Snom 300 - Outgoing Calls dropped after 5 seconds

Discussion in '3CX Phone System - General' started by cdstech, Mar 30, 2012.

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  1. cdstech

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    Hi

    I have a 3CX IP PBX with 4 extensions of which one is internal to 3CX and the other 3 extensions are remote.

    I have had a problem with one of the external Snom 300 extensions, it registers OK, can receive incoming calls from external and internal numbers, however when it makes an outgoing call either internally or externally, the call gets dropped after 5 secs, any idea's, log and SIP Trace below :


    30/3/2012 15:37:02 [INFO ] PHN: WEB: Request 146/429/0: GET /index.htm HTTP/1.1
    30/3/2012 15:37:02 [DEBUG0] TOOLS: Uri Replace >{web_lng_iso_code}< in >http://snom.com:80/{web_lng_iso_code}/AddOn< with >en<
    30/3/2012 15:37:02 [DEBUG0] TOOLS: Uri Replaced result: >http://snom.com:80/en/AddOn<
    30/3/2012 15:37:02 [DEBUG0] TOOLS: Uri Replace >{web_lng_iso_code}< in >http://snom.com:80/{web_lng_iso_code}/AddOn< with >en<
    30/3/2012 15:37:02 [DEBUG0] TOOLS: Uri Replaced result: >http://snom.com:80/en/AddOn<
    30/3/2012 15:37:02 [INFO ] PHN: ETH STATUS: unchanged up
    30/3/2012 15:37:04 [INFO ] PHN: WEB: Request 146/592/0: POST /index.htm HTTP/1.1
    30/3/2012 15:37:04 [INFO ] GUI: GUI: Allocating channel 15, line 0, current 0
    30/3/2012 15:37:04 [INFO ] GUI: find_function_key_for_uri: Looking to match 101@voip.testdomain.com
    30/3/2012 15:37:04 [DEBUG2] GUI: Line 15: state change 'closed' (5) -> 'offhook' (6)
    30/3/2012 15:37:04 [DEBUG2] CSTA: Received event for identity sip:101@voip.testdomain.com
    30/3/2012 15:37:04 [DEBUG2] PHN: RTP: set_port - ports: 49152 65534, con 19
    30/3/2012 15:37:04 [INFO ] PHN: rtp_port::send_ice_rtp_packet(), rtp=true
    30/3/2012 15:37:04 [INFO ] PHN: rtp_port::send_ice_rtp_packet(), rtp=false
    30/3/2012 15:37:04 [INFO ] PHN: SIP: Dialog 15/19/1 going to trying
    30/3/2012 15:37:04 [INFO ] GUI: GUI: Line-Seize subscription was not sent
    30/3/2012 15:37:04 [DEBUG2] GUI: Line 15: Destruct
    30/3/2012 15:37:05 [DEBUG2] GUI: Line 15: state change 'offhook' (6) -> 'calling' (7)
    30/3/2012 15:37:05 [DEBUG2] CSTA: Received event for identity sip:101@voip.testdomain.com
    30/3/2012 15:37:05 [DEBUG0] PHN: Request lookup for number sip:103@voip.testdomain.com;user=phone
    30/3/2012 15:37:05 [DEBUG0] PHN: Request lookup for number 101@voip.testdomain.com
    30/3/2012 15:37:05 [DEBUG2] PHN: TMR: setting up 'scroll_text92' 0x100cc230 to fire periodically every 400 ms
    30/3/2012 15:37:05 [DEBUG2] PHN: TMR: setting up 'gui_state.cpp (738): [static void gui_object::DisplayStateCbAtNextLoop(void *, void *)]' 0x10058b74 to fire periodically every 500 ms
    30/3/2012 15:37:05 [DEBUG2] PHN: RTP: ice_rtp.received_ice still not set
    30/3/2012 15:37:05 [INFO ] PHN: Unknown message attribute type 0x8022
    30/3/2012 15:37:05 [DEBUG2] PHN: RTP: ice_rtcp.received_ice still not set
    30/3/2012 15:37:05 [INFO ] PHN: Unknown message attribute type 0x8022
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: rtp allocated for connection 15, id 19
    30/3/2012 15:37:05 [NOTICE] PHN: SDP: Skipped Codec G723 is not supported for 20 ms packets
    30/3/2012 15:37:05 [DEBUG0] PHN: SIP: set_additional_addresses
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: No special routing, routing to sip:103@voip.testdomain.com;user=phone
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: route pending packet 1000247: entry=url ? sip:103@voip.testdomain.com;user=phone all 1
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: route pending packet 1000247: entry=srv tls _sips._tcp.voip.testdomain.com all 1
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: route pending packet 1000247: entry=tls 123.456.789.123 5061 all 1
    30/3/2012 15:37:05 [DEBUG2] PHN: Allocate SIP connection SIP-83 1
    30/3/2012 15:37:05 [INFO ] PHN: SIP: Dialog 15/19/0 going to trying
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: route pending packet 1000247: entry=tls 123.456.789.123 5061 all 1
    30/3/2012 15:37:05 [INFO ] PHN: SIP: Use Connection tls:192.168.1.61:2138 for packet 1000247
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: send INVITE (1: 4f75c510f353-w79oibxqbfbp) -> tls:123.456.789.123:5061
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: result of get_ip_adr:123.456.789.123 192.168.1.61
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: result of get_ip_adr:123.456.789.123 192.168.1.61
    30/3/2012 15:37:05 [DEBUG2] PHN: TMR: 'gui_state.cpp (736): [static void gui_object::DisplayStateCbAtNextLoop(void *, void *)]' 0x10058b24
