Is inbound fax to email support sipxecs 4.4 dependent on SipXecs-Bridge.
I have tested and am getting connection but no fax tone.
It works perfectly when directed at a trunk using sipxecs-bridge
Gabrial Casey
Telecommunications
Franklin American Mortgage Company
501 Corporate Center
, they must also be t.38 capable for fax mode.
The only time you will hear the fax tone is when the invite or re-invite
negotiates t.38. I somehow think you are not negotiating t.38 or you would be
getting a tone.
On Wed, Aug 17, 2011 at 4:03 PM, Gabe Casey gca...@franklinamerican.com
wrote
Congrats Josh !!! its about time you started getting paid for your help
Gabrial Casey
Telecommunications
Franklin American Mortgage Company
501 Corporate Center Dr.
Suite 400
Franklin, TN 37067
Direct:615-468- 2945
Cell: 615-852-5015
Fax: 615-628-5698
Do the Conference API docs on the WIKI reflect the rest services in 4.2.1 ?
Gabrial Casey
Telecommunications
Franklin American Mortgage Company
501 Corporate Center Dr.
Suite 400
Franklin, TN 37067
Direct:615-468- 2945
Cell: 615-852-5015
Fax: 615-628-5698
County Network Engineer
979.361.4676
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Gabe Casey
Sent: Wednesday, April 06, 2011 2:48 PM
To: Discussion list for users of sipXecs software
Subject: [sipx-users] Silent Fail when
Has anyone had this issue ?
4.2.1
Gabrial Casey
Telecommunications
Franklin American Mortgage Company
501 Corporate Center Dr.
Suite 400
Franklin, TN 37067
Direct:615-468- 2945
Cell: 615-852-5015
Fax: 615-628-5698
Email:gca...@franklinamerican.com
Tw should be able to hand off a sip line if that can be done your job will be
much easier/cheaper I bet the device they drp off is an ata/cable modem anyway
Sent from my iPhone
On Mar 29, 2011, at 4:07 PM, Philippe Laurent p...@ideos.com wrote:
Folks -
Of all the knowledgeable goodness on
I was able to solve this via an asterisk media gateway by setting pedantic
message examination on the peer.
Sent from my iPhone
On Mar 10, 2011, at 10:18 AM, Ly Tran ly.t...@synaptyk.com wrote:
We have been running SipXecs for a little over a year now and had never
experienced this problem
Hey Kyle could you explain how you built the collector i am curious as we have
the productivity suite and would like to turn this feature on.
Gabrial Casey
Telecommunications
Franklin American Mortgage Company
501 Corporate Center Dr.
Suite 400
Franklin, TN 37067
Direct:615-468- 2945
Sans Sipx-Bridge
On Sat, Jan 8, 2011 at 10:25 AM, Gabe Casey gca...@franklinamerican.com
wrote:
I have a question concerning the installation of a cisco UBE. This SBC is
required for an installation.
Typically Music on hold scenarios are handled via the sipx-bridge. I see
I have a question concerning the installation of a cisco UBE. This SBC is
required for an installation.
Typically Music on hold scenarios are handled via the sipx-bridge. I see that
the polycom phones use an sdp event to trigger the invite to the url
~~...@doman.tld
Can anyone explain this
effect on the display.
On Fri, Nov 19, 2010 at 2:01 PM, Matthew Kitchin (public/usenet)
mkitchin.pub...@gmail.com wrote:
No luck. I downgraded to spip_ssip_BootROM_4_2_1_release_sig.zip and I still
had the contrast issue.
On 11/19/2010 12:27 PM, Gabe Casey wrote:
Version 4.2.2
57235
Email:gca...@franklinamerican.com
From: Massimo Vignone massimo.vign...@unimore.it
To: Discussion list for users of sipXecs software
sipx-users@list.sipfoundry.org
Sent: Tuesday, November 16, 2010 8:18:45 AM
Subject: Re: [sipx-users] 1 Ring Diversions to Voicemail
On 11/15/2010 07:28 PM, Gabe
, but the 450s
had the issue. I don's see it with 3.1.3c or 3.2.3.
On 11/19/2010 11:21 AM, Gabe Casey wrote:
Yes after much testing i believe the 3.2.2 would be my recommendation for
Sipxecs 4.2.X
3.1.3 has inbound calling issues as well as zero support for LLDP.
3.2.3 is just unusable
you may
I am having an issue that was reported to be in Polycom Firmware 3.2.3
* 3.2.3 Any latency over approximately 30-40 ms will will cause the call to
divert to voicemail after 1 or less rings.
I am using bootrom 4.2.2 and Sip App 3.2.2 and am seeing it occur
inconsistently throughout the day.
should be using.
On Mon, Nov 15, 2010 at 1:28 PM, Gabe Casey gca...@franklinamerican.com
wrote:
I am having an issue that was reported to be in Polycom Firmware 3.2.3
* 3.2.3 Any latency over approximately 30-40 ms will will cause the call to
divert to voicemail after 1 or less rings.
I
the Proxy.
