This is with Bootrom 2.6.2.0032, SIP 1.5.2.0054.
On Nov 24, 2005, at 3:32 AM, Adam Goryachev wrote:
What firmware version did you use for the polycom phone ??
I just tried it on my IP600, and when I press the park button, it waits
for me to dial an extension number, then I press park again, an
On 11/24/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Without putty, my windows would be meaningless.
>
> PaulH
>
Subtle Paul! but nice! :)
Mike
UK
> - Original Message -
> From: "C F" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Frida
From my original post:
"using ParkAndAnnouce puts the parked call on hold, hangs up the parker
and then immediately calls them back with an announcement of the stall
number"
So, I would say, yes :-)
On Nov 24, 2005, at 11:09 AM, Alvaro Parres wrote:
Hi... I have the polycom 301 with firmw
Title: Linksys SPA-841 Disconnects from Asterisk
Hi all, I wonder if anyone out there has experienced an issue I am having with my Sipura / Linksys SPA-841 phones.
They work fine generally, but occasionally, incoming calls are missed. It's like the SIP registration is expiring. Does anyon
Title: Linksys SPA-841 Disconnects from Asterisk
Check in you console or your logs when this happens. I'm
guessing it's a Stale Nonce
If this is the case, Sipura supposedly fixed the bug on
it's most recent firmware (At least for the SPA-1001 and SPA-2100, but I'm
guessing the SPA-841 also)
Hi,
This is just a reminder to inform you that the asterisk usergroup in
montreal will hold a meeting today at 4h45.
For more information, please visit:
http://amug.modulis.ca/
See you there,
Adrien
--
Adrien Laurent - CIO
(514) 284-2020 x 202
[EMAIL PROTECTED]
www.modulis.ca
_
What version of Asterisk are you using?
I had a similar problem with another make. Since using Asterisk 1.20
this issue has gone.
Dave Morrow wrote:
Hi all, I wonder if anyone out there has experienced an issue I am
having with my Sipura / Linksys SPA-841 phones.
They work fine generally,
The F3000 is also a clamshell, "flip" type phone. I should be receiving
an eval unit shortly and will post my findings after we work it over in
the lab.
Cory Andrews
Senior Partner
+++
VOIPSupply.com
454 Sonwil Drive
Buffalo, NY 14225
+++
voice - 716.630.1555 X22
email - [EMAIL
Hi,
I got the person to force the G729 codec on their Linksys WRT54GP2 and
forced it on Asterisk as well. The person then managed to get a single call
out but all subsequent call set ups failed with the same 488 error.
I went back over my SIP traces and noticed that the Cseq's were often out of
On Thu, November 24, 2005 20:09, Francesco Peeters said:
> On Wed, November 23, 2005 20:29, Francesco Peeters said:
>
> Just made myself a crossed NT1 connection to the NT mode card (as
> described on the PBX4linux site) and connected my phone.
>
> The zaphfc driver shows that layer 1 is activated
hi all, i have asterisk configured and working but the quality is very
poor. i ear noise and braks in the voice when the people talk to me, and
the people that eared me have the same problem any recommendation?
any files you need to post?
--
.-
Pablo Allietti
LACNIC
Dear all.
I am new using asterisk.
I planned to have in my company an asterisk pbx that as a start would
be serving one analog phone, four sip hardphone extensions and two Iax
softphone. The next plan is to integrate asterisk with an old PBX
Alcatel 4100 and 3 lines to the public phone company.
So
On Thu, 2005-11-24 at 16:00 -0500, Adrien Laurent wrote:
> Hi,
>
> This is just a reminder to inform you that the asterisk usergroup in
> montreal will hold a meeting today at 4h45.
So much stuff in Montreal, can't wait to move up there :)
>
> For more information, please visit:
> http://amug.m
On 24 Nov 2005, at 10:26, Julian Lyndon-Smith wrote:I know that's a real newbie question, but I have a problem.I keep getting frame rejects, and a D-channel bouncing up and down. BT say that it is at my end. If I stop asterisk, stop the zaptel service and restart, things seem ok for a while.Pardon
Merci pour ces précisions.
