Re: [sipx-users] Caller ID problem wiith Sipgate

[Douglas Hubler]

On Sat, Mar 5, 2011 at 5:54 AM, Irena Dolovčak  wrote:
> Hi Douglas,
> here is the error that i get when the trunk fails on 4.2.1
> in the Prefered identity field i set this: 385492...@sipconnect.sipgate.de
> and get this error:

Let's focus on 4.4 if we can, you said both "crash" however I was able
to set the Preferred Identity fine with-out "crashes" in 4.4.  Send
your copy of /etc/sipxpbx/sipxbridge.xml to me.
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Re: [sipx-users] sipXecs open source licensing

[Douglas Hubler]

Just to keep people posted, there is a SF board meeting this Thursday
where this will try to be decided and as it stands now, I will have a
strong vote against this as I outlined.  I could change my vote should
new information emerge.   If you have a response on either side that
you haven't already given, try to have it in by then.
 --Thanks
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Re: [sipx-users] sipXecs open source licensing

[Douglas Hubler]

It doesn't seem as though you addressed the folks writing plugins in a
non-hosted environment and that sell packaged solutions.  This change
would affect you as a developer/company in the following way:

  As a developer, you would have to make the source available to your
plugin the instant there was some change in the API that effectively
made derivative project/API AGPL.

The problem I see with this is that it would be ripe with corporate
abuse to gain an unfair competitive advantage.   I would see
infighting on both sides with keeping APIs under a certain license.
Complicate the pot even further, if a single company was the sole
author or the API, they would have joint copyright with SIPfoundry to
the source and they may think they would not have to comply with the
license. Even if they believed this (which i would find very shady but
I can't say is illegal), they could suspiciously discourage any LGPL
contributions to said API so that they wouldn't have to comply with
the AGPL like everyone else.  Even if this is scenario is full of
holes, we can expect these types of conversations to distract
everyone.

My overall engineering perspective on the areas Martin did address...

1.) If someone can change the source and merge in their changes
release after release good for them.  Eventually they will either not
keep up or they will start to show up on the list assuming we can
allow them enough options to keep their business running.  Only
someone who has done thousands of merges in their lifetime can truly
understand the significance of this.

2.) Any developer worth anything want to write what hasn't been
written before.   Any developer that doesn't understand that (and I
know a few of these) there's a very good chance we wouldn't want their
patches because they're of such poor quality.

3.) Forcing someone to contribute is somewhat pointless.  No one
forces them to explain their code, write unit tests, format the
source, document.  It's hard enough to stay on top of the patches we
do get.  I don't foresee FSF winning a landmark case for us, and
getting gigabytes of unexplained source and feeling like a winner.

4.) Any further restriction on the libraries we could use would only
slow engineering down.  Slicing up the code base into LGPL and AGPL to
circumvent licensing can only lead to a giant headache that i'll have
to spend time explaining and maintaining what's what.

5.) Best defense to encourage participation is move forward as fast as we can.

6.) From my experience, contributions of a significant size were
historically and not entirely encouraged.  There was enough
engineering at hand so jeopardizing the stability of a release for a
feature that wasn't on their road-map rarely made sense and especially
not to help someone competing with SCS.  If whatever companies(that
I'm told exist) that are hosting sipXecs and not contributing back
were now contacted and told their contributions are encourage, I'm
sure we'd convert a few of them.

7.) At a time when we're still trying to rebuild from a dramatic
change in control, this would not be a good time for a change even if
it did make sense.

In summary:
 AGPL to protect against hosting reciprocity will not be effective
 Change to AGPL for businesses today legally building off of sipXecs
today will be unfair
 Change to AGPL for future businesses will not be attractive

I respect others will have a different opinion, but this is mine.

vote: -1
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Re: [sipx-users] Caller ID problem wiith Sipgate

[Douglas Hubler]

4.4.0 trace only shows a registration, not an INVITE attempt thru the trunk


On Fri, Mar 4, 2011 at 5:46 AM, Irena Dolovčak  wrote:
> here are the logs from both servers.
> By crash i mean that the configuration test fails. (system -> services ->
> SIP trunking)

ok, this is a different story, you definitely should have led with
this fact, not that PPI wasn't working. Capture a snapshot and make is
available somewhere to download or if it's small enough, email it.
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Re: [sipx-users] Openfire IM file transfers very slow

[Douglas Hubler]

On Fri, Mar 4, 2011 at 10:47 AM, Michael Picher  wrote:
> Jeff,
>
> When we find bugs or would like a feature enhancement the best place to file
> something is in the Jira (http://track.sipfoundry.org).
>
> This sounds like a logical enhancement / change.

Also, I'm not aware of any thing implemented to restrict this, so if
you can debug and found out that would help greatly.  Here's how to
get to openfire admin console

http://wiki.sipfoundry.org/display/sipXecs/The+Openfire+admin+console

keep in mind changes you make to openfire directory would get undone
by sipxconfig, so this is more useful as an investigation technique.
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Re: [sipx-users] Caller ID problem wiith Sipgate

[Douglas Hubler]

On Thu, Mar 3, 2011 at 8:38 AM, Irena Dolovčak  wrote:
> the funny thing: when I try to put the preferred identity, whatever I set in
> that field, the SIP trunking service crashes. (same thing on 4.4.0)

what do you mean by "crash". You have the message?

can you capture siptrace xml or wireshark of message?
http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+Sipviewer
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Re: [sipx-users] Caller ID problem wiith Sipgate

[Douglas Hubler]

On Thu, Mar 3, 2011 at 8:38 AM, Irena Dolovčak  wrote:
> Hi,
> I'm using sipx 4.2.1 with SipGate sip trunk and i can't make the trunk to
> show the caller ID number. In some scenarios it does work thou.
> I managed to make the number to show on mobile phones, but when I call some
> other local provider the called person gets "unknown", and I cannot make it
> work.
> Than I tried the same thing on version 4.4.0. and the result was much worse
> than on 4.2.1. I couldn't even get the phone number to be visible on the
> mobile phones. I think that the main problem begins with the invite message
> when the server (4.4.0) sends the number of the user in the From field. In
> the old version, there was the username.
> I tried to make it work with P-A-I and P-P-I, but nothing helped.
> the guys fron Sipgate told me i should put the number in the P-P-I field and
> it should work..
> To get it more clear here's what I did:
> version 4.2.1.
> - set up sip trunk (SipGate) - username, password, register on
> initialization - enabled
>                                          - disabled "Use Asserted Identity"
>                                          - enabled "Use Preferred Identity"
>                           - Caller ID - enabled "transform extension"
>                                           - caller ID prefix - added the
> needed prefix
>      RESULT: the phone number is visible on the mobile phones but not on the
> IP phones on another VoIP provider (anonymous)
> the funny thing: when I try to put the preferred identity, whatever I set in
> that field, the SIP trunking service crashes. (same thing on 4.4.0)
> version 4.4.0
> -  set up sip trunk (SipGate) - username, password, register on
> initialization - enabled
> - INVITE From ITSP Account - enabled (i couldn't make the call when it was
> disabled)
> - all other settings were the same as above in version 4.2.1 and it didn't
> worked.
> event the setting "transform extension" didn't worked. the number didn't get
> the prefix.
> RESULT: the phone number isn't visible
>
> any idea how to make the callerID to be shown?
> I want to have a diferent number for all users so I cannot put the caller ID
> on the trunk.

did you set the caller id for each user by editing the user first?
It's easy to forget this.

I don't recall anything changing in this area from 4.2.1 to 4.4.0 so
maybe missing a config somewhere, you recreated 4.2.1 configs from
scratch?

PPI should be the callers DID or your ITSP account name?
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Re: [sipx-users] AA needs more hops in 4.4.0

[Douglas Hubler]

On Wed, Mar 2, 2011 at 8:59 PM, Michael Scheidell
 wrote:
> i seem to remember that 4.2.1? needed more 'hops' than 4.2.0

Can you capture the incoming SIP messages?

Specifically, is Max-Forwards specified in the header?  How long is
the Via list?
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Re: [sipx-users] backup 4.2.0, restore to 4.4.0 too many hops

[Douglas Hubler]

On Wed, Mar 2, 2011 at 8:02 PM, Michael Scheidell
 wrote:
> using iso from yesterday, goal is to try to get a working 4.4.0 system as a
> 'cold standby', and if it works, upgrade my 4.2.0 to it.
> (hopefully to get voip.ms to do ip based registration on port 5060!)
>
> so, backup a running 4.2.0 system,  boot the 4.4.0 centos from yesterday,
> restore it (it has NO network connectivity except laptop)
>
> when the day is quiet:  pull network cable from working system, flush arp
> cache on switches, plug in 'cold standby'
>
> reboot it, figuring it had lots of issues with not having running services.
>
> now: inbound call from level 3 (should go to aa)
>
>  73.157884    47.255.134.48 -> 192.72.0.2    SIP/SDP Request: INVITE
> sip:{mydid}@{mypublicip}:5080, with session description
>  73.367254    192.72.0.2 -> 47.255.134.48   SIP Status: 100 Trying
>  73.435351    192.72.0.2 -> 47.255.134.48   SIP Status: 483 Too many hops
>
> yes, went into gateway, rejenned, what did I miss?

did you enter your public ip address as a domain alias in the domain config?
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Re: [sipx-users] backup/restore: test 4.4

[Douglas Hubler]

On Wed, Mar 2, 2011 at 9:36 AM, Geoff Van Brunt  wrote:
> The problem is the domain is included in calculating the hash. So if you 
> restore to a new domain it uses the new domain for calculations to compare 
> against so they don't match. What is needed is to drop the domain from hash 
> calculations for new pins. However this breaks backwards compatibility. I 
> think this can be worked around by storing the "old domain" for the pin in 
> the table so when it comes time do a comparison it can use that for the hash 
> calculation to compare to. Config server should also prompt the user to 
> change it when they log in if possible so it forces them to move to the "new 
> pin format".

yep, exactly the problem and a very usable solution IMHO.
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Re: [sipx-users] backup/restore: test 4.4

[Douglas Hubler]

On Tue, Mar 1, 2011 at 3:29 PM, Tony Graziano
 wrote:
> Agreed, it's a royal pain. Does the JIRA have a "pledge request" (so we can
> track that)? LOL.
> There are probably several ways to skin this, but I think it's been
> discussed before. The single biggest obstacle being how the PIN and DOMAIN
> are tied together for authentication purposes as an encoded value... that's
> a potentially "BIG" change that is sorely needed (3 years ago).

you're right, I rarely care about the PINs because i just reset them
for my purposes.

I once verified PIN can be encoded anyway we want as it's only used
internally. Hurdle was backward compatibility and I couldn't convince
anyone that the time to tackle documentation/explanation/migration but
i'll keep at it.
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Re: [sipx-users] backup/restore: test 4.4

[Douglas Hubler]

On Tue, Mar 1, 2011 at 12:41 PM, Tony Graziano
 wrote:
> Right. Manual iso install, restore users/phones via import. Manually add
> gateways, etc.

My pledge
   "We will not ship another sipXecs with solving back/restore to
separate domain or ip again."

I hit this all the time and this has been an issue for as long as i
can remember.  Looking in jira, this has be "fixed" many times but
some new feature ends up breaking things again.  We need to fix this
in a different way and keep it from breaking.

I usually run backup and restore at the postgres level then change
domain name using psql before starting system. Next time i need to
restore a system, i will create a wiki page that describe this.
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Re: [sipx-users] Q About port 5060 trunking support on 4.4

[Douglas Hubler]

On Tue, Mar 1, 2011 at 1:35 PM, Michael Scheidell
 wrote:
> rumor has it, you can use port 5060 for both trunking and remote users/sip:
> calls.
>
> If I set it up this way, is the src port on sipx also 5060?  I guess it
> would have to be for bydirectional 5060/5060 support.

Only inbound I think.  Proxy detects something for bridge and passes
it along. Joegen may know more.  You have to enable it on the proxy
advanced settings.  "Enable bridge-proxy relay"
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Re: [sipx-users] where can I download 4.4 i386 iso?

[Douglas Hubler]

On Tue, Mar 1, 2011 at 1:24 PM, Michael Scheidell
 wrote:
> On 3/1/11 1:17 PM, Douglas Hubler wrote:
> You caught me in the middle of updating the ISOs.  Refresh page in 5 min.
> what is the memory leak fix?

memory leak was for 4.0.4 systems in HA mode.  fix was already pushed
awhile ago so it shouldn't show up there, so i will remove it.
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Re: [sipx-users] where can I download 4.4 i386 iso?

[Douglas Hubler]

You caught me in the middle of updating the ISOs.  Refresh page in 5 min.

