Hi, Ranga,
I confirm that we see a similar transfer problem with sipxbridge. Our
version is 4.1.6-017914.
I send the snapshot in a personal mail.
Regards,
Gabor
-Original Message-
From: M. Ranganathan [mailto:mra...@gmail.com]
Sent: 14 February 2010 23:49
To: Pizza Napoletana
Cc: sipx-
I tried once OpenSIPS as load balancer, it integrated nicely with SipX.
OpenSIPS is a very powerful tool (it means configuring it properly is a
challenge ;-)).
Regards,
Gabor
-Original Message-
From: Robert B [mailto:d...@spudland.com]
Sent: 08 December 2009 22:44
To: sipx-users@list.sip
- either in the test program or in the JAIN-SIP stack it uses.
I continue the investigation.
Regards,
Gabor
-Original Message-
From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
Sent: 04 December 2009 18:49
To: Dale Worley
Cc: Gabor Paller; sipx-users@list.sipfoundry.org
Subject
al Message-
From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
Sent: 04 December 2009 18:49
To: Dale Worley
Cc: Gabor Paller; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Retransmission timeout values for SipX
On Fri, 2009-12-04 at 13:25 -0500, Dale Worley wrote:
>
> Exactly w
server's performance. My idea is
that I start to fiddle with retransmission timers.
Gabor Paller, Software Architect
OnRelay
Elizabeth House │ 39 York Road, London SE1 7NQ, UK │ +44 (0) 207 902 8151 │
gabor.pal...@onrelay.com <mailto:j...@onrelay.com> │ www.onrelay.com
<http://w
I have just seen such a case and it was caused by UDP retransmission.
-Original Message-
From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
Sent: 01 December 2009 13:14
To: Wen Jun
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Why there are 2 SIP INVITE in a call to ca
load requires quite an amount of
memory (beside CPU load). Is this conclusion correct?
Regards,
Gabor
-Original Message-
From: Kathleen Eccles [mailto:kecc...@nortel.com]
Sent: 27 November 2009 01:19
To: Gabor Paller
Cc: sipx-users
Subject: RE: [sipx-users] removeOldTransactions
On Thu,
age-
From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
Sent: 26 November 2009 13:04
To: Gabor Paller
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] removeOldTransactions
On Thu, 2009-11-26 at 12:57 +, Gabor Paller wrote:
> Hi,
>
>
>
> I am chasing a bug in
lto:tgrazi...@myitdepartment.net]
Sent: 26 November 2009 13:01
To: Gabor Paller; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] removeOldTransactions
Its "normal" to see the memory usage in sipx go up pretty quickly and stay
there. Its the nature of the way java interacts with the system
es this show that we somehow mismanage the transactions in our load test
tool?
Gabor Paller, Software Architect
OnRelay
Elizabeth House │ 39 York Road, London SE1 7NQ, UK │ +44 (0) 207 902 8151 │
gabor.pal...@onrelay.com <mailto:j...@onrelay.com> │ www.onrelay.com
<http://www.onre
h product from the 4 paying Counterpath
products I am aware of
(Bria, Bria Professional, Bria for Microsoft Outlook, eyebeam).
Regards,
Gabor
-Original Message-
From: Dale Worley [mailto:dwor...@nortel.com]
Sent: 18 November 2009 15:46
To: Gabor Paller
Cc: sipx-users@list.sipfoundry.org
Subj
-
From: Dale Worley [mailto:dwor...@nortel.com]
Sent: 17 November 2009 14:22
To: Gabor Paller
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] In-call status?
On Tue, 2009-11-17 at 11:10 +0000, Gabor Paller wrote:
> The problem with it is that many phones - e.g. Cisco 7940 but also
r
-Original Message-
From: Dale Worley [mailto:dwor...@nortel.com]
Sent: 17 November 2009 14:22
To: Gabor Paller
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] In-call status?
On Tue, 2009-11-17 at 11:10 +, Gabor Paller wrote:
> The problem with it is that many phon
t RFC 4235. SipXproxy, however, is aware of the in-call status by
other means, e.g. by monitoring the SIP dialog. Is there an easy way to access
that information from an external program?
