Sean,
Try 'sip show channels' or 'sip show channel channelid' for the drill down. I
believe the codec in use will be displayed with either command.
Dave
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of sean darcy [EMAIL
PROTECTED]
Sent: Thursday,
We've been EXTREMELY happy with the HP 5400ZL series chassis switch. Price per
port is about 1/3 that of Cisco when it comes to POE. Price is about $100 per
port and all ports are 1Gb with POE by default -- you can't get modules that
don't have 1Gb and POE. 10Gb uplinks are available with other
Obviously we don't need 1Gb connections for VOIP :)
Phones support pass through to the desktop and VLAN tagging.
The need for 1Gb ports comes from wanting to have 1Gb at the desktop.
Dave
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson
but then you would not need the 1G uplink.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
David Gibbons
Sent: Monday, October 06, 2008 11:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PoE switch recommendations
Did you check sip.conf to make sure that the port is correctly set to 5060?
Please show the output of Cli sip show peer peernumber and the contents of
your SEPMAC.cnf file.
Dave
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent:
use Cisco7941 without callmanager software but only with SIP support.
Thanks.
--
Salvatore.
- Original Message -
From: David Gibbons [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, October 09, 2008 2:30
Sasa,
You can actually just rename the .cop file to a .tar.gz file. Cisco doesn't
have (to my knowledge) any non-callmanager SIP software. The SIP load is just a
SIP load, not a SIP load unique to generic SIP or callmanager.
Dave
-Original Message-
From: [EMAIL PROTECTED]
.
- Original Message -
From: David Gibbons [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, October 09, 2008 2:59 PM
Subject: Re: [asterisk-users] Cisco 7906g SIP
Sasa,
Sometimes I have to do a hard reset
You need to check out the chan_sccp-b mainling lists on sourceforge. There is
active development in SVN but not in tarball releases.
http://sourceforge.net/mailarchive/forum.php?forum_name=chan-sccp-b-discussion
It is very stable.
Dave
-Original Message-
From: [EMAIL PROTECTED]
but the reset
process is stopped !
Thanks.
--
Salvatore.
- Original Message -
From: David Gibbons [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, October 09, 2008 4:53 PM
Subject: Re: [asterisk-users
-
From: David Gibbons [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, October 13, 2008 3:29 PM
Subject: Re: [asterisk-users] Cisco 7906g SIP
When the 'upgrading' process fails, it means that one or more of the
required
wrote:
Hi Dave,
I don't view nothing in tftp server because the phone is stopped on
start
screen with displayed 'upgrading' and MAC address..I don't understand
what
happened after the reset. phone
Regards.
--
Salvatore.
- Original Message -
From: David Gibbons [EMAIL
Dare I ask why you want to do this?
Dave
On Oct 23, 2008, at 10:00 PM, Stephen Reese wrote:
I was thinking about complicating my Voip setup by adding CME. I found
this example here:
http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express+Integration
and here:
Gibbons
Subject: Re: [asterisk-users] Asterisk and Cisco Call Manager Express (CME)
On Thu, Oct 23, 2008 at 10:57 PM, David Gibbons [EMAIL PROTECTED] wrote:
Dare I ask why you want to do this?
Dave
I know it seems counter intuitive but I've several examples of it
being done and for me it would
You can use the Cisco phones with either the SIP of the SCCP image.
Though I do agree that the SIP image is a bit easier to setup and auto-
provision, the SCCP image is a more native (obviously) implementation.
The chan-sccp-b project has nearly every feature usable on these
phones working
Gordon,
My guess is that you're a contractor so I can understand why you'd want to keep
yourself in high demand by steering clear of the methods that simplify
deployment and redeployment.
As an employee on the other hand, I want to make things as easy and integrated
as I can in order to
Two separate networks? Did I miss something? I feel like I'm taking crazy
pills! Two separate physical networks means twice the hassle, twice the
maintenance, twice the cost, twice the headache. Not to mention the fact that
the whole idea of VOIP is to simplify IT and focus on converging data
is given priority.
Dave
David Gibbons wrote:
Two separate networks? Did I miss something? I feel like I'm taking crazy
pills! Two separate physical networks means twice the hassle, twice the
maintenance, twice the cost, twice the headache. Not to mention the fact that
the whole idea of VOIP
I've never used the Sipura phones but they probably sync with an NTP server.
My guess is that the NTP server is on the asterisk box (you can probably verify
this by checking the config of the phones and finding the option for NTP
server). It is possible that the NTP service isn't running on the
I'm glad I'm not the only one who got that. I sent them a nasty response
earlier this morning...