On Tue, Mar 1, 2011 at 1:14 PM, Michael Scheidell
 wrote:
>
> links to amazon seem broken.
>
> 
>
> gets this no matter where I pull it from:
>
> 
> AccessDeniedAccess
> DeniedCC25DB8E858938AAdjAEknJCy4sFYDWJJMDO1
> 1f2Xk5VVlTqIoc2JuITAmeuZnGuYyh6vQ6ER0OxW2HK
>
>
>
> --
> Michael Scheidell, CTO
> o: 561-999-5000
> d: 561-948-2259
> ISN: 1259*1300
>> | SECNAP Network Security Corporation
>
> Certified SNORT Integrator
> 2008-9 Hot Company Award Winner, World Executive Alliance
> Five-Star Partner Program 2009, VARBusiness
> Best in Email Security,2010: Network Products Guide
> King of Spam Filters, SC Magazine 2008
>
> 
>
> This email has been scanned and certified safe by SpammerTrap®.
> For Information please see http://www.secnap.com/products/spammertrap/
>
> 
>
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Re: [sipx-users] When is 4.4 coming?

[Douglas Hubler]

On Tue, Mar 1, 2011 at 11:27 AM, m...@grounded.net  wrote:
> On Tue, 1 Mar 2011 11:16:18 -0500, Tony Graziano wrote:
>> as always, there is, you just arent looking in the ISO directory
>>
>> http://download.sipfoundry.org/pub/sipXecs/
>> http://download.sipfoundry.org/pub/sipXecs/ISO/
>
> This is where I found the one I posted about earlier. It's the exact same 
> name/version info as the one I picked up a while back.
> I'm not sure how the versioning works I guess but at least I've got the 
> latest.


if you downloaded a file w/exact same file name, then it will be the
exact same file.  Occasionally i build RPMs but not ISO, but updating
from ISO install should be simple "yum update" away.
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Re: [sipx-users] Do 4.0.4 backups include call report data?

[Douglas Hubler]

On Tue, Mar 1, 2011 at 10:50 AM, Tony Graziano
 wrote:
> Is there a JIRA/Improvement to do a periodic backup of "just" CDR data via
> sipxconfig?

If there isn't, there should be.  Also, backing up CDRs should be
there by default.  Can you create one?
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Re: [sipx-users] When is 4.4 coming?

[Douglas Hubler]

4.4 is out of dev. stage so you will be able to get updates to it,
however that doesn't mean a whole lot because i still wouldn't
recommend you upgrade a production system to it yet.By test, i
mean setup a spare box and verify the features you use most still
work.

On Mon, Feb 28, 2011 at 12:16 PM, m...@grounded.net  wrote:
>> y, that was a recent one that went to the top of the list.  the more
>> people that test like Josh, the faster we'll release.  Thanks Josh!
>
> The only reason I've not is because there's no upgrade to stable path from 
> dev.
> I have no problem testing and more than happy to if you need more people 
> doing so.
>
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Re: [sipx-users] When is 4.4 coming?

[Douglas Hubler]

On Mon, Feb 28, 2011 at 12:00 PM, m...@grounded.net  wrote:
> On Mon, 28 Feb 2011 16:27:49 +, Josh M. Patten wrote:
>> Certainly not until http://track.sipfoundry.org/browse/XX-9475 is fixed.
>
> Darn, guess there's no stable version for those running 61 users or less eh? 
> :)

y, that was a recent one that went to the top of the list.  the more
people that test like Josh, the faster we'll release.  Thanks Josh!
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Re: [sipx-users] Billing information from sipX

[Douglas Hubler]

2011/2/25 Sven Evensen :
> A SIP trunk provider we are working with needs to use the fields
> History-Info or Diversion header
>
> to be able to pick up billing info after a forwarded or transferred call.
> PAI cannot be used as in
>
> Switzerland they have something call “special arrangement” which allows the
> true A party (even
>
> if it is external) to be sent as the A number of the transferred call.
> Therefore another field must
>
> be used for billing purpose.
>
>
>
> Any one know if these fields can be enabled/used in sipX 4.2 (or newer)?

The patch on here adds an alert-info header to sip messages using
redirector plugin api
  http://track.sipfoundry.org/browse/XX-5004
may give you an idea. I'm still looking for someone to update the
nearly empty docs
 http://wiki.sipfoundry.org/display/sipXecs/Call+Redirectors
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[sipx-users] reinstalling sipfoundry www/wiki/jira servers ...at some point

[Douglas Hubler]

In my copious spare time, I'm reinstalling the entire atlassian stack
with the latest versions of everything.  Main driver is that we've
found someone that has figured out how to integrate liferay forums
w/atlassian stack but needs a do-over with how everything is setup.

I'm hoping this will address folks getting randomly locked out of
their account.

Priority keeps getting pushed down, but eventually it will get done.
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Re: [sipx-users] Call park with presence in 4.4

[Douglas Hubler]

On Wed, Feb 23, 2011 at 9:43 PM, Jeff Ferrara  wrote:
> When using sipXecs 4.4 with Polycom IP650's is there any way to set up a
> call park with presence indication - i.e. use the line lights to display
> the park status?
> In 4.2 I used a "speed dial with presence" to the call park extension
> which seemed to work, but in 4.4 this has issues (stuck attendances and
> pressing on the speed dial key does not retrieve the call).
>
> I am testing with the latest build available in the CENTOS repo (4.4.0-
> 2011-02-19EST01:01:58 swift)
>
>  Has anyone got this working correctly in the beta?

Call park itself may have issues in 4.4. Works once then fails second
time.  I was supposed to be reviewing a patch that claims TLS broke
it, which seems strange, but i have not had time to review it so I
will try to get some help tomorrow.
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Re: [sipx-users] sipxecs-setup won't wipe out old data

[Douglas Hubler]

On Wed, Feb 23, 2011 at 2:56 PM, Michael Scheidell
 wrote:
> I am sure I. Really borked it up.  Esp if I still had a web interface on 
> secondary.
> I will try a custom config. If all else fails. Ill just put up fs to do aa vm 
> and forward

I you go this route, FS that is available in sipxecs 4.4 is 99.9%
straight from FS project including /etc/init.d script.  Only patches I
made was to get it to build. So you can install it w/o sipxecs.
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Re: [sipx-users] sipxecs-setup won't wipe out old data

[Douglas Hubler]

On Wed, Feb 23, 2011 at 8:59 AM, Michael Scheidell
 wrote:
> I have a remote sipx server I am trying to configure as an ha server.
> previously, I had restored our main sipx config to it (so it thinks its the
> primary server), but to fix it, I tried to run sipxecs-setup
>  from cli.
>
> sipxecs-setup says it will wipe out existing config, but it doesn't.
> (tried twice)

maybe you say it, but it's not clear to me specifically why do believe this

> step #1: create secondary server, get password (use public ip address)
> step #2: on secondary (which has old config) run sipxecs-setup, put in
> public ip of primary
> follow steps, and when get to last screen where you have the option of
> 'start sipx services', command line or reboot, I have tried start sipx
> services, and reboot.
>
> http://newsecondary server still shows the original config, not the new
> secondary config.

what do you mean by "http://newsecondary";  only master runs a web interface.

> the primary shows both, and shows secondary registered.
> (and both show services that fail to start)
>
> so, without my driving back to secondary noc, how do I 'start from scratch'?

Has instructions on reinstalling
http://wiki.sipfoundry.org/display/sipXecs/Uninstalling+sipXecs
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Re: [sipx-users] Lookup name for incoming call

[Douglas Hubler]

On Mon, Feb 21, 2011 at 9:14 PM, Alistair J. Fenning
 wrote:
> As far as I can see it is not currently possible to have sipXecs try and
> match incoming caller ID with a name (either from the inbuilt Phonebook or
> external service) and then re-write the SIP message with the caller’s name.
>
> Could anyone give me an indication of if this is likely to be a
> soon-implemented feature (I’m guessing not as I can’t find much discussion
> of it) or does it stray too far from sipX being, at its heart, a proxy
> server?
>
> Also, I would be interested to know if the architecture of sipXecs is such
> that I might be able to implement this feature myself via API calls (or
> somehow else without otherwise having to compile my own version and
> effectively fork the project)

David Becker just wrote a redirector plugin to add alert-info to
incoming SIP messages. I don't see why you couldn't transform To:
field.

Not a lot of documentation but maybe you can help out.

   http://wiki.sipfoundry.org/display/sipXecs/Call+Redirectors

Here's the patch, it includes WEB UI administration that you may or
may not need.

  http://track.sipfoundry.org/browse/XX-5004
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Re: [sipx-users] Polycom custom config files patch

[Douglas Hubler]

On Mon, Feb 21, 2011 at 5:45 AM, George Niculae  wrote:
> Nice patch - have a question here, should the  within
> generated *-sip.cfg.xml contain the newly introduced custom-configs?
> Right now it appends it as an attribute, (notice
> feature.19.name="custom-configs" and feature.19.enabled=""):

No it should not, i didn't check that and should have.  Frankly
features is the wrong place.  I'll create a new group and move it
there.
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Re: [sipx-users] Polycom custom config files patch

[Douglas Hubler]

On Sat, Feb 19, 2011 at 11:31 AM, Douglas Hubler  wrote:
> On Fri, Feb 18, 2011 at 11:36 PM, Douglas Hubler  wrote:
>> On Fri, Feb 18, 2011 at 10:58 PM, Jim Canfield  wrote:
>>> On Fri, Feb 18, 2011 at 2:46 PM, Josh M. Patten  
>>> wrote:
>>>> Hey if anyone uses the custom config files patch written by Eric Varsanyi
>>>> that I’ve been posting about lately and would like it to appear in 4.4
>>>> without custom modification, please vote on the issue here:
>>>> http://track.sipfoundry.org/browse/XX-8725 For those that don’t know, it
>>>> will allow you to do lots of awesome things with your Polycom phones,
>>>> details here:
>>>> http://sites.google.com/site/sipxecstipsandtricks/polycom-phones
>>>
>>> Big fat +1!  I'm not sure how one can even deploy Polycom properly w/o
>>> this patch?   ...I voted, but still zero votes.  I thought I'd at
>>> least count for a 1/2 vote.
>>
>> I reviewed the patch and looks good. I'll get it in.
>
> Actually, I have one question, why can't we have just one CSV setting value
>  "External Config FIles:"
> And allow groups and phones to override, change order, etc,  This
> eliminates all the exclusive/cumulative technique and honors how phone
> groups work today for all other settings.
>
> Example:
> Group Name :  All Polycoms
> External Configuration Files: custom-polycom.cfg
>
> Group Name :  Lobby Phones (defined only if the lobby phones are different)
> External Configuration Files: custom-polycom-lobby.cfg, custom-polycom.cfg
>
> Phone 004f:  (if John wants to be different)
> External Configuration Files: custom-polycom.cfg, johns-ringtones.cfg
>
> Phone 004f3334:   (if the lobby phone near john wants to be different)
> External Configuration Files: custom-polycom-lobby.cfg,
> custom-polycom.cfg, johns-ringtones.cfg
>
> This seems more simple, allows complete flexibility.  Phones that
> override external file list will need to be updated if any of it's
> groups change their external file list.

I submitted patch as I spec'ed it here to main but I'll back port 4.4
it tomorrow.
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Re: [sipx-users] Polycom custom config files patch

[Douglas Hubler]

On Fri, Feb 18, 2011 at 11:36 PM, Douglas Hubler  wrote:
> On Fri, Feb 18, 2011 at 10:58 PM, Jim Canfield  wrote:
>> On Fri, Feb 18, 2011 at 2:46 PM, Josh M. Patten  
>> wrote:
>>> Hey if anyone uses the custom config files patch written by Eric Varsanyi
>>> that I’ve been posting about lately and would like it to appear in 4.4
>>> without custom modification, please vote on the issue here:
>>> http://track.sipfoundry.org/browse/XX-8725 For those that don’t know, it
>>> will allow you to do lots of awesome things with your Polycom phones,
>>> details here:
>>> http://sites.google.com/site/sipxecstipsandtricks/polycom-phones
>>
>> Big fat +1!  I'm not sure how one can even deploy Polycom properly w/o
>> this patch?   ...I voted, but still zero votes.  I thought I'd at
>> least count for a 1/2 vote.
>
> I reviewed the patch and looks good. I'll get it in.

Actually, I have one question, why can't we have just one CSV setting value
 "External Config FIles:"
And allow groups and phones to override, change order, etc,  This
eliminates all the exclusive/cumulative technique and honors how phone
groups work today for all other settings.

