Displaying 20 results from an estimated 2000 matches similar to: "Re: [asterisk-dev] Is there a way to define an outbound proxy in sip.conf ?"
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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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
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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 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 28
1
[SPAM] [asterisk-dev] Disable 407 proxy authentication for outbound domains
Hello,
I posted a lot of mails may be asterisk is not able to
accept sip calls from internet !?
My english is not fluent i try my best !
My problem I use ser+asterisk.
For local calls there are no problem (PSTN or IP)
Now i wish to receive calls from other internet domain
but asterisk ask for authentication 407.
IS IT possible to Disable authentication for incoming
calls to my sip uri ?
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 Jun 07
1
asterisk-1.2.9 / res_snmp.so
--- hgaillac-sip@yahoo.fr a ?crit :
> hello,
>
> How asterisk could support res_snmp even this module
> don't help to monitor all asterisk features?
>
> monitoring asterisk with snmp would be a good
> thing.
> Which solution ?
>
> Harry
> --- Kristian Kielhofner <kris@krisk.org> a ?crit :
>
> > hgaillac-sip@yahoo.fr wrote:
> > > I
2007 Jan 20
1
SIP registration problem w/ SBC
Hi,
I'm trying to get my * server connected to a softswitch through an SBC. I
get the following error when * trys to register.
Got SIP response 423 "Interval Too Brief" back from xxx.xxx.xxx.xxx
Jan 20 12:43:54 NOTICE[2138]: chan_sip.c:5473 sip_reg_timeout: --
Registration for '1234561234@xxx.xxx.xxx.xxx ' timed out, trying again
(Attempt #9)
Is there something I can
2003 Nov 07
2
Softswitch
Pardon my ignorance, but I was hoping someone could clear up something for
me.
- For a few POTS lines, digium has a single port card for that, or a T1 card
to a channel bank.
- For 10 or more lines, digium has a T1 or E1 card for that too based on PRI
channels
- For 100's to 1000's of lines, I suspect a soft-switch is in order???
A traditional phone company will sell:
- POTS lines for
2007 May 11
1
Fwd: SER as a Session Border Controller
I am curious if it is advisable to use implement Asterisk as a Session
Border Controller for a VoIP reseller environment. Users will terminate
calls SIP to my server, which will authenticate them via RADIUS, perform a
LCR lookup, select an appropriate trunk (based on LCR), and terminate the
call (update RADIUS accounting at end of call). All while acting as a B2BUA
to prevent the users from seeing
2006 Jan 30
1
app_snmp
Hello,
Is there an app_snmp for asterisk-1.2.3 ?
Harry
___________________________________________________________________________
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! D?couvez les tarifs exceptionnels pour appeler la
France et l'international.
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2005 Jan 20
4
softswitch dilemma
Hello everybody,
Im new to the list and also new to asterisk, Im wondering if I could set up asterisk as a softswitch, I guess for what I've been reading that It could be possible but almost all the info and documentation Ive found so far is about asterisk as a PBX, etc.
Im willing to set a small voip wholesale traffic bussiness and Im not quite sure asterisk is the right chocie for that.
2006 May 26
0
SIP call problem
Hello,
I have problem to make SIP calls, i have asterisk +
PC InterP4 + Digium TDM400P
here is the content of the sip.conf:
[SIP_PROVIDER]
type=peer
fromuser=testcomclient
username=testcomclient
secret=testr
host=IP_SIP_PROVIDER
;dtmfmode=rfc2833
context=interne
canreinvite=no
;allerid=Beer
disallow=all
allow=ulaw
allow=gsm
allow=g723.1 ; Asterisk only
2006 Jun 22
2
*** Spam *** Don't use CDRTool From AG-projescts
This post cannot be left without comment. People who don't know you or Adrian might get a wrong impression.
I know Adrian quite well and know that he is one of the real experts in this industry and he and his stuff does not deserve such a treatment.
I would recommend that you change your attitude. It seems like you did not get what you want (for free) and you complain like a small child. An
2020 Feb 28
4
Samba Bind DLZ Slow queries
Thanks Rowland, I have removed from options, and amended the forwarders.
[global]
workgroup = <MYDOMAIN>
realm = <MYDOMAIN>.CORP
netbios name = <HOSTNAME>
server role = active directory domain controller
idmap_ldb:use rfc2307 = yes
idmap config * : range = 3000-7999 ----------> If I remove the
portion I get errors -> idmap
2007 Oct 23
2
register => to let Asterisk register to another softswitch via SIP
Dear Alex;
Thanks alot for your nice help.
This is if I need to let Asterisk register with
another softswitch (so I used register =>), what if I
need asterisk to send call for the softswitch without
register to it (directly)? If I removed the register
=> then how it will distiguish the IP address in the
"host" at the [sip_trunk] is the IP address of the
softswitch that need to
2005 May 25
6
new cisco ip video phone?
Hi,
Just finished watching the season finale of '24' the TV series.
Throughout the series they have been showcasing Cisco hardware
especially Cisco IP phones (7970's).
On the last episode or two they showed what seemed to me a new cisco
IP video phone. It stands just as a 12" lcd screen with the cisco
branding/logo and letters just as the 79xx series.
I wonder if this is a
2006 Feb 02
1
Clash between multiple authentication databases (security problem?)
Hi all,
I just installed dovecot-1.0_beta2 and love it! However, I've run into issues using multiple authentication databases.
I use both pam (for users on my system) and sql (for virtual users). I've noticed that if I have system user "userabc" and virtual user "userabc at otherdomain.com" (two distinct users) and my "auth default" section contains these