On Mon, Aug 6, 2012 at 9:06 AM, Sven Evensen wrote:
> Is the newest jsip version included in sipx 4.6?
no
> We are unsure if we can upgrade jsip without doing a full inhouse test
> first.That will take a little time.
i agree
> Do you have any suggestions what we can do or how we can troubleshoot
Ranga,
Is the newest jsip version included in sipx 4.6?
We are unsure if we can upgrade jsip without doing a full inhouse test
first.That will take a little time.
Do you have any suggestions what we can do or how we can troubleshoot this
more before we update jsip? I dont want to tell customer w
Please update jsip (as I've suggested a few times before ). Then see if
there is issues with it and please post s new trace. I'll post a new jsip
jar here soon but you can roll your own as well
On Aug 2, 2012 12:26 PM, "Sven Evensen" wrote:
> I dont have a whole lot of info, but the customer repo
I dont have a whole lot of info, but the customer reported that the SIP
trunk was blocked and a restart of SIP trunking on our side resolved it.
And this is supposedly 3-4 time it happens.
What I do see in the attached sipxbridge.log is a semaphore timeout
at 2012-07-30T09:05:14.779000Z followed b
Setup an unmanaged gateway so sipXbridge knows to allow those calls.
On Jul 19, 2012 7:01 PM, "Kurt Albershardt" wrote:
> Thanks - in this case it's a carrier account that uses IP-based
> authorization so there is no registration.
>
> How would I associate the source IP with the gateway instance?
Thanks - in this case it's a carrier account that uses IP-based authorization
so there is no registration.
How would I associate the source IP with the gateway instance? The only place
I see fields which *might* be relevant is under Advanced Settings in ITSP
Account.
Or should I configure ano
In general sipxbridge will allow inbound calls from any IP. However,
it will not be able to identify the associated ITSP account unless you
instruct it as to where to expect inbound signaling from. The
implications of this are that when transfers occur, appropriate
rewrite rules will not be applied
Probably, yes...
On Jul 19, 2012 5:33 PM, "Kurt Albershardt" wrote:
> Do I need to configure a separate instance of sipXbridge if my provider
> sends inbound calls from a different IP than the one to which I send
> outbound calls?
>
>
> --thanks
>
> ___
Do I need to configure a separate instance of sipXbridge if my provider sends
inbound calls from a different IP than the one to which I send outbound calls?
--thanks
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.
Yes.
On Fri, Jul 13, 2012 at 4:12 PM, Kurt Albershardt wrote:
> http://wiki.sipfoundry.org/display/sipXecs/SIP+Trunking does not seem to
> describe the 4.4 user experience.
>
> I an option found under Devices/Gateways to add a new gateway called "SIP
> trunk" which has an option to "Use built-i
http://wiki.sipfoundry.org/display/sipXecs/SIP+Trunking does not seem to
describe the 4.4 user experience.
I an option found under Devices/Gateways to add a new gateway called "SIP
trunk" which has an option to "Use built-in SIP Trunk SBC"
Is this the method of managing sipXbridge in 4.4?
--t
On Sat, Jun 16, 2012 at 1:37 AM, Matt White wrote:
George Niculae 06/15/12 5:48 PM >>>
>
>>> Here is the section in the sipxbridge.log. This error occurs after
>>> sipxbridge sends the Re-invite and the Seiemens sends back the 200 OK
>>>
>>>
>>
>>You're running an fairly old version there, s
>>> George Niculae 06/15/12 5:48 PM >>>
>> Here is the section in the sipxbridge.log. This error occurs after
>> sipxbridge sends the Re-invite and the Seiemens sends back the 200 OK
>>
>>
>
>You're running an fairly old version there, seems 4.2.1 (there are
>more logs added in this area in lates
On Sat, Jun 16, 2012 at 12:24 AM, Matt White wrote:
>
George Niculae 06/15/12 5:09 PM >>>
>
> On Fri, Jun 15, 2012 at 11:41 PM, Matt White
> wrote:
> George Niculae 06/15/12 4:34 PM >>>
>>
>>Debug would be the best shot
>>
>>> Perhaps setting the log to trace wouldnt get me that anyway
>>> George Niculae 06/15/12 5:09 PM >>>
On Fri, Jun 15, 2012 at 11:41 PM, Matt White wrote:
George Niculae 06/15/12 4:34 PM >>>
>
>Debug would be the best shot
>
>> Perhaps setting the log to trace wouldnt get me that anyways.
