On 2010-07-14 20:18, Miguel Jesús Díaz Macedo wrote:
> Hello all
>
> Can any one describe an example for the sipx-trace command line
> utility options
means "any string that would be found in the message of interest"
so if you wanted a trace showing all calls from FooBar brand phones, you
mig
On 2010-07-14 17:16, Gary Luca wrote:
> I already posted this on the forum, but haven't gotten any replies
> yet. I know i'm not supposed to send to the list as well, but i'm not
> sure if it got sent out. Now that i'm aware of the list i'll use this
> from now on.
>
> Good evening all,
>
>
On 2010-07-08 11:18, m...@grounded.net wrote:
> And a big fat 'Ah, so something DID change in sipx that made this pronounced
> all of a sudden!'.
>
No. A clock behaving the way yours did would mess up any version of
sipXecs that was ever built (or ever will be - time must be a one-way
system)
On 2010-07-07 21:57, Wen Jun wrote:
> Well, that protection mechanism to avoid surf registration load to SIPX is
> understood. That seems a pretty wise fence.
>
> My another question is how&when SIPX knows to give phones by random
> registration timers ?
It just always randomizes the returned exp
On 2010-07-07 14:19, m...@grounded.net wrote:
>> Yes.
> Ok, wanted to make sure I didn't miss something.
>
>> I made note that the last time the problem happened was around 11:01.
>> Notice that I started your test script to at 10:23, and it ended at 11:47.
>> The output from that test sh
On 2010-07-07 13:30, m...@grounded.net wrote:
> What is interesting is that it happened around 11:01 or so and looking at the
> sipregistry.log file, the times are totally weird as was identified yesterday.
>
> I have ntpd turned off at the moment.
> Looking at the above file, we see when we rol
On 2010-07-07 12:55, m...@grounded.net wrote:
>>> for i in {1..5000}
>>> do
>>> date>>/tmp/timecheck
>>> sleep 1
>>> done
> This completed but of course, it's a large file and hard to know if any
> timing was missed.
> Anyone know of a command or little to check this output file now?
>
On 2010-07-07 12:27, m...@grounded.net wrote:
>>> 1) Set your logging level to INFO (or DEBUG if you like that better).
> It is set to info or, all default at this time.
>
>>> 2) Wait for the problem to be manifest, that is, a phone is first seen
>>> and then later not seen on the Active Re
On 2010-07-07 10:23, m...@grounded.net wrote:
> Just prior to the problem being repeated, everything looked normal, we saw no
> errors anywhere. This is why I didn't post the entire log and only a few
> seconds prior. When we were able to duplicate the problem, we were watching
> wireshark, ru
On 2010-07-07 0:26, m...@grounded.net wrote:
> It happened again about 5 minutes ago. I grabbed a snippet after all of the
> registered phones had dropped. At the time I took this, there were no phones
> registered.
You provided approximately 33 seconds of log data, and did not tell us
the ti
On 2010-07-06 22:17, m...@grounded.net wrote:
> I've not cleared my logs in a day or two now and decided to double check that
> merge-logs in indeed installed, as I'm sure I ran it.
> This time, it took a rather long time to run so figured perhaps it didn't
> have enough logs when I've run it b
On 2010-07-06 22:35, m...@grounded.net wrote:
> I've never seen this on any other server I've got so this is beyond my
> knowledge.
> I'm central US time, I have an offset of -360 set in the GMT Offset in sipx.
> This is what I have set in ntp.conf.
>
> So, I have a UTC setting somewhere which i
On 2010-07-06 22:36, m...@grounded.net wrote:
> So I have a UTC and GMT conflict somewhere?
No - timezones have nothing to do with it.
You are using NTP incorrectly and it is messing up your clock, which is
then messing up registrations. Actually, all kinds of things will be
fowled up by this
On 2010-07-06 21:40, m...@grounded.net wrote:
> On Tue, 6 Jul 2010 23:47:48 +, Matthew Kitchin (Public) wrote:
>> Can you paste output from crontab -l and the file /etc/ntpd.conf
> Sure, nothing in crontab;
>
> # crontab -l
> no crontab for root
>
> And there is no ntpd.conf in /etc/
>
> I
On 2010-07-06 16:47, m...@grounded.net wrote:
> The tools which were suggested to me are both where I'm stuck at this time.
