Outbound Route Not Working for NEW SIP Service - Cox Communications

We have installed a new PBX Free Server for SMB client. Everything is working great except Outbound calling. Quick run down of our setup:

FreePBX 13.0.167
Asterisk ver. 13.10.0

Specific error string when dialing out from extension is below: (abridged output)

[Aug 30 12:11:40] – Executing [[email protected]:23] Dial(“SIP/110-00000017”, “PJSIP/[email protected],300,T”) in new stack
[Aug 30 12:11:40] – Called PJSIP/[email protected]
[Aug 30 12:11:40] == Everyone is busy/congested at this time (1:0/0/1)
[Aug 30 12:11:40] – Executing [[email protected]:24] NoOp(“SIP/110-00000017”, “Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 1”) in new stack
[Aug 30 12:11:40] – Executing [[email protected]:25] GotoIf(“SIP/110-00000017”, “0?continue,1:s-CHANUNAVAIL,1”) in new stack
[Aug 30 12:11:40] – Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
[Aug 30 12:11:40] – Executing [[email protected]:1] Set(“SIP/110-00000017”, “RC=1”) in new stack
[Aug 30 12:11:40] – Executing [[email protected]:2] Goto(“SIP/110-00000017”, “1,1”) in new stack
[Aug 30 12:11:40] – Goto (macro-dialout-trunk,1,1)
[Aug 30 12:11:40] – Executing [[email protected]:1] Goto(“SIP/110-00000017”, “s-INVALIDNMBR,1”) in new stack
[Aug 30 12:11:40] – Goto (macro-dialout-trunk,s-INVALIDNMBR,1)
[Aug 30 12:11:40] – Executing [[email protected]:1] NoOp(“SIP/110-00000017”, “Dial failed due to trunk reporting Address Incomplete - giving up”) in new stack

Appreciate the help in troubleshooting issue.

Tell us more about your trunk set up.

I’m gonna guess that you’ve got something screwed up in the number manipulation or trunk selector parts of either the outbound route or the trunk.

Does inbound work?

Hey Dave - thanks for the input.

Inbound routing works great. Only Outbound is the issue.

SIP registration is success and ISP can see throughput of outbound calls when dialing from SIP_Chan extension. ISP tech suggested CID incomplete config as cause of issues so we verified CID was labeled for extension and outbound route.

The Edgemarc router (provided by ISP for VoIP traffic) sits in the LAN and is assigned an IP within that subnet. Outbound VoIP traffic travels out the Edgemarc through separate circuit from internet. Therefore, PBX points to internal LAN IP for outbound.

See config below:

Global Settings:

UDP Bindaddress:
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: Asterisk PBX 13.10.0
SDP Session Name: Asterisk PBX 13.10.0
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: asterisk
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

Network QoS Settings:

IP ToS RTP audio: CS0
IP ToS RTP video: CS0
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: Enabled using externaddr
Externrefresh: 10

Global Signalling Settings:

Codecs: (ulaw|alaw|gsm|g726|g729)
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
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: Yes
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: No
Language: en
Tone zone:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97

@lee did you ever figure out the problem? I have the same issue.