Recent Posts

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1
YateBTS / Multioperators in YateBTS is possible? (severals MNC)
« Last post by psilvao on April 24, 2019, 06:58:59 AM »
Hi, so far when examining the YateBTS documentation, this allows only one operator (MNC), but I would like to know if I can implement more than one operator.

Regards,
Pablo
2
YateBTS / band indicator = 1800 even though frequency set to 900
« Last post by nonce on March 29, 2019, 11:59:51 AM »
I'm having a problem with the phone connecting to the cell station. I have set the BladeRF station to broadcast on the 900MHz range and verified this with another SDR. My cellphone sees the station but can't connect.

Troubleshooting:
Attempted to connect to station then inspected for rejected IMSI numbers. none found.

Code: [Select]
nipc list rejected
IMSI            No attempts register
--------------- ---------------

Here is my configuration from ybts.conf

Code: [Select]
Radio.Band=900
Radio.C0=975
Identity.MCC=510
Identity.MNC=01

Here is the wired part. I performed a pcap for GMS traffic. I only see downlink traffic no uplink traffic. In the downlink traffic I see the band indicator field says 1800 even though I have set this to 900MHz. Please see attached pcap.

no. 425 > GSM CCCH > SI 6 Rest Octets > Band Indicator = 1800

I remember seeing a bug that was posted a few years ago about the wrong frequency being broadcast. I'm not sure if this is related.

Is this expected? It would of course help if I got a working pcap as well. I would like to review a pcap on a working configuration.

Here are the commands to get a pcap.

Code: [Select]
telnet localhost 5038
mbts config Control.GSMTAP.GSM on
quit
connect phone
Code: [Select]
sudo tcpdump -i any udp port 4729 -w GSMtraffic.pcap


3
Other Yate server issues / Using http proxy to connect provider
« Last post by argentum on March 29, 2019, 08:40:38 AM »
Hi.
I have only one way to connect to provider SIP server - via http proxy (IP:port) with authorization (login-password).
Is there any way to do it with Yate?

Thanks in advance.
4
Yate users hangout place / SS7 + MGCP + Cisco freezing
« Last post by kaoe on March 28, 2019, 11:02:19 AM »
Hello,

I'm configuring an SS7 interconnect using Yate + MGCP + Cisco.
In the tests I did worked fine, but I was using a single port of cisco as audio channels.
When we did the final assembly using 3 ports of cisco we had several problems of timeout and even a freezing of cisco.
I do not know if this is correct but I came to the conclusion that the problem was caused by I having created each port as being a gw, ie
[gw cisco-port-1]
host = 192.168.10.253
user = S0 / SU1 / DS1-0 / 1
...
[gw cisco-port-2]
host = 192.168.10.253
user = S0 / SU1 / DS1-1 / 1
...
[gw cisco-port-3]
host = 192.168.10.253
user = S0 / SU2 / DS1-0 / 1
...

This way I imagine that for cisco it is as if there were 3 callagents controlling it, and at the top of the documentation it is written that it does not support this situation. I'm not sure if this is the problem but it seemed to me the only justification to work with 1 port but to freeze the cisco when I set the 3.
This way I changed the configuration to use one, with the range option. But as my cisco modules are 2 ports, my user string does not take care of the situation because * would only replace the range number and not the module, ie for a user = S0 / SU1 / DS1 - * / 1 and a range = 0-1 I can configure the S0 / SU1 / DS1-0 / 1 and S0 / SU1 / DS1-1 / 1 endpoints, but not S0 / SU2 / DS1-0 / 1.

I wonder if the cisco crash problem can even be caused by having several separate gw configurations. I searched the source code and did not find an alternative to create the endpoint that is in module 2 without creating a totally new separate configuration.

Thank you.
5
Yate bugs / Issue in enabling Video Codecs in Yate
« Last post by Prateek on March 25, 2019, 12:54:14 AM »
Hi Team,
I am trying to get video calls (H323 to SIP) via YATE and still faild to get it up.

