[asterisk-users] Digium board considerations
I need to create an updated Asterisk install. I'm planning on using FreePBX. I have markings on an old Digium board TDM2400P rev A2 TDM2400P Rev B DIGCN01ATDM2400P Is there any reason I shouldn't use this board? Are there better board options that have been improved that I should consider? Thanks, Gary Kuznitz WPM$LEX5.PM$ Description: Mail message body -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SetVar Warning
I had lines 3 and 4 and added line 1 and 2 to extensions.conf
exten = 106,1,SetVar(CALLFILENAME=${TIMESTAMP}_${CALLERID(num)})
exten = 106,2,Monitor(wav,${CALLFILENAME},m)
exten = 106,3,hint,SIP/106
exten = 106,4,Macro(stdexten,106,${HINT})
I received this warning:
WARNING[31463]: pbx.c:1832 pbx_extension_helper: No application 'SetVar' for
extension (voicemenu-custom-4, 106, 1)
I'm running Asterisk/1.4.22.
Does anyone have any idea what I need to do to either make SetVar work or
replace it
with something else?
Thanks you,
Gary
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[asterisk-users] Call hung up?
I currently have in extensions.conf:
exten = 106,1,Set(CALLFILENAME=${TIMESTAMP}_${CALLERID(num)})
exten = 106,n,Monitor(wav,${CALLFILENAME},m)
exten = 106,hint,SIP/106
exten = 106,Macro(stdexten,106,${HINT})
When I called x106 this was logged:
-- Executing [106@voicemenu-custom-4:1] Set(DAHDI/7-1,
CALLFILENAME=_xxx) in new stack
-- Executing [106@voicemenu-custom-4:2] Monitor(DAHDI/7-1, wav|_xxx-xxx-
|m) in new stack
== Auto fallthrough, channel 'DAHDI/7-1' status is 'UNKNOWN'
-- Hungup 'DAHDI/7-1'
When I don't have the first two lines this is in the log:
-- Executing [106@voicemenu-custom-4:1] Macro(DAHDI/7-1,
stdexten|106|SIP/106) in new stack
-- Executing [s@macro-stdexten:1] Set(DAHDI/7-1, __DYNAMIC_FEATURES=)
in new stack
-- Executing [s@macro-stdexten:2] GotoIf(DAHDI/7-1, 0?5:3) in new stack
-- Goto (macro-stdexten,s,3)
-- Executing [s@macro-stdexten:3] Dial(DAHDI/7-1, SIP/106|20|) in new
stack
What did I do wrong in adding the first two lines?
Thank you,
Gary
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[asterisk-users] How to block everyone outside of our lan
I'd like to find out how to block everyone outside of the our LAN. The following phone call got through: 1. accountcode: Blank 2. src: Caller*ID number Blank 3. dst: Destination extension 901185294464086 4. dcontext: Destination context DLPN_DialPlan1 5. clid: Caller*ID with text Blank 6. channel: Channel used SIP/xxx-088c48d8 7. dstchannel: Destination channel DAHDI/1-1 8. lastapp: Last application if appropriate Dial 9. lastdata: Last application data (arguments) Dahdi/g1/01185294464086 10. start: Start of call 2010-12-16 04:49:28 11. answer: Answer of call 2010-12-16 04:49:32 12. end: End of call 2010-12-16 04:49:52 13. duration: Total time in system, 24seconds 14. billsec: Total time call is up, 20seconds 15. disposition: What happened to the call: ANSWERED 16. amaflags: What flags to use: DOCUMENTATION In Sip.conf I have: deny=0.0.0.0/0.0.0.0 permit=192.168.1.201/255.255.255.255 All the other local phones here snip One WanIP address Thank you, Gary Kuznitz -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuring Softphone
Thank you for the reply. On 10 Dec 2010 at 8:53, Jeroen (Jeroen Eeuwes jeroeneeu...@gmail.com) commented about Re: [asterisk-users] Configuring Softphone: Hi Gary, I not using anything to create my dialplan. I'm trying to add a softphone to a dialplan that was created a couple years ago by someone that knew what they were doing. Everything else in the dialplan works. As you can see I don't understand how to create a dialplan and I'm seeing from doing a lot of reading on google that everyone is having a hard time figuring out the dialplan that works with softphones. The part I There is no secret in a dialplan for softphones. In fact Asterisk doesn't care if the SIP-device is a softphone, a hard-phone or even another Asterisk box. Perhaps you are over-complicating the issue? If you have a working dialplan for other phones then why are you trying to set it up differently? Have you tried just using the same settings as a working phone? That is a great suggestion. Yes I did try that. I might be having router issues with a SonicWall. I'm working with a port sniffer now to try to figure it out. When I'm done with making sure the router is forwarding everything correctly I'll try that again. Thank you, Gary Best regards, Jeroen Eeuwes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuring Softphone
On 10 Dec 2010 at 9:31, Jeroen (Jeroen Eeuwes jeroeneeu...@gmail.com)
commented about Re: [asterisk-users] Configuring Softphone:
Hi Gary,
That is a great suggestion. Yes I did try that. I might be having router
issues with a
SonicWall. I'm working with a port sniffer now to try to figure it out.
When I'm
done with making sure the router is forwarding everything correctly I'll
try that
again.
If a router is blocking stuff it is bound not to work. Something else
you could try is to configure a softphone on a PC on the same LAN as
the Asterisk box. That way you are by-passing any router issues.
That's a great idea. Even though it's an hour drive for me I might try that
just to
prove it's defiantly not a router issue.
I believe I have proven the router is forwarding just fine now. I have put
back in the
same configuration we use for in house phones.
[gary-incomming]
exten = 120,hint,SIP/120
exten = 120,1,Macro(stdexten,120,${HINT})
When I make a call from the softphone it
1. Shows it registered.
2. Initiated sip call to: the correct phone number
3. Says call answered
4. A few seconds later the phone rings.
5. I answer it.
6. A few seconds later the phone call disconnects from the called phone.
7. The phone call doesn't disconnect from the softphone. I have to disconnect
it
manually.
