IP331 Configuration
This may help you --
http://www.klaverstyn.com.au/david/wiki/index.php?title=Provision_Polycom
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Johnson
Sent: Monday, 13 February 2012 5:57
I hope this doesn't already exist, but I couldn't find anything to help. I am
installing a brand new Asterisk server, and want to use the Polycom IP331
phones. Does anyone have any steps on how to configure these? I have
softphones working just fine, but for some reason I can't find a clear
; Week of month in which DST stops
8=last week of month
dst_stop_time: 2; Time of day in which DST stops
dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) DST
automatic adjustment
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.
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, what are the specs of the machine you are on? OS?
32 or 64bit? etc...
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asterisk-users mailing list
I'm having a tough time figuring out how to do something. If I have an
operator (which could potentially be in their own context) and an
internal-only context, is it possible to make it so the operator can
call the internal-only context but *NOT* transfer calls to it?
The idea is that the
Noah Miller wrote:
Sort of. You can create a special extension in the operator's context
with a Goto() statement. Something like this:
[operator]
exten = 100,1,Goto(internal,prompt,1)
Then in the internal context:
[internal]
exten = prompt,1,Background(who-do-you-want-to-call)
Rob Schall wrote:
This might sound like an odd question but here it is anyways...
We currently have Polycom 501 phones. We have Asterisk with
Realtime/MySQL running for our SIP/Voicemail/Extensions. When one phone
dials another, the receiving end does in fact see the callers ID. But...
Jason Fuermann wrote:
I'm not sure about the sippeer stuff, or where they get that variable.
We lookup our info in a database to set it. Also to use sipcalledrpid
you'll probably need the patch at
http://bugs2.digium.com/view.php?id=6643 .
I looked at this in the past and never made it work
Has anyone made this combination work together? I've tried everything
and can't seem to get it work right. It all compiles fine, but when
rxfax is called, I get an unknown symbol error. From my reading,
everything points to me having multiple copies of spandsp and it's maybe
calling the
Dan Austin wrote:
I thought I would give the new IMAP support a spin on my home
server, but without much luck so far.
Asterisk 1.4.0
Dovecot 0.99.14
Maildir format
C-client 2006d
The imap server is also the Asterisk server, so connections are
on the localhost.
The error posted to the logs is:
Matt Gibson wrote:
Hi Pavel,
Thanks for the config!
I modified mine so it was more minimal like yours, and it registers
just fine now. So much nicer without those big red X's!
MG
This modified config works sweet!! Any tricks to get the MWI working?
Mark
Andrew Kohlsmith wrote:
On Thursday 13 April 2006 09:02, David Cook wrote:
My cell vm goes to asterisk, not the carrier. Apparently MWI is turned
on/off with specially formatted SMS messages. Anyone know how to do this
on a Treo 600? Having the phone light from Asterisk would be HUGE ...
not
C F wrote:
I recently updated my phones Cisco 7960 phones (3 of them) in a high
volume call place, where the Secretaries use the 7960 phones to answer
inbound calls, as many as 15 simultaneous calls between all three of
them.
Since then I have had only constant problems, mainly that after 3
I upgraded to 1.2.4 today and am having issues and can't figure this
out. Here's the bottom part of a gdb and a backtrace. Any
thoughts? May roll back to 1.2.3?
Mark
Reading symbols from /usr/lib/asterisk/modules/app_saycountpl.so...done.
Loaded symbols for
I just ran into this today, on 1.2.3 with Polycom IP 501 phones.
Message was from a potential customer looking for a PBX too... imagine
that embarrassment :)
Anyone know how to get this resolved?
Thanks,
Nathan
I had this happen today, also. I've seen it happen in the past, but
Jerry Geis wrote:
All,
I had updated to 1.2.4 right when it came out. I had been working just
fine.
Today I seem to be having recuring seg faults. can explain it.
How can I find why?
Anyone else experiencing this?
I am running (2) TDM04B cards (has been working since 1.0.9)
I have a
Mark Johnson wrote:
Anyone have any idea what's causing this or how to debug it? I'm
pretty sure the root cause is with chan_sccp.so, but not sure how to
prove it.