This is my Setup :
I want to make call from H323 to SIP via Yate. Both ends are registered successfully with their respective IP's. The Issue is based on Video Call, only i can make voice call via Yate not a Video one.

This is the information :

1. All setup is inside the LAN.
2. Running YATE 4.2.
3. Both clients can register and can place voice calls.
5. when I try to establish the video call it does not.
6. Codec is H264 and I have set the 
       
       h263-1998/90000=yes
       h264/90000=yes

      h263-1998=yes
      h263=yes
      h264=yes

in ysipchan.conf
7. Only the Audio Codec is seen by default in the Open Logical Channel in the below Wireshark. How to get Video Codecs too.

Best Regards,
Prateek
6
Yate bugs / Re: enable H264 video support in YATE
« Last post by Prateek on March 25, 2019, 12:48:50 AM »
Hi Team,
I am trying to get video calls (H323 to SIP) via YATE and still faild to get it up.

This is my Setup :
I want to make call from H323 to SIP via Yate. Both ends are registered successfully with their respective IP's. The Issue is based on Video Call, only i can make voice call via Yate not a Video one.

This is the information :

1. All setup is inside the LAN.
2. Running YATE 4.2.
3. Both clients can register and can place voice calls.
5. when I try to establish the video call it does not.
6. Codec is H264 and I have set the 
      h263-1998/90000=yes
       h264/90000=yes

      h263-1998=yes
      h263=yes
      h264=yes

in ysipchan.conf
7. Only the Audio Codec is seen by default in the Open Logical Channel in the below Wireshark. How to get Video Codecs too.

Best Regards,
Prateek
7
Yate bugs / SIP Registration Error
« Last post by John on March 21, 2019, 09:48:34 AM »
Hello there,

i have a working yate / yatebts configuration in nipc mode.
the version is: Yate 6.1.1 devel1 r6325
I can call and send message inside network.
Now i'd like to make call outside the network and possibly have incoming connection.

If I setup a connection with SIP provider and with a VOIP software it's work like sharm,
but when i try to use my yate i got this error:

Code: [Select]
<sip:WARN> SIP line 'outbound' logon timeout

my accfile is:
Code: [Select]
[outbound]
enabled=yes
protocol=sip
number=6xxxxx
username=6xxxxx
authname=6xxxxx
password=kxxxxxxxxxxx
registrar=sip.zadarma.com
domain=sip.zadarma.com
interval=120

my regexroute.conf
Code: [Select]
^.*$=line/\0;line=outbound

debug message: debug sip level 10
Code: [Select]
2019-03-21_16:22:16.511155 <sip:INFO> 'udp:0.0.0.0:5060' sending 'REGISTER sip:sip.zadarma.com' 0x7f30bc01d800 to 185.45.152.174:5060 [0x55f52e2b5f60]
------
REGISTER sip:sip.zadarma.com SIP/2.0
Contact: <sip:6xxxxxxxx@192.168.100.4:5060>
Expires: 120
To: <sip:6xxxxxxxx@sip.zadarma.com>
Call-ID: 1604500942@sip.zadarma.com
Via: SIP/2.0/UDP 192.168.100.4:5060;rport;branch=z9hG4bK1517351562
From: <sip:6xxxxxx@sip.zadarma.com>;tag=1838966968
CSeq: 22 REGISTER
User-Agent: YATE/6.1.1
Max-Forwards: 70
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
Content-Length: 0



or with second provider 'messagenet' i have the same result
accfile:
Code: [Select]
[outbound]
protocol=sip
username=1xxxxxx
server=sip.messagenet.it:5061
password=dxxxxxxxxxxx
ip_transport=UDP
enabled=yes
match_port=yes
match_user=yes