Example:
Group Name :  All Polycoms
External Configuration Files: custom-polycom.cfg

Group Name :  Lobby Phones (defined only if the lobby phones are different)
External Configuration Files: custom-polycom-lobby.cfg, custom-polycom.cfg

Phone 004f:  (if John wants to be different)
External Configuration Files: custom-polycom.cfg, johns-ringtones.cfg

Phone 004f3334:   (if the lobby phone near john wants to be different)
External Configuration Files: custom-polycom-lobby.cfg,
custom-polycom.cfg, johns-ringtones.cfg

This seems more simple, allows complete flexibility.  Phones that
override external file list will need to be updated if any of it's
groups change their external file list.
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Re: [sipx-users] Polycom custom config files patch

[Douglas Hubler]

On Fri, Feb 18, 2011 at 10:58 PM, Jim Canfield  wrote:
> On Fri, Feb 18, 2011 at 2:46 PM, Josh M. Patten  
> wrote:
>> Hey if anyone uses the custom config files patch written by Eric Varsanyi
>> that I’ve been posting about lately and would like it to appear in 4.4
>> without custom modification, please vote on the issue here:
>> http://track.sipfoundry.org/browse/XX-8725 For those that don’t know, it
>> will allow you to do lots of awesome things with your Polycom phones,
>> details here:
>> http://sites.google.com/site/sipxecstipsandtricks/polycom-phones
>
> Big fat +1!  I'm not sure how one can even deploy Polycom properly w/o
> this patch?   ...I voted, but still zero votes.  I thought I'd at
> least count for a 1/2 vote.

I reviewed the patch and looks good. I'll get it in.

re:votes
Election is rigged!  Actually I'm not sure what is wrong w/the voting
system on this issue, works fine for other issues. We're upgrading
jira soon, so I won't worry about it for now.
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Re: [sipx-users] Web Portal Modification

[Douglas Hubler]

On Thu, Feb 17, 2011 at 10:52 AM, andrewpitman  wrote:
>
> Content-Type: text/plain;
>  charset="utf-8"
> Content-Transfer-Encoding: 8bit
> Organization: SipXecs Forum
> In-Reply-To: 
> X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <56971>
> Message-ID: 
>
>
>
> Oh, and layout.css would go under there as well.

I'm only seeing Andrew's replies because original poster is not
subscribed to mailing list, but you'll see in this doc

 http://wiki.sipfoundry.org/display/sipXecs/Customizing+Colors,+Layout+and+Logo

you do not need to rebuild sipXconfig RPM or even build any RPM to
change skin.  Simply create key files in key locations as described in
this document.
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Re: [sipx-users] 4.4 - need to raise limits for restore size

[Douglas Hubler]

no issues filling.
it should at least be configuable

On Wed, Feb 16, 2011 at 12:26 PM, Matthew Kitchin (public/usenet)
 wrote:
> I'm not sure if this is true in all versions, but in 4.4, you cannot restore 
> a backup over about 1 GB using the upload feature in the GUI.
> This is the code causing this:
>
> sipXconfig.war/WEB-INF/hivemodule.xml -
>>
>>     
>>     >  service-id="tapestry.multipart.ServletMultipartDecoder">
>>       >
>>
>>  class="org.apache.tapestry.multipart.MultipartDecoderImpl,maxSize=1073741824"
>>  model="threaded" />
>>     
>>
>
> Anyone see any reason I shouldn't open a jira requesting this limit be 
> raised? VM backups are over 1 GB at most of my sites.
>
> Thanks,
> Matthew
>
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>
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Re: [sipx-users] Language selection by dialing

[Douglas Hubler]

On Wed, Feb 16, 2011 at 10:13 AM, Bob Blanchard Jr.  wrote:
> I have found a hack to change the language on a transfer to another
> auto-attendant, involving passing "locale" on a transfer to a URL.
>
> First, create the second-language auto-attendant (in our case french) and
> determine the "schedule_id".  You can determine the schedule_id by creating
> a temporary Dial Plan for the new auto-attendant and checking
> /etc/sipxpbx/mappingrules.xml...  search for action=autoattendant within the
> newly created dial plan.
>
> >From the english auto-attendant, create a dialpad action "Transfer to
> Extension or Other Destination", and put the URL obtained from the
> mappingrules.xml, with the appropriate locale:  eg.
> 
>
> The only problem remaining is that after dialing french option, the message
> "Please hold while I transfer your call" plays before sending you to the
> french auto-attendant.
>
> Is there a way to suppress this message on transfer?

Re-record the wav with silence?

Nice work figuring this out.  If you edit mapping rules directly,
doesn't it get overwritten on changes?  There used to be a way to
write custom mapping rules that gets merged into the mapping rules one
build in the UI.  Would that help you here?

Can you open a feature request to make this prompt localizable as well?
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Re: [sipx-users] Spectralink 8020 up and running (was: Cisco WIP310 - wifi issues)

[Douglas Hubler]

On Tue, Feb 15, 2011 at 1:30 PM, Matthew Kitchin (public/usenet)
 wrote:
> I am up and running with a Polycom Spectralink 8020.
> I had tried this model once before but gave up.
> There were 2 big differences this time.
> 1) I called Polycom sales first and was given the contact info of an
> engineer that started out with Spectralink
> 2) I got the dual battery charging station that allows you to edit the phone
> directly with the handset administration tool.
>
> I will keep test driving it to see if if I have any issues, but it appears
> to work fine. Parts of the interface are a little quirky compared to a
> Soundpoint 450/550/650, but it looks like it will do the job for a basic
> phone.
>
> Let me know if anyone is interested in any tips on this phone.

please add page w/config notes on
  http://wiki.sipfoundry.org/display/sipXecs/Polycom+Phones
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Re: [sipx-users] OpenACD: XX-9201 - proposal design

[Douglas Hubler]

On Tue, Feb 15, 2011 at 12:14 PM, Victor Williams
 wrote:
> Only thing I would change is the font.  Comic Sans looks a bit...comicky.  I
> prefer Arial...easier to read and looks a bit more professional.

Mock-ups are meant to be more like pencil sketches, common technique
in UI mock-ups to keep reviewer focused on layout and content, not on
the fonts and colors.  Had opposite effect in this case  ;)

FYI: My feedback was
- Agent table on each edit agent group apge was redundant with new
agents interface
- I'd like to see the tabs navigation on more pages
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Re: [sipx-users] Major effort cleanup to wiki complete

[Douglas Hubler]

On Mon, Feb 14, 2011 at 4:47 PM, Geoff Van Brunt  wrote:
> Jira does work OK as a Roadmap but it requires a lot of reading to find
> out the details. I can use Jira, but not everyone can... You have to
> read through an awful lot of entries to find out what is included in new
> release. The new features are often buried in amongst, bugfixes,
> enhancements etc. Often what an end user would think of as one issue is
> spread among several tickets etc. I don't know if you can work around
> those limitations to make an automatic RoadMap for the Wiki that most
> users could understand, but it sure would be nice.

good points.

I whipped together some highlights.
  http://wiki.sipfoundry.org/display/sipXecs/Roadmap
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Re: [sipx-users] Major effort cleanup to wiki complete

[Douglas Hubler]

On Mon, Feb 14, 2011 at 2:31 PM, Michal Bielicki
 wrote:
> Isn't there a function in the wiki to read the stuff out of jira ?

yeah, i plan to take advantage of this once jira is cleaned up.
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Re: [sipx-users] Major effort cleanup to wiki complete

[Douglas Hubler]

On Mon, Feb 14, 2011 at 12:51 PM, Geoff Van Brunt
 wrote:
> One thing absent which would be awesome is the re addition of the road
> map... :)

Very true!

Next up is jira cleanup and that will help drive roadmap and release notes.
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[sipx-users] Major effort cleanup to wiki complete

[Douglas Hubler]

The ezuce team: joegen, cristi, mircea, george, laurentui and myself
have just completed a major update to the wiki documentation over 2
full days.  We tried to remove old content, update existing content
and create critically missing content.  We also spent a lot of time
organizing the content into groups that will help find navigating the
site a lot easier and help us decide where new content should go.

We didn't get everything done we wanted to, but we got a lot done.
Those that were familiar with the old wiki are reporting a tremendous
improvement.

Couple Highlights:

-  "Hardware and Interoperability" section is about telephony hardware
and software and organized into easily navigable groups.  Please
contribute your configuration notes for others to appropriate section

  http://wiki.sipfoundry.org/display/sipXecs/Hardware+and+Interoperability

- Cookbooks

  This is somewhat of a catch-all for general tips for configuring a
system.  From how to setup custom SSL certificates to DNS tips and
tricks.  Please consider reviewing/updating and contributing to this
very useful category.

Also, search index is now fixed so as you update the wiki, moments
later search will find the page.  Also the "Recently Changed" feed is
updated.  Unfortunately the plugin management is still broken but this
only seems to affect a single page that used graphviz image.  Someday
I hope to figure out what's wrong with this, but not a blocker.
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Re: [sipx-users] Patton 5200 SBC Template

[Douglas Hubler]

On Fri, Feb 11, 2011 at 5:33 PM, Jim Canfield  wrote:
> Finally have a working Patton 5200 SBC config working on Voip.ms as B2BUA.
> Here's a rough draft template for anyone who might be interested.

There's a great place for your notes in a page added off of this page.

  http://wiki.sipfoundry.org/display/sipXecs/Patton+Gateways
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Re: [sipx-users] Audiocodes IP phones

[Douglas Hubler]

On Thu, Feb 10, 2011 at 9:27 AM, Gmb  wrote:
> Hi all,
> has anyone tried Audiocodes 3x0HD IP phones?
> Is the provisioning mask of these phone in roadmap? I haven't found
> nothing about that...

i have a 320HD i'd love to put to use one day.  only thing i've heard
negative was that the speed-dial legend you fill in with a pencil is
old school. I would see if it's similar to new karel phone plugin, may
get lucky.

have you manually configured phone? if you have any experience, please
share and create a wiki page under here
  http://wiki.sipfoundry.org/display/sipXecs/Phones
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Re: [sipx-users] Update 4.4.0

[Douglas Hubler]

On Thu, Feb 10, 2011 at 8:01 AM, Ben Goodfellow wrote:

> I just tried to update 4.4.0 using the yum update command and receive the
> following error – can anyone shed any light?
>
>
>
> --> Finished Dependency Resolution
>
> dirac-1.0.2-1.el5.rf.x86_64 from installed has depsolving problems
>
>   --> Missing Dependency: libcppunit-1.12.so.0()(64bit) is needed by
> package dirac-1.0.2-1.el5.rf.x86_64 (installed)
>
> Error: Missing Dependency: libcppunit-1.12.so.0()(64bit) is needed by
> package dirac-1.0.2-1.el5.rf.x86_64 (installed)
>
> You could try using --skip-broken to work around the problem
>
> You could try running: package-cleanup --problems
>
> package-cleanup --dupes
>
> rpm -Va --nofiles –nodigest
>

I have no idea what dirac is, but sipxecs building doesn't need it.  sipxecs
> should also work with any cppunit version so dirac if you need dirac and it
> needs a specific version of cppunit, i think you can use --exclude-package
> in yum command to not update cppunit
>
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[sipx-users] Major wiki work by eZuce tomorrow and friday (2/10-2/11)

[Douglas Hubler]

6 guys:joegen, laurentui, cristi, george, mircea and myself are going
to spend two full days  updating the wiki.  We'll be coordination in
IRC #sipx on freenode if you're interested.

Also, I'll be upgrading confluence tonight in hopes it fixes the
indexing problem.
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Re: [sipx-users] Turning on RTCP-XR for Polycom phones with PPS

[Douglas Hubler]

On Tue, Feb 8, 2011 at 6:08 PM, McIlvin, Don
 wrote:
> FYI – Just thought I would mention the following solution for those that
> have Polycom Productivity Suite and want to enable reports (i.e. turn it
> on).
>
> The RTCP-XR-Enable.cfg file simply sets
> voice.qualityMonitoring.rtcpxr.enable to “1”.

Can you open an "improvement request" in jira w/your attached files.
We can add this as optional polycom settings fairly easily.
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Re: [sipx-users] Bilingual voicemail system

[Douglas Hubler]

On Tue, Feb 8, 2011 at 2:25 PM, Bob Blanchard Jr.  wrote:
> Regarding setting up a truly bilingual voicemail system, we found this post:
>
> http://www.mail-archive.com/sipx-dev@list.sipfoundry.org/msg01992.html
>
> But we were not able to get this to work at all.

What are the details of your configuration and results of your tests.
What version of sipXecs are you using?
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Re: [sipx-users] voicemail box is full

[Douglas Hubler]

On Mon, Feb 7, 2011 at 3:40 PM, Michael Scheidell
 wrote:
> as an administrator, I would like to know lusers who let their voicemail box
> get full.

define "full"

> How do I do that?

depends on answer above

> logs? alerts.

