Displaying 20 results from an estimated 752 matches for "yyy".
2018 Oct 03
2
Any idea what causes "Oooh, got a frame with format of g729 on channel 'PJSIP/121-000001d2' when we're sending 'ulaw', switching to match"
...w = ulaw
from_user = 121
acl = acl1
[10/03 11:29:34.240] DEBUG[21414][C-00000226] chan_pjsip.c: Oooh, got a frame with format of g729 on channel 'PJSIP/121-000001d2' when we're sending 'ulaw', switching to match
INVITE sip:2197 at XXX.XXX.XXX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;branch=z9hG4bKxqyYl2Jg84158000
To: <sip:2197@ XXX.XXX.XXX.XXX <sip:2197 at 192.168.11.176> >
From: <sip:STUFF at YYY.YYY.YYY.YYY <sip:STUFF at YYY.YYY.YYY.YYY%20> >;tag=I0n4X7KK
Contact: <sip:STUFF @YYY.YYY.YYY.YYY:5060<sip:STUFF%20 at YYY.YYY.YYY.YYY:...
2009 May 22
3
No response to our critical packet problem
...ints to a NAT problem, but I don't
understand how it could be because:
1) It does not affect calls to internal office extensions (which still
go through asterisk) OR voicemail
2) The other 20+ phones in the same office on the same network have 0 problems.
Here's a SIP trace of the problem.
yyy.yyy.yyy.yyy is the outside NAT IP
xxx.xxx.xxx.xxx is the IP of my PBX
dddddddddd is the dialed phone number
sssssssssss is the source phone number
The peculiar thing is that asterisk sends an OK in response to an INVITE,
then the phone sends back an ACK, which asterisk seems to ignore
because it r...
2010 May 01
2
Average Login based on date
...02/11/09 xxx FeMale 23 5
9 03/11/09 xxx FeMale 25 9
10 03/11/09 xxx FeMale 35 6
11 03/11/09 xxx Male 18 3
12 03/11/09 xxx Male 31 0
13 04/11/09 xxx Male 32 25
14 04/11/09 xxx Male 31 1
15 04/11/09 xxx FeMale 29 0
16 01/11/09 yyy FeMale 25 2
17 01/11/09 yyy FeMale 35 4
18 01/11/09 yyy Male 18 30
19 01/11/09 yyy Male 31 3
20 02/11/09 yyy Male 32 11
21 02/11/09 yyy Male 31 1
22 02/11/09 yyy FeMale 29 1
23 02/11/09 yyy FeMale 23 5
24 03/11/09 yyy FeMale 25...
2010 Jan 20
2
Call Xfer issue between DataCenter and User Site
...on: Parsing <sip:172 at XXX.XXX.XXX.XXX:10036> for address/port to send to
[Jan 20 16:43:38] set_destination: set destination to XXX.XXX.XXX.XXX, port 10036
[Jan 20 16:43:38] Reliably Transmitting (NAT) to XXX.XXX.XXX.XXX:10016:
NOTIFY sip:172 at XXX.XXX.XXX.XXX:10036 SIP/2.0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;branch=z9hG4bK48cff632;rport
Max-Forwards: 70
From: "Steve (NetTech)" <sip:176 at YYY.YYY.YYY.YYY>;tag=as4f7c4d0c
To: <sip:172 at XXX.XXX.XXX.XXX:10036>;tag=726be2fb618280d0i0
Contact: <sip:176 at YYY.YYY.YYY.YYY>
Call-ID: 718a30a4572984a918b88dc64df642e...
2006 May 17
2
no route to host
...and
Samba 3.0.22.
