Don't waste your time asterisk does not support
presence
--- Mark van Kerkwyk [EMAIL PROTECTED] a écrit :
Hi, anyone managed to get a Presence Agent
configuration with Asterisk 1.2
and X-Ten Eyebeam working. I believe this should be
paritally supported
now in 1.2 ?
regards
Mark
-info.org/wiki-Asterisk+phone+snom
A couple of pages down you'll see this:
SNOM SUBSCRIBE/NOTIFY support for monitoring
extension states
The methods and configuration here are also valid
for Eyebeam.
BB
harry gaillac [EMAIL PROTECTED] uttered the
following thing:
Don't waste
Hello,
When asterisk receive a registration with a private
address is it possible to forward the sip request for
this agent to a sip proxy ?
Regards
Harry
___
Appel audio GRATUIT
PROTECTED]
[mailto:[EMAIL PROTECTED] On
Behalf Of harry gaillac
Sent: Thursday, November 24, 2005 7:11 PM
To: users@openser.org;
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] harry's project
Hello,
here is an other diagram for people who don't yet
understand what i expect to do
Hello,
You built asterisk on freebsd ?
Harry
--- Olivier Taylor [EMAIL PROTECTED] a écrit
:
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:
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De
la part de harry gaillac
Envoyé : vendredi 25 novembre 2005 11:24
À : Asterisk Users Mailing List - Non-Commercial
Discussion
Objet : RE: [Asterisk-Users] Asterisk doesn't start
Hello,
You built asterisk on freebsd ?
Harry
--- Olivier
Hello,
Is there a GUI to manage sip users and voicemail with
Asterisk Realtime .
Regards
Harry
___
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez
Hello,
Read the Makefile in apps.
Harry
--- Olivier Taylor [EMAIL PROTECTED] a écrit
:
Hello,
I have compiled asterisk cvs under freebsd, no
problems.
When starting asterisk, I get :
[res_config_mysql.so] = (MySQL RealTime
Configuration Driver)
/libexec/ld-elf.so.1:
Je ne connais pas la signification de sybillines.
Harry
--- Olivier Taylor [EMAIL PROTECTED] a écrit
:
Tes réponses sont aussi sybillines que tes questions
:)
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De
la part de harry gaillac
Envoyé
signifie au figuré Qui est mystérieux obscur,
dont le sens est difficile
à saisir. Il m'a répondu en termes sibyllins. Des
paroles sibyllines. Un
langage sibyllin.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De
la part de harry gaillac
Envoyé : jeudi 24
sybillines que tes
questions
:)
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
De
la part de harry gaillac
Envoyé : jeudi 24 novembre 2005 16:45
À : Asterisk Users Mailing List - Non-Commercial
Discussion
Objet : RE: [Asterisk-Users
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
Doug,
You have ever post this mail.
Harry
Others have tried to explain it too you, but I don't
think you fully
understand. Maybe it is a language issue.
Your follow-up posts come across as demanding. When
I read your
posts, I feel like you are criticizing people for
not having
Dear users,
This letter is addressed to the most experienced users
for the ser openser and asterisk projects.
Advice me and I'll stop to mail my question.
How a session between two user agents behind nat could
stay in the path ?
Harry
Kinds Regards
|register || register
nat support for sip agents behind nat.
Why do you use both? Asterisk can also do NAT
traversal. For how many
users is the setup?
I think asterisk support 255 users
klaus
harry gaillac wrote:
Dear users,
This letter is addressed to the most experienced
users
for the ser
--- Klaus Darilion [EMAIL PROTECTED] a
écrit :
Hi Harry!
As this emails are on-topic you should cc: to the
list.
harry gaillac wrote:
In fact the problem is in contact sip header
field
(private ip)
agent send ReGISTER to SER (outbound proxy) which
one
send REGISTER to ASTERISK
may be you
I agree
--- Patrick [EMAIL PROTECTED] a écrit :
On Wed, 2005-11-23 at 10:34 +0100, harry gaillac
wrote:
Advice me and I'll stop to mail my question.
That almost sounds like a threat. Do you really
think you motivate
people to answer you this way? Since you asked
What are your prices
Harry
--- harry gaillac [EMAIL PROTECTED] a écrit :
may be you
I agree
--- Patrick [EMAIL PROTECTED] a écrit :
On Wed, 2005-11-23 at 10:34 +0100, harry gaillac
wrote:
Advice me and I'll stop to mail my question.