Ultimately it seems alerts would provide the most flexibility on how
to control notification.  Warnings back to the user in the form of
prompts in the TUI and web console would make sense.

quick google indicates it's been asked for before, check tracker though
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Re: [sipx-users] hairpin transfer from/to pstn causes error :SipUtilities.java:837

[Douglas Hubler]

  at
>> org.sipfoundry.sipxbridge.SipUtilities.createInviteRequest(SipUtilities.java:687)
>>        ... 11 more
>>
>> "2011-02-03T10:41:49.164000Z":136258:JAVA:INFO:sipx.creativevalley.nl:PipelineThread-4::sipXbridge:"[SIPDialog.java:2635][SIPServerTransaction.java:1392][CallControlUtilities.java:49][BackToBackUserAgent.java:1922][CallControlManager.java:621][CallControlManager.java:3054][SipListenerImpl.java:451][EventScanner.java:224][SipProviderImpl.java:192][DialogFilter.java:1138][SIPServerTransaction.java:823][TCPMessageChannel.java:515][PipelinedMsgParser.java:361][Thread.java:619]"
>>
>> "2011-02-03T10:41:49.164000Z":136259:JAVA:INFO:sipx.creativevalley.nl:PipelineThread-4::sipXbridge:"[SIPDialog.java:1233][SIPDialog.java:2829][SIPServerTransaction.java:1392][CallControlUtilities.java:49][BackToBackUserAgent.java:1922][CallControlManager.java:621][CallControlManager.java:3054][SipListenerImpl.java:451][EventScanner.java:224][SipProviderImpl.java:192][DialogFilter.java:1138][SIPServerTransaction.java:823][TCPMessageChannel.java:515][PipelinedMsgParser.java:361][Thread.java:619]"
>>
>> "2011-02-03T10:41:49.164000Z":136260:JAVA:INFO:sipx.creativevalley.nl:PipelineThread-4::sipXbridge:"[SipProviderImpl.java:182][SIPServerTransaction.java:1399][CallControlUtilities.java:49][BackToBackUserAgent.java:1922][CallControlManager.java:621][CallControlManager.java:3054][SipListenerImpl.java:451][EventScanner.java:224][SipProviderImpl.java:192][DialogFilter.java:1138][SIPServerTransaction.java:823][TCPMessageChannel.java:515][PipelinedMsgParser.java:361][Thread.java:619]"
>>
>> "2011-02-03T10:41:49.164000Z":136261:JAVA:INFO:sipx.creativevalley.nl:PipelineThread-4::sipXbridge:"[SIPTransaction.java:556][SIPServerTransaction.java:1476][SIPServerTransaction.java:991][SIPServerTransaction.java:1419][CallControlUtilities.java:49][BackToBackUserAgent.java:1922][CallControlManager.java:621][CallControlManager.java:3054][SipListenerImpl.java:451][EventScanner.java:224][SipProviderImpl.java:192][DialogFilter.java:1138][SIPServerTransaction.java:823][TCPMessageChannel.java:515][PipelinedMsgParser.java:361][Thread.java:619]"
>>
>> "2011-02-03T10:41:49.165000Z":136262:OUTGOING:INFO:sipx.creativevalley.nl:PipelineThread-4::sipXbridge:"Sent
>> SIP Message :
>> Remote Host:192.168.4.10 Port: 35387
>> SIP/2.0 500 Server Internal Error
>> Call-ID: D82A556B-2EB811E0-9D39F69A-3FEFE6E@82.148.198.254.1.1
>> CSeq: 1 INVITE
>> To: \"anonymous\" 
>> Via: SIP/2.0/TCP
>> 192.168.4.10;branch=z9hG4bK-XX-e13b6oBPr0Y1x2lZf7pUK1GsGQ;rport=35387
>> Via: SIP/2.0/UDP
>> 192.168.4.10;branch=z9hG4bK-XX-e138ukdlYLPFbFsP6XF6Alqlgg~JsjkAkjH78Fr7WQUFtexUQ
>> Via: SIP/2.0/UDP
>> 192.168.4.10:5090;branch=z9hG4bKc8695c6e66c250a6f8bdf4d43acf0e85333236;sipxecs-id=7591a1e0
>> From: \"anonymous\" ;tag=4066527
>> Server: sipXecs/4.2.1 sipXecs/sipxbridge (Linux)
>> Content-Type: message/sipfrag
>> Content-Length: 73
>>
>> Exception Info Unexpected error creating INVITE  at
>> SipUtilities.java:837END
>> "
>>
>> "2011-02-03T10:41:49.165000Z":136263:JAVA:INFO:sipx.creativevalley.nl:PipelineThread-4::sipXbridge:"[SIPTransaction.java:1197][SIPTransaction.java:1183][SIPServerTransaction.java:1708][EventScanner.java:255][SipProviderImpl.java:192][DialogFilter.java:1138][SIPServerTransaction.java:823][TCPMessageChannel.java:515][PipelinedMsgParser.java:361][Thread.java:619]"
>>
>> "2011-02-03T10:41:49.165000Z":136264:JAVA:INFO:sipx.creativevalley.nl:PipelineThread-4::sipXbridge:"[SIPTransaction.java:1197][SIPTransaction.java:1183][SIPServerTransaction.java:1708][TCPMessageChannel.java:520][PipelinedMsgParser.java:361][Thread.java:619]"
>>
>> "2011-02-03T10:41:49.166000Z":136265:JAVA:INFO:sipx.creativevalley.nl:Thread-7371::sipXbridge:"[SIPTransaction.java:1152][SIPTransactionStack.java:1460][UDPMessageChannel.java:527][UDPMessageChannel.java:459][UDPMessageChannel.java:295][Thread.java:619]"
>>
>> "2011-02-03T10:41:49.166000Z":136266:INCOMING:INFO:sipx.creativevalley.nl:Thread-7371::sipXbridge:"Read
>> SIP Message:
>> Remote Host:192.168.4.10 Port: 5060
>> SIP/2.0 500 Server Internal Error
>> Call-ID: D82A556B-2EB811E0-9D39F69A-3FEFE6E@82.148.198.254.1.1
>> CSeq: 1 INVITE
>> To: \"anonymous\" 
>> Via: SIP/2.0/UDP
>> 192.168.4.10:5090;branch=z9hG4bKc8695c6e66c250a6f8bdf4d43acf0e85333236;sipxecs-id=7591a1e0
>> From: \"anonymous\" ;tag=4066

Re: [sipx-users] hairpin transfer from/to pstn causes error :SipUtilities.java:837

[Douglas Hubler]

On Fri, Feb 4, 2011 at 4:49 AM, Peter van der Salm
 wrote:
> Then SipXproxy sends and INVITE to sipXbridge to setup the call to
>  0642769062
> sipXbridge returns a 500 Server Internal Error.

I would think /var/log/sipxpbx/sipxbridge.log should have an error message.
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Re: [sipx-users] Answering a call with 4.4 and Polycom phones

[Douglas Hubler]

On Thu, Feb 3, 2011 at 3:02 PM, Burleigh, Matt
 wrote:
> I’ve been experimenting with 4.4 and it seems that all of my Polycom phones
> are acting differently then they did with 4.2.1. When a calls comes in I
> used to be able simply pickup the handset and begin speaking to the caller.
> Now with 4.4, I have to press “Answer” button before I’m able to able to
> speaking. What option controls this behavior?

The polycom-config command will flatten out the xml into a file that
can be diff'ed

So if you run
polycom-config
/var/sipxdata/configserver/phone/profile/tftproot/1234567890ab* >
4.4.config


And compare the output w/a 4.2.1 system you see what settings have
changed.  I don't know of any that would cause this, my polycom picks
right up on 4.4
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Re: [sipx-users] How to send media data ?

[Douglas Hubler]

Wrong list and project.if you ultimately find the right list let me know so
I can help other lost folks
On Jan 31, 2011 8:26 PM, "Li, Yvonne"  wrote:
> sipxCallInjectMediaPacket I guess.
>
> 
> From: Li, Yvonne
> Sent: Monday, January 31, 2011 3:15 PM
> To: 'sipx-users@list.sipfoundry.org'
> Subject: How to send media data ?
>
>
> I am interested in using sipXtapi for transporting data. sipXtapi has
sipxConfigSetMediaPacketCallback method for application to process media
data received on its own. However, I can't find a way to send media data. Is
it supported?
>
> Thanks,
>
> Yvonne
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Re: [sipx-users] User Phone Assignment?

[Douglas Hubler]

Only pages that were blatant copies were deleted. If you go to page
hierarchy you can move pages about very easily instead of editing page
parents
On Jan 31, 2011 6:08 PM, "Josh M. Patten"  wrote:
> The main page for user guides for 4.2 seems to have disappeared on the
wiki, but I wrote guides some time back. I'll work on restoring that page
eventually...
>
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org [mailto:
sipx-users-boun...@list.sipfoundry.org] On Behalf Of Douglas Hubler
> Sent: Monday, January 31, 2011 3:26 PM
> To: Discussion list for users of sipXecs software
> Subject: Re: [sipx-users] User Phone Assignment?
>
> On Mon, Jan 31, 2011 at 3:01 PM, Burleigh, Matt <
matt.burle...@eiisolutions.net> wrote:
>> Is it possible to allow a user permission to add a phone to their
account?
>> We had this functionality in ShoreTel where a user can assign his
>> extension to a hard phone. And do this from a phone. Is this possible
with SipX?
>
> Many customers seems to have custom provisioning needs and wind up writing
a custom script. The script running in on your backend server would need
superadmin privs. but you can provide whatever access you want on the front
end. All this provisioning and line management is available via SOAP, so
most computer languages should be able to work for you.
>
>> Also is there a user oriented "User Guide" of sorts that is fairly
current?
>> I can't seem to find one.
>
> There is a user oriented book on amazon. Otherwise, the wiki is
disorganized as far as category, but lots of gems if you search on topics.
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Re: [sipx-users] User Phone Assignment?

[Douglas Hubler]

On Mon, Jan 31, 2011 at 3:01 PM, Burleigh, Matt
 wrote:
> Is it possible to allow a user permission to add a phone to their account?
> We had this functionality in ShoreTel where a user can assign his extension
> to a hard phone. And do this from a phone. Is this possible with SipX?

Many customers seems to have custom provisioning needs and wind up
writing a custom script. The script running in on your backend server
would need superadmin privs. but you can provide whatever access you
want on the front end.  All this provisioning and line management is
available via SOAP, so most computer languages should be able to work
for you.

> Also is there a user oriented “User Guide” of sorts that is fairly current?
> I can’t seem to find one.

There is a user oriented book on amazon.  Otherwise, the wiki is
disorganized as far as category, but lots of gems if you search on
topics.
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Re: [sipx-users] Clearing CSR?