    30/3/2012 15:37:05 [DEBUG2] PHN: TCP connection SIP-83 1 closed by server
    30/3/2012 15:37:05 [ERROR ] PHN: SIP: transport error: 1000247 -> tls:123.456.789.123:5061
    30/3/2012 15:37:05 [NOTICE] PHN: SIP: Add dirty host: tls:123.456.789.123:5061 (60 sec)
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: setup dirty host expiration for (74/0) tls:123.456.789.123:5061 in 60 secs
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: route pending packet 1000247: entry=srv tcp _sip._tcp.voip.testdomain.com all 1
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: route pending packet 1000247: entry=tcp 123.456.789.123 5060 all 1
    30/3/2012 15:37:05 [INFO ] PHN: SIP: Use Connection tcp:192.168.1.61:2134 for packet 1000247
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: send INVITE (1: 4f75c510f353-w79oibxqbfbp) -> tcp:123.456.789.123:5060
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: result of get_ip_adr:123.456.789.123 192.168.1.61
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: result of get_ip_adr:123.456.789.123 192.168.1.61
    30/3/2012 15:37:05 [DEBUG0] GUI: GUI: set_led: nr 1 id 1 state 1
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: recv 407 INVITE (1: 4f75c510f353-w79oibxqbfbp) <- tcp:123.456.789.123:5060
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: clear message repetition 1000247 4f75c510f353-w79oibxqbfbp
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: Routing to explicit plan tcp 123.456.789.123 5060
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: route pending packet 1000248: entry=tcp 123.456.789.123 5060 all 2
    30/3/2012 15:37:05 [INFO ] PHN: SIP: Use Connection tcp:192.168.1.61:2134 for packet 1000248
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: send ACK (1: 4f75c510f353-w79oibxqbfbp) -> tcp:123.456.789.123:5060
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: result of get_ip_adr:123.456.789.123 192.168.1.61
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: result of get_ip_adr:123.456.789.123 192.168.1.61
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: sip::process_auth Identity=0
    30/3/2012 15:37:05 [WARN ] PHN: SIP: process_registrar_packet: 401 needs 128 bit nonce
    30/3/2012 15:37:05 [NOTICE] PHN: SIP: process auth:Match challenge for user=101, realm=3CXPhoneSystem
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: Routing to explicit plan tcp 123.456.789.123 5060
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: route pending packet 1000249: entry=tcp 123.456.789.123 5060 all 3
    30/3/2012 15:37:05 [INFO ] PHN: SIP: Use Connection tcp:192.168.1.61:2134 for packet 1000249
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: send INVITE (2: 4f75c510f353-w79oibxqbfbp) -> tcp:123.456.789.123:5060
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: result of get_ip_adr:123.456.789.123 192.168.1.61
    30/3/2012 15:37:05 [DEBUG2] PHN: SIP: result of get_ip_adr:123.456.789.123 192.168.1.61
    30/3/2012 15:37:06 [DEBUG2] PHN: SIP: recv 100 INVITE (2: 4f75c510f353-w79oibxqbfbp) <- tcp:123.456.789.123:5060
    30/3/2012 15:37:06 [DEBUG2] PHN: SIP: clear message repetition 1000249 4f75c510f353-w79oibxqbfbp
    30/3/2012 15:37:06 [INFO ] PHN: SIP: Dialog 15/19/1 going to proceeding
    30/3/2012 15:37:06 [DEBUG2] GUI: if_state(15, SIP/2.0 100 Trying)
    30/3/2012 15:37:06 [DEBUG2] PHN: Goto Best State from 'Calling' (9), force 0
    30/3/2012 15:37:06 [DEBUG0] PHN: Request lookup for number sip:103@voip.testdomain.com
    30/3/2012 15:37:06 [DEBUG2] PHN: Goto Best State from 'Calling' (9), force 0
    30/3/2012 15:37:06 [DEBUG2] PHN: Goto Best State from 'Calling' (9), force 0
    30/3/2012 15:37:06 [DEBUG0] PHN: Request lookup for number sip:103@voip.testdomain.com;user=phone
    30/3/2012 15:37:06 [DEBUG0] PHN: Request lookup for number 101
    30/3/2012 15:37:06 [DEBUG2] PHN: TMR: setting up 'gui_state.cpp (738): [static void gui_object::DisplayStateCbAtNextLoop(void *, void *)]' 0x10058b74 to fire periodically every 500 ms
    30/3/2012 15:37:06 [DEBUG2] PHN: TMR: 'gui_state.cpp (736): [static void gui_object::DisplayStateCbAtNextLoop(void *, void *)]' 0x10058b24
    30/3/2012 15:37:07 [INFO ] PHN: ETH STATUS: unchanged up
    30/3/2012 15:37:08 [NOTICE] PHN: TPL: Socket 146 idle/connect timeout
    30/3/2012 15:37:08 [DEBUG2] PHN: WEB: HTTP 146/80 disconnected
    30/3/2012 15:37:08 [DEBUG2] PHN: WEB: HTTP 146/80 deleted
    30/3/2012 15:37:08 [DEBUG2] PHN: SIP: recv 180 INVITE (2: 4f75c510f353-w79oibxqbfbp) <- tcp:123.456.789.123:5060
    30/3/2012 15:37:08 [DEBUG2] PHN: SIP: clear message repetition 1000249 4f75c510f353-w79oibxqbfbp