Gabor Paller, Software Architect
OnRelay
Elizabeth House │ 39 York Road, London SE1 7NQ, UK │ +44 (0) 20
There is one place where it has to be "turned on", if the user does not
have voicemail permission ("Permissions" link at the left side of the
user management page), then it can't subscribe to MWI notifications
either. I don't believe that this is your problem, however, because this
permission is gr
I thought that the outgoing call would be a new call.
Regards,
Gabor
-Original Message-
From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
Sent: 20 October 2009 16:52
To: Gabor Paller
Cc: sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] From: manipulation
On Tue, 2009-10-2
rom: Scott Lawrence [mailto:scott.lawre...@nortel.com]
Sent: 20 October 2009 15:52
To: Gabor Paller
Cc: sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] From: manipulation
On Tue, 2009-10-20 at 15:27 +0100, Gabor Paller wrote:
> That's what I thought but I don't know SipX we
al Message-
From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
Sent: 20 October 2009 15:15
To: Gabor Paller
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] From: manipulation
On Tue, 2009-10-20 at 10:36 +0100, Gabor Paller wrote:
> Quick question: is anyone aware of any
Quick question: is anyone aware of any scenario when SipX or any parts of it
manipulates the incoming caller ID (the From: header in SIP messages)?
Regards,
Gabor
Gabor Paller, Software Architect
OnRelay
Elizabeth House │ 39 York Road, London SE1 7NQ, UK │ +44 (0) 207 902 8151
Thanks for your answers.
The source of the problem was eventually some obscure IP-stack problem
on the secondary machine. Even though the machine seemed to operate
correctly, it rejected incoming SIP packets therefore the clients felt
that the failover was not successful. Restarting the machine so
ks for you. Unfortunately I can't help you with
CUBE-specific questions because I don't have any experience with CUBE.
Regards,
Gabor
-Original Message-
From: Ken Fulmer [mailto:kenful...@icstechnologysolutions.com]
Sent: 16 October 2009 14:17
To: Gabor Paller; 'Josh Patten';
" Also, to call the Cisco device "deficient" seems harsh to me."
Cisco IOS handles REFERs starting from 12.3 (as far as I was able to dig
out). I have very good experiences with Cisco gateways handling REFERs.
If, however, REFER is really impossible, you can pull the call through
sipxbridge that
Can that cause problems?
I understood that many clients (e.g. the Cisco gateway) simply ignore those.
Regards,
Gabor
From: Picher, Michael [mailto:mpic...@cmctechgroup.com]
Sent: 14 October 2009 11:11
To: Gabor Paller; M. Ranganathan
Cc: sipx-users
.
;; Query time: 0 msec
;; SERVER: 127.0.0.1#53(127.0.0.1)
;; WHEN: Wed Oct 14 10:23:57 2009
;; MSG SIZE rcvd: 375
[r...@kennedy ~]#
Regards,
Gabor
____
From: M. Ranganathan [mailto:mra...@gmail.com]
Sent: 14 October 2009 10:22
To: Gabor Paller
C
. :-/
Regards,
Gabor
Gabor Paller, Software Architect
OnRelay
Elizabeth House │ 39 York Road, London SE1 7NQ, UK │ +44 (0) 207 902 8151 │
gabor.pal...@onrelay.com <mailto:j...@onrelay.com> │ www.onrelay.com
<http://www.onrelay.com/>
OnRelay in the News:
OnRelay Named i
If the carries supports SMS delivery report feature, you can detect if
an SMS is undelivered.
Regards,
Gabor
-Original Message-
From: Keith Gearty [mailto:ke...@glensound.co.uk]
Sent: 13 October 2009 09:01
To: Dennis Wallen
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Voi
Does the user have voicemail permission? (Go to users/users, click to
the user ID, click permissions on the left pane, both "voice mail" and
"internal voicemail server" must be checked (provided you use internal
voicemail)).
Regards,
Gabor
From: Chris Tres
the caller can hear
the person that answered, but the person that answered cannot hear the
caller
and
4. If the call is not answered, the phones stop ringing for a second,
and then ring again. Apparently I need to do something more so that the
call will go to the automated attendant if no
"MUST fix list:
-Not sure how to configure sipXecs to ring all extensions for
inbound calls. Ideally, I would like it to ring all extensions for 8
rings, and then go to the automated attendant if no one answers. I can
direct inbound calls to any given extension OK, but I'm not sure now t
ibeServerThread::isAuthenticated()
- No Credentials for mailboxUrl=\"sip:4...@10.1.9.23\",
reqRealm=\"sipxecs.test.onrelay.local\", reqUser=\"404\""
Is this normal or a glitch?
