Re: [sipx-users] localization voice prompts from 4.0 still OK for 4.4?

[Douglas Hubler]

On Tue, May 17, 2011 at 5:35 PM, Carl Farrington  wrote:
> Thanks very much. It worked, although there is some inconsistency on the 
> operator AA and voicemail - once says "square" and the other says "hash". 
> I'll probably find the square.wav or whatever, and symlink to hash.wav.

In 4.6 localization packages are rpm based.  This will keep them
moving along with the versions
  https://github.com/dhubler/sipXlang-en_GB
If you notice any improvements that could be made, I can even give you
commit access to the repo and you can make the updates for the next
sipxecs release for everyone to benefit.

There's not need to know how to code, but there are a few things to
learn about git you can find instructions for.

This of course goes for anyone w/localization abilities.
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Re: [sipx-users] Need some help, dropping calls, paging locking up

[Douglas Hubler]

On Tue, May 17, 2011 at 1:03 PM, Matthew Kitchin (public/usenet)
 wrote:
> Ok. This problem. "About 7-8 times a day when the main line rings, I go
> to answer it and no one is there, but on the LED screen the first button
> says “answer”. You cannot pick up the call at all and the caller hangs
> up and calls back."
> This is the user not being able to answer the original inbound call at
> all. Are you saying evidence of this issue should be in sipxpark log?

In short, 7 to 8 calls a day from your ITSP are dead.  What happens
for successful calls i would say is irrelevant, wouldn't you?

"You cannot pick up the call at all"
So the buttons on the polycom buttons appear to not be responding?
Phone continues to ring but user w/extension 10 smashes on the polycom
phone buttons in vain?
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Re: [sipx-users] Need some help, dropping calls, paging locking up

[Douglas Hubler]

On Tue, May 17, 2011 at 12:51 PM, Matthew Kitchin (usenet/public)
 wrote:
> Sorry. We switched from auto attendant to user at ext 10 handling most calls. 
> That is when the problems began. 7 to 8 calls per day now are unable to be 
> answered, and 50% of pages are unsuccessful.

focusing on problem #1 first: Did you find anything in the logs, in
particular sipxpark log?
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Re: [sipx-users] Need some help, dropping calls, paging locking up

[Douglas Hubler]

I'm having trouble understanding when the problem began. Can you
reformat this focusing on the very last change you made when the
problem started to occur.

On Tue, May 17, 2011 at 11:39 AM, Matthew Kitchin (public/usenet)
 wrote:
> This is a fun one. I need some pointers on where to start.
> I have about 15 sites that are all roughly identical in their setup, so
> I know the general design is ok.
> This particular site was Sipx 4.2.1 and Polycom 450/550 firmware 3.2.4
> and bootrom 4.3 up until last week.
> They are now sipx 4.4 (latest) Polycom 450/550 firmware 3.2.5 and
> bootrom 3.2.5
> Cisco 2960 POE 24 port switch
> Sipxbridge, Verizon VoIP services, no IP address NATing
> Verizon MPLS T1
> Everything is QOSd properly. There are no errors on the router or the
> switch interfaces.
> Unfortunately, the problems remain even after the Sipx and Polycom upgrade.
> There are 16 Polycom 450s and 1 550. The 550 is the receptionist, and
> she answers the bulk of the calls. They were on an auto attendant, but
> decided they wanted a live person answering the calls. That is when my
> problems started.
> The path of most calls is answer, park with blind transfer, page with
> paging group, user retrieves call from park on one of the 450s.
>  From everything we can tell, they are doing this properly. They are
> experiencing 2 major problems. I am not sure if one is causing the other
> or if they are related.
>
> Problem 1, straight from the Polycom 550 user "About 7-8 times a day
> when the main line rings, I go to answer it and no one is there, but on
> the LED screen the first button says “answer”. You cannot pick up the
> call at all and the caller hangs up and calls back."
>
> Problem 2, also from the 550 user "about 50% of the time the overhead
> can’t be heard, then when you try to use the overhead a second time it
> rings busy"
> She is referring to the the paging group. Nobody hears anything. The
> paging call then sits in the active calls list within sipx indefinitely.
> Usually, I will have to restart the paging service to get everything
> going again.
>
> This site seems like a simple setup to me. I have several others that
> are identical. I have tried everything I can think of, so I'm ready to
> start at square 1 again. Can anyone give me some tips?
>
> Thanks as always,
> Matthew
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Re: [sipx-users] Create custom dial plan template with custom rules

[Douglas Hubler]

On Mon, May 16, 2011 at 10:55 AM,   wrote:
> I have find in the documentation where are the localized template, under
> /etc/sipxpbx/region_xx
<
> I want to change the region_ch template.
>
> How can I add a custom rule ?

Pierre, sorry for missing your post before.  There is a concept of a
"Custom Rule" but I'm not sure if that's what you mean.   Regions tend
to describe what the non-custom rules are like local calls, emergency
calls,  .  Do you want to make changes to the current one?  Do you
want to define a whole new region?

>
> Where can  I find the list of the properties to use in the xml custom dial
> plan ?


Properties can mean two things too.  Localization of strings seen on
the GUI or properties like default PSTN Prefix. Which do you mean?  If
it's the latter, the best advice I would have is look at the source

https://github.com/dhubler/sipxecs/tree/master-4.2/sipXconfig/neoconf/src/org/sipfoundry/sipxconfig/admin/dialplan

and match-up EmergencyRule.java with defaultEmergencyRule and if class
has a method
  setEmergencyNumber(String )
then you can use property
  
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Re: [sipx-users] localization voice prompts from 4.0 still OK for 4.4?

[Douglas Hubler]

On Mon, May 16, 2011 at 7:51 PM, Carl Farrington  wrote:
> Are the localized voice prompts (i.e. British English) from the 4.0
> directory on the download server still OK for 4.4? Any known issues, or
> files that I might need to rename/symlink or anything?

all i can say is that other 4.0 files have worked:including japanese,
german and italian i've tried.

You shouldn't run into a problem, but keep an eye on
/usr/share/www/doc/stdprompts symlink.  I don't know if it *has* to
change to British prompts to work, but symlink was put in recently for
localization out of the box suppport.

  /usr/share/www/doc/stdprompts -> stdprompts_en
  /usr/share/www/doc/stdprompts_en
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Re: [sipx-users] sipxopenfire.log Errors

[Douglas Hubler]

On Mon, May 16, 2011 at 4:04 PM, James Walsh  wrote:
> A VM I made is spewing "404" errors every six seconds, and a log file nearly
> triple the size of the other machine.

check for errors in registrar.log.  i think it's trying to connect to
a presence plugin
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Re: [sipx-users] 4.4 sipXrls dead

[Douglas Hubler]

On Mon, May 16, 2011 at 7:24 AM, George Niculae  wrote:
> Can you check permissions for var/sipxdata/tmp/imdb.*
> You may also want to try stop sipxrls, delete these files and restart sipxrls

keep in mind if you delete these, you'd loose registrations and phones
wouldn't know to re-register for up to 2 hours.
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Re: [sipx-users] 4.4 sipXrls dead

[Douglas Hubler]

On Sun, May 15, 2011 at 11:27 AM, Josh M. Patten
 wrote:
> It appears my assumption might be correct. That CA is from when this system 
> was originally built and I think the SSL certs being created are based on a 
> different CA. So how do I go about taking the CA that 
> /bin/ssl-cert/gen-ssl-keys.sh creates and putting it into place?

I think sipxecs-setup caches cert data in

  /var/sipxdata/certdb

so I would try

 mv /var/sipxdata/certdb{,.bak}

then re-run

 sipxecs-setup
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Re: [sipx-users] 4.0.4 repomd.xml missing

[Douglas Hubler]

fixed

y, found it missing for 64bit.

On Sat, May 14, 2011 at 7:12 PM, Tony Graziano
 wrote:
> can someone fix it?
>
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Re: [sipx-users] XMPP Clients on 4.4.0

[Douglas Hubler]

yeah, beware of this issue we found w/spark client
  http://community.igniterealtime.org/thread/42976

On Fri, May 13, 2011 at 4:30 PM, Kyle Haefner
 wrote:
> Aaron,
> Did you do anything special.I have not been able to get Spark to
> authenticate at all!
> Thanks!
> Kyle
>
> On Thu, May 12, 2011 at 2:45 PM, Aaron Pursell  wrote:
>>
>> Well I actually got Spark working through XMPP, with Audio/Video and
>> Desktop sharing through the integrated Openfire in sipXecs 4.4.0.
>>
>>
>>
>>
>> Aaron Pursell, Sr.
>> Network Systems Administrator
>> Easter Seals - Goodwill, Northern Rocky Mountain
>> 4400 Central Ave
>> Great Falls, Montana  59405
>>
>> (406) 771-3721
>> aar...@esgw.org
>> >>> "Aaron Pursell"  5/11/2011 3:45 PM >>>
>> Anyone tested XMPP clients with 4.4.0? We're looking for a replacement our
>> conferencing so it needs to support video and if we can find the right
>> working setup, we'll go with it.
>>
>> Also we're not interested in Bria due to the fact that even to talk to
>> them you have to sign a 3 year NDA. So far I've tested Spark (best one so
>> far) and pidgin.
>>
>> Anyone have any suggestions?
>>
>>
>>
>>
>> Aaron Pursell, Sr.
>> Network Systems Administrator
>> Easter Seals - Goodwill, Northern Rocky Mountain
>> 4400 Central Ave
>> Great Falls, Montana  59405
>>
>> (406) 771-3721
>> aar...@esgw.org
>
>
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Re: [sipx-users] YUM UPDATE Error

[Douglas Hubler]

Fresh install worked.

I've have seen this when using a caching proxy.  Do you use a proxy to
download or is there some firewall that might be caching something?


SHA1 calcs
=
[root@finch sipXecs]# pwd
/var/cache/yum/sipXecs
[root@finch sipXecs]# sha1sum primary.xml.gz
4ef61b4c336cfea08db1263ee1ffb140fedfd370  primary.xml.gz
[root@finch sipXecs]# cat repomd.xml

http://linux.duke.edu/metadata/repo";
xmlns:rpm="http://linux.duke.edu/metadata/rpm";>
  1305085354
  
c17a9a153a98031aec083257c326214cdc7f38d7
1305085363
10365
83699
12ab3f839f59a04f188264e98735e9913f12876c

  
  
abfba11a8f799d047af37c1f629767f0bc35dd3e
1305085363
245118
3498424
7ca95c22ddff88b742a4d88f52a70770865ff45f

  
  
4ef61b4c336cfea08db1263ee1ffb140fedfd370
1305085363
54197
467868
2c494aaff224806f9633a45f637c78b7294e6a93

  