    30/3/2012 15:37:08 [INFO ] PHN: SIP: Dialog 15/19/2 going to early
    30/3/2012 15:37:08 [DEBUG2] GUI: if_state(15, SIP/2.0 180 Ringing)
    30/3/2012 15:37:08 [DEBUG2] PHN: Goto Best State from 'Calling' (9), force 0
    30/3/2012 15:37:08 [DEBUG0] PHN: Request lookup for number 103
    30/3/2012 15:37:08 [DEBUG2] PHN: Goto Best State from 'Calling' (9), force 0
    30/3/2012 15:37:08 [DEBUG2] GUI: Line 15: state change 'calling' (7) -> 'ringback' (8)
    30/3/2012 15:37:08 [DEBUG2] CSTA: Received event for identity sip:101@voip.testdomain.com
    30/3/2012 15:37:08 [DEBUG2] PHN: Goto Best State from 'Calling' (9), force 0
    30/3/2012 15:37:08 [DEBUG0] PHN: Request lookup for number sip:103@voip.testdomain.com;user=phone
    30/3/2012 15:37:08 [DEBUG2] PHN: TMR: setting up 'gui_state.cpp (738): [static void gui_object::DisplayStateCbAtNextLoop(void *, void *)]' 0x10058b74 to fire periodically every 500 ms
    30/3/2012 15:37:08 [DEBUG2] PHN: TMR: 'gui_state.cpp (736): [static void gui_object::DisplayStateCbAtNextLoop(void *, void *)]' 0x10058b24
    30/3/2012 15:37:09 [DEBUG2] PHN: TMR: 'registrar.cpp (754): [static void sip::refresh_crlf(void *, void *)]' 0x100668f0
    30/3/2012 15:37:09 [DEBUG2] PHN: Sending Keepalive (CRLF) to tcp:123.456.789.123:5060
    30/3/2012 15:37:09 [DEBUG2] PHN: SIP: recv 200 INVITE (2: 4f75c510f353-w79oibxqbfbp) <- tcp:123.456.789.123:5060
    30/3/2012 15:37:09 [DEBUG2] PHN: SIP: clear message repetition 1000249 4f75c510f353-w79oibxqbfbp
    30/3/2012 15:37:09 [INFO ] PHN: SIP: Dialog 15/19/3 going to confirmed
    30/3/2012 15:37:09 [DEBUG2] GUI: if_state(15, SIP/2.0 200 OK)
    30/3/2012 15:37:09 [DEBUG2] PHN: Goto Best State from 'Calling' (9), force 0
    30/3/2012 15:37:09 [DEBUG2] PHN: SIP: No special routing, routing to sip:103@192.168.1.100:5060;transport=TCP
    30/3/2012 15:37:09 [DEBUG2] PHN: SIP: route pending packet 1000250: entry=url ? sip:103@192.168.1.100:5060;transport=TCP all 4
    30/3/2012 15:37:09 [DEBUG2] PHN: SIP: route pending packet 1000250: entry=TCP 192.168.1.100 5060 all 4
    30/3/2012 15:37:09 [DEBUG2] PHN: Allocate SIP connection SIP-84 1
    30/3/2012 15:37:09 [DEBUG0] PHN: Request lookup for number sip:103@voip.testdomain.com;user=phone
    30/3/2012 15:37:09 [DEBUG2] PHN: Goto Best State from 'Calling' (9), force 0
    30/3/2012 15:37:10 [NOTICE] PHN: RTP: set_destination RP15 adr=123.456.789.123:9046
    30/3/2012 15:37:10 [INFO ] PHN: RTP: set_audible(0/0 -> 1/1): RP15 changed=1, update_req=1, adr=123.456.789.123:9046
    30/3/2012 15:37:10 [INFO ] PHN: sip::gui_connected(15, 0, 1)
    30/3/2012 15:37:10 [DEBUG2] GUI: if_connected(15, 0, 1)
    30/3/2012 15:37:10 [DEBUG2] GUI: if_connected id 15, state ringback
    30/3/2012 15:37:10 [DEBUG2] GUI: Line 15: state change 'ringback' (8) -> 'connected' (0)
    30/3/2012 15:37:10 [DEBUG2] CSTA: Received event for identity sip:101@voip.testdomain.com
    30/3/2012 15:37:10 [DEBUG2] PHN: Goto Best State from 'Calling' (9), force 0
    30/3/2012 15:37:10 [DEBUG2] PHN: TMR: setting up 'gui_state_connected.cpp (753): [void gui_object::enter_on_connected()]' 0x10058dcc to fire periodically every 1000 ms
    30/3/2012 15:37:10 [DEBUG0] PHN: Request lookup for number 103
    30/3/2012 15:37:10 [DEBUG2] PHN: TMR: setting up 'selection_list.cpp (2070): [void gui_object::SelectionListCtx::draw(bool = false)]' 0x1005b26c to fire periodically every 1000 ms
    30/3/2012 15:37:10 [DEBUG2] PHN: TMR: setting up 'gui_state.cpp (738): [static void gui_object::DisplayStateCbAtNextLoop(void *, void *)]' 0x10058b74 to fire periodically every 500 ms
    30/3/2012 15:37:10 [DEBUG2] PHN: SIP: route pending packet 1000250: entry=TCP 192.168.1.100 5060 all 4
    30/3/2012 15:37:10 [INFO ] PHN: SIP: Use Connection TCP:192.168.1.61:2139 for packet 1000250
    30/3/2012 15:37:10 [DEBUG2] PHN: SIP: send ACK (2: 4f75c510f353-w79oibxqbfbp) -> TCP:192.168.1.100:5060
    30/3/2012 15:37:10 [DEBUG2] PHN: SIP: result of get_ip_adr:192.168.1.100 192.168.1.61
    30/3/2012 15:37:10 [DEBUG2] PHN: SIP: result of get_ip_adr:192.168.1.100 192.168.1.61
    30/3/2012 15:37:10 [DEBUG2] PHN: send rtp keep alive!
    30/3/2012 15:37:10 [DEBUG2] PHN: TMR: 'gui_state.cpp (736): [static void gui_object::DisplayStateCbAtNextLoop(void *, void *)]' 0x10058b24
    30/3/2012 15:37:10 [DEBUG2] PHN: SIP: recv 200 INVITE (2: 4f75c510f353-w79oibxqbfbp) <- tcp:123.456.789.123:5060
    30/3/2012 15:37:11 [DEBUG2] PHN: SIP: recv 200 INVITE (2: 4f75c510f353-w79oibxqbfbp) <- tcp:123.456.789.123:5060
    30/3/2012 15:37:12 [DEBUG2] PHN: TCP connection SIP-84 1 closed by server
    30/3/2012 15:37:12 [ERROR ] PHN: SIP: transport error: 1000250 -> TCP:192.168.1.100:5060