Regards,
Gabor
Gabor Paller, Software Architect
OnRelay
Elizabeth House │ 39 Yo
t Lawrence [mailto:scott.lawre...@nortel.com]
Sent: 24 September 2009 12:40
To: Gabor Paller
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] 407 Proxy Authentication Required question
[please remember not to reply to an unrelated message when starting a
new topic]
On Thu, 2009-09-24 Ga
Hi,
I know that this question has been discussed many times on this list but
I lost track of the current state of the rules when SipX challenges an
incoming INVITE for credentials.
So the question is: what are the exact rules which determine whether
SipX challenges an incoming INVITE for authenti
Well, I will have to rewrite a quite tricky part of my program. I will
open an issue but I can't avoid rewrite. :-(
Regards,
Gabor
-Original Message-
From: M. Ranganathan [mailto:mra...@gmail.com]
Sent: 10 July 2009 15:37
To: Gabor Paller
Cc: sipx-users@list.sipfoundry.org
Subjec
rds,
Gabor
-Original Message-
From: M. Ranganathan [mailto:mra...@gmail.com]
Sent: 10 July 2009 15:18
To: Gabor Paller
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] UPDATE method and sipxbridge
On Fri, Jul 10, 2009 at 2:52 AM, Gabor Paller wrote:
> " It does not forward
cannot replace UPDATE in some scenarios, e.g. RFC 3960.
-Original Message-
From: M. Ranganathan [mailto:mra...@gmail.com]
Sent: 10 July 2009 04:40
To: Gabor Paller
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] UPDATE method and sipxbridge
On Thu, Jul 9, 2009 at 11:43 AM, Gabor
.
Regards,
Gabor
Gabor Paller, Software Architect
OnRelay
Elizabeth House | 39 York Road, London SE1 7NQ, UK | +44 (0) 207 902
8151 | gabor.pal...@onrelay.com <mailto:gabor.pal...@onrelay.com> |
www.onrelay.com <http://www.onrelay.com/>
OnRelay in the News:
BT Trials OnRelay
Personal experience: we use both Nokia E65 and E90 over Wi-Fi with SipX
and they work just fine. No additional software is necessary. I guess,
Nokia E category phones all have the SIP client in them but I have
experience only with the two models above.
Regards,
Gabor
__
SipXbridge is used in a scenario when the caller receives a re-INVITE
from the called party. The SIP trunk challenges the re-INVITE. The
result is that sipxbridge throws an exception (see attached log) and
fails the call.
Any idea, how to correct this? It happens in 4.0.1.
Regards, Gabor
AUS-VM
Thanks for the response. The particular SIP trunk provider uses OPTIONS
as keep-alive and they are happy with any response, including error
response.
Regards,
Gabor
-Original Message-
From: M. Ranganathan [mailto:mra...@gmail.com]
Sent: 04 July 2009 02:49
To: Gabor Paller
Cc: sipx-users
Server: sipXecs/4.0.1 sipXecs/sipxbridge (Linux)
Content-Length: 0
Gabor Paller, Software Architect
OnRelay
Elizabeth House | 39 York Road, London SE1 7NQ, UK | +44 (0) 207 902
8151 | gabor.pal...@onrelay.com <mailto:gabor.pal...@onrelay.com> |
www.onrelay.com <http://
Hi,
Create a custom dial plan. The number pattern should match the numbers
used on CCM, the gateway should point to your trunk and only "Local
Dialing" permission should be required.
Regards,
Gabor
From: arda savran [mailto:ardasav...@hotmail.com]
Sen
"It is used by the proxy on sipx1 when it wants to send a request to
"the
registrar"."
Is it so that in HA mode, sipxproxy queries for this quasi-domain to
obtain the registrar's address? So the "rr" comes from the way sipxproxy
implemented?
Regards,
Gabor
___
Spiralling SIP requests due to an erroneous dial plan? (proxy sends back
the SIP requests to itself).