That almost sounds like a threat. Do you
You should read my mail so you would have an idea of
my problem !!!
Harry
--- Patrick [EMAIL PROTECTED] a écrit :
On Wed, 2005-11-23 at 14:36 +0100, harry gaillac
wrote:
What are your prices
Don't have any since I have no idea what your
problem is and how to
solve it so I can't
my name is gaillac not giallac
Harry
--- Steve Totaro [EMAIL PROTECTED] a
écrit :
New rule for email
Sender = harry giallac = deleted
-Original Message-
From: harry gaillac [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 23, 2005 8:33 AM
To: Asterisk Users Mailing List
You've now been asking the same questions on about 5
lists (we're all on) and
it doesn't help your cause.
three lists.
Why do you think i sent and resent my posts just for
playing ?
This (and the other) lists are free resources
provided by the community.
Have a look on the wiki
, but this might take some
work to get right.
Jan
harry gaillac wrote:
Hello open(ser) asterisk users
Here is what i expect to do :
Asterisk: registrar with public ip port=5050
open(ser): outbound proxy with public ip port=5060
Asterisk don't support IM and presence so i want to
use SER
, but this might take some
work to get right.
Jan
harry gaillac wrote:
Hello open(ser) asterisk users
Here is what i expect to do :
Asterisk: registrar with public ip port=5050
open(ser): outbound proxy with public ip port=5060
Asterisk don't support IM and presence so i want to
use SER
hello
___
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur http://fr.messenger.yahoo.com
hello
___
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur http://fr.messenger.yahoo.com
Hello,
Here is my config :
Asterisk as registrar server :public ip:5050
Ser as outbound proxy server :public ip 5060
I wish ser to handle the packets between Nat box
(netfilter) and Asterisk However contact field in
the sip HF don't change from nat box to asterisk which
don't allow to keep
Hello,
Can we configure asterisk in order to send sip
requests to a outbound proxy
when asterisk get AOR of users agents with an private
ip ?
Asterisk AOR:[EMAIL PROTECTED] ip
|
|
sip proxy/nat box---user agent
192.168.0.0/24
Regards
Harry
Hello open(ser) asterisk users
Here is what i expect to do :
Asterisk: registrar with public ip port=5050
open(ser): outbound proxy with public ip port=5060
Asterisk don't support IM and presence so i want to
use SER because of it's a good proxy:
I want user agents behind nat send
You lost me here. Was that a question or a
statement?
I might not be able to help, since my SER usage is
totally diffent,
but let me see if I got this right:
- You want the SER to forward REGISTER messages to
the Asterisk.
- The user agents use private IP addresses.
- You want the
of the
call debug.
Iqbal
harry gaillac wrote:
Hello open(ser) asterisk users
Here is what i expect to do :
Asterisk: registrar with public ip port=5050
open(ser): outbound proxy with public ip port=5060
Asterisk don't support IM and presence so i want to
use SER because of it's
okay, so ALL your users are registering to
asterisk...is that correct.
Correct via ser as outbound sip proxy
If so the problem is howto accept users from behind
a NAT into asterisk,
or am I confusing things further.
the problem is in contact field.
when user agents send register we
best
to find out howto
let ser do all the hardwork and let asterisk only
work when it needs to.
They can work together !
thanks for help
harry
harry gaillac wrote:
not exactly !
something like this :
asterisk
|
ser
ua1| | ua2
ua1
hello,
Give me your price to enable my diagram ASAP
--- harry gaillac [EMAIL PROTECTED] a écrit :
Hello open(ser) asterisk users
Here is what i expect to do :
Asterisk: registrar with public ip port=5050
open(ser): outbound proxy with public ip port=5060
Asterisk don't support IM
Hello,
I try to compile zaptel .
I installed kernel-sources but when i run :
make linux26
/
serveur1:/usr/local/src/ASTERISK/zaptel-1.2.0# make
linux26
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE
-DSTANDALONE_ZAPATA
[EMAIL PROTECTED] a écrit
:
harry gaillac wrote:
Hello,
I try to compile zaptel .
I installed kernel-sources but when i run :
make linux26
/
serveur1:/usr/local/src/ASTERISK/zaptel-1.2.0#
make
linux26
cc -I
-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De
la part de harry gaillac
Envoyé : lundi 21 novembre 2005 13:34
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] Can not build zaptel with
kernel-2.6.12
Hello,
I try to compile zaptel .
