31 Jul
2024
31 Jul
'24
6:13 p.m.
I have integrated Microsoft teams and freepbx using Kamailio version 5.8 as
my SBC.
Calls from freepbx to teams are working fine with audio and I am having
challenges with calls from teams to freepbx. The calls rings and after
answering there is only way audio and the call drops off after thirty
seconds. My pbx is behind a microtik router and I have port forwarded my sip
port and rtp ports to freepbx and have also disabled sip alg. My Kamailio is
hosted in AWS. I have attached both my pcap from Kamailio, freepbx and my
Kamailio.cfg. I am thinking It's a misconfiguration challenge or something
else. Please help I am failing to see my challenge.
Best Regards,
1 Aug
1 Aug
10:42 a.m.
On 31 Jul 2024, at 17:13, palany via sr-users
<sr-users(a)lists.kamailio.org> wrote:
I have integrated Microsoft teams and freepbx using Kamailio version 5.8 as my SBC.
Calls from freepbx to teams are working fine with audio and I am having challenges with
calls from teams to freepbx. The calls rings and after answering there is only way audio
and the call drops off after thirty seconds. My pbx is behind a microtik router and I have
port forwarded my sip port and rtp ports to freepbx and have also disabled sip alg. My
Kamailio is hosted in AWS. I have attached both my pcap from Kamailio, freepbx and my
Kamailio.cfg. I am thinking It’s a misconfiguration challenge or something else. Please
help I am failing to see my challenge.
Without seeing any details, “call hangup after 30 seconds” and “one way audio” are the
most common issues when you have firewalls or NATs in the way and don’t handle it. The
default timer for retransmission is 32 seconds in SIP and after that the INVITE
transaction fails. During that time you will have audio.
Check the packet logs with Wireshark or a similar tool and you will likely see many
retransmits.
Regards,
/O
13 Aug
13 Aug
9:19 a.m.
I think you are dealing with a challenging issue with your SIP setup, I have a some
additional areas to check
You can ensure that your NAT and firewall settings are correctly configured to allow SIP
and RTP traffic through.
You have disabled SIP ALG, make sure that there are no other features or settings on the
router that could be interfering with SIP traffic.
You can verify that the SIP and RTP ports are correctly configured in both FreePBX and
Kamailio; Ensur there are no port mismatches or conflicts.
Review your kamailio.cfg file for any potential misconfigurations related to NAT handling,
SIP forwarding, RTP settings.
You can analyze the logs from Kamailio and FreePBX, and review the PCAP files to identify
any anomalies or errors.
You can Ensure that the codecs being used between Teams & FreePBX are compatible.
Try testing with different call scenarios to determine if the issue is consistent or
specific to certain conditions; This might help narrow down the cause.
14 Aug
14 Aug
8:32 a.m.
Hello @palany I have a suggestion you can consider
You can ensure that the SIP and RTP ports are properly mapped and that no additional
firewall rules are blocking the rtp stream.
You can Verify the SIP and RTP configurations in both Kamailio and FreePBX.
You can Review your kamailio.cfg for any settings that might be affecting the SIP
signaling or media handling.
Analyze the PCAP files you have attached for any anomalies in the SIP signaling or RTP
stream.
You can ensure that the integration with Microsoft Teams is correctly configure.
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participants (3)
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iamolivasmith＠gmail.com -
Olle E. Johansson -
palany