[Oct 21 00:03:28] VERBOSE[21327] logger.c:
JABBER: gtalk_account INCOMING:
[Oct 21 00:03:28] DEBUG[21327] res_jabber.c: JABBER: I Dont have an IQ!!!
[Oct 21 00:03:28] DEBUG[21327] chan_gtalk.c: The client is guest for alloc
[Oct 21 00:03:28] VERBOSE[21327] logger.c:
JABBER: gtalk_account OUTGOING:
[Oct 21 00:03:28] VERBOSE[21327] logger.c:
JABBER: gtalk_account OUTGOING:
[Oct 21 00:03:28] VERBOSE[21327] logger.c:
JABBER: gtalk_account OUTGOING:
[Oct 21 00:03:28] VERBOSE[21383] logger.c: == Starting Gtalk/gtalk_source-3aec at google-in,ast_server@gmail.com,1 failed so falling back to exten 's'
[Oct 21 00:03:28] DEBUG[21383] pbx.c: Launching 'Answer'
[Oct 21 00:03:28] VERBOSE[21383] logger.c: -- Executing [s@google-in:1] Answer("Gtalk/gtalk_source-3aec", "") in new stack
[Oct 21 00:03:28] DEBUG[21383] chan_gtalk.c: Answer!
[Oct 21 00:03:28] VERBOSE[21383] logger.c:
JABBER: gtalk_account OUTGOING:
[Oct 21 00:03:28] DEBUG[21308] channel.c: Avoiding initial deadlock for channel '0x9b43ab0'
[Oct 21 00:03:28] DEBUG[21383] pbx.c: Launching 'Dial'
[Oct 21 00:03:28] VERBOSE[21383] logger.c: -- Executing [s@google-in:2] Dial("Gtalk/gtalk_source-3aec", "SIP/18005551212@voipgateway") in new stack
[Oct 21 00:03:28] DEBUG[21383] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw)
[Oct 21 00:03:28] DEBUG[21383] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP)
[Oct 21 00:03:28] DEBUG[21383] chan_sip.c: Setting NAT on RTP to Off
[Oct 21 00:03:28] DEBUG[21383] rtp.c: Channel 'Gtalk/gtalk_source-3aec' has no RTP, not doing anything
[Oct 21 00:03:28] DEBUG[21383] channel.c: Not copying variable STACK-google-in-s-2.
[Oct 21 00:03:28] DEBUG[21383] channel.c: Not copying variable STACK-google-in-s-1.
[Oct 21 00:03:28] DEBUG[21384] app_queue.c: Device 'Gtalk/gtalk_source' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Oct 21 00:03:28] DEBUG[21383] chan_sip.c: Outgoing Call for 18005551212
[Oct 21 00:03:28] DEBUG[21383] chan_sip.c: Our T38 capability (0), joint T38 capability (0)
[Oct 21 00:03:28] VERBOSE[21383] logger.c: -- Called 18005551212@voipgateway
[Oct 21 00:03:28] DEBUG[21385] app_queue.c: Device 'Gtalk/gtalk_source' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Oct 21 00:03:28] DEBUG[21330] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '57bb5f6939a545a17aa6b00a28768031@210.101.62.52' Request 102: Found
[Oct 21 00:03:28] VERBOSE[21327] logger.c:
JABBER: gtalk_account INCOMING:
[Oct 21 00:03:28] DEBUG[21327] res_jabber.c: JABBER: I Dont have an IQ!!!
[Oct 21 00:03:28] VERBOSE[21327] logger.c:
JABBER: gtalk_account OUTGOING:
[Oct 21 00:03:28] VERBOSE[21327] logger.c:
JABBER: gtalk_account INCOMING:
[Oct 21 00:03:28] DEBUG[21327] res_jabber.c: JABBER: I Dont have an IQ!!!
[Oct 21 00:03:28] VERBOSE[21327] logger.c:
JABBER: gtalk_account OUTGOING:
[Oct 21 00:03:28] VERBOSE[21327] logger.c:
JABBER: gtalk_account INCOMING:
[Oct 21 00:03:28] DEBUG[21327] res_jabber.c: JABBER: I Dont have an IQ!!!
[Oct 21 00:03:28] VERBOSE[21327] logger.c:
JABBER: gtalk_account INCOMING:
[Oct 21 00:03:28] DEBUG[21327] res_jabber.c: JABBER: I Dont have an IQ!!!
[Oct 21 00:03:28] VERBOSE[21327] logger.c:
JABBER: gtalk_account INCOMING:
[Oct 21 00:03:28] DEBUG[21327] res_jabber.c: JABBER: I Dont have an IQ!!!