> The merge-logs command and the regtimes command don't exit on the ISO so I've
> gotten sidetracked in resolving those issues. This has lead to other problems
> while th
On 2010-07-06 17:15, m...@grounded.net wrote:
> One thing I forgot to note is that someone was on the phone when all of the
> registered devices went away but she never lost her call.
> After she hung up, she used her cell phone to call her sipx phone which is
> internal and it went directly to
>> There is a script installed with 4.2 called 'regtimes' that helps to
>> analyze whether or not you are having the problem described at the wiki
> I'd like to try this. I've got many perl modules installed on test systems
> but this is a production box.
> Just wondering before I go ahead an
> First of all, if I didn't cover all of the bases, PLEASE, do not hijack my
> thread about it, either start another or email me personally.
>
> For a couple of weeks or more now, we have started noticing a behavior which
> is that of phones dropping registration.
> Dropping, meaning that they d
> The hardware would be an IBM BladeCenter LS20 blade, dual duo-2.2Ghz, 4GB of
> memory. I've checked for errors on the blade, haven't seen anything obvious,
> everything else works, no errors in the log which usually shows CPU/memory
> errors.
>
> The dropped call was on a LinkSys PAP2 which h
On 2010-07-02 23:32, m...@grounded.net wrote:
> Just now, *while* on a call, the phone dropped registration so it's getting
> worse.
> Another phone with the same number, registered didn't drop.
> It's random and I'm looking at the logs now.
That is most likely just a broken phone, but there is
On 2010-06-23 9:03, Tony Graziano wrote:
> The master controls all the failover and synchronization when service
> is restored for registration. Taking the master proxy offline creates
> an issue and am glad you are not going to do that.
>
No, the registry synchronization is fully meshed, and d
> So, enforce all the RFC's? SIP standards to all exclusion?
> Truly bad idea, however: it IS an open source, GPL system, and anyone
> here has access to the source code and can 'fix' it themselves.
>
>> will go back to "mostly working" for such ITSPs. It is just true that
>> all call flows ha
>
>>
>> Hello,
>>
>> I had previously put in some hacks to compensate for protocol errors
>>
> >from ITSPs such as responses to SDP solicitations that would return
>
>> an OK with NO SDP body. There are even those very badly mannered ITSPs
>> that randomly return OK to INVITE with no con
On 2010-06-13 17:34, Jiann-Ming Su wrote:
> As Dale Worley put it in his reply, "4.2 should have TLS support for
> communicating with phones." We want SIP Trunking over TLS and phone
> registration over TLS so that SIP credentials are not sent in the
> clear.
>
SIP credentials are always sent
On 2010-05-26 9:43, Michael Scheidell wrote:
> If we have a mixture of cisco phones, we can decide on a phone by
> phone, group by group, model by model basis, what firmware we want on it.
> It looks like the polycom phones have only one firmware option, site wide.
>
> If we have EOL phones (501'
On 2010-05-23 4:35, Sven Evensen wrote:
>
> We are running sipX 4.0.4 in Amazon EC2 Singapore with an ITSP located
> in Hong Kong.
>
> When I place an incoming call to a soft phone, when the soft phone
> answers, 200 OK
>
> propagates back to the ITSP. But it is rejected (thus no OK), the rest
>
On 2010-05-19 11:26, Tony Graziano wrote:
> I think you miss his point. Dialing from the "missed calls" on the
> handset shows the +1 (and 10 digits) on the display. The outbound call
> fails because +1 is ALSO being added at the gateway. Not all carriers
> do this, not all phones display the +
On 2010-05-18 14:00, Mossman, Paul (Paul) wrote:
> Hi all,
>
> How often are multiple Voicemail dial plan rules actually used?
>
> I can think of one reason... You have some 3 digit extensions, and some 4
> digit extensions, but you want both 8XXX and 8 to go directory to the XXX
> (or )
On Mon, 2010-05-10 at 07:58 +0200, Rene Pankratz wrote:
> Scott, thanks for your answer.
>
> I set up a site-to-site dialplan rule for the exact match on 298 going
> through my gateway but sipX still wants an authentication if 298 is
> dialed.
