Has anyone else had issues with upgrading from SIP 2.2 to SIP 4.4 on the
Cisco 7940? I'm following the directions outlined by Cisco. Is there a
trick that I'm missing?
Clues welcome.
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Is this the Adtran 624 series channel bank?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Bartosz Jozwiak
Sent: Wednesday, August 20, 2003 9:55 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port
Channel
phone works
without any problems. I think that this is more SIP related
(than phone).
- Original Message -
From: Nathan Littlepage [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 19, 2003 12:31 AM
Subject: [Asterisk-Users] Cisco 7940 7960
Has anyone had any
Title: Message
Has anyone had any
major issues with the Cisco 7940 and or 7960
phones?
Not only that. I'd hate to accidentally lay my had over that 66 block.
DC is not very forgiving no matter what amps.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Mike Ciholas
Sent: Monday, August 18, 2003 4:36 PM
To: [EMAIL PROTECTED]
Subject:
build.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steven Critchfield
Sent: Monday, August 18, 2003 5:03 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Re: LAN switches with PoE? PoE phones?
On Mon, 2003-08-18 at 16:44, Nathan
the Pingtel phones at a customer site. I
should be able to give
a report in a couple of days
Sincerely,
Andy Hester
Consero
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Nathan
Littlepage
Sent: Wednesday, August 13, 2003 8:15 AM
It doesn't make much sense to me, but it appears Robertson intends to
make money just selling pre-configured phone hardware. The sample
units from Grandstream were $60 a while ago, and $75 MSRP. Doesn't
seem like much markup, so I'm curious to see how this plays out.
I would assume they
Smells like it.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Dave Cotton
Sent: Tuesday, August 12, 2003 8:14 AM
To: Asterisk List
Subject: Re: [Asterisk-Users] Sip and One Way Audio
On Tue, 2003-08-12 at 15:05, Adelino Baena wrote:
PROTECTED] On Behalf Of Nathan
Littlepage
Sent: Tuesday, August 12, 2003 8:19 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Sip and One Way Audio
Smells like it.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Dave Cotton
Sent: Tuesday
Has anyone had the opportunity to use a PingTel phone with Asterisk?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut .
Sent: Wednesday, August 13, 2003 2:01 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] IP phone recommendation
If I can convince finance I'll get one of the hard phones and see how it
does. I'm trying to push the nifty features and java dev kit they have
for them. Besides, Mitel phones aren't much different in price than
Cisco 7900 series.
-Original Message-
From: [EMAIL PROTECTED]
Change the allow=all in sip.conf to allow=alaw and see if that works.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Senad Jordanovic
Sent: Friday, August 08, 2003 1:14 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] X-Lite - No sound +
Use erlang calculations to decide which is most idle.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Monday, August 11, 2003 10:27 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] app_queue, fewestcalls and
leastrecent
I inquired to Grandstream about their resellers and they pointed me to
an establishment that never got back to me with a quote, even after
multiple reminders.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
James Sizemore
Sent: Wednesday, August
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Nathan
Littlepage
Sent: 08 August 2003 19:50
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] X-Lite - No sound + chan_sip issue
Change the allow=all in sip.conf to allow=alaw and see if
that works
Since the DSP is software driven for the Wildcard product. Is there a
benchmark out that depicts how much processor is utilized on TDM calls
per codec that's being used?
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