>>
>> I have fixed any VIA header issues I had earlier. But si
On Fri, Jun 15, 2012 at 11:41 PM, Matt White wrote:
George Niculae 06/15/12 4:34 PM >>>
>
>>>On Fri, Jun 15, 2012 at 11:20 PM, Matt White
>>> wrote:
>>> I'm trying to get more info from the sipxbridge logs.
>>>
>>> I've add log4j.category.org.sipfoundry.sipxbridge=trace to the
>>> /etc/sipx
>>> George Niculae 06/15/12 4:34 PM >>>
>>On Fri, Jun 15, 2012 at 11:20 PM, Matt White wrote:
>> I'm trying to get more info from the sipxbridge logs.
>>
>> I've add log4j.category.org.sipfoundry.sipxbridge=trace to the
>> /etc/sipxpbx/log4j.properties file.
>>
>
>Actually /etc/sipxpbx/log4j.prop
On Fri, Jun 15, 2012 at 11:20 PM, Matt White wrote:
> I'm trying to get more info from the sipxbridge logs.
>
> I've add log4j.category.org.sipfoundry.sipxbridge=trace to the
> /etc/sipxpbx/log4j.properties file.
>
Actually /etc/sipxpbx/log4j.properties is for sipxconfig only. For
sipxbridge you
I'm trying to get more info from the sipxbridge logs.
I've add log4j.category.org.sipfoundry.sipxbridge=trace to the
/etc/sipxpbx/log4j.properties file.
Restart the entire server and set the sipxbridge to debug via sipxconfig.
But my logs arent showing any more detail.
Has this method to get t
Ok thanks for the logs. I found the issue. You are correct 863 is
sending a re-invite after it has terminated the call with a bye. This
results to the re-invite getting processed as a new INVITE. This got
sipXbridge confused and kept on looping the invite back to itself. I
have committed a
there is indeed a loop going on there after a hold attempt. It seems
you can now reproduce this at will. sipx bridge debug level log would
help pinpoint where the loop is happening.
On 04/24/2012 05:58 PM, Sven Evensen wrote:
We have this call scenario (sipX 4.4)
Ext calls to int A (860)
A
I've seen this problem too. What I've noticed is that after about 11
minutes the service tries to register again. It might be as simple
as just reducing the timeout before it retries.
Thanks,
Tim Ingalls
Shared Communications, Inc.
801-618-2102 Office
On 11/25/20
Voted.
On Fri, Nov 25, 2011 at 9:42 AM, Tony Graziano wrote:
> http://track.sipfoundry.org/browse/XX-9982
>
>
> On Fri, Nov 25, 2011 at 7:45 AM, Joegen Baclor wrote:
>
>> Open a jira.
>>
>>
>> On 11/24/2011 08:50 PM, Tony Graziano wrote:
>>
>> I have noticed whenever there is an issue with sip
http://track.sipfoundry.org/browse/XX-9982
On Fri, Nov 25, 2011 at 7:45 AM, Joegen Baclor wrote:
> Open a jira.
>
>
> On 11/24/2011 08:50 PM, Tony Graziano wrote:
>
> I have noticed whenever there is an issue with sipx notbeing able to
> authenticate to a sip trunk provider, it gets tuck on "AU
Open a jira.
On 11/24/2011 08:50 PM, Tony Graziano wrote:
I have noticed whenever there is an issue with sipx notbeing able to
authenticate to a sip trunk provider, it gets tuck on "AUTHENTICATING"
and does not have a timeout or recycle mechnim to stop and try again
after xx seconds.
The end
I very much agree with your suggestion. I've had a number of instances
where the authentication process gets 'stuck' or gives up, and I end up
having to restart the SIP trunking by using VPN to a customer's location.
Even if it's the TSP's issue, I sometimes asked why the phone server
doesn't just
I have noticed whenever there is an issue with sipx notbeing able to
authenticate to a sip trunk provider, it gets tuck on "AUTHENTICATING" and
does not have a timeout or recycle mechnim to stop and try again after xx
seconds.
