> From: Michael Picher [mpic...@ezuce.com]
>
> I think the default install has always installed with weight 0 SRV records.
Well, it shouldn't!
Dale
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List Archive: http://list.sipfoundry.org/arch
> From: cyril constantin [cyril.constan...@gmail.com]
>
> Like I have posted three months ago Bria 3.2.1 (the latest) doesn't support
> DNS SRV with Weight of 0 for the three servers [...]
I may be uninformed about the latest changes, but why are you using weight = 0
in DNS SRV?
That has always
The usual cause of this sort of problem is that the proxy does not
recognize that the request-URI is its own address.
In this case, I see that the request URI is .
Strictly speaking, this is *not* the address of the proxy because
the proxy isn't listening on port 0. I believe that you intended to
> From: Marco Colaneri [sipxm...@gmail.com]
>
> I didn't find any new feature request issue open about
> server-side CCBS.
>
> I opened an issue (xx-4983) about server side call
> completion about two years ago that was closed as "Won't
> fix".
>
> Do you think this feature will be included in S
> From: Ari Sonesh [ason...@gmail.com]
>
> I believe that the issue is inherent to mobile data networks
> that are buffering initial (after standby) data traffic
> until it can allocate a channel to transmit the data to a
> receiver. Once channel is allocated the traffic is flowing
> well.
The
> From: Jan Fricke [jan.fri...@iant.de]
>
> I know that there are some end-devices that support CCBS (call
> completion on busy subscriber). E.g. we used it some time ago with
> snom phones. If I’m not wrong the feature is implemented on phone side
> with some kind of event-package (Subscribe/noti
> From: Jeff Gilmore [j...@thegilmores.net]
>
> Would it be safe to manually delete the entries from the file?
It might be safe to do so if no sipX facility was using the related
entries. But the file is regenerated by sipXconfig whenever
sipXconfig thinks it contents should change, so no manual
> From: Jeff Gilmore [j...@thegilmores.net]
>
> However, I notice a residual set of SUBSCRIBEs that are sent by the
> sipxRLS service to each extension that is enabled for voicemail (even
> ones with no phone attached).
>
> I traced one that went to one of my ATA devices, and it came back "501
>
> From: Tony Graziano [tgrazi...@myitdepartment.net]
>
> My question: If the signalling for the call is established with one IP
> address, does a totally different IP address for media break any
> rules?
No, it doesn't. The endpoint has to listen for media at the address
and port specified in th
> From: Tony Graziano [tgrazi...@myitdepartment.net]
>
> if I make a call to a carrier and send the invite and also send sdp,
> is it proper for the carrier to send back aight reinvite with a new
> sdp even though no session has actually been established? and then
> turn around and tell me my offe
> From: pscheep...@epo.org [pscheep...@epo.org]
>
> If the sip domain has more then 1 sip SRV
> records with equal priority then Bria will pick one of them. I do
> remember that newer Bria's always picks the first SRV,
That is to say, Bria does not properly implement SRV records.
You should co
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Roman Gelfand
[rgelfa...@gmail.com]
Much has been said on this forum pertaining shutting off all sip aware
filtering. Is sip aware filtering only an issu
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Staffan Kerker
[ietf-li...@kerker.se]
If I redial, for some reason the initial INVITE has no "Proxy-Authorization":
header and once that request is challen
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of pscheep...@epo.org
[pscheep...@epo.org]
(Maybe more a SIP question then a SipX question)
I am trying to get my Tandberg Cisco C40 codec under control.
Ever
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Carl Farrington
[c...@css-networks.com]
Hi. I now have a trunk set up, and can make outgoing calls.
I am using the sipXbridge (Use built-in SIP Trunk SBC =
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Kyle Haefner
[kyle.haef...@colostate.edu]
I am trying to connect Lync through sipX using a sip trunk and I'm
having trouble getting sipX accept calls from L
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Laurent Schweizer
[laurent.schwei...@peoplefone.com]
if I start the call with a correct domain (the Sipx domain) all is ok.
how can I solve this issue ?