[Oct 21 00:03:29] NOTICE[21383] rtp.c: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 67.82.63.94
[Oct 21 00:03:34] DEBUG[21330] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '57bb5f6939a545a17aa6b00a28768031@210.101.62.52' Request 102: Found
[Oct 21 00:03:34] VERBOSE[21383] logger.c: -- SIP/voipgateway-09b406f0 is making progress passing it to Gtalk/gtalk_source-3aec
[Oct 21 00:03:34] DEBUG[21383] rtp.c: Channel 'Gtalk/gtalk_source-3aec' has no RTP, not doing anything
[Oct 21 00:03:34] NOTICE[21383] chan_gtalk.c: Don't know how to indicate condition '14'
[Oct 21 00:03:34] DEBUG[21383] rtp.c: Ooh, format changed from unknown to ulaw
[Oct 21 00:03:34] DEBUG[21383] rtp.c: Created smoother: format: 4 ms: 20 len: 160
[Oct 21 00:03:35] DEBUG[21330] chan_sip.c: Allocating new SIP dialog for 10e6b5ae7eab823a57673f5875a1a96f@210.101.62.52 - REGISTER (No RTP)
[Oct 21 00:03:35] DEBUG[21330] chan_sip.c: Scheduled a registration timeout for outbound.vitelity.net id #53
[Oct 21 00:03:35] DEBUG[21330] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '10e6b5ae7eab823a57673f5875a1a96f@210.101.62.52' Request 110: Found
[Oct 21 00:03:35] DEBUG[21330] chan_sip.c: Stopping retransmission on '10e6b5ae7eab823a57673f5875a1a96f@210.101.62.52' of Request 110: Match Not Found
[Oct 21 00:03:35] DEBUG[21330] chan_sip.c: Initializing already initialized SIP dialog 10e6b5ae7eab823a57673f5875a1a96f@210.101.62.52 (presumably reinvite)
[Oct 21 00:03:35] DEBUG[21330] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '10e6b5ae7eab823a57673f5875a1a96f@210.101.62.52' Request 111: Found
[Oct 21 00:03:35] DEBUG[21330] chan_sip.c: Stopping retransmission on '10e6b5ae7eab823a57673f5875a1a96f@210.101.62.52' of Request 111: Match Not Found
[Oct 21 00:03:35] DEBUG[21330] chan_sip.c: Registration successful
[Oct 21 00:03:35] DEBUG[21330] chan_sip.c: Cancelling timeout 53
[Oct 21 00:03:35] DEBUG[21383] rtp.c: Ooh, format changed from unknown to ulaw
[Oct 21 00:03:35] DEBUG[21383] rtp.c: Created smoother: format: 4 ms: 20 len: 160
[Oct 21 00:03:35] DEBUG[21330] chan_sip.c: Allocating new SIP dialog for 0bb9fb0b5023d48a344620d90a5de950@210.101.62.52 - REGISTER (No RTP)
[Oct 21 00:03:35] DEBUG[21330] chan_sip.c: Scheduled a registration timeout for inbound.vitelity.net id #58
[Oct 21 00:03:36] DEBUG[21330] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0bb9fb0b5023d48a344620d90a5de950@210.101.62.52' Request 110: Found
[Oct 21 00:03:36] DEBUG[21330] chan_sip.c: Stopping retransmission on '0bb9fb0b5023d48a344620d90a5de950@210.101.62.52' of Request 110: Match Not Found
[Oct 21 00:03:36] DEBUG[21330] chan_sip.c: Initializing already initialized SIP dialog 0bb9fb0b5023d48a344620d90a5de950@210.101.62.52 (presumably reinvite)
[Oct 21 00:03:36] DEBUG[21330] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0bb9fb0b5023d48a344620d90a5de950@210.101.62.52' Request 111: Found
[Oct 21 00:03:36] DEBUG[21330] chan_sip.c: Stopping retransmission on '0bb9fb0b5023d48a344620d90a5de950@210.101.62.52' of Request 111: Match Not Found
[Oct 21 00:03:36] DEBUG[21330] chan_sip.c: Registration successful
[Oct 21 00:03:36] DEBUG[21330] chan_sip.c: Cancelling timeout 58
[Oct 21 00:03:37] DEBUG[21330] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '57bb5f6939a545a17aa6b00a28768031@210.101.62.52' Request 102: Found
[Oct 21 00:03:37] VERBOSE[21383] logger.c: -- SIP/voipgateway-09b406f0 is ringing
[Oct 21 00:03:37] DEBUG[21383] rtp.c: Channel 'Gtalk/gtalk_source-3aec' has no RTP, not doing anything
[Oct 21 00:03:37] NOTICE[21383] chan_gtalk.c: Don't know how to indicate condition '3'
[Oct 21 00:03:37] DEBUG[21383] channel.c: Driver for channel 'Gtalk/gtalk_source-3aec' does not support indication 3, emulating it
[Oct 21 00:03:37] DEBUG[21383] channel.c: Set channel Gtalk/gtalk_source-3aec to write format slin
[Oct 21 00:03:37] DEBUG[21387] app_queue.c: Device 'SIP/voipgateway' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Oct 21 00:03:37] DEBUG[21383] channel.c: Set channel Gtalk/gtalk_source-3aec to write format ulaw
[Oct 21 00:03:40] DEBUG[21330] chan_sip.c: Acked pending invite 102
[Oct 21 00:03:40] DEBUG[21330] chan_sip.c: Stopping retransmission on '57bb5f6939a545a17aa6b00a28768031@210.101.62.52' of Request 102: Match Not Found
[Oct 21 00:03:40] VERBOSE[21383] logger.c: -- SIP/voipgateway-09b406f0 answered Gtalk/gtalk_source-3aec
[Oct 21 00:03:40] DEBUG[21383] rtp.c: Channel 'Gtalk/gtalk_source-3aec' has no RTP, not doing anything
[Oct 21 00:03:40] NOTICE[21383] chan_gtalk.c: Don't know how to indicate condition '-1'
[Oct 21 00:03:40] WARNING[21383] rtp.c: Can't find native functions for channel 'Gtalk/gtalk_source-3aec'
[Oct 21 00:03:40] VERBOSE[21383] logger.c: -- Native bridging Gtalk/gtalk_source-3aec and SIP/voipgateway-09b406f0 ended
[Oct 21 00:03:40] DEBUG[21389] app_queue.c: Device 'SIP/voipgateway' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
At ~4:00
Asterisk server -> Gtalk client off (seem to hear comfort noise on client)
Phone <- voip_gateway <- Asterisk server <- Gtalk client still ok