8. It says Call has disconnected.
9. It says Overall Call Jitter = 0.98 ms
SIP Debug
--- SIP read from SoftPhoneIP:5060 ---
INVITE sip:91phone#cal...@asteriskip SIP/2.0
Via: SIP/2.0/UDP SoftPhoneIP:5060;rport;branch=z9hG4bK103080
To: sip:91phone#cal...@asteriskip
From: gary sip:1...@asteriskip;tag=8826
Call-ID: 1291970614-3080-gar...@softphoneip
CSeq: 880 INVITE
Max-Forwards: 20
User-Agent: NCH Software Express Talk 4.11
Contact: sip:1...@softphoneip:5060
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 380
v=0
o=NCHSoftware-Talk 1291970611 1291970614 IN IP4 SoftPhoneIP
s=Express Talk Call
c=IN IP4 SoftPhoneIP
t=0 0
m=audio 8000 RTP/AVP 0 8 96 3 13 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:13 CN/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=local:192.168.168.7 8000
a=domain:SoftPhoneIP
-
--- (13 headers 16 lines) ---
Sending to SoftPhoneIP : 5060 (NAT)
Using INVITE request as basis request - 1291970614-3080-gar...@softphoneip
UbuntuAsterisk*CLI
--- Reliably Transmitting (no NAT) to SoftPhoneIP:5060 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
SoftPhoneIP:5060;branch=z9hG4bK103080;received=SoftPhoneIP;rport=5060
From: gary sip:1...@asteriskip;tag=8826
To: sip:91phone#cal...@asteriskip;tag=as361b6138
Call-ID: 1291970614-3080-gar...@softphoneip
CSeq: 880 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=0486b332
Content-Length: 0
Scheduling destruction of SIP dialog '1291970614-3080-gar...@softphoneip' in
32000 ms (Method: INVITE)
Found user '120'
--- SIP read from SoftPhoneIP:5060 ---
ACK sip:91phone#cal...@asteriskip SIP/2.0
Via: SIP/2.0/UDP SoftPhoneIP:5060;rport;branch=z9hG4bK103080
To: sip:91phone#cal...@asteriskip;tag=as361b6138
From: gary sip:1...@asteriskip;tag=8826
Call-ID: 1291970614-3080-gar...@softphoneip
CSeq: 880 ACK
Max-Forwards: 20
User-Agent: NCH Software Express Talk 4.11
Content-Length: 0
-
--- (9 headers 0 lines) ---
UbuntuAsterisk*CLI
--- SIP read from SoftPhoneIP:5060 ---
INVITE sip:91phone#cal...@asteriskip SIP/2.0
Via: SIP/2.0/UDP SoftPhoneIP:5060;rport;branch=z9hG4bK113080
To: sip:91phone#cal...@asteriskip
From: gary sip:1...@asteriskip;tag=8826
Call-ID: 1291970614-3080-gar...@softphoneip
CSeq: 881 INVITE
Max-Forwards: 20
User-Agent: NCH Software Express Talk 4.11
Contact: sip:1...@softphoneip:5060
Proxy-Authorization: Digest
username=120,realm=asterisk,nonce=0486b332,uri=sip:91phone#cal...@asteris
kIP,response=fba7a6cc66cf0238dfcc486a5c4f6c73,opaque=,algorithm=MD5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 380
v=0
o=NCHSoftware-Talk 1291970611 1291970614 IN IP4 SoftPhoneIP
s=Express Talk Call
c=IN IP4 SoftPhoneIP
t=0 0
m=audio 8000 RTP/AVP 0 8 96 3 13 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:13 CN/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=local:192.168.168.7 8000
a=domain:SoftPhoneIP
-
--- (14 headers 16 lines) ---
Sending to SoftPhoneIP : 5060 (NAT)
Using INVITE request as basis request - 1291970614-3080-gar...@softphoneip
Found user '120'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 96
Found RTP audio format 3
Found RTP audio format 13
Found RTP audio format 101
Peer audio RTP is at port
Re: [asterisk-users] Asterisk SIP attacks and sshguard
I'm not sure if this is the log entry you are looking for. I had many of these last night. [Dec 9 06:47:51] NOTICE[5630]: chan_sip.c:15593 handle_request_register: Registration from '106 sip:1...@mywanaddress' failed for '121.11.158.174' - Wrong password If you need more information from this Asterisk box let me know. I need to find a way to block these also. Gary On 9 Dec 2010 at 7:57, Joe (Joe Greco asterisk-users@lists.digium.com) commented about [asterisk-users] Asterisk SIP attac: Hello, We had been seeing SIP-guessing attacks on our Asterisk server here. While it wasn't that hard to write a once-a-minute cron job to spank the lusers, that runs once a minute and creates little spikes in the usage and I/O graphs, and is slower to respond than I'd really prefer. I felt that it'd be much cooler to get something more comprehensive put together. We don't use fail2ban because I don't like having to install python. sshguard is a high-performance compiled C application that can run off a log file or a pipe from syslogd to sshguard, meaning that it can respond a lot more quickly than once a minute, and works with very modest overhead on the host system. It also has features such as touchiness, so that it can get tougher on a miscreant as time goes on; my own shell script is naive in that once it passes a threshold, there's just a permanent rule generated. This worries me if I ever have a situation where a legitimate remote client gets messed up and tries the wrong password or something like that; sshguard does a much nicer job in this regard. In any case, my initial attempts to create rules for sshguard didn't work right, quite possibly because I don't often work in LEX/YACC. I submitted a request to the sshguard guys suggesting new rules. http://www.sshguard.net/support/submission/detail/49ce7182028d8b6f3e3d/ and on their mailing list, a little more: http://sourceforge.net/mailarchive/forum.php?thread_name=F4E10075-5D93-43B4-B73A-1FD217BE130D%40sshguard.netforum_name=sshguard-users In particular, they're looking for log examples of some of those messages, but I have no idea how to generate the conditions that would cause these messages. I'm also not sure if there's a way to disable color codes in the Asterisk log files; we log indirectly via BSD's logger # asterisk -vvv 21 | logger -t asterisk so it may be thinking that the console is color-capable. We use this method because this forces them through the syslog mechanism; we need that for centralized logging, and it's handy for things like sshguard too. Specifically looking for examples of (or how to generate) 1).*No registration for peer '.*' (from HOST) 2).*Host HOST failed MD5 authentication for '.*' (.*) 3).*Failed to authenticate user .*@HOST.* If anyone who is more familiar with the attacks or how to generate these messages would give me some assistance, or chime in on the sshguard-users list, that'd be most appreciated. Thanks. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (Fwd) Re: Configuring Softphone
Thank you for the reply.
