Good day everyone,
I am having a bit of a problem executing outbound calls from my FreePBX/Asterisk server to a mobile phone (for test purpouses I am using my own phone number, at this time) using an iQsim CR250 device. The device contains two working SIM cards whose Network connectivity was tested sending an SMS through an http request.
In order to communicate with the devide, I created a SIP trunk from my FreePBX server and I am using a softphone conntected to my PBX server to make the calls. The call starts, but my mobile phone never rings and, after a while, it stops giving out a warning “app_dial.c: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)”. I checked to see if I had any sip registered and it doesn’t look like it, although the peers I have are the following:
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
101/101 xxx.xxx.xxx.xxx D Yes Yes A 32569 OK (3 ms)
sip-trunk/administrator yyy.yyy.yyy.yyy Yes Yes 5160 UNREACHABLE
2 sip peers [Monitored: 1 online, 1 offline Unmonitored: 0 online, 0 offline]
where the first is my softphone (xxx.xxx.xxx.xxx is the IP of my FreeBPX server) and the second one is the trunk I created to connect to the devide (yyy.yyy.yyy.yyy is the IP address).
The return for sip show registry is
Host dnsmgr Username Refresh State Reg.Time
0 SIP registrations.
My sip.conf file is as follows:
[101] //softphone
deny=0.0.0.0/0.0.0.0
secret=seletech
dtmfmode=rfc2833
canreinvite=no
context=outbound
host=dynamic
defaultuser=
trustrpid=yes
user_eq_phone=no
sendrpid=pai
type=peer
session-timers=accept
nat=force_rport,comedia
port=5160
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
force_avp=no
icesupport=no
rtcp_mux=no
encryption=no
namedcallgroup=
namedpickupgroup=
dial=SIP/101
accountcode=
permit=0.0.0.0/0.0.0.0
callerid=A6 <101>
recordonfeature=apprecord
recordofffeature=apprecord
callcounter=yes
faxdetect=no
[sip-trunk] //trunk
disallow=all
type=peer
host=yyy.yyy.yyy.yyy
username=usr
secret=psw
fromuser=usr
port=5160
qualify=yes
dtmfmode=rfc2833
canreinvite=no
allow=ulaw
allow=alaw
allow=gsm
insecure=port,invite
fromdomain=yyy.yyy.yyy.yyy
register=> usr:[email protected]:5160
context=from-trunk-sip-sip-trunk
The dialplan I am using is:
[outbound]
exten => _X.,1,Dial(SIP/sip-trunk/${ESTEN})
Does anyone know how to connect to a device like this one? Which username and passwrd should I use? At hte moment I am using the ones needed to acces the configuration, but I also tried with the login and password of one of the users I created on the device configuration. I also turned on the sip debugging and I will leave the result below.
VERBOSE[2536] chan_sip.c: Reliably Transmitting (NAT) to yyy.yyy.yyy.yyy:5160:
21403 OPTIONS sip:yyy.yyy.yyy.yyy SIP/2.0
21404 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5160;branch=z9hG4bK36931901;rport
21405 Max-Forwards: 70
21406 From: "Unknown" <sip:[email protected]:5160>;tag=as274bd0bc
21407 To: <sip:yyy.yyy.yyy.yyy>
21408 Contact: <sip:[email protected]:5160>
21409 Call-ID: [email protected]:5160
21410 CSeq: 102 OPTIONS
21411 User-Agent: FPBX-15.0.17.55(16.17.0)
21412 Date: Wed, 10 Nov 2021 10:15:28 GMT
21413 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
21414 Supported: replaces, timer
21415 Content-Length: 0
21416
21417
21418 ---
21419 [2021-11-10 10:15:29] VERBOSE[2536] chan_sip.c: Retransmitting #1 (NAT) to yyy.yyy.yyy.yyy:5160:
21420 OPTIONS sip:yyy.yyy.yyy.yyy SIP/2.0
21421 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5160;branch=z9hG4bK36931901;rport
