Sir,
I am using sip fxo for incoming call all is wroking fine but now i want to display callerid on extension but it will display userno of sip channel instead of caller no.
There is any way to display caller no. instead of sip user no.
Rajeev.
you can check if it is in the variable ${CALLERID(DNID)}
put this in you incomming dialscript
exten=s,1,NoOp(CallerID = ${CALLERID(DNID)})
NoOp does nothing it ony shows you the information in the CLI when an incomming call comes in you should watch the cli for CallerID = then you should see the number. if yo do you need to change the value CALLERID(num)
but it also could be some configuration in your FXO gateway
Sir,
It will return the audiocode fxo port number. i want to receive caller number please help.
Rajeev.
Please post a copy of the SIP INVITE you are receiving. It may be that your upstream is not sending the information.
Sir,
I not expert in asterisk will you please tell me how i have to give you SIP INVITE information.
Rajeev.
CLI sip set debug on (you may need to do core set verbose, as well);
wireshark; or
tcpdump
tcpdump can be better than wireshark, as you don’t want to end up using a screen shot, here.
Sir,
Here is the screen shot.
<— SIP read from 192.168.0.173:5060 —>
INVITE sip:453@myproxy SIP/2.0
Via: SIP/2.0/UDP 192.168.0.173;branch=z9hG4bKac891662743
Max-Forwards: 70
From: “Anonymous” sip:900@192.168.0.246;tag=1c891659115
To: sip:453@myproxy
Call-ID: 89165870311200003410@192.168.0.173
CSeq: 1 INVITE
Contact: sip:900@192.168.0.173
Supported: em,100rel,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-118 FXO/v.5.40A.015.004
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 267
v=0
o=AudiocodesGW 891653273 891653157 IN IP4 192.168.0.173
s=Phone-Call
c=IN IP4 192.168.0.173
t=0 0
m=audio 6000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6001 IN IP4 192.168.0.173
<------------->
— (14 headers 12 lines) —
Sending to 192.168.0.173 : 5060 (no NAT)
Using INVITE request as basis request - 89165870311200003410@192.168.0.173
<— Reliably Transmitting (no NAT) to 192.168.0.173:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.173;branch=z9hG4bKac891662743;received=192.168.0.173
From: “Anonymous” sip:900@192.168.0.246;tag=1c891659115
To: sip:453@myproxy;tag=as4d351114
Call-ID: 89165870311200003410@192.168.0.173
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="0e3283b7"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘89165870311200003410@192.168.0.173’ in 32000 ms (Method: INVITE)
Found user ‘900’
<— SIP read from 192.168.0.173:5060 —>
ACK sip:453@myproxy SIP/2.0
Via: SIP/2.0/UDP 192.168.0.173;branch=z9hG4bKac891662743
Max-Forwards: 70
From: “Anonymous” sip:900@192.168.0.246;tag=1c891659115
To: sip:453@myproxy;tag=as4d351114
Call-ID: 89165870311200003410@192.168.0.173
CSeq: 1 ACK
Contact: sip:900@192.168.0.173
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-118 FXO/v.5.40A.015.004
Content-Length: 0
<------------->
— (12 headers 0 lines) —
<— SIP read from 192.168.0.173:5060 —>
INVITE sip:453@myproxy SIP/2.0
Via: SIP/2.0/UDP 192.168.0.173;branch=z9hG4bKac891694777
Max-Forwards: 70
From: “Anonymous” sip:900@192.168.0.246;tag=1c891659115
To: sip:453@myproxy
Call-ID: 89165870311200003410@192.168.0.173
CSeq: 2 INVITE
Proxy-Authorization: Digest username=“900”,realm=“asterisk”,nonce=“0e3283b7”,uri=“sip:453@myproxy”,algorithm=MD5,response="6f10e0be58685576b997b63de538aa53"
Contact: sip:900@192.168.0.173
Supported: em,100rel,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-118 FXO/v.5.40A.015.004
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 267
v=0
o=AudiocodesGW 891653273 891653157 IN IP4 192.168.0.173
s=Phone-Call
c=IN IP4 192.168.0.173
t=0 0
m=audio 6000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6001 IN IP4 192.168.0.173
<------------->
— (15 headers 12 lines) —
Sending to 192.168.0.173 : 5060 (no NAT)
Using INVITE request as basis request - 89165870311200003410@192.168.0.173
Found user '900’
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.173:6000
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.0.173:6000
Looking for 453 in from-sip (domain myproxy)
list_route: hop: sip:900@192.168.0.173
<— Transmitting (no NAT) to 192.168.0.173:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.173;branch=z9hG4bKac891694777;received=192.168.0.173
From: “Anonymous” sip:900@192.168.0.246;tag=1c891659115
To: sip:453@myproxy
Call-ID: 89165870311200003410@192.168.0.173
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:453@192.168.0.246
Content-Length: 0
Rajeev.
They are not sending you the caller ID. The only numbers they have are:
453@myproxy which is presumably your DNID value; and
900@192.168.0.246 which is presumably the account number to which you referred.
If you had the upstream caller ID it would be in this line:
They definitely need to change their end. It is possible that you will need to use insecure=invite, at your end.
Incidentally, you didn’t say that this was Asterisk 1.6.x
Sir,
I am using asterisk 1.4
I thought the change to using proxy-authenticate was between 1.4 and 1.6. I must have been wrong. It doesn’t matter that much though as you have to get your supplier to change their system first.
It would help to be more precise about which version you are using.
Sir,
900 is the port number on which call receive and 453 is extension on which call is forward.
Rajeev.