[Douglas Hubler]

On Sat, Jan 29, 2011 at 8:00 PM, m...@grounded.net  wrote:
> I generated a CSR file using the GUI but it's only 1024bit while godaddy only 
> accepts 2048.

really? hmm, we should bump this up then, please file a bug

> I re-created my file using the correct bit size and got my godaddy cert but 
> now can't seem to find a way to remove the CSR I generated from the GUI.
>
> Can someone tell me which file I should delete in order to get this back to a 
> starting point so I can import my Certificate & Key File.

http://wiki.sipfoundry.org/display/sipXecs/Notes+on+SSL+Keys+and+Keystores+used+by+sipx
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Re: [sipx-users] ACD testing in 4.4

[Douglas Hubler]

On Fri, Jan 28, 2011 at 10:01 PM, Matthew Kitchin (public/usenet)
 wrote:
> I seem to be one of the heavier ACD users. I would definitely like to
> test this out in the latest version and see if I can provide helpful
> feedback.
> Is the ACD role the old ACD and Call Center the new one?That was just a
> guess after looking at the GUI.

y - you'll find details here
http://wiki.sipfoundry.org/display/sipXecs/OpenACD

> Sorry if this has already been answered. Is the new ACD supposed to
> import settings from the old, or is it a from scratch setup?

the two are independent, in fact you could use both at the same time
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Re: [sipx-users] upgrade from 4.2.1 to 4.4.0

[Douglas Hubler]

On Fri, Jan 28, 2011 at 5:52 PM, Tony Graziano
 wrote:
> No it actually doesn't always.
>
> That is why it is a snapshot "development" version.
>
> I've had to start over several times myself. That is expected in a
> non-production release cycle.
>
> Let's not muddy the waters here. 4.2.1 to 4.4 (pre pre beta) is certainly
> not going to work as an in place upgrade at this time... so be patient or...
>
> Install 4.4 separately and export/import your data and be prepared to move
> greetings, voicemail, etc. manually, etc.
>
> I think there were 4 last items being worked on before the upgrade becomes
> beta. Those are still open and being worked on. I prefer to let the
> developers work on those 4 items than to ask them to stop focusing on their
> current tasks to make it upgradeable at every minor version, but that's me.

there are no known issues with upgrading from 4.2.1 to 4.4 so I would
encourage people try this on a *non-production* box and report any
issues.   More reported bugs may delay the release, but it just means
we have a better release.

as far as backing up an older version and restoring onto a newer one,
yes, that *used* to be discouraged, but I think the policy changed on
that at some point.  Can anyone confirm that?  The only problem is
that the hostname, certs and ip address all gets restored so it's PITA
to use IMHO.
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Re: [sipx-users] upgrade from 4.2.1 to 4.4.0

[Douglas Hubler]

On Fri, Jan 28, 2011 at 5:03 PM, Charles Chalekson  wrote:
> Attempting to yum update to 4.4.0 from 4.2.1 and am getting a failure due to 
> some dependency issues.
>
> Am I doing something wrong?
>
> Thanks,
> Charles
>
>
> --> Running transaction check
> ---> Package erlang.i386 0:R13B04-1.3 set to be updated
> ---> Package freeswitch.i386 0:1.0.7-1441.g35129 set to be updated
> --> Processing Dependency: libogg.so.0 for package: freeswitch
> --> Processing Dependency: libtheora for package: freeswitch
> --> Processing Dependency: libtheora.so.0 for package: freeswitch
> --> Processing Dependency: libvorbis.so.0 for package: freeswitch
> --> Processing Dependency: libogg for package: freeswitch
> --> Processing Dependency: libvorbis for package: freeswitch
> --> Finished Dependency Resolution
> freeswitch-1.0.7-1441.g35129.i386 from sipXecs has depsolving problems
>  --> Missing Dependency: libvorbis.so.0 is needed by package 
> freeswitch-1.0.7-1441.g35129.i386 (sipXecs)
> freeswitch-1.0.7-1441.g35129.i386 from sipXecs has depsolving problems
>  --> Missing Dependency: libogg.so.0 is needed by package 
> freeswitch-1.0.7-1441.g35129.i386 (sipXecs)
> freeswitch-1.0.7-1441.g35129.i386 from sipXecs has depsolving problems
>  --> Missing Dependency: libtheora is needed by package 
> freeswitch-1.0.7-1441.g35129.i386 (sipXecs)
> freeswitch-1.0.7-1441.g35129.i386 from sipXecs has depsolving problems
>  --> Missing Dependency: libogg is needed by package 
> freeswitch-1.0.7-1441.g35129.i386 (sipXecs)
> freeswitch-1.0.7-1441.g35129.i386 from sipXecs has depsolving problems
>  --> Missing Dependency: libtheora.so.0 is needed by package 
> freeswitch-1.0.7-1441.g35129.i386 (sipXecs)
> freeswitch-1.0.7-1441.g35129.i386 from sipXecs has depsolving problems
>  --> Missing Dependency: libvorbis is needed by package 
> freeswitch-1.0.7-1441.g35129.i386 (sipXecs)
> Error: Missing Dependency: libvorbis is needed by package 
> freeswitch-1.0.7-1441.g35129.i386 (sipXecs)
> Error: Missing Dependency: libogg.so.0 is needed by package 
> freeswitch-1.0.7-1441.g35129.i386 (sipXecs)
> Error: Missing Dependency: libvorbis.so.0 is needed by package 
> freeswitch-1.0.7-1441.g35129.i386 (sipXecs)
> Error: Missing Dependency: libtheora is needed by package 
> freeswitch-1.0.7-1441.g35129.i386 (sipXecs)
> Error: Missing Dependency: libtheora.so.0 is needed by package 
> freeswitch-1.0.7-1441.g35129.i386 (sipXecs)
> Error: Missing Dependency: libogg is needed by package 
> freeswitch-1.0.7-1441.g35129.i386 (sipXecs)


are those libraries not available on your system? libogg should be in
the base package, did you let yum update your OS first getting you to
CentOS 5.5?
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Re: [sipx-users] SPAMAtuo-Attendant Fails to Pickup

[Douglas Hubler]

On Fri, Jan 28, 2011 at 8:25 PM, Wayne A. Green
 wrote:
> I have a very strange problem associated with the auto-attendant responding.
> I am currently running (4.2.1-018971.dhubler 2010-08-21T04:59:18 build34).
> This is a new installation and has been operational for approximately 1
> week. PSTN calling is operational in both directions and I am directly
> forwarding all incoming calls to 2 extensions. These features are working
> correctly. Yesterday I have a power issue and the system did not return as
> expected. After carefully inspecting the issues it was determined that the
> the second network interface (eth1) was set for dhcp and had over-written
> the DNS entries.

/etc/resolv.conf was overwritten?  which file or files were overwritten?

> The original entries were the sipxecs server.

what does this mean?

> This issue
> was resolved and the service was restored.

all the issues? the overwritten files where restored?

> Since the restoration of services
> the auto-attendant is not responding. When I attempt to retrieve voicemail I
> get the following error message:
> "Request Timeout 101"
> Additionally, if I attempt to connect to the "operation" I get the following
> error:
> "Request Timeout 0"
>
> Any assistance would be greatly appreciated!!!

look at /var/log/sipxpbx/sipxivr.log.

Do all mail users get this or, just one?  Because it's been reported
before that if a message descriptor isn't properly formated (i.e.
corrupt because of power outage), it can take out all sorts of voice
mail renderings.
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Re: [sipx-users] auto-attendant messages are played real fast

[Douglas Hubler]

On Fri, Jan 28, 2011 at 7:25 PM, srinivasa rao  wrote:
> could you please tell me the file names for the template files?

every file in that directory (not sub directories) is a template.
they end in .vm
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Re: [sipx-users] beta

[Douglas Hubler]

On Fri, Jan 28, 2011 at 5:58 PM, Charles Chalekson  wrote:
> I had not noticed even number version labeling as beta previously.  I always 
> thought developmental unstable versions were  odd numbered and when once 
> stable was changed to an even number.  I guess I know now.

development unstable are still odd numbers.  The number changes to
next even as it becomes beta.

As far as the beta release, there are still bugs coming in worthy of
fixing, so it makes sense to delay the release.
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Re: [sipx-users] interconnect with CS1000

[Douglas Hubler]

On Fri, Jan 28, 2011 at 4:24 AM, Nikolay Kondratyev  wrote:
> Hi, all,
> I continued to investigate my problem.
> And i found that when a call with that bloody ";phone-context=cdp.udp" in
> the user part of request uri is coming to "hostname" (which is configured as
> sip domain alias), that is host part of uri is "hostname", then "Strip User
> Parameters" checkbox WORKS! Here is a fragment from registrar log:
> "2011-01-28T08:05:58.650114Z":777:SIP:DEBUG:beaver.sip.nstel.ru:SipRedirectServer-13:B63EFB90:SipRegistrar:"SipRedirectServer::processRedirect
> Starting to process request URI
> 'sip:3892;phone-context=cdp@beaver.sip.nstel.ru'"
> "2011-01-28T08:05:58.650142Z":778:SIP:INFO:beaver.sip.nstel.ru:SipRedirectServer-13:B63EFB90:SipRegistrar:"[090-USERPARAM]
> SipRedirectorUserParam::lookUp stripped parameters from
> '3892;phone-context=cdp.udp' -> '3892'"
> But beaver.sip.nstel.ru is the hostname (configured as sip domain alias),
> while sip domain is sip.nstel.ru.
> But when the call "is coming to sip domain" 'Strip User Parameters' checkbox
> does not work. Here is another fragment from registrar log:
> "2011-01-28T07:42:00.733352Z":13325:SIP:DEBUG:beaver.sip.nstel.ru:SipRedirectServer-13:B62EFB90:SipRegistrar:"SipRedirectServer::processRedirect
> Starting to process request URI
> 'sip:3892;phone-context=cdp@sip.nstel.ru'"
> "2011-01-28T07:42:00.733377Z":13326:SIP:DEBUG:beaver.sip.nstel.ru:SipRedirectServer-13:B62EFB90:SipRegistrar:"[090-USERPARAM]
> SipRedirectorUserParam::lookUp 'sip:3892;phone-context=cdp@sip.nstel.ru'
> not in my domain - not modified"
> Is it a bug or is it expected behaviour?
> Should i create a jira for it?

seems like a bug to me. y, i would create a jira
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Re: [sipx-users] auto-attendant messages are played real fast

[Douglas Hubler]

On Thu, Jan 27, 2011 at 10:16 PM, srinivasa rao  wrote:
> 2) Afterthis, I called bandwidth again; bandwidth asked me to change a
> value of  in the
> freeswitch/conf/sip_profiles/internal.xml file. Therefore, I rebooted the
> server. Also, in the management interface, I clicked on the server and click
> on send profiles as well. This did not fix the problem either.

if you change files in  /etc/sipxpbx/freeswitch/conf then send
profiles, then sipXconfig will over wright your changes.  If you want
to make more permanent changes, you should change the templates in
/etc/sipxpbx/freeswitch then send profiles.
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Re: [sipx-users] ERR in sipregistrar log

[Douglas Hubler]

On Thu, Jan 27, 2011 at 10:44 AM, Worley, Dale R (Dale)
 wrote:
> In regard to the message "outgoing call 1", I believe that is a debugging 
> message that was incorrectly set to log at "ERR" level rather than "DEBUG" 
> level.  A programmer could probably look at the code and determine if that is 
> true very quickly.  (The message will be in 
> sipXtackLib/src/net/SipUserAgent.cpp, search for the string "outgoing call 
> 1".)  There is probably an "outgoing call 2" message somewhere also...

I took care of this, thanks Dale, it has been bothering me.
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Re: [sipx-users] openacd - lazy question

[Douglas Hubler]

great question.  short answer is it must be INVITE, not REFER as it
stays in the media path.

Looking for a REFER based connection to agent would allow for some
insane scalability as OpenACD really only consumes a fraction of the
CPU to do it's job.   OpenACD today uses the FS session to determine
when agents are available based on open channels and such.  Having
said that, OpenACD has a general architecture for queueing anything
like email and voicemail, so i suspect queuing based on REFER would
require tracking/subscribing to the SIP messages to determine call
state to replicate what FS session provides.

but post to
  http://groups.google.com/group/openacd
if you want a more in-depth response

On Tue, Jan 25, 2011 at 9:32 AM, Tony Graziano
 wrote:
> as an incoming call center the call would go to the agent via an invite.
> i am assuming it will bridge the call via its FS config, and there is no
> transfer.
> though i might be wrong and will wait to be corrected...
>
> On Tue, Jan 25, 2011 at 9:28 AM, Matt White 
> wrote:
>>
>> Sorry for being too lazy to test this for myself.  I was just talking with
>> someone and the question came up.
>>
>> Does the new openacd implementation use the same conference method to join
>> the caller to the agent the current ACD uses or does it use a standard
>> refer?
>>
>> -M
>>
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>
>
>
>
>
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Re: [sipx-users] Issue with autoattendant and remote workers (authrules.xml).

[Douglas Hubler]

So glad your checking out ACD, thanks! comments in-line

On Mon, Jan 24, 2011 at 1:16 PM,   wrote:
> I'm running the 4.4.0-53 build from the SIP Foundry site, and it seems that
> the autoattendant gets broken for remote workers when adding dial plans to
> support OpenACD and my SIP trunk to the PSTN.  It looks like calls to the
> autoattendant for either voicemail or the custom IVR I'm putting in front of
> my ACD queues are hitting the catch all reject rule in authrules.xml, and
> near as I can tell there isn't a way to manage this file through
> adding/modifying dial plans in the sipXconfig GUI so that it works.  This
> also affects calls from outside to the DID for the IVR front end to the ACD
> queues in the same manner.

You shouldn't have to change dial plans what-so-ever to get calls to
ACD.  There are lines in "Call Center" configuration pages that let
you define a number.

If however you want ACD lines to be available from your IVR, I did not
try this, but using the extension in the AA menu should work.

NOTE: The ACD implementation decided to register all the numbers as
aliases instead of dial plans and that's why you would not see any
reference to the ACD lines in the generated dial plans XML. We could
have done either but aliases was far easier to implement and we didn't
see a good reason to use dial plans. At least with aliases, you don't
have to restart proxy.

