Call forward with refer and CTI data

Asterisk Asterisk SIP
1

I am working on setting up call forwarding. Currently, a user calls PBX A (not my PBX) from the PSTN, and then PBX A forwards the call to my Asterisk PBX. PBX A and my Asterisk PBX are connected via a SIP trunk. (PBX A registered in Asterisk PBX pjsip as a endpoint)

; pjsip in Asterisk PBX
[PBX_A]
type=endpoint
context=PBX_A
disallow=all
allow=ulaw
outbound_auth=PBX_A
aors=PBX_A

[PBX_A]
type=auth
auth_type=userpass
password=
username=

[PBX_A]
type=aor
contact=sip:XX.XXX.XX.XX:5060
max_contacts=1

The call flow is as follows: User from PSTN → PBX A → Asterisk PBX

Now, I want to forward the user’s call back to an endpoint in PBX A (let say endpoint 5555). I am using the Dial() function in Asterisk. However, PBX A requires that I use a REFER package for call forwarding. How can I achieve this?

Here is the example that PBX A expects:

image
image1338×950 93 KB

Thanks

2

https://docs.asterisk.org/Asterisk_20_Documentation/API_Documentation/Dialplan_Applications/Transfer/

3

Thanks for reply.

I tried Transfer(), but it cannot transfer the call to the PBX A endpoint.
It will make the call end.
Here is my code:

; when user click #, user will goto this Transfer function
exten = 1234,1,Transfer(pjsip/5555@PBX_A)

I used WireShark to catch the SIP package, nothing happend related to PBX_A.
It only stay in the local network. All IP are Asterisk local IP, the final error is SIP/2.0 500 Server Internal Error, which the error is come from Asterisk PBX.

4

chan_pjsip requires a full URI after the /.

Also note that, to use REFER, the call must have been answered, otherwise a 302 status will be used.

5

Sorry I am new in Asterisk.
May I ask for an example of ‘full URI after the /’ ?

If the IP address is XX.XXX.XX.XX, Is it look like this?

pjsip/XX.XXX.XX.XX:5060/5555

Thanks

6

See SIP-URI and SIPS-URI, towards the middle of this page RFC 3261: SIP: Session Initiation Protocol

7

However, you cannot set even valid headers on REFER, and CTI_Data is not a valid header name, as it is not marked as being non-standard, and there appears to be no RFC defining it.

8

Do you mean that I cannot set CTI_Data in the Refer with Asterisk?

9
10

Seem it can transfer the call, but the call changed to on hold, after a few seconds it pop up Time out request
PBX_A still did not receive anything.

Am I missing anything?

Here is my extensions.conf in Asterisk
image
image

Here is my Refer package, which 10.11.30.80 is my Asterisk PBX local IP

ram
ram1706×788 84.8 KB

Thanks

11

Please provide the complete REFER transaction, as plain text, not an image.

12

Sorry about it, here is the plain text
I saw there are a 5524 NOTIFY after the call forward.

In this case, call flow:
1001@10.11.30.26 → (call) → 1000@10.11.30.80 → (call forward) → sip:5555@XX.XXX.XX.XX

<--- Transmitting SIP request (706 bytes) to UDP:10.11.30.26:56759 --->
REFER sip:1001@10.11.30.26:56759;ob SIP/2.0
Via: SIP/2.0/UDP 10.11.30.80:5060;rport;branch=z9hG4bKPj60da3e12-d22f-423a-a861-63603849a018
From: <sip:1000@10.11.30.80>;tag=861253b2-54ff-4312-9b16-08b479ac4b6f
To: <sip:1001@10.11.30.80>;tag=22c973b683c441419e4fe51990df640a
Contact: <sip:10.11.30.80:5060>
Call-ID: 6d805f31f53d4ad7b8ceaf7018b2c202
CSeq: 5881 REFER
Event: refer
Expires: 600
Supported: 100rel, timer, replaces, norefersub
Accept: message/sipfrag;version=2.0
Allow-Events: presence, dialog, message-summary, refer
Refer-To: sip:5555@XX.XXX.XX.XX
Referred-By: <sip:1000@10.11.30.80>
Max-Forwards: 70
User-Agent: Asterisk PBX 18.10.0~dfsg+~cs6.10.40431411-2
Content-Length:  0


<--- Received SIP response (576 bytes) from UDP:10.11.30.26:56759 --->
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 10.11.30.80:5060;rport=5060;received=10.11.30.80;branch=z9hG4bKPj60da3e12-d22f-423a-a861-63603849a018
Call-ID: 6d805f31f53d4ad7b8ceaf7018b2c202
From: <sip:1000@10.11.30.80>;tag=861253b2-54ff-4312-9b16-08b479ac4b6f
To: <sip:1001@10.11.30.80>;tag=22c973b683c441419e4fe51990df640a
CSeq: 5881 REFER
Expires: 600
Contact: <sip:1001@10.11.30.26:56759;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub, trickle-ice
Content-Length:  0


<--- Received SIP response (576 bytes) from UDP:10.11.30.26:56759 --->
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 10.11.30.80:5060;rport=5060;received=10.11.30.80;branch=z9hG4bKPj60da3e12-d22f-423a-a861-63603849a018
Call-ID: 6d805f31f53d4ad7b8ceaf7018b2c202
From: <sip:1000@10.11.30.80>;tag=861253b2-54ff-4312-9b16-08b479ac4b6f
To: <sip:1001@10.11.30.80>;tag=22c973b683c441419e4fe51990df640a
CSeq: 5881 REFER
Expires: 600
Contact: <sip:1001@10.11.30.26:56759;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub, trickle-ice
Content-Length:  0


<--- Received SIP request (576 bytes) from UDP:10.11.30.26:56759 --->
NOTIFY sip:10.11.30.80:5060 SIP/2.0
Via: SIP/2.0/UDP 10.11.30.26:56759;rport;branch=z9hG4bKPj418872ed3dcf4a4a8035dffe3082ea02
Max-Forwards: 70
From: <sip:1001@10.11.30.80>;tag=22c973b683c441419e4fe51990df640a
To: <sip:1000@10.11.30.80>;tag=861253b2-54ff-4312-9b16-08b479ac4b6f
Contact: <sip:1001@10.11.30.26:56759;ob>
Call-ID: 6d805f31f53d4ad7b8ceaf7018b2c202
CSeq: 5524 NOTIFY
Event: refer
Subscription-State: active;expires=600
Allow-Events: presence, message-summary, refer
Content-Type: message/sipfrag;version=2.0
Content-Length:    20

SIP/2.0 100 Trying

Thanks a lot !

13

You will need to debug this on 10.11.30.26. It has accepted responsibility for the REFER and appears to be still executing it when the log ends.

14

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