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:sip:+4312345...@sipxecs.sos-kd.org;user=phone
should be
To:sip:+4312345...@trunk1.at.telgo.cc
and not
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
Thanks a lot, we will do an upgrade to 4.4 as you suggested.
Zitat von Tony Graziano tgrazi...@myitdepartment.net:
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.
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In order to use sip trunks with sipxbridge
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-Trunk
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It is a SiP trunk to the Sipxecs internal SBC.
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I have been using 1-voip as a residential provider for 2
years with sipxecs no major issues.
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Hi all,
My previously working sipx 4.4.0 server went down abruptly
today due to a UPS
-To: Discussion list for users of sipXecs software
sipx-users@list.sipfoundry.org
Subject: [sipx-users] SIP Trunk registration stuck in INIT after reboot
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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:
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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:
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:
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
@list.sipfoundry.org
Sent: Sun Oct 10 10:07:52 2010
Subject: [sipx-users] SIP Trunk Registration
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
-users] SIP Trunk Registration
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
Sent: Sun Oct 10 11:18:35 2010
Subject: Re: [sipx-users] SIP Trunk Registration
Unless I misunderstood you, I deleted sipxbridge log and restarted
sipx trunk service and still I am seeing registration every minute.
On Sun, Oct 10, 2010 at 10:42 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote
://www.myitdepartment.net/gethelp/
- Original Message -
From: sipx-users-boun...@list.sipfoundry.org
sipx-users-boun...@list.sipfoundry.org
To: Discussion list for users of sipXecs software
sipx-users@list.sipfoundry.org
Sent: Sun Oct 10 11:18:35 2010
Subject: Re: [sipx-users] SIP Trunk
Subject: Re: [sipx-users] SIP Trunk Registration
Unless I misunderstood you, I deleted sipxbridge log and restarted
sipx trunk service and still I am seeing registration every minute.
On Sun, Oct 10, 2010 at 10:42 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
I think I pointed
@list.sipfoundry.org
Sent: Sun Oct 10 11:18:35 2010
Subject: Re: [sipx-users] SIP Trunk Registration
Unless I misunderstood you, I deleted sipxbridge log and restarted
sipx trunk service and still I am seeing registration every minute.
On Sun, Oct 10, 2010 at 10:42 AM, Tony Graziano
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
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-users] SIP Trunk call dropped
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
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 build will fix
...@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
*Subject:* Re: [sipx-users] SIP Trunk -- AA-- SIP trunk call flow
I added
, 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)
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
.
-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: Re: [sipx-users] sip trunk works
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.
is counted down 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] sip trunk works
[image: bvoip] 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
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 trunk call
...@ezuce.com;
dhub...@ezuce.commailto:dhub...@ezuce.com;
sipx-users@list.sipfoundry.orgmailto:sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk -- AA-- SIP trunk call flow
So I am not sure this is exactly supported at this time...
http://track.sipfoundry.org/browse/XX-7362
: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 Ujjval
[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 directly
-
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 call flow
If it is available, I assume one could
-
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 call flow
If it is available, I assume one could
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/
http://wiki.sipfoundry.org/display/xecsuserV4r2/SIP+Trunking
On Wed, Sep 22, 2010 at 8:02 AM, mattias m...@mjw.se wrote:
I find how to add my sip trunk to spix
But not where to enter login settings
___
sipx-users mailing list
, 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 m...@mjw.se wrote:
I find how to add my sip trunk to spix
But not where to enter login settings
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 22, 2010 at 8:02 AM, mattias m...@mjw.se wrote:
I find how to add
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
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Sip trunk
Then you are not following the instructions.
Add gatewaysiptrunk, give it a name, choose the sbc route
sipxbridge-1, choose the provider template (if applicable), then
APPLY. After that there is an option
...@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 instructions.
Add gatewaysiptrunk, give
On Fri, Aug 20, 2010 at 3:52 PM, Ujjval Karihaloo
ujj...@simplesignal.com 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
-users-boun...@list.sipfoundry.org
To: 'Douglas Hubler' dhub...@ezuce.com; 'Ujjval Karihaloo'
ujj...@simplesignal.com
Cc: sipx-users@list.sipfoundry.org 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
...@list.sipfoundry.org
To: 'Douglas Hubler' dhub...@ezuce.com; 'Ujjval Karihaloo'
ujj...@simplesignal.com
Cc: sipx-users@list.sipfoundry.org 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
...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 -- AA-- SIP trunk call
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-- SIP trunk call flow
Yes
...@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...
ITSP1AAFWD through ITSP2
It just so happens they have two itsp's
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:
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
] On 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
Customers:
http://www.myitdepartment.net/gethelp/
- Original Message -
From: sipx-users-boun...@list.sipfoundry.org
sipx-users-boun...@list.sipfoundry.org
To: sipx-users@list.sipfoundry.org sipx-users@list.sipfoundry.org
Sent: Fri Aug 13 18:35:52 2010
Subject: [sipx-users] SIP Trunk Template
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
...@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 phantom user just to see
[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 from ITSP
@list.sipfoundry.org
Sent: Fri Aug 13 20:36:14 2010
Subject: RE: [sipx-users] SIP Trunk -- AA-- SIP trunk call flow
Looks my call forward off of an Extension on SIPX to my cell phone is not
working either...
