Rejected - extension not found

Hello,

I am new to Asterisk and recently took over as the Sys Admin for a company with an Asterisk system. I have figured out most of the system, but have one recurring problem. When I setup a new extension, everything works except that I cannot dial that extension. I can make calls from the SIP phone but when attempting to dial the new extension I see the message below.

Asterisk 1.4.17
Exact error message:
Call from ‘3513’ to extension ‘4514’ rejected because extension not found.

I have verified that 4514 is listed in show peers and well as in show users.
Show Peers result: 4514/4290bf3ab92e2d17ef82 66.116.118.50 D 5060 OK (163 ms)
Show Users result: 4514 password main No RFC3581

I have matched the dialing rules to a known working extension.

Please tell me what you need from me to help with this particular issue, as I am at a loss and none of the solutions found in forums and documentation for this problem have worked for me.

Thank you in advance for any help you can provide!

Do you have a dialplan entry for 4514 that can be reached by the 3513 peer?

Here is the dialplan show result for the extension that is not working:

asterisk*CLI> dialplan show 4514@
[ Context ‘amcat-in-hours’ created by ‘pbx_config’ ]
’_XXXX’ => 1. Goto(s|1) [pbx_config]

[ Context ‘from-pstn’ created by ‘pbx_config’ ]
‘4514’ => 1. SetMusicOnHold(usad) [pbx_config]
2. Goto(extensions|${EXTEN}|1) [pbx_config]
[ Included context ‘usad-catch-all’ created by ‘pbx_config’ ]
’_XXXX’ => 1. Macro(route-call|usad|Usad|usad-in-hours|usad-out-of-hours) [pbx_config]

[ Context ‘usad-catch-all’ created by ‘pbx_config’ ]
’_XXXX’ => 1. Macro(route-call|usad|Usad|usad-in-hours|usad-out-of-hours) [pbx_config]

-= 4 extensions (5 priorities) in 4 contexts. =-

Here is the dialplan show for an extension that is working:

asterisk*CLI> dialplan show 4521@
[ Context ‘amcat-in-hours’ created by ‘pbx_config’ ]
’_XXXX’ => 1. Goto(s|1) [pbx_config]

[ Context ‘from-pstn’ created by ‘pbx_config’ ]
‘4521’ => 1. SetMusicOnHold(usad) [pbx_config]
2. Goto(extensions|${EXTEN}|1) [pbx_config]
[ Included context ‘usad-catch-all’ created by ‘pbx_config’ ]
’_XXXX’ => 1. Macro(route-call|usad|Usad|usad-in-hours|usad-out-of-hours) [pbx_config]

[ Context ‘usad-catch-all’ created by ‘pbx_config’ ]
’_XXXX’ => 1. Macro(route-call|usad|Usad|usad-in-hours|usad-out-of-hours) [pbx_config]

-= 4 extensions (5 priorities) in 4 contexts. =-

It looks the same to me. I will also post the debug for a working and a failed call upon request. Honestly, that part of the system I am not yet acquainted enough with to know what to look for other than to spot differences, of which there are many.

Here are the log files for the calls. These are internal calls, from one extension to the other.

Calling from the broken extension (this part works):

asteriskCLI> sip set debug peer 4999
SIP Debugging Enabled for IP: 10.0.0.213:5060
Really destroying SIP dialog ‘64184ffd380cf2dc21ac76aa67eebd97@10.0.0.1’ Method: OPTIONS
asterisk
CLI> sip set debug peer 3504
SIP Debugging Enabled for IP: 10.0.0.178:5060
Reliably Transmitting (no NAT) to 10.0.0.178:5060:
OPTIONS sip:3504@10.0.0.178 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK6f3da475;rport
From: “asterisk” sip:asterisk@10.0.0.1;tag=as20a71b6d
To: sip:3504@10.0.0.178
Contact: sip:asterisk@10.0.0.1
Call-ID: 0ad7412275d08412275aaa51006b4f71@10.0.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 14 May 2014 18:57:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK6f3da475;rport
From: “asterisk” sip:asterisk@10.0.0.1;tag=as20a71b6d
To: sip:3504@10.0.0.178;tag=31C44624-21E00673
CSeq: 102 OPTIONS
Call-ID: 0ad7412275d08412275aaa51006b4f71@10.0.0.1
Contact: sip:3504@10.0.0.178
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘0ad7412275d08412275aaa51006b4f71@10.0.0.1’ Method: OPTIONS
– Executing [3504@main:1] Answer(“SIP/4999-0831e6d0”, “”) in new stack
– Executing [3504@main:2] GotoIf(“SIP/4999-0831e6d0”, “0?reset_name:dial”) in new stack
– Goto (main,3504,4)
– Executing [3504@main:4] AGI(“SIP/4999-0831e6d0”, “dial_setup.agi|3504”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/dial_setup.agi
– AGI Script dial_setup.agi completed, returning 0
– Executing [3504@main:5] Dial(“SIP/4999-0831e6d0”, “SIP/3504|20|wW”) in new stack
Audio is at 10.0.0.1 port 13578
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.0.178:5060:
INVITE sip:3504@10.0.0.178 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178
Contact: sip:4999@10.0.0.1
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 14 May 2014 18:57:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 277

