pjsip with asterisk 13.18.4 got an 416 error

General Help
1

Hi everyone!

I’m using an asterisk 13.18.4 for my voip server with a sip trunk my SP’s IMS server provided.chan_pjsip driver used.
Everything is ok but incoming call.When i received an incoming call i will got an 416 unsupported uri scheme error.It seems that asterisk doesn’t support a tel uri scheme.

Some useful logging here:

freepbxbjCLI>
<— Received SIP request (1223 bytes) from UDP:10.203.253.241:5060 —>
INVITE sip:[email protected]:5060;line=vrkljaj SIP/2.0
Via: SIP/2.0/UDP 10.203.253.241:5060;branch=z9hG4bKe28415501778241ef3d6-E2PtaN0
To: tel:+8610XXXXXXXX
From: tel:XXXXXXXXXXX;tag=ztesiph9ViWnRNGANbNkoNIquneNAI4-12-20481*gahj.4
Call-ID: 983xdBRFkJbzNRg_wvJd4cl9QewUSsSQaaL3HUU12hdei@zteims
CSeq: 1000 INVITE
Max-Forwards: 65
Contact: sip:10.203.253.241:5060;zte-did=4-12-20481-7842-12-639-65535
P-Called-Party-ID: tel:+8610XXXXXXXX
Supported: 100rel,histinfo,timer
P-Early-Media: supported
P-Asserted-Identity: tel:XXXXXXXXXXX
Accept: application/sdp,
application/isup,
multipart/mixed,
application/dtmf,
application/dtmf-relay
X-ZTE-Cookie: 7zs4rm3;[email protected]
Allow: INVITE,PRACK,ACK,UPDATE,CANCEL,BYE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY
Privacy: none
Min-SE: 90
Session-Expires: 600;refresher=uac
Content-Type: application/sdp
Content-Length: 257
Content-Disposition: session

v=0
o=- 576095279 2131747974 IN IP4 10.203.253.249
s=-
c=IN IP4 10.203.253.249
t=0 0
m=audio 23530 RTP/AVP 8 0 96
c=IN IP4 10.203.253.249
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:96 telephone-event/8000/1
a=fmtp:96 0-15
a=sendrecv

<— Transmitting SIP response (421 bytes) to UDP:10.203.253.241:5060 —>
SIP/2.0 416 Unsupported URI Scheme
Via: SIP/2.0/UDP 10.203.253.241:5060;received=10.203.253.241;branch=z9hG4bKe28415501778241ef3d6-E2PtaN0
Call-ID: 983xdBRFkJbzNRg_wvJd4cl9QewUSsSQaaL3HUU12hdei@zteims
From: tel:XXXXXXXXXXX;tag=ztesiph9ViWnRNGANbNkoNIquneNAI4-12-20481gahj.4
To: tel:+8610XXXXXXXX;tag=z9hG4bKe28415501778241ef3d6-E2PtaN0
CSeq: 1000 INVITE
Server: FPBX-14.0.1.36(13.18.4)
Content-Length: 0

<— Received SIP request (415 bytes) from UDP:10.203.253.241:5060 —>
ACK sip:[email protected]:5060;line=vrkljaj SIP/2.0
Via: SIP/2.0/UDP 10.203.253.241:5060;branch=z9hG4bKe28415501778241ef3d6-E2PtaN0
To: tel:+8610XXXXXXXX;tag=z9hG4bKe28415501778241ef3d6-E2PtaN0
From: tel:XXXXXXXXXXX;tag=ztesiph9ViWnRNGANbNkoNIquneNAI4-12-20481gahj.4
Call-ID: 983xdBRFkJbzNRg_wvJd4cl9QewUSsSQaaL3HUU12hdei@zteims
CSeq: 1000 ACK
Max-Forwards: 70
User-Agent: ZTE-SBC
Content-Length: 0

Any solution to fix that?
Thanks everyone!