Check sipXproxy.log in /var/log/sipxpbx.
Regards,
Gabor
-Original Message-
From: Bernardo Ortega [mailto:jbort...@fschad.com]
Sent: 01 July 2009 16:13
To: SIPxecs Support
Subject: [sipx
.example.com.
_sip._tcp.rr.sipx1.example.com. IN SRV 2 100 5070 sipx2.example.com.
_sip._udp.rr.sipx1.example.com. IN SRV 4 100 5070 sipx2.example.com.
What is this rr.sipx1.example.com domain?
Regards,
Gabor
Gabor Paller, Software Architect
OnRelay
Elizabeth House | 39 York Road, London SE1
.
Am I close to the reality or have I overlooked something?
Regards,
Gabor
Gabor Paller, Software Architect
OnRelay
Elizabeth House | 39 York Road, London SE1 7NQ, UK | +44 (0) 207 902
8151 | gabor.pal...@onrelay.com <mailto:gabor.pal...@onrelay.com> |
www.onrelay.com
Have you activated the dial plan?
Also, you can try to run the network analyser on the node itself. Sipx
components talk to each other using SIP (inside the same node), if you
capture that traffic, you will find where the call fails.
Regards,
Gabor
-Original Message-
From: Jiann-Ming Su [
"How is that? and what to do about it?"
Please, check out M. Ranganathan's extremely useful introduction to
Hosted NAT compensation.
http://list.sipfoundry.org/archive/sipx-users/msg14849.html
There is a setting somewhere on sipXecs admin console (I forgot, where,
sorry :-( ) where you can set u
last
sent packet) interferes with the Microsoft Speech Server?
-Original Message-
From: roman gelfand [mailto:rgelfa...@yahoo.com]
Sent: 16 June 2009 16:28
To: Gabor Paller; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Microsoft Exchange UM Voicemail Server Issue
Yes
Is sipxbridge involved somehow in the faulty case?
Sipxproxy does not deal with media streams, hence cannot cause media faults.
-Original Message-
From: roman gelfand [mailto:rgelfa...@yahoo.com]
Sent: 16 June 2009 12:59
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] Microsoft E
Thanks. What is the obstacle?
Regards,
Gabor
-Original Message-
From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
Sent: 15 June 2009 14:26
To: Gabor Paller
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] IPv6 support
On Mon, 2009-06-15 at 14:00 +0100, Gabor Paller
Hi,
I was searching the wiki for any statement about IPv6 support but I
couldn't find anything. What is the situation of the IPv6 support in
sipXecs?
Regards,
Gabor
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive: http://list.sip
s/ids into that field and nothing happened. Is it just a bug in my
system or is there a hidden catch (like a permission)?
Regards,
Gabor
Gabor Paller, Software Architect
OnRelay
Elizabeth House | 39 York Road, London SE1 7NQ, UK | +44 (0) 207 902
8151 | gabor.pal...@onrela
rding at the same time on Snom 300. :-/
Regards,
Gabor
-Original Message-
From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
Sent: 18 March 2009 16:20
To: Gabor Paller
Cc: Dale Worley; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] FW: Forwarding problem with Snom 300
O
" Go to internet calling and select "server behind NAT""
Shame on me. I assumed it was selected and it wasn't. :-(
Regards,
Gabor
-Original Message-
From: M. Ranganathan [mailto:mra...@gmail.com]
Sent: 20 May 2009 16:29
To: Gabor Paller
Cc: Tony Graziano; sipx-
se for RTP
keepalive.
Regards,
Gabor
-Original Message-
From: M. Ranganathan [mailto:mra...@gmail.com]
Sent: 20 May 2009 15:55
To: Gabor Paller
Cc: Tony Graziano; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] sipXbridge+firewall
On Wed, May 20, 2009 at 10:12 AM, Gabor Pal
>
>If you are using a siptrunk and have it set for "Use public address for
>call setup", can it be assumed the HNC is not applicable?
I can state that even if that box is checked, the c= line carries the
local address (1.2.3.4 in the example).
___
sipx-
.20.10 SIP address trunk address. I can't imagine
how the media gets through.
