Displaying 20 results from an estimated 85 matches for "ftag".
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2016 Oct 15
2
Registered successfully, but after a minute or so no SIP messages anymore
...519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
From: "02363361779" <sip:02363361779 at sipgate.de>;tag=as02fa8fcc
To: <sip:2636146e0 at sipgate.de>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49 at sipgate.de
CSeq:...
2005 Jan 07
0
Inbound Pickup Issue - Sipmedia
...*CLI>
linux01*CLI> sip debug peer 203
SIP Debugging Enabled for IP: 10.0.0.46:5060
linux01*CLI> sip debug peer Sipmedia
SIP Debugging Enabled for IP: 69.1.236.33:5060
linux01*CLI>
Sip read:
INVITE sip:s@10.0.0.245:5060 SIP/2.0
Record-Route: <sip:+1Myphonenumber@69.1.236.33;r2=on;ftag=VPSF50603522629637;lr>
Record-Route: <sip:+1Myphonenumber@69.1.236.33;transport=tcp;r2=on;ftag=VPSF50603522629637;lr>
Via: SIP/2.0/UDP 69.1.236.33;branch=z9hG4bK3a8d.e3377395.0;i=2
Via: SIP/2.0/TCP 209.247.16.5:5060;branch=z9hG4bK50603522629637-1100766419164
From: <sip:+1Mycellnumber@20...
2005 Jan 09
0
RE: Inbound calls getting disconnected when I answer the phone, using 'SIP'
...*CLI>
linux01*CLI> sip debug peer 203
SIP Debugging Enabled for IP: 10.0.0.46:5060
linux01*CLI> sip debug peer Sipmedia
SIP Debugging Enabled for IP: 69.1.236.33:5060
linux01*CLI>
Sip read:
INVITE sip:s@10.0.0.245:5060 SIP/2.0
Record-Route: <sip:+1Myphonenumber@69.1.236.33;r2=on;ftag=VPSF50603522629637;lr>
Record-Route: <sip:+1Myphonenumber@69.1.236.33;transport=tcp;r2=on;ftag=VPSF50603522629637;lr>
Via: SIP/2.0/UDP 69.1.236.33;branch=z9hG4bK3a8d.e3377395.0;i=2
Via: SIP/2.0/TCP 209.247.16.5:5060;branch=z9hG4bK50603522629637-1100766419164
From: <sip:+1Mycellnumber@20...
2007 Mar 29
3
Asterisk hangs up SIP call after 6 200 retransmits
...type=peer
host=domain.co.nz
context=from-trunk
canreinvite=no
Note that Asterisk registers with proxy:
6499777777:password@conversant.co.nz/6499777777
sip debug peer DLS
<-- SIP read from 147.202.nnn.nnn:5060:
INVITE sip:6499777777@203.89.nnn.nnn SIP/2.0
Record-Route: <sip:147.202.nnn.nnn;ftag=as4917b107;lr=on>
Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd0e3.48331fd3.0
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK3757e55c;rport=5060
From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>
Contact: <sip:649...
2005 Jan 09
1
Inbound calls getting disconnected when I answer the phone, using 'SIP'.
...*CLI>
linux01*CLI> sip debug peer 203
SIP Debugging Enabled for IP: 10.0.0.46:5060
linux01*CLI> sip debug peer Sipmedia
SIP Debugging Enabled for IP: 69.1.236.33:5060
linux01*CLI>
Sip read:
INVITE sip:s@10.0.0.245:5060 SIP/2.0
Record-Route: <sip:+1Myphonenumber@69.1.236.33;r2=on;ftag=VPSF50603522629637;lr>
Record-Route: <sip:+1Myphonenumber@69.1.236.33;transport=tcp;r2=on;ftag=VPSF50603522629637;lr>
Via: SIP/2.0/UDP 69.1.236.33;branch=z9hG4bK3a8d.e3377395.0;i=2
Via: SIP/2.0/TCP 209.247.16.5:5060;branch=z9hG4bK50603522629637-1100766419164
From: <sip:+1Mycellnumber@20...
2016 Oct 15
3
Registered successfully, but after a minute or so no SIP messages anymore
ping times are fine as well:
[root at freepbx asterisk]# ping sipgate.de
PING sipgate.de (217.10.79.9) 56(84) bytes of data.
64 bytes from sipgate.de (217.10.79.9): icmp_seq=1 ttl=57 time=46.7 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=2 ttl=57 time=46.4 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=3 ttl=57 time=46.7 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=4 ttl=57
2005 Sep 14
2
Starting From Scratch
...port=5060
in_uri=sip:916xxx6000@mysipprovider.com
out_uri=sip:916xxx6000@192.168.4.97 via_cnt==1"
9 headers, 0 lines
asterisk1*CLI>
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.200:5060;received=24.23.48.16;branch=z9hG4bK032524f5
Record-Route: <sip:916xxx6000@192.168.4.16;r2=on;ftag=as0dbb6283;lr=on>
Record-Route: <sip:916xxx6000@66.81.0.87;r2=on;ftag=as0dbb6283;lr=on>
From: "102" <sip:102@192.168.1.200>;tag=as0dbb6283
To: <sip:916xxx6000@mysipprovider.com>;tag=as5f7d7868
Call-ID: 7ae2b4642cfbb601604c2d4734352e64@192.168.1.200
CSeq: 102 INVITE
Us...
