After reading a gazillion posts I have gotten the inbound calls to stop dropping and sound clear. (no nat and codec g729 issue) Thanks for those posts.
I have gotten endpoint manager to recognize the phones and program them ( had to install every version until I got one to work 2.10.3.6 works on 2.8 so far no prob i think lol.) Also you have to enable the phone/model in endpoint configuration as well as add the mac to the oui…
Sorry if I have forgotten anything as
I am new to unix
I am new to freepbx
I am a little frustrated
I appreciate any help from anyone.
(I actually forgot to write the challenge at hand, lol smh)
I have done asterisk -rvvvvvv and when dialing from the phone all it says is this:
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: No
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
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-2.8.1(1.8.7.0)
SDP Session Name: Asterisk PBX 1.8.7.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Legacy userfield parse: No
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
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
Network Settings:
SIP address remapping: Disabled, no localnet list
Externhost:
Externaddr: (null)
Externrefresh: 10
Global Signalling Settings:
Codecs: 0xe (gsm|ulaw|alaw)
Codec Order: ulaw:20,alaw:20,gsm:20
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 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
Force rport: No
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97
I don’t think it has anything to do with the dial plan, because I’m in a similar situation. The dial plan matches the default SPA303 factory settings. I cannot dial out to any number, though the phone worked without issue if it was manually provisioned (not with endpoint manager).
I just verified that manually configuring the phone results in no issues dialing in or out. It is related to the xml config that is posted for the SPA303
I also had a similar problem in setting up my Cisco SPA 303s using Endpoint manager. I was able to receive calls, but not make them from the phone. I accidentally discovered that if I gave the phone a Station name in the Endpoint manager (edit custom template of the phone) reboot, then the phone suddenly worked fine.