Displaying 20 results from an estimated 300 matches similar to: "1.6.2 - Pickup and SIP Replaces header"
2009 Jul 20
0
No subject
<snip>
Replaces: pickup-9582-c0a80101-d-4 at 192.168.101.102
<snip>
This Replaces header refers to RFC3891 which is not yet supported in
Asterisk (see http://www.voip-info.org/wiki/view/Asterisk+SLA)
This INVITE fails with :
<snip>
chan_sip.c: Trying to pick up 7792 at subs
<snip>
app_directed_pickup.c: No target channel found for 7792.
If I'm dialing *87792 instead
2009 May 05
0
asterisk-users Digest, Vol 58, Issue 9
<--- SIP read from 192.168.32.245:5060 --->
SIP/2.0 481 CallLeg/Transaction Does Not Exist
Via: SIP/2.0/UDP 192.168.32.16:5060;branch=z9hG4bK7508a694;rport
From: "asterisk"<sip:asterisk at 192.168.32.16>;tag=as2ff08179
To: <sip:5386 at 192.168.32.245:5060;user=phone>;tag=c0a80101-2ce1bc03
Call-ID: 2fa28b4-c0a80101-d-9acc at 192.168.32.245
CSeq: 143 NOTIFY
2016 Jan 06
2
Asterisk 11 and old Thomson 2030S Hardphone => SIP Register/Auth Problem against V11
Hi!
I wish you all e Happy New Year first!
Allthough, I'm relative new to Asterisk, I got our server up and
Running, Softphones, ISDN, and a brand new Snom 821 are working
flawlessly. :)
Platform is Debian 8/Asterisk Packages (11) from Debian Repo.
But I am running into problems setting up 2 older Hardphones, Thomson
2030S. :(
with in my sip.conf, I have got for this hardphone:
[...]
2009 Dec 20
1
What changed in Directed PickUp between 1.6.1 and 1.6.2 ?
Hi,
I'm banging my head over this.
Usually, I'm using a SIP hardphone feature called "Call Pickup Starcode" to
enhance BLF with Directed Call Pickup :
basically, SIP hardphone (here a Thomson ST2030S) is configured to send an
INVITE message whenever a BLF is pressed while blinking.
The INVITE is build with the extension number (attached to the BLF that was
blinking and pressed)
2009 Jan 16
0
No subject
MWI-related SUBSCRIBE message to send NOTIFY messages changing phones MWI
status.
This is fine for me but I'm wondering what if I were using SIP hardphones
refusing any such NOTIFY without prior SUBSCRIBE (does such phones exist ?)
?
1. In this case, which URI shall use a hardphone to build its SUBSCRIBE
message ?
Here is a hand written example.
Which value should I substitute to foo (in this
2010 Feb 16
2
OT- Using TR-069
Hi,
Phone vendors (Snom, Thomson-Technicolor, ...) are on the way to support
TR-069 (see http://en.wikipedia.org/wiki/TR-069).
Has someone experienced with TR-069 ?
What do you think of this protocol set ?
Regards
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2020 Jun 18
0
Voice "broken" during calls
Hello Luca,
We are still playing with visualization of your data, but I didn't want
you to wait any longer for some results. I think I blame both DT and
the Pi :)
First, a look at the phone side of your Banana Pi. The first thing we
noticed is there were a LOT more packets in one direction (north towards
DT) than the other (towards the phone):
jeff at
2012 Mar 20
1
Which SIP phone "comply" with COLP feature
Hi,
I would like to test the following COLP use case :
Alice and Bob are both using a SIP phone registered on a Asterisk 10 server.
Alice dials Bob's extension.
While Bob's phone is ringing, Asterisk updates Alice phone screen with
Bob's name, so that at a glance, Alice can check she dialed the
correct number.
Before diving into Asterisk documentation, I would be happy to be
2020 Jun 13
0
Voice "broken" during calls
So the call used Alaw as Codec.
> Am 13.06.2020 um 17:23 schrieb Luca Bertoncello <lucabert at lucabert.de>:
>
> Am 13.06.2020 um 13:47 schrieb Michael Keuter:
>
> Hi
>
>> Try "sip show peer <peername>" for a phone.
>
> So:
>
> mobile phone:
> bpi*CLI> sip show peer 0049177xxxxxxx
>
>
>
>
> * Name :
2020 Jun 13
5
Voice "broken" during calls
Am 13.06.2020 um 13:47 schrieb Michael Keuter:
Hi
> Try "sip show peer <peername>" for a phone.