Regards,
Gabor
-Original Message-
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: 20 May 2009 15:07
To: sipx-users@list.sipfoundry.org; Gabor Paller
Subject: Re: [sipx-users] sipXbridge+firewa
Hi,
I have a seemingly very simple configuration with SipX 4.0 (sipxbridge
activated), an authenticated SIP trunk and a firewall. IP addresses are
the following (not actual addresses):
1.2.3.4: IP address of the SipX box, also hosting sipxbridge
1.2.3.40: A SIP endpoint on the internal network, sa
Has anyone noticed this?
The SipX in question is the standard ("pub") SipX 4.0.0. The test
scenario is very simple: I call an extension that has no endpoint
registered from a desk phone. The voicemail triggers and then I leave a
message and immediately after the message I hang up (so there is no
s
" Most components bind to the IP address that you specified explicitly
in
the setup screens"
What is the situation with sipXproxy?
Regards,
Gabor
-Original Message-
From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
Sent: 29 April 2009 14:17
To: Gabor Paller
Cc:
Hi,
I wonder what is the strategy SipX follows when it binds itself to the
local host address. For example, if the host has more than one network
interface, which is the interface that SipX uses to bind itself to?
Regards,
Gabor
___
sipx-users mailing l
Well, for a bug related to forwarding on the desk phones, I got the
following response from Scott Lawrence:
Scott Lawrence, SipX chief architect:
"On Wed, 2009-03-18 at 15:46 +0000, Gabor Paller wrote:
> " Don't do forwarding on the phone -- configure sipX to do it."
Regard
hink it is quite basic scenario.
Regards,
Gabor
-Original Message-
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: 28 April 2009 16:53
To: sipx-users@list.sipfoundry.org; Gabor Paller
Subject: Re: [sipx-users] Immediate forward
>>> On 4/28/2009 at 11:40 AM, in mess
Hi,
One of our customers is interested, how to activate immediate
forwarding. E.g. 200 is forwarded to 202 and it happens immediately, 200
does not ring first, only 202. Is there any way to do it with SipX 4.0?
Regards,
Gabor
___
sipx-users mailing list
Hi,
I just started to move to 3.11/4.0 and I heard that sipxbridge can
handle REFER on behalf of the SIP trunk or gateway. My problem is that I
am not able to activate this feature (the REFER goes to the
trunk/gateway). Is there any common pitfall/trick that I should be aware
of?
Regards,
Gabor
_
", but I haven't seen it written
anywhere that this is allowed."
RFC 3261, section 4:
" For example, if Bob's SIP phone
returned a 486 (Busy Here) response, the biloxi.com proxy server
could proxy the INVITE to Bob's voicemail server. A proxy server can
also send an INVITE to a number o
nes are able to call that external number.
Regards,
Gabor
-Original Message-
From: Dale Worley [mailto:dwor...@nortel.com]
Sent: 18 March 2009 15:35
To: Gabor Paller
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] FW: Forwarding problem with Snom 300
On Wed, 2009-03-18 at
Let's try without the attachment now. The SIP message log is available
on request. I have problems sending mail to the list, I hope all my
attempts don't show up in one batch. :-/
____
From: Gabor Paller
Sent: 17 March 2009 12:26
To: 'sipx-users@list
The analog extension is represented by some gateway and that gateway
must support REFER. Then the transfer works.
From: Andrew Radke [mailto:andrew.ra...@yuruga.com.au]
Sent: 03 March 2009 05:18
To: sipx-users
Subject: [sipx-users] transferring calls on analog
Hi,
I have a problem with MWI NOTIFYs. As I can't unsubscribe these
notifications, whenever I restart some application subscribing to those,
the old subscription lingers around and I receive stray NOTIFYs. As
those NOTIFYs are ignored, they are resent and that puts an unnecessary
load on the syste
e other
systems they have are based on "numbers".
Regards,
Gabor
-Original Message-
From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
Sent: 25 February 2009 17:55
To: Gabor Paller
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Best practices for transfer
On Wed, 200
Well, I am not sure whether this discussion belongs to here or to the
dev list, or maybe somewhere else but I still press on. :-)
"You may want to check out the design document. Please download the
source code from svn and look at sipXbridge/doc/design.tar.gz""
Thanks for the pointer, it is a ve
hones have compatibility
problems with those call flows.
