Call transfer issue

Asterisk Asterisk Support
1

Hi all,

I meet call transfer issue.
I want to connect A to C.

A : GSM mobile (no SIP)
B : phone1 (no SIP)
C : IP phone (user gugiui) rings by calling 022yyy6060

A -> B -> C

C rings but hangup after 1 second.

<------------->
— (10 headers 0 lines) —
sip_route_dump: route/path hop: sip:guigui@10.74.148.134:1915
_ – SIP/guigui-00002c66 is ringing_
Reliably Transmitting (no NAT) to 10.65.9.1:62829:
OPTIONS sip:iphone-dl@10.65.9.1:62829;rinstance=EC92814D SIP/2.0
Via: SIP/2.0/UDP 195.x.x.x:5060;branch=z9hG4bK07597331
Max-Forwards: 70
From: “anonymous” sip:anonymous@195.x.x.x;tag=as2523a92f
To: sip:iphone-dl@10.65.9.1:62829;rinstance=EC92814D
Contact: sip:anonymous@195.x.x.x:5060
Call-ID: 0e9b032250f3256a2bb53b12201eacc1@195.x.x.x:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.10.0
Date: Mon, 19 Jun 2017 09:16:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

—
Scheduling destruction of SIP dialog ‘6084de9c0ef4bdb6313e52d1122df265@195.x.x.x:5060’ in 6528 ms (Method: INVITE)
Reliably Transmitting (NAT) to 195.y.y.y:1915:
CANCEL sip:guigui@10.74.148.134:1915;rinstance=59BD586E SIP/2.0
Via: SIP/2.0/UDP 195.x.x.x:5060;branch=z9hG4bK0c41fb69;rport
Max-Forwards: 70
From: sip:4122xxxxxxx@195.x.x.x;tag=as7243753f
To: sip:guigui@10.74.148.134:1915;rinstance=59BD586E
Call-ID: 6084de9c0ef4bdb6313e52d1122df265@195.x.x.x:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 13.10.0
Content-Length: 0

—
Scheduling destruction of SIP dialog ‘6084de9c0ef4bdb6313e52d1122df265@195.x.x.x:5060’ in 6528 ms (Method: INVITE)
_ == Spawn extension (interco, 6060, 1) exited non-zero on ‘IAX2/interco-6804’_
_ – Hungup ‘IAX2/interco-6804’_
_ _
_ _
SIP Account :
_ _
[guigui]
qualify=1000
nat=force_rport,comedia
callerid=G <4122zzzzzzz>
context=billing-dialout
call-limit=1
canreinvite=no
limitonpeers=yes
secret=xxx
host=dynamic
username=guigui
dtmfmode=rfc2833
allowtransfer=yes
directmedia=yes
type=friend
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g722

There is no issue if we directly call 022yyyyyyy.

Thx for your help,
G

2

The log starts too late. Asterisk has already decided to cancel the call before it starts.

Also, the log only shows SIP activity and you haven’t told us how the “no SIP” devices are connected to Asterisk.

3

Generally there are three reasons why Asterisk would send CANCEL:

  1. the call timed out;
  2. the caller abandoned the call;
  3. the & notation was used when dialling, and one of the other devices answered.