HI
This is my setup of the features.conf but it had not any reaction after I
pushed the *2 while calling was acting ! Could you tell me the reason ? Or
give my the method of the setting.
Thanks!
LARRY
[general]
parkext = 700
parkpos =
Hi,
I've just followed
http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talkinstructions
from wiki,
And i always get my jabber (GoogleTalk account for asterisk server) not
registred:
hi,
i am recieving deadlocks frequently and its calls are getting hanged .
Aug 11 13:13:53 WARNING[6367] channel.c: Avoided initial deadlock for
'0xb6692258', 10 retries!
Aug 11 13:13:54 WARNING[6367] channel.c: Avoided initial deadlock for
'0xb6692258', 10 retries!
Aug 11 13:13:54 WARNING[6367]
Hello !
I am wondering if there is a possibility that Asterisk will be present
on Linuxtag booth in Berlin 2009, and if there will be an option to
take a DCAP exam ?
Kind regards,
Jan Prunk
--
Jan Prunk janprunk AT SPAMFREE gmail DOT com
Website: http://www.prunk.si PGP key: 00E80E86
Hi
One of my softphones is supposed to support g711 , however I am getting
these errors and a 404 not found when I try to make a call from it. However
on xlite it works fine using g711.
Below is the log of the phone that is not working.
Content-Type: application/sdp
Content-Length: 1123
P-hint:
Hi,
I was switching from zaptel to dahdi and got latest SVN from everything.
Compiling works fine.
kernel module dahdi_dummy is loaded.
/dev/dahdi/pseudo exists
Trying to go into a meetme does not work:
[Aug 11 14:04:45] -- Executing [EMAIL PROTECTED]:1]
MeetMe(SCCP/6000-0001, 444|dcIM)
D.J.Sateesh [EMAIL PROTECTED] writes:
hi,
i am recieving deadlocks frequently and its calls are getting hanged .
Aug 11 13:13:53 WARNING[6367] channel.c: Avoided initial deadlock for
'0xb6692258', 10 retries!
We upgrade customers who hit that bug to 1.4... The locking is greatly
improved.
Here's my attempt to explain a quick way of doing an auto dialer with
Scala and the Asterisk-Java library:
http://blogs.reucon.com/asterisk-java/2008/08/10/outbound_message_delive
ry_using_agi_and_ami_in_scala.html
Cheers,
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and
Stefan Gofferje wrote:
Trying to go into a meetme does not work:
[Aug 11 14:04:45] -- Executing [EMAIL PROTECTED]:1]
MeetMe(SCCP/6000-0001, 444|dcIM) in new stack
[Aug 11 14:04:45] WARNING[4184]: app_meetme.c:775 build_conf: Unable to
open pseudo device
[Aug 11 14:04:45] --
Hi,
I'm running asterisk realtime, i had prob when a user does not
unregister properly.
I tested with SPA942 and a PAP2, when phone is registered, i call using
the SPA using x-lite no problem, but when i unplugged the power, it does
not unregister properly, so asterisk think SPA942 is still
On Mon, 2008-08-11 at 11:20 +0200, Jan Prunk wrote:
I am wondering if there is a possibility that Asterisk will be present
on Linuxtag booth in Berlin 2009, and if there will be an option to
take a DCAP exam ?
While Digium typically sends a few people to present at the Linuxtag
show every
Hi Larry -
This is my setup of the features.conf but it had not any reaction after I
pushed the *2 while calling was acting ! Could you tell me the reason ? Or
give my the method of the setting.
Thanks!
LARRY
[general]
parkext =
Kevin P. Fleming schrieb:
Fixed in revision 137188; this module apparently did not get any DAHDI
conversion work at all, but I don't know how it got missed. Thanks for
the testing!
Confirmed. Works fine now under all (extensively) tested conditions.
Terve,
Stefan
--
Last words of a
Hi all,
I am using outgoing call files to place calls. Issue is when that call
is BUSY I dont get the correct DIALSTATUS
from that call when running my AGI and the failed extension.