I recently upgraded from CVS-head to 1.2.1 and the chan_sccp from
12-17-2005. Now, once or twice a week, I get
Alex Ongena wrote:
Hi,
We are using Asterisk 1.2.1 with Cisco 7940 and 7960 phones.
Most things are running fine ;-)
But, when you are calling and you want to Transfer, you need
to press first on the 'more' button (4th), then you have the
label 'Trnsfr' to Transfer.
these are the lables on
Chris Bagnall wrote:
Is this specific to the SIP firmware? I'm using chan_sccp with a few 7960s
and Transfer is definitely on one of the initial softkeys when on a call.
If it's a SIP thing, you might want to consider using SCCP.
Regards,
Chris
Yes, the SIP image did some pretty strange
Peter Fern wrote:
I'm pretty sure I've seen some commits dealing with channel locking
since 1.2.1
Brent Torrenga wrote:
Might it be related to the memory leak bug? Upgrade to 1.2.4? (shot
in the
dark, a brainstorm on my part is all)
Here's what the logfile shows. Any ideas? And is
Anyone have any idea what's causing this or how to debug it? I'm pretty
sure the root cause is with chan_sccp.so, but not sure how to prove it.
I recently upgraded from CVS-head to 1.2.1 and the chan_sccp from
12-17-2005. Now, once or twice a week, I get this on the console:
Jan 31
Anyone out there got a SIP phone (mine's a Cisco 7940) to work through a
VPN with a Netscreen 5gt? It has always worked for me with any ScreenOS
version 4.x. I had the need to upgrade it to ScreenOS 5.x and it breaks
the phone. Here's the goofy part, it works enough to still register
with
Lists Pleasants wrote:
ScreenOS 5.0x and 5.1x has some issues wit SIP. Try the policies I have
listed below.
set policcy id 1001 from Trust to Trust Local Remote SIP
permit log count
set policy id 1001 application IGNORE
set policy id 1002 from Trust to Trust Remote Local SIP
permit log
Lists Pleasants wrote:
ScreenOS 5.0x and 5.1x has some issues wit SIP. Try the policies I have
listed below.
set policcy id 1001 from Trust to Trust Local Remote SIP
permit log count
set policy id 1001 application IGNORE
set policy id 1002 from Trust to Trust Remote Local SIP
permit log
cp wrote:
The example I gave was going over a VPN with tunnel terminating in the
trusted zone. Put the polices how our traffic traverse through the
netscreen. I would config a policy for trust to untrust traffic and for
untrust to trust or untrust to global if you have MIPing going on.
-chip
Polycom User wrote:
i appear to misplaced my password for my cisco 7960 SIP Phone. Does
anyone know the procedure to recover this? I have read in the past
that you can use cisco or Cisco but this does not appear to work.
Thanks in advance.
Is this phone setup using tftp? If so, I
Victor Alvarez wrote:
Hi,
I found a problem when trying to install the module chan_bluetooth
from 'the crazy greek'. Most of installation seems fine,
chan_bluetooth.so is created and located in
/usr/src/asterisk/channels/. But when I try to start up asterisk, I
get the following
Jimmy Smith wrote:
seems every 10 sec something is happeneing on your network...
make sure your router is rebooted often if you have QOS on it has they
tend to get behind on queues..
or UDP crc checksum failing in router.. that happened to me
on a linksys
your ping is ok 60 is good
i
Geoff Manning wrote:
Info relating to the 7.5 firmware version and it failing to register. Thus
needing a reboot to fix:
I don't have any documentation, but I can tell you that the 7.5 image
caused me ALL sorts of headaches. I rolled it out to a few phones to
test, one being our
Matt wrote:
Yes,
Go into sip.conf and add this line:
progressinband=no
Thank you!!! My Cisco 7960's started acting weird with SIP version 7.5,
so I kept them at 7.4 for this reason. Works great now!
Mark
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Technical Support wrote:
Has anyone configured ast_fax (sending faxes via asterisk) with
sendmail? The creation of rules to trap all numbers
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] seems too
complicated. Does anyone have setup details to share? (I don't want
to switch MTA's).
As a
This is off list... I was really concerned about this, too!! It turns
out that it is some sort of clean up routine that runs once an hour. If
you have calls in progress on channels 3 and 4, those won't show up as
restarted!! Good Luck!
