Sip INVITE with tel: extension rejected

Hi,

I am trying to send an INVITE to asterisk with tel:<mobile_no> and want the action to be an outgoing call on ZAP/g1/<mobile_no>.

But on receiving the INVITE I am getting following error:
[Dec 11 19:12:35] WARNING[2019]: chan_sip.c:8727 get_destination: Huh? Not a SIP header (tel:9873175481)?

and asterisk returns a 404 Not Found message.

my sip.conf:

[general]

context=default  ; Default context for incoming calls
allowoverlap=no  ; Disable overlap dialing support. (Default is yes)
bindport=5060    ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes    ; Enable DNS SRV lookups on outbound calls

[9873175481]
type=friend
context=phones
callingpres=allowed_passed_screen      ; Set caller ID presentation
                                ; See README.callingpres for more information

callerid=5481       ; Full caller ID, to override the phones config
                                ; on incoming calls to Asterisk
fromuser=5481

host=172.16.105.226


[5481]
type=friend
context=phones
callingpres=allowed_passed_screen      ; Set caller ID presentation
                                ; See README.callingpres for more information

callerid=Mansi <1234>       ; Full caller ID, to override the phones config
                                ; on incoming calls to Asterisk
host=172.16.105.226

[authentication]

my extensions.conf:

[quote]
[globals]

[general]
autofallthrough=yes

[default]
exten =>3911299,1,Dial(SIP/9873175481/549899999999)
exten =>3911299,n,Playback(vm-nobodyavail)
exten =>3911299,n,Hangup()

exten =>tel:9873175481,1,Dial(Zap/g1/${EXTEN})
exten =>tel:9873175481,n,Playback(vm-nobodyavail)
exten =>tel:9873175481,n,Hangup()
#even “exten =>9873175481,1,Dial(Zap/g1/${EXTEN})” this is not working
#even “exten =>_XXXX9873175481,1,Dial(Zap/g1/${EXTEN})” this is not working

exten =>3911200,1,Verbose(1|Unrouter call handler)
exten =>3911200,n,Answer()
exten =>3911200,n,Background(demo-instruct)
exten =>3911200,n,WaitExten()

exten =>5481,1,Dial(SIP/5481/5481)
exten =>5481,n,Playback(vm-nobodyavail)
exten =>5481,n,Hangup()

[incoming]
exten =>_XXXXXX,1,Dial,Zap/g1/${$EXTEN}
exten =>s,1,Wait(1)
exten =>s,2,Answer()
exten =>s,3,Playback(demo-congrats)
exten =>s,4,Hangup()

[internal]
exten =>3911200,1,Verbose(1|Echo test application)
exten =>3911200,n,Echo()
exten =>3911200,n,Hangup()

[phones]
include => default
~ [/quote]

asterisk logs:

INVITE tel:9873175481 SIP/2.0
Via: SIP/2.0/UDP 172.16.105.110:4060;branch=z9hG4bK3250255488–21384
Route: sip:10.203.166.233:5060;lr
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
From: “1243049600” sip:1243049600@10.203.166.233;tag=as4cd7505e
To: sip:5481@172.16.105.226
Call-ID: 30@172.16.105.24
CSeq: 1 INVITE
Contact: sip:172.16.105.110:4060
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 2014 2014 IN IP4 10.203.166.233
s=session
c=IN IP4 10.203.166.233
t=0 0
m=audio 15274 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
— (13 headers 14 lines) —
Sending to 172.16.105.110 : 4060 (no NAT)
Using INVITE request as basis request - 30@172.16.105.24
Found no matching peer or user for '172.16.105.110:4060’
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 10.203.166.233:15274
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.203.166.233:15274
[Dec 11 19:12:35] WARNING[2019]: chan_sip.c:8727 get_destination: Huh? Not a SIP header (tel:9873175481)?

<— Reliably Transmitting (no NAT) to 172.16.105.110:4060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 172.16.105.110:4060;branch=z9hG4bK3250255488–21384;received=172.16.105.110
From: “1243049600” sip:1243049600@10.203.166.233;tag=as4cd7505e
To: sip:5481@172.16.105.226;tag=as5f8e29ef
Call-ID: 30@172.16.105.24
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------>
[Dec 11 19:12:35] NOTICE[2019]: chan_sip.c:13858 handle_request_invite: Call from ‘’ to extension ‘tel:9873175481’ rejected because extension not found.

Can anyone please tell how to support tel: URI.

TIA
Mansi

Asterisk is not a SIP proxy (not that many SIP proxies support anything but SIP: URIs).

You need to specify an “extension number” as the user part of the SIP URI, and map that extension number to a dahdi/ channel in the dialplan for the context associated with the incoming SIP connection.