We have customers that have migrated to our network from them due to their
reliability issues. Most of them are in the US west and east.
Jared Baxley wrote:
Has anyone had experience using a SIP trunk provided by Paetec over MPLS?
With or without FreePBX
Regards,
Jared Baxley
Kevin
I am using 1.8.x & 10.x
Bryant Zimmerman (ZK Tech Inc./interNetGR)
(616) 855-1030 Ext. 2003
On Apr 25, 2012, at 5:00 PM, "Kevin P. Fleming" wrote:
> On 04/25/2012 07:08 AM, Bryant Zimmerman wrote:
>> I can log the ISDN cause code using ${HANGUPCAUSE} but I also need to
>> track the actu
Thank you for your responses. No where did I say I hate polycom phones. I
personally do not like their approach to sip as a company. Their audio quality
is top notch but for me the rest leaves me wanting. Has anyone used the newer
snom conference room phone?
Bryant Zimmerman
On Jan 8, 2012,
Yes you can. Lync can not do registration and it is a trick to setup.
Bryant
On Apr 14, 2011, at 11:23 AM, Jim Dickenson wrote:
> We have a client that currently has a Microsoft Lync setup. I must admit I
> know nothing about this setup.
>
> What we would like to be able to do is allow the ph
On Mar 5, 2011, at 8:52 AM, brya...@zktech.com wrote:
>
>
> On Mar 5, 2011, at 2:29 AM, Olivier CALVANO wrote:
>
>> Hi
>>
>> I have two Asterisk Server:
>>
>> The first server "A", all phone are connected
>> The Second server "B" only route call to a lot of SIP supplier
>>
>> the server A
On Mar 5, 2011, at 2:29 AM, Olivier CALVANO wrote:
> Hi
>
> I have two Asterisk Server:
>
> The first server "A", all phone are connected
> The Second server "B" only route call to a lot of SIP supplier
>
> the server A sent:
>
> ; Destination: Non connu dans le DialPlan - Apparaitra en UNK
On Jan 20, 2011, at 8:53 PM, Steve Underwood
> On 01/21/2011 06:46 AM, Bryant Zimmerman wrote:
>> On 01/20/2011 11:47 AM, Steve Underwood
>> On 01/20/2011 11:11 PM, Kevin P. Fleming wrote:
>> > On 01/19/2011 02:30 PM, Bryant Zimmerman wrote:
>> >> On 01/19/2011 02:05 PM, Bryant Zimmerman wrote:
>>
I am using the g option and it does not run the next statement or "h" extension
if the caller hangs up before an answers or time out event occurs during a
dial comand.
Bryant
On Dec 24, 2010, at 9:55 AM, Jim Dickenson wrote:
> If on the dial command you add option g, if the call is not answe
If a call is hung up before an answer our "h" extension is not running in our
dial macro
Bryant
On Dec 24, 2010, at 3:47 AM, Vardan Harutyunyan wrote:
> Hello Bryant
> Extension "h" is worked in any case of hangup.
> It not important to that the call was answered or no.
> It also be more flex
I use grandstream with the linksys/cisco adapter.
Bryant
On Dec 17, 2010, at 3:04 PM, "Michael Graves" wrote:
> On Fri, 17 Dec 2010 10:40:00 -0500, Matt wrote:
>
>> I'm looking for a wireless desktop VoIP phone. Does any exist?
>
> I beleive that snom supports the use of a wifi usb dongle in
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