Thanks a lot, we will do an upgrade to 4.4 as you suggested.
Zitat von Tony Graziano :
you really should update to 4.4 as there are better options to
address some ITSP's who want different options.
In your case I "think" 4.2 has the option to change this here.
Use default asserted i
you really should update to 4.4 as there are better options to address some
ITSP's who want different options.
In your case I "think" 4.2 has the option to change this here.
Use default asserted identity(Default: checked)The asserted identity header
may be used by the ITSP to determine the origin
Hello,
we are using sipxecs 4.2.1 and are now trying to enable a SIP trunk.
But we are rejected from our SIP provider, because the wrong TO:
domain is sent by sipxecs
To:\r\nRecord-Route:
\r\nFrom:
\"Kofler Thomas\"
;epid=6AF60C3424;tag=68c63cf6d3\r\nTo:
\r\nCSeq: 163
INVITE\r\nCall-ID:
Content-Type: text/plain;
charset="utf-8"
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
In-Reply-To: <10067.4f1ec...@forum.sipfoundry.org>
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <65713>
Message-ID: <100b1.4f227...@forum.sipfoundry.org>
In order to use sip trunks with sipxb
, January 06, 2012 2:13 PM
> To: sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] Sip-Trunk Registration Confusion
>
>
> Content-Type: text/plain;
> charset="utf-8"
> Content-Transfer-Encoding: 8bit
> Organization: SipXecs Forum
> In-Reply-To:
> X-F
There's a forum?
/under rock
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Picher
Sent: Friday, January 06, 2012 2:13 PM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Sip-
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Organization: SipXecs Forum
In-Reply-To:
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Message-ID:
Well, good luck with this particular config.
The proxy should probably be the outbound proxy and the ITSP
addr
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Organization: SipXecs Forum
In-Reply-To:
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Message-ID:
It is a SiP trunk to the Sipxecs internal SBC. My vendor
1-Voip has a commercial trunks accross the internet.
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Organization: SipXecs Forum
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Message-ID:
I have been using 1-voip as a residential provider for 2
years with sipxecs no major issues. I recently upgraded to
their co
http://www.voip-info.org/wiki/view/STUN
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.326.5325
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: helpd...@voic
Use a different stun server that 1 is no longer accessible.
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.326.5325
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.842
No problem. I tend to ask for help more than give it on this list, so
glad to be able to help!
On 3/29/2011 8:01 PM, Dan Herman wrote:
> Content-Type: text/plain;
>charset="utf-8"
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> In-Reply-To:<1193405663-1301445532-cardhu_decom
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charset="utf-8"
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Organization: SipXecs Forum
In-Reply-To:
<1193405663-1301445532-cardhu_decombobulator_blackberry.rim.net-19962421...@bda015.bisx.prod.on.black>
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <58240>
Message-ID:
HOORAY F
Subject: [sipx-users] SIP Trunk registration stuck in INIT after reboot
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Hi all,
My previously working sipx 4
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Message-ID:
Hi all,
My previously working sipx 4.4.0 server went down abruptly
today due to a UPS failure. Upon restart, my gateways are
t starts up up and registers is the
> >> piece that needs to be inspected.
> >>
> >> Tony Graziano, Manager
> >> Telephone: 434.984.8430
> >> Fax: 434.984.8431
> >>
> >> Email: tgrazi...@myitdepartment.net
> >>
t;> http://www.myitdepartment.net/gethelp/
>>
>> ----- Original Message -
>> From: sipx-users-boun...@list.sipfoundry.org
>>
>> To: Discussion list for users of sipXecs software
>>
>> Sent: Sun Oct 10 11:18:35 2010
>> Subject: Re: [sipx-users
;
> Helpdesk Contract Customers:
> http://www.myitdepartment.net/gethelp/
>
> - Original Message -
> From: sipx-users-boun...@list.sipfoundry.org
>
> To: Discussion list for users of sipXecs software
>
> Sent: Sun Oct 10 11:18:35 2010
> Subject: Re: [sipx-users] SIP Trunk Regis
:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
- Original Message -
From: sipx-users-boun...@list.sipfoundry.org
To: Discussion list for users of sipXecs software
Sent: Sun Oct 10 11:18:35 2010
Subject: Re: [sipx-users] SIP Trunk
4.8426
> Fax: 434.984.8427
>
> Helpdesk Contract Customers:
> http://www.myitdepartment.net/gethelp/
>
> - Original Message -
> From: sipx-users-boun...@list.sipfoundry.org
>
> To: sipx-users@list.sipfoundry.org
> Sent: Sun Oct 10 10:07:52 2010
> Subject:
Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
- Original Message -
From: sipx-users-boun...@list.sipfoundry.org
To: sipx-users@list.sipfoundry.org
Sent: Sun Oct 10 10:07:52 2010
Subject: [sipx-users
On the ITSP gateway, I specified registration interval of 600 ( I
assume seconds as it states so on the page). However, the sipx is
attempting to register with ITSP every minute. Why?