On Fri, May 13, 2011 at 4:08 PM, James Walsh  wrote:
> I used Microsoft's command line utility, and verified with a GUI utility I
> quickly grabbed from here: http://raylin.wordpress.com/downloads/
>
>
>
>
>
> D:\SIP>c:\fciv\fciv -add other.xml.gz -sha1
>
> //
>
> // File Checksum Integrity Verifier version 2.05.
>
> //
>
> c17a9a153a98031aec083257c326214cdc7f38d7 other.xml.gz
>
>
>
> 15d0ff3622127cdcdc6d5b68ebb8c2411c4fbf4c <- repomd.xml
>
>
>
> D:\SIP>c:\fciv\fciv -add filelists.xml.gz -sha1
>
> //
>
> // File Checksum Integrity Verifier version 2.05.
>
> //
>
> abfba11a8f799d047af37c1f629767f0bc35dd3e filelists.xml.gz
>
>
>
> 256899a3aba13738837a50ee92fa85030ceb46a2 <- repomd.xml
>
>
>
> D:\SIP>c:\fciv\fciv -add primary.xml.gz -sha1
>
> //
>
> // File Checksum Integrity Verifier version 2.05.
>
> //
>
> 4ef61b4c336cfea08db1263ee1ffb140fedfd370 primary.xml.gz
>
>
>
> 7c99ce8441e2cb84451baf2d22d1f60170bd63c1 <- repomd.xml
>
>
>
>
>
>
>
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Douglas Hubler
> Sent: May-13-11 3:49 PM
> To: Discussion list for users of sipXecs software
> Subject: Re: [sipx-users] YUM UPDATE Error
>
>
>
> can you show me command and output of calculating SHA?
>
> in the meantime, I'm trying a fresh install...
>
>
>
> On Fri, May 13, 2011 at 3:44 PM, James Walsh  wrote:
>
>> The baseurl was correctly identified at the time I ran the update, and
>
>> specifying x86_64 didn't help.
>
>>
>
>> I calculated the SHA of the files at the listed in the repo, and the
>
>> sums didn't correspond to what was in the repomd.xml - which is why I
>
>> thought it to be a problem at the site, and reported it initially.  I
>
>> checked the x86 repo and the checksums match correctly, so unless the
>
>> x86_64 is calculated differently...
>
>>
>
>> I checked again, and  the checksums just don't add up.
>
>>
>
>> Am I doing something wrong???
>
>>
>
>>
>
>>
>
>>
>
>> -Original Message-
>
>> From: sipx-users-boun...@list.sipfoundry.org
>
>> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Matthew
>
>> Kitchin
>
>> (public/usenet)
>
>> Sent: May-13-11 3:08 PM
>
>> To: sipx-users@list.sipfoundry.org
>
>> Subject: Re: [sipx-users] YUM UPDATE Error
>
>>
>
>> I upgraded two 64 bit 4.2.1 machines yesterday, andI had to change the
>
>> repo file slightly.
>
>> I changed mine to this:
>
>> baseurl=http://download.sipfoundry.org/pub/sipXecs/4.4.0/CentOS_$relea
>
>> sever/
>
>> x86_64/
>
>>
>
>> I had to add the x86_64/ to the end of it. I'm not sure if I should
>
>> have had to do that or not, but it didn't work without it, and that is
>
>> what I visually saw when I looked at the site in a browser.
>
>>
>
>>
>
>> On 5/13/2011 2:03 PM, James Walsh wrote:
>
>>> No joy on those last commands, except, of course, a broken install.
>
>>>
>
>>> I'm just going to point this out for the halibut - I'm using the
>
>>> x86_64 release.  It was 4.2.1 upgraded to 4.4.0
>
>>>
>
>>> My \etc\yum.repos.d\sipxecs.repo:
>
>>>
>
>>> [sipXecs]
>
>>> name=sipXecs software for CentOS $releasever - $basearch
>
>>> baseurl=http://download.sipfoundry.org/pub/sipXecs/4.4.0/CentOS_$rele
>
>>> a
>
>>> sever/
>
>>> $basearch
>
>>> gpgcheck=0
>

Re: [sipx-users] YUM UPDATE Error

[Douglas Hubler]

can you show me command and output of calculating SHA?
in the meantime, I'm trying a fresh install...

On Fri, May 13, 2011 at 3:44 PM, James Walsh  wrote:
> The baseurl was correctly identified at the time I ran the update, and
> specifying x86_64 didn't help.
>
> I calculated the SHA of the files at the listed in the repo, and the sums
> didn't correspond to what was in the repomd.xml - which is why I thought it
> to be a problem at the site, and reported it initially.  I checked the x86
> repo and the checksums match correctly, so unless the x86_64 is calculated
> differently...
>
> I checked again, and  the checksums just don't add up.
>
> Am I doing something wrong???
>
>
>
>
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Matthew Kitchin
> (public/usenet)
> Sent: May-13-11 3:08 PM
> To: sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] YUM UPDATE Error
>
> I upgraded two 64 bit 4.2.1 machines yesterday, andI had to change the repo
> file slightly.
> I changed mine to this:
> baseurl=http://download.sipfoundry.org/pub/sipXecs/4.4.0/CentOS_$releasever/
> x86_64/
>
> I had to add the x86_64/ to the end of it. I'm not sure if I should have had
> to do that or not, but it didn't work without it, and that is what I
> visually saw when I looked at the site in a browser.
>
>
> On 5/13/2011 2:03 PM, James Walsh wrote:
>> No joy on those last commands, except, of course, a broken install.
>>
>> I'm just going to point this out for the halibut - I'm using the
>> x86_64 release.  It was 4.2.1 upgraded to 4.4.0
>>
>> My \etc\yum.repos.d\sipxecs.repo:
>>
>> [sipXecs]
>> name=sipXecs software for CentOS $releasever - $basearch
>> baseurl=http://download.sipfoundry.org/pub/sipXecs/4.4.0/CentOS_$relea
>> sever/
>> $basearch
>> gpgcheck=0
>>
>> Next???
>>
>>
>> -Original Message-
>> From: sipx-users-boun...@list.sipfoundry.org
>> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Douglas
>> Hubler
>> Sent: May-13-11 2:47 PM
>> To: Discussion list for users of sipXecs software
>> Subject: Re: [sipx-users] YUM UPDATE Error
>>
>> I could not duplicate this. On a system w/4.4.0 installed i ran
>>    yum clean all
>>    yum erase sipx*
>>    yum install sipxecs
>>
>> Anyone else?
>>
>> On Fri, May 13, 2011 at 2:22 PM, James Walsh  wrote:
>>> One of the first things I tried, I'm afraid...
>>>
>>>
>>> -Original Message-
>>> From: sipx-users-boun...@list.sipfoundry.org
>>> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Matthew
>>> Kitchin
>>> (public/usenet)
>>> Sent: May-13-11 1:54 PM
>>> To: sipx-users@list.sipfoundry.org
>>> Subject: Re: [sipx-users] YUM UPDATE Error
>>>
>>> On 5/13/2011 12:50 PM, James Walsh wrote:
>>>> Hi all,
>>>>
>>>> Receiving this error the last couple of days trying to update for
>>>> the various fixes that were released:
>>>>
>>>> sipXecs/primary                                          |  53 kB
>>> 00:00
>>>> http://download.sipfoundry.org/pub/sipXecs/4.4.0/CentOS_5/x86_64/rep
>>>> o
>>>> d
>>>> ata/pr
>>>> imary.xml.gz: [Errno -1] Metadata file does not match checksum
>>>>
>>>> Any ideas??
>>>>
>>> try 'yum clean all' and run it again?
>>>> James
>>>>
>>>>
>>>>
>>>>
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Re: [sipx-users] YUM UPDATE Error

[Douglas Hubler]

I could not duplicate this. On a system w/4.4.0 installed i ran
  yum clean all
  yum erase sipx*
  yum install sipxecs

Anyone else?

On Fri, May 13, 2011 at 2:22 PM, James Walsh  wrote:
> One of the first things I tried, I'm afraid...
>
>
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Matthew Kitchin
> (public/usenet)
> Sent: May-13-11 1:54 PM
> To: sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] YUM UPDATE Error
>
> On 5/13/2011 12:50 PM, James Walsh wrote:
>> Hi all,
>>
>> Receiving this error the last couple of days trying to update for the
>> various fixes that were released:
>>
>> sipXecs/primary                                          |  53 kB
> 00:00
>> http://download.sipfoundry.org/pub/sipXecs/4.4.0/CentOS_5/x86_64/repod
>> ata/pr
>> imary.xml.gz: [Errno -1] Metadata file does not match checksum
>>
>> Any ideas??
>>
> try 'yum clean all' and run it again?
>> James
>>
>>
>>
>>
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>
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Re: [sipx-users] Bug fix release update: sipXecs 4.4.0 has been updated

[Douglas Hubler]

On Thu, May 12, 2011 at 4:47 PM, Tony Graziano
 wrote:
> versioning the repo is perhaps the most straightforward method in my mind.
> an archive of the initial release any any patches that are not the most
> recent version.

I'm not sure what you mean here.

The policy now is exactly what you'd find w/CentOS.
- ISOs stays the same
- you update if you want the latest
- if you require only the RPMs that were available for download on the
CentOS mirrors on November 14th, 2010 then you best keep a local copy
- CentOS makes available the latest RPMs for each of these major.minor
versions :  5.0 - 5.6

re: Changelog suggested by Todd
Yep. good idea.
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Re: [sipx-users] Bug fix release update: sipXecs 4.4.0 has been updated

[Douglas Hubler]

On Thu, May 12, 2011 at 4:29 PM, Matthew Kitchin (public/usenet)
 wrote:
> On 5/12/2011 3:20 PM, Douglas Hubler wrote:
>>> I
>>> do wonder why they are not archived in another repo (the previous 4.4.0-192
>>> packages. It makes the onus on the superadmin to have saved it as its no
>>> longer available.
>> Added items #6 to release policy...
>> http://wiki.sipfoundry.org/display/sipXecs/Bug+Fix+Release+Policy
> Is it remotely feasible to have the patch process store a copy of these
> before they are upgraded? If not by default, then maybe as an option?

I'm willing to come up with easy instructions for how folks can keep
local copies.

re:ISO
I might change my mind, but ISO are not updated with new RPMs.  People
would need to be in the habit of updating after installing if they
want the latest fixes.  rationale i gave last time: i'm not uploading
a binary that wasn't tested, and I'm not testing ISOs for every fix.
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Re: [sipx-users] Bug fix release update: sipXecs 4.4.0 has been updated

[Douglas Hubler]

On Thu, May 12, 2011 at 3:15 PM, Tony Graziano
 wrote:
> devel packages. So I think it's only 10 that were pushed out in the patch.

yep

To a certain degree, you have to trust yum is configured correctly and
you tell you if you have the latest.  Paranoid admins (which sometimes
includes me) often double check (as Tony did) the file names in the
download directory.

> I
> do wonder why they are not archived in another repo (the previous 4.4.0-192
> packages. It makes the onus on the superadmin to have saved it as its no
> longer available.

Added items #6 to release policy...
http://wiki.sipfoundry.org/display/sipXecs/Bug+Fix+Release+Policy
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Re: [sipx-users] TLS & IVR

[Douglas Hubler]

On Thu, May 12, 2011 at 11:06 AM, barisyanar  wrote:
> of course a core file is generated.
> here is a sample one:
> http://www.ceng.metu.edu.tr/~e1348093/paylasim/core.21730

Tell us what you've investigated so far by analyzing the core.
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Re: [sipx-users] TLS & IVR

[Douglas Hubler]

On Thu, May 12, 2011 at 8:03 AM, barisyanar  wrote:
> I managed to make TLS signal and SRTP transport to another client.
> But whenever I try to make an IVR call, FreeSwitch crashes.
> Attached are the necessary logs.

nothing in the logs pertaining to a crash that i can see.

By "crash", do you mean core is generated, or call disconnected, or
traveling in a speeding car...

Try to verify SIP and RTP traffic separately.  The sip TLS traffic
thru sipXecs was notorious for getting ports wrong, study the Via
headers to make sure the ports make sense.
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Re: [sipx-users] ConfigAgent & PresenceServer "ConfigurationMismatch" after upgrade

[Douglas Hubler]

On Wed, May 11, 2011 at 11:26 AM, Tony Graziano
 wrote:
> I hear it I think about that upgrading from unstable versions is supported.
>
> well I am making my own point and maybe only to myself, upgrading from
> unstable versions is not implicitly supported until these issues are
> resolved.

Upgrading from unstable versions is not supported.  If someone
(including me) told you otherwise, they probably shouldn't have.
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Re: [sipx-users] Bug fix release update: sipXecs 4.4.0 has been updated

[Douglas Hubler]

On Wed, May 11, 2011 at 11:27 AM, Matthew Kitchin (public/usenet)
 wrote:
> On 5/10/2011 11:24 PM, Douglas Hubler wrote:
>> This is a notice in accordance with bug fix release policy
>>
>>     http://wiki.sipfoundry.org/display/sipXecs/Bug+Fix+Release+Policy
>>
> If it was determined one of the bugfixes caused an issue, is there a way
> to back it out?

keep your rpms and force reinstall them.  i wouldn't knowingly let a
bug fix be submitted unless you could downgrade to any previous set of
rpms in the same major.minor version.


I'll add this to the policy...
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Re: [sipx-users] Bug fix release update: sipXecs 4.4.0 has been updated

[Douglas Hubler]

On Wed, May 11, 2011 at 10:17 AM, Dave Deutschman
 wrote:
> +1

http://track.sipfoundry.org/browse/XX-9609

FYI: It would be very hard to base this on context of bug fixes.
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Re: [sipx-users] ConfigAgent & PresenceServer "ConfigurationMismatch" after upgrade

[Douglas Hubler]

On Wed, May 11, 2011 at 9:26 AM, Tony Graziano
 wrote:
> I had planned that. What I didn't understand is why none of the logging
> components failed to capture the issue and why there was a reported
> "ConfigurationMismatch".
> While everything reported properly in services, the fact that the
> PresenceServer had a log of "0" bytes make me think it was not running, yet
> sipxconfig UI reported it was. To me, it looks like a bug. "How" I got to
> this version is questionable, and I can easily correct with a rebuild. Am I
> overly concerned with service reported running when it probably is not
> though?

