Using a quad core 1.86GHz Xeon CPU here, running Asterisk 1.4.2. Noticed
the following:
Cpu(s): 4.3% us, 95.4% sy, 0.0% ni, 0.2% id, 0.0% wa, 0.0% hi, 0.0% si
30908 asterisk 18 0 188m 10m 5152 S 400 0.3 51051:13 asterisk
Asterisk is eating up all the cores running the CPU at
On 2007-04-19 at 14:56:10, David Olsen [EMAIL PROTECTED] wrote:
Definitely not. The packet dumps reveal an active refusal from the server,
just continued 401/Unauth'd responses.
So, not sure why, but it was due to my install of linux. I run a highly
customized version of Gentoo on my
Just curios, does the CS1000 now support RFC2833? Previously, I know the
NRS can only support SIP-INFO.
Leo
Jerry Geis wrote:
Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming
calls just fine. However, using outgoing call files the CS1000 is
hanging up after I answer the
Steve Finkelstein wrote:
Doug,
Were you also having issues specifically related to SIP authorization,
same as we're experiencing?
I didn't dig that deeply into it, I did a telnet into the phone, did a
show register and saw that it was having problems with the register and
the timers were
On 2007-04-18 at 22:57:39, Steve Finkelstein [EMAIL PROTECTED] wrote:
I was only able to get a stable setup after I moved my 7940 back to SIP
version 7.5
I'm working with Steve on this issue, and the phone was running 8.6 when we
originally tried it. Just for kicks i downgraded it to 7.5 this
David Olsen wrote:
On 2007-04-18 at 22:57:39, Steve Finkelstein [EMAIL PROTECTED] wrote:
I was only able to get a stable setup after I moved my 7940 back to SIP
version 7.5
I'm working with Steve on this issue, and the phone was running 8.6 when we
originally tried it. Just for
On 2007-04-19 at 12:29:50, Doug Lytle [EMAIL PROTECTED] wrote:
I was mistaken. I just checked the phone, it's 7.4
Odd. No dice either there. I must be doing something else wrong.
-d
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David Olsen wrote:
On 2007-04-19 at 12:29:50, Doug Lytle [EMAIL PROTECTED] wrote:
I was mistaken. I just checked the phone, it's 7.4
Odd. No dice either there. I must be doing something else wrong.
Do you see anything weird when logging (telnet to the ip) into the phone
and
On 2007-04-19 at 13:09:51, Doug Lytle [EMAIL PROTECTED] wrote:
Do you see anything weird when logging (telnet to the ip) into the phone
and doing a show register?
I see as follows:
cisco-7960 show reg
LINE REGISTRATION TABLE
Proxy Registration: ENABLED, state: REGISTERING
line APR state
David Olsen wrote:
On 2007-04-19 at 13:09:51, Doug Lytle [EMAIL PROTECTED] wrote:
Do you see anything weird when logging (telnet to the ip) into the phone
and doing a show register?
I see as follows:
cisco-7960 show reg
LINE REGISTRATION TABLE
Proxy Registration: ENABLED, state:
On 2007-04-19 at 14:16:49, Doug Lytle [EMAIL PROTECTED] wrote:
This what mine looked like before the firmware downgrade. Except the
timer and expires were huge numbers.
Firewall issue?
I've also had issues when compiling my own kernel and thinking it was a
good thing to enable 'SIP
I have asterisk 1.4.2 connected to a Nortel CS1000M using SIP.
Version 4.00T.
I can call into the connection just fine. I get my voice prompts and I
thought
everything was working. Then I tried a call out using the outgoing spool
directory.
The call gets placed, the phone rings, when I answer
Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming calls just fine.
However, using outgoing call files the CS1000 is hanging up after I answer the call.
I dont know why?
Thanks, for any assistance.
Jerry
my sip.conf entry is:
[Nortel]
type=friend
Not that it's probably going to help a whole heck of a lot, but...
Cisco7960 show reg
LINE REGISTRATION TABLE
Proxy Registration: ENABLED, state: REGISTERED
line APR state timer expires proxy:port
--- - -- --
On 04/19/07 23:07, Jerry Geis wrote:
I have asterisk 1.4.2 connected to a Nortel CS1000M using SIP.
The call gets placed, the phone rings, when I answer the phone it hangs up.
I have something like this for another hardware. Try to use 1.2 branch
of Asterisk. It works for me.
