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+92.21.529.0381 (3 lines) x200
, etc but nothing
works.
Also- the same phones work perfectly with our Asterisk 1.4.26 office
server.
Help is most appreciated. We are at a loss here.
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CEO
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www.emergen.biz
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of faraz
Sent: Friday, April 18, 2008 6:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom LDAP Corporate Directory
I havent tried it. I have
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Faraz R Khan
Chief Architect
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Protocol (MGCP) yes yes no
Agent Call Agent Proxy Channel yes yes no
--
7 channel drivers registered.
*CLI module reload chan_zap.so
No such module 'chan_zap.so'
*CLI
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there are no grandstream-like problems with these cisco's.
Cisco is much more co-operative than Linksys/Polycom and is working a
feasible deal for me to replace GXPs with Cisco 7905/06G
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Faraz R Khan
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+92.21.111.111.320 x200
www.emergen.biz
actually taken on a policy of not supporting
Grandstream phones. The damage to their reputation has well and truly
been done and will take a long time to repair - if it can be.
Faraz R. Khan wrote:
To be fair to the engineers at grandstream - an update to the latest
1.1.6.16 firmware
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their firmwares WAY too
quickly, earning them a bad reputation.
On Sun, 2008-04-06 at 09:41 +0500, faraz wrote:
Guys thanks a lot. I should be going with a Polycom 650 for all such
jobs.
If grandstream receives such bad reviews- how are they selling anything?
Phones hanging or voice cut-outs
://neobits.com/aastra_-_a1760--10-55_-_560m_p11472.html
Thirdlane PBX Manager: http://www.thirdlane.com/products/pbxmanager
Feel free to contact me off list if I can be of any assistance.
Regards,
Jim
ph: 408-701-9929
Faraz R. Khan wrote:
One of our clients is using
party, he won't be in the dialplan to execute 'h',
so we need some other way to invoke the ringback (step 5).
Thoughts?
Cheers
Tony
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Faraz R Khan
Chief Architect
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+92.21.111.111.320 x200
www.emergen.biz
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becomes 'free'. Other than this scenario using the 'h' extension to read
off a DB (which gets populated on pressing 5) should work just fine.
On Fri, 2008-04-04 at 13:03 +, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Faraz R. Khan [EMAIL PROTECTED] wrote:
Thinking out loud: write
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successfully?
Thank you.
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Chief Architect
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+92.21.111.111.320 x200
www.emergen.biz
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for
extensions/queues.
Quoting Gavin Henry [EMAIL PROTECTED]:
On 17/03/2008, Faraz Khan [EMAIL PROTECTED] wrote:
Good Idea and done. It is now available here:
http://www.voip-info.org/wiki/view/LDAP
The correct LDAP Schema is included:
/asterisk-1.6.0-beta4/contrib/scripts/asterisk.ldap
Just checked 1.6.0beta4 - the res_ldap.conf file still has PBX*
attributes - which I'm guessing would be confusing to any new user.
the schema file looks file though, the missing voicemail/queue part is
what we have added.
Quoting Faraz Khan [EMAIL PROTECTED]:
Did you manage to upload
]
Content-Type: text/plain; charset=iso-8859-1
On Fri, Mar 7, 2008 at 9:52 AM, Faraz Khan [EMAIL PROTECTED]
wrote:
It does work. Did you do the switch statement in extensions.conf?
If not check voip-info for Asterisk Realtime Extensions
Hi Faraz,
I just realised I never replied
A Complete Howto (with sample files) has been uploaded and can be
accessed from Here:
http://www.voip-info.org/wiki/view/LDAP
Quoting Faraz Khan [EMAIL PROTECTED]:
What about pattern matching / QueueMembers? Does that work for you? If
it does please let me know since I might be doing
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find information about how to settle asterisk variables (to get
and to set) with php programming?
I need to make a php page that settles a property of asterisk in runtime.
Is it possible? How do I do it?
Thanks in advance,
Carlos
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Faraz R Khan
Chief Architect
Emergen
:)
You need to define your own schema and mappings in res_ldap.conf. Its
just a parameter to parameter map.
Voicemail global options you cannot do (you can with Static realtime
ldap- but thats ugly)
Quoting Gonzalo Servat [EMAIL PROTECTED]:
On Fri, Mar 7, 2008 at 9:52 AM, Faraz Khan [EMAIL
speaker phone on mute
with Chanspy listening to the far end.
Quoting Thermal Wetland [EMAIL PROTECTED]:
Anyone know of a SIP phone that supports group listen?
Group listen allow you use the handset but what the far end says comes out
the speaker...it is F802 on a Norstar.
Thermal
--
Faraz R
as a 'proxy' so that all
calls are sent through it only (and IAX2 trunked, obviously). Is this
somehow possible or do I HAVE to register my phones to the local
appliance?
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Faraz R Khan
Chief Architect
Emergen Consulting Pvt Ltd
www.emergen.biz
Interesting, since my distributor did ask me to get some (for
testing). maybe he wanted to 'pre-order'
Quoting Darren Nickerson [EMAIL PROTECTED]:
Faraz Khan [EMAIL PROTECTED] wrote:
Dear all,
Just wanted to know if any one had deployed the Grandstream GXW 4024
yet. Wanted to hear any
yeah, boot off a flash card. true. guess use one of the xorcom servers
:) You guys should make one with pre-installed ABE for certain larger
customers.
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
On Sat, Mar 08, 2008 at 07:41:50PM +0500, Faraz Khan wrote:
Just think of a different
old phones do not like the MWI signal and treat it like a ring
tone.
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Dear all,
Just wanted to know if any one had deployed the Grandstream GXW 4024
yet. Wanted to hear any feedback and/or problems with this unit that
you may have experienced.
Thank you.
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Faraz R Khan
Chief Architect
Emergen Consulting Pvt Ltd
www.emergen.biz
to a quintum/grandstream SIP gateway
(completely solid state / no pc required/ consumes little power)
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It does work. Did you do the switch statement in extensions.conf?
If not check voip-info for Asterisk Realtime Extensions
Quoting Gonzalo Servat [EMAIL PROTECTED]:
On Fri, Mar 7, 2008 at 8:46 AM, Gonzalo Servat [EMAIL PROTECTED] wrote:
On Fri, Mar 7, 2008 at 5:50 AM, Faraz Khan [EMAIL
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? Is it stable?
Thanks in advance.
Regards,
Gonzalo
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Is anybody successfully using this code? Pointers would be
appreciated. Thanks.
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