PJSIP conversion

OK - I but back the ; on the second definition so it is just the first.
(thanks didnt reallize that).

Same

Jerry

OK - Made progress…

REGISTER sip:192.168.2.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.14:5060;branch=z9hG4bK7c0e70e4
Max-Forwards: 70
From: sip:production_to_testing@192.168.2.6;tag=as2d66c47d
To: sip:production_to_testing@192.168.2.6
Call-ID: 3347fb425fa730767e766bd4575927ee@192.168.1.14
CSeq: 103 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 18.18.0
Authorization: Digest username=“production_to_testing”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.2.6”, nonce=“1717096619/488f69637fa81dfb01df4e1e262bd654”, response=“906a00dc11d5ef0043ab9aa395359b85”, opaque=“730c33d21b944d8e”, qop=auth, cnonce=“254016f0”, nc=00000001
Expires: 120
Contact: sip:s@192.168.1.14:5060
Content-Length: 0

15:16:59.417652 IP (tos 0x0, ttl 64, id 45817, offset 0, flags [DF], proto UDP (17), length 397)
VMImage.lsi.com.sip > 192.168.1.14.sip: [bad udp cksum 0x85ef → 0xa36b!] SIP, length: 369
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.14:5060;rport=5060;received=192.168.1.14;branch=z9hG4bK7c0e70e4
Call-ID: 3347fb425fa730767e766bd4575927ee@192.168.1.14
From: sip:production_to_testing@192.168.2.6;tag=as2d66c47d
To: sip:production_to_testing@192.168.2.6;tag=z9hG4bK7c0e70e4
CSeq: 103 REGISTER
Server: Asterisk PBX 20.8.1
Content-Length: 0

[production_to_testing]
type=wizard
accepts_registrations=yes
sends_registrations=no
accepts_auth=yes
sends_auth=no
inbound_auth/username=production_to_testing
inbound_auth/password=production_to_testing
endpoint/disallow=all
endpoint/allow=ulaw
endpoint/allow=alaw
endpoint/allow=gsm
endpoint/context=smvoice-mediacontroller

So I am getting a 403 error.
Thoughts on that ?

Jerry

Word order differs

Thanks - but I;m not following - what differs?

Jerry

production_to_testing versus testing_to_production

Ok - instead of typing from the other two systems - I copied/grabbed and posted here.

From Asterisk 20 system PJSIP
[production_to_testing]
type=wizard
accepts_registrations=yes
sends_registrations=no
accepts_auth=yes
sends_auth=no
inbound_auth/username=production_to_testing
inbound_auth/password=production_to_testing
endpoint/disallow=all
endpoint/allow=ulaw
endpoint/allow=alaw
endpoint/allow=gsm
endpoint/context=smvoice

From ASterisk 18. system with chan_sip
general
register => production_to_testing:production_to_testing@192.168.2.6

[production_to_testing]
type=friend
defaultname=production_to_testing
username=production_to_testing
secret=production_to_testing
host=192.168.2.6
context=production_to_testing
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
allow=gsm

THen I am seeing a 403 error with tcpdump on the asterisk 20 system.
What do I not have right ?
Thanks
Jerry

Both of these do the same thing and neither of them is of any use if the host isn’t dynamic. One basic principle you need to understand is that every sample sip.conf is broken. (The other breakage here is using type=friend, when type=peer is sufficient.)

On the other hand, you are defaulting the From user to the caller ID, and, unless the caller ID is production_to_testing, this will not work.

I did figure this out finally. I had an error in my pjsip.conf and nothing was working because of it.
Thanks for the help.

Jerry

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