access to your asterisk server in your database server??
Thanks, you are right, ...
However, I cannot find the right way to GRANT the access. Do you have an
example for me, please?
bye
Ronald
___
You can use super-valet-parking
On Tue, 2005-06-07 at 06:18 -0500, Mike Holloway wrote:
Reading through the code, I don't see a way of exiting the transfer and
regaining the call with the customer, unless the third party hangs up or
maybe doesn't answer and the dialplan doesn't do anything
Shell mysql --user=root mysql
mysql INSERT INTO user
- VALUES('IP_ADDRESS_OF_ASTERISK','USER_ID',PASSWORD('some_pass'),
- 'Y','Y','Y','Y','Y','Y','Y','Y','Y','Y','Y','Y','Y','Y');
mysql FLUSH PRIVILEGES;
Thanks Regards
Ritesh Jalan
Senior Engineer - Test Audit
Net4India Ltd.
If you have phpMyAdmin installed, you can simply go to the webpage, go
into the mysql database, the db table, hit the Browser tab, and clone
the line for your particular asterisk db, changing the Host to %, which
will permit any host to connect to that database (assuming proper
Gary wrote:
Has anyone found a USB phone handset which Actually say even 64Megs of
ram ??
No, unfortunately
would be much more convenient.
Indeed it would
--
Cheers,
Matt Riddell
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Count me in also.We have to join forces.AndreCEO Vink Consultancy- Oorspronkelijk Bericht -Onderwerp:RE: [Asterisk-Users] ENUM NL dead ?Afzender: Florian Overkamp [EMAIL PROTECTED]Aan:'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.comDatum:07-06-2005
Forgot to mention the step to
mysql FLUSH PRIVILEGES;
after making changes.
-mike
Mike Holloway wrote:
If you have phpMyAdmin installed, you can simply go to the webpage, go
into the mysql database, the db table, hit the Browser tab, and clone
the line for your particular asterisk db,
Is this even possible or am I better off getting a voip number for each of
the existing numbers I want to forward.
Thanks!
-Original Message-
From: Geoff Manning [mailto:[EMAIL PROTECTED]
Sent: Friday, June 03, 2005 4:53 PM
To: Asterisk Users (E-mail)
Subject: [Asterisk-Users] Call
Hi all,
I have an asterisk working with several SIP gateways. If I type sip
show peers and sip show users I can see the gateways.
But sip show subscriptions returns 0 records.
Also everytime I place a call the gateways get a 407 message from
asterisk and hate to re-authenticate.
Can you
I already have OH323 support in Asterisk, but have
no clue how to
configure the HiPath.
hi...
oh323 is the only thing you need for Astersik. For the
HiPath it depends on which version you have.
FOR HiPath4000 V1.0
---
for version 1.0 you need a HG3550 V1.1 Board.
-Configure
A Dell 1850 and TE405P didn't work very well for me.
Got a lot of static on ZAP calls. There's a whole thread about it from a
month ago!
Tried everything to fix it and in the end went back to 2 E100P cards which
did the trick.
Aaron
-Original Message-
From: [EMAIL PROTECTED]
fredrik chabot wrote:
The problem is
as follows
I've made a queue and i queue incoming calls in that queue.
The reception log's in as an agent to that queue and gets the calls for
that queue so far so good.
Now I need to transfer the call. I press flash (all granstream bt101
Geoff Manning wrote:
Is this even possible or am I better off getting a voip number for each of
the existing numbers I want to forward.
Thanks!
-Original Message-
From: Geoff Manning [mailto:[EMAIL PROTECTED]
Sent: Friday, June 03, 2005 4:53 PM
To: Asterisk Users (E-mail)
Subject:
When sending a call to a line defined on chan_sccp,
there is an error on the console that says:
Jun 7 08:22:29 WARNING[3924]: sccp_channel.c:79
sccp_channel_send_callinfo: Incoming call SCCP/Line1-0008 doesn't
have CallerId name
Is this because of the changes in the callerid name from
Browsing, I came across this 3com DSS console:
http://www.3com.com/products/en_US/detail.jsp?tab=featurespathtype=purc
hasesku=3C10405A
The ad claims SIP by Summer 2005. Does anyone know anything about this
device and its interoperability? Any potential asterisk
support/integration with
Make sure the PCI revision on the 1850 is at least 2.1
-Matthew
From: Ma Zhiyong [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Tue, 7 Jun 2005 17:22:23 +0800
To: asterisk-users@lists.digium.com
Subject:
The database peers/users are not kept in memory. These are only loaded when
we have a call and then deleted, so there's no support for NAT keep-alives
(qualify=) or voicemail indications for these peers.
