chan_motif nimbuzz calling issue

hello dear friends
i am facing a problum and want solution any ideas wellcome
i have installed centos 6.3
asterisk 11.0.1
and i am able to place call and recieve to my friends when my friends are comes online via {pc,s no metter laptop or desktop} gmail and nimbuzz clients calls in and out working fine
but when my friends comes online using thier mobile nimbuzz clients they call me and my phone also rings but when i answer incoming calls call goes ended,and i want to recieve and place calls to my friends on their nimbuzz clients, please help me
here are my configs
xmpp.conf
[11111]
type=client
serverhost=talk.google.com ;
;pubsub_node=pubsub.astjab.org ;
username=acount1@gmail.com ; U
secret=xxxxxxx ;
priority=1 ; R
port=5222 ; Po
usetls=yes ; Use tl
usesasl=yes ; Use
;buddy=mogorman@

sip.conf
[200]
type=friend
host=dynamic
username=200
secret=xxxxx
context=demo
canreinvite=no

motif.conf
[xxxxxx]
disallow=all ;
allow=ulaw ;
context=incoming- 11111;
connection=xxxxxx

extensions.conf
[incoming-11111]
exten => s,1,NoOp()
exten => s,n,Dial(SIP/200,90,D(:1))
exten => s,n(end),Hangup()
include => demo ;

and these are logs
dropped incoming call from nimbuzz mobile client
Executing [s@incoming-11111:1] NoOp(“Motif/xxxxxx-30e9”, “”) in new stack
– Executing [s@incoming-11111:2] Dial(“Motif/xxxxxx-30e9”, “SIP/200,90,D(:1)”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/200
– SIP/200-00000000 is ringing
– SIP/200-00000000 answered Motif/xxxxxx-30e9
– Sending DTMF ‘1’ to the calling party.
– Locally bridging Motif/xxxxxx-30e9 and SIP/200-00000000
== Spawn extension (incoming-11111, s, 2) exited non-zero on ‘Motif/xxxxxx-30e9’

and this is successfull call nimbuzz desktop client

Executing [s@incoming-11111:1] NoOp(“Motif/xxxxxx-10ba”, “”) in new stack
– Executing [s@incoming-11111:2] Dial(“Motif/xxxxxx-10ba”, “SIP/200,90,D(:1)”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/200
– SIP/200-00000001 is ringing
– SIP/200-00000001 is ringing
– SIP/200-00000001 is ringing
– SIP/200-00000001 answered Motif/xxxxxx-10ba
– Sending DTMF ‘1’ to the calling party.
– Locally bridging Motif/xxxxxx-10ba and SIP/200-00000001
== Spawn extension (incoming-11111, s, 2) exited non-zero on 'Motif/xxxxxx-10ba’
i have edited some email addreses in config and log files
any ideas welcome