    30/3/2012 15:37:12 [NOTICE] PHN: SIP: Add dirty host: TCP:192.168.1.100:5060 (60 sec)
    30/3/2012 15:37:12 [DEBUG2] PHN: SIP: setup dirty host expiration for (75/0) TCP:192.168.1.100:5060 in 60 secs
    30/3/2012 15:37:12 [NOTICE] PHN: SIP: final transport error: 1000250 -> TCP:192.168.1.100:5060
    30/3/2012 15:37:12 [DEBUG2] GUI: SipReportsEndOfIncomingCall(15, 11505281, , ) 1
    30/3/2012 15:37:12 [DEBUG2] GUI: Line 15: state change 'connected' (0) -> 'closed' (5)
    30/3/2012 15:37:12 [DEBUG2] CSTA: Received event for identity sip:101@voip.testdomain.com
    30/3/2012 15:37:12 [INFO ] PHN: SIP: Dialog 15/19/4 going to terminated
    30/3/2012 15:37:12 [DEBUG2] PHN: SIP: mark for deletion 19/4f75c510f353-w79oibxqbfbp, cbi 0, 0
    30/3/2012 15:37:12 [INFO ] PHN: SIP: packet 1000247 stopped retransmission
    30/3/2012 15:37:12 [INFO ] PHN: SIP: packet 1000249 stopped retransmission
    30/3/2012 15:37:12 [DEBUG2] PHN: SIP: delete connection 15/19 in 120 secs
    30/3/2012 15:37:12 [DEBUG2] PHN: Goto Best State from 'Connected' (12), force 0
    30/3/2012 15:37:13 [DEBUG0] TOOLS: Uri Replace >../< in >xml/gui/message.xml< with ><
    30/3/2012 15:37:13 [DEBUG0] TOOLS: Uri Replaced result: >file:///xml/gui/message.xml<
    30/3/2012 15:37:13 [INFO ] PHN: Xpath applies: /SnomIPPhoneBatch/
    30/3/2012 15:37:13 [INFO ] PHN: Xpath contains unrecognized steps, aborting
    30/3/2012 15:37:13 [ERROR ] PHN: SIP: transport error 1000250: generating fake 598
    30/3/2012 15:37:13 [DEBUG2] PHN: SIP: recv 598 ACK (2: 4f75c510f353-w79oibxqbfbp) <- ::0
    30/3/2012 15:37:13 [DEBUG2] PHN: SIP: clear message repetition 1000250 4f75c510f353-w79oibxqbfbp
    30/3/2012 15:37:13 [DEBUG2] PHN: SIP: mark for deletion 19/4f75c510f353-w79oibxqbfbp, cbi 1, 1
    30/3/2012 15:37:13 [INFO ] PHN: SIP: packet 1000247 stopped retransmission
    30/3/2012 15:37:13 [INFO ] PHN: SIP: packet 1000249 stopped retransmission
    30/3/2012 15:37:13 [DEBUG2] PHN: SIP: delete connection 15/19 in 0 secs
    30/3/2012 15:37:13 [INFO ] PHN: Xpath applies: /SnomIPPhoneBatch/
    30/3/2012 15:37:13 [INFO ] PHN: Xpath contains unrecognized steps, aborting
    30/3/2012 15:37:13 [DEBUG2] PHN: TMR: setting up 'gui_state.cpp (738): [static void gui_object::DisplayStateCbAtNextLoop(void *, void *)]' 0x10058b74 to fire periodically every 500 ms
    30/3/2012 15:37:13 [DEBUG0] PHN: SIP: connection 15 terminated
    30/3/2012 15:37:13 [INFO ] PHN: RTP: set_audible(1/1 -> 0/0): RP15 changed=1, update_req=0, adr=123.456.789.123:9046
    30/3/2012 15:37:13 [INFO ] PHN: RTP: set_audible(0/0 -> 0/0): RP15 changed=0, update_req=0, adr=123.456.789.123:9046
    30/3/2012 15:37:13 [INFO ] PHN: rtp_port::release_turn_resources(), bw_check_enabled=0, rtp relay port=0, rtcp relay port=0
    30/3/2012 15:37:13 [INFO ] PHN: RTP: set_audible(0/0 -> 0/0): RP15 changed=0, update_req=0, adr=123.456.789.123:9046
    30/3/2012 15:37:13 [INFO ] PHN: ETH STATUS: unchanged up
    30/3/2012 15:37:13 [DEBUG0] GUI: GUI: set_led: nr 1 id 1 state 0
    30/3/2012 15:37:13 [DEBUG2] PHN: TMR: 'gui_state.cpp (736): [static void gui_object::DisplayStateCbAtNextLoop(void *, void *)]' 0x10058b24
    30/3/2012 15:37:13 [DEBUG2] PHN: SIP: recv 200 INVITE (2: 4f75c510f353-w79oibxqbfbp) <- tcp:123.456.789.123:5060
    30/3/2012 15:37:15 [DEBUG2] PHN: TMR: 'minibrowser_parser.cpp (1221): [void gui_object::MbParserFetch(const xml_tree &)]' 0x10059020
    30/3/2012 15:37:15 [NOTICE] PHN: Fetching URL: snom://mb_exit
    30/3/2012 15:37:15 [INFO ] GUI: Deleting channel 15, identity 1, state closed, total connection count 1
    30/3/2012 15:37:15 [DEBUG2] GUI: Deleting connected_call
    30/3/2012 15:37:15 [DEBUG2] GUI: Line 15: Destruct
    30/3/2012 15:37:15 [DEBUG2] PHN: Goto Best State from 'Terminated' (15), force 1
    30/3/2012 15:37:15 [DEBUG2] PHN: TMR: setting up 'selection_list.cpp (2070): [void gui_object::SelectionListCtx::draw(bool = false)]' 0x1005b26c to fire periodically every 1000 ms
    30/3/2012 15:37:15 [DEBUG2] PHN: TMR: setting up 'gui_state.cpp (738): [static void gui_object::DisplayStateCbAtNextLoop(void *, void *)]' 0x10058b74 to fire periodically every 500 ms
    30/3/2012 15:37:15 [DEBUG2] PHN: TMR: 'gui_state.cpp (736): [static void gui_object::DisplayStateCbAtNextLoop(void *, void *)]' 0x10058b24
    30/3/2012 15:37:15 [DEBUG2] PHN: send lldp advertisment
    30/3/2012 15:37:16 [DEBUG2] PHN: TMR: 'gui_state_idle.cpp (2138): [void gui_object::enter_on_idle()]' 0x10058dcc
    30/3/2012 15:37:17 [DEBUG2] PHN: SIP: recv 200 INVITE (2: 4f75c510f353-w79oibxqbfbp) <- tcp:123.456.789.123:5060
    30/3/2012 15:37:17 [INFO ] PHN: ETH STATUS: unchanged up