Thanks in advance
___
sipx-users mailing list
sipx-users@list.sipfou
flow.
Rgds,
Nikolay.
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
> Roman Gelfand
> Sent: Friday, October 08, 2010 9:03 AM
> To: sipx-users@list.sipfoundry.org
> Subject: [sipx-
A call trace would be helpful.
Normally I would think there is a problem with the siptrunk configuration,
but the call behavior you describe makes me think the phone device itself is
signalling in a way that needs to be scrutinized.
Does this same behavior happen with xlite?
On Fri, Oct 8, 2010
I am using snom 730 phone which is registered with sipx server. It
appears that sip trunc registration between sipx and ITSP was
successfull. My sipx server is behind nat firewall. When I place a
phone call from outside (pstn), the snom 370 is ringing and displays
caller id. When I pick up the
Thx for the lowdown
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Friday, October 01, 2010 3:09 AM
To: Ujjval Karihaloo
Cc: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] SIP Trunk --> AA--> SIP trunk call flow
I'm not sure "any" b
for this setting to take effect..?
>
>
>
> *From:* sipx-users-boun...@list.sipfoundry.org [mailto:
> sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Ujjval Karihaloo
> *Sent:* Tuesday, September 28, 2010 11:48 AM
> *To:* Discussion list for users of sipXecs software; Tony Graziano
lay.
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
> Александр Горбунов
> Sent: Wednesday, September 29, 2010 11:58 AM
> To: Discussion list for users of sipXecs software
> Subject:
t.sipfoundry.org
>> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
>> ?
>> Sent: Tuesday, September 28, 2010 11:57 AM
>> To: sipx-users
>> Subject: [sipx-users] sip trunk works but cisco 7912 can not
>> call through it(sip guru welcome)
&g
:48 AM
To: Discussion list for users of sipXecs software; Tony Graziano
Subject: Re: [sipx-users] SIP Trunk --> AA--> SIP trunk call flow
I added the below entry and it did not work
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
nt: Friday, September 24, 2010 2:27 PM
To: Ujjval Karihaloo
Cc: mar...@ezuce.com<mailto:mar...@ezuce.com>;
dhub...@ezuce.com<mailto:dhub...@ezuce.com>;
sipx-users@list.sipfoundry.org<mailto:sipx-users@list.sipfoundry.org>
Subject: Re: [sipx-users] SIP Trunk --> AA--&
I see, I can try it..Will let everyone know
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Tuesday, September 28, 2010 10:34 AM
To: Ujjval Karihaloo
Cc: mar...@ezuce.com; dhub...@ezuce.com; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk --> AA--> SIP
t;
>
>
>
> *From:* Tony Graziano [mailto:tgrazi...@myitdepartment.net]
> *Sent:* Friday, September 24, 2010 2:27 PM
> *To:* Ujjval Karihaloo
> *Cc:* mar...@ezuce.com; dhub...@ezuce.com; sipx-users@list.sipfoundry.org
>
> *Subject:* Re: [sipx-users] SIP Trunk --> AA--> SIP t
502080]<http://www.simplesignal.com/>
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Friday, September 24, 2010 2:27 PM
To: Ujjval Karihaloo
Cc: mar...@ezuce.com; dhub...@ezuce.com; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk --> AA--> SIP trunk call
own to 0.
Rgds,
Nikolay.
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
> ?
> Sent: Tuesday, September 28, 2010 11:57 AM
> To: sipx-users
> Subject: [sipx-users] si
sipxecs 4.2.1 + sip trunk (nat) to itsp + (Cisco 7911, 7940, Linksys
SPA922) and everything works. Phones can call each other locally,
inside and outside and there are two way audio.
