Ok, so I'm still stuck on this one. I'm not sure what exactly I should
be looking for in the output, but here's a snippet that is relevant I think:
---
-- SIP/LW3086-09e6 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Goto("SIP/518-1acd", "s-CONGESTION|1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing NoOp("SIP/518-1acd", "Dial failed due to CONGESTION")
in new stack
-- Executing Macro("SIP/518-1acd", "dialout-trunk|22|3038943818||")
in new stack
-- Executing GotoIf("SIP/518-1acd", "1?3:2") in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("SIP/518-1acd", "user-callerid") in new stack
-- Executing GotoIf("SIP/518-1acd", "0?report") in new stack
-- Executing GotoIf("SIP/518-1acd", "1?start") in new stack
-- Goto (macro-user-callerid,s,4)
-- Executing NoOp("SIP/518-1acd", "REALCALLERIDNUM is 518") in new
stack
-- Executing Set("SIP/518-1acd", "AMPUSER=518") in new stack
-- Executing Set("SIP/518-1acd", "AMPUSERCIDNAME=Mike Staver") in
new stack
-- Executing GotoIf("SIP/518-1acd", "0?report") in new stack
-- Executing Set("SIP/518-1acd", "CALLERID(all)=Mike Staver <518>")
in new stack
-- Executing NoOp("SIP/518-1acd", "Using CallerID "Mike Staver"
<518>") in new stack
-- Executing Macro("SIP/518-1acd", "record-enable|518|OUT") in new
stack
-- Executing GotoIf("SIP/518-1acd", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/518-1acd",
"recordingcheck|20060714-135108|1152906666.9581") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060714-135108|1152906666.9581: Outbound recording
not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/518-1acd", "No recording needed") in new stack
-- Executing Macro("SIP/518-1acd", "outbound-callerid|22") in new stack
-- Executing GotoIf("SIP/518-1acd", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/518-1acd", "REALCALLERIDNUM is 518") in new
stack
-- Executing Set("SIP/518-1acd", "USEROUTCID=Michael Staver
<303-894-3818>") in new stack
-- Executing Set("SIP/518-1acd", "EMERGENCYCID=") in new stack
-- Executing Set("SIP/518-1acd", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("SIP/518-1acd", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,11)
-- Executing GotoIf("SIP/518-1acd", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,13)
-- Executing GotoIf("SIP/518-1acd", "0?report") in new stack
-- Executing Set("SIP/518-1acd", "CALLERID(all)=Michael Staver
<303-894-3818>") in new stack
-- Executing NoOp("SIP/518-1acd", "CallerID set to "Michael Staver"
<3038943818>") in new stack
-- Executing Set("SIP/518-1acd", "GROUP()=OUT_22") in new stack
-- Executing GotoIf("SIP/518-1acd", "0?108") in new stack
-- Executing Set("SIP/518-1acd", "DIAL_NUMBER=3038943818") in new stack
-- Executing Set("SIP/518-1acd", "DIAL_TRUNK=22") in new stack
-- Executing AGI("SIP/518-1acd", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/518-1acd", "OUTNUM=3038943818") in new stack
-- Executing Set("SIP/518-1acd", "custom=SIP/LW0054") in new stack
-- Executing GotoIf("SIP/518-1acd", "0?16") in new stack
-- Executing Dial("SIP/518-1acd", "SIP/LW0054/3038943818|120|r") in
new stack
-- Called LW0054/3038943818
Transmitting (no NAT) to 10.0.0.121:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
10.0.0.121;branch=z9hG4bKabdafff5314CEDCA;received=10.0.0.121
From: "Mike Staver"
<sip:[EMAIL PROTECTED]>;tag=7B8310C8-DE20AB03
To: <sip:[EMAIL PROTECTED];user=phone>;tag=as665b07ac
Call-ID: [EMAIL PROTECTED]
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
---
-- SIP/LW0054-c1d8 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Goto("SIP/518-1acd", "s-CONGESTION|1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing NoOp("SIP/518-1acd", "Dial failed due to CONGESTION")