The end result is that when it is not authenticated and it fails, it
be
I'm referring to this discussion:
http://www.mail-archive.com/sipx-users@list.sipfoundry.org/msg15101.html
I'm trying to see if the tightening may be causing an issue I'm having.
I've placed this line in several places in /etc/sipxpbx/sipxbridge.xml
false
Per Matt White, this change was committed i
I think you would be able to. I am not a domain expert on this but i
have seen Ranga's work and it is generic enough to stand by itself
together with sipXrelay. Feel free to send in patches if you are able
to improve on it!
On Friday, 17 December, 2010 10:19 PM, Henry Dogger wrote:
Hi a
: vrijdag 17 december 2010 10:59
To: Discussion list for users of sipXecs software
Subject: [sipx-users] sipXbridge and sipXproxy fails
Hi all,
I installed 0.0.4.5.1 to test openACD
But I can't get rid of these errors:
- sipxbridge.log
"2010-12-17T09:37:34.
Hi all,
I would like to know if it is possible to use the sipxbridge as a
standalone service?
So not in combination with sipXecs, but with another telephony server
say asterisk, or any other?
I could provide more intel if this is to vague.
Kind regards,
Henry Dogger
Telecats BV
_
Hi all,
I installed 0.0.4.5.1 to test openACD
But I can't get rid of these errors:
- sipxbridge.log
"2010-12-17T09:37:34.85Z":8:JAVA:ERR:servername:main::sipXbr
idge:"Invalid argument address = 10.10.10.1 port = 5080 transport = udp"
"2010-12-17T09:37:34.854000Z":9:J
Hi all,
i heve made some tests (sipx 4.2.1) with an ITSP, that have DNS A record
pointing to their website, and SRV records, pointing to their SBC (Acme), whose
ip address is different from website.
And i found that sipxbridge sends Register to correct address - address given
by SRV record.
Bu
, 2010 1:49 PM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] SIPXBRIDGE Configuration Woes
No there are not symlinks.
Get thee to a reputable ITSP.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN
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: Thu Nov 04 13:47:42 2010
Subject: Re: [sipx-users] SIPXBRIDGE Configuration Woes
yes, even now when I view the
.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: Thu Nov 04 13:22:51 2010
> Subject: Re: [sipx-users] S
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: Thu Nov 04 13:22:51 2010
Subject: Re: [sipx-users] SIPXBRIDGE Configuration Woes
That is just the thing, the
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: Thu Nov 04 13:05:30 2010
> Subject:
-
From: sipx-users-boun...@list.sipfoundry.org
To: Discussion list for users of sipXecs software
Sent: Thu Nov 04 13:05:30 2010
Subject: [sipx-users] SIPXBRIDGE Configuration Woes
For weeks now my sipx configuration has been working great. The
reason why it worked well is because I set &quo
For weeks now my sipx configuration has been working great. The
reason why it worked well is because I set "loose routing" tag to
false in sipxbridge.xml. So the sip uri didn't have ;lr. Now, no
matter what I do I can't get ;lr out.
Any help is appreciated.
Thanks
_
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Keith [kei...@dakins.ca]
Is there any way to get sipXbridge (or is it sipXrelay?) to
decode audio and generate RFC2833 codes based on the audio?
Am 07.09.2010 um 14:31 schrieb Keith:
>
> Content-Type: text/plain;
> charset="utf-8"
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> Organization: SipXecs Forum
> In-Reply-To: <5f73145c36350425082d8c701989a...@mail.gmail.com>
> X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <51576>
> Message-ID:
>
>
>
>
On Tue, Sep 7, 2010 at 8:31 AM, Keith wrote:
>
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> charset="utf-8"
> Content-Transfer-Encoding: 8bit
> Organization: SipXecs Forum
> In-Reply-To: <5f73145c36350425082d8c701989a...@mail.gmail.com>
> X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <51576>
> Message-ID:
>
>
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Organization: SipXecs Forum
In-Reply-To: <5f73145c36350425082d8c701989a...@mail.gmail.com>
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <51576>
Message-ID:
I'm a bit weak on the architecture, but when using
sipXbridge,
:
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: Tue Sep 07 07:50:28 2010
Subject: [sipx-users] SipXbridge decode of DTMF
Content-Type: text/plain;
charset="utf-8"
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <51573>
Message-ID:
My ITSP is very good, but as with all of them, there is one
feature which is quite annoying. They send all DTMF inband
and d
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Scheidell
[list-s...@secnap.com]
I did find that in the 'advanced settings', I needed to put in something
(like recommended on isn site) a prefix of
On 6/24/10 2:59 PM, Tony Graziano wrote:
Mine works without 012.