On 8 Dec 2010 at 13:38, Danny (Danny Nicholas da...@debsinc.com) commented
about RE: [asterisk-users] Configuring Softphone:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Kuznitz
Sent: Wednesday, December 08, 2010 1:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Configuring Softphone
The phone is finally registering. That's great.
I'm trying to understand what all these lines in Extensions.conf are
defining.
I can't make heads or tails them. I have been reading the manual
AsteriskManualTheFutureOfTelephony2ndEdition.
I'm currently getting an error when placing a call on the cmd line saying:
NOTICE[5630]: chan_sip.c:14383 handle_request_invite: Call from 'Gary' to
extension '91AreaCodePhone#' rejected because extension not found.
What I have in Extensions.conf is:
[gary-incomming]
exten = 1001,1,Dial(IAX2/gogh)
exten = 1001,2,HangUp()
exten = 120,1,Dial(SIP/Gary)
exten = Gary,1,Goto(120,1)
exten = i,1,Playback(invalid)
exten = i,2,Goto(s,1)
exten = s,1,Wait(1)
exten = s,2,Answer()
exten = s,3,NoOp(${CALLERID})
exten = s,4,NoOp(${CALLERIDNUM})
exten = s,5,NoOp(${CALLERIDNAME})
exten = s,6,Wait(4)
exten = s,7,Playback(vm-goodbye)
exten = s,8,Wait(2)
exten = s,9,HangUp()
What I have in Sip.conf is:
[authentication]
[general]
context = default
allowoverlap = no
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
limitonpeers = yes
allowguest=no
nat=yes
[Gary]
type = friend
username = Gary
callerid = 120
secret = password
host = dynamic
defaultip = dynamic
context = gary-incomming
dtmfmode = rfc2833
allow=all
Frustrated,
Gary
Without any other comment, you need
exten = _91.,1,Dial(DAHDI/g1/${EXTEN})
in the gary-incomming context.
As defined now, Gary can
#1 answer a call
#2 call IAX/gogh using 1001
I entered the exten line you suggested:
[gary-incomming]
exten = _91.,1,Dial(DAHDI/g1/${EXTEN})
I removed all other lines in [gary-incomming]
When I place a call I get on the cmd line:
-- Executing [916618579...@gary-incomming:1] Dial(SIP/Gary-08941b20,
DAHDI/g1/916618579191) in new stack
-- Called g1/916618579191
-- DAHDI/1-1 answered SIP/Gary-08941b20
[Dec 9 14:00:37] WARNING[5630]: chan_sip.c:1958 retrans_pkt: Maximum retries
exceeded on transmission 1291829914-5076-gar...@192.168.168.7 for seqno 669
(Critical Response) -- See doc/sip-retransmit.txt.
[Dec 9 14:00:37] WARNING[5630]: chan_sip.c:1980 retrans_pkt: Hanging up call
1291829914-5076-gar...@192.168.168.7 - no reply to our critical packet (see
doc/sip-retransmit.txt).
-- Hungup 'DAHDI/1-1'
== Spawn extension (gary-incomming, 916618579191, 1) exited non-zero on
'SIP/Gary-08941b20'
Do you have any ideas? Would you like to see what is in extensions.conf for a
local
extension?
Thank you,
Gary
--- End of forwarded message ---
WPM$44FF.PM$
Description: Mail message body
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Re: [asterisk-users] (Fwd) Re: Configuring Softphone
On 9 Dec 2010 at 13:31, Gary (Gary Kuznitz docf...@theoffice.la) commented
about
[asterisk-users] (Fwd) Re: Configuring Softphone:
Thank you for the reply.
On 8 Dec 2010 at 13:38, Danny (Danny Nicholas da...@debsinc.com) commented
about RE: [asterisk-users] Configuring Softphone:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Kuznitz
Sent: Wednesday, December 08, 2010 1:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Configuring Softphone
The phone is finally registering. That's great.
I'm trying to understand what all these lines in Extensions.conf are
defining.
I can't make heads or tails them. I have been reading the manual
AsteriskManualTheFutureOfTelephony2ndEdition.
I'm currently getting an error when placing a call on the cmd line saying:
NOTICE[5630]: chan_sip.c:14383 handle_request_invite: Call from 'Gary' to
extension '91AreaCodePhone#' rejected because extension not found.
What I have in Extensions.conf is:
[gary-incomming]
exten = 1001,1,Dial(IAX2/gogh)
exten = 1001,2,HangUp()
exten = 120,1,Dial(SIP/Gary)
exten = Gary,1,Goto(120,1)
exten = i,1,Playback(invalid)
exten = i,2,Goto(s,1)
exten = s,1,Wait(1)
exten = s,2,Answer()
exten = s,3,NoOp(${CALLERID})
exten = s,4,NoOp(${CALLERIDNUM})
exten = s,5,NoOp(${CALLERIDNAME})
exten = s,6,Wait(4)
exten = s,7,Playback(vm-goodbye)
exten = s,8,Wait(2)
exten = s,9,HangUp()
What I have in Sip.conf is:
[authentication]
[general]
context = default
allowoverlap = no
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
limitonpeers = yes
allowguest=no
nat=yes
[Gary]
type = friend
username = Gary
callerid = 120
secret = password
host = dynamic
defaultip = dynamic
context = gary-incomming
dtmfmode = rfc2833
allow=all
Frustrated,
Gary
Without any other comment, you need
exten = _91.,1,Dial(DAHDI/g1/${EXTEN})
in the gary-incomming context.