> When removing the unmanaged GW from another dialing
On Sat, 2010-05-08 at 19:24 +0200, Robert Hoffmann wrote:
> I would like to do a very basic install of sipXecs with the latest
> ISO. I have a public domain with the needed SIP records.
> Now I see that the sipX proxy is not starting because it requires a
> local DNS server with the same SIP recor
On Fri, 2010-05-07 at 12:35 +0200, Rene Pankratz wrote:
> Hello,
> I have a PSTN GW and another unmanaged GW connected to the SipX
> (4.0.4).
>
> Incoming calls from PSTN that match an extension (e.g exact match on
> ext "298") shall be routed to the other unmanaged GW.
> But unfortunately the Sip
On Thu, 2010-05-06 at 19:24 +0200, Eelco Brölman wrote:
> Hi all,
>
> I just installed a Mediant 1000 gateway, uploaded the generated config
> for it (pretty standard), and that works like a charm. But: when a
> call comes in from the PSTN to the gateway with a CLI which is the
> same as a user de
On Thu, 2010-05-06 at 13:46 -0500, Austin Curry wrote:
> Has anyone tested a SIP client Iphone app with sipx?
>
I've used iSip successfully quite a bit.
> I know that 5060 is blocked on parts of AT&T’s network, but would like
> to test out an app that is known to work.
I've actually not encount
On Wed, 2010-05-05 at 21:33 -0400, Mossman, Paul (Paul) wrote:
> Hi all,
>
> XX-7822 [1] mentions "phantom user" usage, and that these requirements
> should be accommodated in the re-organized Dial Plan.
A suggestion for one way to slice this into more bite-sized pieces:
First, deal with inbound
On Wed, 2010-05-05 at 08:54 -0400, Tony Graziano wrote:
> I was testing a case this morning where I created a new 4.2 system,
> add some users so test Bria softphone functions. I deleted my test
> users and had one PC I could not get back to, and see it was
> continuously "refreshing" its registrat
On Tue, 2010-05-04 at 02:42 +0100, Hiral Patel wrote:
> If Dial plan 1 is above dial plan 2 in sipXecs dial plan list, when
> User two places a call to an external PSTN number the call fails, this
> is because there seems to be an issue in sipXecs where USER 2 is
> denied and the call request from
On Mon, 2010-05-03 at 13:23 -0500, Josh Patten wrote:
> Perhaps I could try regenerating all certificates? I'm not sure how to
> do this in a cluster environment. Could someone point me in the right
> direction? I know in a single node environment you can use this:
> http://sipx-wiki.calivia.com
On Sun, 2010-05-02 at 19:39 -0500, m...@grounded.net wrote:
> Is there any way in 4.2 of preventing the web structure overwrite upon
> reboot/restart?
It's not clear what you're referring to here... can you be more specific
about what problem you're seeing?
__
On Mon, 2010-05-03 at 11:11 -0400, Mossman, Paul (Paul) wrote:
> Josh wrote:
> > Is there any way to disable the gravatar feature? Some of the
> > default gravatars are not "fitting," to put it lightly, in a
> > business environment and might be offensive to some users.
>
> There's no sipXecs co
On Sun, 2010-05-02 at 07:03 -0400, Justin Menga wrote:
> Just thought I'd give an update as I ran into this issue yet
> again and it stumped me for a little bit.
>
> The fix is simple - in my case my voice gateway was sending
> INVITE with the To header in the format of
> @ - by default,
> SIPXec
On Sat, 2010-05-01 at 12:51 -0400, Picher, Michael wrote:
> Alright, been fighting with this for a few hours…
>
>
>
> With the new IM integration I can get signed in with Pidgin just fine
> and see status back and forth and things are working across the
> interwebs as well (split DNS is working
On Fri, 2010-04-30 at 16:07 -0400, Andreas (Around the Clock Information
Systems) wrote:
> sipXecs version 4.0.1-015823 is an unstable developer release?!?!? It was
> installed from ISO like six months ago and updated along the way. I didn't
> realize that the developer releases where even availa
On Fri, 2010-04-30 at 11:54 -0500, Gabe Casey wrote:
> Thanks although I have no /var/log/sipxpbx/sipxprovision.log on the
> failed server this is the service that is affecting my system.