Regards,
Gabor
-Original Message-
From: M. Ranganathan [mailto:mra...@gmail.com]
Sent: 25 February 2009 14:30
To: Gabor Paller
Cc: Tony Graziano; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Best practices for transfer
On W
uary 2009 11:08
To: sipx-users@list.sipfoundry.org; Gabor Paller
Subject: Re: [sipx-users] Best practices for transfer
Once the call is answered, your PSTN gateway shouldn't be involved in
the transfer process. When a call is transferred it is handled by the
proxy, and your UA's (phones
Hi,
I am curious about the best practices for transferring external incoming
calls.
The scenario is very common: Source (an external number) calls target1
(an internal number, SipX user). Target1 transfers the call to target2
(another internal number). This models a common case of a secretary
tra
2 has "active"
subscription state instead of having a terminated subscription state.
What do I do wrong?
Regards,
Gabor Paller
SIP messages:
1. Client subscribes after 401 challenge:
853:
SUBSCRIBE sip:4...@sipxecs.test.onrelay.local SIP/2.0
Call-ID: 6d978c1818cd277832c51927ce66d...@10
Hi,
I have the following interesting situation with SipX 3.10.2. There is a
JAIN-SIP application subscribed to MWI notifications. Before the
scenario presented below, there was a successful subscription and it is
expiring, therefore the application sends a new subscribe. The
subscription progresse
]
Sent: 20 November 2008 17:00
To: Gabor Paller
Cc: Nikolay Kondratyev; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] Re-registration problem
On Thu, 2008-11-20 at 09:40 +, Gabor Paller wrote:
> The Cisco 7940 (SIP image, P0S3-08-5-00) is a more complicated issue.
> The
nded periods of time."
Are the rules of this exception handling public? I am curious, after how
much time the branch with SDP would be CANCELed.
Regards,
Gabor
-Original Message-
From: Scott Lawrence [mailto:[EMAIL PROTECTED]
Sent: 25 November 2008 17:08
To: Gabor Paller
Cc: sipx-
You mean registration expiration time of the endpoint or expiration time
of the INVITE message?
Regards,
Gabor
-Original Message-
From: Dale Worley [mailto:[EMAIL PROTECTED]
Sent: 25 November 2008 16:43
To: Gabor Paller
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] CANCEL
Hi,
I am investigating a strange case (see attached SIP log. I edited the
long and clunky SIP log, if anyone is interested, I can provide the
original). There is an incoming call (User 405, from 10.1.5.74) to user
402 which is branched to two endpoints subscribed to user 405
(SIP001E136C6F0.tes
rcome.
Regards,
Gabor
-Original Message-
From: Dale Worley [mailto:[EMAIL PROTECTED]
Sent: 13 November 2008 16:06
To: Gabor Paller
Cc: Nikolay Kondratyev; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Re-registration problem
On Thu, 2008-11-13 at 12:38 +0000, Gabor Paller wrote
It seems to happen with Cisco 7940 (SIP image, P0S3-08-5-00). It is
still unconfirmed, seems to happen spuriously and only for a few
seconds.
Regards,
Gabor
-Original Message-
From: Nikolay Kondratyev [mailto:[EMAIL PROTECTED]
Sent: 13 November 2008 12:34
To: Gabor Paller; sipx-users
es not take into account the
returned value and is therefore unregistered.
Thanks again for the help,
Gabor
-Original Message-
From: Nikolay Kondratyev [mailto:[EMAIL PROTECTED]
Sent: 13 November 2008 12:07
To: Gabor Paller; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-use
Hi,
I am running 3.10.2 and I observed a strange phenomenon about
re-registrations. My endpoint re-registers periodically, with an
expiration value of 1800 seconds. The re-registration is successful,
REGISTER messages and responses are exchanged correctly. Monitoring from
the "Registrations" page
Hello,
I apologize in advance if this is going to be an extremely trivial
question. :-) Is there any way to set the voice mail activation timeout?
(i.e. after how much time the voice mail kicks in)
Regards,
Gabor
___
sipx-users mailing list
sipx-users@l
Hi,
I develop a JAIN-SIP application and use SipX 3.10.2. The scenario is
very simple: there is an incoming call that I would like to reject for
some reason therefore I respond with 603 DECLINE to the incoming INVITE.