debug code : debug sip level 10
Code: [Select]
<sip:INFO> 'udp:0.0.0.0:5060' sending 'REGISTER sip:sip.messagenet.it:5061' 0x7f30bc006d80 to 212.97.59.76:5061 [0x55f52e2b5f60]
------
REGISTER sip:sip.messagenet.it:5061 SIP/2.0
Contact: <sip:5xxxxxxxx@192.168.100.4:5060>
Expires: 600
To: <sip:5xxxxxxxxx@sip.messagenet.it:5061>
Call-ID: 432445012@sip.messagenet.it:5061
Via: SIP/2.0/UDP 192.168.100.4:5060;rport;branch=z9hG4bK1091355561
From: <sip:5406096367@sip.messagenet.it:5061>;tag=1198750290
CSeq: 2 REGISTER
User-Agent: YATE/6.1.1
Max-Forwards: 70
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
Content-Length: 0
8
Yate bugs / SIP REGISTRATION failure 401 Unauthorized
« Last post by Alexey Kononykhin on March 20, 2019, 03:49:48 AM »
Hello

I have a strange situation with line registration to DIDLogic provider.
Initial logon SIP REGISTRATION sequence works fine (REGISTER->401 Unauthorized->REGISTER (with Auth)->200 OK) a lot of the next SIP REGISTRATION too but starting from some moment provider starts to decline the registration.

After some investigation I found the next. The second SIP REGISTRATION packet generated by Yate contains Authorization header:
Quote
Authorization: Digest username="...", realm="sip.nyc.didlogic.net", nonce="XJILtlySCooP26WY/Xl8+TGKaUdK4Hux4B93tYA=", uri="sip:sip.nyc.didlogic.net", response="6bff40c0c46f75122e3d7479a6233061", algorithm=MD5, qop=auth, nc=0000004e, cnonce="b00b8ee42d705f142ed79eec4566bb0c"
with parameter nc=0000004e (at this packet). The value of nonce-count (value of nc=...) is increased after the each SIP REGISTRATION sequence. And as soon as it's increased to 0x00000100 and higher DIDLogic starts to decline authorization.

Of course the problem of nonce-count higher then  0x00000100 is the question to DIDLogic but I'd like to discuss the question of increasing of nonce-count value.
As far as I understand the description of nonce-count at https://tools.ietf.org/html/rfc2617:
Quote
nonce-count
     This MUST be specified if a qop directive is sent (see above), and
     MUST NOT be specified if the server did not send a qop directive in
     the WWW-Authenticate header field.  The nc-value is the hexadecimal
     count of the number of requests (including the current request)
     that the client has sent with the nonce value in this request.  For
     example, in the first request sent in response to a given nonce
     value, the client sends "nc=00000001".  The purpose of this
     directive is to allow the server to detect request replays by
     maintaining its own copy of this count - if the same nc-value is
     seen twice, then the request is a replay.   See the description
     below of the construction of the request-digest value.
The value of nonce-count has to be unique for SIP REGISTRATION requests with the same nonce value. If nonce value is changed (SIP server returned new nonce value) the value of nonce-count has to be started from 0x1 (00000001) back.

Any ideas about it?

Best regards
Alexey
9
YateBTS / Re: BladeRF File or Device I/O failure when running YateBTS
« Last post by goodboytower on March 19, 2019, 04:05:25 PM »
What is the bladerf version and FPGA ?

My firmware version is 1.6.1 and FPGA is 0.1.2
10
YateBTS / Re: VolteLab No packet
« Last post by Monica Tepelus on March 11, 2019, 06:42:21 AM »
Hi,

I NiPC mode the labkit doesn't communicate with the MiniCore. The NiPC contains a basic implementation of the logic for  MSC/VLR, SMSC and HLR functions locally.
To see any communication between Labkit and Minicore use any of the other modes: roaming, dataroam or LTE ENB.
For the first two modes you can look at the SIP, GTP connection between the Labkit and the Minicore. SIP is for registration, calls and sms and GTP is for data.
In the LTE ENB you should look for S1AP connection.
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