The error you're seeing indicates that the service cannot start because
its configuration is not the right ver
On Fri, 2010-04-30 at 10:47 -0500, Gabe Casey wrote:
> I have installed a test system off the 4.0.4 and upgraded to 4.2.
> The only difference in the upgrade were the following files
> polycom_phone1.cfg.rpmnew polycom_sip.cfg.rpmnew
>
> I backed up my production config and placed it on the test
On Thu, 2010-04-29 at 19:37 -0400, Tony Graziano wrote:
> Makes me ask myself if a HA upgrade should have a special bootup
> script/procedure to prevent the profile push but allow for a certificate
> handshake/sunc to the other server before assuming any interactive roles
> with users/gateway.
It'
On Thu, 2010-04-29 at 18:02 -0500, Josh Patten wrote:
> Now that I reread your post Scott, it make much more sense, and this is
> what we ultimately decided to do. One question that may or may not be
> worth asking, is there an easy way to "copy" registrations over so that
> the mock upgrade sys
On Fri, 2010-04-30 at 14:33 +0300, an...@iguanait.com wrote:
>
>
> Do you know why sipxecs has chosen 300 seconds as minimum by default?
Registration can create a lot of load on the system - if you had a few
hundred phones re-registering every 30 seconds that's a lot of messages.
Unless the IP
On Thu, 2010-04-29 at 14:33 -0400, Tony Graziano wrote:
> Can someone explain to me how to move this from Perosnal Space to the
> wiki for general availability?
>
>
> http://wiki.sipfoundry.org/x/D4Fd
Under the Tools menu in the top right of the page, there should be a
'Move' entry that will do
On Thu, 2010-04-29 at 14:51 -0500, Josh Patten wrote:
> Already did those things you recommended with the "First Test System".
> That doesn't "mirror" my current production environment. It only makes
> a separate environment that I can play around in.
>
> What I need to be able to do is have an ex
On Thu, 2010-04-29 at 13:06 -0400, Tony Graziano wrote:
> That's a very good question. MY logic tells me NO (wont work) if the
> DNS ZONE is a different name than the internal one. I may be wrong,
> but look at my question to user/developers on EXACTLY what my logic is
> and asking for some confirm
On Wed, 2010-04-28 at 21:13 -0400, Picher, Michael wrote:
> Well, there might be a way to do that...
>
> It would be a bit strange but if you were to make a custom dial plan
> entry that you would have your users select with a different prefix.
> Then setup a phantom gateway that has the callerid
On Mon, 2010-04-26 at 15:08 -0500, Josh Patten wrote:
> Upon performing an upgrade in my test environment from 4.0.4 to 4.2.0 I
> noticed something that might make life difficult for administrators of
> large, dispersed systems that have many location based dialing rules:
> The "groups" that wer
On Sun, 2010-04-25 at 09:08 -0400, Jermaine Pinder wrote:
> I’m currently testing the new IP 335 HD From Polycom with Firmware
> 3.2.1 and I’m having an issue with very low register time causing the
> phone to expire in 280 seconds and re-register in 60 seconds (default)
> for remote workers.
Are
On Sun, 2010-04-25 at 07:54 -0500, Robert B wrote:
> Scott,
>
> How does this work exactly? I ask because when I was running sipx under
> OpenVZ using venet interfaces, sipx was completely non-workable because
> there was venet0 and venet0:0. The only way I could fix it was to switch
> to bridg
On Sun, 2010-04-25 at 03:22 +0100, Hiral Patel wrote:
> We would like to know if using port bonding on CentOS to provide
> redundant network interfaces would work with sipXecs or if any one has
> tried this?
Yes, it works fine.
___
sipx-users mailing l
On Sat, 2010-04-24 at 20:29 +0800, Rhon wrote:
> Hi Scott,
>
> Thanks for your reply.
>
> On Sat, Apr 24, 2010 at 8:05 PM, Scott Lawrence
> wrote:
>
>
> Do you have a second polycom phone? Can you call between
> those?
>
On Sat, 2010-04-24 at 08:22 -0400, Tony Graziano wrote:
>
>
> On Sat, Apr 24, 2010 at 8:17 AM, Scott Lawrence
> wrote:
> On Fri, 2010-04-23 at 20:10 +, mkitchin.pub...@gmail.com
> wrote:
>
> > . One of the major components of
On Fri, 2010-04-23 at 20:10 +, mkitchin.pub...@gmail.com wrote:
> . One of the major components of Sipx appears to be working on
> becoming more compatible with virtualization.