I installed kernel-sources
___
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur http://fr.messenger.yahoo.com---BeginMessage---
Hello,
Here is my config :
___
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur http://fr.messenger.yahoo.com---BeginMessage---
Remarque : message transféré en pièce jointe.
___
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur
Hello,
Here is my config :
Asterisk as registrar server :public ip:5050
Ser as outbound proxy server :public ip 5060
I wish ser to handle the packets between Nat box
(netfilter) and Asterisk However contact field in
the sip HF don't change from nat box to asterisk which
don't allow to keep
Hello,
Here is my config :
Asterisk as registrar server :public ip:5050
Ser as outbound proxy server :public ip 5060
I wish ser to handle the packets between Nat box
(netfilter) and Asterisk However contact field in
the sip HF don't change from nat box to asterisk which
don't allow to keep
However, it doesn't work consistently. Sometimes it
does, and sometimes
it doesn't. There's a thread on the asterisk-dev
list titled
chan_exosip2 where I am discussing my problems
with Olle.
Yes i posted the chan_exosip2 thread !
Harry
Do you configure BLA bridged line appearence for
presence in asterisk ?
Harry
--- Kevin Hanson [EMAIL PROTECTED] a écrit :
Michael Araba wrote:
I am having the same problems. The polycom phones
the 501 or 601 or
301 will list more more than 7 buddies neither
will the 601 with an
you mean than when the status of your subscribers
change they are notified
busy, away, ...
harry
--- Kevin Hanson [EMAIL PROTECTED] a écrit :
harry gaillac wrote:
Do you configure BLA bridged line appearence for
presence in asterisk ?
Harry
I am using hints in extensions.conf
you mean than when the status of your subscribers
change they are notified
busy, away, ...
harry
--- Kevin Hanson [EMAIL PROTECTED] a écrit :
harry gaillac wrote:
Do you configure BLA bridged line appearence for
presence in asterisk ?
Harry
I am using hints in extensions.conf
Hello,
Can you monitor the buddies status with you polycom
phones ?
Harry
--- Michael Araba [EMAIL PROTECTED] a écrit :
I am having the same problems. The polycom phones
the 501 or 601 or 301 will list more more than 7
buddies neither will the 601 with an expansion
module monitor more than
Hello
How do you configure Polycom for presence please ?
Harry
--- Alvaro Parres [EMAIL PROTECTED] a écrit :
Hi list, i have the next problem:
I create 3 hints.. (111 (SIP/111), 112 (SIP/112),
and 102 (ZAP/35) )
the SIP/111 is a GrandStream ATA
the SIP/112 is a Polycom 301
the ZAP/35 is
http://www.hylafax.org/
Harry
--- Doug Lytle [EMAIL PROTECTED] a écrit :
Jason Brashear wrote:
Receiving faxes with Asterisk.
Is there a good resource for learning how to set
this up?
www.soft-switch.org
Doug
--
Ben Franklin quote:
Those who would give up Essential
/asterisk/full
if /var/log/asterisk is where your log files are
kept.
With that, we can have a better idea of what's
happening/not
happening to give you the issue you're having.
On 11/10/05, harry gaillac [EMAIL PROTECTED]
wrote:
I did
: Operation not permitted
///
--- harry gaillac [EMAIL PROTECTED] a écrit :
Sorry,
Here are some files
Harry
--- BJ Weschke [EMAIL PROTECTED] a écrit :
This is good debugging info you've listed below,
but this isn't a sip
debug/trace
When the polycom ip300 phone (1.6.2) send registration
SUBSCRIBE message is sent to buddies from directory
file so NOTIFY is received from these one.
When I want to change status the ip phone don't send
NOTIFY to subscriber unlike SER which is a proxy!!!
Why?
Harry
--- harry gaillac [EMAIL
Hello,
I try to setup presence with polycom ip phones ip300
(1.6.2) .
I added buddies in directory files all is right for
registration subscription notification but when i want
to change status notify message is not sent to
subscribers ?
I don't understand !
Regards
Harry
Hello,
Asterisk don't support IM presence because of no proxy
function in chan_sip !
Regards
Harry
--- harry gaillac [EMAIL PROTECTED] a écrit :
When the polycom ip300 phone (1.6.2) send
registration
SUBSCRIBE message is sent to buddies from directory
file so NOTIFY is received from
hello,
http://www.egroupware.org/ would be a good choice (
open source).