> The default rule seems to be automatically generated out of code in class
> AuthRules, containing 's for the FreeSWITCH unmanaged gateway
> and my SIP trunk, and would clobber any manual changes to this file.  To
> verify that the issue is indeed with the default, catch all reject rule, I
> commented out the call to generateNoAccess in AuthRules.end() which
> generates the block of XML in question, recompiled and replaced the
> sipxconfig jar, and the autoattendant works again for remote workers and the
> DID for my IVR.  Obviously, this workaround isn't ideal as it might open
> some security holes.
>
> Is there a way to customize my dial plans to allow these calls through the
> System -> Dial Plans section of the sipXconfig web GUI?  It seems that any
> rules generated in authrules.xml are simply based on the gateway(s) the dial
> plan is associated with, and there is no way to get any finer granularity
> here.  Is there something obvious I'm missing?

If you still need to do this, despite my answer above, it's possible
following the instructions on this wiki page where you setup an
unmanaged gateway to dial back your ACD lines would work

http://wiki.sipfoundry.org/display/sipXecs/OpenACD+Setup+for+sipXecs+4.2

One reason might be to add a time schedule to your ACD, but that is
the only reason I can think of at the moment.
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Re: [sipx-users] LDAP attribute change

[Douglas Hubler]

On Mon, Jan 24, 2011 at 10:17 AM, m...@grounded.net  wrote:
> Looks like a git file, looks like I run something like 'git apply patchname' 
> but don't want to break my system so can't run it til I know more.
>
> I did manage to find a little information but it's like taking on a whole new 
> technology, that of trying to learn how to use git and a patching environment 
> :). The 'click here' links don't work so I'm not sure if I'm missing 
> something simpler that just tells me how to run a patch.
> Could it be as simple as running it?

4.2.1 using old method of build RPMs where you need to have sudo
access, but the approach is the same.
 http://wiki.sipfoundry.org/display/sipXecs/Building+RPMs

Because your fix is in Java, it needs to be compiled so you cannot
just "run a patch".
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Re: [sipx-users] HA - remote workers - public port

[Douglas Hubler]

On Mon, Jan 24, 2011 at 5:51 AM, Tony Graziano
 wrote:
> it sounds like an internal dns issue. I "think" there is a JIRA on this.
>
> How long before the local phones stopped being able to register, 30 minutes?

I found the DNS advisor found in the System menu fairly useful to check DNS
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Re: [sipx-users] authorization code feature

[Douglas Hubler]

On Fri, Jan 21, 2011 at 6:32 AM, Rama Krishnam Raju Pakalapati  wrote:

> Thanks for the information, i browsed the wiki pages you have referred and
> it points to the
> http://article.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/29381
> link and explains the cases tested.
>
> I have some more questions to ask regarding the authorization code
> implementation..
>
> 1) Whether the design is completed, because i could not see it in the main
> repository ?
>

Here are some links
https://github.com/dhubler/sipxecs/tree/master-4.2/sipXacccode
https://github.com/dhubler/sipxecs/tree/master-4.2/sipXconfig/neoconf/src/org/sipfoundry/sipxconfig/acccode
https://github.com/dhubler/sipxecs/blob/master-4.2/sipXconfig/web/src/org/sipfoundry/sipxconfig/site/admin/AuthCodesPage.java
https://github.com/dhubler/sipxecs/blob/master-4.2/sipXconfig/web/context/WEB-INF/admin/AuthCodes.html



> 2) I would be interested to extended this functionality to even for the o
> outgoing calls, some one on group working on it..??
>

last one to work on it was from avaya, so no active sipxecs developers ATM.
 It is for outgoing calls already as far as I understand.



> 3) where can i get the latest source code with this implementation ?
>

see #2


> 4) Autho code and Account code are confusing little bit on the forum
> discussions, i hope both are pointing to the same in our case.
>

I only know of the one code a user needs to enter.
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Re: [sipx-users] Call forwarding at phone

[Douglas Hubler]

On Thu, Jan 20, 2011 at 5:28 AM, Henry Dogger  wrote:
> Hm, I tried making a dummy user for the mediant, which has permissions
> to call everything.
> The mediant registered using this dummy user, I can see it is registered
> in the registrations list.
> But somehow my calls keep getting into voicemail.
> Case:
> 401 (aastra) has call forward to a mobile number (00612345678) set on
> the phone itself.
> When I call from outside (or internal) I keep getting in the voicemail.
> The only way to get transferred is to give the internal user permissions
> to call mobile numbers. But outside callers still get send to
> voicemail...
> Some help would be greatly appreciated, since are client really wants to
> get this to work...

FYI: Here's some advice i gave on IRC

(12:23:12 PM) lazyboy: effectively you want to give someone permission
to make an outbound call without outbound permissions, i recall this
being a very old debate
(12:24:01 PM) lazyboy: and from what i recall (not my opinion either
way), once the system starts making exceptions, it opens the door to
call fraud
(12:25:01 PM) lazyboy: everything is possible in software, so, of
course modifying source will address your issue, but you really don't
want to go there
(12:26:50 PM) lazyboy: I suspect you can do some fairly easy
freeswitch dialplan config to bridge a call for you
(12:27:33 PM) lazyboy: once you work out the FS xml,  i can tell you
where you can stick it
(12:28:28 PM) lazyboy: /etc/sipxpbx/freeswitch/default_context.xml.vm
(12:28:48 PM) lazyboy: there are examples on how to do some FS stuff
w/sipxecs on the wiki
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Re: [sipx-users] 4.4.0 - sipxproxy failed

[Douglas Hubler]

As a workaround, If you disable ACD as a role in the server
configuration menu option, then that  process shouldn't start.

On Thu, Jan 20, 2011 at 8:09 AM, Nikolay Kondratyev  wrote:
> Hi, all.
> I started to play with 4.4.0.
> Today I upgraded my test system from 4.2.1 to 4.4.0 (4.4.0-
> 2011-01-18EST13:19:19 swift).
> And now sipxproxy process fails (sipxproc shows "failed" for sipxproxy).
>
> I tried to analize the problem:
> I found the following in the sipXproxy log:
> "2011-01-20T12:42:02.166892Z":39:KERNEL:ERR:sipx4.lab.nstel.ru:pid-17831:B7FC26E0:SipXProxy:"OsServerSocket:
> bind to port 172.23.12.104:5061 failed with error: 98 = 0x62"
> Ok, lets see who's there:
> [root@sipx4 sipxpbx]# netstat -nlp | grep 5061
> tcp    0  0 172.23.12.104:5061  0.0.0.0:*
> LISTEN  17082/sipxacd
> Then i tried to turn ACD off - unchecked ACD on the "server roles"
> configuration page.
> After that, sipXproxy works. At least sipxproc shows "running".
>
> If i turn on ACD on the "server roles" page, the problem comes back.
>
> So... my idea is that acd configuration is wrong in 4.4.0...
>
> The whole snapshot is available at
> ftp://sipx:s...@ftp.nstel.ru/sipxproxy-failed-sipx-configuration-sipx4.lab.nstel.ru.tar.gz
>
> Is my analisys correct?
> Is it known problem?
>
> At the moment i don't need ACD, so i can continue without it... but i think
> the problem is serious...
> Should i raise a jira?
> Is it possible to fix the problem in this release (4.4.0-46.g01752)? Or
> shoud i just wait a while?
>
> Thanks and reagrds,
> Nikolay.
>
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Re: [sipx-users] Upgrade.

[Douglas Hubler]

Technically upgrades from dev releases are not supported but if you get past
yum update and system runs you should be good going forward
On Jan 19, 2011 6:37 PM, "Rolland Hart"  wrote:
> Can this version of Sipx (sipXconfig (0.4.4-5b9dc0c 2010-12-01T21:31:52
> build32)) be upgraded by changing the repo to the lastest 4.4 beta without
> problems or should this not be attempted?
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Re: [sipx-users] authorization code feature

[Douglas Hubler]

On Wed, Jan 19, 2011 at 9:14 AM, Rama Krishnam Raju Pakalapati
 wrote:
> I hope the implementation of account code xx-4824 mentions that, the
> authorization code feature was designed for Outgoing calls and the use case
> is similar to the case what we are testing..
>
> It should also work for Outgoing calls, because this is basic use case for
> testing the authorization code feature!
>
> Can we get the wiki or doc, explaining about the behavior of authorization
> code feature ?

Placeholder w/link to ML that explains a little more
 http://wiki.sipfoundry.org/display/sipXecs/Authorization+Codes

If you need wiki access, please email me off-list w/your user id.
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Re: [sipx-users] Blank VM when forwarding a second time from handset

[Douglas Hubler]

On Tue, Jan 18, 2011 at 1:53 PM, Matthew Kitchin (public/usenet)
 wrote:
> I finally have a significant new piece of info on this. The original
> issue is below. We were having trouble reproducing it. Today, we
> discovered it happens 100% of the time when a VM is forwarded a second time.
> Call in -> User A VM - forward to -> User B VM - forward to - > User C VM
> When we try the above scenario, user C gets the blank message described
> below every time. They claim this sometimes happens on messages that are
> forwarded a single time, but I still need to see proof of that.
> I still do not know how to turn on "debug logging in sipXivr". Nothing
> sticks out at my in the sipxivr log file. I see it creating an xml file
> for the 3rd destination mailbox.
> I wouldn't have thought this type of VM usage would be so common, but
> apparently it is at my company.
> I will be happy to open a bug report if that is what is appropriate now.
> I wanted to check first to see if anyone else has seen this.

Nice job find out how to reproduce this, definitely open a bug.

re:logs
I looked in some code
  
https://github.com/dhubler/sipxecs/tree/master-4.2/sipXivr/src/main/java/org/sipfoundry/voicemail
and i didn't see any calls to log info in CopyMessage.java so it may
just be that it's not seen by looking the logs.
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Re: [sipx-users] Dial by Name Issues

[Douglas Hubler]

On Tue, Jan 18, 2011 at 7:54 PM, Richard Bruce
 wrote:
> Can someone direct me to where the dial by name feature is
> controlled?

https://github.com/dhubler/sipxecs/blob/master-4.2/sipXivr/src/main/java/org/sipfoundry/sipxivr/DialByName.java

and accompanying files.
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Re: [sipx-users] sipXecs 4.4 beta tomorrow

[Douglas Hubler]

On Tue, Jan 18, 2011 at 5:49 AM, Michal Bielicki
 wrote:
> If you wait another week I'll throw in a fresh mod_smp in freeswitch ;)

Good news is freeswitch is independent now (no signifigant
modification made for sipxecs) so folks should be able to update
freeswitch post sipxecs release, the way it should be.

What is mod_smp?

speaking of freeswitch, I built the latest freeswitch as of friday in
the sipxecs 0.0.4.5.2 build.  There were some 600 checkings since we
last updated.sipxecs 4.4 beta still has the older freeswitch
version.  Freeswitch announced 1.0.7 "release", so I feel we should
switch to the sanctioned version. Anyone object?
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Re: [sipx-users] Voicemail IVR Voice Stutters During Playback

[Douglas Hubler]

On Mon, Jan 17, 2011 at 11:24 PM, Tim Ingalls  wrote:
> I'm running BootROM 4.3.0.0246 (which is Rev. B) and SIP 3.2.1.0054. Do I
> need to roll back the BootROM to 4.2.2?
>
> I tried loading 4.2.2 into sipX-config (Devices >> Device files) and
> rebooting the phones, but the same BootROM (4.3.0) is still coming up when I
> go to the phone's status page. What's the correct way to roll back the
> firmware?

Bootrom is notoriously one-way street on upgrades, I'm not sure the
specifics of 4.3.0 though.  Luckily, it's almost never the problem.

> By the way, when you say "firmware" are you referring to the sip.ld
> application or the BootROM?

sip.ld is the "application" firmware.
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[sipx-users] sipXecs 4.4 beta tomorrow

[Douglas Hubler]

I want to do a full build sometime tomorrow and run a smoke test. Then
we can declare beta unless anyone has any concerns on what they've
seen so far.
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Re: [sipx-users] Dial by Name Issues

[Douglas Hubler]

On Sat, Jan 15, 2011 at 8:50 PM, Richard Bruce
 wrote:
>
> Content-Type: text/plain;
>  charset="utf-8"
> Content-Transfer-Encoding: 8bit
> Organization: SipXecs Forum
> In-Reply-To: 
> X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <55999>
> Message-ID: 
>
>
>
> I have been researching this and found the dial by name vmxl
> file in usr/share/www/doc/aa_vmxl.  I noticed the prompt
> requiring 1 digit for minimum digits.  I changed this to 3
> but it had no effect.  I don't believe I am looking at the
> correct dial by name vmxl.  Is there another copy that is
> the active vmxl?

vxml is not even used in 4.2.1 anymore.  That should be cleaned up and
removed from release. Can you create an issue?
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Re: [sipx-users] LDAP attribute change

[Douglas Hubler]

Why can't you just ignore the user group?
On Jan 16, 2011 9:49 AM, "m...@grounded.net"  wrote:
> Are there any other places to look for information about the LDAP
functions? I've spent countless hours searching and can't find the answers
to what I'm trying to achieve. I've been using google, searching the wiki,
other sites,etc.
>
> I'm trying to set up a sipx box so that its user accounts are mainly on an
openLDAP server.
>
> This means being able to map the right attributes along with being able to
create groups/permissions that disable and enable functions.
>
> One small problem is that when users are imported using LDAP import, they
are given a group ldap_imports in addition to any I am using from my LDAP
server. Is there any way of disabling this because I am pulling in my own
group names to put users into.
>
> Mike
>
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Re: [sipx-users] sipxconfig spa942 plugin: 'cfwd busy dest' parameter default value.

