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Messages - manit123

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1
hi,
I am using YATE 6.0.0-1 r6258 (http://YATE.null.ro)
I have done 'type=tcp' in ysipchan.conf
I found that SIP server sends keep alive every 30 seconds.
How can I change this interval ?
Is https://docs.yate.ro/wiki/Accfile relevant to my requirement ?
If you can provide an example, it will be useful.

Thanks.

2
hi,
I am running yate server (YATE 6.1.0-1 r6319) on windows 7 professional 64 bit OS with 4GB RAM .
I added following lines to conf.d/regfile.cof

[<phone_number>]
password=0000

I have observed that phone number of client registers after every 5 minutes.
How can I change this interval ?
Is it possible to have different registration interval for different phone numbers ?

3
Yate users hangout place / using yate to send calls from/to mobile phone
« on: February 09, 2016, 10:30:25 PM »
hi
i am able to call between VOIP phones in a LAN using yate server on windows .
How can I make a VOIP phone call a public number via mobile with GSM network connected via wifi on same LAN .
to illustrate

external world-(GSM)MOBILE-(wifi)LAN -(wired ethernet) VOIP_PHONE

I want to call via VOIP_PHONE to external world .

4
Is it too late ?
Should I post above questions in a new thread ?

5
sorry . could not reply earlier .

suggestion works.
Here is my ysipchan.conf
Code: [Select]
[general]
port=5060
addr=10.0.0.2

Thanks.

6

Sorry for my very late reply

I followed http://docs.yate.ro/wiki/CDR_File_Module
Created cdrfile.conf in C:\Program Files (x86)\Yate\conf.d containing
Code: [Select]
[general]
file=c:\yate-log.csv
format=${time},"${address}","${caller}","${called}",${billtime},${ringtime},${du
ration},"${direction}","${status}"
So far , so good .

Code: [Select]
status regfile
%%+status:regfile
name=regfile,type=misc;create=false,defined=904,users=39;499=sip/sip:499@10.0.0.
2:50682,177=sip/sip:177@10.0.0.34,178=sip/sip:178@10.0.0.34,153=sip/sip:153@10.0
.0.13,154=sip/sip:154@10.0.0.13,162=sip/sip:162@10.0.0.17,161=sip/sip:161@10.0.0
.17,193=sip/sip:193@10.0.0.25:5065,194=sip/sip:194@10.0.0.25:5065,156=sip/sip:15
6@10.0.0.22:5065,159=sip/sip:159@10.0.0.23:5065,160=sip/sip:160@10.0.0.23:5065,1
75=sip/sip:175@10.0.0.98,176=sip/sip:176@10.0.0.98,155=sip/sip:155@10.0.0.22:506
5,189=sip/sip:189@10.0.0.24:5065,190=sip/sip:190@10.0.0.24:5065,201=sip/sip:201@
10.0.0.14:5065,202=sip/sip:202@10.0.0.14:5065,157=sip/sip:157@10.0.0.26:5065,158
=sip/sip:158@10.0.0.26:5065,179=sip/sip:179@10.0.0.19:5065,180=sip/sip:180@10.0.
0.19:5065,191=sip/sip:191@10.0.0.40:5065,192=sip/sip:192@10.0.0.40:5065,195=sip/
sip:195@10.0.0.28:5065,196=sip/sip:196@10.0.0.28:5065,205=sip/sip:205@10.0.0.45:
5065,206=sip/sip:206@10.0.0.45:5065,185=sip/sip:185@10.0.0.38:5065,186=sip/sip:1
86@10.0.0.38:5065,171=sip/sip:171@10.0.0.29:5065,172=sip/sip:172@10.0.0.29:5065,
203=sip/sip:203@10.0.0.27:5065,204=sip/sip:204@10.0.0.27:5065,151=sip/sip:151@10
.0.0.8:5065,152=sip/sip:152@10.0.0.8:5065,187=sip/sip:187@10.0.0.21:5065,188=sip
/sip:188@10.0.0.21:5065
%%-status

QUESTION | Anyway to format above output in a proper table with columns ip address and phone number/s ?

QUESTION | Please provide links for doing these (pretty useful when debugging live traffic)
Code: [Select]
- if you want a custom log in telnet you could add a filter for "call.execute"
- you could even build a custom global module in javascript or php that prints a message when a call between 2 sip participants is made, but I don't see why you would need that.

7
I am running yate 5.5.0-1 on windows 7 professional 64bit .
the desktop has two network cards and a wifi adapter .
Code: [Select]
Wireless LAN adapter Wireless Network Connection:

   Connection-specific DNS Suffix  . :
   IPv4 Address. . . . . . . . . . . : 192.168.7.2
   Subnet Mask . . . . . . . . . . . : 255.255.255.0
   Default Gateway . . . . . . . . . : 192.168.7.1

Ethernet adapter Local Area Connection 1:

   Connection-specific DNS Suffix  . :
   IPv4 Address. . . . . . . . . . . : 10.3.255.102
   Subnet Mask . . . . . . . . . . . : 255.255.255.0
   Default Gateway . . . . . . . . . : 10.3.255.101

Ethernet adapter Local Area Connection 2:

   Connection-specific DNS Suffix  . :
   IPv4 Address. . . . . . . . . . . : 10.0.0.2
   Subnet Mask . . . . . . . . . . . : 255.255.255.0
   Default Gateway . . . . . . . . . : 10.0.0.1

I ran yate client on same pc using sip:499@10.0.0.2
i get
Unregistered account sip:499@10.0.0.2 reason: Not Implemented
Though yate service is running and regfile.conf has entry 499 with password 0000 .
Where am I going wrong ?

Is it possible to tell sip server to associate with ip 10.0.0.2 ?
In asterisk , we used to edit bind address entry in a specific file .

8
sorry for my late reply .
I get a little about sniffer filter .
(1)Can you suggest a string that appears only when server is managing a call between two sip phones ?
(2)Also , is there a command which can show the phone numbers that have registered at sip server . example - my regfile.conf has entry for phone number 100-1003 . Currently in network 500-515 are active . How can I find that via debug or command ?

9
sorry for my late reply.
I did following on computer running sip server
telnet localhost 5038
debug on
debug level 10
debug sip level 10
color on

Now all sort of information is displayed continuously .
Can I filter this so that it only shows current calls running details ?

Thank you.

10
I meant
Yate client is also running on desktop serving as yate server.
Can yate server log the calls as it handles the communication between clients .
Can I get a window regarding yate server which says whether it is sitting idle or client with ip1 is in conversation with client of ip2 ?
Else , can yate server log the calls in file , so that I can reload the file to see updated entry about call activities .

We used asterisk in which if you run 'sip debug' then it shows running call details etc.
The problem was asterisk failed on windows 7 64 bit .

11
hi
I have a network of few computers . Each has yate client installed . Yate server runs on one of them in same network.
Communication is via SIP protocol .
I am not able to discern the activity between two clients from server.
Example
101 is calling 102 . Server is also running a yate client with phone number 499 .
How can a person sitting on server know - which call is ongoing ?

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