--- Patrick [EMAIL PROTECTED] a écrit :
On Wed, 2005-11-09 at 12:45 +, Are wrote:
We want to intergrate AstBill with a Groupeware or
CRM but want input
what people will prefeer.
On our list today we have
[EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Discussion
asterisk-users@lists.digium.com
Sent: Thursday, November 10, 2005 5:25 AM
Subject: Re: [Asterisk-Users] groupware + unified
messagerie +Asterisk
harry gaillac wrote:
it's no what i expect the easier solution
Thanks for your advises
it's no what i expect the easier solution you
provide
the more customers you get !
I don't agree you ! the best solution you provide the
more customers you get (apache projects) !
Indeed. However, I tend to be of the opinion that
you should have enough
money in
Hello,
Does asterisk's team will improve IM and presence in
asterisk-1.2 !
Send Sip MESSAGE is impossible.
When the buddies status change nothing is happened.
How asterisk's team plan to solve this problem ?
Regards
Harry
you
to try and troubleshoot your problem, and you
haven't provided that to
date.
On 11/10/05, harry gaillac [EMAIL PROTECTED]
wrote:
Hello,
Does asterisk's team will improve IM and presence
in
asterisk-1.2 !
Send Sip MESSAGE is impossible.
When the buddies status change nothing
Does asterisk support RFC3265 ?
Harry
--- Matt Riddell [EMAIL PROTECTED] a écrit :
harry gaillac wrote:
nobody has an answer here!
Actually someone asked for you config details.
--
Cheers,
Matt Riddell
___
http://www.sineapps.com
://www.voip-forum.com/news.php?p=184c=1
--- Matt Riddell [EMAIL PROTECTED] a écrit :
harry gaillac wrote:
Hello,
Does sip_message_support.patch is available for
asterisk-1.2-bêta2 ?
Is there an other solution for Sip message ?
pabx*CLI show agi send text
Usage: SEND TEXT text
Somebody would be interested in a such project ?
Harry
--- Kristof Hardy [EMAIL PROTECTED] a
écrit :
harry gaillac wrote:
Is it possible to add a frontend groupware with
All is possible, you're only limited by your
imagination. (always wanted
to say this :p)
I'm not sure there's a(n
)
/
neither SUBSCRIBE, NOTIFY, MESSAGE sip method are ok
:(
Harry
--- Matt Riddell [EMAIL PROTECTED] a écrit :
harry gaillac wrote:
Hello Matt,
In fact I look for messaging an presence between
sip
phones .
http://www.voip-forum.com/news.php?p=184c=1
Should work with current CVS
I'm not a developper !
What do you mean Some parts of it, yes.
harry
--- BJ Weschke [EMAIL PROTECTED] a écrit :
Some parts of it, yes.
On 11/9/05, harry gaillac [EMAIL PROTECTED]
wrote:
Does asterisk support RFC3265 ?
Harry
--- Matt Riddell [EMAIL PROTECTED] a
écrit :
harry
it's no what i expect the easier solution you provide
the more customers you get !
--- Matt Riddell [EMAIL PROTECTED] a écrit :
harry gaillac wrote:
What about egroupware !
We use it, although there is no simple click to
install installation package
for Asterisk integration.
The idea
Olivier,
Oui !!! pour asterisk ou openpbx.
SER est un excellent proxy sip !
Il est evident qu' SER n'offre pas les fonctionalités
d'un ipbx.
je ne pense pas que toneec soit viable , combien
d'opérateurs offrent ces services (Skype)...
Votre pojet stagne!
Vous avez fait le choix de beacoups
nobody has an answer here !!
--- harry gaillac [EMAIL PROTECTED] a écrit :
Hello,
Where may i find documentation about SIP domain
support and dnsmgr.conf ,
Harry
___
Appel
nobody has an answer here!
--- harry gaillac [EMAIL PROTECTED] a écrit :
Hello,
I configure Polycom ip300 for presence but when
status
change notify is no sent to subscriber !?
Why ?
Regards
Harry
Hello,
Sorry here are my sip.conf and extensions.conf
in fact when polycom ip300 send subscribe to buddies
these one send back notify but nothing else when
status change
Regards
Harry
--- Matt Riddell [EMAIL PROTECTED] a écrit :
harry gaillac wrote:
nobody has an answer here!