But there is bunch of Cisco 7912 (Software Version8.0.1(060412A) )
which cannot call out through the trunk. Please
t;
>
> -Original Message-
> From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
> Sent: Sunday, August 22, 2010 7:53 AM
> To: mar...@ezuce.com; dhub...@ezuce.com; Ujjval Karihaloo
> Cc: sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] SIP Trunk --> AA--
--Original Message-
> From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
> Sent: Sunday, August 22, 2010 7:53 AM
> To: mar...@ezuce.com; dhub...@ezuce.com; Ujjval Karihaloo
> Cc: sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] SIP Trunk --> AA--> SIP trunk c
[mailto:tgrazi...@myitdepartment.net]
Sent: Sunday, August 22, 2010 7:53 AM
To: mar...@ezuce.com; dhub...@ezuce.com; Ujjval Karihaloo
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk --> AA--> SIP trunk call flow
If it is available, I assume one could edit the xml file di
:* sipx-users-boun...@list.sipfoundry.org [mailto:
> sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Tony Graziano
> *Sent:* Wednesday, September 22, 2010 2:39 PM
> *To:* Discussion list for users of sipXecs software
> *Subject:* Re: [sipx-users] Sip trunk
>
> Then you are not following the in
: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Sip trunk
Then you are not following the instructions.
Add gateway>siptrunk, give it a name, choose the sbc route
"sipxbridge-1", choose the provider template (if applicable), then
APPLY. After that there is an o
aha ok i missed it
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony
Graziano
Sent: Wednesday, September 22, 2010 2:39 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Sip trunk
boun...@list.sipfoundry.org] *On Behalf Of *Tony Graziano
> *Sent:* Wednesday, September 22, 2010 2:14 PM
> *To:* Discussion list for users of sipXecs software
> *Subject:* Re: [sipx-users] Sip trunk
>
> http://wiki.sipfoundry.org/display/xecsuserV4r2/SIP+Trunking
>
> On Wed, Sep
er 22, 2010 2:14 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Sip trunk
http://wiki.sipfoundry.org/display/xecsuserV4r2/SIP+Trunking
On Wed, Sep 22, 2010 at 8:02 AM, mattias wrote:
I find how to add my sip trunk to spix
But not where to enter login set
http://wiki.sipfoundry.org/display/xecsuserV4r2/SIP+Trunking
On Wed, Sep 22, 2010 at 8:02 AM, mattias wrote:
> I find how to add my sip trunk to spix
> But not where to enter login settings
>
> ___
> sipx-users mailing list
> sipx-users@list.sipfoundry
I find how to add my sip trunk to spix
But not where to enter login settings
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
;Douglas Hubler' ; 'Ujjval Karihaloo'
Cc: sipx-users@list.sipfoundry.org
Sent: Sun Aug 22 09:37:36 2010
Subject: Re: [sipx-users] SIP Trunk --> AA--> SIP trunk call flow
>
> On Fri, Aug 20, 2010 at 3:52 PM, Ujjval Karihaloo
> wrote:
> > Guys:
> >
> &g
un...@list.sipfoundry.org
To: 'Douglas Hubler' ; 'Ujjval Karihaloo'
Cc: sipx-users@list.sipfoundry.org
Sent: Sun Aug 22 09:37:36 2010
Subject: Re: [sipx-users] SIP Trunk --> AA--> SIP trunk call flow
>
> On Fri, Aug 20, 2010 at 3:52 PM, Ujjval Karihaloo
> wrot
>
> On Fri, Aug 20, 2010 at 3:52 PM, Ujjval Karihaloo
> wrote:
> > Guys:
> >
> >
> > Looking for some help on this
.has anyone tried thisL I get no Audio
> in
> > each direction. Tony tried it with 2 different ITSPs and it works
but
> I
> > cannot get it to work with only one ITSP that I have to
On Fri, Aug 20, 2010 at 3:52 PM, Ujjval Karihaloo
wrote:
> Guys:
>
>
> Looking for some help on this….has anyone tried thisL I get no Audio in
> each direction. Tony tried it with 2 different ITSPs and it works…but I
> cannot get it to work with only one ITSP that I have to test.
