Displaying 20 results from an estimated 28 matches for "rinstanc".
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rinstance
2010 Mar 26
2
need help on setup rtp directly between 2 sip clients
...DY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.26
Supported: replaces, timer
Contact: <sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>>
Content-Length: 0
Reliably Transmitting (no NAT) to 192.168.1.2:34312:
INVITE sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2dfa6cce;rport
Max-Forwards: 70
From: "Do Nguyen Ha" <sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5>
>;tag=as2886cf30
To: <sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176>
Contact: &l...
2010 Jul 28
0
what is rinstance parameter in sip header
hello
i was wondering what is the use of "rinstance" in SIP Headers. I noticed
that this parameter is visible only when there is NAT invloved.
I am experiencing one way audio when dialing a registered user by his
IP:port. I this case "rinstance" parameter is missing.
when i dial "SIP/username" audio is fine but when i dia...
2007 Sep 20
0
Video doesn't work for outgoing call?
...the manager API.
There is no video, either.
Any idea why video doesn't work for outgoing call from asterisk?
cwhuang*CLI> sip set debug peer 403
SIP Debugging Enabled for IP: 172.16.148.129:36042
Reliably Transmitting (NAT) to 172.16.148.129:36042:
OPTIONS sip:403 at 172.16.148.129:36042;rinstance=24208107cb163901 SIP/2.0
Via: SIP/2.0/UDP 172.16.148.1:5060;branch=z9hG4bK0dcbc40d;rport
From: "asterisk" <sip:asterisk at 172.16.148.1>;tag=as25cd8375
To: <sip:403 at 172.16.148.129:36042;rinstance=24208107cb163901>
Contact: <sip:asterisk at 172.16.148.1>
Call-ID: 18ce...
2010 Jul 28
2
Nat issue one way audio on IP dial
...is IP:Port (sip
uri), call is connected fine and he can hear the called user but the called
user can not here the caller voice.
If the caller calls the other user by username instead of IP:Port , the
voice is perfect both ways.
what i have noticed is that IP:Port dial is missing a parameter "rinstance"
in "Contact" , "To" headers for adf. what is "rinstance" for? Also something
with "Contact" header seems fishy. or RTP issue.
that is
Dial(SIP/adf,30,r) works fine with bothway audio, but
Dial(SIP/116.18.35.235:28614,30,r) has one way audio....
2006 Dec 28
1
one way rtp stream (Sent alwax to 127.0.0.1)
Asterisk version 1.2.14
I use snom190 and xliteV3 as sip phones.
asterisk send the rtp stream never to the xlite softphone.
Any hits for me?
*CLI> rtp debug
RTP Debugging Enabled
-- Executing Dial("SIP/xlite-007918f0", "SIP/snom") in new stack
-- Called snom
-- SIP/snom-00797110 is ringing
-- SIP/snom-00797110 is ringing
-- SIP/snom-00797110 answered
2016 Sep 02
2
Asterisk 13.11 realtime problem registering phones
..._config_mysql.c:1162 require_mysql:
Realtime table general at ps_contacts: column 'via_port' cannot be type
'int(11)' (need char)
[Sep 2 15:38:46] ERROR[2098]: res_pjsip_registrar.c:411
register_aor_core: Unable to bind contact
'sip:2001 at 192.168.2.203:57776;transport=UDP;rinstance=48d5c7d09b9f2525'
to AOR '2001'
== Contact
2001/sip:2001 at 192.168.2.203:57776;transport=UDP;rinstance=48d5c7d09b9f2525
has been deleted
The mysql warnings have always been there since version 13.0 and
the "Unable to bind contact..." error has also been present...
2010 Jul 22
0
SIP URI Dial has one way audio
...ch it was registered. like this,
Dial(SIP/XYZ at 117.58.x.x:5062,30,rtT)
during conversation audio was one way just like before (calling party can
hear called party but called party can not hear calling).
after taking debug trace of both methods what I found was that a SIP HEADER
parameter "rinstance" was missing in "to" and "INVITE" header fields when
dialing via IP:PORT. I think this parameter is assigned automatically by
asterisk.
*NORMAL DIAL *
INVITE sip:XYZ at xxxxxxxxxxxx:28614;rinstance=0266b8b94f488588 SIP/2.0
To: <sip:XYZ at xxxxxxxxxxxx:28614;rinstance=0...