It's a good point. If someone see's it again, maybe i'll be inspired
to investigate.  Internals of supervisor code is not straightforward
to me to have a good idea of the issue.
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Re: [sipx-users] Bug fix release update: sipXecs 4.4.0 has been updated

[Douglas Hubler]

On Wed, May 11, 2011 at 7:44 AM, Tony Graziano
 wrote:
> Please note: this will repush profiles to end user devices as well as reboot
> them, so shedule/apply accordingly.

Would everyone agree we should change this so it only happens on major
version upgrades?
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Re: [sipx-users] OpenACD

[Douglas Hubler]

Weren't we supposed to get descriptions on those fields?

I think it means call will ring until agent is available. That is, folks
don't get put on hold. Yet another way to think about it: kinda like a be
hunt group.

This is how it was explained to me, I never tried it though.
On May 11, 2011 4:47 AM, "Laurentiu Ceausescu"  wrote:
> On Wed, May 4, 2011 at 11:28 AM, Kumaran <
> thiru.venkateshwa...@ttplservices.com> wrote:
>
>> Hi All,
>> I'm testing OpenACD on the build "sipXconfig (0.0.4.5.2-
>> 2011-04-26EDT08:49:49 domU-12-31-39-05-28-E2)".When Adding a line there
>> is option "supervision check box" which is enabled by default.So what is
>> use of Answer supervision check box?
>> Please guide.
>>
>
> Answer supervision indicates that the call is being answered.
> Douglas explained me some time ago how this option works ... but for my
> shame I forgot.
> Douglas can you explain one more time please?
>
> Thanks,
> Laurentiu
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[sipx-users] Bug fix release update: sipXecs 4.4.0 has been updated

[Douglas Hubler]

This is a notice in accordance with bug fix release policy

   http://wiki.sipfoundry.org/display/sipXecs/Bug+Fix+Release+Policy

Notes
=
- ** No security updates in this update **
- ISO has *not* been rebuilt as decided in release policy. Yum update
after installation is recommended for getting these updates
- Thank you George, Mircea and Domenico for your fixes and everyone
for your bug reports!


Build Log
=

commit 157292d20020af90e1e905ce4859000758bbf581
Author: George Niculae 
Date:   Tue May 10 13:14:42 2011 +0300

XX-9582: SipXbridge not working properly in 4.4. with HA cluster

- set sip trunk associated sbc device when loading EditGateway
page (otherwise will save it always with the built in sbc from
primary)

M   
sipXconfig/web/src/org/sipfoundry/sipxconfig/site/gateway/EditGateway.java

commit 6b3e39ff982a669b19001738e2760cdaf9226131
Author: Mircea Carasel 
Date:   Mon May 9 18:34:45 2011 +0300

XX-9577: LDAP Authentication using user alias is not working

-obtained user's true username id and sent it to ldap

M   
sipXconfig/neoconf/src/org/sipfoundry/sipxconfig/security/ConfigurableLdapAuthenticationProvider.java

commit 909e62ed2259c5005f9bd298612e00ab08da5a50
Author: George Niculae 
Date:   Fri May 6 02:35:07 2011 +0300

XX-9565: record and send message also after 1 is pressed and menu
timeouts after 1 is pressed and wrong option provided

M   sipXivr/src/main/java/org/sipfoundry/voicemail/Deposit.java
M   sipXivr/src/main/java/org/sipfoundry/voicemail/VmMenu.java

commit ca825992f141519912424c7602d810dfbeb10ca1
Author: George Niculae 
Date:   Mon May 9 17:07:56 2011 +0300

Revert "XX-9565: record and send message also after 1 is pressed
and menu timeouts after 1 is pressed and wrong option provided"

This reverts commit f9a297ab811027de76b8ba9964b2a8780c6cca7d.

M   sipXivr/src/main/java/org/sipfoundry/voicemail/Deposit.java
M   sipXivr/src/main/java/org/sipfoundry/voicemail/VmMenu.java

commit f9a297ab811027de76b8ba9964b2a8780c6cca7d
Author: George Niculae 
Date:   Fri May 6 02:35:07 2011 +0300

XX-9565: record and send message also after 1 is pressed and menu
timeouts after 1 is pressed and wrong option provided

M   sipXivr/src/main/java/org/sipfoundry/voicemail/Deposit.java
M   sipXivr/src/main/java/org/sipfoundry/voicemail/VmMenu.java

commit c9044a8087d183ec0945b69149c3b99e575dd306
Author: George Niculae 
Date:   Fri Apr 29 16:30:15 2011 +0300

XX-8063: fixed 'SipTcpServer-3' - no room, ret = 9" error

Patch from Domenico Chierico, thanks!

M   sipXtackLib/src/net/SipClient.cpp
M   sipXtackLib/src/net/SipTcpServer.cpp

commit 3dc3fa1783324615c2f1a03a40f73322bbb1e340
Author: George Niculae 
Date:   Thu Apr 28 17:46:04 2011 +0300

XX-9574: User portal: automatic reload FS when conference change

- when changing conferences from user portal, FS automatically
reloads so changes become effective without other action
- exposed SipxProcessContext.manageServices(Collection< ? extends
SipxService> processes, Command command) - it will issue command on
all locations for the given service collection
- when called from user portal EditConference will issue reload
for FS service

M   
sipXconfig/neoconf/src/org/sipfoundry/sipxconfig/admin/commserver/SipxProcessContext.java
M   
sipXconfig/web/src/org/sipfoundry/sipxconfig/site/conference/EditBridge.java
M   
sipXconfig/web/src/org/sipfoundry/sipxconfig/site/conference/EditConference.java
M   
sipXconfig/web/src/org/sipfoundry/sipxconfig/site/conference/UserConferencesPanel.java

commit 1da88129b3827b5799c980d3b25dd181209eb4ba
Author: George Niculae 
Date:   Wed Apr 27 15:50:01 2011 +0300

XX-9573: remove call rate limit settings from sipxproxy.xml

M   sipXconfig/neoconf/etc/sipxproxy/sipxproxy.properties
M   sipXconfig/neoconf/etc/sipxproxy/sipxproxy.xml

commit 63a7da5198f01244bb1259873d4afc51bb849fe2
Author: George Niculae 
Date:   Thu Apr 21 12:47:46 2011 +0300

HttpMessage: change log level form warning to debug

M   sipXtackLib/src/net/HttpMessage.cpp

commit 3097e494c97776cd2b1e274a492154b51a39a9cf
Author: George Niculae 
Date:   Wed Apr 20 14:28:44 2011 +0300

while parsing nattraversalrules.xml sipXproxy reports ERROR when
- publicaddress not found - this should be ok if address type set
to STUN, changed log level to debug
- mediarelayexternaladdress not found - this is never written in
nattraversalrules by config, changed log level to debug
- secureXMLRPC not found - never written in nattraversalrules.xml
by config, changed log level to debug
- rediscovery-time not found - should be OK if address type set to
public IP, changed log level to debug
- stun-server-address not found - should be OK if address type set
to public IP, changed log level to debug

M   sipXproxy/lib/authplugins/NatTraversalAgent/NatTraversalRules.cpp
__

Re: [sipx-users] DHCP

[Douglas Hubler]

On Mon, May 9, 2011 at 1:32 PM, Charles Chalekson  wrote:
> Ok, I decided to try to keep my Sonicwall TZ-100 as my DHCP server.  I set
> option 66 to my pbx server [string type, no other option], option 128 to the
> IP address of the server [10.10.10.200], and option 150 with my domain name
> [pbx.doctor.local] and I still cant get it to pick up the ftp/tftp server.

My wireshark shows proper options in DHCP traffic.  What does yours show?
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Re: [sipx-users] Paging

[Douglas Hubler]

On Mon, May 9, 2011 at 5:50 AM, Kumaran
 wrote:
>  Please check the scenario which is totally Invalid but to make sipxecs
> more friendly and flexible
>    Scenario:
>       1.Paging Prefix set to 10 from *77 and Paging number Ext =0
>  2.When I dial 100 its going to Paging not  to AA(Paging is validated
> first)for this scenario whether Sipregistar can validate the Paging
> combination while adding it?
>
>  Check some invalid combination:
>   1.added user 980 and Paging prefix 98 and  Paging number Ext=0
>        The call will established to Paging service not to user 980
>   2.added user *770 and Paging prefix 77 and Paging number Ext=0
>        The call will established to Paging service not to user *770


Are you reporting a bug or suggesting a useful setup?
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Re: [sipx-users] firefox 4.4 and GUI Phonebook loading problem

[Douglas Hubler]

On Mon, May 9, 2011 at 8:40 AM, opyzcinr opyzcinr  wrote:
> What's wrong with FF4?

the phonebook was written in gwt which has all sorts of checks for
browser makes and versions.  phonebook app may have to be "recompiled"
for firefox 4.  can you create a bug and reference this thread?
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Re: [sipx-users] OpenACD

[Douglas Hubler]

I would create feature requests then. In the descriptions, try to
describe why you need the feature instead of just mentioning the old
acd server had it.

On Mon, May 9, 2011 at 2:36 AM, Kumaran
 wrote:
> Hi Douglas,
>      In old ACD there is no option called voicemail...So if agent
> refuse the call it rolls to another agent...
>      According to my scenario in old ACD is if 200 refuse the call it
> rolls to 201 and if 201 refuse the call  it  rolls to other agent.And If
> agent not available then the call will be in the queue.
>      But in OpenACD if 200 refuse its going to voicemail.
>
> There are so many option for monitoring the agent calls in old ACD by
>    Agent wrap-up time
>    Agent Non-Responsive time
>    Maximum Bounce Count
> But in OpenACD I never came through any of the above option.
>
> Regards
> Kumaran T
>
>
> Douglas Hubler wrote:
>> On Fri, May 6, 2011 at 4:30 AM, Kumaran
>>  wrote:
>>
>>>    Please check the below scenario and let me know the update.
>>>       1.OpenACD line 650
>>>       2.Agent 200 and 201 are logged in and available
>>>       3.Caller calls the line 650
>>>       4.Agent 200 rings
>>>       5.Don't answer or reject the call
>>>       6.The call rolls to 200 agent VM
>>>  Whether step6 is the valid behavior or The agent 201 should ring  after
>>> agent 200 reject or don't answer the call?
>>>
>>
>> What does the old ACD do?  I would suspect that particular call should
>> roll over to 201.  We'll probably need to stick the "novoicemal" sip
>> url parameter in the outgoing call to clue freeswitch in.
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>>
>
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Re: [sipx-users] DHCP

[Douglas Hubler]

On Sun, May 8, 2011 at 9:51 AM, Charles Chalekson  wrote:
> One question however, I have always used tftp and know the polycoms out of
> the box are defaulted to FTP. Does option 66 also automatically work for FTP
> too?  I know that sipxecs has a ftp server tuned on by default. When you
> upload configuration files to the pbx I know they go to the tftp folder so
> will they get picked up if the phone is set to FTP?

yes, to all your questions.
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Re: [sipx-users] DHCP

[Douglas Hubler]

On Sat, May 7, 2011 at 12:18 PM, Charles Chalekson  wrote:
> I have had SIPx running for several years without the server running DHCP and 
> I have just
> manually configured phones in the past [long story why I did this but many 
> tylenol later
> I have decided to change].

You can still run your own dhcp server, just configure where the tftp
server is located.  For example, if you're running dhcpd on a linux
box, something like this in  /etc/dhcp/dhcpd.conf will do the trick

subnet 10.10.10.0 netmask 255.255.255.0 {
   option tftp-server-name "10.10.10.200";
}

even the simplest of routers will let you configure where the tftp
server is.  "option 66" i believe it's also call.  tftp is actually a
bit of a misnomer now, polycom and snom among others will use this
same setting for lookup for provisioning via ftp and http.
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Re: [sipx-users] OpenACD

[Douglas Hubler]

On Fri, May 6, 2011 at 4:30 AM, Kumaran
 wrote:
>    Please check the below scenario and let me know the update.
>       1.OpenACD line 650
>       2.Agent 200 and 201 are logged in and available
>       3.Caller calls the line 650
>       4.Agent 200 rings
>       5.Don't answer or reject the call
>       6.The call rolls to 200 agent VM
>  Whether step6 is the valid behavior or The agent 201 should ring  after
> agent 200 reject or don't answer the call?

