Hello Johan,
Thank you for reply.
The only NAT problem can be on MS Teams Client, because on Opensips side pretty sure all good.
volga629
Can’t it be a NAT problem? The IP address where the bye is coming from doesn’t seem a pstnhub to me.
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Van: Users <[email protected]> namens volga629 via Users <[email protected]>
Verzonden: Saturday, April 18, 2020 11:01:19 PM
Aan: OpenSIPS users mailling list <[email protected]>; Alexey Vasilyev <[email protected]>
Onderwerp: Re: [OpenSIPS-Users] ms teams ACKHello Alexey,
Thank you on reply,
I undone all changes regard headers changes and MS Teams send BYE directly to asterisk.
No Route header present.
But INVITE ACK 183 180 all travel with proper routing information.
2020/04/18 17:54:28.599711 190.109.70.77:5060 -> 190.109.68.250:5060
BYE sip:[email protected]:5060 SIP/2.0
FROM: <sip:[email protected]>;tag=4d7fb0763c224e39a13a03c669c4b387
TO: <sip:[email protected]>;tag=as41e97ff5
CSEQ: 3 BYE
CALL-ID: [email protected]
MAX-FORWARDS: 69
Via: SIP/2.0/UDP 190.109.70.77:5060;branch=z9hG4bK050e.e400e373.0;i=66c9c603
VIA: SIP/2.0/TLS 52.114.14.70:5061;rport=8208;received=52.114.14.70;branch=z9hG4bK9594cd7
REASON: Q.850;cause=18;text="fcb37a2a-4bc4-49b6-a5e3-aabddc8f7a22;Call Controller timed out while waiting for acknowledgement."
CONTACT: <sip:52.114.14.70:8208;nat=yes;x-i=fcb37a2a-4bc4-49b6-a5e3-aabddc8f7a22;x-c=b5841f98785c5819bb99e57cd0fa7d86/s/1/9b6ba2f8eefa4a67bed29609fd1884ec>
CONTENT-LENGTH: 0
USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.4.13.7 i.ASSE.3
ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY
volga629
On 4/18/20 5:13 PM, Alexey Vasilyev wrote:
Hi volga629, There were nothing special for ACK. You don't need to change To/From/Contact. All the necessary steps were in the article https://blog.opensips.org/2019/09/16/opensips-as-ms-teams-sbc/ and for most people it still works. So I'm not sure, that MS changed anything, because all the hardware SBCs should change behaviour, so they need new firmware. SBC vendors should inform customers to update etc. So this is not so simple process. And it definitely make no sense for anybody. And in the test lab for the article I've used absolutely the same architecture with asterisk, the only difference was RTPEngine to transcode SRTP-RTP. And within test lab I've tested not only calls, but transfers worked fine too.----- --- Alexey Vasilyev -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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