Feel free to comment on the JIR. Though I hate to use prefixes
that are system specific, I never think its a good idea. I can dial
1234 it stays local, I dial 1234*256, it knows its an ISN number, does
the lookup and rou
Mine works without 012. [?]
Feel free to comment on the JIR. Though I hate to use prefixes that are
system specific, I never think its a good idea. I can dial 1234 it stays
local, I dial 1234*256, it knows its an ISN number, does the lookup and
routes the call. I don't think it should be that hard
On 6/24/10 2:31 PM, Tony Graziano wrote:
> Coincidentally, you made my point that I was suggesting enabling that
> should be in DIAL PLAN, it logically makes more sense (at least from
> enable/disable) to be placed there.
>
>
I did find that in the 'advanced settings', I needed to put in someth
Which is why I was looking for input.
My post earlier today...
"What I do find is that ISN dialing, which used to be defined "allow
ISN dialing" was located under Domain or Internet calling. Now it is
under the registrar.
Hurray it works! But, I am wondering "why" it is under registrar? I
under
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
[tgrazi...@myitdepartment.net]
Coincidentally, you made my point that I was suggesting enabling that
should be in DIAL PLAN, it logically make
Coincidentally, you made my point that I was suggesting enabling that
should be in DIAL PLAN, it logically makes more sense (at least from
enable/disable) to be placed there.
On Thu, Jun 24, 2010 at 2:03 PM, Michael Scheidell wrote:
> On 6/24/10 2:00 PM, Tony Graziano wrote:
>
> gimme a break...
On 6/24/10 2:00 PM, Tony Graziano wrote:
gimme a break... are you really asking that? Where are all the
services in sipx listed? I know you know the answer.
brain fried, looking for ^Registrar.. didn't see SIP registerar.
O
--
Michael Scheidell, CTO
Phone: 561-999-5000, x 1259
> *
On 6/24/10 1:57 PM, Michael Scheidell wrote:
where is the registrar in sipx?
never mind, found it.
--
Michael Scheidell, CTO
Phone: 561-999-5000, x 1259
> *| *SECNAP Network Security Corporation
* Certified SNORT Integrator
* 2008-9 Hot Company Award Winner, World Executive Alliance
gimme a break... are you really asking that? Where are all the
services in sipx listed? I know you know the answer.
On Thu, Jun 24, 2010 at 1:57 PM, Michael Scheidell wrote:
> where is the registrar in sipx?
>
>
> On 6/24/10 1:54 PM, Tony Graziano wrote:
>
> Um. If you go to the registrar in sipx
where is the registrar in sipx?
On 6/24/10 1:54 PM, Tony Graziano wrote:
Um. If you go to the registrar in sipx and turn on ISN dialing, you
can dial 1234*256. The "*" means "@" and "256" will resolve to
"loligo.com". That's what its all about!
On Thu, Jun 24, 2010 at 1:48 PM, Michael Scheidel
Um. If you go to the registrar in sipx and turn on ISN dialing, you
can dial 1234*256. The "*" means "@" and "256" will resolve to
"loligo.com". That's what its all about!
On Thu, Jun 24, 2010 at 1:48 PM, Michael Scheidell wrote:
>
>
> On 6/24/10 1:39 PM, Tony Graziano wrote:
>
> No. If you dont
On 6/24/10 1:39 PM, Tony Graziano wrote:
No. If you dont use an ITAD it does not matter.
interesting. yes, I remember this from a while back, forgot all about it.
oh, I can't dial 1234*256
we got a 4 digit extension and 1234 belongs to the CFO :-(
It's a global DB (you configure the r
No. If you dont use an ITAD it does not matter.
