[asterisk-users] Re: some transfers dropped.
Steven
asterisk at tescogroup.com
Wed Oct 25 04:57:40 MST 2006
I am working around this by having the front desk use the "##" transfer.
I am dealing with tech support on the SIP device. (a 24 port SIP to digital handset converter)
I am not sure which is at fault, asterisk or the SIP device.
--
--
Steven
http://www.glimasoutheast.org
"Steven" <asterisk at tescogroup.com> wrote in message news:ehh5qm$3ds$1 at sea.gmane.org...
> Could there be something going on in asterisk to make the first request fail, so that the SIP device cancels and retries the
> transfer(refer)?
>
> Could it be manager overuse?
>
> --
> --
> Steven
>
> http://www.glimasoutheast.org
>
>
>
> "BerkHolz, Steven" <StevenBerkHolz at TESCOGroup.com> wrote in message
> news:03C20CC487B2EF488E22EC586871ABBE02EBD0A3 at tg32.tescogroup.com...
> We are having an issue with transferred calls being dropped.
>
> Looking at the asterisk 1.2.10 logs, it appears that when it is dropped,
> the SIP unit send a CANCEL message to the server.
> On successful transfers this is not seen.
>
> The errors logged in the SIP Unit error log, I believe are from the
> second attempt to transfer the call, after it has actually been
> disconnected.
>
> Nothing is deferent in the logs above the CANCEL request for successful
> or failed transfers.
> So, I am not sure why the CANCEL is being sent.
>
> I can not discern what may be different when it fails.
>
>
>
>
> Thank You,
>
> Steven BerkHolz
> Board member of
> www.glimasoutheast.org
>
>
>
>
>
> ref: from SIP Phone (I think these are the second invite after it is
> hung up)
>
> 2006-OCT-20 17:49:52 GMT +++ Current Timestamp +++
> 2006-OCT-20 17:19:47 GMT NLPA ERROR: sipFailureResponseToRefer: received
> 603 response to REFER
> 2006-OCT-20 15:56:37 GMT NLPA ERROR: sipFailureResponseToRefer: received
> 603 response to REFER
> 2006-OCT-20 15:50:00 GMT NLPA ERROR: sipFailureResponseToRefer: received
> 603 response to REFER
> 2006-OCT-20 15:45:38 GMT NLPA ERROR: sipFailureResponseToRefer: received
> 603 response to REFER
> 2006-OCT-20 15:11:28 GMT NLPA ERROR: sipFailureResponseToRefer: received
> 603 response to REFER
> 2006-OCT-20 15:10:58 GMT NLPA ERROR: sipFailureResponseToRefer: received
> 603 response to REFER
> 2006-OCT-20 14:59:26 GMT NLPA ERROR: sipFailureResponseToRefer: received
> 603 response to REFER
> 2006-OCT-20 12:45:30 GMT NLPA ERROR: sipFailureResponseToRefer: received
> 603 response to REFER
> 2006-OCT-19 19:53:25 GMT NLPA ERROR: sipFailureResponseToRefer: received
> 603 response to REFER
> 2006-OCT-19 18:40:52 GMT NLPA ERROR: sipFailureResponseToRefer: received
> 603 response to REFER
> 2006-OCT-19 18:03:45 GMT NLPA ERROR: sipFailureResponseToRefer: received
> 603 response to REFER
> 2006-OCT-19 17:55:55 GMT NLPA ERROR: sipFailureResponseToRefer: received
> 603 response to REFER
> 2006-OCT-19 15:09:13 GMT NLPA ERROR: sipFailureResponseToRefer: received
> 603 response to REFER
> 2006-OCT-19 15:04:33 GMT NLPA ERROR: sipFailureResponseToRefer: received
> 603 response to REFER
> 2006-OCT-19 14:52:12 GMT NLPA ERROR: sipFailureResponseToRefer: received
> 603 response to REFER
> 2006-OCT-19 14:34:35 GMT NLPA ERROR: sipFailureResponseToRefer: received
> 603 response to REFER
> 2006-OCT-19 14:20:17 GMT NLPA ERROR: sipFailureResponseToRefer: received
> 603 response to REFER
> 2006-OCT-19 13:45:33 GMT NLPA ERROR: sipFailureResponseToRefer: received
> 603 response to REFER
>
>
>
>
>
> ref. from asterisk 1.2.10 logs:
>
> Oct 20 13:19:45 VERBOSE[10652] logger.c: -- Requested transfer
> capability: 0x00 - SPEECH
> Oct 20 13:19:45 DEBUG[8159] channel.c: Avoiding initial deadlock for
> 'Zap/25-1'
> Oct 20 13:19:45 VERBOSE[10652] logger.c: -- Called g2/5155
> Oct 20 13:19:45 VERBOSE[10652] logger.c: Transmitting (no NAT) to
> 172.16.8.200:5065:
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP
> 172.16.8.200:5065;branch=z9hG4bKline0-2425957956;received=172.16.8.200
> From: "From Desk"<sip:5100 at 172.16.200.5>;tag=2425948795
> To: <sip:5155 at 172.16.200.5>;tag=as279eb184
> Call-ID: 2425954456-c4756-5065 at 172.16.8.200
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:5155 at 172.16.200.5>
> Content-Length: 0
>
>
> ---
> Oct 20 13:19:45 DEBUG[10658] app_queue.c: Device 'Zap/25' changed to
> state '2' (In use) but we don't care because they're not a member of any
> queue.
