[asterisk-dev] geo location segmentation fault occurring on inbound call
Joshua C. Colp
jcolp at sangoma.com
Wed Sep 7 16:48:53 CDT 2022
Things like this should be filed on the issue tracker[1] along with a
backtrace[2].
[1] https://issues.asterisk.org/jira
[2] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
On Wed, Sep 7, 2022 at 6:24 PM Dan Cropp <dan at amtelco.com> wrote:
> I have two asterisk VMs setup. One Asterisk VM calls the other, passing
> geo location information.
>
>
>
> On one box, I initiate an Originate via AMI
>
>
>
> I pass in a blank A2 value (attempting to replicate a blank A2 field a
> customer’s SIP passes to them). Everything is fine when I don’t pass the
> A2=, portion.
>
> The easy answer is not to send the A2=, portion.
>
>
>
> However, it seems like it might be important to make sure Asterisk doesn’t
> generate a segmentation fault when a packet like this is received.
>
>
>
>
>
>
>
> Action: Originate^M
>
> ActionID: S29^M
>
> Channel: PJSIP/3333 at 192.168.33.33^M
>
> Exten: createcall^M
>
> Context: IS^M
>
> Priority: 1^M
>
> Timeout: 60000^M
>
> CallerID: John Doe <1234>^M
>
> Variable:
> CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=3,OriginateCallId=3^M
>
> Variable:
> GEOLOC_PROFILE(usage_rules)="retransmission-allowed=true",GEOLOC_PROFILE(allow_routing_use)="yes",GEOLOC_PROFILE(pidf_element)="tuple",GEOLOC_PROFILE(confidence)="pdf=Normal,value=80",GEOLOC_PROFILE(location_info)="country=CA,\
>
> A1=QC,A2=,A3=COOKSHIRE EATON,HNO=160,RD=POPE,PC=J0B 1M0,NAM=CENTRE DE
> SANTE
> COOKSHIRE",GEOLOC_PROFILE(method)="Manual",GEOLOC_PROFILE(format)="civicAddress"^M
>
> Async: true^M
>
> Codecs: ulaw^M
>
> ^M
>
>
>
>
>
> The second box sees the INVITE and starts processing it but then outputs a
> Segmentation fault.
>
>
>
>
>
>
>
> <--- Received SIP request (2262 bytes) from UDP:192.168.33.31:5060 --->
>
> INVITE sip:3333 at 192.168.33.33 SIP/2.0
>
> Via: SIP/2.0/UDP 192.168.33.31:5060
> ;rport;branch=z9hG4bKPj62f47e49-29fd-420e-97fb-d9a1970aba48
>
> From: "John Doe" <sip:1234 at 192.168.33.31
> >;tag=36ae7a5e-8264-4a38-8468-873aa245985e
>
> To: <sip:3333 at 192.168.33.33>
>
> Contact: <sip:asterisk at 192.168.33.31:5060>
>
> Call-ID: 0ab86466-cbca-4d11-a0dd-b8d14b9e687c
>
> CSeq: 1996 INVITE
>
> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE,
> CANCEL, UPDATE, PRACK, MESSAGE, REFER
>
> Supported: 100rel, timer, replaces, norefersub, histinfo
>
> Session-Expires: 1800
>
> Min-SE: 90
>
> Geolocation: <cid:twpre at 192.168.33.31>
>
> Geolocation-Routing: yes
>
> Max-Forwards: 70
>
> User-Agent: Asterisk PBX 18.14.0
>
> Content-Type: multipart/mixed;boundary=1ffe84bc-e538-44a4-a860-788994f54a28
>
> Content-Length: 1479
>
>
>
>
>
> --1ffe84bc-e538-44a4-a860-788994f54a28
>
> Content-Type: application/sdp
>
> Content-Length: 181
>
>
>
> v=0
>
> o=- 812269913 812269913 IN IP4 192.168.33.31
>
> s=Asterisk
>
> c=IN IP4 192.168.33.31
>
> t=0 0
>
> m=audio 10742 RTP/AVP 0
>
> a=rtpmap:0 PCMU/8000
>
> a=ptime:20
>
> a=maxptime:150
>
> a=sendrecv
>
>
>
> --1ffe84bc-e538-44a4-a860-788994f54a28
>
> Content-ID: <twpre at 192.168.33.31>
>
> Content-Type: application/pidf+xml
>
> Content-Length: 1018
>
>
>
> <?xml version="1.0"?>
>
> <presence xmlns="urn:ietf:params:xml:ns:pidf"
> xmlns:ca="urn:ietf:params:xml:ns:pidf:geopriv10:civicAddr"
> xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:fn="
> http://www.w3.org/2005/xpath-functions"
> xmlns:gbp="urn:ietf:params:xml:ns:pidf:geopriv10:basicPolicy" xmlns:gml="
> http://www.opengis.net/gml"
> xmlns:gp="urn:ietf:params:xml:ns:pidf:geopriv10" xmlns:gs="
> http://www.opengis.net/pidflo/1.0"
> xmlns:con="urn:ietf:params:xml:ns:geopriv:conf" xmlns:date="
> http://exslt.org/dates-and-times" entity="PJSIP/192.168.33.33-00000002">
>
> <tuple>
>
> <status>
>
> <gp:geopriv>
>
> <gp:location-info>
>
> <ca:civicAddress xml:lang="en"/>
>
> <con:confidence pdf="Normal">80</con:confidence>
>
> </gp:location-info>
>
> <gp:usage-rules>
>
> <gp:retransmission-allowed>true</gp:retransmission-allowed>
>
> </gp:usage-rules>
>
> <gp:method>Manual</gp:method>
>
> </gp:geopriv>
>
> </status>
>
> <timestamp>2022-09-07T21:10:45Z</timestamp>
>
> </tuple>
>
> </presence>
>
>
>
> --1ffe84bc-e538-44a4-a860-788994f54a28--
>
>
>
> <--- Transmitting SIP response (376 bytes) to UDP:192.168.33.31:5060 --->
>
> SIP/2.0 100 Trying
>
> Via: SIP/2.0/UDP 192.168.33.31:5060