It's a global DB (you configure the records in their system after a
free application is approved). Optionally you can have the lookups
done in your DNS. There's a lookup and DNS component from their
registry when they host the records, and points bac
On 6/24/10 10:00 AM, Tony Graziano wrote:
where is 'registrar'?
I don't remember doing anything to enable or disable ISN/SIP calling
(you can sip me on my extention @, or any alias, including alphanumeric
aliases)
so you are saying this is a function of your ITSP, when you register
y
I think ISN is a great idea. We implement it and so do a lot of
customers. I'm just trying to make it easier to enable for others. ISN
assignments are free, and very handy to dial a sip uri eqivalent from
a numeric keypad.
On Thu, Jun 24, 2010 at 9:56 AM, Michael Scheidell wrote:
> On 6/24/10 8:5
On 6/24/10 8:58 AM, Tony Graziano wrote:
What I do find is that ISN dialing, which used to be defined "allow
ISN dialing" was located under Domain or Internet calling. Now it is
under the registrar.
where is 'registrar'?
I don't remember doing anything to enable or disable ISN/SIP calling
Ok. I am dredging this back up... sorry.
In 4.2 I can do Internet dialing by default using the default sbc,
sipXbridge-1. I do not have to enable Internet dialing by default to
dial by sip URI.
What I do find is that ISN dialing, which used to be defined "allow
ISN dialing" was located under Doma
Thanks guys.. I have already solved the problem.. ;)
On Tue, Jun 15, 2010 at 5:29 PM, Tony Graziano wrote:
>
> Content-Type: text/plain;
> charset="utf-8"
> Content-Transfer-Encoding: 8bit
> Organization: SipXecs Forum
> In-Reply-To: >
> X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <47798>
>
Content-Type: text/plain;
charset="utf-8"
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Organization: SipXecs Forum
In-Reply-To:
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Message-ID:
(hint: subscribe to sip-users and get a wider audience and
faster response)
Remember you will lose your config
Hi,
I have a question.. I have deleted the internal bridge from the server. How
can I configure it again? I must use it with ITSP Account. I have tried to
configure an unmanaged sbc for it but it won't register.. any advice?
--
Irena Dolovčak
___
sipx-
On 11 maj 2010, at 20.27, M. Ranganathan wrote:
> On Tue, May 11, 2010 at 2:02 PM, Staffan Kerker wrote:
>> Hi
>>
>> I just noticed that sipXbridge seems to remove the crytpo-attributes in SDP
>> if a call setup with SRTP information
>> is recieved. The RTP/SAVP profile is still there, but the
On Tue, May 11, 2010 at 2:02 PM, Staffan Kerker wrote:
> Hi
>
> I just noticed that sipXbridge seems to remove the crytpo-attributes in SDP
> if a call setup with SRTP information
> is recieved. The RTP/SAVP profile is still there, but the crypto attributes
> are gone...
Since sipxbridge/sipxr
Hi
I just noticed that sipXbridge seems to remove the crytpo-attributes in SDP if
a call setup with SRTP information
is recieved. The RTP/SAVP profile is still there, but the crypto attributes are
gone...
--- SDP in INVITE sent to sipXbridge (outgoing call)
v=0
o=- 1273600686040637 127360068
On Sun, Apr 25, 2010 at 1:09 PM, Rene Pankratz
wrote:
>> Well, if there is enough demand, I can make the refresher choice
>> dynamic. The other possibility is to suppress Session timer altogether
>> and simply rely on periodic re-INVITE to check for liveness of the
>> session.
>
>
> Is it possible
>
> Well, if there is enough demand, I can make the refresher choice
> dynamic. The other possibility is to suppress Session timer altogether
> and simply rely on periodic re-INVITE to check for liveness of the
> session.
>
Is it possible to achieve one of these possibilities without needing anot
On Fri, Apr 23, 2010 at 7:48 AM, Rene Pankratz
wrote:
> Hello list members,
> we are evaluating a VoIP provider that is used as SIP Trunk (www.qsc.de, the
> product is named "IPFonie").
> Incoming calls are working without any problems.
>
> But when we are trying to place a call the INVITE sent by
Hello list members,
we are evaluating a VoIP provider that is used as SIP Trunk (www.qsc.de, the
product is named "IPFonie").
