Well looks like the phone is sending some data... I was unable to debug the
problem however..
~Shaun
-- SIP read from 68.5.xxx.xxx:1025:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK396e6066
From: 302 sip:[EMAIL
On 04/16/06 10:51 C F said the following:
use feauters.conf and the application map section.
i may be wrong, but that's not the same as background music during a call.
iianm, using playback() or background() in features.conf turns off the call
audio and plays the selected file.
--
I have a call screening system setup, caller calls in runs a macro and sets
a far to track the recording that was taken of the callers name... then the
callee runs a macro also that plays him that recording (pulled from that var
that was set) This works fine until i use a queue in the
On Saturday, April 15, 2006 3:17 PM Remco Barende wrote:
I heard that Junghanns is working on such an interconnection. It is
already possible to connect their PRI cards, and they are working on
BRI-PRI.
Correct. The next driver generation is supposed to support this fully.
I ise their
Shaun wrote:
I have a call screening system setup, caller calls in runs a macro and sets
a far to track the recording that was taken of the callers name... then the
callee runs a macro also that plays him that recording (pulled from that var
that was set) This works fine until i use a
have you tried serveremail or mailcmd ?On 4/15/06, Ronald Wiplinger [EMAIL PROTECTED] wrote:
How can I change this:Asterisk PBX [EMAIL PROTECTED]to:London PBX [EMAIL PROTECTED]
??I tried several settings in voicemail.conf, without success!byeRonald
On 4/15/06, Min Hwan Chang [EMAIL PROTECTED] wrote:
wondering if its stable enough to use. Currently I'm editing my own *.conf
Using it at multiple sites (ranging from 10-50 extensions).
scripts but it sure would be nice if there were some sort of web interface
for other people to use. The
Hi,
Check the vm_general.inc file
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronald
Wiplinger
Sent: Sunday, April 16, 2006 12:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] voicemail email-from
This is how we do faxing, and till now it has bee 100% accurate.
Incoming faxes are handled by chan_capi, using a cheap AVM fritz pci
card.
Outgoing fax is done with an old brother analog fax, connected to a
grandstream 285 (or whatever the number, it's their cheapest).
Asterisk connects
kevin ling wrote:
Hi,
Check the vm_general.inc file
Where should this file be?
bye
Ronald Wiplinger
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronald
Wiplinger
Sent: Sunday, April 16, 2006 12:24 PM
To: Asterisk Users Mailing List
Ronald Wiplinger wrote:
How can I change this:
Asterisk PBX [EMAIL PROTECTED]
to:
London PBX [EMAIL PROTECTED] ??
I tried several settings in voicemail.conf, without success!
I found it!
I had to change the info in the /etc/passwd file
bye
Ronald Wiplinger
begin:vcard
fn:Ronald
Dinesh Nair wrote:
On 04/16/06 10:51 C F said the following:
use feauters.conf and the application map section.
i may be wrong, but that's not the same as background music during a
call. iianm, using playback() or background() in features.conf turns
off the call audio and plays the
Michiel van Baak wrote:
This is how we do faxing, and till now it has bee 100% accurate.
Incoming faxes are handled by chan_capi, using a cheap AVM fritz pci
card.
Outgoing fax is done with an old brother analog fax, connected to a
grandstream 285 (or whatever the number, it's their
Ronald Wiplinger wrote:
kevin ling wrote:
Hi,
Check the vm_general.inc file
Where should this file be?
bye
Ronald Wiplinger
You could probably find it using updatedb and then slocate or
locate or Google the wiki will almost definitely give you the location.
Kevin
-Original
On Apr 16, 2006, at 2:07 PM, Steve Totaro wrote:
How many faxes a day do you average? If it is one a day for six
months then that is one thing. One hundred a day and I would say
that you definitely have a stable setup.
Yeah, only couple faxes a week.
So nothing huge here, but like I
Many PBX's based on the Asterisk code refer to this in their feature
list. Its taken off of the Asterisk.org site. Where is this used in
Asterisk? Is it in the extensions.conf, is it used when logging into the
console, where? How is it used? Is this referring top contexts in the
dial plan?