As defined now, Gary can
#1 answer a call
#2 call IAX/gogh using 1001
I entered the exten line you suggested:
[gary-incomming]
exten = _91.,1,Dial(DAHDI/g1/${EXTEN})
I removed all other lines in [gary-incomming]
When I place a call I get on the cmd line:
-- Executing [916618579...@gary-incomming:1] Dial(SIP/Gary-08941b20,
DAHDI/g1/916618579191) in new stack
-- Called g1/916618579191
-- DAHDI/1-1 answered SIP/Gary-08941b20
[Dec 9 14:00:37] WARNING[5630]: chan_sip.c:1958 retrans_pkt: Maximum retries
exceeded on transmission 1291829914-5076-gar...@192.168.168.7 for seqno 669
(Critical Response) -- See doc/sip-retransmit.txt.
[Dec 9 14:00:37] WARNING[5630]: chan_sip.c:1980 retrans_pkt: Hanging up call
1291829914-5076-gar...@192.168.168.7 - no reply to our critical packet (see
doc/sip-retransmit.txt).
-- Hungup 'DAHDI/1-1'
== Spawn extension (gary-incomming, 916618579191, 1) exited non-zero on
'SIP/Gary-08941b20'
Do you have any ideas? Would you like to see what is in extensions.conf for
a local
extension?
Thank you,
Gary
I'm getting closer. Express Talk is now making the call.
I'm getting an error on the cmd line:
-- Executing [91myareacodepho...@dlpn_dialplan1:1] Macro(SIP/120-
b6003810, trunkdial-failover-0.3|Dahdi/g1/1MyAreaCodePhone#||trunk_1|) in
new
stack
-- Executing [...@macro-trunkdial-failover-0.3:1]
GotoIf(SIP/120-b6003810, 0?1-
fmsetcid|1) in new stack
-- Executing [...@macro-trunkdial-failover-0.3:2]
GotoIf(SIP/120-b6003810, 0?1-
setgbobname|1) in new stack
-- Executing [...@macro-trunkdial-failover-0.3:3] Set(SIP/120-b6003810,
CALLERID(num)=) in new stack
-- Executing [...@macro-trunkdial-failover-0.3:4]
GotoIf(SIP/120-b6003810, 0?1-
dial|1) in new stack
-- Executing [...@macro-trunkdial-failover-0.3:5] Set(SIP/120-b6003810,
CALLERID(all)=) in new stack
-- Executing [...@macro-trunkdial-failover-0.3:6] Goto(SIP/120-b6003810,
1-
dial|1) in new stack
-- Goto (macro-trunkdial-failover-0.3,1-dial,1)
-- Executing [1-d...@macro-trunkdial-failover-0.3:1]
Dial(SIP/120-b6003810,
Dahdi/g1/1MyAreaCodePhone#) in new stack
-- Called g1/1MyAreaCodePhone#
-- DAHDI/1-1 answered SIP/120-b6003810
-- Hungup 'DAHDI/1-1'
== Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 1) exited non-zero
on
'SIP/120-b6003810' in macro 'trunkdial-failover-0.3'
== Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 1) exited non-zero
on
'SIP/120-b6003810'
[Dec 9 18:40:39] WARNING[5806]: chan_sip.c:1958 retrans_pkt: Maximum retries
exceeded on transmission 1291829922-5076-gar...@192.168.168.7 for seqno 287
(Critical Response) -- See doc/sip-retransmit.txt.
I don't know if this has anything to do with Express Talk using Local RTP ports
to
listen 8000-8020 and Asterisk using 1 and up. I
Re: [asterisk-users] Configuring Softphone
Thanks for the reply.
On 9 Dec 2010 at 20:56, Steve (Steve Edwards asterisk@sedwards.com)
commented about Re: [asterisk-users] (Fwd) Re: Configuring Softp:
On Thu, 9 Dec 2010, Gary Kuznitz wrote:
I'm getting closer. Express Talk is now making the call.
I'm getting an error on the cmd line:
-- Executing [91myareacodepho...@dlpn_dialplan1:1] Macro(SIP/120-
b6003810, trunkdial-failover-0.3|Dahdi/g1/1MyAreaCodePhone#||trunk_1|)
in new
stack
-- Executing [...@macro-trunkdial-failover-0.3:1]
GotoIf(SIP/120-b6003810, 0?1-
fmsetcid|1) in new stack
-- Executing [...@macro-trunkdial-failover-0.3:2]
GotoIf(SIP/120-b6003810, 0?1-
setgbobname|1) in new stack
-- Executing [...@macro-trunkdial-failover-0.3:3] Set(SIP/120-b6003810,
CALLERID(num)=) in new stack
-- Executing [...@macro-trunkdial-failover-0.3:4]
GotoIf(SIP/120-b6003810, 0?1-
dial|1) in new stack
-- Executing [...@macro-trunkdial-failover-0.3:5] Set(SIP/120-b6003810,
CALLERID(all)=) in new stack
-- Executing [...@macro-trunkdial-failover-0.3:6]
Goto(SIP/120-b6003810, 1-
dial|1) in new stack
-- Goto (macro-trunkdial-failover-0.3,1-dial,1)
-- Executing [1-d...@macro-trunkdial-failover-0.3:1]
Dial(SIP/120-b6003810,
Dahdi/g1/1MyAreaCodePhone#) in new stack
-- Called g1/1MyAreaCodePhone#
-- DAHDI/1-1 answered SIP/120-b6003810
-- Hungup 'DAHDI/1-1'
== Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 1) exited
non-zero on
'SIP/120-b6003810' in macro 'trunkdial-failover-0.3'
== Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 1) exited
non-zero on
'SIP/120-b6003810'
[Dec 9 18:40:39] WARNING[5806]: chan_sip.c:1958 retrans_pkt: Maximum
retries
exceeded on transmission 1291829922-5076-gar...@192.168.168.7 for seqno 287
(Critical Response) -- See doc/sip-retransmit.txt.