21422 Max-Forwards: 70
21423 From: "Unknown" <sip:[email protected]:5160>;tag=as274bd0bc
21424 To: <sipyyy.yyy.yyy.yyy>
21425 Contact: <sip:[email protected]:5160>
21426 Call-ID: [email protected]:5160
21427 CSeq: 102 OPTIONS
21428 User-Agent: FPBX-15.0.17.55(16.17.0)
21429 Date: Wed, 10 Nov 2021 10:15:28 GMT
21430 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
21431 Supported: replaces, timer
21432 Content-Length: 0
21433
21434
21435 ---
21436 [2021-11-10 10:15:30] VERBOSE[2536] chan_sip.c: Retransmitting #2 (NAT) to yyy.yyy.yyy.yyy:5160:
21437 OPTIONS sip:yyy.yyy.yyy.yyy SIP/2.0
21438 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5160;branch=z9hG4bK36931901;rport
21439 Max-Forwards: 70
21440 From: "Unknown" <sip:[email protected]:5160>;tag=as274bd0bc
21441 To: <sip:yyy.yyy.yyy.yyy>
21442 Contact: <sip:[email protected]:5160>
21443 Call-ID: [email protected]:5160
21444 CSeq: 102 OPTIONS
21445 User-Agent: FPBX-15.0.17.55(16.17.0)
21446 Date: Wed, 10 Nov 2021 10:15:28 GMT
21447 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
21448 Supported: replaces, timer
21449 Content-Length: 0
21450
21451
21452 ---
21453 [2021-11-10 10:15:31] VERBOSE[2536] chan_sip.c: Retransmitting #3 (NAT) to yyy.yyy.yyy.yyy:5160:
21454 OPTIONS sip:yyy.yyy.yyy.yyy SIP/2.0
21455 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5160;branch=z9hG4bK36931901;rport
21456 Max-Forwards: 70
21457 From: "Unknown" <sip:[email protected]:5160>;tag=as274bd0bc
21458 To: <sip:yyy.yyy.yyy.yyy>
21459 Contact: <sip:[email protected]:5160>
21460 Call-ID: [email protected]:5160
21461 CSeq: 102 OPTIONS
21462 User-Agent: FPBX-15.0.17.55(16.17.0)
21463 Date: Wed, 10 Nov 2021 10:15:28 GMT
21464 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
21465 Supported: replaces, timer
21466 Content-Length: 0
21467
21468
21469 ---
21470 [2021-11-10 10:15:32] VERBOSE[2536] chan_sip.c: Retransmitting #4 (NAT) to yyy.yyy.yyy.yyy:5160:
21471 OPTIONS sip:yyy.yyy.yyy.yyy SIP/2.0
21472 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5160;branch=z9hG4bK36931901;rport
21473 Max-Forwards: 70
21474 From: "Unknown" <sip:[email protected]:5160>;tag=as274bd0bc
21475 To: <sip:yyy.yyy.yyy.yyy>
21476 Contact: <sip:[email protected]:5160>
21477 Call-ID: [email protected]:5160
21478 CSeq: 102 OPTIONS
21479 User-Agent: FPBX-15.0.17.55(16.17.0)
21480 Date: Wed, 10 Nov 2021 10:15:28 GMT
21481 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
21482 Supported: replaces, timer
21483 Content-Length: 0
21484
21485
21486 ---
21487 [2021-11-10 10:15:32] VERBOSE[2536] chan_sip.c: Really destroying SIP dialog '[email protected]:5160' Method: OPTIONS
21488 [2021-11-10 10:15:42] VERBOSE[2536] chan_sip.c: Reliably Transmitting (NAT) to 10.182.233.232:5160:
21489 OPTIONS sip:yyy.yyy.yyy.yyy SIP/2.0
21490 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5160;branch=z9hG4bK10aa98d0;rport
21491 Max-Forwards: 70
21492 From: "Unknown" <sip:[email protected]:5160>;tag=as59ccdad0
21493 To: <sip:yyy.yyy.yyy.yyy>
21494 Contact: <sip:[email protected]:5160>
21495 Call-ID: [email protected]:5160
21496 CSeq: 102 OPTIONS
21497 User-Agent: FPBX-15.0.17.55(16.17.0)
21498 Date: Wed, 10 Nov 2021 10:15:42 GMT
21499 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
21500 Supported: replaces, timer
21501 Content-Length: 0
Where xxx.xxx.xxx.xxx = PBX IP and yyy.yyy.yyy.yyy = GATEWAY IP
Can anyone help me figure out why I don’t seem to be able to connect to my gateway?