Is it possible to get sip to listen on two ports (say 5060 and 5061)?
Maybe its not necessary, but I'm trying to get a PAP2 to work with 2
lines configured behind a Linksys router with NAT.
I've noticed the default config in the PAP2 is to use 5060 for line 1
and 5061 for line 2.
I'm guessing
Please disregard this message.
Evidently changing the port required a power cycle on the PAP2.
On 8/26/06, Mr. Jones [EMAIL PROTECTED] wrote:
Is it possible to get sip to listen on two ports (say 5060 and 5061)?
Maybe its not necessary, but I'm trying to get a PAP2 to work with 2
lines
on her phone at the
same time it rings the execs phone, and have one light if he is on
the phone
Also FOP works great
On Jul 23, 2006, at 3:42 PM, Mr. Jones wrote:
Thanks Sebastian -
You're right - I have limited experience in this area :)
I think the idea below is workable, except we actually
Hi Folks,
I'm trying to use the Queue feature to essentially implement a
multiple call appearance situation for some of our executives.
Essentially I have a queue defined per executive like:
exten=9495551212,1, Queue(stever|tTr|||25)
exten=9495551212,2, Goto(druid-users,1212,1)
So the user
Thanks Guido -
I tried that and still have the same problem. The call never seems to
leave the queue.
Any other ideas?
On 9/2/06, Guido Hecken [EMAIL PROTECTED] wrote:
-Ursprüngliche Nachricht-
Von: Mr. Jones [mailto:[EMAIL PROTECTED]
Gesendet: Sonntag, 3. September 2006 01:12
Hi Guido -
Evidently I needed to add a timeout to the queue itself.
Thanks,
Brian
On 9/3/06, Guido Hecken [EMAIL PROTECTED] wrote:
-Ursprüngliche Nachricht-
Von: Mr. Jones [mailto:[EMAIL PROTECTED]
Gesendet: Sonntag, 3. September 2006 06:10
An: Asterisk Users Mailing List - Non
Hi Folks,
I'm getting a lot of these messagse now with the Grandstream phones and Asterisk
Incoming call: Got SIP response 415 Unacceptable Content-Type back
from 192.168.1.X
I don't think I noticed them when I only had one or two phones hooked
up for testing, but I suppose I could have just
Hi Folks,
We're trying to roll Asterisk out to production and are having a few
complications.
Most specifically we have G711 for our inbound origination, but would
prefer G729 for outbound termination, so far so good - it appears that
dtmfmode=auto works in both cases.
The area I'm having
=user for inbound and type=peer for outbound. Have different
codec settings for each of them.
Mr. Jones wrote:
Hi Folks,
We're trying to roll Asterisk out to production and are having a few
complications.
Most specifically we have G711 for our inbound origination, but would
prefer G729
Hi Folks,
Has anyone seen these errors repeatedly in the CLI?
Incoming call: Got SIP response 415 Unacceptable Content-Type back
from 192.168.1.209
We're using GXP-2000s.
TIA,
Brian
___
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, application/simple-message-summary,
application/octet-stream, application/pidf+xml,
message/sipfrag;version=2.0
Content-Length: 0
On 9/25/06, Anthony Cennami [EMAIL PROTECTED] wrote:
Bidirectional SIP trace usually helps in these situations.
On 9/25/06, Mr. Jones [EMAIL PROTECTED] wrote:
Hi
I'm still getting these errors if anyone has any ideas I'd be truly
appreciative.
On 9/25/06, Mr. Jones [EMAIL PROTECTED] wrote:
Could the problem is this: Content-Type: unknown?
Reliably Transmitting (NAT) to 192.168.1.228:5060:
NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP
Hi Folks,
I'm curious if there's anyway to force Asterisk to transcode for
certain handsets.
Specifically we have an inbound SIP origination service which uses g711.
We're having bandwidth issues with a client and would like to force
Asterisk to transcode to g729 until we can get their T1 in
Thanks -
This worked. I swear I was getting a 503 or something weird before
when I did this but it seems to be working now.
