Hi
I’m a complete newbie to freepbx/asterisk but I’ve got a very simple setup at the moment and yet I can’t make any calls.
I’ve got a static external IP, nat turned off and no firewall on (while I’m testing) and one cisco phone. I’m using Sipgate incoming and outgoing.
I’m trying to call the sipgate test call number 10005 from my phone but the call never connects. On the sipgate website my account is showing as active so they think I’m connected at least.
I have nat turned on within my phone (I think I’ve read that is the best way for cisco phones) but I’ve tried turning that off without any success.
Looking in the logs this is the only actual error I can see
WARNING[4097][C-00000023] channel.c: Prodding channel ‘SIP/801-00000033’ failed
I don’t even know where to begin with this.
Here’s the verbose call log of the entire call
[2015-03-28 09:38:42] VERBOSE[1888] chan_sip.c: Sending to 192.168.0.18:5060 (no NAT)
[2015-03-28 09:38:42] VERBOSE[1888][C-00000022] chan_sip.c: Sending to 192.168.0.18:5060 (no NAT)
[2015-03-28 09:38:42] VERBOSE[1888][C-00000022] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 192.168.0.18:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 151.228.56.223:5060;branch=z9hG4bKea3855ca;received=192.168.0.18
From: "801" <sip:[email protected]>;tag=0022555e197500189aa8690a-1b9ad40b
To: <sip:[email protected]>;tag=as08513794
Call-ID: [email protected]
CSeq: 101 INVITE
Server: FPBX-AsteriskNOW-12.0.50.1(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3f736e21"
Content-Length: 0
<------------>
[2015-03-28 09:38:42] VERBOSE[1888][C-00000022] chan_sip.c: Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
[2015-03-28 09:38:42] VERBOSE[1888][C-00000022] chan_sip.c: Sending to 192.168.0.18:5060 (no NAT)
[2015-03-28 09:38:42] VERBOSE[1888][C-00000022] netsock2.c: Using SIP RTP TOS bits 184
[2015-03-28 09:38:42] VERBOSE[1888][C-00000022] netsock2.c: Using SIP RTP CoS mark 5
[2015-03-28 09:38:42] VERBOSE[1888][C-00000022] chan_sip.c: Found RTP audio format 0
[2015-03-28 09:38:42] VERBOSE[1888][C-00000022] chan_sip.c: Found RTP audio format 8
[2015-03-28 09:38:42] VERBOSE[1888][C-00000022] chan_sip.c: Found RTP audio format 18
[2015-03-28 09:38:42] VERBOSE[1888][C-00000022] chan_sip.c: Found RTP audio format 102
[2015-03-28 09:38:42] VERBOSE[1888][C-00000022] chan_sip.c: Found RTP audio format 116
[2015-03-28 09:38:42] VERBOSE[1888][C-00000022] chan_sip.c: Found RTP audio format 101
[2015-03-28 09:38:42] VERBOSE[1888][C-00000022] chan_sip.c: Found audio description format PCMU for ID 0
[2015-03-28 09:38:42] VERBOSE[1888][C-00000022] chan_sip.c: Found audio description format PCMA for ID 8
[2015-03-28 09:38:42] VERBOSE[1888][C-00000022] chan_sip.c: Found audio description format G729 for ID 18
[2015-03-28 09:38:42] VERBOSE[1888][C-00000022] chan_sip.c: Found audio description format L16 for ID 102
[2015-03-28 09:38:42] VERBOSE[1888][C-00000022] chan_sip.c: Found audio description format iLBC for ID 116
[2015-03-28 09:38:42] VERBOSE[1888][C-00000022] chan_sip.c: Found audio description format telephone-event for ID 101
[2015-03-28 09:38:42] VERBOSE[1888][C-00000022] chan_sip.c: Capabilities: us - (ilbc|ulaw|alaw|gsm|g726|g726aal2|g722|g729), peer - audio=(ulaw|alaw|g729|slin16|ilbc)/video=(nothing)/text=(nothing), combined - (ilbc|ulaw|alaw|g729)
[2015-03-28 09:38:42] VERBOSE[1888][C-00000022] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing)
[2015-03-28 09:38:42] VERBOSE[1888][C-00000022] chan_sip.c: Peer audio RTP is at port 151.228.56.223:16386
[2015-03-28 09:38:42] VERBOSE[1888][C-00000022] chan_sip.c: Looking for 10005 in from-internal (domain 192.168.0.20)
