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 endpoint/callerid = "TOOTAi" <00xx xxx xxx xxx> aor/contact = sip:sips.provider.eu:5061 registration/client_uri = sips:OUR_ID at sips.provider.eu registration/server_uri = sips:sips.provider.eu:5061 outbound_auth/username = OUR_ID outbound_auth/password = OUR_PWD identity/match = PROVIDER_IP Thanks for any hint -- Daniel
On Mon, Apr 6, 2020 at 2:06 PM Administrator <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 > endpoint/callerid = "TOOTAi" <00xx xxx xxx xxx> > aor/contact = sip:sips.provider.eu:5061 > registration/client_uri = sips:OUR_ID at sips.provider.eu > registration/server_uri = sips: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. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at sangoma.com and asterisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <lists.digium.com/pipermail/asterisk-users/attachments/20200408/e0f86120/attachment.html>
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 <sips.provider.eu> > endpoint/callerid = "TOOTAi" <00xx xxx xxx xxx> > aor/contact = sip:sips.provider.eu:5061 <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 > <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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <lists.digium.com/pipermail/asterisk-users/attachments/20200408/032ed48d/attachment.html>
Reasonably Related Threads
- Outgoing PJSIP using Kamailio
- Can someone please help with this sip2sip pjsip_wizard "no matching endpoint" issue?
- Asterisk 13.6.0/The simplest TCP configuration does not work
- No matching endpoint found for incoming call from SIP trunk
- No matching endpoint found for incoming call from SIP trunk