What does the old ACD do?  I would suspect that particular call should
roll over to 201.  We'll probably need to stick the "novoicemal" sip
url parameter in the outgoing call to clue freeswitch in.
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Re: [sipx-users] [sipx-dev] DNS and High Availability Problem

[Douglas Hubler]

On Thu, May 5, 2011 at 1:08 PM, Douglas Hubler  wrote:
> This looks like the registration servers are replicating properly.
typo:
  This looks like the registration servers are *not* replicating properly.
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Re: [sipx-users] [sipx-dev] DNS and High Availability Problem

[Douglas Hubler]

2011/5/5 Kemal Eroğlu :
> I have a question about DNS and High Availability. I was testing the High
> Availability before our product release. I set up a primary and a redundant
> server and configured them to share the phone registration load. When all
> the phones were registered to the system; I saw that in the Registrations
> Menu in sipxconfig, only the phones that registered to the primary server
> were shown with their expire values. The phones registered to the secondary
> were not shown. Also I could not make a call between two phones that
> registered different servers. I tried this same scenario with different
> configurations, but nothing changed. Lastly I entered an external DNS as the
> "Nameserver 1" value of primary server and did the setup again. This time
> the problem did not appear. I could not understand why such a change can
> cause a problem like this. So do you have any idea about this problem or is
> it so important for a sipXecs setup to have acces to external DNSs or
> internet?

I'm reporting back to user's list as that would be the best place for
this question.

This looks like the registration servers are replicating properly. DNS
is probably the issue but i don't think you need upstream DNS
configured.  Often each server has a local caching DNS server that is
screwing things up.  Try these commands *from each and every server*
to make sure each server get the appropriate values


  dig -t SRV _sip._tcp.your-domain-here

See

http://wiki.sipfoundry.org/display/sipXecs/DNS+Verification

Ultimately supervisor or registrar logs will have the errors, but
sometimes they can be vague or cryptic, but post them back and i know
how to translate a few of them by now.
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Re: [sipx-users] Dependency failure during upgrade 4.2.1 to 4.4.0

[Douglas Hubler]

On Thu, May 5, 2011 at 12:45 PM, Tony Graziano
 wrote:
> wasn't there a 4.05 release/patch that you needed to go to to properly
> get to 4.2.1?
>
> In any case, why not just erase the offending package/dependency and
> then update and let it fix things?

Tony, that was my quick response too, but you'll notice he's not
getting a conflict, rather a missing package altogether.

Keith, I don't know where yum builds $releasever variable, but check
these commands

[root@finch ~]# head -2 /etc/issue
CentOS release 5.5 (Final)
Kernel \r on an \m

[root@finch ~]# cat /etc/redhat-release
CentOS release 5.6 (Final)

[root@finch ~]# rpm -qa | grep release
centos-release-5-6.el5.centos.1
centos-release-notes-5.6-0

( as you see, mine are a little off but it seems ok)
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Re: [sipx-users] Fwd: Send 180 Ringing before AA picks the call

[Douglas Hubler]

On Wed, May 4, 2011 at 5:18 PM, Julian Sanchez (alertas)
 wrote:
> My ITSP is asking me to send SIP 180 Ringing before establish the audio of
> the call, this in order to avoid an issue with old PSTN switch  were a call
> gets answered by AA but after a few seconds gets dropped if sipXecs don't
> send the 180 at the beginning of the call.
> So, there is any way to force sipxproxy to do this?

sipxproxy just forwards what freeswitch sends it so check into
freeswitch. That should at least help narrow your googling.
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Re: [sipx-users] MWI indicator

[Douglas Hubler]

On Wed, May 4, 2011 at 8:03 PM, Charles Chalekson  wrote:
> Still having issues with persistent MWI lighting up despite no message being
> there.

strange. try sending the user a voicemail and then delete the
voicemail.  are you getting a NOTIFY from the sipXproxy that has a
content body claiming the phone has messages?
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Re: [sipx-users] "Process 'SipXbridge' stopped unexpectedly. Attempting to restart the process."

[Douglas Hubler]

On Wed, May 4, 2011 at 9:38 AM, Solomon Mohamed  wrote:
> I changed the STUN server as suggested by Todd to 'stun.ezuce.com'

I don't think Joegen's stun service is up
  [dhubler@swift sipxecs]$ telnet stun.ezuce.com 3478
  Trying 174.142.82.79...
  telnet: connect to address 174.142.82.79: Connection refused
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Re: [sipx-users] Dependency failure during upgrade 4.2.1 to 4.4.0

[Douglas Hubler]

On Wed, May 4, 2011 at 9:48 AM, Keith Laidlaw  wrote:
> First time I tried it, I got error 404 when I ran “yum update –y” and had to
> change “$releasever” to “5” in the base url of the repo then it worked.

you shouldn't have to do this, but i think it's indicative of the
subsequent problem your getting.


> freeswitch-1.0.7-1441.g35129.i386 from sipXecs has depsolving problems
> --> Missing Dependency: libvorbis.so.0 is needed by package

Somehow your base repo on CentOS is not copasetic because vorbis along
with all the other packages upgrading is complaining about are
available from the base repo.  Curious, did you upgrade this from
4.0.4 sometime back before going to 4.2.1.  I ask because there was
some monkey business messing w/the centos repos in that release that i
think you are running into.

You're /etc/yum.repos/CentOS-Base.repo should look like the one
attached.  If this works, I'll update the wiki.


CentOS-Base.repo
Description: Binary data
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Re: [sipx-users] ITSP registration expires before reregistration, calls are dropped

[Douglas Hubler]

On Mon, May 2, 2011 at 9:22 AM, Anders Mydland  wrote:
> I am testing sipXecs 4.4 with a Norwegian ITSP.
>
> It appears that re-registrations are not sent from sipXecs  until a few
> milliseconds _after_ the configured register interval.
>
> Consequently, any active  calls are terminated by the ITSP since the
> registration has expired.
>
> I see that a similar issue (along with a NAT issue) was discussed a couple
> of years ago in this thread:
> http://forum.sipfoundry.org/index.php?t=msg&goto=12692&S=30854a8c68e6bc12fc40ffe6c5025437
>
> The problem was supposedly fixed. Has this fix been reverted?

it's not reverted, more likely unrelated.

> I am also seeing a similar issue with the session timer reported in
> http://track.sipfoundry.org/browse/XX-9348
> Could the two be somehow related?

doesn't seem related to me

can you capture a network trace showing only the important sip messages?
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Re: [sipx-users] Need to upgrade 4.0.4 machine and it is running Exim instead of Sendmail

[Douglas Hubler]

On Fri, Apr 29, 2011 at 11:39 PM, Matthew Kitchin (public/usenet)
 wrote:
> I have one 4.0.4 server that I built from scratch. Postfix is my MTA of
> choice, but somehow this machine ended up running Exim.
Exim, really, and sipXecs did that?

>  I'm fine with
> the Sendmail setup in the Sipx ISO install as well. I plan to upgrade
> this server to 4.4. How will it handle the email config? Will it leave
> it alone, install sendmail, or something else?

it does attempt to configure something about sendmail.  i only know
this because there were some problems that were fixed in 4.4 early
alpha.

> I would be happy to get
> it on  a standard Sendmail config like my ISO builds.
> On a side note, is it recommended I got from 4.0.4 straight to 4.4, or
> do I need to go to 4.2.1 first?

i recently assisted in an upgrade from 3.10 to 4.2.1 and ran into this
 http://track.sipfoundry.org/browse/XX-9566
and i'm not sure if upgrading from 4.0.5 will hit this too. i think
step from 4.2.1 to 4.4 is not an issue, therefore I would mostly worry
about the XX-9566 issue and following EACH steps mentioned in the wiki
from 3.10 thru 4.4
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Re: [sipx-users] Bug fixes to 4.4.0 release policy open for comments

[Douglas Hubler]

On Fri, Apr 29, 2011 at 2:16 PM, Tony Graziano
 wrote:
> I think it would then be a good idea to be able to display the
> versions installed in the gui "diagnostics>package list..."

agree so i created
http://track.sipfoundry.org/browse/XX-9575
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Re: [sipx-users] Bug fixes to 4.4.0 release policy open for comments

[Douglas Hubler]

On Fri, Apr 29, 2011 at 11:52 AM, Matthew Kitchin (public/usenet)
 wrote:
> I'm far from an expert at yum, repos, etc. Would it be possible for you
> to do something link maintain a mirror of the CentOS yum repositiry and
> point sipx machines to that repository only? You could then only
> "approve" important patches.

this is technically easy to do and may seems like a good idea, but ...
* What's approved by me may not be approved by you.
* CentOS already does a great job of this and doesn't post updates to
patches that aren't relatively safe to apply.
* I wouldn't want to be the bottleneck on getting an important
security patch out.
* I would just end up just republishing everything CentOS does
* I doubt this wouldn't change one thing in practice as far as admin
update policies
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Re: [sipx-users] Bug fixes to 4.4.0 release policy open for comments

[Douglas Hubler]

On Fri, Apr 29, 2011 at 10:10 AM, Matthew Kitchin (public/usenet)
 wrote:
> Will it be easy/obvious for an user to show exactly what version/fixes
> are installed on their system when they are asking for help?

Addressing this and Tony's similar response.  This is a good point.
The revision number should definitely show up in the web UI. I'm not
sure it does now so that would be a bug.  It does report the build
date though which is pretty good indication.  NOTE: I wrote "revision"
not "version".  So you will see:

  1.2.3-456

However, what's different about the new build system, is that we have
the ability to build a single or select set of packages across all
distros.  This is vital as you try to decide if you want to update
your system, if only there is a single package that needs updating,
you can minimize and contain the impact on your system.  The web ui
only reports the version of the sipxconfig package so this is may not
be the complete picture anymore.

The snapshots alway report every version and release of every package,
so in the end snapshot will be the exact source of what packages have
been updated.

> Is it (will it) be recommended to run 'yum update sipxecs' or 'yum update'?
>
> On a fresh 4.4 install, yum update wants to install quite a bit (see
> below). That is fine with me, but seemed to be frowned on by some in the
> past as I was running version of certain packages that were newer than
> those tested with that version of sipxecs.

This policy doesn't say one way or another, but it is a related point.
We can support all models: from those that never update and those that
always update.  For the record, any testing and building done at ezuce
is with the latest binaries made available at the the time of the test
or build.  This is also a departure from past release engineering
where build machines and possibly test machines contained the
version of binaries at the moment they were installed, which is
essentially random and impossible to reconstruct.

To update all
  yum update
To update sipx
  yum update sipx*
Although in the last comment, this would *not* get sipx deps like
freeswitch or openfire which can be a problem.  A better option would
be to use
  yum --disablerepo=* --enablerepo=sipXecs update


> When building from ISO, I
> have just done kernel and major security updates.

Hmmm, that seems like a good middle ground between security and
stability.  Doesn't this require you to review the changelog for
hundreds for system packages? How do you do this efficiently?

As we review this policy, keep in mind sipXecs has been evolving into
more of a distribution of communication packages over a single
monolithic package with the introduction of freeswitch, openfire,
openacd, etc.   As such, we need to consider the tired-and-true
release policies from distributions as well as the release policies of
a single project.
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[sipx-users] Bug fixes to 4.4.0 release policy open for comments

[Douglas Hubler]

I believe bug fixes like the one George just made to allow conference
pin changes to be made on the fly will get pushed to the 4.4.0 build
sooner rather than later so I put together a release policy open for
comments:

http://wiki.sipfoundry.org/display/sipXecs/Bug+Fix+Release+Policy

This fairly standard stuff and inline with opinions that others have
already shared with me.
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Re: [sipx-users] polycom ring

[Douglas Hubler]

On Thu, Apr 28, 2011 at 9:34 AM, Ben Goodfellow  
wrote:
> I really like the new internal/external alert feature. However how can I 
> change the tone generated for external calls? The ring tone set on the phone 
> affects internal calls, but I cannot figure out how to change for external.

you should be able to, if you don't get any responses, i would google
how polycom implements alert-info in RFC 3261 and post back.
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Re: [sipx-users] polycom ring

[Douglas Hubler]

On Thu, Apr 28, 2011 at 12:10 AM, Jeff Ferrara  wrote:
> Is there any documentation available regarding how to use the new alert-info
> feature?  I had a quick look through the wiki and came up only with the 4.4
> features list.
>
> Also, can anyone think of a way to use this feature to change the ringtone
> of call coming from a ring group / queue?  At present its difficult to
> differentiate between a general inbound call and one coming in via a user's
> direct line.

Feature came from David Becker and we should come up w/some
documentation.  Code is in

  sipXproxy/lib/authplugins/CallerAlertInfo.cpp

And I suspect can be adapted to do anything.
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Re: [sipx-users] where could I download source code of sipxacd-devel-4.2.1 source code?