>
> Any other su
gt; ITSP-->PSTN...NO AUDIO either direction.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ujjval Karihaloo
Sent: Monday, August 16, 2010 6:17 PM
To: Tony Graziano
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk -
->PSTN
>
>
>
>
>
>
>
> *From:* Tony Graziano [mailto:tgrazi...@myitdepartment.net]
> *Sent:* Friday, August 13, 2010 6:51 PM
>
> *To:* Ujjval Karihaloo
> *Cc:* sipx-users@list.sipfoundry.org
> *Subject:* Re: [sipx-users] SIP Trunk --> AA--> SIP trunk call
ompted to press 2 which thenroute the call back out
sipX--> SIP trunk --> ITSP-->PSTN
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Friday, August 13, 2010 6:51 PM
To: Ujjval Karihaloo
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk --> AA--
;
> *From:* sipx-users-boun...@list.sipfoundry.org [mailto:
> sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Ujjval Karihaloo
> *Sent:* Friday, August 13, 2010 7:54 PM
>
> *To:* Tony Graziano
> *Cc:* sipx-users@list.sipfoundry.org
> *Subject:* Re: [sipx-users] SIP Trun
: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Friday, August 13, 2010 6:51 PM
To: Ujjval Karihaloo
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk --> AA--> SIP trunk call flow
Yes, this works for this site...
ITSP1>AA>FWD through ITSP2
It just so
...@myitdepartment.net]
Sent: Friday, August 13, 2010 6:51 PM
To: Ujjval Karihaloo
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk --> AA--> SIP trunk call flow
Yes, this works for this site...
ITSP1>AA>FWD through ITSP2
It just so happens they have two itsp's, one
84.8427
>
> Helpdesk Contract Customers:
> http://www.myitdepartment.net/gethelp/
>
> - Original Message -
> From: Ujjval Karihaloo
> To: Ujjval Karihaloo ; Tony Graziano
>
> Cc: sipx-users@list.sipfoundry.org
> Sent: Fri Aug 13 20:36:14 2010
> Subject: RE: [s
: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
- Original Message -
From: Ujjval Karihaloo
To: Ujjval Karihaloo ; Tony Graziano
Cc: sipx-users@list.sipfoundry.org
Sent: Fri Aug 13 20:36:14 2010
Subject: RE: [sipx-users] SIP Trunk --> AA--> SIP
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ujjval Karihaloo
Sent: Friday, August 13, 2010 6:27 PM
To: Tony Graziano
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk --> AA--> SIP trunk call flow
Hi All:
Does anyone have this call flow working.
Call fro
...@myitdepartment.net]
Sent: Wednesday, August 11, 2010 3:07 PM
To: Ujjval Karihaloo
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk --> AA--> SIP trunk call flow
I think it might matter what the softphone is, and the version...
I would "try" the AA using the phan
I got it, Thanks a mill.
I need training :)
-Original Message-
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: 2010年8月14日 6:44
To: he...@tcd.ie; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk Template missing
You have to pic sbc route during
Customers:
http://www.myitdepartment.net/gethelp/
- Original Message -
From: sipx-users-boun...@list.sipfoundry.org
To: sipx-users@list.sipfoundry.org
Sent: Fri Aug 13 18:35:52 2010
Subject: [sipx-users] SIP Trunk Template missing
Hi Guys,
I am using sipx 4.3. I try to create a SIP TRUNK but
Behalf Of Michael
Scheidell
Sent: 2010年8月14日 6:38
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk Template missing
On 8/13/10 6:35 PM, Sen Heng wrote:
> Hi Guys,
>
> I am using sipx 4.3. I try to create a SIP TRUNK but it seems the provider
> template is missing.
On 8/13/10 6:35 PM, Sen Heng wrote:
> Hi Guys,
>
> I am using sipx 4.3. I try to create a SIP TRUNK but it seems the provider
> template is missing...?
> It was on sipx 4.2...
>
you pushed the buttons in the wrong order.
and sipx 4.3 isn't released anyway.
> Thanks,
> Sen
>
>
Hi Guys,
I am using sipx 4.3. I try to create a SIP TRUNK but it seems the provider
template is missing...?
It was on sipx 4.2...
Thanks,
Sen
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/si
e:
> See inline
>
>
>
> *From:* Tony Graziano [mailto:tgrazi...@myitdepartment.net]
> *Sent:* Wednesday, August 11, 2010 2:37 PM
> *To:* Ujjval Karihaloo
> *Cc:* sipx-users@list.sipfoundry.org
> *Subject:* Re: [sipx-users] SIP Trunk --> AA--> SIP trunk call flow
>
See inline
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Wednesday, August 11, 2010 2:37 PM
To: Ujjval Karihaloo
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk --> AA--> SIP trunk call flow
What you are describing is a hairpinned call. You should p
What you are describing is a hairpinned call. You should provide a siptrace
of the call with the proxy at debug as a minimum.