2016 May 16
2
Asterisk PJSIP Multi-tenant
...6 create_out_of_dialog_request: Unable to
create outbound NOTIFY request to endpoint 1001 at sip.domain.com
WARNING[2208]: res_pjsip_mwi.c:379
send_unsolicited_mwi_notify_to_contact: Unable to create unsolicited
NOTIFY request to endpoint 1001 at sip.domain.com URI
sip:1001 at 95.250.29.3:50673;rinstance=1af959e7c0059fc4
Regards
El 16/05/2016 a las 02:52, George Joseph escribi?:
>
>
> On Sun, May 15, 2016 at 12:00 PM, Annus Fictus <annusfictus at gmail.com
> <mailto:annusfictus at gmail.com>> wrote:
>
> Hello List,
>
> following this thread:
>
>...
2007 Feb 01
2
strange caller display
...UzYzBmOTQ2NGUyYTE2OTAxYTRlYTk4ODAxOTA..
CSeq: 2 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Contact: <sip:85236418505@10.201.0.224>.
Content-Length: 0.
.
#
U 10.201.0.224:5060 -> 10.0.0.25:2750
OPTIONS sip:9000220002@10.0.0.25:2750;rinstance=f136277835976893 SIP/2.0.
Via: SIP/2.0/UDP 10.201.0.224:5060;branch=z9hG4bK3069abdf;rport.
From: "asterisk" <sip:asterisk@10.201.0.224>;tag=as77042273.
To: <sip:9000220002@10.0.0.25:2750;rinstance=f136277835976893>.
Contact: <sip:asterisk@10.201.0.224>.
Call-ID: 5dae571...
2008 Jul 21
1
Problems w/Asterisk Realtime + MySQL + SIP
Hi all, Asterisk is great but I'm having issues with setting up
realtime for our call center, which is needed for login integration
with the rest of our applications (telephonists' web interface, etc.).
I have reviewed a large number of previous posts to the mailing list
and the voip-info wiki to no avail.
Setup is as follows:
Linux 2.6.23 (gentoo) / AMD Athlon(tm) 64 Processor 3000+ /
2016 May 15
2
Asterisk PJSIP Multi-tenant
...the
Asterisk console:
ERROR[1748]: res_pjsip.c:2946 create_out_of_dialog_request: Unable to
create outbound OPTIONS request to endpoint 1000 at sip.domain.com
ERROR[1748]: res_pjsip/pjsip_options.c:350 qualify_contact: Unable to
create request to qualify contact
sip:1000 at 95.250.29.3:53570;rinstance=d90827763e4353c0
in the aor section I'm using:
qualify_frequency=30
Any hint?
Regards
2016 Feb 15
2
Asterisk 13.6.0/The simplest TCP configuration does not work
...t=TCP
Transport: TCP
Sent-by Address: 192.168.1.15
Sent-by port: 47053
Branch: z9hG4bK-d8754z-5e3d9f441f1de1d3-1---d8754z-
RPort: rport
transport=TCP
Max-Forwards: 70
Contact: <sip:5678 at 192.168.1.15:47053
;rinstance=bea6f11f37c55605;transport=TCP>
Contact URI: sip:5678 at 192.168.1.15:47053
;rinstance=bea6f11f37c55605;transport=TCP
Contact URI User Part: 5678
Contact URI Host Part: 192.168.1.15
Contact URI Host Port: 47053
Contact...
2007 Sep 25
1
Help with Sip Registration
...d
the messages, I got below. Please help me in solving the
problem.
Thanks in advance,
Treesa
REGISTER sip:192.168.12.160 SIP/2.0
Via: SIP/2.0/UDP 192.168.25.116:52166;branch=z9hG4bK-d87543-b65c86230550ff4f-
1-- d87543-;rport
Max-Forwards: 70
Contact: <sip:1002 at 192.168.25.116:52166;rinstance=1a12ef13351e0ee1>
To: "1002"<sip:1002 at 192.168.12.160>
From: "1002"<sip:1002 at 192.168.12.160>;tag=5f799517
Call-ID: Yjk5ZTFjZjI1Mzc5N2RiMWFhYWRiMjhhMjc0NTU4ZDI.
CSeq: 1 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, S...