v=0
o=root 4793 4793 IN IP4 10.0.0.1
s=session
c=IN IP4 10.0.0.1
t=0 0
m=audio 13578 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


-- Called 3504

asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178;tag=4F70555F-C5692AAA
CSeq: 102 INVITE
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
Contact: sip:3504@10.0.0.178
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Content-Length: 0

<------------->
— (9 headers 0 lines) —
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178;tag=4F70555F-C5692AAA
CSeq: 102 INVITE
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
Contact: sip:3504@10.0.0.178
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Allow-Events: talk,hold,conference
Content-Length: 0

<------------->
— (10 headers 0 lines) —
– SIP/3504-0831a410 is ringing
Scheduling destruction of SIP dialog ‘0fed059875d7fba14ea5b45214429272@10.0.0.1’ in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 10.0.0.178:5060:
CANCEL sip:3504@10.0.0.178 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


Scheduling destruction of SIP dialog ‘0fed059875d7fba14ea5b45214429272@10.0.0.1’ in 6400 ms (Method: INVITE)
== Spawn extension (main, 3504, 5) exited non-zero on 'SIP/4999-0831e6d0’
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178
CSeq: 102 CANCEL
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
Contact: sip:3504@10.0.0.178
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Content-Length: 0

<------------->
— (9 headers 0 lines) —
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178;tag=4F70555F-C5692AAA
CSeq: 102 INVITE
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
Contact: sip:3504@10.0.0.178
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Transmitting (no NAT) to 10.0.0.178:5060:
ACK sip:3504@10.0.0.178 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178;tag=4F70555F-C5692AAA
Contact: sip:4999@10.0.0.1
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


Retransmitting #1 (no NAT) to 10.0.0.178:5060:
CANCEL sip:3504@10.0.0.178 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178;tag=4F70555F-C5692AAA
CSeq: 102 INVITE
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
Contact: sip:3504@10.0.0.178
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Retransmitting #2 (no NAT) to 10.0.0.178:5060:
CANCEL sip:3504@10.0.0.178 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178;tag=4F70555F-C5692AAA
CSeq: 102 INVITE
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
Contact: sip:3504@10.0.0.178
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Retransmitting #3 (no NAT) to 10.0.0.178:5060:
CANCEL sip:3504@10.0.0.178 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178;tag=4F70555F-C5692AAA
CSeq: 102 INVITE
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
Contact: sip:3504@10.0.0.178
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Content-Length: 0

<------------->
— (9 headers 0 lines) —
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178;tag=4F70555F-C5692AAA
CSeq: 102 INVITE
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
Contact: sip:3504@10.0.0.178
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Retransmitting #4 (no NAT) to 10.0.0.178:5060:
CANCEL sip:3504@10.0.0.178 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178;tag=4F70555F-C5692AAA
CSeq: 102 INVITE
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
Contact: sip:3504@10.0.0.178
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Content-Length: 0

<------------->
— (9 headers 0 lines) —
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178;tag=4F70555F-C5692AAA
CSeq: 102 INVITE
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
Contact: sip:3504@10.0.0.178
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Content-Length: 0

<------------->
— (9 headers 0 lines) —
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178;tag=4F70555F-C5692AAA
CSeq: 102 INVITE
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
Contact: sip:3504@10.0.0.178
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Retransmitting #5 (no NAT) to 10.0.0.178:5060:
CANCEL sip:3504@10.0.0.178 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178
CSeq: 102 CANCEL
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Retransmitting #6 (no NAT) to 10.0.0.178:5060:
CANCEL sip:3504@10.0.0.178 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


asterisk*CLI> sip set debug off
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178
CSeq: 102 CANCEL
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Content-Length: 0

