Great share. Thanks!
On Mon, Aug 1, 2011 at 7:03 AM, Max DiOrio m...@universityent.com wrote:
That's fantastic. Thanks for your work. Ill definitely be giving them our
during our roll out this week.
Mark Wood mark.w...@redphonetech.com wrote:
We developed these training aids for our
redownloading the firmware or
get with your audiocodes vendor to determine why you can't upload the file.
On Fri, Jul 22, 2011 at 3:13 AM, Rhon c4rdi...@gmail.com wrote:
Hi,
I have a problem updating my firmware from 5.6 to 6.0. The error shows
when uploading my cmp file:
Unrecognized CMP file
the firmware
or get with your audiocodes vendor to determine why you can't upload the
file.
On Fri, Jul 22, 2011 at 3:13 AM, Rhon c4rdi...@gmail.com wrote:
Hi,
I have a problem updating my firmware from 5.6 to 6.0. The error shows
when uploading my cmp file:
Unrecognized CMP file
Hi,
I have a problem updating my firmware from 5.6 to 6.0. The error shows
when uploading my cmp file:
Unrecognized CMP file was detedted. File Upload aborted
Can anyone here would like to share their MP118 6.0 firmware please?
Thanks
___
sipx-users
Great stuff
Thanks very much Josh!
On Mon, Jul 4, 2011 at 6:22 PM, Michael Picher mpic...@ezuce.com wrote:
Excellent job Josh. A good solution for those who can't wait!
On Sun, Jul 3, 2011 at 3:53 PM, Josh Patten jpat...@ezuce.com wrote:
Just wrote up an active/passive clustering howto
I interested in knowing the status of this topic. Does anyone resolve the
issue?
Thanks
On Thu, Sep 2, 2010 at 3:49 AM, Martin Steinmann mar...@ezuce.com wrote:
Kyle
I don't think this is a known issue. I assume there is no NAT or else
between the gateway and sipXecs. Firmware 5.8 should
/DevelopingBBB#Developing_on_the_bleeding_edgelooks
like it’s already been started
*From:* sipx-users-boun...@list.sipfoundry.org [mailto:
sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Rhon
*Sent:* Thursday, September 23, 2010 12:38 AM
*To:* sipx-users@list.sipfoundry.org
*Subject:* [sipx
Hi,
Has anyone here was able to integrate SipXecs 4.2 and Bigbluebutton 0.7?
My plan is whenever a videoconference is conducted using Bigbluebutton, I
can make a conference call to SipXecs where our remote offices are
connected.
This way we can use Videoconf via Bigbluebutton and have voice conf
Hi Everyone,
I manage to fix our problem. It was a NAT issue. And setting pfsense to NO
NAT takes care of the problem.
Thank you all for your help.
Best regards and have a nice day!
Rhon
On Wed, May 19, 2010 at 12:05 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Either you create
to:
*Automatic outbound NAT rule generation (IPsec passthrough)*
I think NAT is not necessary since traffic is passing thru the GRE Tunnel
and not going out. You can correct me if I'm wrong here.
Hoping for your usual response.
Many thanks and have a nice day!
Rhon
Your diagram (to me) shows your PBX
destination
but still failed.
Any clue what's happening?
Thanks in advance.
Rhon
On Tue, May 18, 2010 at 2:30 PM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
Again, since your connection is site-to-site and your vpn via ipsec is
there, you need to ensure the ipsec is passing/allowing all
...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
- Original Message -
From: Rhon c4rdi...@gmail.com
To: Tony Graziano tgrazi...@myitdepartment.net;
sipx
but not on a hard phone.
I also made an ACL in cisco to open 5060(UDP/TCP) but it's no use.
I will greatly appreciate any inputs here.
Thank you in advance.
On Tue, May 18, 2010 at 3:42 PM, Rhon c4rdi...@gmail.com wrote:
Hello Tony,
Here's my x-lite registration to sipx:
Rhonsip:2
using Configurations tests.
Our networks are setup as seen below:
SITE A SIPX -- PFSENSE -- CISCO -- VIA GRE TUNNEL -- CISCO
-- PFSENSE -- SIPX SITEB
Any thoughts on what the problem could be?
I have bypassed everything on the firewall at the moment.
Thank you in advance.