... and that was one of the motivations for moving more of our media
services to freeswitch in 4.2.
Testing and pr
On Thu, 2010-04-22 at 11:19 +0800, Rhon wrote:
> I found this in the Diagnostics > Registration
>
> sip:1...@domain.com
>
> 1928
>
>
> sipxgw.domain.com
> sip:1...@domain.com
>
> 311
>
>
> sipxgw.domain.com
> sip:1...@domain.com
>
> 2670
> 0004f21e7cdf
> sipxgw.domaincom
>
> sip: 114 an
On Fri, 2010-04-23 at 14:48 -0400, Saint, David (David) wrote:
> >
> > Hi Dave,
> >
> > I have managed to get a wireshark trace of the divert to
> > voicemail. The key bit of the decode are:
> >
> > From: "6667912"
> > mailto:6667...@10.203.105.50>;tag=d19c5205-82bd-44fc-88c4
> > -bf5d3c52feb5
On Fri, 2010-04-23 at 12:59 -0400, Tony Graziano wrote:
> The wiki had said to use the hostname, which did not work. I altered the
> wiki accordingly.
It said 'the DNS name of the other PBX', by which I meant the SIP domain
name (it did not say to use the hostname)... I see how it could have
been
On Fri, 2010-04-23 at 08:52 -0700, Johan Reinalda wrote:
> All,
> Apologies for a lenghty post.
>
> Here is the Scenario: test-lab fresh install of two 4.2.0 servers in
> HA mode, some phones, and a gateway. Intra SIP, and inbound dialing
> from campus PBX works fine (via AudioCodes M1K).
> Tryin
On Fri, 2010-04-23 at 10:52 -0400, Tony Graziano wrote:
> Sounds like it is not just me:
>
>
> http://track.sipfoundry.org/browse/XX-8221
... and you still have not attached the snapshots for both systems.
___
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On Fri, 2010-04-23 at 18:26 +0800, Rhon wrote:
> Hi Tony,
>
> That's exactly what I did.
>
> As said we have site-to-site running already... it's just that people
> have to call the AA first become connecting to the desired extension.
>
> What I'm looking at is to go directly to the desired ext
On Thu, 2010-04-22 at 12:25 -0700, Nathan Nieblas wrote:
> Cisco follows IETF standards for SIP
With all due respect to Cisco, that statement doesn't mean very much.
There are lots of documents that make up 'standards for SIP', and many
ways in which implementations can be incompatible while sti
Ken Fulmer [kenful...@icstechnologysolutions.com]:
> We have a customer that’s using the 4.0.4 version. When we enable call
> forwarding for the user, nothing happens. I perform a capture and see
> the conversation dies between processes on the system.
Worley, Dale R (Dale) wrote:
> You'll have
On Thu, 2010-04-22 at 10:35 -0400, Gary wrote:
> Content-Type: text/plain;
> charset="utf-8"
> Content-Transfer-Encoding: 8bit
> Organization: SipXecs Forum
> X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <45291>
> Message-ID:
>
>
>
> results of yum and current repos file below
>
> I modified
On Thu, 2010-04-22 at 07:22 -0500, Robert B wrote:
> Tony,
>
> Okay -- so you're right, but here's the issue and apparently a bug...
>
> When I restored my configuration, it did not bring over the SBC Route.
> Nor, once a SIP trunk is created, can I change the SBC Route. That
> option does not
On Thu, 2010-04-22 at 15:13 +0400, Nikolay Kondratyev wrote:
> Hi all,
> A bit earlier in the list i found a command for manual log rotating.
> I tried
> logrotate -f /etc/logrotate.d/sipxchange
> And all logs were rotated except sipxcallresolver.log.
> And i noticed that it was never rotated...
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
On Wed, 2010-04-21 at 11:08 -0500, Ken Fulmer wrote:
> When we attempt to forward an internal extension to an external number
> and someone calls that extension from the PSTN, our provider is
> returning a 604 Does Not Exist Anywhere message.