[Douglas Hubler]

On Fri, Jan 14, 2011 at 4:47 AM, Nikolay Kondratyev  wrote:
> Hi all,
> i found that by default sipxconfig plugin sets "Cfwd Busy Dest" incorrectly.
> By default the value is set to 'vm'. This leads to incorrect behaviour when
> DND mode is activated on spa942 phone.
> The thing is that, when DND is set, spa942 replies with "302 moved
> temporarily" with "Cfwd Busy Dest" in Contact field.
> So, sipx can not find number 'vm'. When the parameter is set to
> ~~vm~ (in my case ~~vm~3853) DND mode works as expected, the call is
> forwarded to the users voicemail.
> I think the default value should be changed...
> And in addition to that, Cfwd No Ans Dest parameter may be set to the same
> value too...
> Should i create a jira to track it?

unless a spa942 user comes forward and objects, i would say sure.
Defaults may be accessible in the settings file (phone.xml and
line.xml) so check it out .
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Re: [sipx-users] Intermittent Handset Issues with Polycom 335s

[Douglas Hubler]

On Thu, Jan 13, 2011 at 9:46 AM, Jeff Ferrara  wrote:
> I'm really hoping we end up with support for 3.3.x sooner rather than
> later... Im, guessing this won't make 4.4 - Is it planned for 4.6?

It's hard to believe it wouldn't be in 4.6.

I'd also like to propose developers spend the time to separate polycom
phones support (as well as other device support) into a separate sipx
project so backporting is a possibility going forward. e.g. polycom
5.0 support could be added post sipxecs 4.6 release without taking all
of sipxconfig with you should the need/demand arise even if it's
"unofficial" support.
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Re: [sipx-users] Media Services won't start

[Douglas Hubler]

On Thu, Jan 13, 2011 at 9:17 AM, Ben Goodfellow wrote:

> After playing around with a file 
> ('/etc/sipxpbx/freeswitch/conf/dialplan/sipX_context.xml')
> I managed to stop Media Services from starting, in the details it suggested
> running
>
>
>
> Freeswitch.sh –configtest
>
>
>
> Which I did, after running that command it appears 3 files have been
> deleted
>
>
>
>- Missing resource: file '/etc/sipxpbx/freeswitch/conf/freeswitch.xml'
>- Missing resource: file
>'/etc/sipxpbx/freeswitch/conf/dialplan/sipX_context.xml'
>- Missing resource: file
>'/etc/sipxpbx/freeswitch/conf/sip_profiles/sipX_profile.xml'
>
>
Config test shouldn't make the system unusable after, in fact a test should
make any changes to the system IMHO.  This is a bug.  Please create a bug
and reference this thread in the bug number. URL is
  http://article.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/31935

If sending profiles to server didn't help, try hitting the apply button on
media services page.


As a little background for you or others because this can be a little
confusing

Files in the directory
 /etc/sipxpbx/freeswitch
are template files and are not used until profiles are sent to server or
the apply/OK button is hit on the media services page

Files in the directory, or are a descendant of the directory
 /etc/sipxpbx/freeswitch/conf
Are transient files and if you edit them directly to test something, you
need to keep in mind 2 things:
1.) the file permissions the same otherwise freeswitch --conftest complains
2.) The instant profiles are sent to the server or the apply/OK button is
hit on the media services page, all your changes are gone.  To keep your
changes, you need to put them in respective template files in
/etc/sipxpbx/freeswitch
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Re: [sipx-users] Suspicious calls

[Douglas Hubler]

also check the freeswitch log,

On Wed, Jan 12, 2011 at 9:20 AM, Huw Jones  wrote:
> Thank you (and Tony) for the suggestions. I'll investigate as per your advice.
>
>
> Cheers
>
>
> Huw
>
>
>
 "Nikolay Kondratyev"  01/12/11 11:08 AM >>>
> Somebody (something :) ) from your local lan might just sent invites
> directly to AC.
> Do you collect syslog or cdr from your AC? (that would be for certain).
> "IP to tel calls count" on the AC may be interesting...
> Regards,
> Nikolay.
>
>> -Original Message-
>> From: sipx-users-boun...@list.sipfoundry.org
>> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Huw Jones
>> Sent: Wednesday, January 12, 2011 1:38 PM
>> To: sipx-users@list.sipfoundry.org
>> Subject: [sipx-users] Suspicious calls
>>
>> Hi folks
>>
>> We've had a strange situation in our college. Over the
>> Christmas holiday one of our sites is reported (by BT) to
>> have made thousands of 'suspicious' calls to Sierra Leone!!
>> I've checked the Call Detail Records on the web interface and
>> there's no sign of these calls, I've also searched through
>> the contents of /var/log/sipxpbx/ and there's no sign of the
>> numbers there either.
>>
>> Our SipX server is not visible from outside our own network
>> and it only interfaces with the outside world via three
>> Audiocodes gateways connected to ISDN. If the calls were made
>> maliciously I'm at a loss how they did it. :-(
>>
>> Can anyone suggest where I might look for further
>> information? I'm pretty dubious that these calls really have
>> come from our system but I'd like to be certain!! I'd very
>> much appreciate any advice or suggestions.
>>
>> Happy New Year to you all
>>
>> Huw
>>
>>
>> ***
>> Mae'r e-bost yma ac unrhyw ffeiliau a drosglwyddir oddi fewn
>> iddo yn gyfrinachol, a bwriedir ef ar gyfer yr unigolyn neu'r
>> endidau mae wedi ei gyfeirio ato'n unig. Os ydych wedi derbyn
>> yr e-bost yma trwy gamgymeriad hysbyswch y rheolwr system os
>> gwelwch yn dda.
>>
>> Mae'r troednodyn yma hefyd yn cadarnhau bod y neges e-bost
>> yma wedi cael ei wirio gan MIMEsweeper am unrhyw feirysau
>> cyfrifiadurol oedd yn bodoli.
>> www.mimesweeper.com
>> ***
>> This email and any files transmitted with it are confidential
>> and intended solely for the use of the individual or entity
>> to whom they are addressed. If you have received this email
>> in error please notify the system manager.
>>
>> This footnote also confirms that this email message has been
>> swept by MIMEsweeper for the presence of computer viruses.
>>
>> www.mimesweeper.com
>> ***
>>
>>
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Re: [sipx-users] Message receipt time not correct in TUI

[Douglas Hubler]

On Mon, Jan 10, 2011 at 6:47 PM, Bob  wrote:
>
>>which date is wrong for you and what should it be? what is your timezone?
>
> It's not the date, but the time of the call as played in the TUI. The time
> displayed on the web interface is correct for the messages.
>
>>If I recall correctly,  the timezone recording in the message digest is
> fixed at EST for all systems no matter the
>>timezone on the OS.
>>This was a bug that I believe was purposely ported to sipxivr to avoid
> migrating existing voicemail digest files
>>timestamps in files to GMT.
>>As long as each server is adjusting, it shouldn't matter, but maybe
> something isn't adjusting.
>
>
> Server TZ is EST. Server HW clock is EST.

I looked at the code,

  sipXivr/src/main/java/org/sipfoundry/voicemail/MessageDescriptor.java

and I was incorrect, it _does_ use the systems timezone, but fixes the
_locale_ to English.  So I checked a production system in the eastern
timezone and timestamps are stored in native timezone

 
...
  Tue, 28-Sep-2010 05:41:12 PM EDT

so for some reason java is not in sync with the timezone the OS is on.

I wrote a small test program, let me know what this returns on the system.

To run upload class file to system and in same directory as the class
file, run command:
  java TestTime


TimeTest.class
Description: Binary data
import java.text.*;
import java.util.*;
public class TimeTest {
private static String DATE_FORMAT = "EEE, d-MMM- hh:mm:ss aaa z";
public static void main(String[] x) {
	SimpleDateFormat dateFormat = new SimpleDateFormat(DATE_FORMAT, Locale.ENGLISH);
	System.out.println(dateFormat.format(new Date()));
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Re: [sipx-users] Message receipt time not correct in TUI

[Douglas Hubler]

On Sat, Jan 8, 2011 at 2:15 PM, Bob  wrote:
> I'm running 4.2.1 under RHEL. Basically installed it via YUM, basic
> configuration options, etc. The problem is this:
>
> If you listen to a voicemail received at 10:00 AM Eastern time, if you play
> the message info in the TUI, it says the message was received at 3:00 PM.
> The time zone for the system is at default - Eastern. The message time as
> displayed in the web interface is correct - 10:00 AM. The .xml file
> associated with the specific message contains 15:00 GMT, so I understand why
> it's saying the time wrong. There seems to be some disconnect between the
> web interface and the subsystem that writes the messages and their
> associated info to disk. One other thing... The server time according to the
> date command is Eastern, also.
>
> Any ideas or anyone seen this before?

which date is wrong for you and what should it be? what is your timezone?

If I recall correctly,  the timezone recording in the message digest
is fixed at EST for all systems no matter the timezone on the OS.
This was a bug that I believe was purposely ported to sipxivr to avoid
migrating existing voicemail digest files timestamps in files to GMT.
As long as each server is adjusting, it shouldn't matter, but maybe
something isn't adjusting.
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Re: [sipx-users] Dimdim bought by Salesforce.com

[Douglas Hubler]

On Mon, Jan 10, 2011 at 10:48 AM, Geoff Van Brunt
 wrote:
> I agree. An "install all and disable unneeded" is bad from a security
> perspective as well. As we all know SIP products are in the crosshairs
> of hackers everywhere with recent automated attacks. This is only going
> to get worse. The more that is installed, the more attack surface there
> is.
>
> I also would like to see "lightweight" integration as you mentioned in
> your blog. In the case of openfire there are a lot of features/modules
> we would love to use as a company, but it's too cumbersome to use
> "hidden" features using the integrated version. If not using the
> integrated version, then we lose too much. Essentially we can only use
> what is available out of the box...
>
> My votes for vNext are:
>
> - VMail redundancy in HA config
> - More Branch Support for features, esp. dial plans
> - More modularized so there is less dependencys. Essentially move to an
> install as needed model. Also allow direct control over integrated
> products such as openfire. The modularized model should also make adding
> lightweight integrations easier allowing sipx to work with more
> products. It should also make it easier for "third parties" to write
> integrations with the product. Is this going to be hard? Yes. Is there
> going to be a lot of secure information going back and forth between
> sipx and some third party products (openfire). Yes. I don't think either
> reason is a showstopper though and the benefits are just too good to
> overlook.

