Displaying 20 results from an estimated 4000 matches similar to: "[SPAM] [asterisk-dev] Disable 407 proxy authentication for outbound domains"
2006 Apr 10
1
Call me for testing my system
Dear User,
Anybody could dial these sip uri :
sip:info@nxs.yi.org (french)
sip:music@nxs.yi.org (music 60s)
sip:support@nxs.yi.org (french)
Thanks for help
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2006 Apr 07
2
407 proxy authentication
Hello,
Asterisk sent back 407 proxy authentication .
How can avoid this ?
I set insecure=very without success in sip.conf and my
sql server .
Harry
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2006 Apr 08
0
Re: [asterisk-dev] bug or bad chan_sip.c
Tzafrir,
How did you set sip:tzafrir@local.xorcom.com
I use ser----asterisk
look at my sip.conf and extensions.conf
Regards
Harry
////////////////////////////////////////////////////
[general]
context=sip
realm=nxs.yi.org
bindport=5050
bindaddr=nxs.yi.org
srvlookup=yes
tos=lowdelay
maxexpirey=3600
defaultexpirey=1000
allow=all
musicclass=default
language=fr
insecure=very
allowguest=yes
2006 Apr 08
0
Re: [asterisk-dev] bug or bad chan_sip.c
Tzafrir,
How did you set sip:tzafrir@local.xorcom.com
I use ser----asterisk
look at my sip.conf and extensions.conf
Regards
Harry
////////////////////////////////////////////////////
[general]
context=sip
realm=nxs.yi.org
bindport=5050
bindaddr=nxs.yi.org
srvlookup=yes
tos=lowdelay
maxexpirey=3600
defaultexpirey=1000
allow=all
musicclass=default
language=fr
insecure=very
allowguest=yes
2006 Mar 07
3
Re: [asterisk-dev] Is there a way to define an outbound proxy in sip.conf ?
Hello,
I use both ser/asterisk .
In fact i wish asterisk to forward all the sip
requests which are not handled by domain=domain.tld
in sip.conf
Here is a diagram:
The sip agents use the Sip proxy as an outbound sip
proxy and domain=domain.tld .
When the sip agents dial sip:user@otherdomain.tld so
the request is sent to sip proxy and so to Asterisk.
I wish Asterisk to Look up the
2006 Apr 08
2
HELP !!!!!
Hello,
I wish to set a sip uri sip:info@mydomain.
I use ser for authorization and authentication
(registrar rtpproxy and outbound proxy)
I use asterisk 1.2.5 with realtime .
the info is used as a hunt group so i add in
extension.conf
[info]
exten => info,1,Answer()
exten => info,n,Dial(Sip/84,10)
exten => info,n,Dial(Sip/85,10)
exten => info,n,Hangup
Ser forward sip:info@mydomain
2006 Jan 30
3
How many digium cards per server ?
Hello,
How many digium cards is supported per asterisk
server ?
Regards
Harry
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2006 Mar 02
0
Redirect a sip outbound requests to a sip proxy
Hi all,
Is there a solution to solve this ?
ASTERISK 1.2.4
||
Internet===SER/OPENSER=====Nat==[private net]
|| sip agents
rtpproxy/mediaproxy
Sip agents use SER/OPENSER as an outbound sip proxy
and asterisk as a registar server, pbx functions, ...
SER/OPENSER look for domains in URI. if domains are
handled by SER/OPENSER
2006 Mar 06
0
Outbound Proxy Support
Hi all,
May I have to patch asterisk-1.2.x with this patch
http://bugs.digium.com/bug_view_page.php?bug_id=0002859
to configure an outbound sip proxy in sip.conf ?
Regards
Harry
___________________________________________________________________________
Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et
2006 Jan 30
1
app_snmp
Hello,
Is there an app_snmp for asterisk-1.2.3 ?
Harry
___________________________________________________________________________
Nouveau : t?l?phonez moins cher avec Yahoo! Messenger
! D?couvez les tarifs exceptionnels pour appeler la
France et l'international.
T?l?chargez sur http://fr.messenger.yahoo.com
2004 Sep 08
4
Cisco GW and DTMF problems
I'm stuck running a old copy of asterisk because something strange is going
on in later versions of the CVS..
When I call in from a PSTN into my cisco 2610XM gateway which then routes
the call to my asterisk box via sip.. Asterisk can no longer process DTMF
tones generated by the calling party. This affects DISA, prompts and
menus.. Has anyone else had this problem?? and use.. I DO have
2005 Dec 18
12
ACD with polycom ip phones
Hello,
Polycom ip soundpoint support ACD login/logout .
Can we configure asterisk with polycom ACD support?
Regards
Harry
___________________________________________________________________________
Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international.
T?l?chargez sur http://fr.messenger.yahoo.com
2005 Jul 20
5
Grandstream GXP2000 resetting all the time
All,
I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 phones.
All seems well other than the phones have to be reset up to 5 times per day.
It is like they lose thier ip connection or maybe thier SIP connection. Has
anyone else experienced this issue? I have the phones set for static IP
addresses and that doesnt seem to help either. Any help would be greatly
2005 Sep 02
6
Looking for better "Follow Me"
Hi everybody :)
I am a new member here and hope that someone gives me a hint for my problem:
Let's say I am at work and my SIP phone (KPhone in my case) is connected to my
private Asterisk. I want to call my wife at home so her SIP phone rings. She
does not pick up the phone (maybe she is somewhere in the house and has to
run to the phone) so after 15 seconds her cell phone should ring.
2007 Jun 06
4
Slow list
Wow. My message made it to the list after more than 3 hours.
Philipp
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? -> http://www.das-asterisk-buch.de
Gesch?ftsf?hrer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
2008 Sep 12
2
Setup speed dials on Cisco 7921
I've added lines like this:
speeddial = 123,test
speeddial = 260,Bob
in the [device] section for my 7921, but the speed dials do NOT appear on
the menu (click right from the main screen). Am I missing something obvious
here?
Thanks
MD
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2007 Sep 25
4
Anyone else having problems with the list
I have sent a few emails over the past couple of days that simply have
not arrived on the list (or so it seems).
Is anyone else encountering this ?
Julian
2005 May 23
4
CallerID, TAPI and CTI
I would like to hear stories from people using TAPI, CTI or CallerID
software with Asterisk.
What are you guys using, setup examples, etc.
Has anybody sucessfully integrated SugarCRM with Asterisk and how did you do
it.
Are you running callerid software? Did you stumble into problems like using
tapi and callerid software returned both the callerid and calledid?
Hope you can help me out with
2005 Jul 25
4
Fritz PCI card in ptp mode with chan_misdn
Hello !
I would like to get working a Fritz PCI card using chan_misdn
operating in ptp mode.
Afer compiling mISDN into the kernel and building chan_misdn
Asterisk stops loading with :
[chan_misdn.so] => (Channel driver for mISDN Support (Bri/Pri))
== Parsing '/etc/asterisk/misdn.conf': Found
UnLocking config_mutex
== Registered channel type 'mISDN' (This driver enables
2008 Jan 10
3
OT - Is handover included in DECT GAP ?
Hi,
Do you if a DECT-GAP (or DECT-CAP) compliant handset MUST or MAY support
roaming and handover and are these functions transparent for handset (then,
these functions are implemented in DECT base stations) ?
Regards
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