__
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ben Goodfellow
[b...@btg-computers.co.uk]
MoH does not play for inbound calls dialled from a UK mobile.
I would fi
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Irena Dolovčak
[irena.dolov...@gmail.com]
our logs show that this is happening: call comes from the itsp's server to the
bridge on sipx1. sipxbridge
forwar
> From: Irena Dolovčak [irena.dolov...@gmail.com]
>
> to be more precise, exactly every second call goes through. by
> checking the logs, it seems that the remaining sipxecs server (one is
> down) does load balancing, that is, once it processes the call
> himself, once it sends the call to the oth
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Sven Evensen
[sven.even...@onrelay.com]
There is no Trying from sipxBridge as that thread is hanging in the DNS lookup.
The second INVITE from sipxProxy DOE
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Bryan Anderson
[branderso...@msn.com]
If there are three phones on three lan's
LAN A
LAN B
LAN C
phone 1 is on LAN A with the SipXecs server
Phone 2 and 3
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Douglas Hubler
[dhub...@ezuce.com]
back to the topic... I once posted the idea of embedding sip viewer
right into admin ui, where one could drill into calls
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jeff Gilmore
[j...@thegilmores.net]
That is basically my point. There is no reference to sipx-response-correlator
anywhere on the WIKI, and a google searc
You should make sure you have reviewed what can be done with sipviewer,
merge-logs, sipx-snapshot, sipx-trace, and sipx-response-correlator.
Dale
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sipx-users mailing list
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List Archive: http://list.sipfoundry.org/archive
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Niklas
[niklas.rehnb...@gmail.com]
Have a issue that some calls is disconnected before B--part
has answer,
Therefor I need to use sipx-trace.
The tool shoul
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Borginger Rikard
[rikard.borgin...@combitech.se]
I have a system where a sipXecs HA cluster is installed at different geographic
locations. On each site th
INVITE sip:mailto:18005551...@losangeles.voip.ms;sipxecs-lineid=2 SIP/2.0\r
___
I don't know where that "mailto" URI came from, but it is deeply wrong.
I don't know of any way that could be introduced through nor
On Fri, Feb 18, 2011 at 4:18 PM, Burden, Mike
mailto:m...@lynk.com>> wrote:
Is it possible to configure the sipXproxy to add some number N to the Max
Forwards when a call is received, to make sure that there are always enough
forwards left to allow the s
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Brett Jones
[brettjo...@ozemail.com.au]
I've tried to argue with them but they simply say that the service is for
consumer ATA's not a comms system.
__
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Bob Blanchard Jr.
[bl...@dainty.ca]
Is it possible with sipXecs, to have auto-attendant dial options for language
selection? IE. "Press 1 for english, pre
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Henry Dogger
[h.dog...@telecats.nl]
I checked with wireshark and the number is dialed correctly, I can ping my
server from the gateway, and the other way a
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Peter van der Salm
[peter.vanders...@smart-future.nl]
Thanks a lot! Then we are not going to dig into this ERR issue. I have phones
that from time to time
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Irena Dolovčak
[irena.dolov...@gmail.com]
I have a two server HA sipx cluster.
When I take down the primary server, everything still works with the secondar
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Peter van der Salm
[peter.vanders...@smart-future.nl]
We have some hard to research registration problems between SipX 4.2.1 and
Unidata WPU-7700.
With 4.0
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Henry Dogger
[h.dog...@telecats.nl]
I’m looking for some tips concerning dividing server capacity in my ha-setup.
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nikolay Kondratyev
[k...@nstel.ru]
i encountered "duplicated challenge" from sipx.