NOTE: As of CVS-HEAD 3/16/05, if you enable RealTime caching in your
sip.conf, Voicemail
On Monday 06 June 2005 22:20, Obaid Siddiqui wrote:
I tested IAXy with my asterisk server in US, using
both DSL. It was
working
fine.
As someone else stated, first try and do a trace route
from your friends end to your * box. Once it is confirmed
that he can even reach your IP, try
http://www.voip-info.org/wiki-Asterisk+h+extension
This might help
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Eric Smith - Fruitcom
|Sent: Tuesday, June 07, 2005 1:02 AM
|To: Asterisk Users
Hi
I have a PHP agi-bin scripted called callhander.php and its setup to
answer anything that comes into the PBX,
In the script I am trying to the get the system to play a file called home
which I know works, as I can get the Play function to work from the
extensions.conf file. However within
On superviced you cancel on a zap channel you can cancel the transfer by
a hook flash this will send you back to the original caller.
On sip phones you hit the cancel button or if you have line buttons you just
pick the original callers line.
--
From: Mike Holloway[SMTP:[EMAIL
hi,
can you someone post tftp template for gxp-2000?
like
http://www.grandstream.com/DOWNLOAD/Configuration_Tool/Windows/Grandstream_Configuration_File_Template_1.0.6.x.txt
thanks
---
Marek Cervenka
===
Mike Holloway wrote:
If you have phpMyAdmin installed, you can simply go to the webpage, go
into the mysql database, the db table, hit the Browser tab, and clone
the line for your particular asterisk db, changing the Host to %,
which will permit any host to connect to that database
Does anyone know if there is any way to make the 'services' key do
anything?
How about a way to remap it?
Chris Coulthurst
[EMAIL PROTECTED]
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Hi
all,
i'm testing my
asterisk and without warning i can not hear any audio file (the files situated
under /var/lib/asterisk/sounds).
I don't hear no
audio and i get this message on CLI:
*CLI -- Executing Dial("SIP/2339-4e1d",
"SIP/2391|60|Ttr") in new stack -- Called
2391 --
Hi, is there any way I could make this work without having to explicitly
perform port forwarding for RTP traffic at my NAT? (i.e. NAT
transparently sets up the RTP channel for the internal SIP UA with the
external SIP UA) Thanks Eric Date: Fri, 03 Jun 2005 09:13:18 -0500 From:
Eric if they
Read /var/log/asterisk/debug for errors
-Matthew
From: Ronald Wiplinger [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Tue, 07 Jun 2005 18:26:54 +0800
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hello! Continuing my PRI saga - I have a PRI setup and appears to be
answering calls OK, but my carrier is cutting all the calls after 15
seconds. For example, when I call from my cell phone, it goes
straight to a busy signal - however the CLI shows the call coming in
and being answered.
That was the problem! Thank you very much!
It happens that [EMAIL PROTECTED] comes only the r in the Dial String, leaving
out t and T...
Thanks again
M.G
- Original Message -
From: Jay Milk [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Matthew Boehm wrote:
Read /var/log/asterisk/debug for errors
I am not sure what I am looking for there?
I tried to add the databases from the localhost to the database server
and changed the every /etc/asterisk/*.conf from host=localhost to
host=192.168.10.10
(my dababase server)
When
Hello all,
Has anyone tried 8xE1 in one box using Asterisk and Digium boards ?
What is the CPU needed for sustained performance in this capacity ?
Is this affected if G.729 codec is used ?
Regards,
Jorge A.
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Cleyverson P. Costa wrote:
Hi,
What is the easier way to configure 2 Asterisk to comunicate one with
each order?