> The provider is PaeTec and they need a SIP Diversio
On Wed, 2010-04-21 at 11:13 -0400, Joseph Modi wrote:
> I am unable to upgrade to 4.2, it installs but then when I refresh, it
> defaults back to 4.1, any ideas.
Please be more specific.
What did you do to upgrade?
___
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On Wed, 2010-04-21 at 07:10 -0500, Robert B wrote:
> I am getting pretty sick of Sun's JVM using up so many resources...
>
> What about alternatives such as IBM's j9 or Apache Harmony? Does
> anyone have any experience with these and SipX?
That's really a topic for the sipx-dev list, not this one
> 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, 2010-04-21 at 10:02 +0200, r.vanv...@raffel.nl wrote:
> You are absolutely right, that was my initial thought too. I checked
> with the ITSP today and they do not support re-INVITE's. On the other
> hand, they DO support REFER's.
Who is this and what SIP system are they using (answer off
On Wed, 2010-04-21 at 07:31 -0400, Tony Graziano wrote:
> To go from 3.10 to 4.2 would require two steps? 4.0 first right?
In theory, no, but that's the way I'd do it.
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List Archive: http://list
On Wed, 2010-04-21 at 05:29 +0100, Hiral Patel wrote:
> What improvements have been made to the voicemail system in sipXecs
> 4.2?
The list of issues has been posted several times in the last couple of
weeks, and is easily found in the tracker.
The most important change is that it is now based
On Wed, 2010-04-21 at 11:30 +1000, Graeme Allen wrote:
>
> Some further information:
>
> I got SipX to start, by changing /etc/init.d/sipxpbx
All that you did was disable the test that was designed to keep you from
hitting the problems you're hitting now.
As Josh said in another post, 4.2 has a
On Tue, 2010-04-20 at 22:18 +0200, r.vanv...@raffel.nl wrote:
>
> 1.) SipXbridge gets the REFER from sipxecs' proxy, and correctly
> translates it to re-INVITE for the ITSP.
> 2.) The ITSP replies with 100, trying, followed by 200 OK.
> 3.) SipXbridge replies with BYE, Reason: Protocol error 200 O
On Tue, 2010-04-20 at 22:18 +0200, r.vanv...@raffel.nl wrote:
> Hi all,
>
> Sipxecs seems to perform a little weird on transferring calls from the
> IVR when they're coming from an ITSP trunk. Incoming calls work fine,
> and are (by DID) answered by the auto attendent/IVR. When the user
> selects
On Tue, 2010-04-20 at 13:19 -0400, Roman Gelfand wrote:
> It appears that configuration files were not upgraded. Is there a way
> to reset all configuration and start from clean slate?
To be clear ... upgrading into or out of development versions ( X.Y.Z
where Y is odd ) is Unsupported.
Yes, th
On Tue, 2010-04-20 at 11:57 -0400, Roman Gelfand wrote:
> My topology is...
>
> The sipx server has wan ip address. It is behind transparent
> firewall. Can you point me appropriate configuration sample?
Consider the following excellent advice when requesting help...
http://www.chiark.greene
On Tue, 2010-04-20 at 09:42 -0400, Michael Scheidell wrote:
>
>
> On 4/20/10 9:37 AM, Tony Graziano wrote:
> > Wrong. You want to send to them on port 5060, you want them to send calls to
> > you on 5080.
> >
> > When you say "inbound and outbound" it makes me think you think the two
> > paths
On Mon, 2010-04-19 at 15:21 -0400, Abdul Mayat wrote:
> I noticed the new branch setting in 4.2, and wanted to
> clarify its usage with the forum...Currently we are able to
> apply the branch setting to users, phones, gateways. If
> this was extended to dialplans too, would it be a way of
> suppo
On Mon, 2010-04-19 at 10:42 -0400, Michael Scheidell wrote:
> On 4/19/10 10:21 AM, M. Ranganathan wrote:
> >
> >
> > If you are seeing a REFER sent out to the ITSP then it likely is a
> > misconfiguration on your sipxecs.
>
> sipxbridge shows
> > Are you sure that the signaling is
> > being SE
On Mon, 2010-04-19 at 10:18 -0400, Michael Scheidell wrote:
> >
> > > Trying to regress test 4.12. ok, this didn't work on 4.04 either, I
> > > just decided to wait.