[Douglas Hubler]

On Thu, Apr 28, 2011 at 2:09 AM, envelopes envelopes
 wrote:
> 1st post on the mail-list. I couldn't find the source code repository.

http://wiki.sipfoundry.org/display/sipXecs/Source

branch release-4.2
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Re: [sipx-users] migration from ant to automake?

[Douglas Hubler]

I meant to put this on dev list.

Thanks for the response and support ;)
On Apr 28, 2011 6:39 AM, "Michal Bielicki" 
wrote:
> +1
> but why is this on the user list ?
>
>
> Am 27.04.2011 um 23:50 schrieb Nathaniel Watkins:
>
>> + 0.25
>>
>> I don’t know what we are talking about…but it sounds important and wanted
to show my support.
>>
>>
>> From: sipx-users-boun...@list.sipfoundry.org [mailto:
sipx-users-boun...@list.sipfoundry.org] On Behalf Of George Niculae
>> Sent: Wednesday, April 27, 2011 5:48 PM
>> To: Discussion list for users of sipXecs software
>> Subject: Re: [sipx-users] migration from ant to automake?
>>
>>
>> On Thu, Apr 28, 2011 at 12:37 AM, Douglas Hubler 
wrote:
>> When trying to add config files for stunnel, I really didn't want to
>> add to the ant mess in neoconf for installing the new config files so
>> i converted much of the install target for neoconf/etc to automake and
>> it's much cleaner IMHO.
>>
>> +1 install should really be an automake matter - don't see any side
effect on converting more (not sure if precommit would be affected though)
>>
>> George
>>
>> This message and any files transmitted with it are intended only for the
individual(s) or entity named. If you are not the intended individual(s) or
entity named you are hereby notified that any disclosure, copying,
distribution or reliance upon its contents is strictly prohibited. If you
have received this in error, please notify the sender, delete the original,
and destroy all copies. Email transmissions cannot be guaranteed to be
secure or error-free as information could be intercepted, corrupted, lost,
destroyed, arrive late or incomplete, or contain viruses. Garrett County
Government therefore does not accept any liability for any errors or
omissions in the contents of this message, which arise as a result of email
transmission.
>>
>>
>> Garrett County Government,
>> 203 South Fourth Street, Courthouse, Oakland, Maryland 21550
www.garrettcounty.org
>> ___
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>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
> Michal Bielicki
> Geschäftsführer / CEO
>
> Seventh Signal Ltd. & Co. KG
> Weigandufer 45, Büro 115, D-12059 Berlin
> Voice: +49 30 60988730
>
> Amtsgericht Charlottenburg HRA 44413 B
> Ust.-ID: DE266981999
> Geschäftsführer: Michal Bielicki
> Persönlich Haftende Gesellschafterin:
> Seventh Signal Ltd, 69 Great Hampton St. Birmingham,
> B18 6EW, GB, Company Nr.: 06889439
> WWW.: http://www.seventhsignal.de
>
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[sipx-users] migration from ant to automake?

[Douglas Hubler]

When trying to add config files for stunnel, I really didn't want to
add to the ant mess in neoconf for installing the new config files so
i converted much of the install target for neoconf/etc to automake and
it's much cleaner IMHO.Would anyone object if i had to convert
more?  Maven i know was a long term goal, but at some point maven
would have to integrate with top-level automake and i see that as
messy as the ant integration was.
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[sipx-users] fixed: jira was not sending out emails

[Douglas Hubler]

I had the email notifier off by accident.  I was missing some great
bug reports against 4.4.0 reported, thanks!
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Re: [sipx-users] sipx 4.4 SSL received a record that exceeded the maximum permissible length.

[Douglas Hubler]

On Wed, Apr 27, 2011 at 9:36 AM, Deon Vermeulen
 wrote:
> Attached is a print screen of what I get.

do you use a corporate proxy?
did you modify the internal apache configs at all?
are all the sipxecs running properly?  What is returned from command
 sipxproc
Did you try restarting sipxecs?
 service sipxecs restart
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Re: [sipx-users] sipx 4.4 SSL received a record that exceeded the maximum permissible length.

[Douglas Hubler]

is this a message reported by your browser?  Do you have a screenshot?

On Wed, Apr 27, 2011 at 4:25 AM, Deon Vermeulen
 wrote:
> Hi All
>
> I installed sipxecs 4.4 last night, and everything worked 100%.
> I could access the web gui and did the services tests, especially for my srv 
> dns records.
>
> This morning I get the following error when trying to access the gui.
>
> SSL received a record that exceeded the maximum permissible length.
>
>
> I've searched on the net but still can't find any information to help me 
> resolve this problem.
>
> Any help will be appreciated.
>
> Thank you
>
> Regards
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>
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Re: [sipx-users] polycom ring

[Douglas Hubler]

On Wed, Apr 27, 2011 at 7:40 AM, Matthew Kitchin (usenet/public)
 wrote:
> When a phone is set to monitor a parking spot by presence/speed dial, it does 
> a little chirp when a call goes into the parking spot. I have to set that 
> line to silent ring. Would it affect that too?
> It would be a hack, but you could upload a silent wav file, and use it for 
> the ringtone, right?

yeah, i probably shouldn't say ringtone, it's really just an
alert-info header added the the INVITE messages.
 ;info=alert-external;x-line-id=0
so I'm sure each polycom could be configured to play a different tone.
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Re: [sipx-users] polycom ring

[Douglas Hubler]

Unfortunately that is bad news. The internal/external ringtone thing is
global setting so one cannot use it if there are any phones configured for
silent ringing.

Is there another polygon setting that can be set to ignore ringtones via
alert info?
On Apr 26, 2011 6:31 PM, "Charles Chalekson"  wrote:
> That took care of it. Thanks!
>
> Is it only for external calls?
>
> If so, there was a feature added to 4.4. to distinguish internal calls
> from external calls by different ringtones. Maybe polycom is taking
> that over silent ring.
>
> You can check/disable these settings here
>
> System/Servers/name-of-server/SIP Proxy/Show Advanced Setting
>
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Re: [sipx-users] attachments gone from track.sipfoundry.org

[Douglas Hubler]

false alarm, drive didn't mount after reboot, added to fstab now and
mounted. enjoy

On Tue, Apr 26, 2011 at 6:03 PM, Douglas Hubler  wrote:
> I will try to restore them from few weeks back.  I'll keep you posted.
>
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[sipx-users] attachments gone from track.sipfoundry.org

[Douglas Hubler]

I will try to restore them from few weeks back.  I'll keep you posted.
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Re: [sipx-users] polycom ring

[Douglas Hubler]

On Tue, Apr 26, 2011 at 12:27 AM, Charles Chalekson  wrote:
> Unsure how this can be related to 4.4 but I just upgraded, and have several 
> polycom 650s.  Prior to upgrade to 4.4, I had several phone lines on my 
> polycoms set to silent ring [no audio ring, flashing light only].  Now with 
> the upgrade, the phone rings when that line is dialed despite that on the 
> polycom phone settings, it still states it is set to silent ring.

Is it only for external calls?

If so, there was a feature added to 4.4. to distinguish internal calls
from external calls by different ringtones.  Maybe polycom is taking
that over silent ring.

You can check/disable these settings here

 System/Servers/name-of-server/SIP Proxy/Show Advanced Settings
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Re: [sipx-users] IM client does not show presence

[Douglas Hubler]

2011/4/26 Sven Evensen :
> We are trying out the Pidgin client for corporate IM. We also want to pick
> up phone presence, at least
>
> the users who have proper desk phones. But if a user logs off their IM
> client (thus showing status Offline),
>
> the “On the phone” presence does not work anymore.

someone can confirm this, but I think it's a known issue.  i cannot
seem to find tracker item though.

I recently tried to figure out how unified presence to was designed
and it looked very complicated. The openfire plugin had the bulk of
the code along w/registrar plugin.  Here's part of the puzzle

  http://wiki.sipfoundry.org/display/sipXecs/sipXopenfire+architecture

Having an EXACT value of the EXACT state of a phone/user at all times
is also incredibly important to many parts of the system.  So it lead
to conversation about looking into AMQP so various parts of the system
can help determine a user's state.
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Re: [sipx-users] Conferencing in 4.4 problems

[Douglas Hubler]

On Mon, Apr 25, 2011 at 2:19 PM, George Niculae  wrote:
> On Mon, Apr 25, 2011 at 9:10 PM, Burleigh, Matt
>  wrote:
>>
>> Understood, but if a user changes his PIN in the middle of the night,
>> they’ll have to wait until I notice that the restart needs to happen. That
>> sucks.
>
> Agree, really annoying, please file a JIRA

George, is this a regression?  if so, how did it work before?
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[sipx-users] sipfoundry sites back up.

[Douglas Hubler]

track/wiki/www.sipfoundry.org are back online.  www is missing a few
images, but I've let the right people know about that.  let me know if
you have any issues.  sorry for the outage.
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Re: [sipx-users] SIPFoundry download site is available

[Douglas Hubler]

On Thu, Apr 21, 2011 at 10:01 PM, Outback Dingo  wrote:
> isnt there an install iso someplace ?

http://download.sipfoundry.org/pub/sipXecs/ISO/
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[sipx-users] SIPFoundry download site is available

[Douglas Hubler]

It actually never went down
  http://download.sipfoundry.org/pub/sipXecs/
It is on a different host.
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Re: [sipx-users] sipfoundry download site down?

[Douglas Hubler]

On Thu, Apr 21, 2011 at 1:17 PM, Josh M. Patten  wrote:
> Perhaps this is why: 
> http://slashdot.org/story/11/04/21/1515238/Major-Outage-At-the-Amazon-Web-Services

This is exactly why.  It started w/freezing when accessing the EBS
volumes and now i cannot even take a snapshot of the volume to start a
different server in a different zone.  We have new servers somewhat
ready but not very useful if i cannot get my latest production data
out.

I've used AWS for last 4 years and never had a problem like this or
even in this very zone before.
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Re: [sipx-users] sipfoundry download site down?

[Douglas Hubler]

it's down and does not look good.  system is wedged, won't even reboot
properly.  it may take a while to get it back online, i'll keep folks
posted

On Thu, Apr 21, 2011 at 8:48 AM, Tony Graziano
 wrote:
> Was trying to test an update to 4.4. Can't reach the site.
>
> --
> ==
> Tony Graziano, Manager
> Telephone: 434.984.8430
> sip: tgrazi...@voice.myitdepartment.net
> Fax: 434.326.5325
>
> Email: tgrazi...@myitdepartment.net
>
> LAN/Telephony/Security and Control Systems Helpdesk:
> Telephone: 434.984.8426
> sip: helpd...@voice.myitdepartment.net
>
> Helpdesk Contract Customers:
> http://support.myitdepartment.net
> Blog:
> http://blog.myitdepartment.net
>
> Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
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Re: [sipx-users] Open ACD (Call Center) - not showing all the tabs?

[Douglas Hubler]

On Wed, Apr 20, 2011 at 7:22 PM, Nathaniel Watkins
 wrote:
> I thought those screen shots were from sipXecs…now I realize those are
> openACD screens J

However, all is not lost. You can safely edit queues, recipes, etc. on
the admin interface on port  that comes with OpenACD.  Do not
however edit agents in OpenACD admin interface otherwise your systems
will be out of sync.  If the admin interface on port  is acting
flakey, try logging out, closing web browser and logging back in.
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[sipx-users] sipXecs 4.4.0 Officially Released

[Douglas Hubler]

Finally!  Thanks to everyone for making this happen

  http://download.sipfoundry.org/pub/sipXecs/

Instructions
  http://wiki.sipfoundry.org/display/sipXecs/Upgrade+to+Latest+Stable+Version

Going forward, there will be updates to this build for bug fixes as
they are fixed.  I will publish a changelog so you can decide if and
when you want to update.  I have not decided how to publish this log,
but I'll post back here when I determine the details.
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Re: [sipx-users] Basic OpenACD questions on 4.4

[Douglas Hubler]

On Tue, Apr 19, 2011 at 4:37 PM, Jim Canfield  wrote:
> Still not sure how the endpoint settings should be setup though, when
> I go "available" my phone rings, but i cannot answer, it just gets
> stuck in limboland.  After it's stuck, I see the agent in queue with
> timer running even after the call is ended.  Only way to clear is
> restart media services.  Is this a polycom thing again?