[2015-03-28 09:38:42] VERBOSE[1888][C-00000022] sip/route.c: sip_route_dump: route/path hop: <sip:[email protected]:5060;user=phone;transport=udp>
[2015-03-28 09:38:42] VERBOSE[1888][C-00000022] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.0.18:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 151.228.56.223:5060;branch=z9hG4bKe36238f2;received=192.168.0.18
From: "801" <sip:[email protected]>;tag=0022555e197500189aa8690a-1b9ad40b
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-AsteriskNOW-12.0.50.1(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
[2015-03-28 09:38:42] VERBOSE[63793][C-00000022] pbx.c: Executing [10005@from-internal:1] ResetCDR("SIP/801-00000032", "") in new stack
[2015-03-28 09:38:42] VERBOSE[63793][C-00000022] pbx.c: Executing [10005@from-internal:2] NoCDR("SIP/801-00000032", "") in new stack
[2015-03-28 09:38:42] VERBOSE[63793][C-00000022] pbx.c: Executing [10005@from-internal:3] Progress("SIP/801-00000032", "") in new stack
[2015-03-28 09:38:42] VERBOSE[63793][C-00000022] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.0.18:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 151.228.56.223:5060;branch=z9hG4bKe36238f2;received=192.168.0.18
From: "801" <sip:[email protected]>;tag=0022555e197500189aa8690a-1b9ad40b
To: <sip:[email protected]>;tag=as1d8721dc
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-AsteriskNOW-12.0.50.1(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
[2015-03-28 09:38:42] VERBOSE[63793][C-00000022] pbx.c: Executing [10005@from-internal:4] Wait("SIP/801-00000032", "1") in new stack
[2015-03-28 09:38:43] VERBOSE[63793][C-00000022] pbx.c: Executing [10005@from-internal:5] Progress("SIP/801-00000032", "") in new stack
[2015-03-28 09:38:43] VERBOSE[63793][C-00000022] pbx.c: Executing [10005@from-internal:6] Playback("SIP/801-00000032", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
[2015-03-28 09:38:43] VERBOSE[63793][C-00000022] file.c: <SIP/801-00000032> Playing 'silence/1.gsm' (language 'en')
[2015-03-28 09:38:44] VERBOSE[63793][C-00000022] file.c: <SIP/801-00000032> Playing 'cannot-complete-as-dialed.gsm' (language 'en')
[2015-03-28 09:38:47] VERBOSE[63793][C-00000022] file.c: <SIP/801-00000032> Playing 'check-number-dial-again.gsm' (language 'en')
[2015-03-28 09:38:49] VERBOSE[63793][C-00000022] pbx.c: Executing [10005@from-internal:7] Wait("SIP/801-00000032", "1") in new stack
[2015-03-28 09:38:50] VERBOSE[63793][C-00000022] pbx.c: Executing [10005@from-internal:8] Congestion("SIP/801-00000032", "20") in new stack
[2015-03-28 09:38:50] VERBOSE[63793][C-00000022] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 192.168.0.18:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 151.228.56.223:5060;branch=z9hG4bKe36238f2;received=192.168.0.18
From: "801" <sip:[email protected]>;tag=0022555e197500189aa8690a-1b9ad40b
To: <sip:[email protected]>;tag=as1d8721dc
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-AsteriskNOW-12.0.50.1(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
[2015-03-28 09:38:50] WARNING[63793][C-00000022] channel.c: Prodding channel 'SIP/801-00000032' failed
[2015-03-28 09:38:50] VERBOSE[63793][C-00000022] pbx.c: Spawn extension (from-internal, 10005, 8) exited non-zero on 'SIP/801-00000032'
[2015-03-28 09:38:50] VERBOSE[63793][C-00000022] pbx.c: Executing [h@from-internal:1] Hangup("SIP/801-00000032", "") in new stack
[2015-03-28 09:38:50] VERBOSE[63793][C-00000022] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/801-00000032'
[2015-03-28 09:38:50] VERBOSE[1888] chan_sip.c: Really destroying SIP dialog '[email protected]' Method: ACK