WHERE can I make a change in the code so that the DIALSTATUS when the
call ended can be
added as a variable in
If a phone is unplugged, it's not likely to have time to send
notification of this to Asterisk before it powers off. There's nothing
you can add to your dialplan to overcome this, however you *can* set the
qualify parameter within sip.conf (or it's equivalent realtime table)
to overcome this.
Works ok on my side. Any debug messages from your console you could post here?
Thanks,
Philippe
On Mon, Aug 11, 2008 at 9:16 AM, vivek rastogi [EMAIL PROTECTED] wrote:
Hi,
I've just followed
http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talkinstructions
from wiki,
And i
Thank you for your reply sir. I tried setting qualify=yes my CPU spiked
to 113%
i continuously see this on my CLI
Aug 11 23:31:56] NOTICE[15207]: chan_sip.c:12669
handle_response_peerpoke: Peer '118555' is now Reachable. (388ms / 2000ms)
[Aug 11 23:31:56] NOTICE[15207]: chan_sip.c:12669
Benny Amorsen wrote:
D.J.Sateesh [EMAIL PROTECTED] writes:
hi,
i am recieving deadlocks frequently and its calls are getting hanged .
Aug 11 13:13:53 WARNING[6367] channel.c: Avoided initial deadlock for
'0xb6692258', 10 retries!
That is actually more of a debug message, and is not
Stefan Gofferje wrote:
[Aug 8 23:30:13] SSL certificate ok
[Aug 8 23:30:13] == Problem setting up ssl connection:
error::lib(0):func(0):reason(0)
[Aug 8 23:30:13] WARNING[23835]: tcptls.c:463 ast_make_file_from_fd:
FILE * open failed!
First, try the latest code in the Asterisk
Stefan Gofferje wrote:
I have configured all IAX clients with encryption. I use Zoiper as a
softphone. When I make a call in the LAN from desktop-PC to *, the call
is - according to wireshark not encrypted. Wireshark identifies the
packets as normal G.711 mu-law packets. However, * reports the
Hi all,
I'm writing an application to Queue and Manage AMI Originate actions.
Basically, callfiles on steroids if you may :)
I'm facing the following challenges, and any ideas or pointers will be
hugely appreciated.
1. When I successfully Queue an Originate... (Response is Success), how
will
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED],
altrimenti vi risponderò al mio rientro.
Dimitri Osler
I
Hi all,
I have an interesting problem that I am looking for a solution for. I
want to be able to call into an asterisk server and have what I say be
broatcast over a streaming web radio station. I imagine using
something like icecast for that. Does anybody have any pointers on how
to get started?
Off the top of my head... you could probably route the audio of a
softphone (like Zoiper/X-Lite) to something like Nicecast (Mac) or
Icecast.
On 11 Aug, 2008, at 9:21 AM, Andrew Niemantsverdriet wrote:
Hi all,
I have an interesting problem that I am looking for a solution for. I
want to
Hi,
The only reliable solution I've found for this is to set a custom
variable with the Originate action and query new channels for that
variable when they appear.
We've also used this strategy successfully when implementing
Asterisk-Java's live API.
Depending on which language you are going to
Essien Ita Essien wrote:
Hi all,
I'm writing an application to Queue and Manage AMI Originate actions.
Basically, callfiles on steroids if you may :)
I'm facing the following challenges, and any ideas or pointers will be
hugely appreciated.
1. When I successfully Queue an Originate...
Russell Bryant schrieb:
You'd have to provide a packet capture to see exactly what is happening.
It sounds like on the call leg between your client and Asterisk, it
isn't offering encryption as a capability, so it doesn't get used.
However, when your friend calls you, and Asterisk makes
This is how it sounds:
http://stefan.gofferje.net/chan_mobile_distorted.wav
Terve,
Stefan
--
Last words of a stormchaser:
Where is that rotation on the radar?!
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008
If you want inbound speech then you can use one of the web based
softphones (mexuar etc), just through them into a conference room with
all but the 'moderators' into a silenced one way conference room and use
DTMF to raise hands etc.
But for bandwidth capacity issues you can use something as
SIP wrote:
When calling from our SIP proxy through Asterisk to the PSTN provider,
we support reINVITES which tend to work seamlessly.