> ;rport=5060;received=192.168.33.31;branch=z9hG4bKPj62f47e49-29fd-420e-97fb-d9a1970aba48
>
> Call-ID: 0ab86466-cbca-4d11-a0dd-b8d14b9e687c
>
> From: "John Doe" <sip:1234 at 192.168.33.31
> >;tag=36ae7a5e-8264-4a38-8468-873aa245985e
>
> To: <sip:3333 at 192.168.33.33>
>
> CSeq: 1996 INVITE
>
> Server: Asterisk PBX 18.14.0
>
> Content-Length: 0
>
>
>
>
>
> Segmentation fault
>
>
>
>
>
>
>
> [09/07 16:10:45.813] DEBUG[102650] res_pjsip/pjsip_distributor.c: Could
> not find matching transaction for Request msg INVITE/cseq=1996
> (rdata0x7f509c002578)
>
> [09/07 16:10:45.813] DEBUG[102650] res_pjsip/pjsip_distributor.c:
> Calculated serializer pjsip/distributor-00000059 to use for Request msg
> INVITE/cseq=1996 (rdata0x7f509c002578)
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_endpoint_identifier_ip.c:
> Source address 192.168.33.31:5060 does not match identify 'identify55'
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_endpoint_identifier_ip.c:
> Source address 192.168.33.31:5060 does not match identify 'identify57'
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_endpoint_identifier_ip.c:
> Source address 192.168.33.31:5060 does not match identify 'identify59'
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_endpoint_identifier_ip.c:
> Source address 192.168.33.31:5060 matches identify 'identify61'
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_endpoint_identifier_ip.c:
> Identify 'identify61' SIP message matched to endpoint 192.168.33.31
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_session.c: (null session)
> Request: INVITE
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_session.c: Request:
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip/pjsip_distributor.c:
> Calculated serializer pjsip/distributor-00000059 to use for Request msg
> INVITE/cseq=1996 (rdata0x7f509c016b98)
>
> [09/07 16:10:45.813] DEBUG[102651] chan_pjsip.c: 192.168.33.31
>
> [09/07 16:10:45.813] DEBUG[102651] chan_pjsip.c: Direct media no glare
> mitigation
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_session.c: 192.168.33.31
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_session.c: 192.168.33.31:
> Call (UDP:192.168.33.31:5060) to extension '3333' sending 100 Trying
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_session.c: 192.168.33.31:
> Method is INVITE, Response is 100 Trying
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_session.c: 192.168.33.31
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_session.c: 192.168.33.31
> Event: TSX_STATE Inv State: INCOMING
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_session.c: 192.168.33.31:
> Source of transaction state change is TX_MSG
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_session.c:
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_session.c: 192.168.33.31 TSX
> State: Proceeding Inv State: INCOMING
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_session.c: Nothing delayed
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_session.c: 192.168.33.31 TSX
> State: Proceeding Inv State: INCOMING
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_session.c: Topology:
> Pending: (null topology) Active: (null topology)
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_session.c:
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_session.c: 192.168.33.31:
> Media count: 1
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_session.c: 192.168.33.31:
> Processing stream 0
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_session.c: 192.168.33.31:
> Using audio-0 for new stream name
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_session.c: 192.168.33.31:
> Using new stream 0:audio-0:audio:sendrecv (nothing)
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_session.c: 192.168.33.31
> Adding position 0
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_session.c: Creating new
> media session
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_session.c: Setting media
> session as default for audio
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_session.c: Done
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_session.c: 192.168.33.31:
> Negotiating incoming SDP media stream 0:audio-0:audio:sendrecv (nothing)
> using audio SDP handler
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_sdp_rtp.c: 192.168.33.31
>
> [09/07 16:10:45.813] DEBUG[102651] res_pjsip_sdp_rtp.c: Transport
> transport1 bound to 0.0.0.0: Using it for RTP media.