1 Like
2

Any solutions？

1 Like
3

自 pjsip with asterisk 13.18.4 got an 416 error 继续讨论：

I got same problem .

22:17:12.837033 IP (tos 0x88, ttl 250, id 14266, offset 0, flags [none], proto UDP (17), length 1268)
172.128.3.4.sip > 172.x.x.x.sip: SIP, length: 1240
INVITE sip:[email protected]:5060;line=hiqvbip SIP/2.0
Via: SIP/2.0/UDP 172.128.3.4:5060;branch=z9hG4bK83268f47936f1dc56668-E5PtaN0
To: tel:+8622xxxxxx
From: tel:186xxxxxxxx;tag=ztesipe_bXAOqq1-7-20481dhag.1
Call-ID: g8pJ8wffq2sa_nnJ7becb@zteims
CSeq: 1000 INVITE
Max-Forwards: 64
Contact: sip:172.128.3.4:5060;zte-did=1-7-20481-3165-12-276
P-Called-Party-ID: sip:[email protected]
Supported: 100rel,timer
P-Early-Media: supported
P-Asserted-Identity: tel:186xxxxxx
Accept: application/sdp,
application/isup,
multipart/mixed,
application/dtmf,
application/dtmf-relay
X-ZTE-Cookie: 7zs4rm4;[email protected]
Allow: INVITE,PRACK,ACK,UPDATE,CANCEL,BYE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY
Privacy: none
Min-SE: 90
Session-Expires: 300;refresher=uac
Content-Type: application/sdp
Content-Length: 304
Content-Disposition: session

v=0
o=- 604275436 568159253 IN IP4 172.128.3.33
s=-
c=IN IP4 172.128.3.33
t=0 0
m=audio 29870 RTP/AVP 8 0 4 18 96
c=IN IP4 172.128.3.33
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:96 telephone-event/8000/1
a=fmtp:96 0-15
a=sendrecv

22:17:12.869222 IP (tos 0x60, ttl 64, id 27133, offset 0, flags [DF], proto UDP (17), length 401)
172.141.x.x.sip > 172.128.3.4.sip: SIP, length: 373
**SIP/2.0 416 Unsupported URI Scheme**
Via: SIP/2.0/UDP 172.128.3.4:5060;received=172.128.3.4;branch=z9hG4bK83268f47936f1dc56668-E5PtaN0
Call-ID: g8pJ8wffq2sa_nnJ7becb@zteims
From: <tel:186xxxxxxx>;tag=ztesipe_bXAOqq*1-7-20481*dhag.1
To: <tel:+8622xxxxxxx>;tag=z9hG4bK83268f47936f1dc56668-E5PtaN0
CSeq: 1000 INVITE
Server: FPBX-14.0.5.25(15.5.0)
Content-Length:  0

22:17:12.873682 IP (tos 0x88, ttl 250, id 14275, offset 0, flags [none], proto UDP (17), length 410)
172.128.3.4.sip > 172.141.x.x.sip: SIP, length: 382
ACK sip:[email protected]:5060;line=hiqvbip SIP/2.0
Via: SIP/2.0/UDP 172.128.3.4:5060;branch=z9hG4bK83268f47936f1dc56668-E5PtaN0
To: tel:+8622xxxxxxxx;tag=z9hG4bK83268f47936f1dc56668-E5PtaN0
From: tel:186xxxxxxxx;tag=ztesipe_bXAOqq1-7-20481dhag.1
Call-ID: g8pJ8wffq2sa_nnJ7becb@zteims
CSeq: 1000 ACK
Max-Forwards: 70
User-Agent: ZTE-SBC
Content-Length: 0

4

Asterisk isn’t accepting the TEL URI scheme for this. It doesn’t support it. SIP/2.0 416 Unsupported URI Scheme covers it. You would need a media gateway to do the conversions or see if the provider can send these calls with standard SIP URIs instead.

5

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