    Sent to tls:123.456.789.123:5061 at 30/3/2012 15:32:24:225 (1206 bytes):

    INVITE sip:103@voip.testdomain.com;user=phone SIP/2.0
    Via: SIP/2.0/TLS 192.168.1.61:2136;branch=z9hG4bK-uxi0z3yh0o7g;rport
    From: "User" <sip:101@voip.testdomain.com>;tag=qxukzfzsu2
    To: <sip:103@voip.testdomain.com;user=phone>
    Call-ID: 4f75c3f7ee5b-66wik1e4d7iv
    CSeq: 1 INVITE
    Max-Forwards: 70
    Contact: <sip:101@192.168.1.61:2136;transport=tls;line=dnb9r8vc>;reg-id=1
    X-Serialnumber: 0004133BD122
    P-Key-Flags: keys="3"
    User-Agent: snom300/8.7.3.7
    Accept: application/sdp
    Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
    Allow-Events: talk, hold, refer, call-info
    Supported: timer, 100rel, replaces, from-change
    Session-Expires: 3600;refresher=uas
    Min-SE: 90
    Content-Type: application/sdp
    Content-Length: 421

    v=0
    o=root 1531094312 1531094312 IN IP4 86.180.168.114
    s=call
    c=IN IP4 86.180.168.114
    t=0 0
    m=audio 58008 RTP/AVP 9 0 8 3 99 108 18 101
    a=rtpmap:9 G722/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:99 G726-32/8000
    a=rtpmap:108 AAL2-G726-32/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    a=rtcp:61091
    a=sendrecv
    Sent to tcp:123.456.789.123:5060 at 30/3/2012 15:32:24:344 (1206 bytes):