On 9/28/06, Andres [EMAIL PROTECTED] wrote:
Mr. Jones wrote:
Hi Folks,
I'm curious if there's anyway to force Asterisk to transcode for
certain handsets.
All you
Hi Folks,
I'm not sure if this is possible, but I'd like to give users the
option of transfering to an employee's cell phone when they get to
their greeting. This is a feature that is common on Nortel KSUs.
Is there an easy way to do this on a per employee basis? I can see
how it can be done
or be transfred to
thier
cell phone.
I also created a macro where users can dial an extension and set thier
mobile number. Let me know if you want it.
- Original Message -
From: Mr. Jones [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
I'm having problems with conference calls (3-way) when I have my codec
forced to g729 in sip.conf.
I'm using Grandstream 2000s.
If enable both g711 and g729 then 3 way calling and transfers work.
I'm not sure why this would matter?
Here's the error:
Oct 13 13:54:45 NOTICE[31184] chan_sip.c:
Is there some way I can tell?
On 10/16/06, Thomas Kenyon [EMAIL PROTECTED] wrote:
Mr. Jones wrote:
I'm having problems with conference calls (3-way) when I have my codec
forced to g729 in sip.conf.
I'm using Grandstream 2000s.
If enable both g711 and g729 then 3 way calling and transfers
Right -
I get the error on the console - I just can't tell how many
transcodes are occuring at any given point in time...
On 10/18/06, Thomas Kenyon [EMAIL PROTECTED] wrote:
Mr. Jones wrote:
Is there some way I can tell?
On 10/16/06, Thomas Kenyon [EMAIL PROTECTED] wrote:
Mr. Jones wrote
Hello Folks,
I'm an Asterisk newbie, that being said I have managed to get an
SPA941 working with 1.2.8. I've got some issues (like getting the
voicemail button to work as it should, and making the message
indicator light work) but overall I'm pretty happy.
I'm now trying to get a PAP2-NA to
Please disregard - sometimes I think posting these emails is the key
to solving all my problems ;)
Evidently I wasn't reloading the files (I thought I was).
Brian
On 5/31/06, Mr. Jones [EMAIL PROTECTED] wrote:
Hello Folks,
I'm an Asterisk newbie, that being said I have managed to get
Has anyone fed a Nortel BCM from Asterisk?
I'm interested in switching our company over, but don't want to
replace all the handsets in one fell swoop.
I imagine some of the PRI cards can emulate a switch?
I'd still like to pass CallerID into the Nortel, etc but all the
external traffic would
Excellent. -
So I can basically make a crossover cable to my Nortel, and pass calls
to the old phones from the PTSN (via my VOIP originator ) in to it?
I guess I'm off to look for sample configs.
Thx
Brian
On 6/3/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
- Mr. Jones [EMAIL
Hi Folks,
I'm trying to test out Asterisk overall.
I'm having some problems with DTMF. Currently I'm playing with DISA,
but I'm worried this will happen when I get to implementing AAs etc.
I have a free SIP trunk from IPKall that I'm trying to make work.
I'm able to receive calls, and I've
codec.
inband DTMF does not work with any other codec.
Mr. Jones wrote:
Hi Folks,
I'm trying to test out Asterisk overall.
I'm having some problems with DTMF. Currently I'm playing with DISA,
but I'm worried this will happen when I get to implementing AAs etc.
I have a free SIP trunk from
notification.
A couple of other alternatives maybe to create a queue, or possibly go
with a side car type device.
I'm open to any and all input.
Best,
Mr. Jones.
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asterisk-users mailing list
Thanks Sebastian -
You're right - I have limited experience in this area :)
I think the idea below is workable, except we actually want it to work
in the other direction - sort of.
Essentially we want the receptionist to screen the calls when she's
available. The executive should have option
Yes this is what I want.
I guess the question is what is the best way to do it?
Use a Queue? or something else?