Gabrial Casey
Telecommunications
Franklin American Mortgage Company
501 Corporate Center Dr.
Suite 400
Franklin, TN 37067
Direct:615-468- 2945
Cell: 615-852-5015
Fax: 615-628-5698
Email:gca...@franklinamerican.com
From: Gabe Casey gca...@franklinamerican.com
Gabrial Casey
Telecommunications
Franklin American Mortgage Company
501 Corporate Center Dr.
Suite 400
Franklin, TN 37067
Direct:615-468- 2945
Cell: 615-852-5015
Fax: 615-628-5698
Email:gca...@franklinamerican.com
From: Gabe Casey gca...@franklinamerican.com
To: Discussion list
Gabrial Casey
Telecommunications
Franklin American Mortgage Company
501 Corporate Center Dr.
Suite 400
Franklin, TN 37067
Direct:615-468- 2945
Cell: 615-852-5015
Fax: 615-628-5698
Email:gca...@franklinamerican.com
From: Gabe Casey gca...@franklinamerican.com
To: Discussion list for users
is there a setting or ssl crt that needs to be adjusted when doing an external
rewrite of the url to the end user portal from outside the lan.
My goal is to allow access of the user portal outside the network, however the
browser is throwing errors and forcing a logout.
voicemail.example.com
and not a rewrite.
On Mon, Nov 1, 2010 at 3:27 PM, Gabe Casey gca...@franklinamerican.com wrote:
is there a setting or ssl crt that needs to be adjusted when doing an
external rewrite of the url to the end user portal from outside the lan.
My goal is to allow access of the user portal
at the aliases file directly
cat /var/sipxdata/sipdb/aliases.xml
im betting its assigned somewhere
On Mon, Nov 1, 2010 at 5:23 PM, Gabe Casey gca...@franklinamerican.com
wrote:
I am getting the error The user ID or alias 1000 duplicates an existing alias
for a user or service
I
I am having an issue when backing up and restoring a system from 4.2.1 -
4.2.1.
This is a 2 server system in which the Second server serves ACD , Conference
Rooms and a Secondary Sip Proxy.
When i restore from back up i seem to lose my conference rooms. I had
instructed the system to assign a
requires a soap
call.
- Original Message -
From: Nathaniel Watkins
To: Gabe Casey gca...@franklinamerican.com
Sent: Thu Jul 08 14:38:42 2010
Subject: RE: [sipx-users] Recommended free virtualization platform for sipx?
I’m out the rest of today – I’ll do some testing tomorrow/weekend
I am having some issues adding a new server to the HA environment since my
upgrade to 4.2
I currently have 2 that were configured together 4.0.4 and then upgraded.
Currently I have:
installed 4.2 Centos ISO to a 3rd server
Created the server in the master
Run the setup on the 3rd box
Is there an easy way to configure a group invite to a conference room ?
Warm Regards
Gabrial Casey
Telecommunications
Franklin American Mortgage Company
501 Corporate Centre Dr.
Suite 400
Franklin, TN 37067
Direct:615-468- 2945
Cell: 615-693-2833
Fax: 615-628-5698
: 615-693-2833
Fax: 615-628-5698
Email:gca...@franklinamerican.com
From: Michael Scheidell scheid...@secnap.net
To: sipx-users@list.sipfoundry.org
Sent: Thursday, July 1, 2010 4:15:38 PM
Subject: Re: [sipx-users] Group Invite to Conference
On 7/1/10 4:37 PM, Gabe Casey wrote
...@franklinamerican.com
From: Nathaniel Watkins nwatk...@garrettcounty.org
To: Gabe Casey gca...@franklinamerican.com, Michael Scheidell
scheid...@secnap.net
Cc: sipx-users@list.sipfoundry.org
Sent: Thursday, July 1, 2010 5:02:46 PM
Subject: RE: [sipx-users] Group Invite to Conference
Message -
From: M. Ranganathan mra...@gmail.com
To: ROBERT JOLY (ROBERT) rj...@avaya.com
Cc: Gabe Casey gca...@franklinamerican.com, sipx-users
sipx-users@list.sipfoundry.org
Sent: Monday, May 17, 2010 9:10:48 AM
Subject: Re: [sipx-users] Ghost Calls
On Mon, May 17, 2010 at 9:46 AM, JOLY
I am seeing a strange issue periodically in 4.2 where a call is getting stuck
in the system and recalling the outside number every 30 min. From the Logs each
call has the same Call-ID as the original. Anyone else seen this ? It has
happened 2 or 3 times now. The only way to get rid of it seems
Corporate Centre Dr.