Mark
Matthew T. O'Connor wrote:
Hey, I'm up and
Ok... I asked a question a few months back about a 7960 that a user
claims to be shocking her in her ear from time to time. A few others
indicated they had similiar issues and alot of them seemed to stem from
power over ethernet. Here's what we've done... We replaced the phone,
ran two new
Jason wrote:
Hey all, I have set up my cisco 30vip using chan_skinny because
chan_sccp wont register. The problem I am having is, everytime a call
is sent to the phone Skinny/[EMAIL PROTECTED] it rings once, then asterisk
segfaults.
Man... Use chan_sccp from Sergio at:
Andres wrote:
Help is on the way:)
This is quite simple to achieve on Sipura units. There is a
parameter called
Dial Tone: [EMAIL PROTECTED],[EMAIL PROTECTED];10(*/0/1+2)
It defines the frequencies and duration of the tone. The 10 you
see near the end is the duration. Simply change
Hi Mark,
I've done this using SPA-2000, SPA-2000 can generate polarity reversal
signal, The pay-phone detects call answer and hangup by revesal
signal.
also the pay-phone must be supported polarity reversal detection.
Hi Mark,
I've done this using SPA-2000, SPA-2000 can generate polarity reversal
signal, The pay-phone detects call answer and hangup by revesal
signal.
also the pay-phone must be supported polarity reversal detection.
Anyone got any suggestions? I need to know what piece of hardware I
need
I have an interesting problem. I am attempting to install a payphone
utilizing a Cisco ATA-188. The payphone actually works, but there are
some timing issues. What happens is you pick up the payphone and the
ATA grabs a line and goes offhook. While you monkey with putting money
in and
Hi Mark,
I think ATA-188 supports polarity reversal.
Cheers,
~Madhawa
I hope I don't sound stupid, but what does that mean? I can't find a
definition for polarity reversal and how it would help me. I do see the
188 supports it, but I'm not sure what to do with it.
Thanks!!
Mark
Bryce Chidester wrote:
The CallerID that is seen by others on calls originating from your
PRI is set by your PRI provider; you have no control from Asterisk
about this as it gets overridden by the provider. You must contact
your carrier and ask them to set the CallerID for all PRI lines to
Chee Foong Chiew wrote:
Hello,
I have the following situation:
I have a PRI with 200 DID numbers and I have set up
200 sip extensions that matches the last 4 digit of
the corresponding DID numbers so that when any of the
200 DID number is called, asterisk can pass the call
to the respective
Chee Foong Chiew wrote:
Hey Mark,
Have you tested on doing transfer (blind and
attended)? Are the extensions in the CDR still
correct?
CCF
--- Mark Johnson [EMAIL PROTECTED] wrote:
Actually, I don't think they are. That was something I wanted to
research a little farther. I wish
I am looking into using a Cisco T1 device that uses MGCP. Asterisk is
talking to it fine, but I am having a hard time figuring out how to
handle channel grouping like Zap does. With Zap, I can take channels
1-23 and make a group g1 out of it and then simply dial Zap/g1. Does
MGCP have this
I have read of people attempting to do this, and I just wanted everyone
to know about what we've discovered about the Cisco 7750. If you don't
know what it is, it's basically a blade server. I have 1 power blade, 1
alarm processor, 2 system processing engines and 1 multi-service route
Trey Scarborough wrote:
- Original Message - From: Mark Johnson
[EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, June 21, 2005 8:56 AM
Subject: [Asterisk-Users] Cisco 7750
I have read of people attempting to do this, and I just wanted
everyone to know about
I have a pretty strange problem. I have about 100 DID's that come down
a PRI from SBC in the United States. On Friday afternoon, one of my
DID's flipped out. When you call it, the SBC operator comes on and says
that the line has been disconnected. I contacted them and they ran test
and
Chris Coulthurst wrote:
If you have a loopback plug, I would take that PRI down, unplug the NIU from
the Asterisk box, and plug that RJ45 loopback plug in to the NIU, and call
the telco, have them run a loop test on your circuit. Out here in
Qwest-land they can usually get a tester on it and
Joseph wrote:
On Thu, 2005-06-09 at 02:24 +1000, Julien Goodwin wrote:
On 8/06/2005 11:37 PM, Sergio Chersovani wrote:
Joseph ha scritto:
When sending a call to a line defined on chan_sccp, there is an
error on the console that says:
Jun 7 08:22:29 WARNING[3924]:
Joseph wrote:
On Thu, 2005-06-09 at 11:57 -0400, Mark Johnson wrote:
I just downloaded the latest chan_sccp and am having problems with
internal to internal calls with callerid. I added a few debug lines to
the code to help sort it out, but here's what happens... Exten 581
calls 580
I am trying out the new atxfer feature from CVS-HEAD. I set atxfer
equal to *7 and it seems to work OK. I am having a problem getting it
to work the way a receptionist would want. If an extension calls me, I
hit *7 and I hear the voice say transfer. I dial another extension.