2010 Mar 16
1
Asterisk hangup all incoming calls after 10 seconds
...working perfectly from the ATA (linksys pap2t)
but not from asterisk, because it hangs up after 10 seconds.
Some LOGS:
[Mar 16 15:11:12] DEBUG[13311] acl.c: ##### Testing 192.168.20.113 with
192.168.20.0
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 0: OPTIONS
sip:241 at 192.168.20.113:15956;rinstance=542e2865b2c6abe1 SIP/2.0 (71)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 1: Via: SIP/2.0/UDP
192.168.20.249:5060;branch=z9hG4bK29f64fe1;rport (65)
[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 2: From: "asterisk" <
sip:asterisk at 192.168.20.249 <sip%3Aasterisk at 192.16...
2010 Oct 07
2
401 Unauthorized with Snom but not with Zoiper softphone
...tration fails.
Another remark : when using a Zoiper softphone, the registration goes
very well :
REGISTER sip:sip.domain.tld;transport=UDP SIP/2.0
Via: SIP/2.0/UDP
192.168.114.20:5060;branch=z9hG4bK-d8754z-fab4a5effbf90a05-1---d8754z-
Max-Forwards: 70
Contact:
<sip:test3 at public_ip:51363;rinstance=b6fd38105c91b9bf;transport=UDP>
To: <sip:test3 at sip.domain.tld;transport=UDP>
From: <sip:test3 at sip.domain.tld;transport=UDP>;tag=db1a5018
Call-ID: NzBlZDMyN2U0YTEzZDk4Y2M2N2NmNzMxYTk4OWUxYTY.
CSeq: 1 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE...
2016 Feb 15
2
Asterisk 13.6.0/The simplest TCP configuration does not work
Thanks for the mighty quick response, Joshua!
I am using Zoiper on Linux softclient:
REGISTER sip:<ipAddr>;transport=TCP SIP/2.0
Changed the port back to 5060.
On Mon, Feb 15, 2016 at 1:40 PM, Joshua Colp <jcolp at digium.com> wrote:
> Sonny Rajagopalan wrote:
>
> <snip>
>
>
> *CLI> pjsip set logger on
>> PJSIP Logging enabled
>> [Feb 15
2009 Jul 02
1
need help, service unavailable, registered but call does not get through
...-- Executing [702 at from-internal:2] NoCDR("SIP/701-0864f1b8", "") in new
stack
-- Executing [702 at from-internal:3] Wait("SIP/701-0864f1b8", "1") in new
stack
Retransmitting #1 (NAT) to 123.456.789.000:9855:
OPTIONS sip:701 at 123.456.789.000:37587;rinstance=9428b8620cd7a907 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK782c5851;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown at 192.168.1.4 <sip%3AUnknown at 192.168.1.4>
>;tag=as43db5836
To: <sip:701 at 123.456.789.000:37587;rinstance=9428b8620cd7a907>
Contact...
2008 Oct 02
1
OT - Is sip.instance useful ?
Hi,
I've seen some hardphones or Softswitchs now support this sip.instance
feature :
http://www.softarmor.com/wgdb/docs/draft-jennings-sipping-instance-id-01.txt
I don't really see any convincing use of this draft but I would be curious
to share thoughts on it.
Cheers
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2011 Feb 24
1
Using a Virtual IP Line
...eone can help me to use this line with my asterisk.
These are the traces of my Xllite
REGISTER sip:Xlite release 1100l stamp 49022 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.221:22818;branch=z9hG4bK-d8754z-e322ee549824f666-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:8887776666 at 10.0.0.221:22818;rinstance=570ac597afa82c9a>
To: "8887776666"<sip:8887776666 at Xlite release 1100l stamp 49022>
From: "8887776666"<sip:8887776666 at Xlite release 1100l stamp 49022>;tag=fb1acd4f
Call-ID: NjUyN2Q5ODEzZmZiNjRhY2FmNzRlZjljNDRjMjg5Yjk.
CSeq: 2 REGISTER
Expires: 3600
Allow:...
2006 Feb 21
2
Authorization Plugin for Rails
I''ve posted a lengthy description of an authorization plugin for Rails on my
blog:
http://www.billkatz.com/authorization
It describes a proposed DSL for authorization, a pattern for use that
describes conventions, and a reference implementation that lets you test the
some of the ideas. I hope that some subset of Rails programmers gravitate
toward a common DSL for authorization, which