Displaying 20 results from an estimated 335 matches for "5061".
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5060
2004 Jul 09
2
vonage.ca * integration possible?
...risk spoof
the User-Agent field and register itself?
Any thoughts/feedback? Thanks.
>
> No. Time Source Destination Protocol Info
> 222 53.601179 172.21.5.102 216.115.25.187 SIP Request: REGISTER sip:bspgroup1.bsp.vonage.net:5061
>
> Frame 222 (622 bytes on wire, 622 bytes captured)
> Ethernet II, Src: 00:0f:9f:86:42:d4, Dst: 00:06:25:db:aa:25
> Internet Protocol, Src Addr: 172.21.5.102 (172.21.5.102), Dst Addr: 216.115.25.187 (216.115.25.187)
> User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5061 (5...
2015 Nov 20
2
SIP calls dropping at 15 minutes
...(
'02' # set-id (in dispatcher.list)
, '4' # algorithm (4 = round-robin)
);
forward();
return;
}
where set-id 02 is 'sip:Asterisk:5061'
The 'Flow' diagram from Wireshark from a tcpdump on the OpenSIPS host
follows, hopefully the email clients will not mung it too much.
|Time | Client | Asterisk |
| | | OpenSIPS |
|7.158764 | INVI...
2011 Aug 16
1
Asterisk -> Office 365 Unified Messaging... anyone done it?
...r
PBX system with Office 365, we do not support specific setups for PBX systems
for Unified Messaging. Please contact the vendor for more specific instructions
and configurations.
Here is a SIP debug:
[2011-08-11 23:00:26] VERBOSE[17000] chan_sip.c: Reliably Transmitting (no NAT) to 65.55.174.100:5061:
OPTIONS sip:um.outlook.com SIP/2.0
Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK16884162
Max-Forwards: 70
From: "Unknown" <sip:Unknown at 1.2.3.4>;tag=as438c582c
To: <sip:um.outlook.com>
Contact: <sip:Unknown at 1.2.3.4:5061;transport=TLS>
Call-ID: 67f260947dae7c27121ca3...
2005 May 13
0
asterisk dials random number when receiving incoming call
...<sip:Unavailable@213.166.5.129>;tag=as3606b893
To: <sip:448715046363@iptel.tgfslp.dalmany.co.uk>
Call-ID: 1f9465ed1482e9804b089a351a4174a4@213.166.5.129
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:448715046363@82.70.154.145:5061>
Content-Length: 0
---
-- Executing Dial("SIP/4563-5e36",
"SIP/448715046363@192.168.4.5:5061|60|r")
We're at 192.168.4.3 port 35002
Answering/Requesting with root capability 0x4 (ulaw)
Answering with capability 0x2 (gsm)
Answering with capability 0x8 (alaw)
Answerin...
2005 Mar 18
1
Registration issues with Sipura SPA-841
...401 Unauthorized"
Any ideas what is happening and how to fix it?
Phone is configured to register every 10 minutes.
I updated the SPA-841 to the latest firmware. It says Software Version:
3.1.1(a)
192.168.1.101 is the Asterisk server
192.168.1.199 is the phone
Mar 18 16:00:57 sipura199 [1:5061]->192.168.1.101:5060
Mar 18 16:00:57 sipura199
Mar 18 16:00:57 sipura199 [1:5061]<<192.168.1.101:5060
Mar 18 16:00:57 sipura199 SIP/2.0 100 Trying
Mar 18 16:00:57 sipura199 [1:5061]<<192.168.1.101:5060
Mar 18 16:00:57 sipura199 SIP/2.0 401 Unauthorized
Mar 18 16:00:57 sipura199 [1:50...
2003 Nov 11
1
Unable to use voicemail
...1001 1002
Authorization User same as username
PAssword 1001 1002
Domain/Realm 192.168.0.120
SIP Proxy 192.168.0.120
Here is call flow on this test.
(c)2003 Xten Networks Inc. All rights reserved.
Private build: 1008
SIP: 192.168.0.125:5061
RTP: 192.168.0.125:8000
NAT: 210.253.186.126
PXY#0: 192.168.0.120:5060
RECEIVE << 192.168.0.120:5060
NOTIFY sip:1002@192.168.0.125:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.120:5060;branch=z9hG4bK375605f3
From: "asterisk" <sip:asterisk@192.168.0.120>;tag=as633f7afa
To: <sip:...
2005 Jul 25
0
Outgoing SIP to SER causes LOOP BACK message
> Hello fellow asterisk people!
>
> I have Asterisk listening on port 5061 and SER on port 5060.
>
> Asterisk is configured as a gateway for ISDN/Analog/H323 and also SIP.
