Hi, Starting in Asterisk 1.8.0, Asterisk supports connected line updates. This is fantastic for SIP. How can I prevent them from being sent to a PRI channel?
I'm having problems when a call is answered by an internal SIP extension, then transferred (blind or attended) to another internal SIP extension. One of my PRI providers can't handle the ROSE_ETSI_EctInform APDU and drops the call. Looking through the chan_dahdi and sig_pri code, I don't see any configuration flag to block the updates from going through. I was even hoping I could set __CONNECTED_LINE_CALLER_SEND_MACRO to something bogus, but it looks like if the macro fails to execute in any way, the code will just go ahead and update the connected line data. I wouldn't be surprised if the provider has the same issue with redirecting updates. Here's what happens when external number 87133306 calls into my PRI, extension 1111 answers, does an attended transfer to 0102, and completes the transfer. The provider eventually hangs up with "Message not compatible with call state (101)". channel.c: Released clone lock on 'SIP/1111-00000007<ZOMBIE>' channel.c: Done Masquerading DAHDI/i1/87133306-3 (6) chan_dahdi.c: Requested indication 26 on channel DAHDI/i1/87133306-3 chan_dahdi.c: Requested indication 17 on channel DAHDI/i1/87133306-3 channel.c: Bridge stops because we're zombie or need a soft hangup: c0=SIP/1111-00000007<ZOMBIE>, c1=SIP/1111-00000006, flags: Yes,Yes,No,No channel.c: Bridge stops bridging channels SIP/1111-00000007<ZOMBIE> and SIP/1111-00000006 chan_dahdi.c: Requested indication 22 on channel DAHDI/i1/87133306-3 sig_pri.c: Received AST_CONTROL_CONNECTED_LINE on DAHDI/i1/87133306-3 chan_dahdi.c: 1 Adding facility ie contents to send in FACILITY message: chan_dahdi.c: 1 ASN.1 dump chan_dahdi.c: 1 Context Specific/C [1 0x01] <A1> Len:24 <18> chan_dahdi.c: 1 Integer(2 0x02) <02> Len:1 <01> chan_dahdi.c: 1 <03> - "~" chan_dahdi.c: 1 OID(6 0x06) <06> Len:6 <06> chan_dahdi.c: 1 <04 00 82 71 01 05> - "~~~q~~" chan_dahdi.c: 1 Sequence/C(48 0x30) <30> Len:11 <0B> chan_dahdi.c: 1 Enumerated(10 0x0A) <0A> Len:1 <01> chan_dahdi.c: 1 <01> - "~" chan_dahdi.c: 1 Context Specific/C [0 0x00] <A0> Len:6 <06> chan_dahdi.c: 1 Context Specific [0 0x00] <80> Len:4 <04> chan_dahdi.c: 1 <30 31 30 32> - "0102" chan_dahdi.c: 1 ASN.1 end chan_dahdi.c: 1 INVOKE Component Context Specific/C [1 0x01] chan_dahdi.c: 1 invokeId Integer(2 0x02) = 3 0x0003 chan_dahdi.c: 1 operationValue OID(6 0x06) = 4.0.369.1.5 chan_dahdi.c: 1 operationValue = ROSE_ETSI_EctInform chan_dahdi.c: 1 EctInform Sequence/C(48 0x30) chan_dahdi.c: 1 callStatus Enumerated(10 0x0A) = 1 0x0001 chan_dahdi.c: 1 redirectionNumber PresentedNumberUnscreened chan_dahdi.c: 1 Explicit Context Specific/C [0 0x00] chan_dahdi.c: 1 presentationAllowedNumber PartyNumber chan_dahdi.c: 1 unknownPartyNumber Context Specific [0 0x00] = "0102" chan_dahdi.c: 1 chan_dahdi.c: 1 > DL-DATA request chan_dahdi.c: 1 > Protocol Discriminator: Q.931 (8) len=34 chan_dahdi.c: 1 > TEI=0 Call Ref: len= 2 (reference 