- Can you elaborate with more technical details on "cannot answer".
- Can you review logs for errors.
  /var/log/sipxpbx/openacd/*.log
  /var/log/sipxpbx/freeswitch.log
- Once you log into agent console on port 5050, try a ring test. Does
your phone ring and do you hear few seconds of silence.
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Re: [sipx-users] Openfire sip plugin

[Douglas Hubler]

On Wed, Apr 20, 2011 at 7:59 AM, Tony Graziano
 wrote:
> to what end? openfire sip plugin will register, but it cannot transfer
> calls or make insternal calls either (does not support refer)

Good question, Darth, what is your desired goal?


> On Wed, Apr 20, 2011 at 7:44 AM, Darth Zejdr  wrote:
>> Hi
>>
>> I'm trying to get openfire sip plugin running on sipx server. The
>> Problem is, i can't fins plugin version 1.0.5, and 1.0.6 works only with
>> openfire 3.7.0.
>>
>> Has anyone tried updating openfire to 3.7.0 and does it work?
>>
>> Also if someone has sip.jar 1.0.5 version, could you mail it to me?

Only info i have on the openfire upgrade is that the authentication
*may* not support DNS SRV because it would use a newer version of the
smack library.

  http://community.igniterealtime.org/thread/42976

But the upgrade of openfire has to happen at some point and so I'd
encourage your effort to do so.
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Re: [sipx-users] Basic OpenACD questions on 4.4

[Douglas Hubler]

On Tue, Apr 19, 2011 at 3:37 PM, Michael Scheidell
 wrote:
> On 4/19/11 3:20 PM, Douglas Hubler wrote:
> If you make someone a supervisor, login to interface on port 5050 then add
> supervisor dashboard you should see all sorts of info.

sorry, this is not clear at all !  Let me restate this:
 - Change an agent's Security level to "SUPERVISOR" as found in the
Call Center Agent's Edit page.  Do *not* make the user superadmin
privledges in the User Edit page.
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Re: [sipx-users] Basic OpenACD questions on 4.4

[Douglas Hubler]

If you make someone a supervisor, login to interface on port 5050 then add
supervisor dashboard you should see all sorts of info.

The format for the commands is quasi freeswitch but on your initial tests,
ignore those commands and log into interface on port 5050 and go available
from there to take calls.
On Apr 19, 2011 2:53 PM, "Jim Canfield"  wrote:
> Hi folks,
>
> Wanted to start getting familiar with openACD this morning and I have
> a couple basic questions:
>
> 1) Where do you see queue status?
>
> 2) When writing commands, what is the proper format?  For example the
> wiki has the following for "Login":
>
> agent_dialplan_listener testme@fedorabox agent_login
> ${sip_from_user} pstn ${sip_from_uri}
>
> If my hostname is "sipx" and the what would the user side be?
>
> agent_dialplan_listener whatgoeshere@sipx agent_login
> ${sip_from_user} pstn ${sip_from_uri}
>
>
> So far, the acd line answers and the caller is put in queue, but I'm
> still at a loss when it comes to logging into the queue and answering
> calls.
>
> Thanks,
>
> --Jim
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Re: [sipx-users] hairpinned calls? media?

[Douglas Hubler]

Ugh, this should have been made available from the UI.  Sven,
hopefully you can get by, by editing your customer configs?

I created this...

 http://track.sipfoundry.org/browse/XX-9562

and selected for 4.6

On Fri, Apr 15, 2011 at 10:54 AM, Sven Evensen  wrote:
> Media release (or not anchoring media) works fine when using unmanaged SIP
> trunk. But when you use sipxbridge,
>
> that does not work.
>
>
>
> Ranga did a fix to this about a year ago and here is from that email:
> (18-apr-2010)
>
>
>
> By default  sipxbridge/sipxrelay will ALWAYS anchor media.
>
>
>
> There is a flag which is not supported via sipxconfig ( i.e.
>
> experimental flag ) which will allow you to remove the media anchor
>
> for forwarded calls.
>
>
>
> It is called   Look in the sipxbridge.xsd file for
>
> a description. By default that flag is true and it is hidden. You can
>
> set it to false by editing sipxbridge.xml and if you do so, then for
>
> hairpinned forwarded or blind transferred calls, the media relay will
>
> be removed from the media path AFTER transfer.
>
>
>
> We really want thus feature, as most of our servers are in the cloud, but
> have not had time to test it yet.
>
>
>
> Keep us updated how it goes!
>
>
>
> Regards,
>
> Sven
>
>
>
> 
>
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael
> Scheidell
> Sent: 15 April 2011 15:06
> To: sipx-users@list.sipfoundry.org users
> Subject: [sipx-users] hairpinned calls? media?
>
>
>
> I seem to remember a 'tweak/hack' that could be done to the itsptrunkxml
> file to keep the rtp on the itsp side if both inbound and outbound of the
> call were to/from the same itsp.
>
> is this something I remember wrong?
>
> if not, what is it, and where is the file I hack?
>
> --
> Michael Scheidell, CTO
> o: 561-999-5000
> d: 561-948-2259
> ISN: 1259*1300
>> | SECNAP Network Security Corporation
>
> · Best Intrusion Prevention Product, Networks
> Product Guide
>
> · Certified SNORT Integrator
>
> · Hot Company Award, World Executive Alliance
>
> · Best in Email Security, 2010 Network Products
> Guide
>
> · King of Spam Filters, SC Magazine
>
>
>
> 
>
> 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] Polycom custom config issues in 4.4?

[Douglas Hubler]

On Fri, Apr 15, 2011 at 3:15 AM, Nathaniel Watkins
 wrote:
> Is anyone having trouble using Eric & Josh's Custom polycom config - 
> http://sites.google.com/site/sipxecstipsandtricks/polycom-phones

This is how 4.4 implemented the custom configs
 http://article.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/32740
making the custom polycom config patch unnec.  Please try it out and
tell me if you have any issues.
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Re: [sipx-users] Yumming from 4.2.1 to 4.4 beta

[Douglas Hubler]

On Fri, Apr 15, 2011 at 11:42 AM, Tony Graziano
 wrote:
> Seeing a lot of dependency issues... I also don't think this is hard
> to overcome for some users, but wanted to share it.
> ***
> --> Finished Dependency Resolution
> sipxecs-4.2.1-18591.9.1.i386 from installed has depsolving problems
>  --> Missing Dependency: sipx-freeswitch-codec-passthru-g729 is
> needed by package sipxecs-4.2.1-18591.9.1.i386 (installed)

I've not seen this on my test update and doesn't make a lot of sense
because sipxecs-4.2.1 should get selected for uninstalling.  In a
subsequent email you said upgrade is on and that ISO is wrong.  So is
this still happening?
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Re: [sipx-users] Restoring backups from 4.2.1 to 4.4.0

[Douglas Hubler]

On Sat, Apr 16, 2011 at 11:59 AM, Tony Graziano
 wrote:
> Fyi - last ISO 4.4 does not run sipxconfig web interface...

32 or 64 bit?  just tried 64 bit ISO and didn't have any issues w/ui
coming up.

I'm rebuilding now because we want to get the latest fixes in.
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Re: [sipx-users] Ubuntu sipXecs Repository

[Douglas Hubler]

No binaries published for ubuntu. Best luck is googling for compiling from
source
On Apr 16, 2011 2:46 PM, "Aza Tek"  wrote:
> Is there a stable repo for sipxecs 4? I'm using Ubuntu 10.10.
>
> Thanks for sipXecs...
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Re: [sipx-users] official 4.4.0 release anticipated this friday

[Douglas Hubler]

Sorry, /etc/yum.repos.d is the right dir. I'll correct it when I get a
chance unless someone beats me to it.
 On Apr 13, 2011 8:06 AM, "Claas Hilbrecht" <
claas.hilbrecht+maillinglists.sipx...@linum.com> wrote:
>> I refactored the upgrade instructions extensively to make upgrade
>> process straightforward as possible
>>
>> http://wiki.sipfoundry.org/display/sipXecs/Upgrade+to+Latest+Stable+Versi
>> on
>
> Just a note, in the section "Before upgrading from 4.2.x" the command "rm
> sipxecs-*.repo" has no information attached WHERE to remove those files.
> And searching on my sipXecs 4.2.1 system I found no "sipxecs-*.repo"
files.
> Just the /etc/yum.repos.d/sipxecs.repo I probably should remove. Since I
> run debian system everywhere else I'm not 100% sure if the
> /etc/yum.repos.d/sipxecs.repo has to be removed or not.
>
> --
> Claas Hilbrecht
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[sipx-users] official 4.4.0 release anticipated this friday

[Douglas Hubler]

If all goes well. Lingering issues that need to be cleaned-up, but
this is mostly a waiting period for any new issues at last chance
fixes to come in.

I refactored the upgrade instructions extensively to make upgrade
process straightforward as possible
  http://wiki.sipfoundry.org/display/sipXecs/Upgrade+to+Latest+Stable+Version

P.S. The dragging out of the release of 4.4.0 is related to getting
the release process down. Each release phase should be significantly
faster. Thanks for hanging in there.  Folks that are already running
4.4.0 thank you for your help and a simple update once 4.4.0 is
released is all that you need to do at your convenience.

P.P.S. 4.6.0 is not too far behind (several months?) 4.4.0 due to
specific eZuce needs.  This is subject to change, but unlikely.
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Re: [sipx-users] Openfire admin console on 4.2.1

[Douglas Hubler]

On Fri, Apr 8, 2011 at 4:35 PM, Jim Canfield  wrote:
> Can anyone confirm that the Openfire admin access is broken on 4.2.1
> http://wiki.sipfoundry.org/display/sipXecs/The+Openfire+admin+console
> This procedure no longer seems to work.

do you have any details on what didn't "work"?
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Re: [sipx-users] Design proposal for http://track.sipfoundry.org/browse/XX-9554

[Douglas Hubler]

What if you combine Replication and Status columns with 3 images: red,
yellow, green

[red]  errors registering
  details >>

[yellow]  registered
  details >>

[green] - configured
  details >>


On Fri, Apr 8, 2011 at 3:34 PM, Mircea Carasel  wrote:
> Hi,
>
> I opened a new JIRA ticket: XX-9554 that addresses the server replication.
> We would like to offer more consistent information to the user, to have an
> overall image on how send profiles went for each server
> Please study the attached screenshot
>
> 1. I propose a new column to the server table called: Replication, that will
> contain information about replication on each of the configured servers
> 2. The column will contain:
>  a) an Image that will represent the replication status: green bullet if
> replication was successful, red bullet if something failed
>          During "send profiles" running we will show instead of the image a
> progress bar  to reflect that replication is in running state
>  b) a collapsible panel that will show:
>  - date/time when last replication was performed
>          - in case of replication error, show files that were not replicated
>          -show errors that determined replication failure (disk full,
> XML-RPC problem, certificate handshake problem, etc...)
>
> Opinions welcomed!
>
> Thanks,
> Mircea
>
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[sipx-users] jira (track.sipfoundry.org) overhaul

[Douglas Hubler]

There are 1000+ issues/feature requests/improvements/tasks in jira in
various states of being valid.  They are also are not in any order of
importance.  I'm going to be cleaning up jira and defining a process
that will make jira more useful as a tool for us. I have a pretty good
idea of what needs to be done, essentially major simplification of the
forms/fields, and coming up w/a policy to keep it useful as a tool
going forward.  ezuce has offered a few days for their engineer's time
after 4.4 is released to jira cleanup of issue reports.  This will
happen after I get jira in a state where we can get ahead of it.   I
have imported jira data into a new staging server as I practice the
clean up.  Those that have an interest in reorg of the bug tracker,
feedback welcomed.
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Re: [sipx-users] ERR in sipregistrar log

[Douglas Hubler]

FYI: Just wanted to let folks know, I'm listening even if i don't have
the time to address each issue.

As you can imagine, sparse documentation makes things very difficult
for ezuce engineers as well so we feel your pain too. Good news is
that now that the wiki is organized, ezuce is starting to get more
stuff up there, but it takes time.

I would encourage folks to think of the wiki as your own.  There are
cookbook sections that are really meant for semi-structured notes as
you dig into something.  They in-turn can be harvested for more
structured info elsewhere in the wiki. Additionally, they can save
others a lot of time.

back to the topic... I once posted the idea of embedding sip viewer
right into admin ui, where one could drill into calls by user,
destination, exactly as you describe here.  I got very little interest
but maybe it was just bad timing on my post.