>
> [09/07 16:10:45.813] DEBUG[102651] rtp_engine.c: Using engine 'asterisk'
> for RTP instance '0x7f507003eef0'
>
> [09/07 16:10:45.813] DEBUG[102651] res_rtp_asterisk.c: (0x7f507003eef0)
> RTP allocated port 14734
>
> [09/07 16:10:45.813] DEBUG[102651] res_rtp_asterisk.c: (0x7f507003eef0)
> ICE creating session 0.0.0.0:14734 (14734)
>
> [09/07 16:10:45.813] DEBUG[102651] res_rtp_asterisk.c: (0x7f507003eef0)
> ICE create
>
> [09/07 16:10:45.814] DEBUG[102651] res_rtp_asterisk.c: (0x7f507003eef0)
> ICE add system candidates
>
> [09/07 16:10:45.814] DEBUG[102651] res_rtp_asterisk.c: (0x7f507003eef0)
> ICE add candidate: 192.168.33.33:14734, 2130706431
>
> [09/07 16:10:45.814] DEBUG[102651] rtp_engine.c: RTP instance
> '0x7f507003eef0' is setup and ready to go
>
> [09/07 16:10:45.814] DEBUG[102651] res_rtp_asterisk.c: (0x7f507003eef0)
> ICE stopped
>
> [09/07 16:10:45.814] DEBUG[102651] res_rtp_asterisk.c: (0x7f507003eef0)
> RTCP setup on RTP instance
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_sdp_rtp.c: 192.168.33.31
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_sdp_rtp.c: 192.168.33.31
>
> [09/07 16:10:45.814] DEBUG[102651] rtp_engine.c: Setting tx payload type 0
> based on m type on 0x7f503ca182b0
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_sdp_rtp.c:
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_session/pjsip_session_caps.c:
> '192.168.33.31' Caps for incoming audio call with pref 'local' - remote:
> (ulaw) local: (ulaw) joint: (ulaw)
>
> [09/07 16:10:45.814] DEBUG[102651] rtp_engine.c: Crossover copying tx to
> rx payload mapping 0 (0x7f5070020ed8) from 0x7f503ca182b0 to 0x7f503ca182b0
>
> [09/07 16:10:45.814] DEBUG[102651] rtp_engine.c: Copying rx payload
> mapping 0 (0x7f5070020ed8) from 0x7f503ca182b0 to 0x7f507003f0c8
>
> [09/07 16:10:45.814] DEBUG[102651] rtp_engine.c: Copying tx payload
> mapping 0 (0x7f5070020ed8) from 0x7f503ca182b0 to 0x7f507003f0c8
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_sdp_rtp.c:
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_sdp_rtp.c:
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_session.c: 192.168.33.31:
> Media stream 0:audio-0:audio:sendrecv (ulaw) handled by audio
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_session.c: 192.168.33.31:
> Done with stream 0:audio-0:audio:sendrecv (ulaw)
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_session.c: 192.168.33.31:
> Handled? yes
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_session.c: 192.168.33.31
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_session.c: 192.168.33.31:
> Processing streams
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_session.c: 192.168.33.31:
> Processing stream 0:audio-0:audio:sendrecv (ulaw)
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_session.c: 192.168.33.31
> Adding position 0
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_session.c: Using existing
> media_session
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_session.c: 192.168.33.31
> Stream: 0:audio-0:audio:sendrecv (ulaw)
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_sdp_rtp.c: 192.168.33.31
> Type: audio 0:audio-0:audio:sendrecv (ulaw)
>
> [09/07 16:10:45.814] DEBUG[102651] res_rtp_asterisk.c: (0x7f507003eef0)
> RTCP ignoring duplicate property
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_sdp_rtp.c: RC: 1
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_session.c: Had handler