You should also describe your environment...
what kind of phone/ua (firmware software version might be relevant), whether
the UA or sipx is behind a nat or if the user is
I have a call coming in via a Sip trunk to an extension assigned to an AA.
AA plays the prompts user to dial 1 or 2...
In either case I send the call back out over the SIP trunk to a Cell PSTN
number. The call connects but I have no Audio either way.
Which Logs should I collect and provide to t
l' ;
sipx-users@list.sipfoundry.org
Sent: Mon Aug 09 18:47:02 2010
Subject: RE: [sipx-users] SIP Trunk Setup questions
Thx for the help.
Since I can change the listen port of asterisk , I just changed that to
something else for now...to validate sipX.
So interprocess communication on sipX will use DNS SR
) [mailto:dwor...@avaya.com]
Sent: Monday, August 09, 2010 4:02 PM
To: Ujjval Karihaloo; Todd Hodgen; 'Tony Graziano'
Cc: 'Michael Scheidell'; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] SIP Trunk Setup questions
From: U
From: Ujjval Karihaloo [ujj...@simplesignal.com]
I was using port 5050 for SIP Proxy as I am also running asterisk on this
server...I changed it back to port 5060, shutdown asterisk and it worked...
Looks like the sipXBridge only talks to the SIPXProxy pr
e) [mailto:dwor...@avaya.com]
> Sent: Monday, August 09, 2010 1:23 PM
> To: Ujjval Karihaloo; Todd Hodgen; 'Tony Graziano'
> Cc: 'Michael Scheidell'; sipx-users@list.sipfoundry.org
> Subject: RE: [sipx-users] SIP Trunk Setup questions
>
> __
7;
Cc: 'Michael Scheidell'; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] SIP Trunk Setup questions
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ujjval Karihaloo
[ujj...@simplesignal.com
From: Ujjval Karihaloo [ujj...@simplesignal.com]
I am not getting any ANSWER SECTION back for that.
I will make sure its hostname is resolvable through DNS and then try
What are my options if the hostname of the sipX server is not a FQDN...that
maps
) [mailto:dwor...@avaya.com]
Sent: Monday, August 09, 2010 1:23 PM
To: Ujjval Karihaloo; Todd Hodgen; 'Tony Graziano'
Cc: 'Michael Scheidell'; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] SIP Trunk Setup questions
F
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ujjval Karihaloo
[ujj...@simplesignal.com]
I am getting following error for the SIP Proxy Service….I am not using DNS SRV
for now.
* SIP route to SIPXC
users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Monday, August 09, 2010 10:36 AM
To: Ujjval Karihaloo
Cc: Michael Scheidell; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk Setup questions
yes.
On Mon, Aug 9, 2010 at 1:33 PM, Ujjval Karihaloo
wrote:
Thx,
Is t
@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk Setup questions
yes.
On Mon, Aug 9, 2010 at 1:33 PM, Ujjval Karihaloo
mailto:ujj...@simplesignal.com>> wrote:
Thx,
Is this what I do to get a SIP trace that will help?
http://sipx-wiki.calivia.com/ind
gt;
>
>
>
> *From:* Tony Graziano [mailto:tgrazi...@myitdepartment.net]
> *Sent:* Monday, August 09, 2010 11:33 AM
>
> *To:* Ujjval Karihaloo
> *Cc:* Michael Scheidell; sipx-users@list.sipfoundry.org
> *Subject:* Re: [sipx-users] SIP Trunk Setup questions
>
>
>
> I think a si
@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk Setup questions
I think a siptrace would be most useful first, I am not sure a full snapshot
would be required at this time. I would send the trace to the list, a JIRA is
premature.
On Mon, Aug 9, 2010 at 1:23 PM, Ujjval Karihaloo
mailto:ujj
...@list.sipfoundry.org] On Behalf Of Ujjval
Karihaloo
Sent: Monday, August 09, 2010 10:23 AM
To: Tony Graziano
Cc: Michael Scheidell; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk Setup questions
BTW, I am the ITSP and looking to test SipX as many of our Customers use it
idell; sipx-users@list.sipfoundry.org
> *Subject:* Re: [sipx-users] SIP Trunk Setup questions
>
>
>
> You are not providing much information. What is the UA? I see the ITSP is
> sending G722, is that for real? Who is the ITSP?