2009 Mar 16
1
Could Asterisk be rewriting an incoming invite?
I'm not getting inbound audio from bandwidth.com. Their engineer said the
invite that they're sending me looks like this:
INVITE sip:+15129616808 at 67.198.16.18:5060;transport=udp SIP/2.0.
Record-Route: <sip:216.82.224.202;lr;ftag=VPSF506071629460>.
Record-Route: <sip:4.79.212.229;lr;ftag=VPSF506071629460>.
Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK6314.4f7b5d05.0.
Via: SIP/2.0/UDP 4.79.212.229;branch=z9hG4bK6314.15486fb6.0.
Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1207516720501.
From: "...
2004 Dec 11
2
ACK from asterisk not matched to transaction by SER / LCS2005
...): No Route headers found
9(2890) loose_route(): There is no Route HF
9(2890) parse_headers: flags=2048
9(2890) check_via_address(192.168.4.39, 192.168.4.39, 0)
9(2890) Sending:
INVITE sip:chriz@karlshorst.net SIP/2.0
Max-Forwards: 10
Record-Route:
<sip:chriz@192.168.4.39;transport=tcp;r2=on;ftag=as47998c2b;lr>
Record-Route: <sip:chriz@192.168.4.39;r2=on;ftag=as47998c2b;lr>
Via: SIP/2.0/TCP 192.168.4.39;branch=0
Via: SIP/2.0/UDP 192.168.4.39:5082;branch=z9hG4bK61c24316
From: "10" <sip:10@fedora.karlshorst.net>;tag=as47998c2b
To: <sip:chriz@karlshorst.net>
Cont...
2005 Sep 29
1
Cisco AS5300 --> [SIP] --> Asterisk - NO AUDIO
...have been struggling with this for weeks with no luck.
Any help would be appreciated.
Steven Ducat.
*********************************************************************
<-- SIP read from 203.88.192.42:5160:
INVITE sip:84104214@70.84.200.204 SIP/2.0
Record-Route: <sip:203.88.192.42:5160;ftag=1CA65AC-9C8;lr=on>
Via: SIP/2.0/UDP 203.88.192.42:5160;branch=z9hG4bK5aeb.8f7e6027.0
Via: SIP/2.0/UDP 211.147.240.237:5060;rport=57786
From: <sip:0017911@211.147.240.237>;tag=1CA65AC-9C8
To: <sip:84104214@203.88.192.42>
Date: Thu, 29 Sep 2005 20:14:40 GMT
Call-ID: 805AF00B-305C11DA-...
2009 Jul 20
0
Error: Invalid SIP message - rejected , no call id
...oints are equipped for G.729a, so I
don't believe that's the issue here. Their logs are as follows:
U 2009/07/16 20:15:19.097706 66.23.129.253:5060 -> 24.136.116.102:5060
INVITE sip:18888106944 at 24.136.116.102 SIP/2.0..
Record-Route: <sip:18888106944 at 66.23.129.253:5060;nat=yes;ftag=SDju4rb01-gK0665046d;lr=on
>..
Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bKf06e.46f9bd3.0..
Via: SIP/2.0/UDP
67.16.97.188:5060;branch=z9hG4bKfv7a0p204o3hfkg9r100.1..
From: <sip:8502296800 at 67.16.97.188;isup-oli=0;pstn-
params=808481808882>;tag=SDju4rb01-gK0665046d..
To: <si...
2003 Dec 20
2
More beginner questions
...ow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:30342@65.121.72.14>
Content-Length: 0
10 headers, 0 lines
-- SIP/fwd.pulver.com-43fd is ringing
Sip read: CLI>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372
Record-Route: <sip:612@192.246.69.223;ftag=as1f0e4544;lr=on>
From: "calisto" <sip:91184@82.38.193.149>;tag=as1f0e4544
To: <sip:612@fwd.pulver.com>;tag=as2046b5cb
Call-ID: 4336592d02a1a1ba7173c842778e8b8e@82.38.193.149
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:...
2008 Nov 07
1
Outgoing SIP calls dropped after 30 seconds.
...ndwidth.com TRM (bw7.gold.13)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
FreePBX*CLI>
<--- SIP read from 216.82.224.202:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP public IP:5060;branch=z9hG4bK6ea30a1a;rport=5060
Record-Route: <sip:216.82.224.202;lr;ftag=as3ed791f3>
From: "8881231234" <sip:+18881231234 at public IP>;tag=as3ed791f3
To: <sip:+18005551212 at 216.82.224.202>;tag=VPST50603522629853
Call-ID: 28aa1a24047e1bdc3328f645766ddbbb at public IP
CSeq: 102 INVITE
Contact: <sip:+18005551212 at 209.247.16.221:5060;transpo...
2005 Oct 10
1
Incoming SIP getting in, but not ringing.
...ster=6698221:(MYSECRET)@sipgate.co.uk/6698221
Port on my IPCOP box as follows:
UDP/5060
UDP/10000:20000
UDP/8000:8012
UDP-TCP/3478
Thanks for your time.
