@russelMcNuggets wrote:
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 isHost 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:psw@yyy.yyy.yyy.yyy:5160 context=from-trunk-sip-sip-trunkThe 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:administrator@xxx.xxx.xxx.xxx:5160>;tag=as274bd0bc 21407 To: <sip:yyy.yyy.yyy.yyy> 21408 Contact: <sip:administrator@xxx.xxx.xxx.xxx:5160> 21409 Call-ID: 656fc9067ef0e38009d24a57025d4bb9@xxx.xxx.xxx.xxx: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:administrator@xxx.xxx.xxx.xxx:5160>;tag=as274bd0bc 21424 To: <sipyyy.yyy.yyy.yyy> 21425 Contact: <sip:administrator@xxx.xxx.xxx.xxx:5160> 21426 Call-ID: 656fc9067ef0e38009d24a57025d4bb9@xxx.xxx.xxx.xxx: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:administrator@xxx.xxx.xxx.xxx:5160>;tag=as274bd0bc 21441 To: <sip:yyy.yyy.yyy.yyy> 21442 Contact: <sip:administrator@xxx.xxx.xxx.xxx:5160> 21443 Call-ID: 656fc9067ef0e38009d24a57025d4bb9@xxx.xxx.xxx.xxx: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:administrator@xxx.xxx.xxx.xxx:5160>;tag=as274bd0bc 21458 To: <sip:yyy.yyy.yyy.yyy> 21459 Contact: <sip:administrator@xxx.xxx.xxx.xxx:5160> 21460 Call-ID: 656fc9067ef0e38009d24a57025d4bb9@xxx.xxx.xxx.xxx: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:administrator@xxx.xxx.xxx.xxx:5160>;tag=as274bd0bc 21475 To: <sip:yyy.yyy.yyy.yyy> 21476 Contact: <sip:administrator@xxx.xxx.xxx.xxx:5160> 21477 Call-ID: 656fc9067ef0e38009d24a57025d4bb9@xxx.xxx.xxx.xxx: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 '656fc9067ef0e38009d24a57025d4bb9@10.181.233.205:5160' 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:administrator@xxx.xxx.xxx.xxx:5160>;tag=as59ccdad0 21493 To: <sip:yyy.yyy.yyy.yyy> 21494 Contact: <sip:administrator@xxx.xxx.xxx.xxx:5160> 21495 Call-ID: 667fb92608ad96e22da0225e2887f144@xxx.xxx.xxx.xxx: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: 0Where 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?
Posts: 1
Participants: 1