Suite 400
Franklin, TN 37067
Direct:615-468- 2945
Cell: 615-693-2833
Fax: 615-628-5698
Email:gca...@franklinamerican.com
- Original Message -
From: Scott Lawrence xmlsc...@gmail.com
To: Gabe Casey gca...@franklinamerican.com
Cc: sipx-users@list.sipfoundry.org
: Tony Graziano tgrazi...@myitdepartment.net
To: Gabe Casey gca...@franklinamerican.com
Cc: sipx-users@list.sipfoundry.org
Sent: Friday, April 30, 2010 12:28:11 AM
Subject: Re: [sipx-users] Provisoning Server Errors after 4.2 Upgrade
depending on the service that is failing...
post
: Gabe Casey gca...@franklinamerican.com
To: Tony Graziano tgrazi...@myitdepartment.net
Cc: sipx-users@list.sipfoundry.org
Sent: Friday, April 30, 2010 9:26:46 AM
Subject: Re: [sipx-users] Provisoning Server Errors after 4.2 Upgrade
{FreeSWITCH=Running,
sipXmrtg=Running,
SipXrest=Running
-693-2833
Fax: 615-628-5698
Email:gca...@franklinamerican.com
- Original Message -
From: Scott Lawrence xmlsc...@gmail.com
To: Gabe Casey gca...@franklinamerican.com
Cc: sipx-users@list.sipfoundry.org
Sent: Friday, April 30, 2010 10:56:12 AM
Subject: Re: [sipx-users] Provisoning
Try this
cat /etc/yum.repos.d/sipxecs-stable-centos.repo
[centos-5-base]
name=CentOS-5 - Base
mirrorlist=http://mirrorlist.centos.org/?release=5arch=$basearchrepo=os
#baseurl=http://mirror.centos.org/centos/5/os/$basearch/
gpgcheck=1
Hey After Upgrade from 4.0.4 i am getting some errors in my provisioning
service.
Message from sipXecs
Alarm: SPX00032
Reported on: plsipx01.franklinamerican.com
Reported at: 2010-04-29T20:10:21.746582Z
Severity: WARNING
Alarm Text: The configuration data for process 'sipXprovision' ()
-5698
Email:gca...@franklinamerican.com
- Original Message -
From: Gabe Casey gca...@franklinamerican.com
To: sipx-users@list.sipfoundry.org
Sent: Thursday, April 29, 2010 3:22:55 PM
Subject: [sipx-users] Provisoning Server Errors after 4.2 Upgrade
Hey After Upgrade from 4.0.4 i
-
From: Gabe Casey gca...@franklinamerican.com
To: sipx-users@list.sipfoundry.org
Sent: Thursday, April 29, 2010 3:22:55 PM
Subject: [sipx-users] Provisoning Server Errors after 4.2 Upgrade
Hey After Upgrade from 4.0.4 i am getting some errors in my provisioning
service.
Message from
I was reading the wiki on SipXBride
Typically ITSPs do not handle certain types of SIP requests such as REFER
which is used in Call Transfer operations. To implement call transfer,
SipXbridge does signaling translation, converting a REFER request to an INVITE
request to the call transfer
for now ?
Gabrial Casey
Telecommunications
Franklin American Mortgage Company
Direct:615-468- 2945
Cell: 615-693-2833
Email:gca...@franklinamerican.com
- Original Message -
From: M. Ranganathan mra...@gmail.com
To: Gabe Casey gca...@franklinamerican.com
Cc: sipx-users sipx-users
Gabe Casey wrote:
yeah its asterisk 1.6.2.0-rc4 which is something i am willing to
downgrade at anytime
I have a call trace but its at work right now. I will supply it as
soon as i get there however the dialog was
normal until exactly 30min into the call when SBC sends another invite
for PRI Trunks
Gabrial Casey
Telecommunications
Franklin American Mortgage Company
Direct:615-468- 2945
Cell: 615-693-2833
Email:gca...@franklinamerican.com
- Original Message -
From: Tony Graziano tgrazi...@myitdepartment.net
To: Gabe Casey gca...@franklinamerican.com
I have seen a couple of open tickets on the 30 min session reinvite issue. This
causes the media gateway to respond with a 488 and then a disconnect.
What are some possible configuration options here ? It happens only on outbound
calls
polycom 450 frimware 1.3.1
sipxecs 4.04
asterisk /
polycom app version is 3.1.3 bootrom is 4.2 yes
Gabrial Casey
Telecommunications
Franklin American Mortgage Company
Direct:615-468- 2945
Cell: 615-693-2833
Email:gca...@franklinamerican.com
- Original Message -
From: Tony Graziano tgrazi...@myitdepartment.net
To:
This actually does not happen pstn - gw - sipxecs - phone
but happens consistently polycom - sipxecs - gw - pstn
Gabrial Casey
Telecommunications
Franklin American Mortgage Company
Direct:615-468- 2945
Cell: 615-693-2833
Email:gca...@franklinamerican.com
- Original Message -
with a 488 and then the call ends.
Gabrial Casey
Telecommunications
Franklin American Mortgage Company
Direct:615-468- 2945
Cell: 615-693-2833
Email:gca...@franklinamerican.com
- Original Message -
From: Tony Graziano tgrazi...@myitdepartment.net
To: Gabe Casey gca
Guys I have tried and tried to find a reasonable solution to a PRI media
gateway in the OS realm Asterisk Freeswitch ect but still can not find a
solution to forwards to mobile devices via a users forwarding options as well
as unpredictable assisted transfers
Note there is no nat in this
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