> Oct 20 13:19:45 DEBUG[8159] devicestate.c: Changing state for Zap/25 -
> state 2 (In use)
> Oct 20 13:19:45 DEBUG[10659] app_queue.c: Device 'Zap/25' changed to
> state '2' (In use) but we don't care because they're not a member of any
> queue.
> Oct 20 13:19:45 DEBUG[8167] chan_zap.c: Enabled echo cancellation on
> channel 25
> Oct 20 13:19:45 VERBOSE[10652] logger.c: -- Zap/25-1 is ringing
> Oct 20 13:19:45 DEBUG[8159] devicestate.c: Changing state for Zap/25 -
> state 6 (Ringing)
> Oct 20 13:19:45 DEBUG[10660] app_queue.c: Device 'Zap/25' changed to
> state '6' (Ringing) but we don't care because they're not a member of
> any queue.
> Oct 20 13:19:45 DEBUG[8171] chan_sip.c: Header 0: (0)
> Oct 20 13:19:46 VERBOSE[8171] logger.c:
> <-- SIP read from 172.16.8.200:5065:
> CANCEL sip:5155 at 172.16.200.5 SIP/2.0
> Via: SIP/2.0/UDP 172.16.8.200:5065;branch=z9hG4bKline0-2425957956
> To: <sip:5155 at 172.16.200.5>
> From: "From Desk"<sip:5100 at 172.16.200.5>;tag=2425948795
> Call-Id: 2425954456-c4756-5065 at 172.16.8.200
> Max-Forwards: 70
> CSeq: 2 CANCEL
> Content-Length: 0
>
>
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 0: CANCEL
> sip:5155 at 172.16.200.5 SIP/2.0 (36)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 1: Via: SIP/2.0/UDP
> 172.16.8.200:5065;branch=z9hG4bKline0-2425957956 (65)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 2: To:
> <sip:5155 at 172.16.200.5> (27)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 3: From: "From
> Desk"<sip:5100 at 172.16.200.5>;tag=2425948795 (55)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 4: Call-Id:
> 2425954456-c4756-5065 at 172.16.8.200 (43)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 5: Max-Forwards: 70 (16)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 6: CSeq: 2 CANCEL (14)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 7: Content-Length: 0 (17)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 8: (0)
> Oct 20 13:19:46 VERBOSE[8171] logger.c: --- (8 headers 0 lines)Oct 20
> 13:19:46 VERBOSE[8171] logger.c: --- (8 headers 0 lines)---
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: **** Received CANCEL (14) -
> Command in SIP CANCEL
> Oct 20 13:19:46 VERBOSE[8171] logger.c: Sending to 172.16.8.200 : 5065
> (non-NAT)
> Oct 20 13:19:46 VERBOSE[8171] logger.c: Reliably Transmitting (no NAT)
> to 172.16.8.200:5065:
> SIP/2.0 487 Request Terminated
> Via: SIP/2.0/UDP
> 172.16.8.200:5065;branch=z9hG4bKline0-2425957956;received=172.16.8.200