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards
Sent: Thursday, November 06, 2008 11:05 AM
To: Asterisk Users Mailing List - Non-Commercial
.
David Gibbons wrote:
I'm glad I'm not the only one who got that. I sent them a nasty response
earlier this morning...
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards
Sent: Thursday, November 06, 2008 11:05 AM
To: Asterisk Users Mailing
How about a call queue using the roundrobin strategy?
http://www.voip-info.org/wiki/view/Asterisk+call+queues
Dave
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christophorus
Laube
Sent: Friday, November 14, 2008 11:29 AM
To:
Hello,
When I execute parkandannounce() in the dialplan, is the extension that the
call is parked to stored in a variable? I would like to send it to an AGI
script but can't seem to figure out where the 'announcer' gets its information.
Thanks
Dave
snip
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas
According to lists.digium.com/pipermail/asterisk-dev/2006-March/019516.html
the value is stored in ${PARKEDAT}
/snip
*grin*
I guess I deserved that.
Thanks for checking.
Dave
Last I checked, Lynch mobs don't shoot people.
snip
I wonder if there would be interest in organizing a bounty for a lynching
mob, that would track down these !...@#$# silly excuses for human beings and
shoot them. If we all chipped in a few dollars I bet we could hire
someone.
/snip
--Dave
Ken,
An empty conference call or a parking lot with MOHMP3 both come to mind.
--Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio
Sent: Thursday, January 08, 2009 4:15 PM
To:
I'm confused as to why you think leaving a phone off the hook is better than
parking the call and hanging up the phone. The phone that's off the hook can't
receive any more calls after you've 'pulled' the one it was on the line with,
assuming you don't walk back to that phone and subsequently
snip
My understanding is that Charter 'telephone' doesn't use IP at all but
rather uses some additional frequency spectrum on their cable network.
Hence, the reason why faxing with their service is reliable unlike other
providers who are *actually* using VoIP.
/snip
I think what you're referring
snip
I'd be willing to bet *TWO* pennies that you're correct. I certainly was not
coming into the conversation as an expert, just stating what I'd read/heard of
their service... hence the My understanding is that... beginning to the
email. :-)
/snip
Fair enough. I get worked up when I hear the
snip
One problem to overcome is that your competitors are:
1) Literate.
2) Post to the right mailing lists.
Meftah Tayeb wrote:
/snip
Ha ha ha ha.
So, you're saying you don't want the job?
LOL.
___
-- Bandwidth and Colocation Provided by
The higher you raise the barrier for entry to the mailing list, the more you
decrease the amount good the mailing list is actually capable of doing.
(barrier height is inversely related to how much help we can provide to the
people that need help the most)
I agree with you regarding the
How pompous are we now?
What happened to the 'open source community'?
There's a give and take involved; you answer questions you know how to answer
in the hopes that someone with greater experience and knowledge of the software
will answer your questions.
Yikes.
-Original Message-
If your provider has two different IP addresses at its endpoint, you could use
iproute2 (source based routing) with two local source addresses to make sure
that there is a one-to-one mapping of source address to destination address.
Then you could have two peer definitions and an
Which firmware load? We had all kinds of trouble with 8.4.x, after being stable
for a few months on 8.3.x. Going back to 8.3.x made all of the weirdness
disappear. While we're on the cisco note, I have script to remotely reboot the
SIP firmware load Ciscos and to provision the phones based on
-Commercial Discussion'; David Gibbons
Subject: RE: [asterisk-users] Sending Calls via SIP trunk from two different IP
addresses from same Asterisk Machine
My provider has one IP and one port ONLY, I need to send for him the calls from
different IP's on the same Asterisk machine, how?
Regards
snip
Problem is that its crashing for seemingly no reason at all, no errors
on the console, no logs (that I can find), nothing in /var/lib/messages
- its puzzeling! Management is screaming like banshees, calls are
dropping like flies, and all hell is about to break loose if I can't
stop asterisk
snip
Anyone use CIsco 1760 with Asterisk
/snip
No, but I'm using 7941G-GE and 7961G-GE in a deployment of ~80 phones. Did you
have a question about implementation or are you just curious?
--Dave
___
-- Bandwidth and Colocation Provided by
I've got SIP load SIP41.8-4-1S running w/o problems in a stable environment.
I'll provide SEPMAC.cnf.xml's if requested off-list.
--Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack
Sent:
snip
On a similar subject, I have been able to get a 7961 to switch to a SIP
firmware, has anyone had any luck with this?