    INVITE sip:103@voip.testdomain.com;user=phone SIP/2.0
    Via: SIP/2.0/TCP 192.168.1.61:2134;branch=z9hG4bK-uxi0z3yh0o7g;rport
    From: "User" <sip:101@voip.testdomain.com>;tag=qxukzfzsu2
    To: <sip:103@voip.testdomain.com;user=phone>
    Call-ID: 4f75c3f7ee5b-66wik1e4d7iv
    CSeq: 1 INVITE
    Max-Forwards: 70
    Contact: <sip:101@192.168.1.61:2134;transport=tcp;line=dnb9r8vc>;reg-id=1
    X-Serialnumber: 0004133BD122
    P-Key-Flags: keys="3"
    User-Agent: snom300/8.7.3.7
    Accept: application/sdp
    Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
    Allow-Events: talk, hold, refer, call-info
    Supported: timer, 100rel, replaces, from-change
    Session-Expires: 3600;refresher=uas
    Min-SE: 90
    Content-Type: application/sdp
    Content-Length: 421

    v=0
    o=root 1531094312 1531094312 IN IP4 86.180.168.114
    s=call
    c=IN IP4 86.180.168.114
    t=0 0
    m=audio 58008 RTP/AVP 9 0 8 3 99 108 18 101
    a=rtpmap:9 G722/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:99 G726-32/8000
    a=rtpmap:108 AAL2-G726-32/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    a=rtcp:61091
    a=sendrecv
    Received from tcp:123.456.789.123:5060 at 30/3/2012 15:32:24:646 (506 bytes):

    SIP/2.0 407 Proxy Authentication Required
    Via: SIP/2.0/TCP 192.168.1.61:2134;branch=z9hG4bK-uxi0z3yh0o7g;rport=57987;received=86.180.168.114
    Proxy-Authenticate: Digest nonce="414d535c058654f900:11f403b2ac256e4bf727f4da5a763ba1",algorithm=MD5,realm="3CXPhoneSystem"
    To: <sip:103@voip.testdomain.com;user=phone>;tag=a9410b4e
    From: "User"<sip:101@voip.testdomain.com>;tag=qxukzfzsu2
    Call-ID: 4f75c3f7ee5b-66wik1e4d7iv
    CSeq: 1 INVITE
    User-Agent: 3CXPhoneSystem 10.0.23053.0
    Content-Length: 0

    Sent to tcp:123.456.789.123:5060 at 30/3/2012 15:32:24:670 (412 bytes):

    ACK sip:103@voip.testdomain.com;user=phone SIP/2.0
    Via: SIP/2.0/TCP 192.168.1.61:2134;branch=z9hG4bK-uxi0z3yh0o7g;rport
    From: "User" <sip:101@voip.testdomain.com>;tag=qxukzfzsu2
    To: <sip:103@voip.testdomain.com;user=phone>;tag=a9410b4e
    Call-ID: 4f75c3f7ee5b-66wik1e4d7iv
    CSeq: 1 ACK
    Max-Forwards: 70
    Contact: <sip:101@192.168.1.61:2134;transport=tcp;line=dnb9r8vc>;reg-id=1
    Content-Length: 0

    Sent to tcp:123.456.789.123:5060 at 30/3/2012 15:32:24:707 (1438 bytes):