>
> My problems are with SIP. I can make incoming calls from SIP to asterisk
> and to any of the other networks, but when I try to make an outgoing call
> from Asterisk to SER I see...
2006 Jan 09
0
SIP-SIP transfer via the REFER/NOTIFY method
...Asterisk do the SIP-SIP transfer with the REFER / NOTIFY methods according to RFC-3515 ?
Below is X-Pro Softphone's debug output of a successful transfer according to RFC-3515 :
--------------------------------------------------------
SEND TIME: 26251796
SEND >> sphone.vopr.vonage.net:5061
REGISTER sip:sphone.vopr.vonage.net SIP/2.0
Via: SIP/2.0/UDP aster.autoattendant.eu:5060;rport;branch=z9hG4bKAD14EC1C2A2542DF877A3DF9D1358091
From: Vonage User <sip:pstn_6523567@sphone.vopr.vonage.net>;tag=1997595095
To: Vonage User <sip:pstn_6523567@sphone.vopr.vonage.net>
Contact: &qu...
2005 Jun 19
1
*67 with Sipura 3000
.... If I dial a
number without doing *67 it's working fine...
sip 221 being the extension of my Cisco phone and 192.168.1.6 being
the IP of my asterisk server...
I have my outgoing context configure like this :
[outgoing]
ignorepat => 9
exten => _9.,1,Dial(SIP/${EXTEN:1}@192.168.1.20:5061)
exten => _9.,2,Dial(SIP/${EXTEN:1}@192.168.1.21:5061)
exten => _9.,3,Dial(SIP/${EXTEN:1}@192.168.1.22:5061)
exten => _9.,4,Dial(SIP/${EXTEN:1}@192.168.1.23:5061)
exten => _9.,5,Playback(invalid)
exten => _9.,6,Hangup
When I do 9*67 and the number it take a while and then it will pl...
2004 Nov 29
2
SPA-2000 Dropped calls
...0.5>
WWW-Authenticate: Digest realm="asterisk", nonce="47af5efb"
Content-Length: 0
to 192.168.0.20:5060
Scheduling destruction of call '76662903-a6afea65@192.168.0.20' in 15000
ms
pbx*CLI>
Sip read:
REGISTER sip:192.168.0.5 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.20:5061;branch=z9hG4bK-9723ca95
From: 8445983 <sip:8445983@192.168.0.5>;tag=342babdb37a0856do1
To: 8445983 <sip:8445983@192.168.0.5>
Call-ID: 5039486b-2415121d@192.168.0.20
CSeq: 1 REGISTER
Max-Forwards: 70
Contact: 8445983 <sip:8445983@192.168.0.20:5061>;expires=3600
User-Agent: Sipura/S...
2006 Feb 06
2
Uniden UIP200 and Asterisk v1.2.4: problem not registering
...d useragent "Uniden SIP Phone p2 Ver BS4.70" for peer uip200
but after about 5s I will get something like:
UIP200 is now unreachable.
htpc*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
uip200/uip200 192.168.10.104 D 5061 UNREACHABLE
I have tried the latest firmware (v4.70) and the previous one we've been
running for over 18 months (v4.59) without any luck
Here is the sip.conf I've created on a test server where Asterisk is using
the port 5061 , same for the UIP200 using port 5061. There is no NAT, th...
2006 Feb 20
2
spa3000
...hentication from
the spa3000. Asterisk answers forbidden (SIP/2.0 403 Forbidden) and I
think I placed the username and password correctly...
Sip.conf says this:
[linea2]
username=linea2
type=peer
secret=1111
context=from-pstn
auth=md5
[PSTN_2]
username=line2
type=peer
secret=1111
qualify=yes
port=5061
nat=no
host=192.168.0.20
context=from-pstn
canreinvite=no
auth=md5
The sip debug says this:
<-- SIP read from 192.168.0.20:5061:
INVITE sip:s@192.168.0.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.20:5061;branch=z9hG4bK-fc209f89
From: <sip:987073366@192.168.0.254>;tag=be447a1af149c461o1
T...
2005 Jun 16
2
Multiple Sipura 3000
...y the zap channel
configure so like this :
[outgoing]
ignorepat => 9
exten => _9.,1,Dial(Zap/g2/${EXTEN:1})
exten => _9.,2,Playback(nomoreline)
exten => _9.,3,Hangup
I tried this and it's working :
[outgoing]
ignorepat => 9
exten => _9.,1,Dial(SIP/${EXTEN:1}@10.0.1.111:5061)
exten => _9.,2,Playback(nomoreline)
exten => _9.,3,Hangup
10.0.1.111:5061 is the IP and SIP port of the Sipura 3000 device. So
that would work great if I had only one Sipura but if I have multiple
I would do it that way ? :
[outgoing]
ignorepat => 9
exten => _9.,1,Dial(SIP/${EX...