Actually
Hello,
Is it possible to add a frontend groupware with
asterisk in order to Provide send receive fax to mail,
sms to mail, voice messages .
Asterisk or openpbx could be the server of the unified
messagerie .
click to dial contact in address book ,...
Harry
What about egroupware !
Harry
--- Kristof Hardy [EMAIL PROTECTED] a
écrit :
harry gaillac wrote:
Is it possible to add a frontend groupware with
All is possible, you're only limited by your
imagination. (always wanted
to say this :p)
I'm not sure there's a(n Open-source) project like
+xml
192.168.0.20 84 61c23b4e-3d 86
Idle xpidf+xml
2 active SIP subscriptions
--- BJ Weschke [EMAIL PROTECTED] a écrit :
Ok. What does sip show subscriptions from the CLI
show you?
On 11/8/05, harry gaillac [EMAIL PROTECTED]
wrote:
Hello,
Sorry here
thanks Matt for your answer
Does asterisk-1.2-stable will provide this features ?
Harry
PS:
Who are the main developpers for the sip channels ?
--- Matt Riddell [EMAIL PROTECTED] a écrit :
harry gaillac wrote:
nobody has an answer here !!
Where may i find documentation about SIP domain
[mailto:[EMAIL PROTECTED]
Sent: Tuesday, November 08, 2005 12:23
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] SIP domain support for
authentication and
virtualhosting
harry gaillac wrote:
thanks Matt for your answer
Does asterisk-1.2-stable
Hello,
Does sip_message_support.patch is available for
asterisk-1.2-bêta2 ?
Is there an other solution for Sip message ?
http://juraj.bednar.sk/work/software/asterisk/messaging/sip_message_support.patch
Hello,
I configure Polycom ip300 for presence but when status
change notify is no sent to subscriber !?
Why ?
Regards
Harry
___
Appel audio GRATUIT partout dans le monde avec le
Hello,
Where may i find documentation about SIP domain
support and dnsmgr.conf ,
Harry
___
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette
No !
Asterisk should send the invite request to sip proxy .
Harry
--- Walter Willis [EMAIL PROTECTED] a écrit :
the ser an asterisk run in the same box???
redirect host + port :)
2005/11/4, harry gaillac [EMAIL PROTECTED]:
Hello,
I wish to setup this scheme:
ser-0.9.4
Hello,
I wish to setup this scheme:
ser-0.9.4
asterisk-1.2-bêta
polycom ip300 phones
asterisk 5050--
|firewall+nat|-192.168.
ser 5060---
My ip phones use ser as outbound sip proxy and
asterisk as sip registrar server.
Ser Forward REGISTER requests to asterisk however
Harry
--- Walter Willis [EMAIL PROTECTED] a écrit :
the ser an asterisk run in the same box???
redirect host + port :)
2005/11/4, harry gaillac [EMAIL PROTECTED]:
Hello,
I wish to setup this scheme:
ser-0.9.4
asterisk-1.2-bêta
polycom ip300 phones
asterisk 5050
attributes,
radius failover, LCR, Call failover, Codec based
routing and other useful
things.
rafael
On 11/2/05, harry gaillac [EMAIL PROTECTED]
wrote:
Ok I trust you but does asterisk support radius ?
Harry
--- Daryl Sanders [EMAIL PROTECTED] a
écrit :
Asterisk works fine
Hello asterisk users,
I want to register sip agents (polycom ip 300) and
asterisk on Ser (sip express router)
sip user1--SERsip user2
|
|
Asterisk
How may i configure Ser+Asterisk in order to provide
Moh to sip agents
Hello,
Is this possible to send SIP messages (MESSAGE,
SUBSCRIBE, NOTIFY) to a sip proxy from asterisk ?
Regards
Harry
___
Appel audio GRATUIT partout dans le
Hello,
I read roadmap on www.openpbx.org.
Does chan_exosip2 will be able to provide a real sip
proxy ?
What about asterisk solutions ?
Harry
___
Appel audio GRATUIT partout dans le
: No authentication occurs (this is
presumably done outside by a SIP
proxy), incoming calls just get thrown into a
context. Outgoing calls are
down via SIP URI.
Joshua Colp
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Behalf Of harry gaillac
Sent: Tuesday
In fact I want to forward SIP MESSAGES to any sip
proxy
Unless chan_exosip2 is able to relay IM presence via
SIMPLE .
