Call drops after 15mins Asterisk-SBC-IMS Scenario

Asterisk Asterisk SIP
1

Hello All,

I’ve a case which I saw several times posted in different forums. My Asterisk 16.1 sends outbound calls to an Athera SBC then to the Operator IMS. All calls drop exactly at 15mins.

Now it’s clear that SIP Timers r causing the issue from all the previous cases i checked on the Internet. Yet i tried almost all the proposed solutions without getting a successful result.
Example:
timers=no
qualify=no
timers=refuse
etc…
(Details r masked)
The 1st Bye Rcvd
W2: U 01/01/1970 21:07:14:890 sbc0: SIP-MSG:<<< SBC Rx from 10.241.94.112:5061 (transport=tls):BYE sip:bshtp-4f-aa9-cb02@10.191.0.46:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 10.241.94.112:5061;branch=z9hG4bKszk1esb1htzsf0kdzhk12hktr;Role=3;Hpt=8fd8_16
Call-ID: 09386A1E-0001176A0007B001-741D8530
From: sip:0123456@ims.xx.yy;user=phone;tag=4f8q6dn6
To: sip:+77777777@ims.xx.yy;user=phone;tag=56771611-0001176A0007AFE8-741D8530
CSeq: 1 BYE
Max-Forwards: 65

10.241.94.112 = IMS Interface Remote IP
10.191.0.46 = The SBC WAN IP

Regards,

Ali

2

[Nov 14 14:26:24] WARNING[1715]: chan_sip.c:30528 sip_send_keepalive: sip_send_k eepalive to 192.168.2.8:5061 returned 0: No such file or directory

This is the Keepalive reply that i get from SBC. Calls still dropping at 15mins.

3

It’s trying to send it on a TCP connection that no longer exists.

Note that chan_sip is unsupported, and is not included in the latest version of Asterisk.

4

I will switch to PJSIP and see how it goes. Thanks

5

Hello!

I had this same problem with the operator stfc and it only resolved after I switched to pjsip, as it turns out that chan_sip does not have the UPDATE method, so the re-invite used to update the session was unknown due to a device in the operator’s environment that disconnected the call exactly at 15 minutes.

6

Thank you dear. I will test with Pjsip and update the case if solved.

7

you should check for session timer , default is 15 min.

8

Hi. Thnx for ur reply. As mentioned above , non of the previously proposed solutions worked, including the timers tweak or on/off. The SIP UPDATE method is not supported in old sip module. I’m switching to PJSIP. Lets see.

9

Thank you all guys . The PJSIP solved the issue totally. The SIP UPDATE is now received and ACK.
CASE SOLVED.

Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
UPDATE sip:127.0.0.1:5080;transport=udp SIP/2.0
CSeq: 25184 UPDATE
CSeq: 25184 UPDATE
Allow: INVITE, ACK, OPTI

10

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