Harry
--- Olivier Taylor <[EMAIL PROTECTED]> a écrit
:
> SIBYLLIN, INE. adj. Qui appartient aux sibylles. Il
> n'est guère usité au
> sens propre que dans ces locutions : Les oracles,
> les livres, les vers
> sibyllins, Les oracles, les livres, les vers des
> sibylles.
Je ne donne pas de réponse !
Il me semble t'avoir suggèrer asterisk comme système
de messagerie vocale au lieu d'SEMS, avoir fourni
quelques fichiers de configuration, ce n'étaient pas
des devinettes.
Conbien de fois on ma répondu "personne n'est obligé
de faire ton tavail, tu n'as qu'a payé pour
Hi Jerry & List,
I have the following registrations in sip_additional.conf
register=02820:@202.177.XXX.XXX/02820
[02820]
type=user
secret=
host=202.177.XXX.XXX
context=from-pstn
sip_additional.conf is (or should be) included from sip.conf
Any other suggestions? Unfortuantely I was
Hi Julian,
I think the Dell poweredge2850 servers are not too compatible with the zaptel cards..
Thanks
krishna
On 11/24/05, Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote:
I know that's a real newbie question, but I have a problem.I keep getting frame rejects, and a D-channel bouncing up and d
Title: Linksys SPA-841 Disconnects from Asterisk
I don t have problems, after upgrade the
firmware to the latest version.
Alex
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave Morrow
Sent: Thursday, November 24, 2005
3:49 PM
To: Asterisk
Users Mailing Lis
In article <[EMAIL PROTECTED]>,
Steven Langley <[EMAIL PROTECTED]> wrote:
>
> I have been using Asterisk 1.0.9 fairly successfully. I have Iax2 softphones
> based on the IaxClient library that are dialing into Meetme conferences. I
> am using a Zaptel card as a timing source.
>
> I am now trying
Adam Goryachev wrote:
> see the zapata.conf for callprogress=yes
> However, this is unreliable, and could provide incorrect results. For
> accurate information you will need to get a BRI or PRI and related
> interface card. These provide the information Out Of Band, and as such
> are accurate.
I'v
Hi,
Asterisk 1.2 on FC4, all is right, I'm happy. But when I try to load
chan_misdn after a successful install, I get it :
# asterisk -vvvgc
[...]
[chan_features.so] => (Feature Proxy Channel)
== Registered channel type 'Feature' (Feature Proxy Channel Driver)
[chan_misdn.so] => (Channel driv
Can some on help me find the problem here please:
I'm using asterisk 1.2.0 with Grandstream GXP-2000
This is the debugging output from asterisk:
<-- SIP read from 10.0.3.21:5060:
REGISTER sip:10.0.3.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.3.21;branch=z9hG4bK5c77f205e9f991de
From: ;tag=aea38200ad3c1539
T
Hi Alfie.
Did you try setting up a "username=100" in your [100] context and a
"username=101" in your [101] context?
That should do the trick..
Michel Belleau
SERVICES INFORMATIQUES MALAIWAH.COM
(418) 261-6412 -- http://www.malaiwah.com
Alfie Viechweg a écrit :
> Can some on help me find the p
Yoann,
I am going through a similar problem you reported in a past posting:
Nov 24 17:49:31 ERROR[9326] chan_misdn.c: Unable to initialize mISDN
Nov 24 17:49:31 WARNING[9326] loader.c: chan_misdn.so: load_module
failed, returning -1
Nov 24 17:49:31 WARNING[9326] chan_misdn.c: cb_log called with
ou
Michel Belleau (malaiwah.com) wrote:
Hi Alfie.
Did you try setting up a "username=100" in your [100] context and a
"username=101" in your [101] context?
That should do the trick..
Michel Belleau
SERVICES INFORMATIQUES MALAIWAH.COM
(418) 261-6412 -- http://www.malaiwah.com
Alfie Viechweg a é
Hi there,
We have PolyCom IP501s in a context that allows long-distance dialing,
but we want to prevent those same phones from being forwarded to
long-distance numbers using the softkey on the phone (without disabling
the key itself).
Does anyone have any PolyCom/dialplan tricks to accomplis
Jose,
I met so many problems these last 8 days that I don't remember exactly
which config was mine at that time, so I can't testify the answer...