Harry
--- Roy Sigurd Karlsbakk [EMAIL PROTECTED] a écrit :
I read roadmap on www.openpbx.org.
Does chan_exosip2 will be able to provide a real
sip
proxy ?
What about
Hello,
I think you have to contact digium for software to
support SS7 with digium cards.
Harry
--- Usman [EMAIL PROTECTED] a écrit :
anyone running SS7 with Asterisk ? Please help me
out.
I need to know the hardware used for SS7 with Digium
E1 cards...
Thanks,
Look at http://www.ss7box.com/ too.
Harry
--- Goran Skular [EMAIL PROTECTED] a écrit
:
anyone running SS7 with Asterisk ? Please help me
out.
I need to know the hardware used for SS7 with
Digium E1 cards...
I can point you to one company in Austria. They
deployed SS7 on Asterisk,
but
Hello,
I 'll ask to my reseller
Harry
--- [EMAIL PROTECTED] a écrit :
thanks for that, i knew already but it misses the
actual version
Jesse Keating wrote:
On Fri, 2005-10-07 at 11:17 +0200, Kib Eki wrote:
Hello,
can anybody tell me where to get the latetest SIP
Firmware 1.6.2
Hello,
What do you think of this project www.openpbx.org ?
Something like ser and openser !
Kinds Regards
Harry
___
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo!
Hello,
Try insecure=very in [sip.philonline.com]
Harry
--- Ryan Pagquil [EMAIL PROTECTED] a écrit :
Hi,
I'm setting up Asterisk as a voicemail with
SER. My problem is,
when a caller that is not registered with asterisk
(no username and
password in sip.conf) it prompts 403,
Hello,
In order I try to fix my configuration please to call
me at :
sip:[EMAIL PROTECTED]
or
sip:[EMAIL PROTECTED]
or
sip:[EMAIL PROTECTED]
or
sip:[EMAIL PROTECTED]
Regards
Harry
___
Hello,
I have two polycom ip300.
I patched Asterisk However it don't show status of
phones when I press busy, Away, ...
So I use Sip Express Router (proxy sip) for IM and
Presence SIMPLE.
Harry
--- Adam Goryachev
[EMAIL PROTECTED] a écrit :
Hi all,
I've just updated to current CVS, and
Remarque : message transféré en pièce jointe.
___
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur
Mr Richardson,
I sympathize with american people after this disaster.
However If i was God I would feel remorse for all
people in the world in destitution because of
diseases, wars, starvation, ...
God should really feel remorse .
Thinks to all people in destitution in the world .
Harry
---
hello,
I read
http://www.voip-info.org/tiki-index.php?page=Asterisk+at+large
anybody could tell me more about this ?
Is it available with ARA ?
Regards
Harry
Method 3
Q: If you have your SIP phones registered with SER but
your voicemail is handled by asterisk, how do you get
the MWI
Hello,
I set SER as sip proxy and ASTERISK as voicemail
server (ARA) and serweb as TUI (telephone user
interface) .
Serweb
|
Ua---ser---asterisk voicemail
| |
Mysql DB
I add user agents with address sip:[EMAIL PROTECTED] +
aliases
] a écrit :
These questions should be sent to Asterisk-Users
this is not a developer
issue.
Cheer's
Steve McMahon
- Original Message -
From: harry gaillac [EMAIL PROTECTED]
To: asterisk-dev@lists.digium.com
Sent: Monday, August 29, 2005 9:18 AM
Subject: [Asterisk-Dev
] a écrit :
These questions should be sent to Asterisk-Users
this is not a developer
issue.
Cheer's
Steve McMahon
- Original Message -
From: harry gaillac [EMAIL PROTECTED]
To: asterisk-dev@lists.digium.com
Sent: Monday, August 29, 2005 9:18 AM
Subject: [Asterisk-Dev
Am i alone with this problem ?
I just rewrote voicemessages table because of errors.
I read app_voicemail.c to fix my problem.
However app_voicemail.c support many schemes to query
the tables.
Harry
--- Jerris, Michael MI [EMAIL PROTECTED] a écrit :
harry gaillac
I agree you however i
Am i alone with this problem ?
I just rewrote voicemessages table because of errors.
I read app_voicemail.c to fix my problem.
However app_voicemail.c support many schemes to query
the tables.
Harry
--- Jerris, Michael MI [EMAIL PROTECTED] a écrit :
harry gaillac
I agree you however i
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