If the newly
Garth Brown wrote:
I have my Asterisk server all setup. But have an odd problem and hope
someone here can help.
I have a Polycom IP 300, a Grandstream GXP-2000, and an installation
of X-Lite. They can each call each other just fine
(extension-to-extension). I can also dial-in from the
Vikram Rangnekar wrote:
static ! Get your carpets washed and use static guard on it.
Thank you everyone for the replies. After doing some testing, it has
been determined that it was the phone that was the cause of the user
being shocked. We could relocate the phone, switch to a power
Ok, guys... Please be gentle with me. I have what is going to be the
strangest question you will have ever heard, but I have no idea what to
tell this person.
I set up Asterisk 3 or 4 weeks ago, everything is running smooth. My
receptionist has told me on two different occasions that she
Dan Austin wrote:
Yup. I even suspected it was a 7960 before I got that far in your
email.
It hasn't happened to any of my users, but I heard about it at
a Cisco users group meeting, from a number of people representing
a different companies.
Cisco was present and stumped, I have heard any
Eric Alexander wrote:
Are you using POE from a 3550? We have had similar problems, upgrading the
firmware on the switch has reduced the occurrences. The Cisco phones are not
always nice in an environment with a lot of static electricity.
POE is coming from a 3500XL I think. It just weird
Marty Mastera wrote:
Can anyone help me to understand what the significance of this output is?
May 17 10:50:23 DEBUG[2030]: Didn't get a frame from channel: SIP/105-1ae4
May 17 10:50:23 DEBUG[2030]: Bridge stops bridging channels
SIP/105-1ae4 and SIP/outbound-7dc3
I searched for these phrases
Mark Brown wrote:
Hi Everyone!
Is there any hope for us newbie plebs who want to also get hold of the
updated Cisco firmware?
I need to get a 7910G updated to work on SIP..
Any help on obtaining the updated firmware quickly and painlessly
would be great J
Cheers
M
7910 does not have a SIP
Joseph wrote:
On Tue, 2005-05-17 at 14:30 +0100, Mark Brown wrote:
Thanks for that Mark... doesn't sound promising then :(
7910 does not have a SIP image and looks like it never will. I have
about 40 of these stupid things that I can't get to work 100% with
skinny or sccp. If you ever figure
Manjit Riat wrote:
I am going to buy some IP phones from them but I sent them an email
couple of weeks ago and got no reply. Has anyone ordered anything from
them? Any other places that I can buy from? Sorry if its a wrong post.
I have ordered from them with their web shopping cart and it went
I am having trouble with zttest on a Tyan board, dual AMD Opteron's on
FC3. Tried different kernels, no IRQ sharing, everything looks in
order. My zaptel modules load fine, but if I run zttest, it just
hangs. Below is the strace output and you can see where it stops.
Anyone have any ideas?
Damian Funnell wrote:
1. Check that the TDMP is on it's own IRQ (much to our
embarrassment our card wasn't at the time, so we had to play
with it a bit to get it to occupy a unique IRQ).