Ronald Wiplinger wrote:
Steve Totaro wrote:
Ronald Wiplinger wrote:
kevin ling wrote:
Hi,
Check the vm_general.inc file
Where should this file be?
bye
Ronald Wiplinger
You could probably find it using updatedb and then slocate or
locate or Google the wiki will almost definitely
jennyw wrote:
We've been reasonably happy with Polycom SoundPoint phones, but we
only have them installed on the LAN. I've read that they have problems
working across NAT. So ... I guess I have a few questions. First, is
there a way to get Polycoms to work well over NAT? If not, then are
Thanks for posting it back to the list
kevin ling wrote:
Hi,
Check the vm_general.inc file
Where should this file be?
You could probably find it using updatedb and then slocate or
locate or Google the wiki will almost definitely give you the
location.
Short, you do not know it,
Hi, Is there a way to limit the duration of a call in the Dial command? Mainly for perpay account. Thanks__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
On Saturday 15 April 2006 22:37, C F wrote:
That is until you run into problems, while they do work, I wouldn't
say that Polycoms work EXEPTIONALLY well, Cisco, and SPA work *MUCH*
better.
Can you detail some problems? Just about any off-the-shelf router seems to
work with these. There may
On Sunday 16 April 2006 08:07, Steve Totaro wrote:
How many faxes a day do you average? If it is one a day for six months
then that is one thing. One hundred a day and I would say that you
definitely have a stable setup.
Does 14526 faxes to date (since 06 Jan 2006) count as stable? My setup
Google is your friend, or you enemy, either way they usually have an answer:
/Dial a single destination, ringing for a maximum of 20 seconds. Limit
the call length to 60 seconds, warning the caller when only 20 seconds
remain:/
exten = 200,1,Dial(SIP/1234,20,L(6:2))
--
Ronald Wiplinger wrote:
Thanks for posting it back to the list
No problem, not sure why you would think I would like to correspond with
you directly. I am into the community thing. Why send me a direct
email with some crappy process to become a verified sender just to be
able to send
Shaun wrote:
Well looks like the phone is sending some data... I was unable to debug the
problem however..
Looking for 9011905326471222 in default (domain 204.10.xxx.xxx)
Do you have a pattern in the default context that will match
9011905326471222 ?
2006/4/16, Koopmann, Jan-Peter [EMAIL PROTECTED]:
The drivers for their BRI/PRI cards are totally independant from thestandard HFC driver. I own a QuadBRI card and do not have any timingproblems whatsoever.We used this card and still got faxing problems (roughly 95% of faxes worked OK but 5% of
What firewall was the problem user running? We have Polycoms behind
Linux, Mikrotik, Linksys, Dlink, Netgear, etc all without any problems.
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Mason (Lists)
Sent: Sunday, April 16, 2006 9:21 AM
To:
On Tuesday 11 April 2006 21:45, Begumisa Gerald M wrote:
Hi,
I've been battling with a similar issue:
a) I wrote a script to periodically run the command cat
/proc/interrupts and figure out the interrupts per second. I run this
script for over 24 hours and never once did the difference
Hi users:
astcc script exits when dialing an uncomplete or wrong
number. What changes need to be made for astcc.agi to
allow dialing phone numbers more than one wrong
attempt.
Regards;
Chawki Hammoud
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail
Steve Totaro wrote:
Ronald Wiplinger wrote:
Thanks for posting it back to the list
No problem, not sure why you would think I would like to correspond
with you directly. I am into the community thing. Why send me a
direct email with some crappy process to become a verified sender just
2006/4/14, Rich Adamson [EMAIL PROTECTED]:
Factually, the sangoma cards integrate with the pci bus in a much morestable/usable way then does the digium TDM card (and I believe the te110if it uses the TigerJet pci chip).Could you elaborate ?
What are the pos's and con's of each PCI bus integration
What are you doing with GPS and what is the moringa tree?