That call, or at least, the las
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Henry Dogger
[h.dog...@telecats.nl]
I have found an issue and want to know if this is normal behavior
When I configure a call forward at my phone (so not at
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
[tgrazi...@myitdepartment.net]
Probably. I've noticed reliability breaks down with Polycom phones if
the same line is registered/used more the
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of shariksaigal
[sharik.sai...@globaledgesoft.com]
Thanks for the reply but when I Configure From header which is *not* a
configured user of sipX Authenticati
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nikolay Kondratyev
[k...@nstel.ru]
I know that there are many sip gurus in the list...
I encountered the registrar, that uses status 407 in reply to Registe
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Burden, Mike
[...@lynk.com]
Is there a better way to get “incoming calls ring all extensions” functionality
without taking such a hit on Max Forwards? Be
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Burden, Mike
[...@lynk.com]
When I run a siptrace of an inbound call, I see that before an INVITE is sent
to any of our phones, the following sequence is r
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Josh M. Patten
[jpat...@co.brazos.tx.us]
There is one forwarding scenario that this doesn't cover: unconditional
forwarding. There are scenarios where it i
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Roman Gelfand
[rgelfa...@gmail.com]
Is it possible to forward calls using phone rather than web portal?
___
It
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Roman Gelfand
[rgelfa...@gmail.com]
Are you required to have split dns when using remote workers?
___
In princi
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Roman Gelfand
[rgelfa...@gmail.com]
Is it possible to have several phones pointing to the same mailbox and
have all phones show mwi? I have tried using voi
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nathaniel Watkins
[nwatk...@garrettcounty.org]
My network guy apparently moved some servers around - which includes my DNS
server for sipx (it is not being
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jesse Reynolds
[je...@va.com.au]
Has anyone else seen this? Am I just a crazy fool to try and run sipx on
virtualbox on a mac?
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of David Becker
[david.bec...@itison-ikt.de]
Hm, closer inspection reveals that the Siemens phone doesn't send an
algorithm=md5 parameter. Also the contact add
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Burden, Mike
[...@lynk.com]
The ITSP says that the calls arrive at their switch that way.
Looking at the sipxtrace, it looks like the modular architecture
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of shariksaigal
[sharik.sai...@globaledgesoft.com]
Scenario:Every INVITE gives 407 Proxy Authentication Required for
version 4.2.1-018971 of SIPX Serve
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Roman Gelfand
[rgelfa...@gmail.com]
Let's say my phone extension is 201. Does sipx allow for me to be
able to pick up the call to( while it is ringing) to
It's though to diagnose authorization problems, because the computation of the
hash is fairly complex.
Set the logging level to DEBUG and see if /var/log/sipxpbx/sipstatus.log
reports anything interesting.
One thing to check is that when the UA re-sends the SUBSCRIBE after being
challenged, th
From: Staffan Kerker [ietf-li...@kerker.se]
Wouldn't it be nice to make GRUU support in SipX a configurable item? At least
giving the admin the possibility
to turn the feature OFF for remote users needing remote NAT traversal?
_
From: Staffan Kerker [ietf-li...@kerker.se]
The patches applied to sipXregistry so far, are they now in the main/devel
branch so that GRUU now and forward on works at least
when remote users are not behind NAT?
I b
From: Staffan Kerker [ietf-li...@kerker.se]
Note, this is a slightly different call scenario due to some changes in my
setup, but the problem is the same. The call is initiated from an incoming SIP
trunk, forks to two phones behind NAT, one answers (200 O
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Rene Pankratz
[rene.pankratz.l...@iant.de]
Thanks a lot for your answer... though it is somehow nonsatisfying...
Y
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Rene Pankratz
[rene.pankratz.l...@iant.de]
I have a dialplan rule with several gateways associated (a VoIP provider and a
patton ISDN gateway).
Now, if I
From: Staffan Kerker [ietf-li...@kerker.se]
The sipXregistrar now seems to do the right stuff, but now it looks to me as if
the sipXproxy
is messing up the Request-URI and not using the correct one (public IP address)
that is
received from sipXregistry.
