[asterisk-users] Outgoing PJSIP using Kamailio
Administrator
admin at tootai.net
Wed Apr 8 09:16:42 CDT 2020
Hi Joshua
Le 08/04/2020 à 15:28, Joshua C. Colp a écrit :
> On Mon, Apr 6, 2020 at 2:06 PM Administrator <admin at tootai.net
> <mailto:admin at tootai.net>> wrote:
>
> Hello,
>
> We have a provider which is using Kamailio as front end. Our asterisk
> 13/chan_sip server has no problem to register and pass/receive calls
> form this provider.
>
> Now we want to move to asterisk 16/pjsip and face problem.
> Registration
> is OK but when we pass a call our INVITE never receive answer from
> the
> provider. We opened a ticket to their support but in the mean time we
> want to know if someone is using successfully a PJSIP channel against
> Kamailio.
>
> Another one: despite the fact that they use 5061 port, it's not
> TLS but
> UDP. Our asterisk16 has no TLS configured.
>
> We use wizard which looks like:
>
> [Provider-tootai](!)
> ;
> type = wizard
> sends_auth = yes
> sends_registrations = yes
> accepts_auth = no
> accepts_registrations = no
> endpoint/call_group = 1
> endpoint/pickup_group = 1
> endpoint/accountcode = TOOTAi
> endpoint/language = fr
> endpoint/allow = !all,ulaw,alaw,g729
> endpoint/context = incoming-Provider
> endpoint/direct_media = no
> endpoint/dtmf_mode = inband
> registration/retry_interval = 20
> registration/max_retries = 0
> registration/expiration = 3600
> registration/transport = transport-udp
> aor/max_contacts = 2
> aor/qualify_frequency = 2000
>
> [Provider](Provider-tootai)
> ;
> remote_hosts = sips.provider.eu <http://sips.provider.eu>
> endpoint/callerid = "TOOTAi" <00xx xxx xxx xxx>
> aor/contact = sip:sips.provider.eu:5061 <http://sips.provider.eu:5061>
> registration/client_uri = sips:OUR_ID at sips.provider.eu
> <mailto:sips%3AOUR_ID at sips.provider.eu>
> registration/server_uri = sips:sips.provider.eu:5061
> <http://sips.provider.eu:5061>
> outbound_auth/username = OUR_ID
> outbound_auth/password = OUR_PWD
> identity/match = PROVIDER_IP
>
>
> Your server URI For registration and calling differs in that one uses
> "sips" and the other "sip" for URI scheme. Is there a particular
> reason they differ? I'd also expect "sips" not to be used at all if
> it's strictly UDP. You could also compare chan_sip and chan_pjsip
> traffic to see what the difference is.
Yes, someone point this error and I correct it. As said in my previous
message, I had to add outbound_proxy to make it work in UDP. Provideer
support gave me false information by saying that port 5061 was for UDP
but it was as usually for TLS. I correct all the stuff, had to modify
openssl.cnf and downgrade it to TLSv1 as they still use this one and now
connection is OK in UDP as well as TLS.
Thanks for your support
--
Daniel
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