-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Guenther Boelter
I have 3 Grandstream Budge-Tone 100 with Firmware
1.0.7.11beta, and they are working very well since more then
4 month now.
I'm using two Grandstream Budgetone 101 without
Hi,
I need to connect two sites and two Asterisk servers sharing their dialplan.
In fact users usually can be moved at different offices and carry their
phone number.
What's the best way to do this?
- switch statement
- DUNDI
?
Thanks for any help
Mimmus
Hello,
I agree with Damon's comments below.
Just for information. Eicon do have the Diva Server Analog range of cards that
will work with asterisk. You can plug these into Analog lines and then use them
with Asterisk via the CAPI interface of the Diva Server driver.
If you have CLIP (The
I'm going to try,
Thanks very much
Giordano
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Remco Barende
Inviato: lunedì 30 gennaio 2006 20.04
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [Asterisk-Users] Kirk IP600
Hi!
Any suggestions on how to go about
this?
so person calls, recording: "press2 to call
cell phone", user presses 2, call forwards to my cell phone.
Thank you
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Asterisk-Users mailing list
To
Try setting the Callerpresentation to something else:
http://www.voip-info.org/wiki/page_history.php?page_id=1682preview=2
SetCallerPres(prohib) actually worked! Thanks!
Regards,
Jan
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Hi,
I'm looking for an interface card for termination of Euro-ISDN2 (BRI)
lines.
That is ISDN lines from the telco into my Asterisk box.
Any recommendations, good/bad expiriences ?
At present I'm looking at cards from BeroNet and Junghanns.
Cheers,
John
Faroese Telecom
Hello,
Call sip:[EMAIL PROTECTED]
Regards
harry
--- Jimmy Smith [EMAIL PROTECTED] a écrit :
anyone having weird problems on latest cdrtool?
#!/usr/bin/php4
*Fatal error*: Class
webservice_ngnprocdrtool_ngnprocdrtool: Cannot
inherit
from undefined class soap_client in
execute the dial command from AGI.
e.g.
exec(dial(SIP/provider/2394892348))
you may want to reset or fork the cdr so you can have the record for
the IVR interaction and a different record for the call you are
connecting.
See ForkCDR and ResetCDR
hope this helps,
Cristi
On 1/31/06, [EMAIL
Look here for the updated firmware: http://www.grandstream.com/BETATEST/
Don't ask me why, but you really have to use capital-letters for the word
BETATEST!!
If you are interested in 1.0.7.11beta, i can gsend you a copy via email
because it's not on the server anymore.
Guenther
Mimmus
On 31 Jan 2006, at 09:12, John Jensen wrote:
Hi,
I'm looking for an interface card for termination of Euro-ISDN2 (BRI)
lines.
That is ISDN lines from the telco into my Asterisk box.
Any recommendations, good/bad expiriences ?
At present I'm looking at cards from BeroNet and Junghanns.
I'm
Hello,
Is there an app_snmp for asterisk-1.2.3 ?
Harry
___
Nouveau : téléphonez moins cher avec Yahoo! Messenger
! Découvez les tarifs exceptionnels pour appeler la
France et
Hi all,
look at these lines.
I created a queue named info when a caller (extension
86) place a call he is put on queue he sould hear MOH
.
What's the meaning of :
Jan 29 14:35:30 WARNING[2591]: file.c:509
ast_openstream_full: File 100 does not exist in any
format
Jan 29 14:35:30 WARNING[2591]:
I installed the chan_sccp and configured the sccp.conf, but when try to start
asterisk I get this error
[chan_sccp.so]Jan 31 10:31:15 WARNING[19727]: loader.c:258 ast_load_resource:
/usr/lib/asterisk/modules/chan_sccp.so: undefined symbol: ast_park_call
Jan 31 10:31:15 WARNING[19727]:
HI,
all newer HFC-S cards will do. Depending on your application and system,
you could easily ebaying an used Fritz!Card PCI or some active AVM B1
controller. Depending on the card you want to use you must se ZAPHFC or
mIISDN/chan_isdn or chan_capi or mixtures with 2 different cards ...
+++ Strain Jer [30/01/06 01:29 +]:
I was searching thru the internet and I found a wide variety of different
web interfaces for asterisks
I was curious which one is best suited for asterisks. Thanks
___
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PB == Phil Blundell [EMAIL PROTECTED] writes:
PB Right now I'm still using their Java thing, but it's slow enough
PB that one of these days I guess I'll crack and reimplement that
PB stuff directly in python. I think the algorithm is described on
PB the voip-info.org wiki someplace.