Rhon
:
sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Rhon
*Sent:* Monday, May 17, 2010 9:14 AM
*To:* sipx-users@list.sipfoundry.org
*Subject:* [sipx-users] No Voice/IVR on Site-to-Site
Hi,
I have a problem with our deployment with SipXecs 4.2 which was installed
fresh using ISO build
Also, I set the pfsense *to Manual Outbound NAT rule generation (Advanced
Outbound NAT (AON))*
My NAT rules below for my voice VLAN:
WAN172.16.3.0/24 * * * * * NO
Thanks again for your help.
Brgds,
Rhon
On Mon, May 17, 2010 at 3:33 PM, Rhon c4rdi...@gmail.com wrote:
Hello Michael
I'm using IPSEC GRE and pfsense interfaces have private IPs. should I still
need NAT for that matter?
Thanks
On Tue, May 18, 2010 at 3:03 AM, Picher, Michael
mpic...@cmctechgroup.comwrote:
It should be set to manual and yes.
*From:* Rhon [mailto:c4rdi...@gmail.com]
*Sent:* Monday, May 17
I'm using IPSEC GRE and pfsense interfaces have private IPs. should I still
need NAT for that matter?
Thanks
On Tue, May 18, 2010 at 3:03 AM, Picher, Michael
mpic...@cmctechgroup.comwrote:
It should be set to manual and yes.
*From:* Rhon [mailto:c4rdi...@gmail.com]
*Sent:* Monday, May 17
.noarch: failure: rubygems-1.2.0-2.noarch.rpm from
sipxecs-stable: [Errno 256] No more mirrors to try.
xerces-c-2.8.0-2.x86_64: failure: xerces-c-2.8.0-2.x86_64.rpm from
sipxecs-stable: [Errno 256] No more mirrors to try.
*Any idea on how to fix it?
Thanks in advance
Rhon
*
*
On Tue, Apr 27
Hi,
I fixed the problem by individually installing the packages.
Rhon
On Tue, May 4, 2010 at 8:24 PM, Matthew Kitchin (public/usenet)
mkitchin.pub...@gmail.com wrote:
What is the output from cat /etc/yum.repos.d/*
On 5/4/2010 7:00 AM, Rhon wrote:
Hi Everyone,
I followed
We've been working with CME and SipXecs.
Could you please post the result of sh run so we can evaluate it?
Regards,
2010/3/26 Александр Горбунов m...@golex.nsk.ru
Can anyone share positive experience how to interconnect Cisco CM and
sipXecs 4.0.4?
It seems to me that I cannot make right CM
Hi Tony,
Sorry for hijacking the post. I'm just curious if DNS will be installed also
when installing sipxecs via yum?
How about CentOS 5.4 64bit? Any known issues with the install/sipxecs rpm
packages?
Thanks
Rhon
On Mon, Apr 26, 2010 at 6:35 PM, Tony Graziano tgrazi...@myitdepartment.net
Hi Matthew,
I noticed you turn of sipxecs and postgresql services, why?
Aren't those required by sipxecs?
Rhon
On Mon, Apr 26, 2010 at 10:03 PM, Matthew Kitchin (public/usenet)
mkitchin.pub...@gmail.com wrote:
These are the steps I use for installing sipx on CentOS 5.4. I think the
only
the polycom phone
and that's where we've got to start.
Actually, that's the reason my siptrace are empty. Once a call is made,
you'll get a busy tone instantly.
Rhon
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this?
Best regards,
Rhon
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Ok, that's clear enough. I'll post whatever finding made on monday.
Thanks again Tony for your assistance.
Rhon
On Sat, Apr 24, 2010 at 8:44 PM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
ex:
user 1234
1...@mydomain.com
not 1...@192.168.10.x or 1...@sipx.mydomain.com
Hi Tony,
That's exactly what I did.
As said we have site-to-site running already... it's just that people have
to call the AA first become connecting to the desired extension.
What I'm looking at is to go directly to the desired extension without
passing thru AA always.
Rhon
On Fri, Apr 23
Hmn.. that's strange. I already have this options in placed but still can't
call directly to extension without passing AA.
Thank you for your patience.
Rhon
On Fri, Apr 23, 2010 at 6:33 PM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
You dialplan needs to say, example:
Prefix 3 and 2
to the operator or dialing 200.
This can be accomplished the other way around.