On Thu, Mar 31, 2011 at 3:20 PM, Jeff Gilmore  wrote:
> +1 on this approach!
>
> On Mar 31, 2011, at 12:50 PM, Matt White wrote:
>
> Nathaniel Watkins  03/31/11 11:41 AM >>>
 I'm sure it wouldn't be hard for someone to write some code that would 
 give you an overview of the recent calls from the >>logs - you select one 
 of the calls you want to trace - and hit generate trace/etc button...which 
 then lets you download the >>xml file stright to your browser...
>>
>>
>>
>>
>> I've always though the call records page should have an advanced column that 
>> list the calls call-id (which is the most useful string to use with 
>> sipx-trace).  I would easily regain at least 2 weeks of my life over the 
>> course of a year in search for a failed calls id ;-)
>>
>>
>>> From there it wouldn't be hard to make that call-id clickable to 
>>> autogenerate the xml trace file and pop for the user to open with sipviewer.
>>
>>
>> But I don't think the call-id is stored in the CDR tables.
>>
>>
>> -M
>>
>>
>>
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Re: [sipx-users] stun01.sipphone.com is gone...

[Douglas Hubler]

On Tue, Mar 29, 2011 at 10:55 PM, Joegen Baclor  wrote:
> Since the broken STUN server has bitten someone again today, there is an
> urgent need for an interim solution to correct this in 4.6.  At the very
> least we should avoid the surprises brought about by the default STUN server
> which nobody might suspect to even be there if they've never had prior
> problems with it after months of being in production.   Here is the interim
> proposal and I would love to hear some comments.
>
> Version <= 4.4
>
> 0.  So as not to open up new doors for bugs to come in for 4.4 and below, it
> makes sense to just vote for a replacement value.  I declare the floor open
> for this.

counterpath seems fine.  sipfoundry is using nettica for hosting DNS
and I've found it to be very reliable over the last 3 years using it
so I'm all for setting DNS records in sipfoundry's DNS to point to
counterpath should we need to change this.

I researched and stun4j used by sipxbridge. It does not seem to
support SRV records out of the box.


> -
>
> Version 4.6
>
> 1.  Add a new entry "No NAT" in the "Address Type" field and have it as the
> default.  This address type would mean that sipXconfig would insert the
> internal address as the value of the external address.  This would mimic the
> behavior of "Use STUN" in cases where sipXecs is not sitting behind a NAT
> where STUN would yield the same value for the external address

We might find time would be better spent on redesigning how NAT is
configured today instead of tacking something on to what we have.
Biggest issue I have (beyond that it's configure in 3 different
places) is that when it breaks, folks didn't even know they were using
it.  Your suggestion doesn't really address that, just provides more
options once your at the UI.  Also, I could be wrong, but enough (if
not most) sipxecs are behind a NAT so default behavior to silently not
work may be worse than what we have now.

> 2.  Change the Description of "Use STUN" to include a list of known STUN
> servers.  The user may simply copy and paste a server to their liking.  This
> is chosen over having a list-box of known STUN servers to avoid further
> confusion that the ones listed are the only working/compatible servers with
> sipX.

Once we redesign, i think this would be a great option on the ultimate
text field.
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Re: [sipx-users] Sipx 4.4.0, failed external forward, Verizon VoIP

[Douglas Hubler]

On Tue, Mar 29, 2011 at 9:45 PM, Matthew Kitchin (public/usenet)
 wrote:
> The notes next to that setting indicate:
> "Set the From header to ITSP account information for outgoing INVITE
> messages. If you need to set this, ITSP is not following RFC and should be
> notified to this fact."
> I understand the general concept of what is happening here, but not all the
> details. Is there a particular stock explanation, rfc, etc I should send to
> Verizon?

AFAIK: verizon should be using the Contact header, not the From.  From
is informational and not to be used in routing.  You'll see the
contact header is correct in both cases.

> I'm going to assume this setting/issue is the same reason the caller ID of
> the actual calling party doesn't show up on the call after it has been
> forwarded?

not sure.

> Thanks again. Do you want to close the jira or should I?

closed
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Re: [sipx-users] Sipx 4.4.0, failed external forward, Verizon VoIP

[Douglas Hubler]

On Tue, Mar 29, 2011 at 9:05 PM, Matthew Kitchin (public/usenet)
 wrote:
> Is it significant that it is showing different numbers? 4.2.1 is showing the
> cell phone I'm calling in on. 4.4 is showing the DID alias for the extension
> that is being forwarded.
> As for the ITSP setting, is that 4.4 only? I don't see it in 4.2.1. I will
> swap over and try it on the 4.4 machine if that is what you are looking for.
> It is currently set to whatever the default is, and I do not use a ITSP
> template.

It may be new in 4.4, for you, it appears you should have it  - Unchecked.
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Re: [sipx-users] Sipx 4.4.0, failed external forward, Verizon VoIP

[Douglas Hubler]

Only difference I found was

WORKS (4.2.1)
=
From: "WIRELESS CALLER" ;tag=8452481748283643453

DOESN'T WORK (4.4)

From: "WIRELESS CALLER" ;tag=1035674599848618864

In ITSP Account advanced settings, do you have  this set?
  INVITE From ITSP Account:  X


On Tue, Mar 29, 2011 at 8:29 PM, Matthew Kitchin (public/usenet)
 wrote:
> On 3/29/2011 6:55 PM, Douglas Hubler wrote:
>>
>> On Tue, Mar 29, 2011 at 5:36 PM, Matthew Kitchin (public/usenet)
>>   wrote:
>>>
>>> On 3/29/2011 4:11 PM, Michael Scheidell wrote:
>>>
>>> On 3/29/11 4:41 PM, Douglas Hubler wrote:
>>>
>>> Obvious answer is that there are no credentials, sipxecs is never
>>> challenged by verizon.  In addition, sipxecs never challenges on the
>>> original incoming INVITE either.   The original SIP trace is suspect,
>>> can you (or anyone) think of any reason why "407 Proxy Authentication
>>> Required" would be missing from your original message?
>>> ___
>>>
>>> verizon is doing ip based authentication?
>>>
>>> Yes. I guess I should have picked up on that. I do not
>>> authenticate/register
>>> with Verizon.
>>
>> I still need to see a wireshark trace from your 4.2.1 system that
>> exhibits the full scenario that works.
>
> It is attached. The only difference is the forward destination number is now
> 6155914780, because I'm at home now instead of the office. This is a
> wireshark capture of a successful forward on 4.2.1. It is the same sip trunk
> that was used for the previous captures on the 4.4 machine.  The calling
> number is 6155008073. I dialed 6159253938. That alias is assigned to
> extension 200. Extension 200 has no handset assigned to it and is set to
> forward to 6155914780. It works. Let me know if there is anything else I can
> tell you.
>
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Re: [sipx-users] Sipx 4.4.0, failed external forward, Verizon VoIP

[Douglas Hubler]

On Tue, Mar 29, 2011 at 5:36 PM, Matthew Kitchin (public/usenet)
 wrote:
> On 3/29/2011 4:11 PM, Michael Scheidell wrote:
>
> On 3/29/11 4:41 PM, Douglas Hubler wrote:
>
> Obvious answer is that there are no credentials, sipxecs is never
> challenged by verizon.  In addition, sipxecs never challenges on the
> original incoming INVITE either.   The original SIP trace is suspect,
> can you (or anyone) think of any reason why "407 Proxy Authentication
> Required" would be missing from your original message?
> ___
>
> verizon is doing ip based authentication?
>
> Yes. I guess I should have picked up on that. I do not authenticate/register
> with Verizon.

I still need to see a wireshark trace from your 4.2.1 system that
exhibits the full scenario that works.
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Re: [sipx-users] Sipx 4.4.0, failed external forward, Verizon VoIP

[Douglas Hubler]

On Tue, Mar 29, 2011 at 4:35 PM, Douglas Hubler  wrote:
> On Tue, Mar 29, 2011 at 3:38 PM, Douglas Hubler  wrote:
>> On Tue, Mar 29, 2011 at 3:16 PM, Matthew Kitchin (public/usenet)
>>  wrote:
>>> Sure. Merged logs are attached. This is a call from 6159253938 (sipx 4.4
>>> test) to 6154670142 (sipx 4.2.1 production). The 2 systems are on seperate
>>> Verizon sip trunks.
>
> I'm no expert so here are the final messages, maybe someone has an idea.
> Also: It occurred to me we really need one more trace:  call
> forwarding working to verizon from your 4.2.1 system.
>
> DOESN'T WORK (far-end ultimately send CANCEL after 20 seconds.
> Generated from call forwarding)
> 
> INVITE sip:6154670142@172.30.216.62;sipx-noroute=Voicemail;user=phone SIP/2.0
> Call-ID: BW102147333250311467067431@63.77.76.250-1
> CSeq: 1 INVITE
> From: "WIRELESS CALLER" ;tag=1035674599848618864
> To: 
> Via: SIP/2.0/UDP
> 10.81.3.5:5080;branch=z9hG4bKaeb1e386274092e542c00ac082f20b35333432
> Max-Forwards: 70
> User-Agent: sipXecs/4.4.0 sipXecs/sipxbridge (Linux)
> P-Asserted-Identity: "WIRELESS CALLER" 
> 
> Contact: 
> Route: 
> Session-Expires: 1800;refresher=uac
> References: 
> BW102147333250311467067431@63.77.76.250-0;rel=chain;sipxecs-tag=request-invite-z9hg4bk-xx-a7c7hxqzqrk5`emcdlkjkxchrq
> Allow: INVITE,BYE,ACK,CANCEL,OPTIONS
> Supported: timer
> Content-Type: application/sdp
> Content-Length: 185

Obvious answer is that there are no credentials, sipxecs is never
challenged by verizon.  In addition, sipxecs never challenges on the
original incoming INVITE either.   The original SIP trace is suspect,
can you (or anyone) think of any reason why "407 Proxy Authentication
Required" would be missing from your original message?
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Re: [sipx-users] Sipx 4.4.0, failed external forward, Verizon VoIP

[Douglas Hubler]

On Tue, Mar 29, 2011 at 3:38 PM, Douglas Hubler  wrote:
> On Tue, Mar 29, 2011 at 3:16 PM, Matthew Kitchin (public/usenet)
>  wrote:
>> Sure. Merged logs are attached. This is a call from 6159253938 (sipx 4.4
>> test) to 6154670142 (sipx 4.2.1 production). The 2 systems are on seperate
>> Verizon sip trunks.

I'm no expert so here are the final messages, maybe someone has an idea.
Also: It occurred to me we really need one more trace:  call
forwarding working to verizon from your 4.2.1 system.

DOESN'T WORK (far-end ultimately send CANCEL after 20 seconds.
Generated from call forwarding)

INVITE sip:6154670142@172.30.216.62;sipx-noroute=Voicemail;user=phone SIP/2.0
Call-ID: BW102147333250311467067431@63.77.76.250-1
CSeq: 1 INVITE
From: "WIRELESS CALLER" ;tag=1035674599848618864
To: 
Via: SIP/2.0/UDP
10.81.3.5:5080;branch=z9hG4bKaeb1e386274092e542c00ac082f20b35333432
Max-Forwards: 70
User-Agent: sipXecs/4.4.0 sipXecs/sipxbridge (Linux)
P-Asserted-Identity: "WIRELESS CALLER" 
Contact: 
Route: 
Session-Expires: 1800;refresher=uac
References: 
BW102147333250311467067431@63.77.76.250-0;rel=chain;sipxecs-tag=request-invite-z9hg4bk-xx-a7c7hxqzqrk5`emcdlkjkxchrq
Allow: INVITE,BYE,ACK,CANCEL,OPTIONS
Supported: timer
Content-Type: application/sdp
Content-Length: 185

v=0
o=sipxbridge 4725027341631093188 1 IN IP4 10.81.3.5
s=-
c=IN IP4 10.81.3.5
t=0 0
m=audio 30502 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20


WORKS (Enters ringing state.  Straight INVITE thru sipXbridge)

INVITE sip:6154670142@4.sipxt.voipt;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.81.3.254;branch=z9hG4bK661f71454BF4BD84
From: "ID: 272" ;tag=D012051C-AE0C1E7
To: 
CSeq: 2 INVITE
Call-ID: 5c69e470-9d6b90cb-328ec102@10.81.3.254
Contact: 
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_450-UA/3.2.4.0244
Accept-Language: en
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest
username="~~id~sipXprovision/0004f22cea3b", realm="4.sipxt.voipt",
nonce="ab6954891c162b30f6caa5632465a72f4d92389c", qop=auth,
cnonce="vRSCyQc4L5i5Pgy", nc=0001,
uri="sip:6154670142@4.sipxt.voipt;user=phone",
response="fe2f380662f28c73bcf87149c81d7be4", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 292

v=0
o=- 1301426889 1301426889 IN IP4 10.81.3.254
s=Polycom IP Phone
c=IN IP4 10.81.3.254
t=0 0
a=sendrecv
m=audio 2258 RTP/AVP 9 0 8 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
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Re: [sipx-users] Sipx 4.4.0, failed external forward, Verizon VoIP

[Douglas Hubler]

On Tue, Mar 29, 2011 at 3:16 PM, Matthew Kitchin (public/usenet)
 wrote:
> Sure. Merged logs are attached. This is a call from 6159253938 (sipx 4.4
> test) to 6154670142 (sipx 4.2.1 production). The 2 systems are on seperate
> Verizon sip trunks.
> Would you like to see a wireshark capture as well?