A trick
I'm looking at version.h installed by Asterisk 1.2.3/4 - and the default
value is 00. I thought the value should be 010200. I know many
people have problems compiling chan_bluetooth because of this
inconsistency. Anyone has the last word on this?
On Tue, 31 Jan 2006, Jens Vagelpohl wrote:
On 31 Jan 2006, at 09:12, John Jensen wrote:
Hi,
I'm looking for an interface card for termination of Euro-ISDN2 (BRI)
lines.
That is ISDN lines from the telco into my Asterisk box.
Any recommendations, good/bad expiriences ?
At
Hi,
is there a way to executing commands in the dialplan regardless which number is
dialed before
the pattern matching starts ?
when a call enters the first context it would be nice if i can set some
variable or manipulate
a callerid, or what ever before the patternmatching starts.
a solution
we are using the beronet cards together with mISDN, works stable
on system with digium and beronet we use bristuff
John Jensen wrote:
Hi,
I'm looking for an interface card for termination of Euro-ISDN2 (BRI)
lines.
That is ISDN lines from the telco into my Asterisk box.
Any recommendations,
On 31 Jan 2006, at 10:06, Armin Schindler wrote:
I'm very happy with an Eicon Diva Server V-BRI that I bought a
couple months
ago. The only drawback is that it doesn't do any fax traffic
apparently. It
works with chan_capi-cm from Sourceforge.
The 'V' version of that card is for (V)oice.
On Monday 30 January 2006 21:48, [EMAIL PROTECTED] wrote:
On Mon, 30 Jan 2006, Dmitry Ivanov wrote:
I have created dynamic CGI-like TFTP server so I will create
config files on-the-fly. Now we use this system (dynamic tftp
server and Perl CGI script) for country-wide Sipura 3000
Dave Morrow a écrit :
Hi all. I am trying to find out what the most popular soft phone for
Windows is for use with Asterisk. SIP or IAX?
If you have the choice, go with IAX. I'm using IaxComm and Diax. They
work great, Diax is multi language, IaxComm works Windows and Linux, no
FW issues,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Guenther Boelter
Look here for the updated firmware:
http://www.grandstream.com/BETATEST/
Don't ask me why, but you really have to use capital-letters
for the word BETATEST!!
If you are
Can anyone explain me differences among:
- chan_capi (and chan_capi-cm)
- bristuff
- mISDN
?
Thanks
Mimmus
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kib Eki
Sent: Tuesday, January 31, 2006 11:12 AM
we are using the beronet cards together
From the usual place, http://www.grandstream.com/BETATEST/GXP2000/
Note, there are two (and it took me a bit of a while to figure out)
images to be loaded. Copy the ...a.bin's and the .bin's to your http
provisioning directory, and reboot. The phone _must_ load the .bin
files before it
Hello all,
Just a question, on asterisk box :
I looking on the web , for asterisk at large , and 'asterisk future of
telephonie' ...
If we would like to change our OLD PABX 600 phone with 4 E1, to install a
asterisk with full ip phone in SIP, Could we use 1 Box for asterisk with
voicemail,
Hi All,
I
would be happy if anyone can tell me how does asterisk interact with the
telephony boards.what files or APIs are used by it to interact with
them.
thanks and regards
krishna
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Hi all,
I'm experiencing a problem with meetme i can't resolve.
This is my scenario:
A iax client, say IaxComm, make a call through a zap channel. When it
answers it is tranfered to a conference room.
Then the iax client make a second call though a second zap channel, at
the other side there
A user has set in his phone to transfer each call to another number. Is
it possible to configure Asterisk not to transfer the calls? Or is it
only phone setting?
--
Best regards,
Bartosz Piec
___
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On Tue, 31 Jan 2006, Mimmus wrote:
Can anyone explain me differences among:
- chan_capi (and chan_capi-cm)
If your card and its driver support a CAPI 2.0 interface,
you should use chan_capi-cm.
Eicon DIVA Server, AVM and some other which I don't know.
- bristuff
I'm not the expert here, but
Fabrice a écrit :
Hello all,
Just a question, on asterisk box :
I looking on the web , for asterisk at large , and 'asterisk future of
telephonie' ...