> From: "From Desk"<sip:5100 at 172.16.200.5>;tag=2425948795
> To: <sip:5155 at 172.16.200.5>;tag=as279eb184
> Call-ID: 2425954456-c4756-5065 at 172.16.8.200
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:5155 at 172.16.200.5>
> Content-Length: 0
>
>
> ---
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: *** SIP TIMER: Initalizing
> retransmit timer on packet: Id #51636
> Oct 20 13:19:46 VERBOSE[8171] logger.c: Transmitting (no NAT) to
> 172.16.8.200:5065:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 172.16.8.200:5065;branch=z9hG4bKline0-2425957956;received=172.16.8.200
> From: "From Desk"<sip:5100 at 172.16.200.5>;tag=2425948795
> To: <sip:5155 at 172.16.200.5>;tag=as279eb184
> Call-ID: 2425954456-c4756-5065 at 172.16.8.200
> CSeq: 2 CANCEL
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:5155 at 172.16.200.5>
> Content-Length: 0
>
>
> ---
> Oct 20 13:19:46 DEBUG[10652] channel.c: Hanging up channel 'Zap/25-1'
> Oct 20 13:19:46 DEBUG[10652] chan_zap.c: zt_hangup(Zap/25-1)
> Oct 20 13:19:46 DEBUG[10652] chan_zap.c: Set option AUDIO MODE, value:
> ON(1) on Zap/25-1
> Oct 20 13:19:46 DEBUG[10652] chan_zap.c: Hangup: channel: 25 index = 0,
> normal = 38, callwait = -1, thirdcall = -1
> Oct 20 13:19:46 DEBUG[10652] chan_zap.c: Not yet hungup... Calling
> hangup once with icause, and clearing call
> Oct 20 13:19:46 DEBUG[10652] chan_zap.c: disabled echo cancellation on
> channel 25
> Oct 20 13:19:46 DEBUG[10652] chan_zap.c: Set option TDD MODE, value:
> OFF(0) on Zap/25-1
> Oct 20 13:19:46 DEBUG[10652] chan_zap.c: Updated conferencing on 25,
> with 0 conference users
> Oct 20 13:19:46 DEBUG[10652] chan_zap.c: Set option AUDIO MODE, value:
> OFF(0) on Zap/25-1
> Oct 20 13:19:46 DEBUG[10652] chan_zap.c: disabled echo cancellation on
> channel 25
> Oct 20 13:19:46 VERBOSE[10652] logger.c: -- Hungup 'Zap/25-1'
> Oct 20 13:19:46 DEBUG[10652] app_dial.c: Exiting with DIALSTATUS=CANCEL.
> Oct 20 13:19:46 DEBUG[10652] app_macro.c: Spawn extension
> (macro-dial,s,12) exited non-zero on 'SIP/5100-9f70a9b0' in macro 'dial'
> Oct 20 13:19:46 DEBUG[10652] app_macro.c: Spawn extension
> (macro-dial,s,12) exited non-zero on 'SIP/5100-9f70a9b0' in macro
> 'exten-vm'
> Oct 20 13:19:46 DEBUG[10652] pbx.c: Spawn extension (macro-dial,s,12)
> exited non-zero on 'SIP/5100-9f70a9b0'
> Oct 20 13:19:46 DEBUG[8159] devicestate.c: Changing state for Zap/25 -
> state 0 (Unknown)
> Oct 20 13:19:46 DEBUG[10652] cdr_addon_mysql.c: cdr_mysql: inserting a
> CDR record.