In order of ease:
1. hire somebody to do it - an email on asterisk-biz will sort you out
2. hire somebody to give you some basic training to get started (that's
what I did)
OK, I found the solution
custom-did-route,${EXTEN},1
the call to the custom application has to be done in this way
Now I am happy, becouse with only one line in extensions_custom.conf and 1
DID Route in AMP I can
route all of the external extensions (01012345xx) to the internal
extensions (5xx)
Hi, I want to install * and te405p on Dell Poweredge 1850. Can I do
that successfully? Any one has successful experience on that scenario?
Thanks.
I do. The only thing I would like to inform you, is the total lack of
other Linux distros support than RedHat.
I'm running a very slimmed 1850 with
On 7/06/2005 10:27 PM, Joseph wrote:
When sending a call to a line defined on chan_sccp,
there is an error on the console that says:
Jun 7 08:22:29 WARNING[3924]: sccp_channel.c:79
sccp_channel_send_callinfo: Incoming call SCCP/Line1-0008 doesn't
have CallerId name
Is this because of the
Hello everyone,
This is is my first email to this group.
I'm am writing a small php program to pull some info out of our
Asterisk's queue_log. I'm having trouble figuring out what some of the
parameters mean.
Here's an example:
1118098465|1118098465.47|salesq|NONE|ENTERQUEUE||Ray Balbin 25
Julien Goodwin ha scritto:
On 7/06/2005 10:27 PM, Joseph wrote:
When sending a call to a line defined on chan_sccp, there is an error
on the console that says:
Jun 7 08:22:29 WARNING[3924]: sccp_channel.c:79
sccp_channel_send_callinfo: Incoming call SCCP/Line1-0008 doesn't
have
Title: AAH 1.1 - CRM Setup
Hello All,
Has anyone successfully gotten the Click to Dial to work in SugarCRM in the latest AAH?
I keep getting 'Invalid Channel' but I cannot figure out why.
Thanks!
Wiley
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On Jun 6, 2005, at 4:39 PM, [EMAIL PROTECTED] wrote:
I want to manage this dialplan variable for each extension separately,
unfortunately this doesn't work:
**77,hint,DS/splat${CALLERIDNUM}
Do you have an idea for that?
Is there an easy place to patch it in asterisk 1.0.7 stable?
Will it be
Hi Tom,
Have you tried replacing your Answer command with Ringing, some PRI
requires the Ringing signal before?
It worked for us
Francois Lambert
COO
Atelka/Aheeva Inc.
Tel. : 514-448-4905 #2200
Cel. : 514-570-4797
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It's pretty obvious from the wiki that realtime and
Nat don't befriend quite well. As It is obvious the necesity of both of them,
mainly have clients under nat talking to an asterisk server. The question I
would like to throw away is.. What would you do to have both of them? I have two
Those of you who monitor how do you handle outgoing
faxes?
I have my * box connected to my legacy PBX via T1
and all calls going out of my PBX are going Voip via asterisk but my monitor
script is making my fax transmissions fail.
Is there a way to determine if the call is an
outgoing fax
AAH is just a collection of projects. A very good one at that.
So there are no config files from AAH per se. They are coming from AMP.
Download the amportal tar.gz file from the AMP project. It is a lot smaller.
http://sourceforge.net/project/showfiles.php?group_id=121515package_id=132595
Although I have not really tried much IAX stuff yet I am on Rogers in
Ontario, Canada. So if you need someone to do a bit of troubleshooting with
you I'd be glad to help.
The only ports I know that Rogers blocks are 139 and the 1433.
They don't block 25 (as I run a mail server and everything
Why would realtime have a problem with NAT ? its just another flag in
asterisk.
Victor Alvarez wrote:
It's pretty obvious from the wiki that realtime and Nat don't befriend
quite well. As It is obvious the necesity of both of them, mainly have
clients under nat talking to an asterisk server.
Hi,
Has anyone used the SS7 link from Digium? If so, how did it work for
you? Any issues? Anything to be aware of? Do I just need a T1 card
like the PRI card I have now from Digium?