If we would like to change our OLD PABX 600 phone with 4 E1, to install a
asterisk with full ip phone in SIP, Could we use 1 Box for
On Tue, Jan 31, 2006 at 11:24:54AM +0100, Vikram Rangnekar wrote:
+++ Strain Jer [30/01/06 01:29 +]:
I was searching thru the internet and I found a wide variety of different
web interfaces for asterisks
I was curious which one is best suited for asterisks. Thanks
Check out
Hi,
i have diffirent provider example(3 single account in deltathree, 4
account in packet8 and so on) . How this possible to make the three
individual sip account in deltathree act as trunk so that i cannot get a
busy call. If line one fail goto line 2 then line 3 or another trunk
line 1
Can someone tell me the advantage in using an active card such as the AVM-B1
do they have echo cancelling built in?
Just that I've got three pots lines and keep thinking I should convert over
to ISDN but I don't want to get echo issues.
Chris
- Original Message -
From: Armin
Jolly M. Recto wrote:
Hi,
i have diffirent provider example(3 single account in deltathree, 4
account in packet8 and so on) . How this possible to make the three
individual sip account in deltathree act as trunk so that i cannot get
a busy call. If line one fail goto line 2 then line 3 or
Hi all, I am resending this message, so far no one has helped me with this incoming call issue. there is no problem with outbound call but there is no inbound call to my sip phone. the only message I get when I call from pstn is "unable to create local channel for call forward to 'Local/[EMAIL
On 1/31/06, John Jensen [EMAIL PROTECTED] wrote:
Any recommendations, good/bad expiriences ?
At present I'm looking at cards from BeroNet and Junghanns.
Only have experience with junghanns cards, but they are the same..
beronet doesn't use bristuff.. but you can also use junghanns cards
the
On 1/31/06, Giordano Grandis [EMAIL PROTECTED] wrote:
[chan_sccp.so]Jan 31 10:31:15 WARNING[19727]: loader.c:258
ast_load_resource: /usr/lib/asterisk/modules/chan_sccp.so: undefined symbol:
ast_park_call
Jan 31 10:31:15 WARNING[19727]: loader.c:391 load_modules: Loading module
Hello,
i've recently switched over from 1.0.9 to 1.2.3.
I've experienced some (to me) weird behaviour.
This is the config for an example queue.conf:
[654]
wrapuptime=30
timeout=20
strategy=ringall
retry=5
queue-youarenext=queue-youarenext
queue-thereare=queue-thereare
Hi
how about SIP friend to SIP Friend
even it taking gsm
ram
On 1/31/06, JP Carballo [EMAIL PROTECTED] wrote:
ram wrote: Hi as per the list people guidence i have downloaded the Codec and installe
my Pc is P4, but i have downloaded the P2.so file and copied in specific directory whe i see
Giordano Grandis ha scritto:
I installed the chan_sccp and configured the sccp.conf, but when try to start asterisk I get this error
[chan_sccp.so]Jan 31 10:31:15 WARNING[19727]: loader.c:258 ast_load_resource:
/usr/lib/asterisk/modules/chan_sccp.so: undefined symbol: ast_park_call
Jan 31
John Jensen schrieb:
I'm looking for an interface card for termination of Euro-ISDN2 (BRI)
lines.
That is ISDN lines from the telco into my Asterisk box.
Any recommendations, good/bad expiriences ?
At present I'm looking at cards from BeroNet and Junghanns.
How many lines do you want to
does it registers well?
although i think you have to add context=default to the stargate1 section.
try that and see what happens.
2006/1/31, abc def [EMAIL PROTECTED]:
Hi all, I am resending this message, so far no one has helped me with this
incoming call issue. there is no problem with
Are you using a SIP Softphone or an ATA?
2006/1/31, Facundo Ameal [EMAIL PROTECTED]:
does it registers well?
although i think you have to add context=default to the stargate1 section.
try that and see what happens.
2006/1/31, abc def [EMAIL PROTECTED]:
Hi all, I am resending this message,
On Tue, 31 Jan 2006, Chris Stenton wrote:
Can someone tell me the advantage in using an active card such as the AVM-B1
do they have echo cancelling built in?
Just that I've got three pots lines and keep thinking I should convert over to
ISDN but I don't want to get echo issues.
The active
I'm looking for an interface card for termination of Euro-ISDN2 (BRI)
lines.
That is ISDN lines from the telco into my Asterisk box.
Any recommendations, good/bad expiriences ?
At present I'm looking at cards from BeroNet and Junghanns.
How many lines do you want to terminate?