I currently have in extensions.conf:
[gary-incomming]
exten = s,1,Wait(1)
exten = s,2,Answer()
exten = s,3,NoOp(${CALLERID})
exten = s,n,NoOp(${CALLERIDNUM})
exten = s,n,NoOp(${CALLERIDNAME})
exten = s,n,Wait(4)
exten = s,n,Playback(tt-weasels)
exten = s,n,Voicemail(11...@vm-test)
exten = s,n,Wait(2)
exten = s,n,Playback(vm-goodbye)
exten = s,n,Wait(2)
exten = s,n,HandUp()
exten = 120,1,Dial(SIP/gary)
exten = gary,1,Goto(120,1)
exten = i,1,Playback(invalid)
exten = i,2,Goto(s,1)
Does it seem odd that your console output does not match your dialplan?
I would suggest discarding PIAF or Elastix or whatever created your
dialplan and start from scratch.
I not using anything to create my dialplan. I'm trying to add a softphone to a
dialplan
that was created a couple years ago by someone that knew what they were doing.
Everything else in the dialplan works. As you can see I don't understand how
to
create a dialplan and I'm seeing from doing a lot of reading on google that
everyone is
having a hard time figuring out the dialplan that works with softphones. The
part I
don't understand is why I'm not getting better answers on this list. I know
there are
lots of experts on this list. I'd be happy to hear from someone that gives me
a
private reply that says something like, I'd be happy to help you resolve your
issue if
you are willing to pay me for my time. I don't know what other secrete there
may be
to get help to resolve this issue.
Once you master the concepts and interaction between sip.conf and
extensions.conf you will be in a better place to evaluate the merits of
using a GUI to create your dialplan or continue growing your own.
I'm not using a GUI. It would probably do a much better job than I am. The
entries
I am trying are all found on Google. I'm amazed with all the experts in the
world that
there aren't lots of examples that work. With my trial and error I'm not
having a lot
of luck. Either finding examples that work or finding rules to create a
dialplan.
Thanks for your input,
Gary
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] Audio ports
I see in sip debug it says Audio is at port 10342 Express Talk suggests Audio at UDP 8000-8020 I've tried changing Express Talk to 1 and forwarded 1-10400. Is there a possibility Express Talk won't work in the 1 range? Is it possible to limit Asterisk to 8000-8020? Thank you, Gary -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audio ports
On 9 Dec 2010 at 22:32, Gary (Gary Kuznitz docf...@theoffice.la) commented about [asterisk-users] Audio ports: I see in sip debug it says Audio is at port 10342 Express Talk suggests Audio at UDP 8000-8020 I've tried changing Express Talk to 1 and forwarded 1-10400. Is there a possibility Express Talk won't work in the 1 range? Is it possible to limit Asterisk to 8000-8020? I see in rpt.conf rtpstart = 8000 rtpend = 8020 Is Audio port 10342 in sip debug not related to rtp ports? It sounds like Express Talk should be configured for 8000-8020 Thanks, Gary Thank you, Gary -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Configuring Softphone
The phone is finally registering. That's great.
I'm trying to understand what all these lines in Extensions.conf are defining.
I can't make heads or tails them. I have been reading the manual
AsteriskManualTheFutureOfTelephony2ndEdition.
I'm currently getting an error when placing a call on the cmd line saying:
NOTICE[5630]: chan_sip.c:14383 handle_request_invite: Call from 'Gary' to
extension '91AreaCodePhone#' rejected because extension not found.
What I have in Extensions.conf is:
[gary-incomming]
exten = 1001,1,Dial(IAX2/gogh)
exten = 1001,2,HangUp()
exten = 120,1,Dial(SIP/Gary)
exten = Gary,1,Goto(120,1)
exten = i,1,Playback(invalid)
exten = i,2,Goto(s,1)
exten = s,1,Wait(1)
exten = s,2,Answer()
exten = s,3,NoOp(${CALLERID})
exten = s,4,NoOp(${CALLERIDNUM})
exten = s,5,NoOp(${CALLERIDNAME})
exten = s,6,Wait(4)
exten = s,7,Playback(vm-goodbye)
exten = s,8,Wait(2)
exten = s,9,HangUp()
What I have in Sip.conf is:
[authentication]
[general]
context = default
allowoverlap = no
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
limitonpeers = yes
allowguest=no
nat=yes
[Gary]
type = friend
username = Gary
callerid = 120
secret = password
host = dynamic
defaultip = dynamic
context = gary-incomming
dtmfmode = rfc2833
allow=all
Frustrated,
Gary
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[asterisk-users] Configuring Softphone
Hi, I'm trying to get a softphone configured. In Sip.conf [general] I found an example that said I need: nat=yes localnet=192.168.xxx.xxx Is localnet supposed to be a LAN IP or a WAN IP? Thank you, Gary -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Configuring Softphone
I have no idea the correct way to configure this software phone.
It's called Express Talk
The Asterisk box is at IP = WanLocation
Software phone is at IP = WanSoftware
They are not on the same LAN.
