Can't call out error: channel.c: Prodding channel ' ' Failed

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:801[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 [[email protected]:1] ResetCDR("SIP/801-00000032", "") in new stack
[2015-03-28 09:38:42] VERBOSE[63793][C-00000022] pbx.c: Executing [[email protected]:2] NoCDR("SIP/801-00000032", "") in new stack
[2015-03-28 09:38:42] VERBOSE[63793][C-00000022] pbx.c: Executing [[email protected]: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 [[email protected]:4] Wait("SIP/801-00000032", "1") in new stack
[2015-03-28 09:38:43] VERBOSE[63793][C-00000022] pbx.c: Executing [[email protected]:5] Progress("SIP/801-00000032", "") in new stack
[2015-03-28 09:38:43] VERBOSE[63793][C-00000022] pbx.c: Executing [[email protected]: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 [[email protected]:7] Wait("SIP/801-00000032", "1") in new stack
[2015-03-28 09:38:50] VERBOSE[63793][C-00000022] pbx.c: Executing [[email protected]: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 [[email protected]: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

Hi, did you find a solution to this. I am also a newbie trying to setup with sipgate and I have the same error in my log. Prodding Channel …

Internal extension work; sipgate tells me my system is online … but I cannot get an outbound call to work?

Hi Robert!

Obth/Jared was last seen on May 18, '15 so I seriously doubt you will get an answer from him…

If you want people to be able to help please post your logs while attempting a call…

That said, SIPgate seems to be require a more complex setup than most providers as you need to set the P-Preferred-Identity header to your caller id in E164 format.

For outgoing calls, please enter the sender number in E.164 format (i.e. international format without the leading zeros or plus (+) sign) as a new header P-Preferred-Identity:

SIPAddHeader(P-Preferred-Identity: sip:[email protected])

(or sipgate.de or any of the other possible countries…)

from https://teamhelp.sipgate.co.uk/hc/en-gb/articles/207414635-How-Do-I-Configure-Asterisk-for-sipgate-trunking-

If you think this might be your problem than take a look at

The solution posted there will not work if you are using PJSIP but you can easily adjust things so that they work with the using the information @lgaetz provided in that thread…

Good luck and have a nice day!

Nick