If I try to connect from NEW to itself by using smbclient I got the
shared resources list correctly. If I try to connect to NEW from OLD,
always using smbclient, I receive the message:
added interface ip=XXX.XXX.XXX.XXX bcast=XXX.XXX.X.255 nmask=255.255.255.0
error connecting to YYY.YYY.YYY.YYY:139 (No route to host)
Error connecting to YYY.YYY.YYY.YYY (No route to host)
Connection to YYY.YYY.YYY.YYY failed
Supposing that XXX.XXX.XXX.XXX is the OLD server address and
YYY.YYY.YYY.YYY is the NEW server address.
I try to find in documentation and in other resources but I found...
2006 Feb 01
1
Unable to Register to Asterisk through Proxy
...==========================================================
sip.conf for user test
[test]
type=friend
host=dynamic
nat=yes
canreinvite=no
username=test
secret=test
==========================================================================
Failure REGISTER through Proxy:
xxx.xxx.xxx.xxx = Asterisk
yyy.yyy.yyy.yyy = Proxy
zzz.zzz.zzz.zzz = User Agent Public IP
192.168.1.2 = User Agent Private IP
<-- SIP read from yyy.yyy.yyy.yyy:5060:
REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bKCefqJfo5hAO/paxgvw/iR7owic4=
Via: SIP/2.0/UDP 192.168.1.2:5066;received...
2006 Apr 12
33
DUNDi with SIP
Anyone out there have a functional DUNDi configuration using SIP for the
inter-Asterisk transport? I've gotten it to work with IAX2, but if I
change it to SIP it does not pass the call over even though it knows
where to send it. Thanks.
The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally
2006 Jan 05
1
Apache reverse proxy authentication problem on RHEL based distribs only
...SLEngine On
SSLCertificateFile /etc/httpd/conf/ssl.crt/server.crt
SSLCertificateKeyFile /etc/httpd/conf/ssl.key/server.key
RequestHeader set Front-End-Https "On"
ProxyRequests Off
ProxyPreserveHost On
LogLevel debug
<Location /exchange>
ProxyPass http://yyy.yyy.yyy.yyy/exchange
ProxyPassReverse http://yyy.yyy.yyy.yyy/exchange
SSLRequireSSL
</Location>
<Location /exchweb>
ProxyPass http://yyy.yyy.yyy.yyy/exchweb
ProxyPassReverse http://yyy.yyy.yyy.yyy/exchweb
SSLRequireSSL
</Location>
<Locatio...
2010 May 07
2
Problems with the IMAP proxy after upgrading from dovecot 1.1.16 to 1.211
...o the MySQL database after a timeout, after
sending the SQL query. Here's a piece of my log file:
May 7 14:46:32 ttt dovecot: auth(default): new auth connection: pid=5136
May 7 14:46:42 ttt dovecot: auth(default): client in:
AUTH^I1^IPLAIN^Iservice=imap^Isecured^Ilip=xxx.xxx.xxx.xxx^Irip=yyy.yyy.yyy.yyy^Ilport=sss^Irport=26480^Iresp=<hidden>
May 7 14:46:42 ttt dovecot: auth-worker(default):
sql(uid,yyy.yyy.yyy.yyy): query: SELECT NULL as password, destuser,
host, 'zzz' as port, 'Y' as proxy, '0' as proxy_timeout, 'Y' as
nopassword, 'Y...
2017 Jan 06
3
Issue with handling of 480 DND
...to our main incoming server (zzz.zzz.zzz.zzz) as "181 call
is being forwarded".
Is this a bug or a feature? :-) How could we handle this correctly?
SIP and Asterisk debug log below. Any help would be appreciated!
Markus
SIP:
#
U 2017/01/06 11:38:29.515836 xxx.xxx.xxx.xxx:45731 ->
yyy.yyy.yyy.yy:5060
SIP/2.0 480 Do Not Disturb.
v: SIP/2.0/UDP yyy.yyy.yyy.yy:5060;branch=z9hG4bK749dbc68;rport=5060.
f: "0160XXXXXXX" <sip:0160XXXXXXX at yyy.yyy.yyy.yy>;tag=as4ef364e1.
t: <sip:4120089 at 192.168.178.70:45731;line=8lln9qsq>;tag=0380h4r478.
i: 7568eb9e7c148e535166...