/snip
Yes, I have several 7961s and 7971s running SIP, same firmware generation as
the 41s
--Dave
___
-- Bandwidth and
snip
Certainly a sobering thought. Have others had to deal with this in PBX
replacement scenarios? Its a giant cost savings in this case - they are
dropping about 12 POTS lines in favor of utilizing (an underutilized) T1
trunk that was already in place.
/snip
Yes -- our alarm monitoring company
snip
We will be testing the ADT connection heavily this week. The modem
connections to my understanding are 2400 baud. Over G.711U and a T1 I
don't see why this wouldn't be as solid as a POTS line, but our tests will
tell!
/snip
We do *fax* in this way and it works like a charm. We can hit much
You could use the XML browser on the cisco 79xx series.
--Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Chamberlain
Sent: Thursday, February 19, 2009 2:18 PM
To: Asterisk Users Mailing List -
Is this a question?
Haha.
Computer won't doesn't turn on. Got blck scrn.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chuck Coleman
Sent: Wednesday, February 25, 2009 3:11 PM
To: asterisk-users@lists.digium.com
Subject:
I have several Dell boxes running onboard Broadcom and Intel NICs any haven't
had any issues. It's preposterous to make a blanket statement like that about
all Dell hardware.
Maybe you should re-compile your drivers. Or have prosupport come put a new
mobo in for you :).
-Dave
snip
Not at
Harry,
Chill on the duplicate posts. Sometimes the listserv takes time to forward the
message.
-Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Harry Vangberg
Sent: Thursday, March 26, 2009 3:25 PM
To:
This is the outgoing callerid. If you have 1200 DIDs in a range, you probably
only need to outpulse 4 digits (they already know the first six). If you want
to be able to make your callerid anything that may or may not be one of your
DIDs, you probably want 7 or 10. I pick 10 no matter what for
I had a similar situation a while ago and the fix was setting up
indications.conf:
http://www.voip-info.org/wiki-Asterisk+config+indications.conf
-Dave
snip
I configured a SIP registration with my SIP provider (SIPCall).
Everything works fine except the ring tone for the caller. The caller
The fact that you sent this again (what is that -- 3 times now?) AND with high
importance, will likely cause people to ignore your messages rather than trying
to help you.
There are few things that annoy me more than messages sent with high importance
(same category of annoyance as messages
Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
And don't top post ;)
On 3 Apr 2009, at 14:38, David Gibbons wrote:
The fact that you sent this again (what is that -- 3 times now?) AND
with high importance, will likely cause people to ignore
Yes, you can flash them back and forth as you require.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Shauger
Sent: Monday, May 04, 2009 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
snip
...routing via satellite adds about a quarter second of latency to the path.
Is that too much?
/snip
Eric,
I believe that you are mistaken. Routing via satellite adds about a quarter
second of latency PER TRIP from earth to orbit. This is simply due to the
distance a satellite is from
snipOf course, that's assuming your satellite is in geosynchronous orbit. If
It's in LEO.../snip
Singer,
You are of course correct, low earth orbit will have lower latency. I was
assuming that this user would be using a stationary link on the ground, not a
portable sat phone or an aimable
Redirect traffic with iptables like this:
Host ~# iptables -t nat -I PREROUTING -d OLD_PUBLIC_IP -j DNAT --to
NEW_PUBLIC_IP
I'm not sure if this will work for SIP. You may need the proxy to change info
in the sip messages between server and client.
--Dave
From:
Tunnel samba or nfs through ssh, rather than using sshfs, then mount using once
of those more ubiquitous standards.
-Dave
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Elliot Murdock
Sent: Wednesday, May 13, 2009 1:09 PM
To:
,
including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY
14225 USA.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: Tuesday, May 26, 2009 10:33 AM
To: Asterisk Users
I could be wrong but I don't think the cat5 limit of 100 meters applies to any
analog signaling over that copper. I believe it only applies to Ethernet
signaling.
-Dave
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Cory,
Precisely what do you mean by 'Anything other than Callmanager will essentially
be a hack'?
I've got nearly 100 Cisco 79x1s in production using Asterisk against the SIP
image. They're not 'hacked', they're set up properly against the Cisco provided
SIP image and are rock-solid stable. I
Assuming you mean the firewall in front of the client, you don't need to open
any ports as long as the VPN client is tunneling all traffic to and from the
Asterisk server.
I set NAT=yes in the config file for the extensions behind a VPN.
-Dave
From: asterisk-users-boun...@lists.digium.com
or telephone and then delete this message,
including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY
14225 USA.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent
Danny,
Just out of curiosity, can you elaborate? Anything in use for asterisk should
be in cache by the time it's needed for a SIP stream. And nothing related to a
SIP stream should ever be read directly from the disk...