Thank you again for all your support. \
Rhon
On Fri, Apr 23, 2010 at 8:37 PM, Picher, Michael
mpic...@cmctechgroup.comwrote:
How are you reaching the remote AA?
You really aren’t giving us much to go on.
*From
going to the operator or dialing 200.
Thanks
On Fri, Apr 23, 2010 at 8:46 PM, Rhon c4rdi...@gmail.com wrote:
Hi Michael,
Sorry for that.
Both sites are connected via IPSEC GRE tunnel.
To reach the remote site I dial 200 (i changed the default operator = 200),
I will then dial the extension
on HQ Sipx and
vice versa.
siptrunk is disable in our setup.
Rhon
On Fri, Apr 23, 2010 at 8:48 PM, Scott Lawrence xmlsc...@gmail.com wrote:
On Fri, 2010-04-23 at 18:26 +0800, Rhon wrote:
Hi Tony,
That's exactly what I did.
As said we have site-to-site running already... it's just
Hi Everyone,
Tony is right, changing the Address field to domain name fixed it.
Closing this thread.
Many thanks to Tony!
Rgds,
Rhon
On Sat, Apr 24, 2010 at 1:07 AM, Picher, Michael
mpic...@cmctechgroup.comwrote:
Ah, well, that never worked unless that is how the PBX is setup (not
using
+1 for me.
Best regards,
Rhon
On Sat, Apr 24, 2010 at 8:00 AM, gabriel g...@bayintegrated.net wrote:
Nathan, I feel your pain, I got over it :)
we should have a sipx-users-cisco list where we (the cisco users) can help
each other out without having to deal with the classic trow it out buy
and would like to thanks everyone for their inputs.
Hope you all bear with me and hoping for your usual support.
Best regards,
Rhon
On Wed, Apr 21, 2010 at 11:52 AM, Rhon c4rdi...@gmail.com wrote:
Hi Scott,
I turned the logging level to debug. Here's the result of siptrace.
Calling the cisco
There was an article to update your repos.
Kindly check this
Linkhttp://sipxecs.blogspot.com/2009/12/how-to-fix-yum-repos-file-for-sipxecs.html
Hope this helps.
Rhon
On Fri, Apr 23, 2010 at 12:12 AM, Scott Lawrence xmlsc...@gmail.com wrote:
On Thu, 2010-04-22 at 10:35 -0400, Gary wrote
Hi Tony,
Just a few question about your implementation.
Seeing your scenario means you only have 1 sipxecs server? Is the IPSEC
tunnel created via pfsense or from your router?
Best regards,
Rhon
On Thu, Apr 22, 2010 at 9:18 PM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
On Thu
into
interoperability issues like this. I just hope it will be very soon. :)
Thank you for your attention.
Best regards,
Rhon
On Fri, Apr 23, 2010 at 5:31 AM, Scott Lawrence xmlsc...@gmail.com wrote:
On Thu, 2010-04-22 at 12:25 -0700, Nathan Nieblas wrote:
Cisco follows IETF standards for SIP
With all due
operator
and was successful. Just don't know how to start with this scenario.
Hope you can help me.
Thanks in advance.
Rhon
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from cisco to cisco and everything are working well, except when
you introduce polycom on the scene. You can't call from a polycom phone to a
cisco 7970g phon.
Does the entries above contribute to our problem?
Best regards,
On Wed, Apr 21, 2010 at 11:52
AM, Rhon c4rdi...@gmail.com wrote:
Hi
Hi Scott,
Thank you for your reply.
Here's the new siptrace.
Thanks in advance.
Rhon
On Mon, Apr 19, 2010 at 7:52 PM, Scott Lawrence xmlsc...@gmail.com wrote:
On Mon, 2010-04-19 at 10:35 +0800, Rhon wrote:
Hi Scott,
Attached is the siptrace for your reference. Tried to interpret
Hi Scott,
I turned the logging level to debug. Here's the result of siptrace.
Calling the cisco 7970g from polycom will redirect to IVR with an error The
owner of extension 112 is not available.
Thanks in advance.
Rhon
On Tue, Apr 20, 2010 at 9:45 PM, Scott Lawrence xmlsc...@gmail.com wrote
I figured this out. :)
Thanks
On Mon, Apr 19, 2010 at 1:05 PM, Rhon c4rdi...@gmail.com wrote:
Hi Tony,
Thank you for your reply.