Paul.
-------------- next part --------------
Sip read:
INVITE sip:6698221@MY_ISP_IP:5060 SIP/2.0
Record-Route: <sip:6698221@217.10.79.219;ftag=as6a04ebdf;lr=on>
Max-Forwards: 9
Record-Route: <sip:6698221@217.10.79.8;ftag=as6a04ebdf;lr=on>
Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKeafc.0df1fb45.0
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKeafc.dd5c7802.0
Via: SIP/2.0/UDP 217.10.79.218:5060;branch=z9hG4bK66be0c1c
From: "07...
2020 Sep 24
2
Negotiates g729 but RTP contains g711
...an_iax) but then transmits g711a media to +27888888888 (chan_sip).
Herewith the scrubbed logging with SIP debug:
[2020-09-19 23:42:19] VERBOSE[2637] chan_sip.c:
<--- SIP read from UDP:41.11.11.12:5060 --->
INVITE sip:0100000000 at 52.22.22.22:5160 SIP/2.0
Record-Route: <sip:41.11.11.12;lr;ftag=as40fe2614>
Via: SIP/2.0/UDP 41.11.11.12:5060;branch=z9hG4bK4df7.5bc77035.0
Via: SIP/2.0/UDP 41.11.11.11:5070;received=41.11.11.11;branch=z9hG4bK0cb77ea3;rport=5070
From: "+27888888888" <sip:+27888888888 at 41.11.11.11:5070>;tag=as40fe2614
To: <sip:0100000000 at 52.22.22.22:51...
2005 Feb 16
1
Strict Routing vs Loose Routing
...h my
comments.
This is a 200 OK (INVITE) received by Asterisk
=========================
U 2005/02/10 16:41:55.065538 143.173.202.82:5060 ->
143.173.202.83:5070
SIP/2.0 200 OK..Via: SIP/2.0/UDP
143.173.202.83:5070;branch=z9hG4bK42c78895..Record-Route:
<sip:001178612341106@143.173.
202.81;ftag=as182aa61c;lr>..Record-Route:
<sip:001178612341106@143.173.202.82:5060>..From: "Call
Center 1" <sip:55512@si
p.trdc.telenova.com.br>;tag=as182aa61c..To:
<sip:001178612341106@sip.com>;tag=281B1720-1D3C..Call-ID:
3
e3b393c407188e9311d01a03e43f144@sip.com..CSeq: 103...
2009 Dec 02
2
Variable Name needed
...n I stripped
out the IPs
<--- Transmitting (no NAT) to:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP;branch=z9hG4bK6631.98b11517.0;received=
Via: SIP/2.0/UDP;branch=z9hG4bK6631.f18e9a97.0
Via: SIP/2.0/UDP :5060;branch=z9hG4bK506071629460-1256675031807
Record-Route: <sip:;lr;ftag=VPSF506071629460>
Record-Route: <sip:;lr;ftag=VPSF506071629460>
From: "BEAUMONT TX"
<sip:+14096798092@;isup-oli=0>;tag=VPSF506071629460
To: <sip:+14098383113@>;tag=as4b59d217
Call-ID: DALMGC0520091202194656056692@
CSeq: 1 INVITE
User-Agent: Asterisk PB...
2020 Sep 22
2
Negotiates g729 but RTP contains g711
Hi,
We have a scenario where inbound calls from an upstream provider (chan_sip) sent downstream (chan_iax2) negotiates only g729 yet RTP media contains g711. Both the upstream and downstream trunks are limited to only offering g729 whilst the initial invite from our upstream provider offers both g711 and g729. Asterisk presumably simply forwards the media from iax2 trunk encapsulation to sip
2007 May 16
0
NO ANSWER, When openser make an oubound SIP call to my asterisk
...byline -d
any port 5060" ####################
interface: any
filter: (ip) and ( port 5060 )
#
U 2007/05/17 13:31:35.908163 my.openser.ip.addr:5060 -> my.asterisk.ip.addr
:5060
INVITE sip:03939749001@my.asterisk.ip.addr:5060;transport=udp SIP/2.0.
Record-Route: <sip:my.openser.ip.addr;lr;ftag=3840196923;nat=yes>.
Via: SIP/2.0/UDP my.openser.ip.addr;branch=z9hG4bKab01.ae06a6e5.0.
Via: SIP/2.0/UDP 192.168.11.9:57536;received=61.217.xxx.xxx
;rport=57536;branch=z9hG4bK834BA777F3C7439EBBC7C4DAECC52FD4.
From: 101 <sip:101@my.openser.domain.name>;tag=3840196923.
To: <sip:0028863939...
2023 Nov 11
1
New syntax for positional-only function parameters?
6 ?????? 2023 ?. 22:54:24 GMT+03:00, mikkmart via R-devel <r-devel at r-project.org> ?????:
>The pattern of functions accepting other functions as inputs and
>passing additional ... arguments to them is prevalent throughout
>the R ecosystem. Currently, however, all such functions must one
>way or another tackle the problem of inadvertently passing arguments
>meant to go to