I am currently running Asterisk 13.1.0-1

I have a chan_sip configuration that works fine with a 3rd party. Third party does not use authentication or registration, it’s ip based authentication…

When I try switching to PJSIP.conf, I seeing 488 responses from the Asterisk side. What has me really baffled is the debugging indicates

[Jul 14 17:28:24] DEBUG[3620] pjsip: sip_endpoint.c Processing incoming message: Request msg INVITE/cseq=222 (rdata0x7f9e98129f38)
[Jul 14 17:28:24] DEBUG[3620] netsock2.c: Splitting ‘xxx.xxx.xxx.xxx:1662’ into…
[Jul 14 17:28:24] DEBUG[3620] netsock2.c: …host ‘xxx.xxx.xxx.xxx’ and port ‘1662’.
[Jul 14 17:28:24] DEBUG[3620] netsock2.c: Splitting ‘0.0.0.0:5060’ into…
[Jul 14 17:28:24] DEBUG[3620] netsock2.c: …host ‘0.0.0.0’ and port ‘5060’.
[Jul 14 17:28:24] DEBUG[3620] netsock2.c: Splitting ‘xxx.xxx.xxx.xxx’ into…
[Jul 14 17:28:24] DEBUG[3620] netsock2.c: …host ‘xxx.xxx.xxx.xxx’ and port ”.
[Jul 14 17:28:24] DEBUG[3614] threadpool.c: Increasing threadpool SIP’s size by 5
[Jul 14 17:28:24] DEBUG[3689] pjsip: sip_endpoint.c Distributing rdata to modules: Request msg INVITE/cseq=222 (rdata0x7f9e98085848)
[Jul 14 17:28:24] DEBUG[3689] res_pjsip_endpoint_identifier_user.c: Retrieved endpoint 3400
[Jul 14 17:28:24] DEBUG[3689] pjsip: tsx0x25e0538 ..Transaction created for Request msg INVITE/cseq=222 (rdata0x7f9e98085848)
[Jul 14 17:28:24] DEBUG[3689] pjsip: tsx0x25e0538 .Incoming Request msg INVITE/cseq=222 (rdata0x7f9e98085848) in state Null
[Jul 14 17:28:24] DEBUG[3689] pjsip: tsx0x25e0538 ..State changed from Null to Trying, event=RX_MSG
[Jul 14 17:28:24] DEBUG[3689] pjsip: dlg0x6e879f8 …Transaction tsx0x25e0538 state changed to Trying
[Jul 14 17:28:24] DEBUG[3689] pjsip: dlg0x6e879f8 .UAS dialog created
[Jul 14 17:28:24] DEBUG[3689] pjsip: dlg0x6e879f8 .Module mod-invite added as dialog usage, data=0x25c4778
[Jul 14 17:28:24] DEBUG[3689] pjsip: dlg0x6e879f8 ..Session count inc to 2 by mod-invite
[Jul 14 17:28:24] DEBUG[3689] pjsip: inv0x6e879f8 .UAS invite session created for dialog dlg0x6e879f8
[Jul 14 17:28:24] DEBUG[3689] pjsip: dlg0x6e879f8 .Module Session Module added as dialog usage, data=(nil)
[Jul 14 17:28:24] DEBUG[3689] pjsip: dlg0x6e879f8 ..Session count inc to 2 by Session Module
[Jul 14 17:28:24] DEBUG[3689] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21
[Jul 14 17:28:24] DEBUG[3689] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116
[Jul 14 17:28:24] DEBUG[3689] res_pjsip_session.c: Negotiating incoming SDP media stream ‘audio’ using audio SDP handler
[Jul 14 17:28:24] DEBUG[3689] res_pjsip_sdp_rtp.c: Endpoint has no codecs for media type ‘audio’, declining stream

The actual packet is as follows and it clearly has audio settings….
17:28:24.631135 IP (tos 0x0, ttl 63, id 15854, offset 0, flags [none], proto UDP (17), length 801)
xxx.xxx.xxx.xxx.1662 > yyy.yyy.yyy.yyy.sip: SIP, length: 773
INVITE sip:446@yyy.yyy.yyy.yyy:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bKxHCO97tu8412a000
To: >
From: “Oper 57 PAUL”>;tag=BqB1jc5P
Contact: >
Call-ID: fja7CsJi-0001-@xxx.xxx.xxx.xxx
CSeq: 222 INVITE
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,REFER,NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 333

v=0
o=- 11264001 11264001 IN IP4 xxx.xxx.xxx.xxx
s=-
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 32770 RTP/AVP 0 2 8 18 110 120 100
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:110 G723/5300
a=rtpmap:120 G723/6300
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sendrecv

sip.conf…

[general]
context = ABC
srvlookup = no callcounter = yes

[3400]
type = friend qualify = no nat = no host = xxx.xxx.xxx.xxx incominglimit = 32
accountcode = 1
port = 5060
context = DEF
dtmfmode = inband insecure = invite

I am trying to make it work with PJSIP.

My pjsip.conf looks like…

[transport1]
type = transport bind = 0.0.0.0
protocol = udp

[3400]
type = aor max_contacts = 1
remove_existing = yes contact=sip:xxx.xxx.xxx.xxx

[3400]
type = endpoint context = DEF
transport = transport1
aors = 3400
accountcode = 1
dtmf_mode = inband device_state_busy_at = 32

Dan Cropp Senior Software Engineer, R&D Software Dept. AMTELCO, 4800 Curtin Drive, McFarland, WI 53558-9424
608 838-4197 ext. 291
1-800-238-5275 ext 291
www.amtelco.com

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