I posted another issue here which apparently shouldn’t be an issue.
Apparently I should already be able to log into a queue using *500 on my phone. 500 being the queue number. This isn’t working for me as my call fails.
As i stated in the other thread absolutely everything else appears to be working perfectly except logging into queues.
Here’s the system info:
Now the obligatory system information:
CentOS 2.6.9-34.0.2
Asterisk 1.2.12.1
FreePBX 2.2.0beta2
Here’s a what comes up in /var/log/asterisk/full when dialing *500
[code:1]Nov 15 18:19:42 DEBUG[3026] chan_sip.c: Setting NAT on RTP to 524288
Nov 15 18:19:42 DEBUG[3026] chan_sip.c: Stopping retransmission on ‘[email protected]’ of Response 2121865887: Match Found
Nov 15 18:19:42 DEBUG[3026] chan_sip.c: Setting NAT on RTP to 524288
Nov 15 18:19:42 DEBUG[3026] chan_sip.c: Checking SIP call limits for device 201
Nov 15 18:19:43 DEBUG[3026] chan_sip.c: Stopping retransmission on ‘[email protected]’ of Response 2121865888: Match Found[/code:1]
Nothing comes up in the Asterisk CLI at 6
and in the Asterisk CLI with sip debug on (ip’s have been replaced):
[code:1]
<-- SIP read from <>:5060:
INVITE sip:*500@<>:5060 SIP/2.0
Via: SIP/2.0/UDP <>;branch=z9hG4bKae1d35712
Max-Forwards: 70
Content-Length: 587
To: *500 <sip:*500@<>:5060>
From: Tech Support <sip:201@<>:5060>;tag=443e0279c7efac8
Call-ID: 605b8689cdcfdb63a2faf96857bd1cc8@<>
CSeq: 1000915564 INVITE
Supported: timer
Allow-Events: talk,hold,conference
Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
Content-Type: application/sdp
Contact: Tech Support <sip:201@<>>
Supported: replaces
User-Agent: Aastra 480i/1.4.0.1048 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26
v=0
o=MxSIP 0 2034638098 IN IP4 <>
s=SIP Call
c=IN IP4 <>
t=0 0
m=audio 3000 RTP/AVP 0 18 96 102 107 104 105 106 4 97 98 2 99 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 BV16/8000
a=rtpmap:102 BV32/16000
a=rtpmap:107 L16/16000
a=rtpmap:104 PCMU/16000
a=rtpmap:105 PCMA/16000
a=rtpmap:106 L16/8000
a=rtpmap:4 G723/8000
a=rtpmap:97 G726-16/8000
a=rtpmap:98 G726-24/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:99 G726-40/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=silenceSupp:on - - - -
— (15 headers 24 lines)—
Using INVITE request as basis request - 605b8689cdcfdb63a2faf96857bd1cc8@<>
Sending to <> : 5060 (non-NAT)
Reliably Transmitting (NAT) to <>:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP <>;branch=z9hG4bKae1d35712;received=<>
From: Tech Support <sip:201@<>:5060>;tag=443e0279c7efac8
To: *500 <sip:*500@<>:5060>;tag=as0856e78d
Call-ID: 605b8689cdcfdb63a2faf96857bd1cc8@<>
CSeq: 1000915564 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:*500@<>>
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="1a40c705"
Content-Length: 0
Scheduling destruction of call ‘605b8689cdcfdb63a2faf96857bd1cc8@<>’ in 15000 ms
Found user '201’
asterisk1*CLI>
<-- SIP read from <>:5060:
ACK sip:*500@<>:5060 SIP/2.0
Via: SIP/2.0/UDP <>;branch=z9hG4bKae1d35712
Max-Forwards: 70
Content-Length: 0
To: *500 <sip:*500@<>:5060>;tag=as0856e78d
From: Tech Support <sip:201@<>:5060>;tag=443e0279c7efac8
Call-ID: 605b8689cdcfdb63a2faf96857bd1cc8@<>
CSeq: 1000915564 ACK
User-Agent: Aastra 480i/1.4.0.1048 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26