>
>
>
> I see they are sending to port 5080,
-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk Setup questions
You are not providing much information. What is the UA? I see the ITSP is
sending G722, is that for real? Who is the ITSP?
I see they are sending to port 5080, which is good. Is the UA at
a.b.c.161:5060? If so, what do
conference rooms.
Just point a DID to that auto attendant specifically.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael
Scheidell
Sent: Monday, August 09, 2010 9:05 AM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk
Karihaloo
Cc: Michael Scheidell; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk Setup questions
You are not providing much information. What is the UA? I see the ITSP is
sending G722, is that for real? Who is the ITSP?
I see they are sending to port 5080, which is good. Is the
a=rtpmap:8 PCMA/8000
>
> a=rtpmap:18 G729/8000
>
> a=fmtp:18 annexb=no
>
> a=rtpmap:101 telephone-event/8000
>
>
>
>
>
>
>
> *From:* Tony Graziano [mailto:tgrazi...@myitdepartment.net]
> *Sent:* Monday, August 09, 2010 10:25 AM
rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Monday, August 09, 2010 10:25 AM
To: Ujjval Karihaloo
Cc: Michael Scheidell; sipx-users@list.sipfoundry.org
Subjec
a trying back from SIPX and then nothing…
>
>
>
> *From:* sipx-users-boun...@list.sipfoundry.org [mailto:
> sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Michael Scheidell
> *Sent:* Monday, August 09, 2010 10:05 AM
>
> *To:* sipx-users@list.sipfoundry.org
> *Subject:* R
you have 4 digit internal extensions?
and, siptrace shows it going where?
and are you sure the user answers? (before you try to fwd it to the conf
bridge, make sure you got the right user)
I think I would have a 'normal' 4 digit user, '1000' with an alias of 562*.
also, watch 'job status' whe
...
I see a trying back from SIPX and then nothing...
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Scheidell
Sent: Monday, August 09, 2010 10:05 AM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk Setup
On 8/9/10 11:37 AM, Tony Graziano wrote:
OR you can assign the DID to a separate Auto Attendant and let people
choose the conference they want (1 for sales conf, 2 for management
conf, etc.).
I set it up so that 'special' users were given their own conference
'rooms', with a two digit prefix(70
Thx a lot. I will take a look
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Monday, August 09, 2010 9:38 AM
To: Ujjval Karihaloo
Cc: thod...@verizon.net; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk Setup questions
You need to configure a dial plan to send
appreciated.
>
> -Original Message-
> From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
> Sent: Monday, August 09, 2010 5:08 AM
> To: thod...@verizon.net; Ujjval Karihaloo; sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] SIP Trunk Setup questions
>
> Yo
Documentation/wiki links will be appreciated.
-Original Message-
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Monday, August 09, 2010 5:08 AM
To: thod...@verizon.net; Ujjval Karihaloo; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk Setup questions
You
On 8/8/10 10:56 PM, Ujjval Karihaloo wrote:
How do I register my SIPX with a ITSP. I do not see any way to put in
a username and passwd...only address (FQDN).
delete and start over.
its TRICKY and if you don't do it EXACTLY right, the options never show up.
as soon as you enter gateway ->a
: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
- Original Message -
From: sipx-users-boun...@list.sipfoundry.org
To: 'Ujjval Karihaloo' ;
sipx-users@list.sipfoundry.org
Sent: Mon Aug 09 02:05:05 2010
Subject: Re: [sipx-users] SIP Trunk Setup question
that screen, it will give you some additional choices!
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ujjval
Karihaloo
Sent: Sunday, August 08, 2010 7:56 PM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] SIP Trunk Setup ques
Used the following instructions to install SIPX.
http://sipx-wiki.calivia.com/index.php/Installing_sipXecs_on_Fedora_and_Centos
(BTW - Import Yum repository for CentOS does not work)
How do I register my SIPX with a ITSP. I do not see any way to put in a
username and passwd...only address (F
: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
- Original Message -
From: sipx-users-boun...@list.sipfoundry.org
Cc: sipx-users@list.sipfoundry.org
Sent: Wed Jul 28 08:14:55 2010
Subject: Re: [sipx-users] sip trunk - ITSP timed out
On 7/28/10 7:58 AM
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