2017 Jan 24
2
Asterisk 14.2.1 PJSIP - is it possible to retrieve a PJSIP header To field for the SIP OK response to Trying?
...I even tried reading multiple layers of the To header, but it still didn't retrieve the newer dialog To headers.
I am including the SIP messages reported by Asterisk for the call coming in...
*** Phone sends INVITE to Asterisk ***
INVITE sip:333 at xxx.xxx.xxx.xxx SIP/2.0^M
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5063;branch=z9hG4bK-18e552c3^M
From: "1004" <sip:1004 at xxx.xxx.xxx.xxx>;tag=79e7940882a792ao2^M
To: <sip:333 at xxx.xxx.xxx.xxx>^M
Call-ID: 3162d378-ea2b2452 at yyy.yyy.yyy.yyy^M
CSeq: 102 INVITE^M
Max-Forwards: 70^M
Authorization: Digest username="1004"...
2010 Oct 19
1
FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
....
These are the logs:
<--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 --->
INVITE sip:1234567 at 10.0.0.3:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8
From: <sip:98765432 at xxx.xxx.xxx.xx8:5060>;tag=gK0d817deb
To: "Fax" <sip:1234567 at yyy.yyy.yyy.yyy>;tag=as0ddeacb5
Call-ID: 74ca1e4e3e86a1b873428773477e201f at yyy.yyy.yyy.yyy
CSeq: 1785 INVITE
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,MESSAGE,PUBLISH
Accept: application/sdp, application/isup, application/dtmf,
application/dtmf-relay, multipart/mixed...
2011 Feb 21
1
File writing strangeness
...by this piece of software on files
stored on this share work absolutely fine.
Any help will be appreciated.
Kind Regards,
Chris
-------------- next part --------------
No. Time Source Destination Protocol Info
1033 *REF* 192.168.0.XX 192.168.0.YYY SMB Trans2 Request, QUERY_PATH_INFO, Query File Basic Info, Path: \Mp5data\Mp5Backups\Backup 2004-2011
1034 0.000066 192.168.0.YYY 192.168.0.XX SMB Trans2 Response, QUERY_PATH_INFO
1035 0.000540 192.168.0.XX 192.168.0.YYY SMB Tran...
2008 Sep 17
2
Slow "run as ...", firewall issues.
...ted. Wait one minute though
(about, 30 seconds is not long enough, 45 seconds to 1 minute usually
is) and the next time it will be slow once more.
Working back through this it turned out that the firewall rule which had
previously allowed 137-138 access:
ACCEPT tcp -- xxx.xxx.xxx.xxx yyy.yyy.yyy.yyy tcp
dpts:137:139 state NEW
ACCEPT udp -- xxx.xxx.xxx.xxx yyy.yyy.yyy.yyy udp
dpts:137:139 state NEW
ACCEPT tcp -- xxx.xxx.xxx.xxx yyy.yyy.yyy.yyy tcp
dpts:137:139
ACCEPT udp -- xxx.xxx.xxx.xxx yyy.yyy.yyy.yyy udp
dpts:137...
2009 Oct 27
1
RTP timestamps
...like this before and
knows what is the cause and how to fix this?
I've tried many changes in config and upgraded to 1.6.1 but it didnt
change anything, currently running asterisk 1.4.26.1 on 64 bit intel
platform with opensuse.
Here is the tcpdump view from wireshark, xxx is providers ip and yyy is
asterisk:
6218 207.717454 xxx.xxx.xxx.xxx yyy.yyy.yyy.yyy RTP
PT=ITU-T G.711 PCMA, SSRC=0x85048346, Seq=54364, Time=1987711680
6219 207.717481 yyy.yyy.yyy.yyy xxx.xxx.xxx.xxx RTP
PT=ITU-T G.711 PCMA, SSRC=0x35276954, Seq=22826, Time=2202453496
6220 207.737442...