<------------->
— (8 headers 0 lines) — off

Calling to the broken extension - this is where I’m having the problem that the extension is not found
asteriskCLI> sip set debug peer 4999
SIP Debugging Enabled for IP: 10.0.0.213:5060
asterisk
CLI> sip set debug peer 3504
SIP Debugging Enabled for IP: 10.0.0.178:5060
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
INVITE sip:4999@10.0.0.1:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.178;branch=z9hG4bK273c44d1E83C10CA
From: “Alfreda Benford” sip:3504@10.0.0.1;tag=9009CFAB-EC311066
To: sip:4999@10.0.0.1;user=phone
CSeq: 1 INVITE
Call-ID: bacc6ff-2fc5f41d-e7c1fa00@10.0.0.178
Contact: sip:3504@10.0.0.178
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 245

v=0
o=- 1400093487 1400093487 IN IP4 10.0.0.178
s=Polycom IP Phone
c=IN IP4 10.0.0.178
t=0 0
a=sendrecv
m=audio 2258 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
<------------->
— (14 headers 11 lines) —
Sending to 10.0.0.178 : 5060 (no NAT)
Using INVITE request as basis request - bacc6ff-2fc5f41d-e7c1fa00@10.0.0.178
asterisk*CLI>
<— Reliably Transmitting (no NAT) to 10.0.0.178:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.0.178;branch=z9hG4bK273c44d1E83C10CA;received=10.0.0.178
From: “Alfreda Benford” sip:3504@10.0.0.1;tag=9009CFAB-EC311066
To: sip:4999@10.0.0.1;user=phone;tag=as37fea988
Call-ID: bacc6ff-2fc5f41d-e7c1fa00@10.0.0.178
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="7eb74102"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘bacc6ff-2fc5f41d-e7c1fa00@10.0.0.178’ in 32000 ms (Method: INVITE)
Found user '3504’
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
ACK sip:4999@10.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.178;branch=z9hG4bK273c44d1E83C10CA
From: “Alfreda Benford” sip:3504@10.0.0.1;tag=9009CFAB-EC311066
To: sip:4999@10.0.0.1;user=phone;tag=as37fea988
CSeq: 1 ACK
Call-ID: bacc6ff-2fc5f41d-e7c1fa00@10.0.0.178
Contact: sip:3504@10.0.0.178
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Max-Forwards: 70
Content-Length: 0

<------------->
— (11 headers 0 lines) —
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
INVITE sip:4999@10.0.0.1:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.178;branch=z9hG4bKcf6397c8386D929
From: “Alfreda Benford” sip:3504@10.0.0.1;tag=9009CFAB-EC311066
To: sip:4999@10.0.0.1;user=phone
CSeq: 2 INVITE
Call-ID: bacc6ff-2fc5f41d-e7c1fa00@10.0.0.178
Contact: sip:3504@10.0.0.178
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username=“3504”, realm=“asterisk”, nonce=“7eb74102”, uri=“sip:4999@10.0.0.1:5060;user=phone”, response=“f6ae3c2a135d402882adda36b41b27f0”, algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 245

v=0
o=- 1400093487 1400093487 IN IP4 10.0.0.178
s=Polycom IP Phone
c=IN IP4 10.0.0.178
t=0 0
a=sendrecv
m=audio 2258 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
<------------->
— (15 headers 11 lines) —
Sending to 10.0.0.178 : 5060 (no NAT)
Using INVITE request as basis request - bacc6ff-2fc5f41d-e7c1fa00@10.0.0.178
Found user '3504’
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 10.0.0.178:2258
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.0.0.178:2258
Looking for 4999 in main (domain 10.0.0.1)
asterisk*CLI>
<— Reliably Transmitting (no NAT) to 10.0.0.178:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.0.178;branch=z9hG4bKcf6397c8386D929;received=10.0.0.178
From: “Alfreda Benford” sip:3504@10.0.0.1;tag=9009CFAB-EC311066
To: sip:4999@10.0.0.1;user=phone;tag=as37fea988
Call-ID: bacc6ff-2fc5f41d-e7c1fa00@10.0.0.178
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------>
[May 14 13:51:32] NOTICE[4875]: chan_sip.c:13922 handle_request_invite: Call from ‘3504’ to extension ‘4999’ rejected because extension not found.
Scheduling destruction of SIP dialog ‘bacc6ff-2fc5f41d-e7c1fa00@10.0.0.178’ in 32000 ms (Method: INVITE)
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
ACK sip:4999@10.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.178;branch=z9hG4bKcf6397c8386D929
From: “Alfreda Benford” sip:3504@10.0.0.1;tag=9009CFAB-EC311066
To: sip:4999@10.0.0.1;user=phone;tag=as37fea988
CSeq: 2 ACK
Call-ID: bacc6ff-2fc5f41d-e7c1fa00@10.0.0.178
Contact: sip:3504@10.0.0.178
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Proxy-Authorization: Digest username=“3504”, realm=“asterisk”, nonce=“7eb74102”, uri=“sip:4999@10.0.0.1:5060;user=phone”, response=“f6ae3c2a135d402882adda36b41b27f0”, algorithm=MD5
Max-Forwards: 70
Content-Length: 0