This is what I see in the SipXeconfig:Conference New Conference
Name:
Extension:
Description:
Conference Owner
Participant PIN:
Maximum legs:
Music
Hi,
We're running on the same problem. And we've been in that dilemma in 2 weeks
now.
Any chance you make any progress please share them.
Best regards,
Rhon
On Tue, Apr 20, 2010 at 8:44 AM, Nathan Nieblas nathan.nieb...@sacatech.com
wrote:
Ran into some firmware compatibility issues after
for DimDim
5.5 was fixed for SipXecs 4.2. Are you referring to hosting the conference
via Dim Dim site and not locally installed?
2. Are there any wiki about SipXecs and Dim Dim?
Thanks in advance
Rhon
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that was provision automatically by Sipx is my Polycom
650 phone.
I also noticed, even with the release of SiXecs 4.2 the problem with auto
provisioning remained.
The wiki seem to be lacking an important part in the configuration to make
things work flawlessly.
- Rhon
Hi Michael,
We always make sure that whenever we resolved issues, we post it on the
mailing list for those who might encounter the same problem in the future.
We will be documenting our project and will contribute to the community
whatever we've accomplished.
Best regards,
Rhon
On Sun, Apr 18
accomplish in our project documentation.
Best regards,
Rhon
On Mon, Apr 19, 2010 at 1:26 AM, Todd Hodgen thod...@verizon.net wrote:
The wiki is completely built by community involvement. We all have an
obligation to fix what is not accurate, or create articles for the mutual
benefit
Hi Tony,
Thank you for your reply.
This is what I see in the SipXeconfig:Conference New Conference
Name:
Extension:
Description:
Conference Owner
Participant PIN:
Maximum legs:
Music On Hold source:
Could you please explain what are the correct values?
Thanks in advance.
Rhon
On Sun, Apr
Hi Michael,
Sipx to Sipx are working now. Thanks for all your help.
Now our problem is how we can directly call an extension without passing the
AA.
Any thoughts on how to accomplish this?
Thanks in advance.
Rhon
On Thu, Apr 15, 2010 at 7:45 PM, Picher, Michael
mpic...@cmctechgroup.comwrote
g711ulaw as the default codec on both phones but makes no
difference. Though, I doubt this would be the culprit since cisco phone can
call polycom 650 without problem.
Any assistance will be very much appreciated.
Many thanks.
Rhon
On Fri, Apr 9, 2010 at 10:49 PM, Rhon c4rdi...@gmail.com wrote
the siptrace tomorrow.
Any thoughts on how to fix it? We are so delayed hope you can help us
resolve this.
Many thanks and greatly appreciate all your assistance.
Rhon
subscribe
Time: 2010-04-14T11:35:12.582000Z
Frame: 1 sipxopenfire.xml:28388
Source: sipxgw.ourcompany.lan
)? Are they behind their own NAT's?
Both sides are not NATed.
Thanks for your reply.
Best regards,
Rhon
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manipulation:
Destination Prefix: 9
Stripped Digits Number: 1
Prefix (Suffix) to Add: blank
Manipulation Table:
Source Prefix: *
Source IP: *
Stripped digits from left: 1
Number of Digits to Leave is blank.
Any thoughts on how to fix our problems?
Thanks in advance.
Rhon
On Mon, Apr 12, 2010 at 8
Hi Winson,
Thank you for your reply, appreciate it.
I will try your suggestion soon as I get in the office tom. Actually, we are
stuck with this problem. ;((
For our PSTN, we use FXO interface (RJ11) and not PRI. We're not using any
converter.
Thanks and best regards,
Rhon
On Mon, Apr 12
The extension is not available.
Any thoughts on how to fix this?
Thanks in advance.
Rhon
On Fri, Apr 9, 2010 at 2:54 PM, Picher, Michael mpic...@cmctechgroup.comwrote:
They need to have at least one of the same codecs selected. You might
want to capture some of this traffic and inspect the SIP
and after a few seconds dropped the call.
I can call any extension from the PSTN though.
Please help.
Rhon
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Hi Tony,
I also did that but it's still unable to make outgoing calls.
Are there any settings that I have to manually configure on my Audiocodes
Gateway?