(just for fun : my linux box is having 3 hd with a different distro on
each of them and I plug the cable on the hd I want to boot depending
on my mood ;
Hello,
here is an other diagram for people who don't yet
understand what i expect to do.
Look at sip_call_flow.png file i wish to substitute
ondo sip server with ser and ondo pbx with asterisk .
ondo sip server is able to do far-end near-end nat I
guess ser too.
I do hope i will find some peop
Found auth problem!
Installing asterisk 1.2.0 with INSTALL_PREFIX set will copy this
variables into your config file - asterisk.conf and result int things
like failed sip user information etc. If you do you own install (LFS :)
poeple) beware! Try using DESTDIR instead. The docs and Makefile is
Hi;
We're looking to standardise on a single family of E1 PRI cards.
I guess our options are :
Digium / Zaptel / libpri
Sangoma/ Zaptel / Wanpipe
AVM/ CAPI
eIcon / CAPI
Junghanns / Bristuff
Can anyone share any comparative experience of these, please ? Do they
differ muc
This is how I just did it (finally):
### First grab the mqueue branch of mISDN to the folder which is hard-coded
in the chan_misdn Makefile
mkdir /usr/src/mqueue
cd /usr/src/mqueue
cvs -d :pserver:[EMAIL PROTECTED]:/i4ldev login
(password: readonly)
cvs -d :pserver:[EMAIL PROTECTED]:/i4ldev co
Well, I kinda answered this myself, and I'll post how I did it in case
a) it might cause other problems and 2) anyone else finds it useful.
To recap, a Polycom phone will let you enter anything as a call
diversion number - this is obviously a problem in that someone can
forward their phone to
http://www.automated.it/guidetoasterisk.htm
I don't think you even require SER in that case.
That will be $100.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of harry gaillac
Sent: Thursday, November 24, 2005 7:11 PM
To: users@openser.org; as
Adrian A wrote:
Does anyone know what exactly the option transmit_silence_during_record in
asterisk.conf does? Is this useful for voicemail recording?
Could the option be named any more explicitly? It does _exactly_ what it
says it does.
___
--Band
On Thu, Nov 24, 2005 at 10:16:40PM +0100, Francesco Peeters wrote:
> On Thu, November 24, 2005 20:09, Francesco Peeters said:
> > On Wed, November 23, 2005 20:29, Francesco Peeters said:
>
> >
> > Just made myself a crossed NT1 connection to the NT mode card (as
> > described on the PBX4linux site
hi everybody:
I use Asterisk and SER(with nathelp moudle) in on box, SER as sip
registrar and sip proxy, Asterisk as media gw and pstn connector. Here
is my configuration: SER use 192.168.2.10:5060,Asterisk use
192.168.2.10:5065,my pstn gw is 192.168.2.20:5060
in ser.cfg
if (method=="INVI
Have you tried the "soft hangup" command?On 11/24/05, Paradise Dove <[EMAIL PROTECTED]> wrote:
hi,how can i hangup such calls without restarting asterisk?the Zap channel on this case is busy for more than 7 hours
some logs are followed.thanks,Paradise Dove-Nov 23 16:59:49 NOTICE
On Nov 24, 2005, at 12:14 PM, Bharath wrote:
I found out that I have a faulty Belkin Router which was causing
the problem. I tried forwarding ports as well as DMZ'd the Sip
device but still could'nt not hear the voice. So i plugged the sip
device directly to the cable modem & it worked fine
Hi,
What would be a recommended PCI latency timing for server running TE406P
card?
Thanks
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/a
Are there any kind of patches or experimental libraries that I can use
to pull caller ID info off a japanese pots line?
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lis
Hi all
iam setting PBX for outgoing calls at this moment
once iam success this , iam planning to do config inbound to
So iam start configuring with Outbound calls
Ring now my config looks like follow
Lan Users-- Astrix--- VoIP provider
I have one account with VoIP provider, i can make
Adrian A wrote:
Does anyone know what exactly the option
transmit_silence_during_record in asterisk.conf does? Is this useful
for voicemail recording?
Could the option be named any more explicitly? It does _exactly_ what
it says it does.