What I have in Extensions.conf is:
[gary-incomming]
exten = 1001,1,Dial(IAX2/gogh)
exten = 1001,2,HangUp()
exten = 120,1,Dial(SIP/Gary)
exten = Gary,1,Goto(120,1)
exten = i,1,Playback(invalid)
exten = i,2,Goto(s,1)
exten = s,1,Wait(1)
exten = s,2,Answer()
exten = s,3,NoOp(${CALLERID})
exten = s,4,NoOp(${CALLERIDNUM})
exten = s,5,NoOp(${CALLERIDNAME})
exten = s,6,Wait(4)
exten = s,7,Playback(vm-goodbye)
exten = s,8,Wait(2)
exten = s,9,HangUp()
What I have in Sip.conf is:
[authentication]
[general]
context = default
allowoverlap = no
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
limitonpeers = yes
allow = all
allowguest=no
nat=yes
[Gary]
type = friend
username = Gary
callerid = 120
secret = 5351
host = dynamic
defaultip = dynamic
context = gary-incomming
dtmfmode = rfc2833
allow=all
When I reload the dialplan I get an error from Asterisk saying:
[Dec 7 22:01:48] NOTICE[5630]: chan_sip.c:15593 handle_request_register:
Registration from 'sip:g...@wanlocation' failed for 'WanSoftware' - No
matching
peer found
The Softphone SipTrace log says:
17:25:35 UDP Packet Received from WanLocation:5060
SIP/2.0 404 Not found
Via: SIP/2.0/UDP
192.168.168.7:5060;branch=z9hG4bK03856;received=WanSoftware;rport=16699
From: sip:g...@wanlocation;tag=1424
To: sip:g...@wanlocation;tag=as214040c6
Call-ID: 1291771532-3856-gar...@localip
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
Any ideas on how to configure it better are welcome.
Thank you,
Gary
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[asterisk-users] Asterisk ports
Shouldn't Asterisk be listening on UDP port 5060? I'm working with an Asterisk installation running in Ubuntu. Asterisk is running but non of the phone are connecting. I ran netstat -a and I didn't see 5060. Am I supposed to see something listening? Thank you, Gary -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ports
On 2 Dec 2010 at 15:11, Danny (Danny Nicholas da...@debsinc.com) commented about RE: [asterisk-users] Asterisk ports: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Kuznitz Sent: Thursday, December 02, 2010 3:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk ports Shouldn't Asterisk be listening on UDP port 5060? I'm working with an Asterisk installation running in Ubuntu. Asterisk is running but non of the phone are connecting. I ran netstat -a and I didn't see 5060. Am I supposed to see something listening? Thank you, Gary Try netstat -anp|grep ast This will show you all of the ports and addresses asterisk is using (if it is running). Thank you for the reply. Does this look correct? I don't know what port the sip phones are supposed to be communicating on. tcp0 0 0.0.0.0:50380.0.0.0:* LISTEN 5382/asterisk tcp0 0 0.0.0.0:20000.0.0.0:* LISTEN 5382/asterisk tcp0 0 0.0.0.0:80880.0.0.0:* LISTEN 5382/asterisk udp0 0 0.0.0.0:27270.0.0.0:* 5382/asterisk udp0 0 0.0.0.0:45200.0.0.0:* 5382/asterisk udp0 0 0.0.0.0:45690.0.0.0:* 5382/asterisk unix 2 [ ACC ] STREAM LISTENING 180595382/asterisk /var/run/asterisk.ctl unix 2 [ ACC ] STREAM LISTENING 205225768/fast-user-swit /tmp/orbit-docfxit/linc-1688-0-54225d8adde37 unix 2 [ ] DGRAM325885382/asterisk unix 3 [ ] STREAM CONNECTED 207295768/fast-user-swit unix 3 [ ] STREAM CONNECTED 207285768/fast-user-swit unix 3 [ ] STREAM CONNECTED 207275768/fast-user-swit /tmp/orbit-docfxit/linc-1688-0-54225d8adde37 unix 3 [ ] STREAM CONNECTED 205395768/fast-user-swit /tmp/orbit-docfxit/linc-1688-0-54225d8adde37 unix 3 [ ] STREAM CONNECTED 205265768/fast-user-swit unix 3 [ ] STREAM CONNECTED 205255768/fast-user-swit /tmp/orbit-docfxit/linc-1688-0-54225d8adde37 unix 3 [ ] STREAM CONNECTED 205205768/fast-user-swit unix 3 [ ] STREAM CONNECTED 205085768/fast-user-swit Thank you, Gary -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ports
Thank you for the reply. On 2 Dec 2010 at 16:23, Jeff (Jeff LaCoursiere j...@sunfone.com) commented about Re: [asterisk-users] Asterisk ports: On Thu, 2 Dec 2010, Gary Kuznitz wrote: Shouldn't Asterisk be listening on UDP port 5060? I'm working with an Asterisk installation running in Ubuntu. Asterisk is running but non of the phone are connecting. I ran netstat -a and I didn't see 5060. Am I supposed to see something listening? Thank you, Gary You probably see it as: udp0 0 *:sip *:* I don't see this. That could certainly be why the phones are connecting. Why wouldn't that port be listening? Thank you, Gary j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ports
Thanks for the reply. On 2 Dec 2010 at 14:11, Steve (Steve Edwards asterisk-users@lists.digium.com) commented about Re: [asterisk-users] Asterisk ports: On Behalf Of Gary Kuznitz Shouldn't Asterisk be listening on UDP port 5060? Yes. Unless configured otherwise, that's the SIP port. It's set in sip.conf. What does 'sip show settings' show? The first 2 settings (1.6.2.5) should be: UDP SIP Port: 5060 UDP Bindaddress:0.0.0.0 In sip.conf bindport = 5060 'Sip show settings' doesn't work in 1.4.22 I have re-booted this machine. What else could I look for as to why UDP 5060 isn't listening? Thanks, Gary unless you know what you're doing. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ports
Thanks for the reply. On 2 Dec 2010 at 15:57, Steve (Steve Edwards asterisk-users@lists.digium.com) commented about Re: [asterisk-users] Asterisk ports: On Thu, 2 Dec 2010, Steve Edwards wrote: What does 'sip show settings' show? The first 2 settings (1.6.2.5) should be: UDP SIP Port: 5060 UDP Bindaddress:0.0.0.0 On Thu, 2 Dec 2010, Gary Kuznitz wrote: In sip.conf bindport = 5060 'Sip show settings' doesn't work in 1.4.22 I don't have access to a '1.4' instance right now, but 'sip show settings' works in 1.2 and 1.6 so I'm guessing it should work in 1.4 as well. You may have an error that prevents the SIP channel driver from loading. What do you get with 'unload chan_sip.so' followed by 'load chan_sip.so'? You get extra points today. I think you found where the problem is. It found /etc/asterisk/sip.conf Warning parse error: No category context for line 1 of /etc/asterisk/sip.conf Unable to load config sip.conf. This is what is in sip.conf. [authentication] [general] context = default allowoverlap = no bindport = 5060 bindaddr = 0.0.0.0 srvlookup = yes limitonpeers = yes allow = all allowguest=yes What doesn't it like? Thanks, Gary -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ports
Thank you very much for the reply.
