[Asterisk-Users] Asterisk 1.2.7.1 with Polycom 501 on SIP ->
Conf Calling
Mike Staver
staver at fimble.com
Fri Jul 14 12:58:22 MST 2006
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:518 at token.globaltaxnetwork.com>;tag=7B8310C8-DE20AB03
To: <sip:83038943818 at token.globaltaxnetwork.com;user=phone>;tag=as665b07ac
Call-ID: 6abbb3a4-55570366-b333a8b1 at 10.0.0.121
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:83038943818 at 10.0.0.12>
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:518 at 10.0.0.12>;tag=7B8310C8-DE20AB03
To: <sip:83038943818 at 10.0.0.12;user=phone>;tag=as665b07ac
Call-ID: 6abbb3a4-55570366-b333a8b1 at 10.0.0.121
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:83038943818 at 10.0.0.12>
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:83038943818 at 10.0.0.12 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.121;branch=z9hG4bKc3197eaeB793628B
From: "Mike Staver" <sip:518 at 10.0.0.12>;tag=7B8310C8-DE20AB03
To: <sip:83038943818 at 10.0.0.12;user=phone>;tag=as665b07ac
CSeq: 2 ACK
Call-ID: 6abbb3a4-55570366-b333a8b1 at 10.0.0.121
Contact: <sip:518 at 10.0.0.121>
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:83038943818 at 10.0.0.12:5060;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
>>>> staver at fimble.com
>>>> mstaver at globaltaxnetwork.com
>>>> _______________________________________________
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>>
>> --
>> -Mike Staver
>> staver at fimble.com
>> mstaver at globaltaxnetwork.com
>> _______________________________________________
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> Asterisk-Users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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--
-Mike Staver
staver at fimble.com
mstaver at globaltaxnetwork.com
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