Thanks
Rhon
On Fri, Apr 9, 2010 at 4:53 PM, Tony Graziano
tgrazi...@myitdepartment.netwrote:
Dont dial the first 9 at the phone. Just
)
v. Profile ID = does not seam to matter, set to “0” or “1”
Endpoint Phone Number Table
i. Channel(s) = “1-8” (for MP118 or “1-4” for MP114)
ii. Phone Number = my *POTS telephone number*
iii. Hunt Group ID = 1 (must set)
Thank you in advance.
Rhon
On Fri, Apr 9, 2010 at 5:42 PM, Rhon c4rdi
mean by that? Could you please elaborate?
Looking forward for your usual response.
Regards,
Rhon
On Fri, Apr 9, 2010 at 5:49 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
None that I know of. If you cannot make the call from any phone
(Cisco/Polycom/softphone), I would ensure
error message or reason. If you do not see the call getting to the AC, then
looking at your dialplan in the proxy and permissions would be the next
step.
On Fri, Apr 9, 2010 at 5:42 AM, Rhon c4rdi...@gmail.com wrote:
Hi Tony,
I also did that but it's still unable to make outgoing calls
:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
- Original Message -
From: Rhon c4rdi...@gmail.com
To: Tony Graziano tgrazi...@myitdepartment.net;
sipx-users@list.sipfoundry.org sipx-users@list.sipfoundry.org
Sent: Fri Apr
-Gateway-MP-118 FXO/v.5.60A.007.002
Reason: Q.850 ;cause=111 ;text=local
Content-Length: 0
Thank you in advance.
Rhon
On Fri, Apr 9, 2010 at 11:04 PM, Rhon c4rdi...@gmail.com wrote:
Tony,
201 is a Polycom 650 ip phone. I'm not sure why it's using an ip rather
than a domain.
Could it be the tftp
(cisco/polycom)
communicate with each other?
Thanks in advance.
Rhon
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during firmware update. I even saw the
term70.default.loads completely transferred. Only that it gave an error and
reboot.
TIA,
RHON
On Wed, Apr 7, 2010 at 1:40 PM, Ben Wannan b...@wannan.org wrote:
Rhon,
ensure that the term70.default.loads file is directing the phone to the
following files
on a sip system.
Yes, I know. I did this using my Cisco router + a tftp running on my
machine. Loading of sccp firmware loaded successfully. Same procedure didn't
work well with SIP firmware though.
Thanks
Rhon
1) Have tftpd32 running as both tftp and dhcp server
2) boot the phone and hold down
? I came across a site
that says you cannot upgrade from 8.5.3 to latest without a lower firmware
installed. I'm unsure if that's true.
Any help on the issue are very much appreciated.
Thanks a lot!
Rhon
On Wed, Apr 7, 2010 at 2:55 PM, Tony Graziano
tgrazi...@myitdepartment.netwrote
I finally got the firmware loaded. whew!!
Thanks for all your help!
On Wed, Apr 7, 2010 at 4:41 PM, Ben Wannan b...@wannan.org wrote:
That may be the case, I upgraded 8.3(1) to 8.3(5) to 8.5(4) no problems.
Regards,
Ben
On Wed, Apr 7, 2010 at 5:55 PM, Rhon c4rdi...@gmail.com wrote
to PSTN which is initiated on the other end via IP?
Which means a specific IP address or remote phone ext can call ex. and
be given a dial tone to the PSTN where he can dial a Local no?
Thank you in advance.
Rhon
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Hi Everyone,
We have a problem calling a local number. We use 3 digits extension (ext
400) number for the local ext.
But the dialplan as seen below don't give us valid results.
DIALTEMPLATE
TEMPLATE MATCH=#... Timeout=5 User=Phone /
TEMPLATE MATCH=0 Timeout=3 / !-- Local operator--
in advance.
Rhon
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FYI,
We are using Cisco as Gateway + FXO, SipXecs 4.1.7.
Calls were made from/to outside number
Thanks.
On Tue, Apr 6, 2010 at 11:29 PM, Rhon c4rdi...@gmail.com wrote:
Hi,
We noticed after calling or receiving a call there was at least more that
10secs of delay before getting a new dial
Cisco as Gateway + FXO, SipXecs 4.1.7. Calls were made from/to
a pstn number.
Thanks in advance.
- Rhon
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between pc and phone (cross-cable) but it's
no use. I'm getting the same error. Could it be the firmware?
We loaded the phone firmware to the router as last resort but we got
an *Auth Failed error
* whenever loading the firmware. We can see it in the cisco ip phone screen.