This is missing the INVITE, something went wrong w/the capture.  Why
don't you send the wireshark trace.  In the Wireshark Capture filter
(not the Wireshark Display Filter) specify:

  port sip and host 10.81.3.5

This will capture all SIP traffic in/out of that vm
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Re: [sipx-users] Sipx 4.4.0, failed external forward, Verizon VoIP

[Douglas Hubler]

On Fri, Mar 25, 2011 at 11:30 AM, Matthew Kitchin (public/usenet) <
mkitchin.pub...@gmail.com> wrote:

> The wireshark shows it pretty clear to me, if I"m reading it correctly. It
> is attached, and I will attach it to the jira to.
> Call is from 6155008073 to 6159253938. It should forward to 6154670142.
> Sipx server is at 10.81.3.5. 172.X is verizon voip servers. It seems to be
> that sipx is canceling the call. Maybe I am missing something.
>

Actually, you're definitely missing something ;)   The  CANCEL you see in
that sip trace from sipx was just a CANCEL that was originally sent from
Verizon in the re-INVITE fork because verizon could not contact the final
callee.  You can see two voip call flows in the trace you sent.

The CANCEL from Verizon comes after 20 seconds so they probably had trouble
delivering the re-INVITE.  I think we need to compare a straight INVITE
when 6159253938 calls 6154670142  with the SIP trace you already sent that
didn't work.  Can you capture that and send it to me on or off list?

I also verified
- (as others have) that call forwarding is fine w/at least voip.ms in 4.4.
- the code that was put into 4.2.1 to fix your original issue on 4.0.4 is in
the 4.4.0 code base, so chances are this is a new problem.
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Re: [sipx-users] All calls were failing... why? stun01.sipphone.com is gone...

[Douglas Hubler]

On Tue, Mar 22, 2011 at 4:44 PM, Matt White  wrote:
>
>
 On 3/22/2011 at 04:19 PM, in message <4d890466.6060...@secnap.com>, Michael
> Scheidell  wrote:
>> where are you specifying an stun server?  what version of sipx is this?
>> I didn't think sipx (4.+) used an stun server for anything.
>>
>
> STUN is the default for determining the public NAT address in 4.x
>
> Its set under the NAT screen.  But usually with entering a static IP is much 
> safer.  How runs sipx behind a NAT of a dynamic ip?
>
> I would entire a jira  so this STUN server is removed from future versions.  
> Perhaps all stun servers should be removed and just allow the user to enter 
> in a stun server.

+1
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Re: [sipx-users] Cordless wifi - Spectralink 8020 up and running - need some help

[Douglas Hubler]

Questions:
 Can explain the simplest scenario that fails?
 Explain exactly what "external calls" is? incoming, outgoing or both
 Does the scenario start with a call between a polycom hardphone and
spectralink registered as two different extensions?


On Mon, Mar 21, 2011 at 12:39 PM, Matthew Kitchin (public/usenet) <
mkitchin.pub...@gmail.com> wrote:

>  * Polite bump request. *
> If there is anyone who could please help me with the issue below, I would
> greatly appreciate it. I can open a paid support ticket if that is a
> possible route.
>
>  Original Message   Subject: Re: Spectralink 8020 up and
> running - need some help  Date: Thu, 17 Mar 2011 10:41:07 -0500  From: Matthew
> Kitchin (public/usenet) 
>   To:
> sipx-users 
>
> The wireshark capture can be downloaded here:
> http://www.dsi-corp.com/mk/transfer030811.zip
>
> On 3/17/2011 10:33 AM, Matthew Kitchin (public/usenet) wrote:
>
> I post the email below 2 days ago, but I'm thinking it didn't make it to
> the list because the size was 710 kb. Is there anyone out there who will
> give me a hand with this? I think it could be helpful for many who are
> looking to use these phones. I will try and find a place to upload the the
> file.
>
>  Original Message   Subject: Re: [sipx-users] Spectralink
> 8020 up and running  Date: Tue, 15 Mar 2011 13:33:58 -0500  From: Matthew
> Kitchin (public/usenet) 
>   To:
> Discussion list for users of sipXecs software
>  
>
> Update, and I need some help.
> I have been working with polycom and tested several new firmwares and we
> have troubleshot together. The only remaining issue I still have is
> transfers with external calls.
> Their preferred method of diagnosing is wireshark to capture at the wifi
> level (yes, that has made it a little more tedious for us). That is what I
> am attaching, because that is what their notes below refer too.
> I'm showing them that blind transfers work with other phones, especially
> Polycom Soundpoint phones, but they still asked if I could shed some lite on
> their observations.
> They are being extremely helpful, responsive, interested, and turning out
> updated firmware within a day or 2 every time I give more captures of what
> went wrong. We have at least fixed it so these phones do work (obviously
> with the exception of transfers) with sipx 4.2.1.
> In the attached pcap file, it encrypted with WPA2 and will need to be
> decrypted within Wireshark (I don’t know of any way to save a decrypted
> capture). The ssid is DSI-VOIP and the passphrase is ‘adadadadad’. So you
> can enter ‘wpa-pwd:adadadadad:DSI-VOIP’ as the key under
> Edit/Preferences/Protocols/IEEE 802.11 in Wireshark and enable decryption.
> The info for the call is:
> Spectralink 8020: Ext , IP=10.82.20.223
> Sipx 4.2.1 server: 10.87.20.5
> Verizon Voip, sipxbridge, No IP NAT
>
> The capture is an outbound call placed from the Spectralink 8020 and then a
> failed transfer/
> If anyone can help me explain what is or isn't happening, I would greatly
> appreciate it and I am confident we can get a wifi phone that is 100%
> compatible with sipx and made by polycom.
>
> Thanks,
> Matthew
>
> Polycom Spectralink engineer notes:
> This looks like a problem with the SIPX server.  Basically, the SIPX is
> never sending us the 200 OK for the transfer (INVITE 104):
>
> 2152 send INVITE 102 from  to 95171139 gets 407 and Acks that (gets
> another and Acks that too)
>
> 2195 send INVITE 103 from  to 95171139
> 2203 recv 100 to INVITE 103
> 2653 recv 183 to INVITE 103
>
> 3652 recv 200 ok for INVITE 103
> 3747 send ACK 103
>
> 
>
> 3795 recv another 200 ok for INVITE 103
> 3859 send another ACK 103
>
> 4074 recv another 200 ok for INVITE 103
> 4139 send another ACK 103
>
> 4636 recv another 200 ok for INVITE 103
> 4697 send another ACK 103
>
> 5690 recv another 200 ok for INVITE 103
> 5749 send another ACK 103
>
> 6046 send INVITE 104 from  to to 95171139 with sendonly
> 6054 recv 100 to INVITE 104
>
> 6512 recv another 200 ok for INVITE 103
> 6524 send another ACK 103
>
> 7202 recv another 200 ok for INVITE 103
> 7227 send another ACK 103
>
> 7287 recv BYE 1
> 7299 send 200 ok to BYE 1
>
> Do you have a SIPX contact you can escalate this to?
>
>
>
> On 2/24/2011 2:36 PM, Matthew Kitchin (public/usenet) wrote:
>
> This was their response as far as me sharing the firmware:
> "Excellent news.  Let me know what you find on the transfer issue.
> Now that we have confirmation of the fix we will be rolling this into our
> next GA release.  Feel free to let the user group know that a fix is
> available, but direct them to Polycom support to obtain it for the time
> being."
>
> The Polycom support ticket realted to this is 'Case# 1-103152086'
>
> If anyone has any trouble getting the firmware before it is released, let
> me know.
>
> I will be glad to write a wiki article on how to get these up and running.
>
> On 2/24/2011 1:57 PM, Matthew Kitchin (public/usenet) wrote:
>
> Success. I 

Re: [sipx-users] Update for the downloads page, stale link "Get Started"

[Douglas Hubler]

Fixed, ISO page will fix itself on next ISO update.

On Fri, Mar 18, 2011 at 2:26 PM, Tony Graziano
 wrote:
> The download.sipfoundry.org site has a link on all the pages that evidently
> became stale when the wiki was reworked.
>
> The "sipXecs Start Here" link points to:
>
> http://wiki.sipfoundry.org/display/sipXecs/ISO+CD+Installation+of+sipXecs
>
> but should insted point to:
> http://wiki.sipfoundry.org/display/sipXecs/Installing+from+CD
>
> So either the download page should be updated or the wiki can simply forward
> it.
>
> I don't think it's urgent because the wiki assumes that is what you want and
> suggests the proper page, you just need to click one more time is all.
> --
> ==
> Tony Graziano, Manager
> Telephone: 434.984.8430
> sip: tgrazi...@voice.myitdepartment.net
> Fax: 434.326.5325
>
> Email: tgrazi...@myitdepartment.net
>
> LAN/Telephony/Security and Control Systems Helpdesk:
> Telephone: 434.984.8426
> sip: helpd...@voice.myitdepartment.net
>
> Helpdesk Contract Customers:
> http://support.myitdepartment.net
> Blog:
> http://blog.myitdepartment.net
>
> Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
>
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>
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Re: [sipx-users] Caller ID problem wiith Sipgate

[Douglas Hubler]

On Wed, Mar 16, 2011 at 5:02 AM, Irena Dolovčak
 wrote:
> Thanks Douglas,
> it's working now.. but still not as I want to..
>
> the number is shown when I put it in the preferred identity like this:
> 385...@sipconnect.sipgate.de
> is there a way to make every user to use a different number? the caller id
> in user settings don't change the preferred identity. it always stays the
> same.
> can some setting affect the number part in preferred identity?
Is sipconnect.sipgate.de your SIP domain?

If not, you can try to set 385...@sipconnect.sipgate.de as the
caller id of each user, but I doubt it will work.  If it doesn't then
this is not supported.  sipXbridge is a network of "if this flag do
this else do this" and I think we need something that lets the admin
customize.

If it is, setting 385111 as the caller id on the user should work.
 If it's not capture a siptrace and be sure to actually capture a
call.  It should have an INVITE in the message.
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Re: [sipx-users] wiki help please:

[Douglas Hubler]

On Wed, Mar 9, 2011 at 4:12 PM, Michael Scheidell
 wrote:
> so, how do I put it on a submenu, and is cookbooks/alerts and notifications
> best place?

yes, i think so

2 ways:
1.) Edit page and you will see Change location toward bottom of edit page
2.) Tools/View Hierarchy and you can drag/drop page where you want
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Re: [sipx-users] Caller ID problem wiith Sipgate

[Douglas Hubler]

On Tue, Mar 8, 2011 at 3:12 AM, Irena Dolovčak  wrote:
> Great!
> Thank you very much Douglas.
> Can you tell me the exact version where this is fixed? (just to be sure, is
> it the version sipxecs-4.4.0-129.ga7ccc-x86_64.iso ?)

It's building now actually, there was a build yesterday but it didn't
make it in.   Few more hours.

version:  4.4.0-132.gcec97

You can install from ISO or just run yum update when it's available.
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Re: [sipx-users] Caller ID problem wiith Sipgate

[Douglas Hubler]

On Mon, Mar 7, 2011 at 4:31 PM, Douglas Hubler  wrote:
> On Mon, Mar 7, 2011 at 4:04 PM, Douglas Hubler  wrote:
>> 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.
>
> ah ha,  I now see the config-test failure and will look into this.
> Please open an issue in jira.
>
>  http://wiki.sipfoundry.org/display/sipXecs/How+to+help

Never mind, I created one and will have fix in next 4.4 build by tomorrow
 http://track.sipfoundry.org/browse/XX-9497
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Re: [sipx-users] Caller ID problem wiith Sipgate

[Douglas Hubler]

On Mon, Mar 7, 2011 at 4:04 PM, Douglas Hubler  wrote:
> 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.

ah ha,  I now see the config-test failure and will look into this.
Please open an issue in jira.

  http://wiki.sipfoundry.org/display/sipXecs/How+to+help
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