2006 Feb 23
3
register => 2345:password@sip_proxy doesn't care about port
Hi,
to register my Asterisk with a SIP provider I use the following
syntax, as shown in the default sip.conf:
register => 2345:password@sip_proxy
where
[sip_proxy]
type=peer
context=from-messagenet
host=sip.messagenet.it
port=5061 <------------- please note this one!!!
5061 is provider's port I have to register to.
This also would work for me:
register => 2345:password@sip.messagenet.it:5061
but I need the other syntax 'cause I *have* to specify a different
context for incoming calls rather than the default...
2005 Jul 25
0
SER & Asterisk & SIP =513 "Message Too Big"
Using Asterisk 1.0.9
When I try to make an outgoing call with SIP I get the message " 513 Message
too big" back from SER. Any ideas what I am doing wrong?
Debug below.
SER and Asterisk are running on the same Server
SER is on port 5060
Asterisk is on port 5061
In my extension.conf I have the line
SERADDRESS=192.219.85.57:5060
in Globals
and am using
exten =>_5XXX,2,Dial(sip/${EXTEN:1}@${SERADDRESS})
to dial out.
Here is the sip debug.
-- Executing Ringing("H323/ip$192.219.85.57:2680/5746", "") in new stack
-- Executi...
2003 Nov 12
1
SPA 2000 and 404 not found
I have a Sipura SPA2000 2 line SIP FXS box with line 1 on port 5060 and line 2
on 5061. The SPA2000 is on IP address 192.168.17.6, and the asterisk box is
on 102.168.17.2. Both SPA2000 ports(5060 and 5061) share the same IP address.
Every minute I repeatedly get the following output:
SIP Debugging Enabled
10 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:192.168.17.6 SIP/2.0...
2014 Oct 07
1
Grandstream GXP2160 + SRTP
...th SRTP, I get stuck. There is an
initial INVITE which is anwered with a 401. There should follow a new
INVITE with a nonce, but this does not happen. Any idea why ? Is it the
Grandstream IP-phone ??
<--- SIP read from TLS:my.pub.lic.ip:53416 --->
INVITE sip:0123123123 at ast.ser.ver.ip:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.1.104:5068;branch=z9hG4bK60724585;rport;alias
From: <sip:testacc77005 at ast.ser.ver.ip:5061>;tag=263162018
To: <sip:0123123123 at ast.ser.ver.ip:5061>
Call-ID: 1695864968-5068-8 at BJC.BGI.B.BAE
CSeq: 50 INVITE
Contact: <sips:testacc77005 at 192.168....
2006 Oct 26
0
Can't Register Client - Multiple Subnets
...nnected
(same subnet). The VPN is not NAT'ing anything... and all other
connections work fine across it (i.e. http, ssh, scp, ftp, etc). In
fact, the asterisk logs show the connections, so its getting to the
server.
I am getting the following errors:
moe*CLI>
<-- SIP read from 10.4.5.6:5061:
REGISTER sip:10.4.6.4 SIP/2.0
CSeq: 1 REGISTER
Via: SIP/2.0/UDP
10.4.5.6:5061;branch=z9hG4bK4e847b36-8663-db11-81de-000e7beef61c;rport
User-Agent: Ekiga/2.0.1
From: <sip:211@10.4.6.4>;tag=86747b36-8663-db11-81de-000e7beef61c
Call-ID: b22a7b36-8663-db11-81de-000e7beef61c@dailyplanet
To: <s...
2004 Jul 13
1
SIP authentication bug with insecure= lines?
...42939 128.151.224.35 -> 128.151.224.11 SIP Request: BYE sip:18005551212@128.151.224.11
9.043038 128.151.224.11 -> 128.151.224.35 SIP Status: 200 OK
[root@app1 asterisk]#
Example #1: works (AsteriskPBX -> SER -> AsteriskPRI)
Message Header
Via: SIP/2.0/UDP 128.151.224.35:5061
To: 18005551212<sip:18005551212@128.151.224.35>
From: "19544342000" <sip:19544342000@128.151.224.35:5061>;tag=fa103954da5f640e9acb74f156cc0d02
SIP from address: "19544342000" <sip:19544342000@128.151.224.35:5061>
SIP tag:...
2005 Jun 16
4
Sipura 3000 help
Anyone know what I need to do to get the FXO port on the SPA 3000 to
forward calls to Asterisk? My Asterisk is running on port 5061 and I
set the dial plan on the device to forward to s@asteriskip:5061 but
Asterisk is not picking it up. I can see on tcpdump traces that the
Invite packets do go to through to the asterisk machine on port 5061,
but it's not picking them up. sip debug does not show any packets
either.