> > >
> > Michael... please try to be more careful in your descriptions. There is
> > no such thing as sipXecs 4.12 or 4.04 (
On Mon, 2010-04-19 at 09:46 -0400, Michael Scheidell wrote:
> Trying to regress test 4.12. ok, this didn't work on 4.04 either, I
> just decided to wait.
Michael... please try to be more careful in your descriptions. There is
no such thing as sipXecs 4.12 or 4.04 (get the dots right).
> Call f
On Mon, 2010-04-19 at 08:45 -0400, Tony Graziano wrote:
> Well, there are two things that should make their way to the codebase IMO.
>
> One would be registration attempts, the other invites. I think people
> probably see more invites as a way of probing than they do registration
> attempts.
>
>
On Mon, 2010-04-19 at 10:35 +0800, Rhon wrote:
> Hi Scott,
>
> Attached is the siptrace for your reference. Tried to interpret it,
> but just don't know what it means. :(
You did not reset your logging to INFO level - none of the messages from
the phones are in the trace (you must restart the co
On Sun, 2010-04-18 at 18:12 +0800, Rhon wrote:
> Hi Everyone,
>
> I'm a little confused about the definition on Managed Phones and
> Gateways in SipXecs. Does the word "Managed" mean "Plug and Play"?
It means that there is at least some support for generating
configuration for them.
As others ha
On Sat, 2010-04-17 at 21:20 -0400, Tony Graziano wrote:
> On Sat, Apr 17, 2010 at 9:29 AM, Scott Lawrence wrote:
> > On Sat, 2010-04-17 at 08:58 -0400, Tony Graziano wrote:
> >>
> >> Setting it up to dial directly "proxy-to-proxy" on port 5060 does not
> &
On Sat, 2010-04-17 at 09:56 -0400, Jeff Gilmore wrote:
>
> Any thoughts on what sipx expects to see with a SUBSCRIBE that might
> lead to this behavior? The same ATA device registers this same user
> ("101" in this example) with no problems. We are using DNS SRV
> records.
>
> Even if you don't
On Sat, 2010-04-17 at 08:58 -0400, Tony Graziano wrote:
>
> Setting it up to dial directly "proxy-to-proxy" on port 5060 does not
> work (no audio). Dialing via xlite sip uri from 4.0.4 works fine, no
> audio issues, but dialing from 4.2 from xlite via sip uri dies NOT
> work (no audio).
>
> Sett
On Fri, 2010-04-16 at 15:30 -0700, Charles wrote:
> enabled=0
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On Fri, 2010-04-16 at 16:10 -0400, Michael Scheidell wrote:
> We were waiting for 4.2 to 'go live', which it is, and along with
> regression testing, we will be testing inbound DID.
> previously, we just took the odd DID and put it in as the alias for
> the user, now that we have a 100 block of DID
On Fri, 2010-04-16 at 15:35 -0500, Ken Fulmer wrote:
> I’d asked a question regarding how many servers could exist in a
> cluster. The reply was no more than 5 had been tested.
>
>
>
> Has this number been scaled higher in 4.2?
There have been no significant changes in that area.
On Fri, 2010-04-16 at 15:29 -0500, Ken Fulmer wrote:
> Sorry if already asked:
>
>
>
> Is there a specific area in the wiki for 4.2 information?
There will be early next week I need a tool installed on the wiki to
clone the 4.0 documentation so that we can update it.
__
On Fri, 2010-04-16 at 15:18 -0400, Francis Tinio wrote:
> If we upgrade to 4.2, do we need to use bootrom 3.2.3 for polycom?
> That is the latest version for IP500.
We've done most of our testing with the 3.2.2 firmware (which is not the
same as the bootrom), I believe.
On Fri, 2010-04-16 at 14:49 -0400, Francis Tinio wrote:
>
> i'll try to update to 3.2.3 and update sipx to 4.2 again later.
I should have put this in the release notes and forgot...
There is an outstanding problem with Polycom version 3.2.3
http://track.sipfoundry.org/browse/XTRN-1046
they a
On Fri, 2010-04-16 at 10:44 -0500, Josh Patten wrote:
> Any update on the 64 bit build?
I don't expect we'll have that posted any earlier than the middle of
next week
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