    INVITE sip:103@voip.testdomain.com;user=phone SIP/2.0
    Via: SIP/2.0/TCP 192.168.1.61:2134;branch=z9hG4bK-pjmej191wzr4;rport
    From: "User" <sip:101@voip.testdomain.com>;tag=qxukzfzsu2
    To: <sip:103@voip.testdomain.com;user=phone>
    Call-ID: 4f75c3f7ee5b-66wik1e4d7iv
    CSeq: 2 INVITE
    Max-Forwards: 70
    Contact: <sip:101@192.168.1.61:2134;transport=tcp;line=dnb9r8vc>;reg-id=1
    X-Serialnumber: 0004133BD122
    P-Key-Flags: keys="3"
    User-Agent: snom300/8.7.3.7
    Accept: application/sdp
    Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
    Allow-Events: talk, hold, refer, call-info
    Supported: timer, 100rel, replaces, from-change
    Session-Expires: 3600;refresher=uas
    Min-SE: 90
    Proxy-Authorization: Digest username="101",realm="3CXPhoneSystem",nonce="414d535c058654f900:11f403b2ac256e4bf727f4da5a763ba1",uri="sip:103@voip.testdomain.com;user=phone",response="0c7ae62baf79e05b55f73dbb497d8162",algorithm=MD5
    Content-Type: application/sdp
    Content-Length: 421

    v=0
    o=root 1531094312 1531094312 IN IP4 86.180.168.114
    s=call
    c=IN IP4 86.180.168.114
    t=0 0
    m=audio 58008 RTP/AVP 9 0 8 3 99 108 18 101
    a=rtpmap:9 G722/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:99 G726-32/8000
    a=rtpmap:108 AAL2-G726-32/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    a=rtcp:61091
    a=sendrecv
    Received from tcp:123.456.789.123:5060 at 30/3/2012 15:32:24:981 (305 bytes):

    SIP/2.0 100 Trying
    Via: SIP/2.0/TCP 192.168.1.61:2134;branch=z9hG4bK-pjmej191wzr4;rport=57987;received=86.180.168.114
    To: <sip:103@voip.testdomain.com;user=phone>
    From: "User" <sip:101@voip.testdomain.com>;tag=qxukzfzsu2
    Call-ID: 4f75c3f7ee5b-66wik1e4d7iv
    CSeq: 2 INVITE
    Content-Length: 0

    Received from tcp:123.456.789.123:5060 at 30/3/2012 15:32:27:375 (412 bytes):

    SIP/2.0 180 Ringing
    Via: SIP/2.0/TCP 192.168.1.61:2134;branch=z9hG4bK-pjmej191wzr4;rport=57987;received=86.180.168.114
    Contact: <sip:103@voip.testdomain.com;user=phone>
    To: <sip:103@voip.testdomain.com;user=phone>;tag=95737a2e
    From: "User"<sip:101@voip.testdomain.com>;tag=qxukzfzsu2
    Call-ID: 4f75c3f7ee5b-66wik1e4d7iv
    CSeq: 2 INVITE
    User-Agent: 3CXPhoneSystem 10.0.23053.0
    Content-Length: 0

    Received from tcp:123.456.789.123:5060 at 30/3/2012 15:32:29:880 (922 bytes):

    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 192.168.1.61:2134;branch=z9hG4bK-pjmej191wzr4;rport=57987;received=86.180.168.114
    Contact: <sip:103@192.168.1.100:5060;transport=TCP>
    To: <sip:103@voip.testdomain.com;user=phone>;tag=95737a2e
    From: "User"<sip:101@voip.testdomain.com>;tag=qxukzfzsu2
    Call-ID: 4f75c3f7ee5b-66wik1e4d7iv
    CSeq: 2 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    Content-Type: application/sdp
    Supported: replaces
    User-Agent: 3CXPhoneSystem 10.0.23053.0
    Content-Length: 368

    v=0
    o=3cxPS 443438596096 486891585537 IN IP4 123.456.789.123
    s=3cxPS Audio call
    c=IN IP4 123.456.789.123
    t=0 0
    m=audio 9042 RTP/AVP 9 0 8 3 18 101
    c=IN IP4 123.456.789.123
    a=rtpmap:9 G722/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=sendrecv
    Sent to TCP:192.168.1.100:5060 at 30/3/2012 15:32:30:108 (412 bytes):

    ACK sip:103@192.168.1.100:5060;transport=TCP SIP/2.0
    Via: SIP/2.0/TCP 192.168.1.61:2137;branch=z9hG4bK-moof1rv9ad19;rport
    From: "User" <sip:101@voip.testdomain.com>;tag=qxukzfzsu2
    To: <sip:103@voip.testdomain.com;user=phone>;tag=95737a2e
    Call-ID: 4f75c3f7ee5b-66wik1e4d7iv
    CSeq: 2 ACK
    Max-Forwards: 70
    Contact: <sip:101@192.168.1.61:2137;transport=TCP;line=dnb9r8vc>;reg-id=1
    Content-Length: 0

    Received from tcp:123.456.789.123:5060 at 30/3/2012 15:32:30:564 (922 bytes):