On 2 Dec 2010 at 17:06, Steve (Steve Edwards asterisk@sedwards.com)
commented about Re: [asterisk-users] Asterisk ports:
On Thu, 2 Dec 2010, Gary Kuznitz wrote:
You get extra points today. I think you found where the problem is. It
found /etc/asterisk/sip.conf Warning parse error: No category context
for line 1 of /etc/asterisk/sip.conf Unable to load config sip.conf.
This is what is in sip.conf.
[authentication]
[general]
context = default
allowoverlap = no
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
limitonpeers = yes
allow = all
allowguest=yes
Running out of clues here :)
I can load the above fine in my 1.2 instance. Any chance the file was
edited on Windows and needs to be 'unixfied?'
What does 'hexdump -C sip.conf' look like?
Does commenting (';') out line 1 change anything?
This fixed the problem. There was some garbage in line 1.
You are great. Thank you very much.
Gary
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-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] Trying to configure a SIP software phone
I have been told that my logic in extentions.conf is wrong in trying to configure a SIP software phone called Express Talk (remote) . I'd like to make outgoing calls and calls to local extensions. Could someone please look at my configuration files at http://pastebin.com/ajp62wqF and see what I did wrong? Thank you, Gary -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying to configure a SIP software phone
Thank you for the reply. Comments below... On 30 Nov 2010 at 20:21, Warren (Warren Selby wcse...@selbytech.com) commented about Re: [asterisk-users] Trying to configure a SIP so: On Tue, Nov 30, 2010 at 8:04 PM, Gary Kuznitz docf...@theoffice.la wrote: I have been told that my logic in extentions.conf is wrong in trying to configure a SIP software phone called Express Talk (remote) . I'd like to make outgoing calls and calls to local extensions. Could someone please look at my configuration files at http://pastebin.com/ajp62wqF and see what I did wrong? Thank you, Gary That pastebin shows a lot of things that seem wrong. From the error message at the bottom of the pastebin, it looks like you've configured your softphone to register using the username 120, however you've configured your sip peer in sip.conf as Gary for the username. You'll need to match those up for starters. The extensions.conf snippet has a lot of odd logic to it as well, but before we begin to tackle that, let's get the phone registered first. I changed the user in the softphone to Gary. This is the new log. 20:07:48 SIP Public IP is: 75.xxx.xxx.xxx:4582 20:07:48 SIP Number: g...@75.xxx.xxx.xxx:4582 20:07:48 Attempting to register sip:g...@208.xxx.xxx.xxx 20:08:30 Server 208.xxx.xxx.xxx did not respond to register (user sip:g...@208.xxx.xxx.xxx) 20:08:30 Check server details for that line -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Someone has hacked into our system
On 23 Nov 2010 at 16:54, Joseph (Joseph syscon...@gmail.com) commented about Re: [asterisk-users] Someone has hacked into our : On 11/23/10 14:18, Gary Kuznitz wrote: Thank you for the reply... Comments below... On 22 Nov 2010 at 17:23, Tilghman (Tilghman Lesher asterisk- us...@lists.digium.com) commented about Re: [asterisk-users] Someone has hacked into our : On Monday 22 November 2010 17:10:31 Gary Kuznitz wrote: I have the log now. I'd like to know what to look for in trying to figure out how the calls are getting originated. I'd be happy to shere all the information. I just don't want to post information on this public list that might show other people how to get in to our box. allowguest=yes in sip.conf, with a context= in the [general] section that is permitted to make outbound calls? I'm trying to understand exactly what this means. I found a sip.conf in /etc/asterisk I have a [general] section. I don't have allowguest=yes. Is that good or am I supposed to have it? Look for allowguest default is yes I change it to allowguest=no In addition you might want to restrict some countries in your dial-plan, here is my list: This would be great. Can I put this anyplace in extensions.conf? Or does it need to go after [DLPN_DialPlanl] ? Thanks, Gary Kuznitz [blocked-numbers] ;block bahamas, etc exten = _91900.,1,congestion; N11 exten = _91XXX976.,1,congestion ; N11 exten = _91XXX555.,1,congestion ; N11 exten = _91X11.,1,congestion; N11 exten = _91867.,1,congestion; Yukon (sorry mike) ;exten = _1NPA Country exten = _91232.,1,congestion; Sierra Leone exten = _91242.,1,congestion; BAHAMAS exten = _91246.,1,congestion; BARBADOS exten = _91264.,1,congestion; ANGUILLA exten = _91268.,1,congestion; ANTIGUA/BARBUDA exten = _91284.,1,congestion; BRITISH VIRGIN ISLANDS exten = _91345.,1,congestion; CAYMAN ISLANDS exten = _91441.,1,congestion; BERMUDA exten = _91473.,1,congestion; GRENADA exten = _91649.,1,congestion; TURKS CAICOS ISLANDS exten = _91664.,1,congestion; MONTSERRAT exten = _91758.,1,congestion; ST. LUCIA exten = _91767.,1,congestion; DOMINICA exten = _91784.,1,congestion; ST. VINCENT GRENADINES exten = _91809.,1,congestion; DOMINICAN REPUBLIC exten = _91829.,1,congestion; DOMINICAN REPUBLIC exten = _91868.,1,congestion; TRINIDAD AND TOBAGO exten = _91869.,1,congestion; ST. KITTS AND NEVIS exten = _91876.,1,congestion; JAMAICA -- Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Someone has hacked into our system