On 25 Jul 2006 13:25:45 +0200, Benny Amorsen [EMAIL PROTECTED] wrote:
J == Jones [EMAIL PROTECTED] writes:
J Hi Folks, We're migrating from a conventional KSU/PBX to Asterisk
J
I have had the same experience with a Grandstream order from them - 7
days and no product.
They even told me it was shipping Monday, but couldn't produce a
tracking number on Tuesday.
Pretty lame.
On 8/9/06, Tom [EMAIL PROTECTED] wrote:
Is anyone else having problems with them? Order placed
I'm trying to get inbound DIDs working via SIP.
I have 20 DIDs coming in via a single SIP profile in sip.conf.
I was hoping to have these matched in extensions.conf, so I have setup
lines like this:
exten=949271,1, Goto(mainmenu,s,1)
Unfortunately these aren't getting matched and I'm
PROTECTED] wrote:
Perhaps the context in sip.conf doesn't match the context in the dial plan.
From: [EMAIL PROTECTED] on behalf of Mr. Jones
Sent: Fri 8/11/2006 2:34 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Inbound Calls SIP/2.0 404 Not Found
of Mr. Jones
Sent: Fri 8/11/2006 2:34 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Inbound Calls SIP/2.0 404 Not Found
I'm trying to get inbound DIDs working via SIP.
I have 20 DIDs coming in via a single SIP profile in sip.conf.
I was hoping to have these matched
/index.php?page=Asterisk+Dialplan+Patterns
Kevin
Mr. Jones wrote:
I'm trying to get inbound DIDs working via SIP.
I have 20 DIDs coming in via a single SIP profile in sip.conf.
I was hoping to have these matched in extensions.conf, so I have setup
lines like this:
exten=949271,1, Goto(mainmenu
Yeah...
I tried the NoOp function someone gave me above and I'll I'm getting is s
I'll go back to the provider
On 8/11/06, C F [EMAIL PROTECTED] wrote:
s, means that it got an incoming call, but no exten came with it.
On 8/11/06, Mr. Jones [EMAIL PROTECTED] wrote:
I double checked
Actually it looks like I am getting the number but its coming through weird:
This is what sip debug gives me:
Looking for s in test-context (domain 9495551212)
So clearly I am getting the number, just not sure if its formated ok?
On 8/11/06, Mr. Jones [EMAIL PROTECTED] wrote:
Yeah...
I
.
On 8/11/06, Hermann Wecke [EMAIL PROTECTED] wrote:
Mr. Jones wrote:
I have 20 DIDs, some I want to send to a menu, most directly to an
extension.
sip debug is (really) your friend. It should give you the [context]
where your DID is being send to and the 404 not found error also.
A particular
Ok - now maybe we're getting somewhere.
I didn't know I had to register them?
This is inbound only and the provider doesn't require that - so do I
just makeup a username?
I currently have the provider as a SIP peer.
On 8/11/06, Rushowr [EMAIL PROTECTED] wrote:
Uh, what's your Register
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of Mr. Jones
Sent: Friday, August 11, 2006 10:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Inbound Calls SIP/2.0 404 Not Found
Actually it looks like I am
So it looks like the information is coming through in the SIP header.
Is there anyway to avoid the register command - at the end of the
day I may have 100s of DIDs and I don't want to have to set them up by
hand.
Is it possible to fix what Asterisk thinks the extension is be resetting it?
This is essentially a follow-up to my previous email on the 404 I was
seeing with my DIDs.
I think it maybe more involved with the SIP headers I'm receiving from
the company providing my origination.
Here's what's interesting.
I have inbound 800 service and outbound termination from provider
Hi Rich,
I'm using a wholesale voip origination provider - they don't deal with
end users. As such they have statically defined my Asterisk box on
their end - there's no registration or authentication by my system
with theirs - other than them hardcoding the destination IP of my
server in their
Thanks Rich -
Maybe I'll try the dev mailing list.
I'm not that familiar with the protocol level as well.
I'm thinking its related to one of those two items (user=phone or the
Contact: being blank).
I've looked through all the configs and I don't seem to see any way to
have the Contact fall
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