It's both sides with this problem. The polycom 321s behind the server are
pulling programming/provisioning from sipxecs (dhcp and tftp). The grandstream
I have here at home I've configured manually. Again, I can successfully make
calls to/from all extensions, audio is working bidirectionally, an
its a grandstream having the problem or both?
Can you tell me how the phones were programmed? I think I see the hostname
being called "instead of" the domain name. Why would that be?
Did you properly populate the intranet subnets?
On Sat, Jun 16, 2012 at 3:07 PM, m...@mattkeys.net wrote:
> Hi T
On Sat, 16 Jun 2012 11:25:57 -0400, Tony Graziano wrote:
>No, you cannot run 4.4 on Centos 6.x.
Thank you.
> On Sat, Jun 16, 2012 at 11:01 AM, m...@grounded.net
> wrote:
>
>> On Sat, 16 Jun 2012 04:56:33 -0400, Michael Picher wrote:
>> Just to clarify..
>>
>> sipXecs / openUC 4.4 - CentOS or
I think this was asked and answered already. Actually it has been brought
up many times, yet the answer has not changed. No, you cannot run 4.4 on
Centos 6.x.
On Sat, Jun 16, 2012 at 11:01 AM, m...@grounded.net wrote:
> On Sat, 16 Jun 2012 04:56:33 -0400, Michael Picher wrote:
> > Just to clar
You are not really providing enough information...
Is the phone local or remote? If it is remote, did you change any of the
phone parameters manually (I ask because the SRV records are missing for
the domain name and that will always cause issues)...
It would really help if you explained the call
On Sat, 16 Jun 2012 04:56:33 -0400, Michael Picher wrote:
> Just to clarify..
>
> sipXecs / openUC 4.4 - CentOS or RedHat 5.x
> sipXecs / openUC 4.6 - CentOS or RedHat 6.x
I guess that brings up another question. Any problem running sipx 4.4 on Centos
6.x?
The problem I'm having is that I can
Karoo seems a good fit. However, where are decent docs? I know
FreeSwitch, I know Javascript... it all sounds good...
in this case, I really just need upper registration. (ISP with phones on
one side/ITSP on the other).
SipXecs is way overkill for what I need.
Is there any doc other than the s
Hello,
I'm having the same problem as this user :
http://forum.sipfoundry.org/index.php?t=msg&th=13827&goto=49331&S=dadc367f6ec44a61108be26ea0301f39#msg_49331
I can make calls from extension to extension, however if I let it go to
voicemail the system returns a "408 Timeout" to the phone. Calls
Did you try karoo?
On Jun 16, 2012 8:17 AM, "Gerry Hull" wrote:
> Hi Joegen,
>
> Sorry for the OT. Hey, you wrote a useful piece of code... so why not use
> it? I use sipXecs for other projects, but in this case I just need an SBC.
>
> I'll shutup now.
>
> Gerry
>
> On Fri, Jun 15, 2012 at 10:
On 6/15/2012 5:02 PM, gabriel wrote:
> hello sipx-users,
>
> I am running a brand new sipx install (4.4.0) with polycom 670 phones and
> a few sip trunks. I want to connect an analog fax machine to the system and
> was wondering which analog adapters you guys recommend.
> I have used cisco ata186
Hi Joegen,
Sorry for the OT. Hey, you wrote a useful piece of code... so why not use
it? I use sipXecs for other projects, but in this case I just need an SBC.
I'll shutup now.
Gerry
On Fri, Jun 15, 2012 at 10:26 PM, Joegen Baclor wrote:
> Todd,
>
> I wrote OpenSBC so I guess the OP is ju
Not at this time.
This is called Call Admission Control and there's no concept of it at this
point.
Personally I believe this belongs at the network layer rather than at the
application layer with hard limits. Cisco had done some work around this
early but I'm not sure it's current state in thei
Just to clarify..
sipXecs / openUC 4.4 - CentOS or RedHat 5.x
sipXecs / openUC 4.6 - CentOS or RedHat 6.x
Thanks,
Mike
On Fri, Jun 15, 2012 at 4:10 PM, m...@grounded.net wrote:
> I was... afraid of that... but appreciate the reply. Thanks very much.
>
>
> On Fri, 15 Jun 2012 16:03:44 -04
hmmm... another workaround then might be to use a Patton gateway as the
go-between. Josh did some work recently around using the Patton as a
termination point for SIP trunks. I think the config is in the Wiki.
Mike
On Fri, Jun 15, 2012 at 3:04 PM, Matt White wrote:
> >>> Michael Picher 06/15
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