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On Tue, 2005-06-07 at 16:49 +0200, Sergio Chersovani wrote:
Julien Goodwin ha scritto:
On 7/06/2005 10:27 PM, Joseph wrote:
When sending a call to a line defined on chan_sccp, there is an error
on the console that says:
Jun 7 08:22:29 WARNING[3924]: sccp_channel.c:79
On 07/06/05 11:30 -0400, Matt wrote:
Hi,
Has anyone used the SS7 link from Digium? If so, how did it work for
you? Any issues? Anything to be aware of? Do I just need a T1 card
like the PRI card I have now from Digium?
Hi Matt,
There are some links to user reports on the wiki:
Hi folks --
For those of us trying to develop interactive web-based Asterisk
applications, it can be a challenge getting Asterisk integrated in a
cgi/mod_perl/php environment. The load associated with making multiple
connections to asterisk via the manager port, having to teach our
Hello i would like to receive incoming faxes thru' asterisk as tiff
files thru' the rxfax application.
I setup extensions 101 like this
exten= 101,1,rxfax(/tmp/fax.tif)
then from CLI i run:
dial 101
and rxfax send me his scream about the fax ^^
instead when i send a real fax
Good lord people. You claim that you all read the Wiki, but you're not
reading ENOUGH!
READ AGAIN!! RealTime DOES NOT HAVE A PROBLEM WITH NAT!!
Let me rephrase that to something a little simpler:
NAT WORKS WITH REALTIME!!
MWI WORKS WITH REALTIME!!
QUALIFY WORKS WITH REALTIME!!
Sorry, I'm not seeing user reports on the wiki.. I look at that
before.. just posted here to try to get some input from users on their
thoughts on it.. and if it works as advertised..
On 6/7/05, Jason Stewart [EMAIL PROTECTED] wrote:
On 07/06/05 11:30 -0400, Matt wrote:
Hi,
Has anyone used
Chris Coulthurst wrote:
Does anyone know if there is any way to make the 'services' key do
anything?
How about a way to remap it?
Look in the Polycom SIP Administrators guide. You can map the button to
a variety of other things.
I haven't tried it, but the documentation appears fairly
This is not the problem
There is an example for faxing. you should set some other parms. ie.
unique filename etc., you also need permissions etc.
Look at NVBackground detect from Newman Telecom. Use the wiki and search
for NVFaxDetect
HTH
Hello i would like to receive incoming faxes thru'
Has anybody tried to use a Polycom phone (I have 500s and 600s) with a
netmask shorter than /24? (A network bigger than 255.255.255.0). We've
run out of IPs in our initial /24 network, and I'd like to expand it to
255.255.248.0.
When I set it to 255.255.248.0 I can ping the phone while the
Joseph ha scritto:
sccp_channel_send_callinfo: Incoming call SCCP/Line1-0008 doesn't
have CallerId name
I am using your new format in sccp.conf and that works fine for outbound
calls. The problem is inbound calls.
Ok got it in the sourcce code. I will fix it tomorrow.
Sergio
sipgetheader(or_To=To)
Cut(or_To,or_To,:,2)
Cut(or_To,or_To,@,1)
That works! Thanks!
Correction to the cut command below, replaced , with = :
sipgetheader(or_To=To)
Cut(or_To=or_To,:,2)
Cut(or_To=or_To,@,1)
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On Tue, Jun 07, 2005 at 11:30:27AM -0400, Matt wrote:
Hi,
Has anyone used the SS7 link from Digium? If so, how did it work for
you? Any issues? Anything to be aware of? Do I just need a T1 card
like the PRI card I have now from Digium?
FYI, I don't think that Digium has an SS7 link :-)
or use the g option in the dial command
On 6/7/05, Eric Smith - Fruitcom [EMAIL PROTECTED] wrote:
How do I run an external script on completing a call?
Like if I want to send email to the caller.
Thansk
Eric
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I just assigned one yesterday in a 10.X.X.X network with a netmast of
255.0.0.0, and had no problems..FYI.
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Charlie Watts
|Sent: Tuesday, June 07, 2005
Can you try a 'more standard' boundary?
Like 255.255.0.0 or 255.0.0.0 ???
If so it (polycom) may not understand CIDR.
You may want to look at implementing a VLAN...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Charlie
Watts
Sent: Tuesday, June 07,
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