This is detailed log of a failed call

[Dec 8 12:03:33] DEBUG[3523]: logger.c:1285 ast_create_callid: CALL_ID [C-00000000] created by thread.
[Dec 8 12:03:33] DEBUG[3523]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting ‘0.0.0.0’ into…
[Dec 8 12:03:33] DEBUG[3523]: netsock2.c:192 ast_sockaddr_split_hostport: …host ‘0.0.0.0’ and port ‘’.
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:283 ast_rtp_instance_new: Using engine ‘asterisk’ for RTP instance ‘0xb732b164’
[Dec 8 12:03:33] DEBUG[3523]: res_rtp_asterisk.c:1732 ast_rtp_new: Allocated port 19592 for RTP instance ‘0xb732b164’
[Dec 8 12:03:33] DEBUG[3523]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting ‘192.168.15.208’ into…
[Dec 8 12:03:33] DEBUG[3523]: netsock2.c:192 ast_sockaddr_split_hostport: …host ‘192.168.15.208’ and port ‘’.
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:292 ast_rtp_instance_new: RTP instance ‘0xb732b164’ is setup and ready to go
[Dec 8 12:03:33] DEBUG[3523]: res_rtp_asterisk.c:3848 ast_rtp_prop_set: Setup RTCP on RTP instance ‘0xb732b164’
[Dec 8 12:03:33] DEBUG[3523]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting ‘192.168.15.208’ into…
[Dec 8 12:03:33] DEBUG[3523]: netsock2.c:192 ast_sockaddr_split_hostport: …host ‘192.168.15.208’ and port ‘’.
[Dec 8 12:03:33] DEBUG[3517]: devicestate.c:342 _ast_device_state: No provider found, checking channel drivers for Motif -xxxxxx
[Dec 8 12:03:33] DEBUG[3517]: devicestate.c:460 do_state_change: Changing state for Motif/xxxxxx - state 2 (In use)
[Dec 8 12:03:33] DEBUG[3517]: devicestate.c:440 devstate_event: device ‘Motif/xxxxxx’ state ‘2’
[Dec 8 12:03:33] DEBUG[3556]: app_queue.c:1751 handle_statechange: Device ‘Motif/xxxxxx’ changed to state ‘2’ (In use) but we don’t care because they’re not a member of any queue.
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 102 based on m type on 0xb723e620
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 100 based on m type on 0xb723e620
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0xb723e620
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0xb723e620
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0xb723e620
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 106 based on m type on 0xb723e620
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 0 from 0xb723e620 to 0xb732b310
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 8 from 0xb723e620 to 0xb732b310
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 100 from 0xb723e620 to 0xb732b310
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0xb723e620 to 0xb732b310
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 102 from 0xb723e620 to 0xb732b310
[Dec 8 12:03:33] DEBUG[3523]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 106 from 0xb723e620 to 0xb732b310
[Dec 8 12:03:33] DEBUG[3523]: res_xmpp.c:3508 xmpp_client_receive: XML parsing successful
[Dec 8 12:03:33] DEBUG[3523][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:33] DEBUG[3523][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting ‘195.211.49.73’ into…
[Dec 8 12:03:33] DEBUG[3523][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: …host ‘195.211.49.73’ and port ‘’.
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: pbx.c:4410 pbx_extension_helper: Launching ‘NoOp’
– Executing [s@incoming-11111:1] NoOp(“Motif/xxxxxx-3a32”, “”) in new stack
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: pbx.c:4410 pbx_extension_helper: Launching ‘Dial’
– Executing [s@incoming-11111:2] Dial(“Motif/xxxxxx-3a32”, “SIP/200,90,D(:1)”) in new stack
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:29048 sip_request_call: Asked to create a SIP channel with formats: (ulaw)
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:8401 sip_alloc: Allocating new SIP dialog for 20fe81b17dabebbf26ac47f52f39c768@192.168.15.208:5060 - INVITE (No RTP)
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: rtp_engine.c:283 ast_rtp_instance_new: Using engine ‘asterisk’ for RTP instance ‘0xb750c73c’
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: res_rtp_asterisk.c:1732 ast_rtp_new: Allocated port 14872 for RTP instance ‘0xb750c73c’
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting ‘192.168.15.208’ into…
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: …host ‘192.168.15.208’ and port ‘’.
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: rtp_engine.c:292 ast_rtp_instance_new: RTP instance ‘0xb750c73c’ is setup and ready to go
[Dec 8 12:03:33] DEBUG[3523][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:33] DEBUG[3523]: res_xmpp.c:3508 xmpp_client_receive: XML parsing successful
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: res_rtp_asterisk.c:3848 ast_rtp_prop_set: Setup RTCP on RTP instance ‘0xb750c73c’
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting ‘192.168.15.208’ into…
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: …host ‘192.168.15.208’ and port ‘’.
== Using SIP RTP CoS mark 5
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:5420 do_setnat: Setting NAT on RTP to Off
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: acl.c:979 ast_ouraddrfor: For destination ‘192.168.15.200’, our source address is ‘192.168.15.208’.
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:3744 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.15.208:5060
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: format_pref.c:339 ast_codec_choose: Could not find preferred codec - Going for the best codec
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:7589 sip_new: *** Our native formats are (ulaw)
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:7590 sip_new: *** Joint capabilities are (ulaw)
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:7591 sip_new: *** Our capabilities are (gsm|ulaw|alaw|h263|testlaw)
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:7592 sip_new: *** AST_CODEC_CHOOSE formats are ulaw
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:7594 sip_new: *** Our preferred formats from the incoming channel are (ulaw)
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:7620 sip_new: This channel will not be able to handle video.
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: channel_internal_api.c:860 ast_channel_callid_set: Channel Call ID changing from [C-00000000] to [C-00000000]
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: channel.c:6473 ast_channel_inherit_variables: Not copying variable DIALEDTIME.
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: channel.c:6473 ast_channel_inherit_variables: Not copying variable ANSWEREDTIME.
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: channel.c:6473 ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME.