100% in agreement with all your points and suggestions.

re:secure info
this spec doesn't solve everything, but this could go a long way to
remove as much of the security burden from each module if they rely on
web services for integration
http://wiki.sipfoundry.org/display/sipXecs/Web+Service+API+Unification

re:The modularized model
I am inspired by how the drupal project manages this.  You submit your
project name and within minutes, you get a CVS repos and after you
upload your module is there for all to use at their discretion. It
could be as simple shell script that shows diagnostics to full blown
SIP server.  The one thing i'd change (beside using git instead of
CVS) is the technical difference between core v.s. non-core (but it's
a minor point).

re:vmail HA config
ezuce's plan is to introduce mongodb both with it's distributed
database capabilities and its gridfs support will take us there.
we'll pay attention to making these distributed capabilities to all
applications based on freeswitch. For example it might mean building
mod_mongo as alternative to sqlite/postgres native transparent service
distribution.  ezuce team is well aware we need to fully understand
and test things like asynchronous disk flushes and data loss v.s.
performance.  I went to mongodb conference and from what i understand,
flexibility is alway in user's control.
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Re: [sipx-users] sipXecs 4.4 ready for testing

[Douglas Hubler]

I will upload them tonight. I noticed build didn't include them and I meant
to upload them
On Jan 8, 2011 7:37 AM, "Matt White"  wrote:
>>>> Douglas Hubler  01/07/11 11:58 PM >>>
>>>There are still outstanding bugs
>>>
> >>http://bit.ly/gCCsCX
>>>
>>>CentOS repo file
>
> Are the srpm's available? I'm using srpms from about 2 weeks ago for the
SuSe builds and I'd like to update them.
>
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[sipx-users] sipXecs 4.4 ready for testing

[Douglas Hubler]

There are still outstanding bugs

  http://bit.ly/gCCsCX

CentOS repo file
 http://download.sipfoundry.org/pub/sipXecs/sipx-4.4.0-centos.repo
ISO (look for file labeled 4.4.0)
  http://download.sipfoundry.org/pub/sipXecs/ISO/

(ignore openacd ones, they are slated for next release) but we've done
our best to squash the blockers.
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Re: [sipx-users] Phone groups not picking up digit map in latest 4.2.1

[Douglas Hubler]

On Fri, Jan 7, 2011 at 12:34 AM, Roy Walker  wrote:
> Hate to dig up an old thread, but it looks like I ran into
> this as well... strange thing was that the digit map was in
> the config, but the phone did not use it (Polycom 650
> running 3.2.3).  I manually put the digit map into the UI on
> each phone to get it working, so I know the digit map was
> valid.  Possible it is a problem in the firmware...

reset all settings the phone itself completely.  Polycom phones keep
settings you set on them manually (thru their polycom phone web ui or
thru polycom phone menul) like an elephant.
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Re: [sipx-users] Dimdim bought by Salesforce.com

[Douglas Hubler]

On Thu, Jan 6, 2011 at 7:34 PM, m...@grounded.net  wrote:
>> Bloating? I don't consider sipx bloated, but everyone has an opinion (me
>> too).
>
> Not saying it is bloated, saying not to let it go that way.

it's all about perspective.  I think sipxecs is on the heavy side, but
i wouldn't suggest stopping anytime soon, I just like to see a
different approach
  http://bit.ly/fvcdQB
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Re: [sipx-users] How to disable Call Forwarding?

[Douglas Hubler]

On Thu, Jan 6, 2011 at 4:39 PM, Nathaniel Watkins
 wrote:
> I've not had the pleasure of using tomcat/java servlet at all - but I'm 
> assuming that if one knew where to find the servlets for the page, you could 
> modify the code to check to see if that user is part of a specific group - if 
> not - don't display the tab?
>
> This is total speculation on my part...

that's about right, there is a fairly robust permission layer, once
you commit to opening the sipxconfig java developer can.
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Re: [sipx-users] auto-attendant messages are played real fast

[Douglas Hubler]

On Wed, Jan 5, 2011 at 4:40 PM, srinivasa rao  wrote:
> We have auto attendants setup in our sipx setup. For the last few weeks, these
> auto attendant messages are played very very fast. For example, our main 
> message
>
> used to take 40 seconds to complete it, and it takes very less time to 
> complete
> it.
>
> However, when I play the auto attendant messages manually in the sipx
> administration page, the auto attendent messages are played correctly.
>
> This is a non profit agency, and I would really appreciate your feed back in
> resolving the issue.

Run this command on the *.wav files
  file myfile.wav

Example:

file tryagain.wav
  tryagain.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM,
16 bit, mono 8000 Hz


it will show you the format details.  Check the bitrate and encoding
on a file that come with sipxecs.
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[sipx-users] FYI: looking to shut down old wiki

[Douglas Hubler]

effectively gone.   http://sipx-wiki.calivia.com.  We can put in
global redirect to new wiki wiki.sipfoundry.org.
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Re: [sipx-users] Per-process fails

[Douglas Hubler]

On Wed, Jan 5, 2011 at 9:15 AM, Henry Dogger  wrote:
> A client of ours is having problems with the sipXecs config ui, it is very
> instable and gives a lot of internal server errors.

I would review the /var/log/sipxpbx/sipxconfig.log for meaningful
error messages.  If you cannot determine what the issue is, post error
back to the list.
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Re: [sipx-users] Caller-id settings not working

[Douglas Hubler]

On Wed, Jan 5, 2011 at 12:18 AM, Roy Walker  wrote:
> Paul,
>
> I wish it was a configuration issue (I am crystal clear on how Caller ID is
> supposed to work in sipXecs now).  I have tried EVERY possible configuration
> I can think of.  I have deleted the Gateway, added it back... I can not get
> the transform or user specific Caller ID to work, it always just spits out
> the 4 digit extension.  Can one of the devs help me debug this so I can
> submit a bug report?

I would start with submitting bug with file merged.xml captured by
following this doc
  
http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+Sipviewer

In bug report, report only the simplest thing that is not working.
i.e. if you think user's caller id simply isn't working, just mention
that.  Do not mention anything about your ultimate goals such as
number or display name transformation.

Also, attach screenshots of all relevant UI w/advanced settings shown.
 If you want to send me offlist, results of
 sipx-backup --configuration
that is fine too.
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Re: [sipx-users] 4.4 release delay

[Douglas Hubler]

On Tue, Jan 4, 2011 at 11:07 AM, Douglas Hubler  wrote:
> On Tue, Jan 4, 2011 at 8:43 AM, Matt White  wrote:
>>>>> Douglas Hubler  12/30/10 1:55 PM >>>
>>>>>Building using mock is currently on 4.5 codebase so instead of
>>>>>backporting it, I recommend we cut the new 4.4 release from 4.5 HEAD
>>>>>and begin to stabilize this branch.
>>
>> I second the recommendation to stabilize 4.4 without mock for now.
>>
>> We are close to getting SuSe builds to compelete and it will be easier to
>> get them out with 4.4 then retooling for mock right now.
>
> I just pushed changes incl. mock to 4.4 release branch in git.   I
> assume you've followed my notes about OBS being unusable in the end.
> Are there mock templates for suse, I cannot seem to find them. Cross
> distro compiling has been a godsend.

Found some old one's here, may be worth a try to update them, what do you think?
 https://build.opensuse.org/package/files?package=mock&project=home:repabuild

Also, for suse, we can upload srpms to OBS and official suse builds
can still come from there, never hitting mock.
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Re: [sipx-users] 4.4 release delay

[Douglas Hubler]

On Tue, Jan 4, 2011 at 8:43 AM, Matt White  wrote:
>>>> Douglas Hubler  12/30/10 1:55 PM >>>
>>>>Building using mock is currently on 4.5 codebase so instead of
>>>>backporting it, I recommend we cut the new 4.4 release from 4.5 HEAD
>>>>and begin to stabilize this branch.
>
> I second the recommendation to stabilize 4.4 without mock for now.
>
> We are close to getting SuSe builds to compelete and it will be easier to
> get them out with 4.4 then retooling for mock right now.

I just pushed changes incl. mock to 4.4 release branch in git.   I
assume you've followed my notes about OBS being unusable in the end.
Are there mock templates for suse, I cannot seem to find them. Cross
distro compiling has been a godsend.
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[sipx-users] 4.4 release delay

[Douglas Hubler]

There is a bit of a delay on 4.4 for at least 2 of the following reasons:

 1.) Switch from using OBS to mock for building rpms
 2.) FreeSWITCH in a "bridge" setup address call transfer to AA/VM but
opened a slew of other issues ezuce it still working on

ezuce developers were working on these two issues in last couple
months in addition to integration with openacd for 4.5

Building using mock is currently on 4.5 codebase so instead of
backporting it, I recommend we cut the new 4.4 release from 4.5 HEAD
and begin to stabilize this branch.  Tony G. and Michal B. have been
diligent about testing 4.5 and surfaced some blockers that will get
addressed.  The good news is that sipXecs will include much improved
openacd integration. No where near complete, but far easier than it
was before.  Openacd support itself should not have introduced any
significant instabilities and that is why i recommend we cut a new
branch.  It did introduce a number of dependencies : erlang, mongo,
boost, but those only get installed if you explicitly install openacd
post sipxecs installation.  For folks that build their own spin of
sipXecs, we can make building of these dependencies optional too if
you wish.
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Re: [sipx-users] Cannot see cdrs

[Douglas Hubler]

On Tue, Dec 28, 2010 at 9:15 AM, Eda Ercan  wrote:
> The owner of sipx installation directory is a user other than sipxchange, it
> is the user that makes the build process, but everyone has the rights to
> read/write (777) - does it matter?

Couple things
1.) CDR report processing is not part of default source based
installs. In order to build and install it automatically, you need to
add these switches to your call to configure script before building

 --enable-reports --enable-agent

2.) When you install from source, some things are not setup for you
automatically such as crons.  There is a cron that processes the raw
call events into actual calls. I always forgot what this is, so I went
onto a running system and typed

 find /etc/cron*

where I saw

  /etc/cron.d/sipxconfig-report-crontab

So you need to run this manually to process all the call events into
call details.   The path will be different for source installs, if you
cannot find it, let me know.
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Re: [sipx-users] SIPXREST and voice codecs.

[Douglas Hubler]

On Thu, Dec 23, 2010 at 12:03 PM, Alexander Shvaryov  wrote:
>
> Apologies for the typo, I meant " ... other than the 711U"?

I'm not 100% positive, but i think the INVITE is generated here.

https://github.com/dhubler/sipxecs/raw/master-4.2/sipXcallController/src/main/java/org/sipfoundry/callcontroller/SipServiceImpl.java

maybe it's a default that should be parameterized or system configurable.
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Re: [sipx-users] Passive installation

[Douglas Hubler]

On Thu, Dec 16, 2010 at 4:46 AM, Xavier D.  wrote:
> Hello guys,
>
> Does someone already make a 'passive' installation ?
> I mean without any interactions with the machine, only by scripting all the
> installation (adding servers, configuring them, ...) ?
> I'm interested in any feedback.
>
> BRs

sdog on IRC in #sipx channel on irc.freenode.net is trying to do this.
 gui actions will be challenging, only a percentage of the operations
are available via REST.  Although some may baulk at this, using a
screen scraping utility to automate what is not available in REST is
totally doable because sipxconfig has a full battery of unit tests
that do just this.  This would be a stop gap until REST APIs would be
built.

I am very, very interested in this helping people be successful in
this as it's very key to many things good.
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Re: [sipx-users] testing t38 fax with sipx.

[Douglas Hubler]

fix is in source control, but there's no build yet.  i will let folks
know when it's ready.  Yes, automating build is still WIP.

On Thu, Dec 16, 2010 at 11:22 AM, Cristi Starasciuc
 wrote:
> Hi,
>
> The fix for fax is in, and ready to be tested.
>
> Regards,
> C
>
> On 11/26/2010 02:07 PM, Nikolay Kondratyev wrote:
>> I just tested if 0.4.4 can receive fax.
>> The  situation is the same, I can see in freeswitch.log, that fax is
>> received, but nothing is sent via email.
>> Meanwhile sipx successfully sends other email notifications: vm and backup
>> notifications, and I see records in /var/log/maillog regarding vm and backup
>> notifications, and there is nothing in maillog when fax is received.
>>
>> Rgds,
>> Nikolay.
>>
>>
>>> -Original Message-
>>> From: sipx-users-boun...@list.sipfoundry.org
>>> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
>>> Cristi Starasciuc
>>> Sent: Monday, November 22, 2010 10:20 PM
>>> To: sipx-users@list.sipfoundry.org
>>> Subject: Re: [sipx-users] testing t38 fax with sipx.
>>>
>>> Hi,
>>>
>>> I got a chance to look at fax.
>>> I have attached a patch (http://track.sipfoundry.org/browse/XX-8645)
>>> that enables proxy media for fax application only.
>>>
>>> I have looked at the email thing. I couldn't figure it out
>>> what happens, but it may not be sipx related. If I look in my
>>> inbox on my machine I get a lot of failed messages from
>>> "postmaster" trying to deliver the fax mails. Some tweaks on
>>> one's system might do the trick.
>>> I am going on vacation tomorrow, but I'll take a look at the
>>> email thing once I get back.
>>> (http://track.sipfoundry.org/browse/XX-9235)
>>>
>>> C
>>>
>>> On 11/17/2010 08:44 AM, Nikolay Kondratyev wrote:
>>>
 Yes.
 Rgds,
 Nikolay.




> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf
>
>>> Of Cristi
>>>
> Starasciuc
> Sent: Tuesday, November 16, 2010 6:47 PM
> To: sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] testing t38 fax with sipx.
>
> Were the successful tests  made using proxy media to true?
>
> * I get same results, I don't know why mails are not being
>
>>> sent. I'll
>>>
> look into it.
>
>
>
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