Thanks for the video Nick! It helped push me over the hump I've been stuck with the past few months trying to figure out getting OPTIONS responses to work. I just sent you an email with a link to a diff that lets you set the FQDN from an environment variable rather than hard coding it in the source. I'd prefer to load it from a channel variable, but I haven't been able to figure out how to find the channel reference inside the particular function in res_pjsip_nat.c from the passed tdata. Now to just find a fix for the "Consult then transfer" issue... Lol
Hi Nick.. Might need your assist.. after I edit "pjsip.conf" file and run the certificate, I have this error. "[Aug 20 09:24:09] ERROR[188973]: res_pjsip.c:903 ast_sip_set_tpselector_from_transport_name: Unable to retrieve PJSIP transport 'transport-tls'" Please help me.
Hi Nick, really thanks for the video. The Set-CsUser needs a phonenumber, that means the call is placed from asterisk to PSTN and then to the teams user (costs money)? or is placed to an internal sip trunk and the phone number is only to locate the user (free costs)?
Calls between the Asterisk and Teams are free. You bring your own SIP trunk and are only bound to the costs of the SIP trunk for inbound/outbound calls.
I've patched my asterisk, ensured there's no other versions installed, but I'm still getting 403 forbidden on the registration requests. it's driving me a bit nuts.
Thanks for the video. One question, how did you make call to the teams? In the video, it seems like the video has been skipped and a call is made directly. Can you elaborate on the steps that you did? Thanks
Attended transfer I never spent time on. So no, that’s still broken. Parking calls should work. Not tested. Ring groups I also didn’t test but I don’t see a reason why it shouldn’t work
I have followed all the sets but my direct routing is still not able to connect, it just shows Inactive on the Teams admin page. Any suggestions, my sip packets also look different to your on the web site.
Check out the logs in the asterisk console (turn on the pjsip logger) You should see packets coming in from MS. If you don't, then check your firewall and TLS certificate. If you do, then check how Asterisk responds. It may be the pjsip configuration. I am also available for consulting. You can contact my company at info#bouwhuisit.nl (replace # with @)
Hi Nick, good work on video guide. Excellent guide. i do have 1 question, what happen if i have multiple phone numbers, do i have to specific all the phone numbers in the pjsip?
Dear Nick, Thanks for the very good video. I followed it step by step and i really can not figure out why i have this problem? [Jul 18 08:15:55] ERROR[1804]: res_pjsip_session.c:937 handle_incoming_sdp: teams-in: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing) Please... :) Any suggeestions? Regards Fred
Hm interesting. Haven’t seem that before but it looks like it could be a codec negotiation issue. Check the sip respons with the pjsip logger in Asterisk or capture it with sngrep
Thanks for the video Nick! It helped push me over the hump I've been stuck with the past few months trying to figure out getting OPTIONS responses to work. I just sent you an email with a link to a diff that lets you set the FQDN from an environment variable rather than hard coding it in the source. I'd prefer to load it from a channel variable, but I haven't been able to figure out how to find the channel reference inside the particular function in res_pjsip_nat.c from the passed tdata. Now to just find a fix for the "Consult then transfer" issue... Lol
Hi Nick.. Might need your assist.. after I edit "pjsip.conf" file and run the certificate, I have this error.
"[Aug 20 09:24:09] ERROR[188973]: res_pjsip.c:903 ast_sip_set_tpselector_from_transport_name: Unable to retrieve PJSIP transport 'transport-tls'"
Please help me.
Hi Nick, really thanks for the video. The Set-CsUser needs a phonenumber, that means the call is placed from asterisk to PSTN and then to the teams user (costs money)? or is placed to an internal sip trunk and the phone number is only to locate the user (free costs)?
Calls between the Asterisk and Teams are free. You bring your own SIP trunk and are only bound to the costs of the SIP trunk for inbound/outbound calls.
@@NickBouwhuis and that phone number is "real" or is for internal use only, like an extension? I mean, have i to pay for that number?
@@JuanMol Teams is designed to use real phone numbers. So yes, you most likely need to pay for that number with your provider.
Hi Nick, we have a multi tenant real time asterisk, I guess this will only work for one tenant?
I've patched my asterisk, ensured there's no other versions installed, but I'm still getting 403 forbidden on the registration requests. it's driving me a bit nuts.
Thanks for the video.
One question, how did you make call to the teams? In the video, it seems like the video has been skipped and a call is made directly. Can you elaborate on the steps that you did?
Thanks
I placed a call from a different phone to the number I linked to Teams. That call came in to the Asterisk SBC and got routed to Teams that way.
Let's Encrypt certificates do seem not to work anymore. With a Comodo certificate I had no problems.
Disappointing that Microsoft throws up these arbitrary restrictions
Were you able to get attended transfer working? What about Parking a call instead of attended transfer
Can a phone number be assigned to multiple users? Kinda like a ring group?
Attended transfer I never spent time on. So no, that’s still broken. Parking calls should work. Not tested. Ring groups I also didn’t test but I don’t see a reason why it shouldn’t work
I have followed all the sets but my direct routing is still not able to connect, it just shows Inactive on the Teams admin page. Any suggestions, my sip packets also look different to your on the web site.
Check out the logs in the asterisk console (turn on the pjsip logger)
You should see packets coming in from MS. If you don't, then check your firewall and TLS certificate. If you do, then check how Asterisk responds. It may be the pjsip configuration.
I am also available for consulting. You can contact my company at info#bouwhuisit.nl (replace # with @)
Hi Nick, good work on video guide. Excellent guide. i do have 1 question, what happen if i have multiple phone numbers, do i have to specific all the phone numbers in the pjsip?
If all the numbers are from the same provider, you only have to add them in extensions.conf
@@NickBouwhuis Seems you have removed your post
@@knerduno5942 sorry! The post wasn’t removed but the url has slightly changed. I updated the description
Dear Nick, Thanks for the very good video. I followed it step by step and i really can not figure out why i have this problem?
[Jul 18 08:15:55] ERROR[1804]: res_pjsip_session.c:937 handle_incoming_sdp: teams-in: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing)
Please... :)
Any suggeestions?
Regards Fred
Hm interesting. Haven’t seem that before but it looks like it could be a codec negotiation issue. Check the sip respons with the pjsip logger in Asterisk or capture it with sngrep
@@NickBouwhuis My solution to this problem was to install the libsrtp and libsrtp-devel packages, then configure and build again
Tks!!!!
Does this work for Chan_SIP and or PJ_SIP ?
I didn't test chan_sip since it's deprecated. I used PJSIP in this video.-
good work
Dus Nick Bouwhuis, onze wegen kruisen weer eens. Dit keer niet op een gta samp server….