So:
mobile phone:
bpi*CLI> sip show peer 0049177xxxxxxx
* Name : 0049177xxxxxxx
Description :
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : default
Record On feature : automon
2007 Sep 26
1
Busy problem
Hi,
I've a huge problem with the following:
Setup:
Asterisk 1.4.11
I've got two Thomson ST2030s in an queue. After a while Asterisk logs the
following if somebody calls the queues number:
- Got SIP response 486 "Busy Here" back from 172.10.3.31
-- SIP/office1-0823d190 is busy
-- Nobody picked up in 0 ms
The phones are NOT busy (show channels show nothing). Also
2007 Oct 09
0
Thomson ST2030 firmware upgrade
Hello,
I'm trying to upgrade a Thomson ST2030 phone froms its default 1.42
firmware to the latest version (1.56) through tftp.
The phone loads the .inf file, then the correct firmware file (as stated
in the ST2030S.inf), then it reboots and loops doing these same things
again and again. The firmware version on the phone stays at 1.42.
Is there a special intermediate firmware version to
2008 Aug 05
0
When shall SIP phone reply "480 Temporarily Unavailable"
Hello,
When sending this AMI request ...
192.168.64.5 -> Action: Originate
192.168.64.5 -> Channel: SIP/9122
192.168.64.5 -> Async: True
192.168.64.5 -> Callerid: 9122 Guest2 <9122>
192.168.64.5 -> Exten: 9123
192.168.64.5 -> Context: local
192.168.64.5 -> Priority: 1
... I've got this INVITE from Asterisk
INVITE sip:9121 at 192.168.100.198:5060;user=phone SIP/2.0
2006 Nov 21
2
Handle Options Method
Hi,
I have an Alteon in test (a sip/rtp load balancer).
This Alteon sends to the asterisk box a "SIP OPTIONS" to know if
asterisk is alive.
However, asterisk sends me a 404 message and not a response like, for
example, a Thomson (200 + SDP)
I wrote a very little script (you can find it at the end of the email)
to send an Options message to asterisk/phones to try.
It works
2002 Jul 16
1
pxelinux problem
Hi. I'm hoping that someone on this list can help me with my
problem. I've been looking on my own for a question for the past few
hours at least, so hopefully this isn't just a FAQ.
Anyhow, my problem is simple (to describe). The NIC's boot agent gets
an IP address (verified with dhcpd), gets the pxelinux boot image and
correct configuration file (verified with tftpd), and then
2011 May 08
3
Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS
Hello all,
I have installed the .deb packages of the Asterisk v1.8.3.3 from the
upstream project on my Debian GNU/Linux Squeeze server and bought the
Comodo's PossitiveSSL SSL certificate to be used for my SIP/TLS
exercise. After setting up everything and trying to fix this problem,
I am still getting a 401 Unauthorized SIP message. So as of this
writing, I still cannot successfully REGISTER
2006 Dec 18
1
Thomson ST2030S and BLF
Hello.
Once again, I came up with a problem for which
I can't seem to find a solution.
I'm not able to make BLF work with Thomson ST2030 phones
and Asterisk (1.2.13).
I've set up hints in dialplan, as well as Subscibe keys
on the phone. The LED status gets updated according to
the associated line status.
However, when a phone is ringing, If I try to pickup
the call by pressing the
2009 Jul 17
0
Rsync problem : stops unexpectedly
Hello.
My problem is Rsync stops when I use it between 2 of my servers (2 NAS
Synology) ( named "*.22*" and "*.6*" ).
The problem continue...
For example :
_ Rsync run correctly between my server "*.22*" and ".6" ( in the 2
directions )
_ Rsync run correctly between my server ".6" and "*.8*" ( in the 2
directions )
_ Rsync *doesn't
2009 Jul 15
0
Rsync stops in the middle of a transfer
Hello.
My problem is Rsync stops when I use it between 2 of my servers (2 NAS
Synology) ( named "*.22*" and "*.6*" ).
For example :
_ Rsync run correctly between my server "*.22*" and ".6" ( in the 2
directions )
_ Rsync run correctly between my server ".6" and "*.8*" ( in the 2
directions )
_ Rsync *doesn't run* correctly between my
2010 Apr 26
0
How to disable dialog-info based call pickups (Was: Re: 1.6.2 - Pickup and SIP Replaces header)
Hello,
I searched this list archives and couldn't find any practical way to disable
newly introduced dialog-info based call pickups (see CHANGES file).
Suggestions ?
Regards
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