[asterisk-users] pjsip.conf question

Tue Jul 14 13:56:36 CDT 2015

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 at yyy.yyy.yyy.yyy:5060<sip:446 at 192.168.8.122:5060> SIP/2.0
        Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bKxHCO97tu8412a000
        To: <sip:446 at yyy.yyy.yyy.yyy<sip:446 at 192.168.8.122>>
        From: "Oper 57 PAUL"<sip:3400 at xxx.xxx.xxx.xxx<sip:3400 at 192.168.33.18>>;tag=BqB1jc5P
        Contact: <sip:3400 at xxx.xxx.xxx.xxx:5060<sip:3400 at 192.168.33.18:5060>>
        Call-ID: fja7CsJi-0001- at xxx.xxx.xxx.xxx<mailto:fja7CsJi-0001- at 192.168.33.18>
        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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