Two to
I have a digium TDM400P with 1 FXO and 1 FXS port. I have a standard analog
phone connected to the FXO port and place calls to the PTSN phone line. The
analog trunk is accessed via the standard 9 and area code (if needed) and
of course the phone number. The error is as follows. I dial
Hello friends,
I am using asterisk with sip phones and sip fxo box. My problem is that my
dtmf is recognised internally only if I use dtmf=inband and outside to the pstn
lines work only if I use dtmf=info. The result is that I cant transfer any
calls from and to pstn. How do I fix this.
Hi all, Some friends of mine have an asterisk box which they use for
outgoing IAX2 via voipbuster.com.
They have been told that they now have an incoming number 0044117***
The thing is I cant seem to get any debug info on the incoming.
I have tried both sip and IAX trunks but dont see any
1. I want to call somebody and, as soon (and not before) a playback
should be played. How can I do that?
2. How can I accept dtmf tones with such calls?
Example:
System calls all staff and ask them a question. The staff will answer
with a digit!
The playback should start when the staff picks
What version of the firmware?
Jerry Glomph Black wrote:
Have just done a deployment of 45 of these puppies.
They are doing their main job quite well, but of course there are minor
kinks.
A not-so-minor one is that if one attempts to plug a PC into the 2nd
RJ-45 jack, as soon as you send
Hi!
When adjusting the rxgainand txgain inAsterisk 1.2.1 do I need to restart Asterisk or is it enough to just reload Asterisk in order to apply the new setting?
-- Morten Isaksenhttp://www.misak.dk/blog/
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Using the Buddy Watch functionality on the IP601 you can watch up to
6 people. The expansion modules are not good for much more than speed
dials due to this limitation.
After talking to our vendor, the reason it is limited to 6 is due to the
current version of Asterisk Business Edition's
Just curious,
I have had issues with the number of monitored phones and getting out
of sync with reloads. Have you had similar issues? Which version of *?
On Jan 30, 2006, at 7:04 PM, Saul Diaz wrote:
Damon Estep wrote:
Anyone successfully set up one of the polycom soundpoint ip
ram wrote:
Hi
how about SIP friend to SIP Friend
even it taking gsm
ram
Check the [general] section of your sip.conf
Most likely there is an allow=gsm line there.
Just allow=ulaw on your end so you can connect to voipjet.
--
JP Carballo
http://www.netfone2x.com
Bringing the world
my setup is
client--registers-- ser-redirect---client ---invite-- asterisk -- pstn
when this happens
i configured the ser.cfg with the rewriteuri and redirect logic and i
am seeing 300 redirect being passed to the client registerd to ser but
when it sends a invite to asterisk,
with incoming lines only maybe are active capi dual/quad-port cards from
AVM an alternative - but I've no experience with them together with
asterisk/chan_capi
an other way with 4 isdn-lines is to think about to order an partial E1
line with 8 channels...
[EMAIL PROTECTED] wrote on 31.01.2006
When adjusting the rxgain and txgain in Asterisk 1.2.1 do I need to restart
Asterisk or
is it enough to just reload
Asterisk in order to apply the new setting?
Need to stop asterisk and restart it. A reload will not take the new setting
into consideration. There is no need to stop/start
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jerry Glomph Black
Sent: Monday, January 30, 2006 11:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Polycom IP301: Pass-through ethernet
Hi
yes i have added all of them in allow
one by one like this
allow=g729
allow=gsm
allow=ulaw
allow=alaw
ram
On 1/31/06, JP Carballo [EMAIL PROTECTED] wrote:
ram wrote: Hi how about SIP friend to SIP Friend even it taking gsm
ramCheck the [general] section of your sip.confMost likely there
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Damon Estep
Sent: Tuesday, January 31, 2006 1:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] dialing 2 channels at the
On 1/31/06, Rich Adamson [EMAIL PROTECTED] wrote:
When adjusting the rxgain and txgain in Asterisk 1.2.1 do I need to restart Asterisk oris it enough to just reload
Asterisk in order to apply the new setting?Need to stop asterisk and restart it. A reload will not take the new settinginto
Yes you must prefix a variabel with __ that's (2) _ underscores so
that
it cross channels.
Aah, the magic formula - documented where? :)
Thanks a million, have a great day.
Damon
___
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I have Asterisk 1.2 and a generic Wildcard single FXO card (a cheapo from
eBay). I have read about many people who have used these cards without an
issue and I'm just testing to work up a new system.
The problem I have is that if I call the telephone number of the line attached
to that card
On Tue, Jan 31, 2006 at 08:44:18AM -0600, Rich Adamson wrote:
When adjusting the rxgain and txgain in Asterisk 1.2.1 do I need to restart
Asterisk or
is it enough to just reload
Asterisk in order to apply the new setting?
