Logging into Queues busted

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:*[email protected]<>:5060 SIP/2.0
Via: SIP/2.0/UDP <>;branch=z9hG4bKae1d35712
Max-Forwards: 70
Content-Length: 587
To: *500 <sip:*[email protected]<>:5060>
From: Tech Support <sip:[email protected]<>:5060>;tag=443e0279c7efac8
Call-ID: [email protected]<>
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:[email protected]<>>
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 - [email protected]<>
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:[email protected]<>:5060>;tag=443e0279c7efac8
To: *500 <sip:*[email protected]<>:5060>;tag=as0856e78d
Call-ID: [email protected]<>
CSeq: 1000915564 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:*[email protected]<>>
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="1a40c705"
Content-Length: 0


Scheduling destruction of call ‘[email protected]<>’ in 15000 ms
Found user '201’
asterisk1*CLI>
<-- SIP read from <>:5060:
ACK sip:*[email protected]<>:5060 SIP/2.0
Via: SIP/2.0/UDP <>;branch=z9hG4bKae1d35712
Max-Forwards: 70
Content-Length: 0
To: *500 <sip:*[email protected]<>:5060>;tag=as0856e78d
From: Tech Support <sip:[email protected]<>:5060>;tag=443e0279c7efac8
Call-ID: [email protected]<>
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:*[email protected]<>:5060 SIP/2.0
Via: SIP/2.0/UDP <>;branch=z9hG4bKb4ef96043
Max-Forwards: 70
Content-Length: 587
To: *500 <sip:*[email protected]<>:5060>
From: Tech Support <sip:[email protected]<>:5060>;tag=443e0279c7efac8
Call-ID: [email protected]<>
CSeq: 1000915565 INVITE
Supported: timer
Allow-Events: talk,hold,conference
Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
Contact: Tech Support <sip:[email protected]<>>
Content-Type: application/sdp
Supported: replaces
Proxy-Authorization:Digest response=“dc6c4d025b078d17e113218e69d411e6”,username=“201”,realm=“asterisk”,nonce=“1a40c705”,algorithm=MD5,uri="sip:*[email protected]<>: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 - [email protected]<>
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:[email protected]<>:5060>;tag=443e0279c7efac8
To: *500 <sip:*[email protected]<>:5060>;tag=as0856e78d
Call-ID: [email protected]<>
CSeq: 1000915565 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:*[email protected]<>>
Content-Length: 0


asterisk1*CLI>
<-- SIP read from <>:5060:
ACK sip:*[email protected]<>:5060 SIP/2.0
Via: SIP/2.0/UDP <>;branch=z9hG4bKb4ef96043
Max-Forwards: 70
Content-Length: 0
To: *500 <sip:*[email protected]<>:5060>;tag=as0856e78d
From: Tech Support <sip:[email protected]<>:5060>;tag=443e0279c7efac8
Call-ID: [email protected]<>
CSeq: 1000915565 ACK
Proxy-Authorization:Digest response=“479923970b8187d20925f3125e33fafe”,username=“201”,realm=“asterisk”,nonce=“1a40c705”,algorithm=MD5,uri="sip:*[email protected]<>: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?

[quote=“Peri”]It’s due to ip dialing, which is enabled by default on the Aastra phones, with no toggle to turn it off.

I’ve created some work around extensions.[/quote]

I get the same problem on some safecom phones.
What was this workaround you did?

Thanks

I created some quick login and logout extensions in extensions_custom.conf

[ext-queues-custom]
exten => 5211,1,Macro(agent-add,521,)
exten => 5212,1,Macro(agent-del,521,521)

The above example is for a queue numbered 521.

Here is what we use. 999* to login, 999** to logout, 2 queues at the same time. just need to repeat the exten => 999*,n,Macro(easy-agent-add,100,) line to add more queues

[code:1]

[ext-queues-custom]
; Support queue logger for isp customer 45 french and english queues

; login to queue 100 (FR) and 101 (EN)
exten => 999*,1,Wait(1)
exten => 999*,n,Macro(easy-agent-add,100,) ;
exten => 999*,n,Macro(easy-agent-add,101,) ;
exten => 999*,n,Playback(agentloginfr)
exten => 999*,n,Hangup()

; logout from queue 100 (FR) and 101 (EN)
exten => 999**,1,Wait(1)
exten => 999**,n,Macro(easy-agent-del,100,) ;
exten => 999**,n,Macro(easy-agent-del,101,) ;
exten => 999**,n,Playback(agentlogoutfr)
exten => 999**,n,Hangup()

[macro-easy-agent-del]
exten => s,1,NoOp
exten => s,2,SetVar(CALLBACKNUM=${CALLERIDNUM})
exten => s,3,GotoIf($[foo${CALLBACKNUM} = foo]?2))
exten => s,4,RemoveQueueMember(${ARG1}|Local/${CALLBACKNUM}@from-internal)
;Allows agents to log out without typing their id
;Arg1 = queue number

[macro-easy-agent-add]
exten => s,1,NoOp
exten => s,2,SetVar(CALLBACKNUM=${CALLERIDNUM})
exten => s,3,GotoIf($[foo${CALLBACKNUM} = foo]?2)) ; if still no number, start over
exten => s,4,AddQueueMember(${ARG1}|Local/${CALLBACKNUM}@from-internal) ; using chan_local

[/code:1]

Steve
mousepad99 at gmail.com

It’s due to ip dialing, which is enabled by default on the Aastra phones, with no toggle to turn it off.

I’ve created some work around extensions.

Aha
I have solved my own problem.

The Aastra 480i phone i’m using sending the * as a . is exactly the problem.

