[Asterisk-Users] Asterisk 1.2.7.1 with Polycom 501 on SIP -> Conf
Calling
Lacy Moore - Aspendora
aspendora at gmail.com
Fri Jul 14 18:44:42 MST 2006
>From your first message, it sounds like you're doing some sort of one to one
mapping. But, from this message, it appears you are using FreePBX. You may
have to post your FreePBX configuration on the FreePBX list or forum.
I'm sure it is something relatively simple, but in my experience FreePBX has
only complicated things for me. You could try posting the relevant sip.conf,
and whatever extensions file that FreePBX writes its database output to
(can't remember offhand, and honestly don't want to anymore).
On 7/14/06, Mike Staver <staver at fimble.com> wrote:
>
> 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
> >>>> _______________________________________________
> >>>> --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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> >>> http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >> --
> >> -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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> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
>
> -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
>
--
Lacy Moore
Aspendora, Inc.
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