2013 Aug 28
6
redirecting web requests from localhost
...the programmer
available to debug it) are getting the production server IP.
So, how can I redirect for instance 443 traffic to a specific IP to
the local IP address of the local server? I have tried this:
iptables -t nat -A PREROUTING -d XXX.XXX.XXX.XXX -p tcp --dport 80
-j DNAT --to YYY.YYY.YYY.YYY
XXX.XXX.XXX.XXX - IP of production server
YYY.YYY.YYY.YYY - local IP of the test server
Thanks
Miguel
This message and any attachments are intended for the use of the addressee or addressees only. The unauthorised disclosure, use, dissemination or copying (either in...
2004 Apr 26
0
Record-route Issues
Could some please confirm that this behavior is incorrect. I am seeing
issues where it appears that asterisk is not following the Record-route on
it's reply messages. Please let me know if you require any other
information.
Thanks
Example:
xxx.yyy.154.243(PSTN-GW) <--sip--> xxx.yyy.77.23(Asterisk) <--sip-->
xxx.yyy.91.74(SNOM or SER proxy) <--sip----> xxx.yyy.165.201(ATA186)
1) Call place from PSTN to xxx9931211
2) Asterisk via rules in extension.conf sends call to xxx.yyy.91.74
3) xxx.yyy.91.74 sends call to xxx.yyy.165.2...
2010 May 07
2
extract required data from already read data
...02/11/09 xxx FeMale 23 5
9 03/11/09 xxx FeMale 25 9
10 03/11/09 xxx FeMale 35 6
11 03/11/09 xxx Male 18 3
12 03/11/09 xxx Male 31 0
13 04/11/09 xxx Male 32 25
14 04/11/09 xxx Male 31 1
15 04/11/09 xxx FeMale 29 0
16 01/11/09 yyy FeMale 25 2
17 01/11/09 yyy FeMale 35 4
18 01/11/09 yyy Male 18 30
19 01/11/09 yyy Male 31 3
20 02/11/09 yyy Male 32 11
21 02/11/09 yyy Male 31 1
22 02/11/09 yyy FeMale 29 1
23 02/11/09 yyy FeMale 23 5
24 03/11/09 yyy FeMale 25...
2006 Jun 22
3
Showing Current Calls
...4079-e7f2)
SIP/2944079-e7f2 <mailto:2944093@one_start:2> 2944093@one_start:2 Up Dial(SIP/2944093|36|tr)
2 active channels
1 active call
hestia*CLI>
hestia*CLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message
xxx.yyy.128.115 (None) e77bba33-cc 00101/02261 unkn No Rx: REGISTER
xxx.yyy.128.110 (None) 739f4603-e8 00101/00778 unkn No Rx: REGISTER
xxx.yyy.128.86 (None) 56caad3a-eb 00101/01046 unkn No Rx: REGISTER
xxx.yyy.128.115 (None) 91ea0410-60 00101/0...
2013 Mar 15
1
Asterisk uses 3 seconds to send ACK after OK
...eb47c2bae6b101ba1aa at 62.109.37.34:5088
[Mar 15 13:16:05] DEBUG[27947] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE
[Mar 15 13:16:05] DEBUG[27947] chan_sip.c: Header 6 [ 51]: Contact: <sip:b at FQDNz:5060>
[Mar 15 13:16:05] DEBUG[27947] chan_sip.c: Header 7 [117]: Record-Route: <sip:yyy.yyy.yyy.yyy;lr=on;ftag=as6b9fcf86;d=b49.b0ae2a82;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA6NTA4OA-->
[Mar 15 13:16:05] DEBUG[27947] chan_sip.c: Header 8 [ 69]: Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE
[Mar 15 13:16:05] DEBUG[27947] chan_sip.c: Header 9 [ 24]: Suppo...