Incoming calls are working without any problems.
But when we are trying to place a call the INVITE sent by SipX contains the
Session-expires header with the value
"Session-
Scott, Dale,
Thanks a lot for the help.
Nikolay.
> -Original Message-
> From: Scott Lawrence [mailto:xmlsc...@gmail.com]
> Sent: Wednesday, April 21, 2010 10:13 PM
> To: Nikolay Kondratyev
> Cc: sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] sipxbridge and
On Wed, 2010-04-21 at 17:00 +0400, Nikolay Kondratyev wrote:
>
> > Support for re-INVITE (no SDP) in order to solicit a SDP
> > offer is mandatory. There is no way to avoid this.
> Can you please point me to the appropriate rfc? 3261?
RFC 3261
Section 13.2.1 Creating the Initial INVITE (page 79
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nikolay Kondratyev
[k...@nstel.ru]
> Support for re-INVITE (no SDP) in order to solicit a SDP
> offer is mandatory. There is no way to avoid this.
Can you
On Wed, Apr 21, 2010 at 9:00 AM, Nikolay Kondratyev wrote:
>
>
>> Support for re-INVITE (no SDP) in order to solicit a SDP
>> offer is mandatory. There is no way to avoid this.
> Can you please point me to the appropriate rfc? 3261?
>
>> Ranga
> Thanks and regards,
> Nikolay.
Please search the a
> Support for re-INVITE (no SDP) in order to solicit a SDP
> offer is mandatory. There is no way to avoid this.
Can you please point me to the appropriate rfc? 3261?
> Ranga
Thanks and regards,
Nikolay.
___
sipx-users mailing list sipx-users@list.si
> On Wed, Apr 21, 2010 at 2:10 AM, Nikolay Kondratyev wrote:
> > Hi all,
> > i have a question regarding "late media" use in sipxbridge...
> > When incoming call is going through sipxbridge and is transferred by the
> > phone or by AA, sipxbridge converts Refer into re-Invite without sdp.
> > I h
On Wed, Apr 21, 2010 at 2:10 AM, Nikolay Kondratyev wrote:
> Hi all,
> i have a question regarding "late media" use in sipxbridge...
> When incoming call is going through sipxbridge and is transferred by the
> phone or by AA, sipxbridge converts Refer into re-Invite without sdp.
> I have installat
Hi all,
i have a question regarding "late media" use in sipxbridge...
When incoming call is going through sipxbridge and is transferred by the phone
or by AA, sipxbridge converts Refer into re-Invite without sdp.
I have installation where the equipment to which sipx is connected does not
support
esday, April 14, 2010 1:42 PM
To: Sipx-users list
Subject: [sipx-users] sipxbridge with one server, two branches
I am trying to understand what would be the simplest way to use two
branches for one system with sipxbridge to be able to separate the
trunks/dialplans and ultimately the gateway the
I am trying to understand what would be the simplest way to use two
branches for one system with sipxbridge to be able to separate the
trunks/dialplans and ultimately the gateway they use.
Short of putting a second server in, I have a customer consolidating
two branches for a period of time. I don
dì, 17 marzo 2010 13:21:38 GMT +01:00
Amsterdam/Berlino/Berna/Roma/Stoccolma/Vienna
Oggetto: [sipx-users] sipxbridge does not send authorization
Thank you Michael for your suggestion. I did it but that was not the problem.
I discovered the problem is probably related to the firewall. When I ge
Deidda" , sipx-users@list.sipfoundry.org
Inviato: Martedì, 9 marzo 2010 13:12:52 GMT +01:00
Amsterdam/Berlino/Berna/Roma/Stoccolma/Vienna
Oggetto: RE: [sipx-users] sipxbridge does not send authorization
I would upgrade to the newer version of scs... this will bring you up
to 4.0.4.
Mike
-Origina
On 3/9/10 11:34 AM, M. Ranganathan wrote:
You are correct.
Specifically, if you want two sipxbridge instances,
1. Specify different external ports for each.
2. Make sure you do not register with the same ITSP for both instances
for the same account (unless your ITSP allows that -- and few do
Thanks!