<------------->
— (12 headers 0 lines) —
asterisk*CLI> sip set debug peer 4999
SIP Debugging Enabled for IP: 10.0.0.213:5060
Reliably Transmitting (no NAT) to 10.0.0.213:5060:
OPTIONS sip:4999@10.0.0.213 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK01a6e809;rport
From: “asterisk” sip:asterisk@10.0.0.1;tag=as662d3c34
To: sip:4999@10.0.0.213
Contact: sip:asterisk@10.0.0.1
Call-ID: 4f53c004403b3bbf7839e5471c563c58@10.0.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 14 May 2014 18:51:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


asterisk*CLI>
<— SIP read from 10.0.0.213:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK01a6e809;rport
From: “asterisk” sip:asterisk@10.0.0.1;tag=as662d3c34
To: sip:4999@10.0.0.213;tag=7D8ABBEE-50B33D1D
CSeq: 102 OPTIONS
Call-ID: 4f53c004403b3bbf7839e5471c563c58@10.0.0.1
Contact: sip:4999@10.0.0.213
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.7.0134
Accept-Language: en
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Supported: 100rel,timer,replaces
Content-Length: 0

<------------->
— (14 headers 0 lines) —

Please, can anyone help with this??? Here is a show peer listing for the non-working extension. I have highlighted the differences between this and my working extensions in the hopes that someone can tell me if one of these would cause the extension to not be found and where to go to change that setting on the peer:

  • Name : 4514
    Secret :
    MD5Secret :
    Context : main
    Subscr.Cont. :
    Language : en
    AMA flags : Unknown
    Transfer mode: open
    CallingPres : Presentation Allowed, Not Screened
    Callgroup :
    Pickupgroup :
    Mailbox : 3514@default
    VM Extension : asterisk
    LastMsgsSent : 0/4
    Call limit : 0
    Dynamic : Yes
    Callerid : “Jeremy Estle” <3514>
    MaxCallBR : 384 kbps
    Expire : 3591 (-1 on working lines)
    Insecure : no
    Nat : RFC3581
    ACL : No
    T38 pt UDPTL : No
    CanReinvite : No
    PromiscRedir : No
    User=Phone : No
    Video Support: No
    Trust RPID : No
    Send RPID : No
    Subscriptions: Yes
    Overlap dial : Yes
    DTMFmode : rfc2833
    LastMsg : 0
    ToHost :
    Addr->IP : External IP Here Port 5060 ("(Unspecified) Port 0" on working lines)
    Defaddr->IP : 0.0.0.0 Port 5060
    Def. Username: 4290bf3ab92e2d17ef828d720fc24bc4 (Blank on Working Lines)
    SIP Options : (none)
    Codecs : 0xe (gsm|ulaw|alaw)
    Codec Order : (ulaw:20,alaw:20,gsm:20)
    Auto-Framing: No
    Status : OK (165 ms) (Unknown on Working Lines)
    Useragent : PolycomSoundPointIP-SPIP_501-UA/3.1.7.0134 (Blank on Working Lines)
    Reg. Contact : sip:4290bf3ab92e2d17ef828d720fc24bc4@External IP Here:5060 (Blank on Working Lines)

Can anyone offer suggestions on this? I’ve received nothing and I really need some help figuring this out since everything seems to be there when you view the dial plan and the config files.

The trace shows 4999, not 3514, being rejected. It looks like you may have domain specific contexts.

I was testing with multiple extensions. The 4999 was doing the same thing as the 4514. I have now resolved the issue on my own. Thank you for your response.