--
Hi all,
I've recently upgraded to Asterisk 1.4.2, coming from 1.2.14. Using my
existing Cisco 7960G handset(s). I've tried multiple installs of
asterisk 1.4.2 with multiple handsets and SIP will not authorize my
phone. I'll include some verbose log messages below to show a VALID
registration and
Steve Finkelstein wrote:
Hi all,
I've recently upgraded to Asterisk 1.4.2, coming from 1.2.14. Using my
existing Cisco 7960G handset(s). I've tried multiple installs of
asterisk 1.4.2 with multiple handsets and SIP will not authorize my
User-Agent: Cisco-CP7960G/8.0
I was only
Running 1.4.2 with a 7960G v. 8.6 and all is well...
Doug Lytle wrote:
Steve Finkelstein wrote:
Hi all,
I've recently upgraded to Asterisk 1.4.2, coming from 1.2.14. Using my
existing Cisco 7960G handset(s). I've tried multiple installs of
asterisk 1.4.2 with multiple handsets and SIP will
Doug,
Were you also having issues specifically related to SIP authorization,
same as we're experiencing?
I noticed other folks on the list aren't having any issues either. Did
it just work out of the box?
Thanks ..
- sf
Doug Lytle wrote:
Steve Finkelstein wrote:
Hi all,
I've recently
Hi all,
I upgrade my asterisk from 1.2.11 to 1.4.2 changing my entire dial plan
but I have the following errors and I'm not able to call anymore. Do you
know what can I have to do?
My Asterisk is connected to a patton with a SIP trunk.
[Mar 22 10:19:03] WARNING[16462]: chan_sip.c:12311
Bill Hackensack wrote:
On 3/21/07, *Jeremiah Millay* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
chan_zap won't compile in asterisk 1.4.2 when used with zaptel 1.4.0.
The changelog has this entry:
And you missed all the other discussions about it not working? Or, are
Hi,
(Salut thomas ;) )
I have a working trunk with a SIP provider on asterisk 1.4.1 and 1.4.2,
without problems.
Send us your sip peers configuration to try to solve your problem.
Greg.
Hi all,
I upgrade my asterisk from 1.2.11 to 1.4.2 changing my entire dial plan
but I have the
thomas,
the dialplan is quite different in 1.4.x... they use a users.conf file
for, I think, all endpoints (phones not providers)... there is no
documentation, that I have seen yet, on this... if someone has seen it
please post on the list so we can get more info on the configuration...
I just
On 3/22/07, dave cantera [EMAIL PROTECTED] wrote:
thomas,
the dialplan is quite different in 1.4.x... they use a users.conf file for,
I think, all endpoints (phones not providers)... there is no documentation,
Somebody forgot to tell me I had to use the users.conf file. Hmmm
Guess
lacy,
had the same problem with DBget... was hard to track down too... if
you are only using SIP, 1.4.x is much easier... extensions.conf loads
users.conf but does nothing with it...
this was the only reference where I found out about users.conf
; User context is where entries from users.conf
The Asterisk and Zaptel development teams have released Asterisk 1.4.2.
In addition to minor bug fixes, this release includes:
- improved SLA support, sample configurations and documentation
- fixes for incoming DTMF handling in the IAX2 channel driver
There are also two security-related
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Looks like there is a requirement for a later Zaptel the the 1.4.0
release for the zap_chan driver to build...
Stu
- --
Randomly Generated Fortune Tag:
They spell it da Vinci and pronounce it da Vinchy. Foreigners
always spell better than they
Trying to use:
Asterisk 1.4.2
Zaptel 1.4.0
chan_zap won't compile in asterisk 1.4.2 when used with zaptel 1.4.0.
The changelog has this entry:
* channels/chan_zap.c, configure, configure.ac: If we receive
ZT_EVENT_REMOVED, destroy the specified channel. (issue #7256,
Jeremiah Millay wrote:
Am I to understand that to be able to build chan_zap with this new
version of asterisk I also need to have a new version of zaptel (which
has yet to be released) that includes ZT_EVENT_REMOVED in zaptel.h? I'm
able to compile chan_zap under asterisk 1.4.1 against zaptel
On 3/21/07, Jeremiah Millay [EMAIL PROTECTED] wrote:
chan_zap won't compile in asterisk 1.4.2 when used with zaptel 1.4.0.
The changelog has this entry:
And you missed all the other discussions about it not working? Or, are you
just special and wanted your own thread?
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