Please help.
Rhon
@Michael,
If I will load it as unmanaged tftp files, what files should be uploaded and
how are my cisco phones know to load it?
@Ben,
We'll try as you suggested and post back any progress here.
Many thanks to both of you.
Rhon
On Tue, Apr 6, 2010 at 8:39 AM, Picher, Michael mpic
It was mentioned earlier. It is a Cisco 2811 router.
Regards,
On Mon, Mar 29, 2010 at 9:17 PM, JOLY, ROBERT (ROBERT) rj...@avaya.comwrote:
Just wanted to update everyone that I did get it working.
It was the SIP ALG on my Cisco router.
Here was the fix:
no ip nat service sip
Hi Staffan,
Please allow me to comment on your answer to my colleague's post.
Our problem become apparent after Cisco Router forwarded an incoming call to
SipXecs server. I observed that AA indeed
processed the call transfer of the dialed extension: only that the call
transfer don't get thru for
a busy tone. But
can receive calls from internal.
I'll try 8.5.4 soon as I get back to the office and will post any progress.
Thank you so much for your help.
Regards and have a nice day!
--Rhon
On Sat, Mar 27, 2010 at 2:30 AM, gabriel g...@bayintegrated.net wrote:
just add your sipx server
Hi Gabriel,
Once I load the latest firmware. Other than the processNodeName, are there
any other configurations which I have to consider manually changing in
SEP.xml file? In your experience?
Thanks in advance and looking forward for your usual response!
Best regards,
Rhon
On Sat, Mar 27
Hi Ben and All,
Could you please share with me your Cisco 7970G sip *8.5(3) *firmware?
Thanks in advance.
On Sun, Mar 28, 2010 at 3:59 PM, Ben Wannan b...@wannan.org wrote:
Rhon,
Are the handsets behind a NAT, ? i've had trouble with these models in that
configuration (with the sipx box
On Thu, Mar 25, 2010 at 7:35 PM, Scott Lawrence xmlsc...@gmail.com wrote:
On Thu, 2010-03-25 at 12:09 +0800, Rhon wrote:
Also, how can we trace logs in sipxecs when establishing a call? We
cannot see what's happening in the backend. All outgoing calls didn't
get through and we don't have any
have issues so good luck
with that ;)
-gabriel
On Fri, 26 Mar 2010, Rhon wrote:
Hi,
Thanks for the link. I've already seen it and will try the soonest. In the
meantime I need a working SEP.xml file for Cisco 7970G phone. SipXecs is
unable to
load the phone by adding managed phones.
I'm
Hi Mate,
Thanks for the advise and will probably do it in binary.
Brgds,
On Fri, Mar 26, 2010 at 12:29 AM, Eric Varsanyi sip...@eljv.com wrote:
On Mar 25, 2010, at 6:34 AM, Scott Lawrence wrote:
On Thu, 2010-03-25 at 10:03 +0800, Rhon wrote:
Hi,
I'm planning to install SipXecs
Hi,
I'm planning to install SipXecs-4.0.4 into a new server. Does anyone here
were able to successfully install SipXecs-4.0.4 on Fedora 11 64bit machine?
Thanks
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Hi Everyone,
We've been having a hard time making our Cisco 7970G work with SipXecs-4.0.4
and we are already running out of time for the deployment of our project.
I wonder if someone can share with us a working SEP file to start with so we
can isolate and minimize problem trace.
Also, how can
Hi,
I bought an AudiioCodes MP-118FXO with 4.8 firmware. I have issues getting
out of my PSTN and other things.
Anyone here would like to share the AudioCodes MP-118FXO 5.4 SIP Firmware
please?
I will appreciate it very much!
Thanks and best regards,
james
Hi,
I download the from the pub repo sipfoundry-4.1.7-018 and hoping to get the
latest build, the download all went fine.
However, after the installation I noticed the version was 4.0.4. How come?
This is also true by downloading the fedora build but downloading CentOS?
one more thing the
Hi,
I'm trying to install ntp on my sipxecs 4.0.4 installation but are getting
some error as shown below:
[r...@sipxecs ~]# yum install ntp
Loading downloadonly plugin
Loading fastestmirror plugin
Loading basearchonly plugin
Loading mirror speeds from cached hostfile
* centos-5-addons:
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