Now to find out why my Aastra’s are sending .'s

I think you must disable the phone features to use the Asterisk features
I could be wrong I have never messed with that brand

it would be 500* to login and 500** to logout

with the * after the extension:

[code:1]<-- SIP read from <>:5060:
INVITE sip:500. SIP/2.0
Via: SIP/2.0/UDP <>;branch=z9hG4bKfe961dce0
Max-Forwards: 70
Content-Length: 589
To: 500. sip:500.
From: Exten 200 <sip:[email protected]<>:5060>;tag=45c7b9f87dd0298
Call-ID: [email protected]<>
CSeq: 342227126 INVITE
Route: <sip:<>:5060;lr>
Supported: timer
Allow-Events: talk,hold,conference
Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
Content-Type: application/sdp
Contact: Exten 200 <sip:[email protected]<>>
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 1667910624 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 - [email protected]<>
Sending to <> : 5060 (non-NAT)
Reliably Transmitting (NAT) to <>:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP <>;branch=z9hG4bKfe961dce0;received=<>
From: Exten 200 <sip:[email protected]<>:5060>;tag=45c7b9f87dd0298
To: 500. sip:500.;tag=as69be0778
Call-ID: [email protected]<>
CSeq: 342227126 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]<>>
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="17a89b21"
Content-Length: 0


Scheduling destruction of call ‘[email protected]<>’ in 15000 ms
Found user '200’
asterisk1*CLI>
<-- SIP read from <>:5060:
INVITE sip:500. SIP/2.0
Via: SIP/2.0/UDP <>;branch=z9hG4bKfe961dce0
Max-Forwards: 70
Content-Length: 589
To: 500. sip:500.
From: Exten 200 <sip:[email protected]<>:5060>;tag=45c7b9f87dd0298
Call-ID: [email protected]<>
CSeq: 342227126 INVITE
Route: <sip:<>:5060;lr>
Supported: timer
Allow-Events: talk,hold,conference
Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
Content-Type: application/sdp
Contact: Exten 200 <sip:[email protected]<>>
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 1667910624 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)—
Ignoring this INVITE request
Retransmitting #1 (NAT) to <>:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP <>;branch=z9hG4bKfe961dce0;received=<>
From: Exten 200 <sip:[email protected]<>:5060>;tag=45c7b9f87dd0298
To: 500. sip:500.;tag=as69be0778
Call-ID: [email protected]<>
CSeq: 342227126 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]<>>
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="17a89b21"
Content-Length: 0


asterisk1*CLI>
<-- SIP read from <>:5060:
INVITE sip:500. SIP/2.0
Via: SIP/2.0/UDP <>;branch=z9hG4bKfe961dce0
Max-Forwards: 70
Content-Length: 589
To: 500. sip:500.
From: Exten 200 <sip:[email protected]<>:5060>;tag=45c7b9f87dd0298
Call-ID: [email protected]<>
CSeq: 342227126 INVITE
Route: <sip:<>:5060;lr>
Supported: timer
Allow-Events: talk,hold,conference
Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
Content-Type: application/sdp
Contact: Exten 200 <sip:[email protected]<>>
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 1667910624 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)—
Ignoring this INVITE request
Retransmitting #2 (NAT) to <>:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP <>;branch=z9hG4bKfe961dce0;received=<>
From: Exten 200 <sip:[email protected]<>:5060>;tag=45c7b9f87dd0298
To: 500. sip:500.;tag=as69be0778
Call-ID: [email protected]<>
CSeq: 342227126 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]<>>
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="17a89b21"
Content-Length: 0


asterisk1*CLI>
<-- SIP read from 24.137.193.55:5066:
NOTIFY sip:<> SIP/2.0
Via: SIP/2.0/UDP 24.137.193.55:5066;branch=z9hG4bK-2a3ccc3f
From: Technical Support <sip:[email protected]<>>;tag=ffe16fa68407d66o1
To: <sip:<>>
Call-ID: [email protected]
CSeq: 510 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Linksys/RT31P2-3.1.3(LI)
Content-Length: 0


Destroying call '[email protected]
asterisk1*CLI>
<-- SIP read from <>:5060:
INVITE sip:500. SIP/2.0
Via: SIP/2.0/UDP <>;branch=z9hG4bKfe961dce0
Max-Forwards: 70
Content-Length: 589
To: 500. sip:500.
From: Exten 200 <sip:[email protected]<>:5060>;tag=45c7b9f87dd0298
Call-ID: [email protected]<>
CSeq: 342227126 INVITE
Route: <sip:<>:5060;lr>
Supported: timer
Allow-Events: talk,hold,conference
Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
Content-Type: application/sdp
Contact: Exten 200 <sip:[email protected]<>>
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 1667910624 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)—
Ignoring this INVITE request
Retransmitting #3 (NAT) to <>:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP <>;branch=z9hG4bKfe961dce0;received=<>
From: Exten 200 <sip:[email protected]<>:5060>;tag=45c7b9f87dd0298
To: 500. sip:500.;tag=as69be0778
Call-ID: [email protected]<>
CSeq: 342227126 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]<>>
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="17a89b21"
Content-Length: 0
[/code:1]

As you can see the * becomes a . which may well be intended, however the call still fails and the phone disconnects. Again, nothing useful in /var/log/asterisk/full

FYI

for linksys phones the built in dialplan must be modified to accept *

Alex

Linksys dialplan example (xxx*.|xxx**.). The DND setting have to be changed as well. Go to the IP of the phone using a broswer.

Go to the “Regional” tab and find the “Vertical Service Activation Codes” section.
DND Act Code: *78
DND Deact Code: *79