Thank you for the reply. On 23 Nov 2010 at 18:51, John (John Novack jnov...@stromberg-carlson.org) commented about Re: [asterisk-users] Someone has hacked into our : Gary Kuznitz wrote: Thank you for the reply... Comments below... On 22 Nov 2010 at 17:23, Tilghman (Tilghman Lesherasterisk- us...@lists.digium.com) commented about Re: [asterisk-users] Someone has hacked into our : On Monday 22 November 2010 17:10:31 Gary Kuznitz wrote: I have the log now. I'd like to know what to look for in trying to figure out how the calls are getting originated. I'd be happy to shere all the information. I just don't want to post information on this public list that might show other people how to get in to our box. allowguest=yes in sip.conf, with a context= in the [general] section that is permitted to make outbound calls? I'm trying to understand exactly what this means. I found a sip.conf in /etc/asterisk I have a [general] section. I don't have allowguest=yes. Is that good or am I supposed to have it? I believe what you SHOULD have is; allowguest=no Not sure if that is the default behavior or not If I'm supposed to have it can it go any place in the [general] section? I have in the [general] section a line with: context = default Is this where I would remove default and enter the IP addresses that are allowed to make calls? Your default context in extensions.conf should basiclly lead nowhere. I have mine set up to play an insane laugh then hangup Probably safe to say NEVER use context default for any outbound calling I don't have any context in extensions.conf I do have context = default in sip.conf Should I remove that line? Could you give me an example of what you have in your extensions.conf? Thank you, Gary Kuznitz You should also have, in general: alwaysauthreject=yes This seems pretty effective in stopping some hacking These are simple fixes. I will let others comment on other more detailed firewalling John Novack What would a line with IP address look like? Could you give me an example? If that isn't where the IP address that are allowed supposed to be where would I put them? Thank you, Gary Kuznitz Just a guess, but there have been more than a few such discussions on the list about that configuration, plus a README-SERIOUSLY.bestpractices.txt in the root directory of every Asterisk source tree. You DID read that file, right? -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Someone has hacked into our system
Thank you for the reply... Comments below... On 22 Nov 2010 at 17:23, Tilghman (Tilghman Lesher asterisk- us...@lists.digium.com) commented about Re: [asterisk-users] Someone has hacked into our : On Monday 22 November 2010 17:10:31 Gary Kuznitz wrote: I have the log now. I'd like to know what to look for in trying to figure out how the calls are getting originated. I'd be happy to shere all the information. I just don't want to post information on this public list that might show other people how to get in to our box. allowguest=yes in sip.conf, with a context= in the [general] section that is permitted to make outbound calls? I'm trying to understand exactly what this means. I found a sip.conf in /etc/asterisk I have a [general] section. I don't have allowguest=yes. Is that good or am I supposed to have it? If I'm supposed to have it can it go any place in the [general] section? I have in the [general] section a line with: context = default Is this where I would remove default and enter the IP addresses that are allowed to make calls? What would a line with IP address look like? Could you give me an example? If that isn't where the IP address that are allowed supposed to be where would I put them? Thank you, Gary Kuznitz Just a guess, but there have been more than a few such discussions on the list about that configuration, plus a README-SERIOUSLY.bestpractices.txt in the root directory of every Asterisk source tree. You DID read that file, right? -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Someone has hacked into our system
Someone has hacked into our system and is making calls overseas. How can I: 1. Find out the where the calls are originating from? 2. Block all calls that are not authorized? Our system is in the USA. Only calls from inside our LAN are allowed. Thank you, Gary Kuznitz -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Someone has hacked into our system
Thank you very much for help in finding the log. I have the log now. I'd like to know what to look for in trying to figure out how the calls are getting originated. I'd be happy to shere all the information. I just don't want to post information on this public list that might show other people how to get in to our box. Thanks you, Gary Kuznitz On 22 Nov 2010 at 13:11, Danny (Danny Nicholas da...@debsinc.com) commented about RE: [asterisk-users] Someone has hacked into our : From: Gary Kuznitz [mailto:docf...@theoffice.la] Sent: Monday, November 22, 2010 12:20 PM To: Danny Nicholas Subject: Re: [asterisk-users] Someone has hacked into our system Thank you for the quick response. Comments below... I am not familiar with navigating Asterisk. Would you please help me understand how to see the CDR? Thank you, Gary Kuznitz By default, Asterisk keeps the CDR as a flat-file in /var/log/asterisk/cdr-csv/Master.csv which you can open in Excel for easy viewing. If you have a custom cdr (see /etc/asterisk/cdr.conf or /etc/asterisk/cdr_custom.conf for more information), your CDR might be stored in a MYSQL table or some other place.I would start under the assumption that you have the flat file available.Once you have it open, use this link as a guide http://www.voip-info.org/wiki/view/Asterisk+cdr+csv Fields * accountcode: What account number to use: Asterisk billing account, (string, 20 characters) * src: Caller*ID number (string, 80 characters) * dst: Destination extension (string, 80 characters) * dcontext: Destination context (string, 80 characters) * clid: Caller*ID with text (80 characters) * channel: Channel used (80 characters) * dstchannel: Destination channel if appropriate (80 characters) * lastapp: Last application if appropriate (80 characters) * lastdata: Last application data (arguments) (80 characters) * start: Start of call (date/time) * answer: Answer of call (date/time) * end: End of call (date/time) * duration: Total time in system, in seconds (integer) * billsec: Total time call is up, in seconds (integer) * disposition: What happened to the call: ANSWERED, NO ANSWER, BUSY, FAILED * amaflags: What flags to use: see amaflags::DOCUMENTATION, BILL, IGNORE etc, specified on a per channel basis like accountcode. You will want to see if there are any peculiar src fields on your international calls (dst). WPM$68B7.PM$ Description: Mail message body -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