> Oct 20 13:19:46 DEBUG[10652] cdr_addon_mysql.c: cdr_mysql: SQL command
> as follows: INSERT INTO cdr
> (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,dura
> tion,billsec,disposition,amaflags,accountcode,uniqueid) VALUES
> ('2006-10-20 13:19:44','5100','5100','5155','from-internal',
> 'SIP/5100-9f70a9b0','Zap/25-1','Dial','zap/g2/5155|15|r',2,0,'NO
> ANSWER',3,'','1161364784.6637')
> Oct 20 13:19:46 DEBUG[10652] pbx.c: Function result is '5100'
> Oct 20 13:19:46 DEBUG[10652] pbx.c: Function result is '5100'
> Oct 20 13:19:46 DEBUG[10652] pbx.c: Function result is '5155'
> Oct 20 13:19:46 DEBUG[10652] pbx.c: Function result is 'from-internal'
> Oct 20 13:19:46 DEBUG[10652] pbx.c: Function result is
> 'SIP/5100-9f70a9b0'
> Oct 20 13:19:46 DEBUG[10652] pbx.c: Function result is 'Zap/25-1'
> Oct 20 13:19:46 DEBUG[10652] pbx.c: Function result is 'Dial'
> Oct 20 13:19:46 DEBUG[10652] pbx.c: Function result is
> 'zap/g2/5155|15|r'
> Oct 20 13:19:46 DEBUG[10652] pbx.c: Function result is '2006-10-20
> 13:19:44'
> Oct 20 13:19:46 DEBUG[10652] pbx.c: Function result is '(null)'
> Oct 20 13:19:46 DEBUG[10652] pbx.c: Function result is '2006-10-20
> 13:19:46'
> Oct 20 13:19:46 DEBUG[10652] pbx.c: Function result is '2'
> Oct 20 13:19:46 DEBUG[10652] pbx.c: Function result is '0'
> Oct 20 13:19:46 DEBUG[10652] pbx.c: Function result is 'NO ANSWER'
> Oct 20 13:19:46 DEBUG[10652] pbx.c: Function result is 'DOCUMENTATION'
> Oct 20 13:19:46 DEBUG[10652] pbx.c: Function result is '(null)'
> Oct 20 13:19:46 DEBUG[10652] pbx.c: Function result is '1161364784.6637'
> Oct 20 13:19:46 DEBUG[10652] pbx.c: Function result is '(null)'
> Oct 20 13:19:46 DEBUG[10652] channel.c: Hanging up channel
> 'SIP/5100-9f70a9b0'
> Oct 20 13:19:46 DEBUG[10652] chan_sip.c: Hangup call SIP/5100-9f70a9b0,
> SIP callid 2425954456-c4756-5065 at 172.16.8.200)
> Oct 20 13:19:46 DEBUG[10652] chan_sip.c: update_call_counter(5100) -
> decrement call limit counter
> Oct 20 13:19:46 DEBUG[10652] chan_sip.c: Updating call counter for
> incoming call
> Oct 20 13:19:46 DEBUG[8159] chan_sip.c: Checking device state for peer
> 5100
> Oct 20 13:19:46 DEBUG[8159] devicestate.c: Changing state for SIP/5100 -
> state 2 (In use)
> Oct 20 13:19:46 DEBUG[8159] chan_sip.c: Checking device state for peer
> 5100
> Oct 20 13:19:46 DEBUG[10661] app_queue.c: Device 'Zap/25' changed to
> state '0' (Unknown) but we don't care because they're not a member of
> any queue.
> Oct 20 13:19:46 DEBUG[10662] app_queue.c: Device 'SIP/5100' changed to
> state '2' (In use) but we don't care because they're not a member of any
> queue.
> Oct 20 13:19:46 VERBOSE[8171] logger.c:
> <-- SIP read from 172.16.8.200:5065:
> ACK sip:5155 at 172.16.200.5 SIP/2.0
> Via: SIP/2.0/UDP 172.16.8.200:5065;branch=z9hG4bKline0-2425957956
> To: <sip:5155 at 172.16.200.5>;tag=as279eb184
> From: "From Desk"<sip:5100 at 172.16.200.5>;tag=2425948795
> Call-Id: 2425954456-c4756-5065 at 172.16.8.200
> Max-Forwards: 70
> CSeq: 2 ACK
> Contact: sip:5100 at 172.16.8.200:5065
> Content-Length: 0
>
>
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 0: ACK
> sip:5155 at 172.16.200.5 SIP/2.0 (33)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 1: Via: SIP/2.0/UDP
> 172.16.8.200:5065;branch=z9hG4bKline0-2425957956 (65)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 2: To:
> <sip:5155 at 172.16.200.5>;tag=as279eb184 (42)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 3: From: "From
> Desk"<sip:5100 at 172.16.200.5>;tag=2425948795 (55)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 4: Call-Id:
> 2425954456-c4756-5065 at 172.16.8.200 (43)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 5: Max-Forwards: 70 (16)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 6: CSeq: 2 ACK (11)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 7: Contact:
