I have an inbound route setup with the did in the exact format of the invite from the provider. I have the call terminated with a hold forever for testing to eliminate endpoint issues. There is also an any/any inbound route.
Asterisk is not accepting inbound calls.
Sitting at asterisk cli with sip debug on I see no activity. Wireshark pcap shows 3 invites and then a cancel all on inbound leg of call with no response from asterisk.
I am not trying, I am telling you how to configure it. You should really treat the help you are getting here with more respect !
The UNMONITORED status is not good because you do not know for sure if the provider responses to the OPTIONS from the trunk.
If you see UNKNOWN status, it means that you have problem with your firewall or NAT port forwarding settings.
If you will use the context=default make sure that it is really exists in your pbx or delete it in order to use the default one of the Freepbx. If you get invites from your provider please send the logs to get a clearer view of what happening.
I would also suggest to check your firewall rules and port forwarding settings.
I didn’t realize that thanking you for your suggestion and trying it out was disrespectful. The unknown was after applying your suggested settings. With the settings I was using, I get unmonitored. see below:
Firewall has all sip ports open to both 64.136.174.30 and 64.136.173.31. Since it is hosted at RentPBX there is no hardware firewall and therefore no port forwarding.
I did a pcap and it shows a pattern of 3 invites, then a cancel, then 3 more invites etc. until it times out. So it doesn’t appear to be the trunk. Something with Asterisk?
I do have 6 other hosts set up with this provider with the same settings (see what they specify below) and all works just fine on other systems. I was trying context as a suggestion and it did not make any difference. I have tried default which should work and from-pstn which does exist and should also work.
CLI> sip set debug peer inbound-VoIP-Innovations
SIP Debugging Enabled for IP: 64.136.173.31
There was no response on the CLI>
I then exited the cli and ran tshark on eth0 and captured udp packets and saw the data from the provider. Somehow asterisk does not appear to be listening on eth0?
I’ll post a link for you to look at the dump from tshark if you like. I just sent you a link via email.
I also went into advanced settings and set: SIP channel driver to chan_sip instead of both chan_sip and chan_pjsip. I then restarted asterisk with an amportal resrtart.
I went to the asterisk cli and set:
ip set debug peer inbound-VoIP-Innovations
SIP Debugging Enabled for IP: 64.136.173.31
Yes I did and I am now able to route the inbound call (after 5 to 10 second delay into music on hold) so great there is some progress. But it also created a new problem as my phone will no longer register and the cli shows:
[2015-11-12 11:46:50] NOTICE[23064]: chan_sip.c:27875 handle_request_register: Registration from ‘“Abc Defdef” sip:[email protected]’ failed for ‘50.198.133.1:5062’ - Wrong password
Good to hear that you progressed. Check your password in the extension settings and in your phone again. Can you send here your trunk settings screenshot?
UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Path support : No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: FPBX-12.0.74(13.5.0)
SDP Session Name: Asterisk PBX 13.5.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: 4294967295
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No
Network QoS Settings:
IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Codecs: (ulaw|alaw|gsm|g726)
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 30
RTP Hold Timeout: 300
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Sub. min duration 60 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Outbound reg. retry 403:0
Notify ringing state: Yes
Include CID: No
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy:
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70
Default Settings:
Allowed transports: UDP
Outbound transport: UDP
Context: from-sip-external
Record on feature: automon
Record off feature: automon
Force rport: No
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: No
Language:
Tone zone:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97
I’ll email the logs