Need to stop asterisk and restart it. A reload will not
I have a Polycom IP501 phone and have set it up to download the config from an
FTP server, it did this once and now is in an endless loop of trying to contact
the FTP server, failing, then rebooting.
When I watch the FTP server logs it looks like the phone starts a session, ends
it, starts it,
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Tuesday, January 31, 2006 8:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] dialing 2 channels
These were configured with the MACADDR-directory.xml and the 6
extension limitation has been verified by several vendors.
Don't get me wrong, they are a nice looking unit, and once the
monitoring of more than 6 people is available they will be a great
replacement for the Snoms.
Right now we
well. Im supposing you mean a SIP phone. Transfers with SIP phones
happens to be a method called REFERRER. Im not sure if its a feature of
Asterisk to allow the administrator to ban the referrers, but if is not
a feature, letme know, may be i can make a patch soon.
To look for a feature like
please consider posting this as a Job offer in asteriskhelpdesk,
because of your lack of information i can tell you are really stuck :DOn 1/30/06, Naren Koka [EMAIL PROTECTED]
wrote:I need to configure / migrate Asterisk server from 0.9 to the latest
version with some upgrades. Please help!Thank
Need to stop asterisk and restart it. A reload will not take the new
setting
into consideration. There is no need to stop/start the zaptel drivers,
just
asterisk itself.
OK.
If I set the gain to a negative number then i decrease the volume? And a
positive number
When adjusting the rxgain and txgain in Asterisk 1.2.1 do I need to
restart Asterisk
or
is it enough to just reload
Asterisk in order to apply the new setting?
Need to stop asterisk and restart it. A reload will not take the new setting
into consideration. There is no
Sorry for the duplicate post but I have hit a brick wall trying to get
this to work. Is there anyone who can help me?
I am having trouble trying to register with a voip
provider using sip. I am able to connect using xlite softphone. in
xlite i use
domain/realm: providerdomain.com
sip
Don't feal bad about not reading. I yell at my 10 y.o. about it all the
time. READ, NO more TV, READ!!!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Damon Estep
Sent: Tuesday, January 31, 2006 10:23 AM
To: Asterisk Users Mailing List -
If I do a supervised forward on a call (Polycom 501, Asterisk 1.2.1), all
goes well until the second time I hit forward (to join the caller with the
extension); then, the caller's MoH goes away (making them think they've
been hung up on), and the server spits out:
asterisk-cw*CLI
-- SIP read from
Sounds like the phone cannot log into the FTP server. Did you create the proper user with the correct login? It's set up in the FTP/TFTP menu.Also, you can end the loop by just going into the config menu and nuking the FTP info and then you'll get a message that says it could not contact the boot
On Tue, Jan 31, 2006 at 08:18:34AM -0700, [EMAIL PROTECTED] said:
I have a Polycom IP501 phone and have set it up to download the config from
an FTP server, it did this once and now is in an endless loop of trying to
contact the FTP server, failing, then rebooting.
When I watch the FTP
Sharon a écrit :
my setup is
client--registers-- ser-redirect---client ---invite--
asterisk -- pstn
when this happens
i configured the ser.cfg with the rewriteuri and redirect logic and i
am seeing 300 redirect being passed to the client registerd to ser but
when it sends
i understand.. anyone know how much is basic support from them ?
On 1/31/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hello,Call sip:[EMAIL PROTECTED]
Regardsharry--- Jimmy Smith [EMAIL PROTECTED] a écrit : anyone having weird problems on latest cdrtool? #!/usr/bin/php4
*Fatal error*: Class
Here is my problem, at this point the IVR doesn't hear the dtmf sended
by the iax client, even if it can hear the dtmf sended by the first zap
channel.
I donot know if IaxComm has inband dtmf mode available, if so enable
it and see if it works.
___
When a call arrives on our PRI from a UK domestic number, the presented
caller ID looks something like 1223123456. In my dialplan, I stick
90 on the front in order to turn this into a valid number for outward
dialling, and everything works fine.
However, when a call comes in from an
Hi,
I have a running asterisk 1.2.1 (bristuffed) with a TDM400 Board. Connected
to 1 port is am ordanary Fax Machine. Everything 'seems' to work,
however receiving faxes is very unreliable.