My work with GPS is on a need to know basis, and the moringa tree is
another thing that can readily be found vi google.
http://www.google.com/search?sourceid=navclientie=UTF-8rls=GGLG,GGLG:2006-13,GGLG:enq=moringa+tree
Andrew Kohlsmith wrote:
On Sunday 16 April 2006 08:07, Steve Totaro wrote:
How many faxes a day do you average? If it is one a day for six months
then that is one thing. One hundred a day and I would say that you
definitely have a stable setup.
Does 14526 faxes to date (since 06 Jan
Hi!
First I want to say hello to the complete list. I am new and I have a
problem, where I did not find an answer in the archives and on
voip-info.org ...
I am subscribed to two SIP Providers:
1. SIPCall.at
2. SIPGate.at
SIPCall works fine with signalling (incoming and outgoing calls), but
Alex Mosburger schrieb:
...
SIPGate works also fine (better than SIPCall) but the ECHO is terrible.
My side (* server) connected with X-Lite Softphone has a great quality,
but the PSTN caller hears his voice with an echo.
Did anybody already had such a problem? Do you think that my * server
...
I forgot in my previous email one further issue:
Maybe the ping round trip to your SIP provider and thus
to the PSTN gateway is too long. The echo cancellation
is typically limited to a reasonable echo run time.
If the time is too long, echo cancellation will fail,
because it would be too
Thanks fort he fast feedback Roger!
It is not my end hearing or producing echo. My voice is heard correctly
without any echo, but the other side hears his OWN voice several msec
later... MM... a switch to another provider is in this case a good
idea... ;)
Alex
-Original Message-
Alex Mosburger schrieb:
...
It is not my end hearing or producing echo. My voice is heard correctly
without any echo, but the other side hears his OWN voice several msec
...
Yes, this is, what I meant.
The other's voice is fed back by your device and running
back to the other side. That's why
On Sun, Apr 16, 2006 at 11:26:56AM +0200, stoffell wrote:
On 4/15/06, Min Hwan Chang [EMAIL PROTECTED] wrote:
wondering if its stable enough to use. Currently I'm editing my own *.conf
Using it at multiple sites (ranging from 10-50 extensions).
scripts but it sure would be nice if there
John Novack wrote:
Damon Estep wrote:
There is some kind of issue with SIP transfer interaction between some
SIP phones and asterisk, I have personal experience with Polycom
phones not being able to do a blind xfer using the feature key.
Our receptionist does both blind and attended
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
chawki hammoud wrote:
With a few fairly minor programming revisions to the script this would
be possible. At present, ASTCC does not support that though.
Darren Wiebe
[EMAIL PROTECTED]
Hi users:
astcc script exits when dialing an uncomplete or
Eric ManxPower Wieling wrote:
John Novack wrote:
Damon Estep wrote:
There is some kind of issue with SIP transfer interaction between
some SIP phones and asterisk, I have personal experience with
Polycom phones not being able to do a blind xfer using the feature key.
Our
Hi John,
If you enter show application dial when logged into the Asterisk console,
you can read that help (extract only regarding dial option) :
L(x[:y][:z]) - Limit the call to 'x' ms. Play a warning when 'y' ms are
left. Repeat the warning every 'z' ms. The following special
What happens if you remove the r option? r is almost NEVER useful.
Steve Feinstein wrote:
I've been pulling my hair out over this one trying to understand it.
If you have a very simple extension:
exten = 1,n,Dial(IAX2/Steve|24|r)
Everything I've seen says this should tell the IAX phone (our
I know it's registering properly because i can use the phone for
internal/local/long distance calls... I suppose it could be a problem in my
dial plan.
I have the following
[default]
exten = _9011.,5,Dial,1,Goto(outgoing-call,${EXTEN:1},1)
[outgoing-call]
exten = _.,n,Dial(SNIPPED)
I
Shaun
I agree with you - I think your dial plan is the problem.
you are stripping off the initial 9 in 'default' thus passing '011xx
' etc to 'outgoing call'.