2. Disable hyper threading on the Xeon CPU.
3. Uninstall our SCSI hardware and replace it
Can someone please help me. I am currently HEAD as of about 5 days ago
(stable was giving me all sort of problems, upgraded per other users
suggestions) on an Intel mainboard using a mix of Cisco 7960/40 SIP and
7910 SCCP. Can someone please explain what the following means? When
this
Is there a way to get a download of asterisk from cvs-head as of like 3
weeks ago? Having some weird problems and most people say that alot of
these things have been introduced over the last few weeks.
Mark
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Sean Kennedy wrote:
Anton Krall wrote:
Guys, what does hint do in a dialplan and how do you use it?
I have been trying to figure this out for a while now, even posted a
question on the list, to which no one replied.
Any details would be apprecaited if you find this one out. I want to
use
Andrew Kohlsmith wrote:
On May 4, 2005 12:05 pm, Matthew Boehm wrote:
May 4 10:57:04 WARNING[25650]: chan_zap.c:4394 my_zt_write: Write
returned -1 (Resource temporarily unavailable) on channel 2 - audio may
have been lost
I think that something in asterisk (not zaptel) changed in the
Steve Hanselman wrote:
I think it's displaying the name of the line that the call is coming in on,
but you're expecting the name of the calling party (as I was!)
Steve
I looked and there was a change in the sccp.conf file for head with the
addition of cid_num and cid_name. I am going to test
I am running Asterisk HEAD and the latest mayday version of
chan_sccp. Everything is going fairly smooth but every once in a while
I get a 7910 to lock up. If I do a show channels in the CLI, I get
the following and it never goes away. While this is happening, the
phone can not be reached.
I upgraded to CVS Head last night to help fix my SCCP problems and now
my SIP installation is having issues. If I restart Asterisk, my SIP
phones may take up to an hour to register correctly so I can place calls
to them. They immediately go to voicemail as being busy. If I do a
sip reload I
Julien Goodwin wrote:
Then why haven't you sent a backtrace? If I can see why it's crashing
then I can fix it.
Thanks,
Julien
chan_sccp project lead
The general consensus was that I needed to be running HEAD to make this
work properly. I upraded last night to HEAD and my SCCP stuff seems to
Mark Johnson wrote:
Julien Goodwin wrote:
Then why haven't you sent a backtrace? If I can see why it's crashing
then I can fix it.
Thanks,
Julien
chan_sccp project lead
The general consensus was that I needed to be running HEAD to make
this work properly. I upraded last night to HEAD and my
Mark Johnson wrote:
I upgraded to CVS Head last night to help fix my SCCP problems and now
my SIP installation is having issues. If I restart Asterisk, my SIP
phones may take up to an hour to register correctly so I can place
calls to them. They immediately go to voicemail as being busy
Adam Goryachev wrote:
The Polycom IP 600, Cisco 7960, and apparently the SNOM (some model)
phones can all do what he wants. ie, have multiple lines with blinking
red lights when a call arrives on that line.
The polycom ip600 and cisco 7960 both have 6 lines available.
Regards,
Adam
I am
Joseph wrote:
The cisco 7960 works well with * and SIP.
Out of curiosity I loaded the ccm version 7.1 and tested it briefly with
CVS HEAD * and latest chan_sccp.
The interface when using ccm load on the phone is certainly different.
Things I don't see how to fix are:
o Setting the date and time on
Joseph wrote:
What if you run it on HEAD?
I've been scared to try it. I just went live with this last week.
Everything is great except the 7910's. I'm downloading HEAD as we
speak. Anything to be aware of or look out for?
Mark
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I just had a very successful installation of Asterisk and have a
question. On my 7910's using the Skinny protocol, the user does not
hear ringing when they make another call. I found a patch that makes
the ringing work, but something is still wrong with it. If I use the
7910 to make
Michael Welter wrote:
Do SIP-SIP calls have static? If you don't have SIP phone then you
can use X-lite.
Arrange you dial plan so an incoming PSTN call can call an outside
number--from outside dial your system and then make an outside call.
This call will be bridged on the Digium card. Do
Jorge Mendoza wrote:
Mark,
Could you please post the models of your first and second mobo?
Thanks
The first, that didn't work correctly the the TE400P was an Asus P4 2.4
Ghz. The model that does work correctly is an AOpen P4 2.0 Ghz.