Some providers terminate the connection if nothing is
On Fri, November 25, 2005 3:27, Tzafrir Cohen said:
> On Thu, Nov 24, 2005 at 10:16:40PM +0100, Francesco Peeters wrote:
>> > Seems to me there's an issue in that area: chan_zap, maybe libpri,
>> etc.
>
> So what do you have in zapata.conf?
>
I posted that a few posts back in this thread... No ne
Actually, exactly now I am trying to do that also...
Isamar
On Fri, 25 Nov 2005, Aaron Anderson wrote:
Are there any kind of patches or experimental libraries that I can use
to pull caller ID info off a japanese pots line?
___
--Bandwidth and Co
as i said before, i've ran "soft hangup" on both sip and zap channels
on this call several times but no success.
by exploring the code in chan_sip.c it shows that * also attempts to
run softhangup on this call.
is this probably be a bug?
thanks,
paradise dove
On 11/25/05, tracinet <[EMAIL PROTECT
Lee Archer wrote:
Nov 24 10:50:15.02: [ 8222]: <-- data [1031]
Nov 24 10:51:15.01: [ 8222]: MODEM TIMEOUT: writing to modem
Nov 24 10:51:15.01: [ 8222]: MODEM WRITE SHORT: sent 1031, wrote 984
Nov 24 10:51:15.01: [ 8222]: SEND end page
What's going on with the iaxmodem output (on stdout/stder
Merci
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de harry gaillac
Envoyé : jeudi 24 novembre 2005 23:25
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : RE: RE : RE : [Asterisk-Users] What does it mean?
Je ne donne pas de r
also add winscp and ultraedit to your windows system, it works great.
http://winscp.net/eng/index.php
http://www.ultraedit.com/
Regards
Guido Hecken
> > Without putty, my windows would be meaningless.
> >
> > PaulH
> >
> Subtle Paul! but nice! :)
> Mike
> UK
___
Hello
Whan starting astersik(1.2) (asterisk -vvc), I get this message :
[res_config_mysql.so] => (MySQL RealTime Configuration Driver)
/libexec/ld-elf.so.1: /usr/lib/asterisk/modules/res_config_mysql.so:
Undefined s
ymbol "ast_config_load"
What did I forgot to do?
Olivier
___
Make sure that you compile misdnuser with gcc3.x, gcc4 did
not work for me.
Hans
Yoann Le Bihan schrieb:
Jose,
I met so many problems these last 8 days that I don't remember exactly
which config was mine at that time, so I can't testify the answer...
(just for fun : my linux box is having 3 hd
You can also try busydetect=yes, busycount=4 in zapata.conf.
;; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D; etc, it can be useful to perform busy detection either in an effort to; detect hangup or for detecting busies;
busydetect=yes;; On trunk interfaces (FXS) it ca
asterisk183 schrieb:
I have download the bristuff-0.3.0-PRE-1.tar.gz and I followed the
instruction in INSTALL:
[..]
5. modprobe zaptel
6.
But when I doing insmod qozap.o
and ztcfg don't start because in /qozap directory I don't have qozap.o
files. Why?
Try
modprobe qozap
inste
Thanks Adam, I will do all that you suggested.
Julian
Adam Goryachev wrote:
On Thu, 2005-11-24 at 10:26 +, Julian Lyndon-Smith wrote:
I posted a similar problem a couple of days ago, and one of the
responses suggested that the TE4xxP may be on it's way out.
Is there any way of testing thi
Thanks for your help Tim:
Comments inline:
tim panton wrote:
On 24 Nov 2005, at 10:26, Julian Lyndon-Smith wrote:
I know that's a real newbie question, but I have a problem.
I keep getting frame rejects, and a D-channel bouncing up and down. BT
say that it is at my end. If I stop asterisk,
Hi Krishna,
Thanks for the heads up, but I thought that Digium recommended the 2850
for the ABE ?
Julian
Krishna Sumanth Chava wrote:
Hi Julian,
I think the Dell poweredge2850 servers are not too compatible with the
zaptel cards..
Thanks
krishna
On 11/24/05, Julian Lyndon-Smith <[EMAIL P
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