Unless I'm mistaken.
Thanks
Dave
snip
Since this is internal SIP, I'd
2mb is small potatoes... unless you mean MegaBytes instead of Megabits...
I am assuming you've already implemented QOS? That is likely the problem if the
intermittent quality issue is only on calls between internal and external
parties.
If someone tries to access the yahoo homepage while
the PC LAN physically (2
separate switches), by design. So theres no web browsing etc on that 2mb
circuit.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Gibbons
Sent: 02 June 2009 15:09
To: 'Asterisk Users
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Gibbons
Sent: 02 June 2009 15:09
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Call quality - how to debug
2mb is small potatoes... unless you mean MegaBytes instead of
Megabits...
I
I've found that different types of TFTP servers return differing errors when a
file doesn't exist. You don't need the TLV file, but you do need a distro that
tells the phone it's not there correctly. I have not had ANY luck with windows
tftp servers, only linux.
-Dave
-Original
And you shouldn't need the tlv file.
-Jonathan
On Fri, Jun 19, 2009 at 8:25 AM, Sasa s...@shoponweb.it wrote:
David Gibbons wrote:
I've found that different types of TFTP servers return differing errors
when a file doesn't exist. You don't need the TLV file, but you do
need
a
distro
Mike,
1. Remove the 'line 2' entries completely from the SEPXX.XML file.
2. Change the 'Version' tag in the SEPXX.XML file. You need only change
one digit; I usually just increment the last digit.
(version1.0.0.0-9/version).
3. Restart the phone (Settings - **#**).
4. This
This may be a stupid question, but IS THERE a message waiting against your PSTN
lines?
-Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Tuesday, August 04, 2009 1:33 PM
To:
I was having the same problem with about half of my POTS lines.
I switched the polarity on the connections for those lines and the problem
disappeared.
-Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
How about a shell script on the monitoring server:
#!/bin/sh
trunk=`ssh aster...@astbox asterisk -r -x 'sip show registry' | grep USERNAME`
state=`echo $trunk | awk '{print $4}'
if state is 'Registered', yay!
else, UHOH!
EOF
Based on that ssh/shell script framework (you'd obviously need host
Yes each extension needs to be configured separately in the cisco CNF file.
I use a distinct extension on each phone (2 phones can't register to one
'extension' afaik) and ring them in order:
1,1,Dial(SIP/xx)
1,n,Dial(SIP/xx1)
1,n,Dial(SIP/xx2)
Or ring them at the same time:
and
then fall straight through to voicemail after the timeout.
Anyone have some thoughts on getting it to work that way?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: Tuesday, August 11
I am using the phones quite successfully, though I have not tried non-English
menus.
-Dave
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Wednesday, August 12, 2009 12:33 AM
To: Asterisk Users Mailing List -
I fail to see how Obama has ANYTHING to do with this.
Danny, please DO elaborate so that I don't have to go on believing that you're
a fool.
-Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
I know I'm missing something here (been a long day)...
How can I specify more than one channel in a call file?
I want to dial SIP/trunk_x and fail to SIP/trunk_y and fail to ZAP/g1...
Thanks
Dave
___
-- Bandwidth and Colocation Provided by
)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: Wednesday, August 12, 2009 3:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Call File Channel
I
then
dial out as you write it.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: Wednesday, August 12, 2009 4:10 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re
] Call File Channel
If you use a Local channel to dial it then it will fall under the same rules
Channel: Local/numbertod...@the-context-you-want
This gets a CDR produced, it does pay to check everything works the same
but it should be fine
Cheers Duncan
David Gibbons wrote:
Context: is what
My messages go through rather quickly (minutes).
Unless the lists.digium.com server is running on an Atari, it's probably NOT an
overload issue...
-Dave
snip
Are there any plans to beef up the mailing list server so that messages
can get through with less of a delay?
/snip
You probably want to set the option
CURLOPT_SSL_VERIFYPEER to FALSE.
Especially with chained certificates (cheapos from godaddy, etc), I have had
lots of trouble with CURL being able to validate a cert. That's probably
because I didn't tell it where the root certs were... but either way.
snip
the IAX quality is at best 40-50% of a SIP connection.
/snip
How is this calculated?
Thanks
Dave
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Tuesday, October 20, 2009 4:46 PM
To: 'Asterisk Users Mailing
snip
What say you to the proposal that some approaches to seeking help are
so ridiculous they should not be tolerated?
/snip
I say give me a break.
Pre-judging people doesn't work on mailing lists given the inherent language
barriers, etc.