    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 192.168.1.61:2134;branch=z9hG4bK-pjmej191wzr4;rport=57987;received=86.180.168.114
    Contact: <sip:103@192.168.1.100:5060;transport=TCP>
    To: <sip:103@voip.testdomain.com;user=phone>;tag=95737a2e
    From: "User"<sip:101@voip.testdomain.com>;tag=qxukzfzsu2
    Call-ID: 4f75c3f7ee5b-66wik1e4d7iv
    CSeq: 2 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    Content-Type: application/sdp
    Supported: replaces
    User-Agent: 3CXPhoneSystem 10.0.23053.0
    Content-Length: 368

    v=0
    o=3cxPS 443438596096 486891585537 IN IP4 123.456.789.123
    s=3cxPS Audio call
    c=IN IP4 123.456.789.123
    t=0 0
    m=audio 9042 RTP/AVP 9 0 8 3 18 101
    c=IN IP4 123.456.789.123
    a=rtpmap:9 G722/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=sendrecv
    Received from tcp:123.456.789.123:5060 at 30/3/2012 15:32:31:639 (922 bytes):

    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 192.168.1.61:2134;branch=z9hG4bK-pjmej191wzr4;rport=57987;received=86.180.168.114
    Contact: <sip:103@192.168.1.100:5060;transport=TCP>
    To: <sip:103@voip.testdomain.com;user=phone>;tag=95737a2e
    From: "User"<sip:101@voip.testdomain.com>;tag=qxukzfzsu2
    Call-ID: 4f75c3f7ee5b-66wik1e4d7iv
    CSeq: 2 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    Content-Type: application/sdp
    Supported: replaces
    User-Agent: 3CXPhoneSystem 10.0.23053.0
    Content-Length: 368

    v=0
    o=3cxPS 443438596096 486891585537 IN IP4 123.456.789.123
    s=3cxPS Audio call
    c=IN IP4 123.456.789.123
    t=0 0
    m=audio 9042 RTP/AVP 9 0 8 3 18 101
    c=IN IP4 123.456.789.123
    a=rtpmap:9 G722/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=sendrecv
    Received from tcp:123.456.789.123:5060 at 30/3/2012 15:32:33:580 (922 bytes):

    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 192.168.1.61:2134;branch=z9hG4bK-pjmej191wzr4;rport=57987;received=86.180.168.114
    Contact: <sip:103@192.168.1.100:5060;transport=TCP>
    To: <sip:103@voip.testdomain.com;user=phone>;tag=95737a2e
    From: "User"<sip:101@voip.testdomain.com>;tag=qxukzfzsu2
    Call-ID: 4f75c3f7ee5b-66wik1e4d7iv
    CSeq: 2 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    Content-Type: application/sdp
    Supported: replaces
    User-Agent: 3CXPhoneSystem 10.0.23053.0
    Content-Length: 368

    v=0
    o=3cxPS 443438596096 486891585537 IN IP4 123.456.789.123
    s=3cxPS Audio call
    c=IN IP4 123.456.789.123
    t=0 0
    m=audio 9042 RTP/AVP 9 0 8 3 18 101
    c=IN IP4 123.456.789.123
    a=rtpmap:9 G722/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=sendrecv
    Received from tcp:123.456.789.123:5060 at 30/3/2012 15:32:37:677 (922 bytes):

    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 192.168.1.61:2134;branch=z9hG4bK-pjmej191wzr4;rport=57987;received=86.180.168.114
    Contact: <sip:103@192.168.1.100:5060;transport=TCP>
    To: <sip:103@voip.testdomain.com;user=phone>;tag=95737a2e
    From: "User"<sip:101@voip.testdomain.com>;tag=qxukzfzsu2
    Call-ID: 4f75c3f7ee5b-66wik1e4d7iv
    CSeq: 2 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    Content-Type: application/sdp
    Supported: replaces
    User-Agent: 3CXPhoneSystem 10.0.23053.0
    Content-Length: 368

    v=0
    o=3cxPS 443438596096 486891585537 IN IP4 123.456.789.123
    s=3cxPS Audio call
    c=IN IP4 123.456.789.123
    t=0 0
    m=audio 9042 RTP/AVP 9 0 8 3 18 101
    c=IN IP4 123.456.789.123
    a=rtpmap:9 G722/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=sendrecv


    Thanks
    Kev
     
  2. albertc

    Joined:
    Nov 21, 2011
    Messages:
    89
    Likes Received:
    0
    Looks like you are using TCP as the transport protocol on your Snom 300. Go into your Snom phone web config. Within the Line Configuration page under Transport, change it from Auto to UDP. See how that works.
     
  3. cdstech

    Joined:
    Mar 30, 2012
    Messages:
    2
    Likes Received:
    0
    Hi

    Thanks for your reply.

    I can't find the option you described, I don't have a Line Configuration or Transport option, my firmware Version is snom300-SIP 8.7.3.7

    I can find a setting under Identity 1, RTP called Media Transport Offer: which is set to UDP, is that what you mean ?

    Thanks
    Kev
     
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