43/0x2B) (Sent to originator) chan_dahdi.c: 1 > Message Type: FACILITY (98) chan_dahdi.c: 1 TEI=0 Transmitting N(S)=43, window is open V(A)=43 K=7 chan_dahdi.c: 1 chan_dahdi.c: 1 > Protocol Discriminator: Q.931 (8) len=34 chan_dahdi.c: 1 > TEI=0 Call Ref: len= 2 (reference 43/0x2B) (Sent to originator) chan_dahdi.c: 1 > Message Type: FACILITY (98) chan_dahdi.c: 1 > [1c 1b 91 a1 18 02 01 03 06 06 04 00 82 71 01 05 30 0b 0a 01 01 a0 06 80 04 30 31 30 32] chan_dahdi.c: 1 > Facility (len=29, codeset=0) [ 0x91, 0xA1, 0x18, 0x02, 0x01, 0x03, 0x06, 0x06, 0x04, 0x00, 0x82, 'q', 0x01, 0x05, '0', 0x0B, 0x0A, 0x01, 0x01, 0xA0, 0x06, 0x80, 0x04, '0102' ] channel.c: Hanging up channel 'SIP/1111-00000006' app_dial.c: Exiting with DIALSTATUS=ANSWER. pbx.c: Spawn extension (inbound_all,s,2) exited non-zero on 'SIP/1111-00000007<ZOMBIE>' pbx.c: == Spawn extension (inbound_all, s, 2) exited non-zero on 'SIP/1111-00000007<ZOMBIE>' channel.c: Soft-Hanging up channel 'SIP/1111-00000007<ZOMBIE>' channel.c: Hanging up zombie 'SIP/1111-00000007<ZOMBIE>' chan_sip.c: Stopping retransmission on '4e4d6d5a1152cff85ed7c18c6ed2b...@172.20.45.10' of Request 103: Match Found res_rtp_asterisk.c: No remote address on RTP instance '0x7f3a46298488' so dropping frame res_rtp_asterisk.c: No remote address on RTP instance '0x7f3a46298488' so dropping frame res_rtp_asterisk.c: No remote address on RTP instance '0x7f3a46298488' so dropping frame chan_sip.c: Stopping retransmission on '565a02b66888481b6862e89b53af3...@172.20.45.10' of Request 103: Match Found res_rtp_asterisk.c: No remote address on RTP instance '0x7f3a46298488' so dropping frame res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xa29ef8' res_rtp_asterisk.c: No remote address on RTP instance '0x7f3a46298488' so dropping frame chan_dahdi.c: 1 chan_dahdi.c: 1 < Protocol Discriminator: Q.931 (8) len=9 chan_dahdi.c: 1 < TEI=0 Call Ref: len= 2 (reference 43/0x2B) (Sent from originator) chan_dahdi.c: 1 < Message Type: DISCONNECT (69) chan_dahdi.c: 1 < [08 02 82 e5] chan_dahdi.c: 1 < Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the local user (2) chan_dahdi.c: 1 < Ext: 1 Cause: Message not compatible with call state (101), class = Protocol Error (e.g. unknown message) (6) ] chan_dahdi.c: 1 Received message for call 0xa285f0 on 0x7f3a46225b30 TEI/SAPI 0/0, call->pri is 0x7f3a46225b30 TEI/SAPI 0/0 chan_dahdi.c: 1 -- Processing IE 8 (cs0, Cause) chan_dahdi.c: 1 -- Found active call: 0xa285f0 cref:43 chan_dahdi.c: 1 q931.c:7201 post_handle_q931_message: Call 43 enters state 12 (Disconnect Indication). Hold state: Idle sig_pri.c: Span: 1 Processing event: PRI_EVENT_HANGUP_REQ sig_pri.c: -- Channel 0/7, span 1 got hangup request, cause 101 My chan_dahdi.conf: signalling = pri_cpe switchtype=euroisdn pridialplan=unknown prilocaldialplan=national resetinterval = 3600 usecallerid=yes threewaycalling=no facilityenable=no transfer=no context = from_e1_provider1 group=0 channel => 1-15,17-31 Thanks, Mike -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