Since you're getting a definitive symptom (phones fail to register), set your
logging level to INFO. Wait until the problem shows up and take a snapshot.
Then dig into the recorded logs to find out exactly where the processing of the
REGISTER requests is failing.
Dale
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Robert Hoffmann
Is that because there are two virtual entities processing the message inside
the proxy?
Yes, si
From: sipx-dev-boun...@list.sipfoundry.org
[sipx-dev-boun...@list.sipfoundry.org] On Behalf Of Laurentiu Ceausescu
[lauren...@ezuce.com]
I'm working on http://track.sipfoundry.org/browse/XX-8614 "Add NAT Traversal
support for TLS port in sipXconfig" ...
INVITE sip:1...@test.com;user=phone SIP/2.0
Via: SIP/2.0/TCP 192.168.0.1:1029;branch=z9hG4bK-4oh6gt1acd3g;rport
From: "George" ;tag=1nozggwi04
To:
Call-ID: 3c2670b590cb-lcifsq0o4kpa
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <"sip:~~gr~r9rdk4oxajt_c...@test.c
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Mark Theis
[mth...@socaltelephone.com]
Both the office and my home are private ip's behind nat. The sipXecs boxes are
at a co-lo off site. I would complet
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of mattias [...@mjw.se]
How to solve it
___
Here is good advice about reporting problems:
http://www.chiark.green
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of m...@grounded.net
[m...@grounded.net]
I've noticed that iptables on sipx is always disabled when I install a server.
I was wondering if there are any probl
___
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Mark Theis
[mth...@socaltelephone.com]
What decides what the registration string will be when a device registers with
sipXecs (or any IP PBX for that matter
From: Joegen Baclor [jbac...@ezuce.com]
Is there a way to not authenticate requests and configure trusted ip
addresses in sipXproxy making it a generic relay? From what I am
seeing, all transactions hitting the sipXproxy will traverse through all
the authe
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Massimo Vignone
[massimo.vign...@unimore.it]
I've found that my Polycom phones respond to INVITEs after 1.7 seconds,
some t
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Hiral Patel
[hiral.pa...@onrelay.com]
The problem seems to be that the TRYING response to the INVITE is received late
approx. between 0.01 to 0.08 seconds
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Hiral Patel
[hiral.pa...@onrelay.com]
We have a regular occurrence with our current SIP trunk where now and then the
network responds with a TRYING event a
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Massimo VIGNONE
[massimo.vign...@unimore.it]
In a HA scenario, with two servers running Sipxecs 4.2.1-018932, users
registered on one server can't call user
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ali EL MOUSSAWI
[mouss...@gmail.com]
Is there any way to configure sipXecs as a SIP outbound proxy?
Yes, you can u
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Rene Pankratz
[rene.pankratz.l...@iant.de]
I have a requierement for some users that shall not be able to call outside (no
problem) and shall not be reacha
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Joe Micciche
[jmicc...@redhat.com]
I've got a repeatable case where a Polycom 501 or 601 running 3.2.3.0002
can call out while not registered.
_
From: Rene Pankratz [rene.pankratz.l...@iant.de]
Sorry that I am posting twice but I want to add folliwing information:
First of all I forgot to thank for your answers :)
I created a pcap and I could see that the redundant server does not send any
traffi
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nikolay Kondratyev
[k...@nstel.ru]
My analisys of the problem is as following:
ACK must be sent to the URI from the Contact header of the 200OK message. I
From: Staffan Kerker [ietf-li...@kerker.se]
Seems like a new minor issue happends after this patch is applied. The ACK is
now lost
in the sipxregistrar again.
Yes, the Registrar is not handling Route headers in ACK
From: Staffan Kerker [ietf-li...@kerker.se]
Snapshot linked below:
http://www.kerker.se/files/sipx-snapshot-sipx.kerker.se-patch2.tar.gz
I'm getting a 404 on that URL (at Wed Sep 8 19:25:14 UTC 2010).
