[OpenSIPS-Users] I am facing soem issue while taking inbound call from a TLS gateway .

[Sasmita Panda]

Hi All ,

Having some issue in tls socket connection .

I have added source IP in my gateway table . as below .
mysql> select * from dr_gateways;
+----+-------+------+----------------------+-------+------------+-------+------------+-------+----------------------+---------------------+
| id | gwid  | type | address              | strip | pri_prefix | attrs |
probe_mode | state | socket               | description         |
+----+-------+------+----------------------+-------+------------+-------+------------+-------+----------------------+---------------------+
|  1 | gw1   |    3 | 1.1.1.1:5080 |     0 |            |       |
 0 |     0 |                      | 1st_gw              |
|  4 | gw4   |    3 | 2.2.2.2:5061  |     0 | NULL       | NULL  |
 0 |     0 | tls:3.3.3.3:5061 | tls testing gateway |

When I am calling from gw1 , in the config it comes under the function
*is_from_gw* . But when I am calling from gw4 with TLS , it does not
come under the *is_from_gw *function .

Outgoung call to gw4 working fine . But Opensips is not accepting call from
the same gateway .


Incoming invite from the TLS gateway looks like below .





























*INVITE sip:8484 at 3.3.3.3:5061 <http://sip:8484@3.3.3.3:5061> SIP/2.0Via:
SIP/2.0/TLS 2.2.2.2:5061;branch=z9hG4bK33fcbf5a;rportMax-Forwards: 69From:
"Anonymous" <sip:anonymous at anonymous.invalid>;tag=as0c2aa7d9To:
<sip:8484 at 3.3.3.3:5061 <http://sip:8484@3.3.3.3:5061>>Contact:
<sip:anonymous at 2.2.2.2:5061;transport=tls>Call-ID:
0e2bb879483361cf685af7cd1520ff80 at 2.2.2.2:5061
<http://0e2bb879483361cf685af7cd1520ff80@2.2.2.2:5061>CSeq: 102
INVITEUser-Agent: Asterisk PBX 17.5.1Date: Tue, 05 Jul 2022 12:12:51
GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGESupported: replaces, timerContent-Type:
application/sdpContent-Length: 340v=0o=root 1038073425 1038073425 IN IP4
2.2.2.2s=Asterisk PBX 17.5.1c=IN IP4 2.2.2.2t=0 0m=audio 18936 RTP/SAVP 0
101a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:Q9p7r6xw4RgSXmuL3jAX4FMFbFnWVRwZc6PAqphya=rtpmap:0
PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101
0-16a=ptime:20a=maxptime:150a=sendrecv*

In the request URI transport=TLS parameter is mandatory for TLS connection
here ? Please suggest what can be done here .

*Thanks & Regards*
*Sasmita Panda*
*Senior Network Testing and Software Engineer*
*3CLogic , ph:07827611765*
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