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:[email protected]:5060<sip:[email protected]:5060> SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bKxHCO97tu8412a000
To: <sip:[email protected]<sip:[email protected]>>
From: "Oper 57
PAUL"<sip:[email protected]<sip:[email protected]>>;tag=BqB1jc5P
Contact: <sip:[email protected]:5060<sip:[email protected]:5060>>
Call-ID:
[email protected]<mailto:[email protected]>
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<http://www.amtelco.com/>
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