___
--
snip
Here is a link to a reboot script http://www.dave.vc/wordpress/?p=14
that uses your ability to press keys on the phone. You could apply
the same idea to press the correct buttons to change the background
without rebooting.
I can't find the script that I found to do this, but I'll keep
snip
There are some other methods to display content on the phone screen without
editing local configs. Check http://www.ciptec.co.uk/ - commercial site but
shows the way.
/snip
If you just want to display user info on the phone, why not use the idle url
feature:
snip
Not trying to be a smart-a$$, just hoping to find something a little smoother.
Is there a better way, or is help as useless as it is starting to appear?
/snip
If you're actually 'sitting' at the *nix console, use CTRL+PageUP to scroll
back up in the buffer.
If you ARE using the console
snip
Customers in Europe all have mobile phones, while senders in North America
rarely have them ( they have answering machines, though ).
/snip
What planet/year are you/your clients living on/in? I don't know anyone who
doesn't have a mobile. Maybe it's just that they call it a cell phone
snip
Thanks for the reply. I am not getting any output from the Asterisk CLI when I
place the call. The phone give busy signal as soon as I push the first digit
of the extension #. When I call the 7961 from another extension I get the
following on the CLI - that works fine.
/snip
If the
I recently implemented a vmware host using SSDs for the VM storage.
I wish you could see the grin on my face right now. It's so fast.
Remember thought that all SSDs are NOT created equal... Be careful what you buy.
snip
On a closely related note, has anyone built a normal (not embedded)
snip
And? Noticed any significant performance advantage?
/snip
Massive increase in performance on mysql VMs with database sizes that exceed
memory size (file caching). Boot times on VMs (windows and linux) under 10
seconds.
There is no noticeable change in performance for normal operations on
I use two 'lines' though 'Line appearances' would be a better term, though
still confusing in my book.
One line for incoming, one line that auto-answers for paging.
Cisco really has so many line appearances on their phones to enable BLF using
SIP over TCP.
-Dave
From:
snip
Cisco 7960 does not do BLF (at least not on the SIP firmware) but the
7961 might. It's a shame they haven't added such features, but there we
go.)
/snip
Are you sure about this? I believe the 79xx series on 8x SIP firmware loads
does BLF with SIP/TCP, just not SIP/UDP.
-Dave
snip Cisco 7960 does not do BLF (at least not on the SIP firmware) but the
7961 might. It's a shame they haven't added such features, but there we
go.)
It does with the skinny firmware :)
The skinny channel driver also comes with the 'random crash' feature ;-p. But
truth be told I only every
snip
If you had 1gb of memory, a 200mb load with everything else would be pretty
taxing. Hope this is helpful.
/snip
What distro are you using?? If linux is using 800Mb of memory in an idle state
for anything other than file system caching, there's a problem...
-Dave
snip
My question is, Does anyone know what variable I would use to get the
information for To from these SIP calls, the below is the actual SIP packet
obtained from the CLI with SIP Debug On. Other than I stripped out the IPs
/snip
The variable you are seeking is ${SIP_HEADER(TO)}
I parse the
snip
A client has two offices in the Virgin Islands that MUST maintain data
connectivity, and there are no available leased line options to run
a P2P link between them.
snip
Is there line of sight? I've been wanting to do a long-shot wifi link and my
company would give it a shot if you want :).
snip
Just a guess, but the connection probably went from full to half duplex.
/snip
Full vs. Half duplex networking would NOT cause half duplex phone calls.
-Dave
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snip
I have multiple trunks to the same ITSP. Incoming calls to any trunk
go to the last incoming label defined in those trunks' contexts in
sip.conf.
My ITSP insists on insecure=very in the trunk context; is this the cause?
/snip
Your provider is probably sending the DID in the SIP header TO:
This may belong on -biz, but does anyone have experience with a decent and
cheap IVR/prompt recording house?
Are decent and cheap mutually exclusive?
A nice *sounding* lady would be nice... you can keep any burly voice studios to
yourself :)
Thanks
Dave
In that case, you're going to have to talk to your provider.
They SHOULD be able to easily send the DID with the call...
-Dave
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Taylor
Sent: Tuesday, December 15, 2009 5:17 AM
To:
Gmail DOES process those headers...
And a proper mail client will also parse the headers and provide unsubscribe
information/buttons based on that
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asterisk-users mailing list
I haven't had a good mailing list war in a while.
Yes, gmail DOES default to top posting, because bottom posting is silly (in
general, but especially for a client that hides quoted text (like gmail)). Top
posting is modern. And better. And doesn't make me scroll through 10 thousand
messages
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