Dale
__
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Danny Shay
[ds...@norlemtc.com]
BLF worked when we first installed 4.2.0, but failed shortly thereafter. I
think it failed after I set the phones to reque
From: Melcon Moraes [mel...@gmail.com]
[regarding
https://wiki.openscs.org/display/xecsuserV4r0/Manually+Configuring+Phone+BLF]
Is this wiki still closed? I couldn't access it.
We're still having problems getting
From: Rene Pankratz (JIRA) [no-re...@track.sipfoundry.org]
Snom RLS subscription fails in SipX 4.2.1
-
Key: XX-8652
URL: http://track.sipfoundry.org/browse/XX-8652
Proje
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jean-Hugues Royer
[jhro...@joher.com]
When sipXecs proxies an INVITE to a phone that answers by a 302
(redirection) it automatically sends an INVITE to the
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Talbot, Peter
[peter.tal...@nrtnortheast.com]
1. Why are the calls being directed to SIPX2 in the first place, if DNS is
set up correctly?
_
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?
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Talbot, Peter
[peter.tal...@nrtnortheast.com]
shouldn’t SBC1 be trying to send that traffic off to SIPX1 based on DNS SRV
Records?
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of David Becker
[david.bec...@itison-ikt.de]
Okay, turns out the ifcfg-eth0 settings didn't include the system's own DNS
server. It might be useful to make si
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Todd Hodgen
[thod...@verizon.net]
I’m wanting to run a sipXecs system without internet access, or access to an
external NTP server for Demonstration purpos
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Burden, Mike
[...@lynk.com]
The switch vendor is pointing out that they are not getting a REFER from us.
This, of course, is because sipXbridge translate
From: Melcon Moraes [mel...@gmail.com]
Maybe a misconfigured DNS, slowing your name resolution. Even when receiving
the replies every 10 seconds, do you get them in order? I mean, no packet is
lost?
For instance:
$ ping 172.20.0.20
PING 172.20.0.20 (172.
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Rene Pankratz
[rene.pankratz.l...@iant.de]
For example if I ping one of the machines i don't get an answer for about 10
seconds but then all pings get a re
From: Staffan Kerker [ietf-li...@kerker.se]
Hopefully better this time... Updated on tracker and on the following link:
http://www.kerker.se/files/sipx-snapshot-sipx.kerker.se-2.tar.gz
OK, the problem there is that
I've looked at the snapshot, and it looks like the INVITE is getting to
Freeswitch correctly, the display name is not getting damaged. Other people
here have been looking at this problem; apparently it is a known bug in
Freeswitch. See Freeswitch Jira FSCORE-605; there should be a patch for th
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Staffan Kerker
[ietf-li...@kerker.se]
http://www.kerker.se/files/sipx-snapshot-sipx.kerker.se.tar.gz
I tried to narrow the snapshot down to the very few mi
From: Staffan Kerker [staf...@kerker.se]
> The Request-URI in the ACK returned from sipxregistry is still the long
> name-addr and not a URI.
However, I've noted that the ACK is different now from the previous trace.
Please see tracefile below (adding
to
From: Alexey Kanukhin [box4b...@gmail.com]
I captured a voice mail session by the Wireshark tool. The captured
packets are in the attached file.
I want to see a sipXecs *snapshot*, which captures far more information
Can you capture a snapshot of a call that leaves such a voicemail? The first
step is to verify that the From header is getting to the voicemail system
correctly.
Dale
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Can you apply the attached patch and see if it eliminates the problem? (I'm
not in a good position to test the patch myself right now.)
Dale
19049.diff
Description: 19049.diff
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From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Staffan Kerker
[ietf-li...@kerker.se]
See the Sipviewer trace:
http://www.kerker.se/files/tandberg-polycom-ack-misrouted.xml
The sipXproxy.log gives error
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