Any ideas befor eI start digging into traceswith sipX using so many
internal ports (5090
:36:14 2010
Subject: RE: [sipx-users] SIP Trunk -- AA-- SIP trunk call flow
Looks my call forward off of an Extension on SIPX to my cell phone is not
working either...
Any ideas befor eI start digging into traceswith sipX using so many
internal ports (5090, 5080, 5090..15060)...the trace
...@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...
ITSP1AAFWD through ITSP2
It just so happens they have two itsp's, one for inbound
..
*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 Trunk -- AA-- SIP trunk call flow
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
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
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 provide
...@simplesignal.comwrote:
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
Sent: Mon Aug 09 02:05:05 2010
Subject: Re: [sipx-users] SIP Trunk Setup questions
I think you are missing a few options on the page. When you open
Device/gateway you will get a menu on the left hand side of the screen.
Click on ITSP account. You can enter your username and password on that
screen
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
to 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
.
-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 need to make sure you choose sbc route
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
...
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
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'
…
*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 questions
On 8/9/10 11:37 AM, Tony
09, 2010 10:25 AM
To: Ujjval Karihaloo
Cc: Michael Scheidell; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk Setup questions
I don't understand. Your userID DOE NOT need to be identical.
Example, user 200 has an ALIAS of 5625551000
Creating users with the same value of a DID
*Cc:* Michael Scheidell; sipx-users@list.sipfoundry.org
*Subject:* Re: [sipx-users] SIP Trunk Setup questions
I don't understand. Your userID DOE NOT need to be identical.
Example, user 200 has an ALIAS of 5625551000
Creating users with the same value of a DID really hampers you
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
to 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
-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
*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 the CDR logs show
...@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
@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
ujj
...@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 siptrace would be most useful first, I am not sure a full
snapshot would be required at this time
@list.sipfoundry.org
Subject: Re: [sipx-users] SIP Trunk Setup questions
yes.
On Mon, Aug 9, 2010 at 1:33 PM, Ujjval Karihaloo
ujj...@simplesignal.commailto:ujj...@simplesignal.com wrote:
Thx,
Is this what I do to get a SIP trace that will help?
http://sipx-wiki.calivia.com/index.php
...@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 ujj...@simplesignal.com
wrote
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
) [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
From: sipx-users-boun
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]
I am getting following
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
From: sipx-users-boun...@list.sipfoundry.org [
sipx-users-boun
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
Graziano' tgrazi...@myitdepartment.net
Cc: 'Michael Scheidell' michael.scheid...@secnap.com;
sipx-users@list.sipfoundry.org 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
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
Hi,
I have removed the subnet 10.0.0.0. from the list..
In system - internet calling - intranet subnets is just the subnet
192.168.1.0/24
that didn't resolved the problem..
i have one more concern..
Do I have to tell the provider to return the sip packets to sbc on port 5080
or can I translate
The provider should send you the packets on port 5080 if possible.
5080 is for receiving calls from ITSP's, while when sending calls to the
ITSP you send them on 5060.
Please see:
http://blog.myitdepartment.net/wp-content/uploads/2009/11/Call-Setup-Example-sipXecs-through-ITSP1.pdf
On Wed,
On 7/28/10 7:58 AM, Tony Graziano wrote:
The provider should send you the packets on port 5080 if possible.
5080 is for receiving calls from ITSP's, while when sending calls to
the ITSP you send them on 5060.
Please see:
Subject: Re: [sipx-users] sip trunk - ITSP timed out
On 7/28/10 7:58 AM, Tony Graziano wrote:
The provider should send you the packets on port 5080 if possible.
5080 is for receiving calls from ITSP's, while when sending calls to
the ITSP you send them on 5060.
Please see:
http
Hi,
See XX-8499. I tightened up error handling on the 4.2 and main
branches. If you are using 4.0.4, you can look at the diff
corresponding to this issue and apply and rebuild sipxbridge.jar.
Regards,
Ranga
On Tue, Jun 1, 2010 at 2:10 PM, Sven Evensen sven.even...@onrelay.com wrote:
Ranga,
My conclusion regarding 401 being ignored is the fact that two 401 are received
on same trunk, then a 200 OK is received, but the trunk is still disabled.
And why does the trunk not try to register again, ever. Until sipXBridge
restarted?
I will try to use IP addresses like you suggest, see if
June 2010 00:38
To: Sven Evensen; mra...@gmail.com
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIP trunk stops registering
portahk.cordiaip.netx
It can't resolve the A record for the above entry. If it can't find it, it
can't register to it. Check the DNS resolver for your amazon
@list.sipfoundry.org
Subject: Re: [sipx-users] SIP trunk stops registering
portahk.cordiaip.netx
It can't resolve the A record for the above entry. If it can't find it, it
can't register to it. Check the DNS resolver for your amazon host, you might
also check the dns timing (latency) from your host
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Sven Evensen
[sven.even...@onrelay.com]
The problem is a little more subtle, I will try to explain. And I will send you
a snapshot too.
sipX sends
I see I have a typo in my scenario below.
should be cseq=2 in line 5, which is a fault from the ITSP?:
sipX sends REGISTER cseq=1.
after 500ms sipX resends REGISTER cseq=1.
a few ms later 401 cseq=1 is received for first REGISTER.
sipX sends new REGISTER with nonce cseq=2.
sipX receives 401
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