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_session.c: 192.168.33.31:
> Stream 0:audio-0:audio:sendrecv (ulaw) added
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_session.c: 192.168.33.31:
> Done with 0:audio-0:audio:sendrecv (ulaw)
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_session.c: 192.168.33.31:
> Adding bundle groups (if available)
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_session.c: 192.168.33.31:
> Copying connection details
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_session.c: 192.168.33.31:
> Processing media 0
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_session.c: 192.168.33.31:
> Media 0 reset
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_session.c: 192.168.33.31
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_session.c: 192.168.33.31:
> Method is INVITE
>
> [09/07 16:10:45.814] DEBUG[102651] chan_pjsip.c: 192.168.33.31
>
> [09/07 16:10:45.814] DEBUG[102651] chan_pjsip.c: 192.168.33.31
>
> [09/07 16:10:45.814] DEBUG[102651] channel_internal_api.c:
> <initializing>: Formats: (none)
>
> [09/07 16:10:45.814] DEBUG[102651] channel_internal_api.c: Channel is
> being initialized or destroyed
>
> [09/07 16:10:45.814] DEBUG[102651] stasis.c: Creating topic. name:
> channel:1662585045.0, detail:
>
> [09/07 16:10:45.814] DEBUG[102651] stasis.c: Topic 'channel:1662585045.0':
> 0x7f5070046e20 created
>
> [09/07 16:10:45.814] DEBUG[102651] channel.c: Channel 0x7f507005c950
> 'PJSIP/192.168.33.31-00000000' allocated
>
> [09/07 16:10:45.814] DEBUG[102651] chan_pjsip.c: Topology:
> <0:audio-0:audio:sendrecv (ulaw)> Formats: (ulaw)
>
> [09/07 16:10:45.814] DEBUG[102651] chan_pjsip.c: Compatible? yes
>
> [09/07 16:10:45.814] DEBUG[102651] channel_internal_api.c:
> PJSIP/192.168.33.31-00000000: MultistreamFormats: (ulaw)
>
> [09/07 16:10:45.814] DEBUG[102651] channel_internal_api.c: Set native
> formats but not topology
>
> [09/07 16:10:45.814] DEBUG[102651] channel_internal_api.c:
> PJSIP/192.168.33.31-00000000: <0:audio-0:audio:sendrecv (ulaw)>
>
> [09/07 16:10:45.814] DEBUG[102651] channel_internal_api.c: Used provided
> topology
>
> [09/07 16:10:45.814] DEBUG[102651] chan_pjsip.c:
>
> [09/07 16:10:45.814] DEBUG[102651] chan_pjsip.c:
> PJSIP/192.168.33.31-00000000
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_geolocation.c:
> PJSIP/192.168.33.31-00000000
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_geolocation.c:
> PJSIP/192.168.33.31-00000000: Geolocation: <cid:twpre at 192.168.33.31>
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_geolocation.c: Processing
> URI '<cid:twpre at 192.168.33.31>'
>
> [09/07 16:10:45.814] DEBUG[102651] res_pjsip_geolocation.c: Processing
> PIDF-LO '<cid:twpre at 192.168.33.31>'
>
> [09/07 16:10:45.814] DEBUG[102651] res_geolocation/geoloc_eprofile.c:
> PJSIP/192.168.33.31-00000000
>
> [09/07 16:10:45.815] DEBUG[102651] res_geolocation/geoloc_eprofile.c:
> PJSIP/192.168.33.31-00000000
>
> [09/07 16:10:45.815] DEBUG[102651] res_geolocation/geoloc_eprofile.c:
> PJSIP/192.168.33.31-00000000
>
> [09/07 16:10:45.815] DEBUG[102651] res_geolocation/geoloc_eprofile.c:
> PJSIP/192.168.33.31-00000000
>
> [09/07 16:10:45.815] DEBUG[102651] res_geolocation/geoloc_eprofile.c:
> PJSIP/192.168.33.31-00000000: Done
>
>
>
> Dan
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
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>
--
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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