— (9 headers 0 lines)—
asterisk1*CLI>
<-- SIP read from <>:5060:
INVITE sip:*500@<>:5060 SIP/2.0
Via: SIP/2.0/UDP <>;branch=z9hG4bKb4ef96043
Max-Forwards: 70
Content-Length: 587
To: *500 <sip:*500@<>:5060>
From: Tech Support <sip:201@<>:5060>;tag=443e0279c7efac8
Call-ID: 605b8689cdcfdb63a2faf96857bd1cc8@<>
CSeq: 1000915565 INVITE
Supported: timer
Allow-Events: talk,hold,conference
Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
Contact: Tech Support <sip:201@<>>
Content-Type: application/sdp
Supported: replaces
Proxy-Authorization:Digest response=“dc6c4d025b078d17e113218e69d411e6”,username=“201”,realm=“asterisk”,nonce=“1a40c705”,algorithm=MD5,uri="sip:*500@<>:5060"
User-Agent: Aastra 480i/1.4.0.1048 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26
v=0
o=MxSIP 0 2034638098 IN IP4 <>
s=SIP Call
c=IN IP4 <>
t=0 0
m=audio 3000 RTP/AVP 0 18 96 102 107 104 105 106 4 97 98 2 99 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 BV16/8000
a=rtpmap:102 BV32/16000
a=rtpmap:107 L16/16000
a=rtpmap:104 PCMU/16000
a=rtpmap:105 PCMA/16000
a=rtpmap:106 L16/8000
a=rtpmap:4 G723/8000
a=rtpmap:97 G726-16/8000
a=rtpmap:98 G726-24/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:99 G726-40/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=silenceSupp:on - - - -
— (16 headers 24 lines)—
Using INVITE request as basis request - 605b8689cdcfdb63a2faf96857bd1cc8@<>
Sending to <> : 5060 (NAT)
Found user '201’
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 102
Found RTP audio format 107
Found RTP audio format 104
Found RTP audio format 105
Found RTP audio format 106
Found RTP audio format 4
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 2
Found RTP audio format 99
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port <>:3000
Found description format PCMU
Found description format G729
Found description format BV16
Found description format BV32
Found description format L16
Found description format PCMU
Found description format PCMA
Found description format L16
Found description format G723
Found description format G726-16
Found description format G726-24
Found description format G726-32
Found description format G726-40
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x55d (g723|ulaw|alaw|g726|slin|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for *500 in from-internal (domain <>)
Reliably Transmitting (NAT) to <>:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP <>;branch=z9hG4bKb4ef96043;received=<>
From: Tech Support <sip:201@<>:5060>;tag=443e0279c7efac8
To: *500 <sip:*500@<>:5060>;tag=as0856e78d
Call-ID: 605b8689cdcfdb63a2faf96857bd1cc8@<>
CSeq: 1000915565 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:*500@<>>
Content-Length: 0
asterisk1*CLI>
<-- SIP read from <>:5060:
ACK sip:*500@<>:5060 SIP/2.0
Via: SIP/2.0/UDP <>;branch=z9hG4bKb4ef96043
Max-Forwards: 70
Content-Length: 0
To: *500 <sip:*500@<>:5060>;tag=as0856e78d
From: Tech Support <sip:201@<>:5060>;tag=443e0279c7efac8
Call-ID: 605b8689cdcfdb63a2faf96857bd1cc8@<>
CSeq: 1000915565 ACK
Proxy-Authorization:Digest response=“479923970b8187d20925f3125e33fafe”,username=“201”,realm=“asterisk”,nonce=“1a40c705”,algorithm=MD5,uri="sip:*500@<>:5060"
User-Agent: Aastra 480i/1.4.0.1048 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26
[/code:1]
What more can i provide?