However, when creating a call file which essentially connects a call
from the SIP proxy to the SIP proxy, Asterisk wants to stay in the RTP
media path. I
Hi,
Are there Asterisk CLI commands I can use to add/manage extensions and dial
plans when using Asterisk Realtime with MySQL? I know about the database
put and database get commands, but from what I've read they apply to
AstDB and not to the Asterisk Realtime database. So far I've been
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED],
altrimenti vi risponderò al mio rientro.
Dimitri Osler
I
Michael Graves schrieb:
Which Asterisk systems provide automatic provisioning of phones?
Gemeinschaft ( http://www.amooma.de/gemeinschaft/ ) does.
Grüße,
Philipp Kempgen
--
http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied -
Russell Bryant schrieb:
Interesting. Here are a couple more sanity checks you can do. First,
double check to ensure that your entry in iax.conf has encryption=yes
set. Also, when you make the call into Asterisk, set the verbose
setting up a bit. You should see output from chan_iax2
On Monday 11 August 2008 12:07:47 J.M. wrote:
Are there Asterisk CLI commands I can use to add/manage extensions and dial
plans when using Asterisk Realtime with MySQL? I know about the database
put and database get commands, but from what I've read they apply to
AstDB and not to the Asterisk
Hi all
Back in the 1.2 days I think, there were some discussions about how two
asterisk
servers can share channel variables through an IAX protocol.
I don't see anything in 1.4 at least to be able to make it done.
Thanks
___
-- Bandwidth and
Robor Oghene schrieb:
Please let someone throw more light on this command and it usage.. i
tried a search but can't to get anything useful.
asterisk -rx 'manager show command Originate'
Grüße,
Philipp Kempgen
--
http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com
Ruddy Gbaguidi wrote:
Hi all
Back in the 1.2 days I think, there were some discussions about how two
asterisk
servers can share channel variables through an IAX protocol.
I don't see anything in 1.4 at least to be able to make it done.
Thanks
Back in 1.2 you had to use type 'friend' to
I am thinking about a change to our company's phone layout and would like
to get comments from people who have done something similar.
Currently, we have 3 locations - each with their own Asterisk PBX. The
corporate office has a PRI. Each remote location has a SIP provider for
5 channels of SIP
Hi list
I got one HP ProLiant DL380 G5 - Quad-Core Xeon E5440 2.83 with 4 gig RAM
I install Centos 5.2 64 bit and it is rumming pretty well and I need to use it
as voice
conferencing application (Meetme) server for high number of users fit to 8 E1
links
(240 users ) with echo
It doesn't seems to be working ...
What I wanted to do is on the first server, Set a channel variable...
then dial the number.
When I received the call on the remote server, use that variable ...
Is it possible ?
Richard Lyman wrote:
Ruddy Gbaguidi wrote:
Hi all
Back in the 1.2 days I
aymen warfalli wrote:
Hi list
I got one *HP* ProLiant *DL380 G5* - *Quad*-*Core* Xeon E5440 2.83
with 4 gig *RAM*
I install Centos 5.2 64 bit and it is rumming pretty well and I need
to use it as voice
conferencing application (Meetme) server for high number of users fit
to 8 E1
TP'n to follow broken flow.
As i stated, you must use a 'user to user' (friend) as the iax2_user
structure has struct ast_variable *vars, the iax2_peer (and
iax2_trunk_peer) do NOT.
Therefore you cannot pass *channel variables* when using peer-user
setups, only user-user setups.
Which means,
Erm, I've been out of the loop, but in 1.6 there's
the IAXVAR dialplan function that does _exactly_ what you want.
I don't know if it's been backported to 1.4, but I think there was a
patch
at one point.
Tim.
On 11 Aug 2008, at 20:43, Richard Lyman wrote:
TP'n to follow broken flow.
As i
My phone rings once and stops before playing message; how to stop this
behavior.
Could it have something to do with this error:
channel_find_locked: Avoided initial deadlock for '0x81c04d0', 9 retries!
Here is the dial plan:
exten = s,1,Wait(2)
exten = s,2,Answer
exten =
On Mon, Aug 11, 2008 at 02:45:18PM -0400, aymen warfalli wrote:
Hi list
Hi. Please don't thread-jack.