> sip:5100 at 172.16.8.200:5065 (35)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 8: Content-Length: 0 (17)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 9: (0)
> Oct 20 13:19:46 VERBOSE[8171] logger.c: --- (9 headers 0 lines)Oct 20
> 13:19:46 VERBOSE[8171] logger.c: --- (9 headers 0 lines)---
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: **** Received ACK (6) - Command
> in SIP ACK
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: ** SIP TIMER: Cancelling
> retransmit of packet (reply received) Retransid #51636
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Stopping retransmission on
> '2425954456-c4756-5065 at 172.16.8.200' of Response 2: Match Found
> Oct 20 13:19:46 VERBOSE[8171] logger.c: Destroying call
> '2425954456-c4756-5065 at 172.16.8.200'
> Oct 20 13:19:46 VERBOSE[8171] logger.c:
> <-- SIP read from 172.16.8.200:5065:
> REFER sip:Unknown at 172.16.200.5 SIP/2.0
> Via: SIP/2.0/UDP 172.16.8.200:5065;branch=z9hG4bKline0-2425953069
> To: "Unknown" <sip:Unknown at 172.16.200.5>;tag=as25d7f97e
> From: <sip:5100 at 172.16.8.200:5065>;tag=2425967614
> Call-Id: 761da63b5eedcc6b125ad03042960be0 at 172.16.200.5
> Max-Forwards: 70
> CSeq: 2 REFER
> Contact: sip:5100 at 172.16.8.200:5065
> Refer-To:
> Referred-By: sip:5100 at 172.16.8.200:5065
> User-Agent: Citel-Handset-Gateway (DTS_E_DTP_32D)
> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, NOTIFY, PRACK, REFER
> Content-Length: 0
>
>
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 0: REFER
> sip:Unknown at 172.16.200.5 SIP/2.0 (38)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 1: Via: SIP/2.0/UDP
> 172.16.8.200:5065;branch=z9hG4bKline0-2425953069 (65)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 2: To: "Unknown"
> <sip:Unknown at 172.16.200.5>;tag=as25d7f97e (55)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 3: From:
> <sip:5100 at 172.16.8.200:5065>;tag=2425967614 (49)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 4: Call-Id:
> 761da63b5eedcc6b125ad03042960be0 at 172.16.200.5 (54)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 5: Max-Forwards: 70 (16)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 6: CSeq: 2 REFER (13)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 7: Contact:
> sip:5100 at 172.16.8.200:5065 (35)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 8: Refer-To: (10)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 9: Referred-By:
> sip:5100 at 172.16.8.200:5065 (39)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 10: User-Agent:
> Citel-Handset-Gateway (DTS_E_DTP_32D) (49)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 11: Allow: INVITE, ACK,
> CANCEL, BYE, OPTIONS, NOTIFY, PRACK, REFER (62)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 12: Content-Length: 0
> (17)
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 13: (0)
> Oct 20 13:19:46 VERBOSE[8171] logger.c: --- (13 headers 0 lines)Oct 20
> 13:19:46 VERBOSE[8171] logger.c: --- (13 headers 0 lines)---
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: **** Received REFER (9) -
> Command in SIP REFER
> Oct 20 13:19:46 DEBUG[8171] chan_sip.c: SIP call transfer received for
> call 761da63b5eedcc6b125ad03042960be0 at 172.16.200.5 (REFER)!
> Oct 20 13:19:46 WARNING[8171] chan_sip.c: Refer-to: Huh? Not a SIP
> header ()?
> Oct 20 13:19:46 VERBOSE[8171] logger.c: Transmitting (no NAT) to
> 172.16.8.200:5065:
> SIP/2.0 603 Declined
> Via: SIP/2.0/UDP
> 172.16.8.200:5065;branch=z9hG4bKline0-2425953069;received=172.16.8.200
> From: <sip:5100 at 172.16.8.200:5065>;tag=2425967614
> To: "Unknown" <sip:Unknown at 172.16.200.5>;tag=as25d7f97e
> Call-ID: 761da63b5eedcc6b125ad03042960be0 at 172.16.200.5
> CSeq: 2 REFER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:Unknown at 172.16.200.5>
> Content-Length: 0
> X-Asterisk-HangupCause: Normal Clearing
>
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