Sometimes I receive a normal page, without problems. Sometimes
half of a page and the rest is scrambled,
Ok anyone have latest cdrtool running 4.1 i think..
ill pay for install
On 1/31/06, Jimmy Smith [EMAIL PROTECTED] wrote:
i understand.. anyone know how much is basic support from them ?
On 1/31/06, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hello,Call
sip:[EMAIL PROTECTED]
Regardsharry---
Hi,
I`ve been trying to
figure out voicemail, but there is something that is obviously escaping me.
Using * 1.2.3, standard built with asterisk-addons.
I have two
voicemails, one is 702 and one is 705. Both in different contexts, but
that doesn`t matter (I think). The point is in the
Ronald,
I've been experimenting with something similar. You might want to check
this out:
http://www.voip-info.org/wiki-Asterisk+auto-dial+out+deliver+message
What kind of trunks do you have for your outbound calls?
(BRI/PRI/analog POTS/SIP/IAX etc.) I'm using PRI and it works very well
- the
FYI I just tested on * 1.2.1 a reload chan_zap.so
It takes the new settings from zapata.conf.
I know because I changed the context and after a reload it showed the new
context.
I can only assume that the gain settings are also changed.
Regards Allan Gee
Phone: +27 21 4644400 Ext. 103
i have ser and asterisk on 2 different boxes.
my ser.cfg
if (method==REGISTER) {
if(!www_authorize(ser domain name, subscriber)){
www_challenge(ser domain name, 0);
break;
}
sl_send_reply(200, ok);
break;
};
rewritehostport (ip addr of asterisk box:5060);
sl_send_reply (300,
I had this same issue with 601s, and I was able to fix it by defining:
progressinband=yes in sip.conf.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken D'Ambrosio
Sent: Tuesday, January 31, 2006 11:20 AM
To:
I had the /exact/ same problem. Turns out it's the FTP server; in the
docs, there are several FTP servers specified as being compatible;
proftp is the one I went with, and it fixed it right up. (Note that I
was using the default Debian FTP server when it was rebooting, so it's
not just a 'doze
[EMAIL PROTECTED] is believed to have said:
The active cards do the ISDN protocol stuff on board, so the host CPU/driver
does not need to do that - better performance, less interrupts.
The AVM cards do not have such DSPs on board, so no echo-cancel.
But the Eicon DIVA Server cards do. They do
Alex Ongena wrote:
I have a running asterisk 1.2.1 (bristuffed) with a TDM400 Board. Connected
to 1 port is am ordanary Fax Machine. Everything 'seems' to work,
however receiving faxes is very unreliable.
Anyone have any idea what's causing this or how to debug it? I'm pretty
sure the root cause is with chan_sccp.so, but not sure how to prove it.
I recently upgraded from CVS-head to 1.2.1 and the chan_sccp from
12-17-2005. Now, once or twice a week, I get this on the console:
Jan 31
All we have a deal on Canadian termination.
Rate: $0.0039 US Dollars
Billing: 1/1
Protocol: SIP or H323
Codec: G729
Terms: Prepaid Only.
We have a real-time web interface where you can
monitor or download your CDR's.
Please e-mail me offlist if you are interested: [EMAIL PROTECTED]
To monitor who is doing what we writing a program that
every user can have on thier windows desktop to see
the status of all phones on the system. It's AIM
style. Has several groups. On the phone, off,
Available, Away etc.
Managers can scroll the mouse over the user and see
what call they are on
Generaly you get what you pay for (with very few
exceptions such as asterisk). Also as far as a web
interface goes its really one that you get used to and
like. There are lots out there. You goto find one that
works for you.
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Jan 31, 2006 at
A)When you say stop asterisk from transfering the call
what do you mean ? oNot to send it to VM if the user
is away ?
B)I think it depends on the phone. I know with the
Polycoms you can program it directly in to the phone.
(Done it in the past).
--- Bartosz Piec [EMAIL PROTECTED] wrote:
A user
i've tested it with this config files and i worked:
extensions.conf
exten = 55,1,Dial(SIP/2271,20)
sip.conf
[2271]
type=friend
host=dynamic
secret=sip
allow=all
qualify=200
nat=no
Instead of 2271 you can put whatever you want.
good luck.
2006/1/31, Facundo Ameal [EMAIL PROTECTED]:
Are
Totally uneducated guess: If your version has the _expression_ parser, it has the leak.
On 1/30/06, Damon Estep [EMAIL PROTECTED] wrote:
Does anyone know what date this memory leak was introduced and/or how tocheck source code for it?
I am running a pre-1.2 CVS head version and would like to know
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