I got one HP ProLiant DL380 G5 - Quad-Core Xeon E5440 2.83 with 4
gig RAM I install Centos 5.2 64 bit and it is rumming pretty well
and I need to use it as voice conferencing
Jerry Geis wrote:
Call files spawn a completely new channel that your AGI most likely
isn't going to be able to track. Since the call is a completely new
channel, the DIALSTATUS variable for this channel will not be visible
to your original channel. You may want to look at using the
On 08/11/08 14:38, Joseph wrote:
My phone rings once and stops before playing message; how to stop this
behavior.
I think it has something to do with Linksys SPA 3201 with Setting under:
PSTN-To-VoIP Gateway.
PSTN-To-VoIP Gateway Enable: Yes
PSTN Ring Thru Line 1: Yes
PSTN Answer Delay: 3
On 8/11/08, aymen warfalli [EMAIL PROTECTED] wrote:
I got one HP ProLiant DL380 G5 - Quad-Core Xeon E5440 2.83 with 4 gig RAM
I install Centos 5.2 64 bit and it is rumming pretty well and I need to
use it as voice
conferencing application (Meetme) server for high number of users fit to 8
On 8/11/08, aymen warfalli [EMAIL PROTECTED] wrote:
I got one HP ProLiant DL380 G5 - Quad-Core Xeon E5440 2.83 with 4 gig RAM
I install Centos 5.2 64 bit and it is rumming pretty well and I need to
use it as voice
conferencing application (Meetme) server for high number of users fit to 8
I disabled logging of NOTIFY on the CLI and it does not show anymore,
however CPU is still very high, latency as well goes up when it is
trying to poke my phone here, my phone(SPA942) also keeps on rebooting
is there a way to increase the time of sending the qualify? TIA
regards
nhadie
Nhadie
Hi All,
I've been testing reliability with t.38 faxing pass through with * 1.4.21.1,
Linksys ATA's 2102 x 2, Sharp UX-B800SE and Cannon ImageClass D880.
cannon 2102 #1 SIP * SIP 2102 #2 sharp
Started out with default settings on all devices, configured Asterisk to
handle T.38 pass through, the
dear list
I'm very new in telephony and asterisk especial. so, it would be great
if somebody see at my dialplan. it works (except e1 which is untested at
this day), but I think it's not perfect. thanks
alexander
p.s. in Russia national prefix is 8 and international 810
Joseph wrote:
On 08/11/08 14:38, Joseph wrote:
My phone rings once and stops before playing message; how to stop this
behavior.
I think it has something to do with Linksys SPA 3201 with Setting under:
PSTN-To-VoIP Gateway.
PSTN-To-VoIP Gateway Enable: Yes
PSTN Ring Thru Line
Bill Andersen wrote:
I am thinking about a change to our company's phone layout and would like
to get comments from people who have done something similar.
Currently, we have 3 locations - each with their own Asterisk PBX. The
corporate office has a PRI. Each remote location has a SIP
On Aug 11, 2008, at 12:04 PM, SIP wrote:
SIP wrote:
When calling from our SIP proxy through Asterisk to the PSTN
provider,
we support reINVITES which tend to work seamlessly.
However, when creating a call file which essentially connects a call
from the SIP proxy to the SIP proxy,
On Aug 11, 2008, at 2:03 AM, larry wrote:
This is my setup of the features.conf but it had not any reaction
after I
pushed the *2 while calling was acting ! Could you tell me the
reason ? Or
give my the method of the setting.
Thanks!
On August 11, 2008 06:59:23 pm JR Richardson wrote:
So my question is this: Can I setup Asterisk to only allow t.38 pass
through from these ATA's, without the need to use the #99 in every dial
string from the fax machine?
Can you use disallow/allow with UDPTL? I'm not sure, I've never played
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED],
altrimenti vi risponderò al mio rientro.
Dimitri Osler
I
I configured parkcalls and can see on cli Playing digits/7 etc but I
cannot hear them in the phone.
I have:
features.conf.
[general]
parkext = 700
parkpos = 701-720
extensions.conf:
[extensions]
include = parkedcalls
What did I miss?
--
#Joseph
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