Outgoing call context needs a 1 in the first priority. the 'n' priority
only seems to work for subsequent steps in the dial
I'm really not interested to look back, but IIRC, when using just one
Polycom phone behind NAT we didn't have any problems, but when using
more than one behind the same NAT that is when problems started,
qualify=somethingbutno seemed to help it a bit, but didn't eliminate
the problem.
On 4/16/06,
FreePBX is good to get you started. Their dialplans are a good start, and it's nice to even load up AAH in a virtual machine and then play with to learn how to do certain things. You then look at the code generated, and you can copy that code into your own dialplan, or at least get a better
Vikram Rangnekar wrote:
+++ Doug Lytle [11/04/06 07:58 -0400]:
Vi
I hope you can reconsider your decision now.
I have,
I setup a test system with a copy of my configurations. Here are some
suggestions:
1.) Put together a installation manual, it kept searching the website,
only
Hi,
I have a Cisco 7941 phone running SIP, and for a variety of reasons [1] have
configured this in the short term to use my local Asterisk server to register
to, and then have my local Asterisk server in turn register to the upstream SIP
proxy at my voice provider.
Voicemail is supplied by my
On Thu, Apr 13, 2006 at 01:04:37PM -0400, Jon-o Addleman spake thusly:
I'm trying to set up a system so that I can record a conversation over
SIP. Monitor and the like don't work so well for me, because I need to
pipe the conversation to other programs in realtime, rather than record
to a
I want send from asterisk to fxo line (tdm400p) a number (232) and Flash Key
For example 232R (R in italy is flash key)
I wrote this configuration in extension.conf:
exten = 377,1,Dial(Zap/7/232)
exten = 377,2,Flash()
exten = 377,3,Hangup()
But this config don't work
Help me
Actually it makes no difference. I tried it in an attempt to get
something to happen.
Thanks,
-Steve
Eric ManxPower Wieling wrote:
What happens if you remove the r option? r is almost NEVER useful.
Steve Feinstein wrote:
I've been pulling my hair out over this one trying to understand it.
There are two approaches to get NAT working properly:
- Use UDP and send and receive from the same port. This is extremly
simple, however some phones do (by default) send and recieve from a
different ports. Then you have to tell explicity no no, dont do that;
use the same port. There are even
Thanks for the great replies, after taking the newest AAH2.8 for a spin I'm beginning to realize that running FreePBX will be well worth it. The system we have running in the office currently is limited by my horrible dialplan so having something autogenerated will be nice. Because we don't need a
Hi,
is it possible to remove the no timeout combo box in flash operator panel?
How can I reduce the flash area? I set small buttons and half of the
area is white and I want to resize it.
Comment transfer_timeout in op_server.cfg. To reduce the flash area
you will have to play with the .html
+++ Doug Lytle [16/04/06 18:23 -0400]:
Vikram Rangnekar wrote:
+++ Doug Lytle [11/04/06 07:58 -0400]:
Vi
I hope you can reconsider your decision now.
I have,
I setup a test system with a copy of my configurations. Here are some
suggestions:
1.) Put together a
Damon Estep wrote:
There is some kind of issue with SIP transfer interaction between some
SIP phones and asterisk, I have personal experience with Polycom phones
not being able to do a blind xfer using the feature key.
We have to use the asterisk # blind xfrer functionality for blind
transfers
On Fri, 14 Apr 2006 23:48:31 -0300, Paulo Scardine wrote
I use Unicall and app_rxfax to receive (works very well) and
iaxmodem+hylafax for sending (took a little longer than I would
expect, like 50%).
Just use a recent iaxmodem (make it static). I use a libspandsp snapshot.
Did you
On 17/04/06, Vikram Rangnekar [EMAIL PROTECTED] wrote:
you can fix issue number 3 by running the install script
sh ./install.sh
or manually running the command
touch /var/log/asterisk/druid
chmod 777 /var/log/asterisk/druid
You'll have difficuly persuading any professional unix admin that
60 matches
Mail list logo
