Re: [asterisk-users] Paetec SIP Trunk

2012-09-27 Thread bryantz
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 jared.bax...@gmail.com wrote: Has anyone had experience using a SIP trunk provided by Paetec over MPLS? With or without FreePBX

Re: [asterisk-users] Hangup Cause and SIP Response Code

2012-04-25 Thread BryantZ
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 kpflem...@digium.com 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

Re: [asterisk-users] Best non polycom SIP conference room phone

2012-01-08 Thread BryantZ
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,

Re: [asterisk-users] Microsoft Lync server and Asterisk access

2011-04-14 Thread BryantZ
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 dicken...@cfmc.com 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

Re: [asterisk-users] Asterisk, Sent accountcode between 2 asterisk

2011-03-05 Thread BryantZ
On Mar 5, 2011, at 2:29 AM, Olivier CALVANO o.calv...@gmail.com 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

Re: [asterisk-users] Asterisk, Sent accountcode between 2 asterisk

2011-03-05 Thread BryantZ
On Mar 5, 2011, at 8:52 AM, brya...@zktech.com wrote: On Mar 5, 2011, at 2:29 AM, Olivier CALVANO o.calv...@gmail.com 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

Re: [asterisk-users] res_fax

2011-01-20 Thread BryantZ
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

Re: [asterisk-users] DIALSTATUS on CANCEL

2010-12-24 Thread BryantZ
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 hvarda...@gmail.com 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

Re: [asterisk-users] DIALSTATUS on CANCEL

2010-12-24 Thread BryantZ
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 dicken...@cfmc.com wrote: If on the dial command you add option g, if the

Re: [asterisk-users] Wireless Desktop VoIP Phone?

2010-12-17 Thread BryantZ
I use grandstream with the linksys/cisco adapter. Bryant On Dec 17, 2010, at 3:04 PM, Michael Graves mgra...@mstvp.com 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