-Original Message-
From: M. Ranganathan [mailto:mra...@gmail.com]
Sent: Tuesday, March 09, 2010 10:34 AM
To: Scott Lawrence
Cc: Ken Fulmer; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] sipXbridge Redundancy
On Tue, Mar 9, 2010 at 11:30 AM, Scott Lawrence
wrote:
>
On Tue, Mar 9, 2010 at 11:30 AM, Scott Lawrence wrote:
> On Tue, 2010-03-09 at 11:19 -0500, M. Ranganathan wrote:
>> On Tue, Mar 9, 2010 at 11:10 AM, Ken Fulmer
>> wrote:
>> > The design doc mentioned one sipXbridge per cluster. The link above
>> > suggests
>> > two can be used in a HA setup. Ca
On Tue, 2010-03-09 at 11:19 -0500, M. Ranganathan wrote:
> On Tue, Mar 9, 2010 at 11:10 AM, Ken Fulmer
> wrote:
> > The design doc mentioned one sipXbridge per cluster. The link above suggests
> > two can be used in a HA setup. Can anyone clarify?
> >
> >
> >
> > http://www.sipfoundry.org/componen
On Tue, Mar 9, 2010 at 11:10 AM, Ken Fulmer
wrote:
> The design doc mentioned one sipXbridge per cluster. The link above suggests
> two can be used in a HA setup. Can anyone clarify?
>
>
>
> http://www.sipfoundry.org/component/content/article/29-test-read-more-link-page.html
Sipxbridge distribut
The design doc mentioned one sipXbridge per cluster. The link above suggests
two can be used in a HA setup. Can anyone clarify?
http://www.sipfoundry.org/component/content/article/29-test-read-more-link-p
age.html
Thanks,
Ken Fulmer
___
@list.sipfoundry.org
Subject: [sipx-users] sipxbridge does not send authorization
I have a SIPX 4.0.1-015823 (SCS 3.0) and I tried to setup a SIP trunk
towards Eutelia.
This is not the first SIP trunk I setup with SIPXBridge and Eutelia so I
expected an easy job but.. the trunk doesnt register.
Looking at the
I have a SIPX 4.0.1-015823 (SCS 3.0) and I tried to setup a SIP trunk towards
Eutelia.
This is not the first SIP trunk I setup with SIPXBridge and Eutelia so I
expected an easy job but.. the trunk doesnt register.
Looking at the capture pcap file I see this:
24 172.16.172.283.211.227.21
Does sipxbridge have to be behind nat in order for remote users or trunking
to work? For trunking I can get inbound calls and make calls, whether or not
I use public address for call setup, but have consistent audio issues.
I've been trialling a standby system that I could use at a central locatio
On Sun, 2010-02-07 at 15:02 +, Sven Evensen wrote:
> A does not do anything, that is external mobile.
> Our app uses REFER to get from B to C, then a redirect (INVITE) to get
> from C to D. When A hangs up, the call does not drop because sipXBridge
> ignores the BYE.
You're going to have to ex
-Original Message-
From: Dale Worley [mailto:dwor...@avaya.com]
Sent: 06 February 2010 14:15
To: Sven Evensen
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] sipXbridge ignoring a BYE
On Sat, 2010-02-06 at 13:22 +, Sven Evensen wrote:
> A Mobile 07795951717
>
> B
On Sat, 2010-02-06 at 13:22 +, Sven Evensen wrote:
> A Mobile 07795951717
>
> B Internal phantom user 8116
>
> C Internal phantom user 8216
>
> D Mobile 07791788997
>
>
>
> B calls A, A answers
>
> B transferred to C
>
> C transferred to D
>
> A hangs up before D answers
When you say
I have this same issue or very similar issue.
A - call comes from anywhere to my VoIP DID
B - Call is routed to a Phantom ext using SIP Alias of the DID in the Alias
section of the Phantom DID
C - Call is forwarded always to the Auto Attendant setup for this DID
D - User selects an option inside t
On Thu, 2010-02-04 at 15:59 +, Sven Evensen wrote:
> Thanks for all the good advice guys.
>
> We solved it by adding a tag to the Refer-To URI. The INVITE that
> eventually comes from the sipXBridge contained that tagged and our
> algorithm worked well.
>
> We still need a generic solution th
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