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Michael Welter wrote:
Mark Johnson wrote:
Michael Welter wrote:
Try 'vmstat 1'--are you getting 40% system utilization every n
seconds? If so, unload the wcfxo and wcfxs modules and test again.
Does anyone have some suggestions on how to get rid of this static on my
Digium card? I am supposed
Andrew Kohlsmith wrote:
On April 26, 2005 06:19 pm, Mark Johnson wrote:
Does anyone have some suggestions on how to get rid of this static on my
Digium card? I am supposed to go live tomorrow night and will get shot
if it's like this!!
Lack of planning on your part does not constitute
Andrew Kohlsmith wrote:
Try these things:
Software:
- don't play with gains on PRI or T1 unless you have echo or too loud/quiet.
Static isn't caused by screwy gains and on digital circuits it technically
shouldn't ever need to be adjusted
- turn echocancel off for now
- I notice you've got
Matt Klein wrote:
ask your upstream.
Not sure what you mean. This T1 is in good working order with a
different system. Do you mean call the telco or Digium?
Mark
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Michael Welter wrote:
Do SIP-SIP calls have static? If you don't have SIP phone then you
can use X-lite.
Arrange you dial plan so an incoming PSTN call can call an outside
number--from outside dial your system and then make an outside call.
This call will be bridged on the Digium card. Do
Michael Welter wrote:
Mark Johnson wrote:
I tested and I do in fact get from 40-50% system util every 5 seconds
or so. After removing the wctdm module, the system util drops to 0
and stays there. I have not loaded the wcfxs and wcfxo modules
because I could never get them to work right. I
I need some serious help!! I have been in the process of building an
Asterisk system to replace a Cisco Call Manager. I have most everything
setup, but only got to test the PRI today. To make a long story short,
my Call Manager is half broken and I need to go live with * a lot sooner
than I
Michael Welter wrote:
Try 'vmstat 1'--are you getting 40% system utilization every n
seconds? If so, unload the wcfxo and wcfxs modules and test again.
I tested and I do in fact get from 40-50% system util every 5 seconds or
so. After removing the wctdm module, the system util drops to 0 and
Derek Conniffe wrote:
Would anyone know why Voicemail in * doesn't get the DTML keypresses
from my Cisco 7940 running SIP (POS3-07-3-00) ? Is it something to
do with dtmf_avt_payload: 101 setting in SIPDefault.cnf in the tftp
server?
Thanks for any help!
Derek
I have the same line in my
I am building a click to dial and CRM type web page and I'm having
trouble with something. I can make everything in the manager api work
as documented, but I can't seem to get a grip on how to tell what the
callerid is of an active call. Example: I know that on phone SIP/101
that there is
Here is what I am attempting to do (which works well on Cisco Call
Manger). I have some 7960's that have multiple lines on them. The
second line specifically is a helpdesk line that is shared among
multiple phones. Here is how I am making that line ring on multiple
phones, maybe you have
Rich Adamson wrote:
Here is what I am attempting to do (which works well on Cisco Call
Manger). I have some 7960's that have multiple lines on them. The
second line specifically is a helpdesk line that is shared among
multiple phones. Here is how I am making that line ring on multiple
Adi Linden wrote:
I believe the current implementation for vm notification is to use
a sip 'notify' message to turn on the mwi, and the sip protocol
implementation within * does not support sending 'notify' messages
to multiple phones. (E.g., how would * even know how many phones
you are trying to
Brian M. Arlinghaus wrote:
I've got 25 7960s with different mailboxes set for different lines.
The MWI indicator (red light) comes on if there are messages in either
of the mailboxes. However, on the display, an envelope shows up next
to the line that has the voicemail waiting. Therefore I
Doug Lytle wrote:
Mark Johnson wrote:
This may be OT, but I can't seem to find how to do this. I have
7940/7960's with Skinny on them. When you start pressing numbers on
the dialpad, you start building a number to dial. When I install
SIP, that functionality goes away. You have to hit
Asterisk wrote:
when booting the cisco 7960 with SIP image 7.3, the Configuring VLAN
takes in order of minutes before it issues a DHCP request .
Does anyone else have this problem - is there any way of disabling the
VLAN configuration at all ?
We are not using Cisco switches.
Julian
I upgraded
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