in new stack
-- Executing Macro("SIP/518-1acd", "outisbusy|") in new stack
-- Executing Playback("SIP/518-1acd", "all-circuits-busy-now") in
new stack
We're at 10.0.0.12 port 16460
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.0.121:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.0.0.121;branch=z9hG4bKabdafff5314CEDCA;received=10.0.0.121
From: "Mike Staver" <sip:[EMAIL PROTECTED]>;tag=7B8310C8-DE20AB03
To: <sip:[EMAIL PROTECTED];user=phone>;tag=as665b07ac
Call-ID: [EMAIL PROTECTED]
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[EMAIL PROTECTED]>
Content-Type: application/sdp
Content-Length: 232
v=0
o=root 3042 3042 IN IP4 10.0.0.12
s=session
c=IN IP4 10.0.0.12
t=0 0
m=audio 16460 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Playing 'all-circuits-busy-now' (language 'en')
asterisk1*CLI>
<-- SIP read from 10.0.0.121:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.0.0.121;branch=z9hG4bKc3197eaeB793628B
From: "Mike Staver" <sip:[EMAIL PROTECTED]>;tag=7B8310C8-DE20AB03
To: <sip:[EMAIL PROTECTED];user=phone>;tag=as665b07ac
CSeq: 2 ACK
Call-ID: [EMAIL PROTECTED]
Contact: <sip:[EMAIL PROTECTED]>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.6.0036
Proxy-Authorization: Digest username="518", realm="asterisk",
nonce="2f91440c", uri="sip:[EMAIL PROTECTED]:5060;user=phone",
response="ae6b67e078bbd47433af49559828c0ca", algorithm=MD5
Max-Forwards: 70
Content-Length: 0
--- (12 headers 0 lines)---
-- Executing Playback("SIP/518-1acd", "pls-try-call-later") in new
stack
-- Playing 'pls-try-call-later' (language 'en')
-- Executing Macro("SIP/518-1acd", "hangupcall") in new stack
-- Executing ResetCDR("SIP/518-1acd", "w") in new stack
-- Executing NoCDR("SIP/518-1acd", "") in new stack
-- Executing Wait("SIP/518-1acd", "5") in new stack
asterisk1*CLI>
Basically, what happens in that I have an outbound route with a bunch of
trunks in it. For whatever reason, let's say I have 5 extensions online
in my office. Then let's say I have only 3 outgoing trunks set up.
Even though nobody is on the phone and I have 3 trunks wide open -
asterisk only allows the first 3 phones to register with the server to
call out. The other 2 get this busy message. How can I fix this?
Ideally, I'd like to have more extensions than outgoing trunks for
obvious reasons.
Jerry Jones wrote:
asterisk -r
set verbose 3
On Jun 28, 2006, at 3:23 PM, Mike Staver wrote:
Yes, I have more than one call per line enabled on the phone itself.
I have a value of 3 entered there, and that should be sufficient I
would think. So, the message I'm getting is coming from Asterisk.
How do I see what the console is saying?
Jerry Jones wrote:
Do you have more than one call per line enabled on the Poly? Is it
the phone or asterisk returning the busy? What does the console say?
On Jun 27, 2006, at 5:29 PM, Mike Staver wrote:
I have one extension setup for each Polycom 501 I have, and when I
try to call out on a conference call, I get "all circuits busy" for
the second call. I have one sip trunk set up for each DID that I
have through our VoIP provider. Each trunk is capable of having one
call placed on it at one time. So, I'm thinking I need a way to
tell Asterisk to have the second call go out on one of the other
empty trunks at the time if one exists, which more than likely, it
will. Is this possible?
-- -Mike Staver
[EMAIL PROTECTED]
[EMAIL PROTECTED]
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--
-Mike Staver
[EMAIL PROTECTED]
[EMAIL PROTECTED]
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--
-Mike Staver
[EMAIL PROTECTED]
[EMAIL PROTECTED]
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