The trouble is that the call runs to voicemail as if I responded 486
Busy here.
ionality is not critical for us.
Regards,
Gabor
-Original Message-
From: Scott Lawrence [mailto:[EMAIL PROTECTED]
Sent: 24 September 2008 19:02
To: Gabor Paller
Subject: Re: [sipx-users] SipX voicemail SIP URIs
On Wed, 2008-09-24 at 14:55 +0100, Gabor Paller wrote:
> Hi,
>
>
>
&
Hi,
As far as I know, there is a direct SIP URI to address the voicemail of
a SipX user but I don't know the format. :-)
The functionality I would like to achieve is to
1. call a SIP URI and the SipX voicemail immediately takes the
call, with the voicemail prompt.
2. call a SIP URI
-
From: Scott Lawrence [mailto:[EMAIL PROTECTED]
Sent: 07 August 2008 13:37
To: Gabor Paller
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] 3.10.2 issue: SIP proxy delays INVITE to
aJAIN-SIP application
On Thu, 2008-08-07 at 13:01 +0100, Gabor Paller wrote:
> Dear list,
>
> I
Dear list,
I have just upgraded to 3.10.2 from 3.10.1 and I ran into an issue with
the new SIP proxy. I have a SIP application based on JAIN-SIP (currently
using version 1.2.76). The application registers on behalf of a SIP
user, accepts INVITEs and initiates calls when incoming INVITEs arrive.
Wi
Thanks, really.
Min-Expires: 300
Regards,
Gabor
-Original Message-
From: Scott Lawrence [mailto:[EMAIL PROTECTED]
Sent: 22 July 2008 16:25
To: Gabor Paller
Subject: Re: [sipx-users] Registration too brief
On Tue, 2008-07-22 at 16:15 +0100, Gabor Paller wrote:
> Hi,
>
> I t
Hi,
I test a SIP application therefore I set the expiration time to a small
value like 10 sec or 100 sec. The SipX proxy protests with 423
Registration Too Brief response. What the lowest limit for valid
expiration time?
Regards,
Gabor
___
sipx-users m
]
Sent: 09 July 2008 17:14
To: Gabor Paller
Cc: Dale Worley; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] GJTAPI/sipprovider registration error
On Wed, 2008-07-09 at 16:06 +0100, Gabor Paller wrote:
> " Now in your case, it looks like your UA is trying to register
against
>
CTED]
Sent: 09 July 2008 15:39
To: Gabor Paller
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] GJTAPI/sipprovider registration error
On Wed, 2008-07-09 at 10:28 +0100, Gabor Paller wrote:
> It seems that the registrar uses the From (or To) header to extract
the
> principal and
I can confirm that it works if I remove the port part from the From/To
header fields. I still feel that something is not right here.
Regards,
Gabor
-Original Message-
From: Gabor Paller
Sent: 09 July 2008 10:28
To: 'sipx-users@list.sipfoundry.org'
Subject: GJTAPI/s
Hi,
I am struggling to get GJTAPI's SIP provider to work with SipX and I ran
into a strange scenario (registrar and SIP message log attached, the SIP
message log is copied after the registrar log excerpt).
The client tries to register but GJTAPI SIP provider constructs a
somewhat complicated user
Hi,
I think the key is the "invalid nonce" message in the log.
There was a similar report just recently and the solution was this:
http://list.sipfoundry.org/archive/sipx-users/msg10806.html
Regards,
Gabor
From: Nikolay Kondratyev [mailto:[EMAIL PROT
16:22
To: Gabor Paller
Cc: Tony Graziano; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Multiple appearance - missed call
On Fri, 2008-06-20 at 14:59 +0100, Gabor Paller wrote:
> I looked up a bit among the standards and found RFC 3326. According to
> this standard, the CANCE
ginal Message-
From: Tony Graziano [mailto:[EMAIL PROTECTED]
Sent: 20 June 2008 12:55
To: sipx-users@list.sipfoundry.org; Gabor Paller
Subject: Re: [sipx-users] Multiple appearance - missed call
Not that I know of.
In 3.10.1, since it is the same line, if the user is logged into the UI
portal, th
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