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: channel.c:6473 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER.
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: channel.c:6473 ast_channel_inherit_variables: Not copying variable DIALSTATUS.
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:6028 sip_call: Outgoing Call for 200
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:6352 update_call_counter: Updating call counter for outgoing call
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:12670 add_sdp: This call needs video offers, but there’s no video support enabled!
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:12718 add_sdp: ** Our capability: (gsm|ulaw|alaw|h263|testlaw) Video flag: False Text flag: False
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:12719 add_sdp: ** Our prefcodec: (ulaw)
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:12856 add_sdp: – Done with adding codecs to SDP
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:13059 add_sdp: Done building SDP. Settling with this capability: (gsm|ulaw|alaw|h263|testlaw)
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:3230 initialize_initreq: Initializing initreq for method INVITE - callid 5714f99d46e148ef1f5008f82e570a72@192.168.15.208:5060
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: chan_sip.c:3587 __sip_xmit: Trying to put ‘INVITE sip:’ onto UDP socket destined for 192.168.15.200:5070
– Called SIP/200
[Dec 8 12:03:33] DEBUG[3563][C-00000000]: res_rtp_asterisk.c:3893 ast_rtp_remote_address_set: Setting RTCP address on RTP instance ‘0xb732b164’
[Dec 8 12:03:33] DEBUG[3528]: chan_sip.c:8798 find_call: = Looking for Call ID: 5714f99d46e148ef1f5008f82e570a72@192.168.15.208:5060 (Checking To) --From tag as70de1a28 --To-tag 8087
[Dec 8 12:03:33] DEBUG[3528][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:33] DEBUG[3528][C-00000000]: chan_sip.c:4320 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on ‘5714f99d46e148ef1f5008f82e570a72@192.168.15.208:5060’ Request 102: Found
[Dec 8 12:03:33] DEBUG[3528][C-00000000]: chan_sip.c:22002 handle_response_invite: SIP response 180 to standard invite
[Dec 8 12:03:33] DEBUG[3528][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
– SIP/200-00000000 is ringing
[Dec 8 12:03:33] DEBUG[3517]: devicestate.c:342 _ast_device_state: No provider found, checking channel drivers for SIP - 200
[Dec 8 12:03:33] DEBUG[3517]: chan_sip.c:28948 sip_devicestate: Checking device state for peer 200
[Dec 8 12:03:33] DEBUG[3517]: devicestate.c:460 do_state_change: Changing state for SIP/200 - state 1 (Not in use)
[Dec 8 12:03:33] DEBUG[3517]: devicestate.c:440 devstate_event: device ‘SIP/200’ state ‘1’
[Dec 8 12:03:33] DEBUG[3556]: app_queue.c:1751 handle_statechange: Device ‘SIP/200’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Dec 8 12:03:33] DEBUG[3528]: chan_sip.c:8798 find_call: = Looking for Call ID: 5714f99d46e148ef1f5008f82e570a72@192.168.15.208:5060 (Checking To) --From tag as70de1a28 --To-tag 8087
[Dec 8 12:03:33] DEBUG[3528][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:33] DEBUG[3528][C-00000000]: chan_sip.c:4320 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on ‘5714f99d46e148ef1f5008f82e570a72@192.168.15.208:5060’ Request 102: Found
[Dec 8 12:03:33] DEBUG[3528][C-00000000]: chan_sip.c:22002 handle_response_invite: SIP response 180 to standard invite
[Dec 8 12:03:33] DEBUG[3528][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
– SIP/200-00000000 is ringing
[Dec 8 12:03:33] DEBUG[3524]: res_xmpp.c:3508 xmpp_client_receive: XML parsing successful
[Dec 8 12:03:33] DEBUG[3524]: res_xmpp.c:3508 xmpp_client_receive: XML parsing successful
[Dec 8 12:03:33] DEBUG[3523]: res_xmpp.c:3508 xmpp_client_receive: XML parsing successful
[Dec 8 12:03:34] DEBUG[3528]: chan_sip.c:8798 find_call: = Looking for Call ID: 5714f99d46e148ef1f5008f82e570a72@192.168.15.208:5060 (Checking To) --From tag as70de1a28 --To-tag 8087
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:4241 __sip_ack: Acked pending invite 102
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:4279 __sip_ack: Stopping retransmission on ‘5714f99d46e148ef1f5008f82e570a72@192.168.15.208:5060’ of Request 102: Match Found
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:22002 handle_response_invite: SIP response 200 to standard invite
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:9679 process_sdp: Processing session-level SDP v=0… UNSUPPORTED OR FAILED.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:9679 process_sdp: Processing session-level SDP o=NCHSoftware-Talk 1354947830 1354947832 IN IP4 192.168.15.200… UNSUPPORTED OR FAILED.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:9679 process_sdp: Processing session-level SDP s=Express Talk Call… UNSUPPORTED OR FAILED.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting ‘192.168.15.200’ into…
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: …host ‘192.168.15.200’ and port ‘’.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:9679 process_sdp: Processing session-level SDP c=IN IP4 192.168.15.200… OK.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:9679 process_sdp: Processing session-level SDP t=0 0… UNSUPPORTED OR FAILED.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0xb6753708
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0xb6753708
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 3 based on m type on 0xb6753708
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0xb6753708
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:10085 process_sdp: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000… OK.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:10085 process_sdp: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000… OK.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:10085 process_sdp: Processing media-level (audio) SDP a=rtpmap:3 GSM/8000… OK.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:10085 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000… OK.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:10085 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-16… UNSUPPORTED OR FAILED.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:10085 process_sdp: Processing media-level (audio) SDP a=sendrecv… OK.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:10085 process_sdp: Processing media-level (audio) SDP a=direction:active… UNSUPPORTED OR FAILED.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: res_rtp_asterisk.c:3893 ast_rtp_remote_address_set: Setting RTCP address on RTP instance ‘0xb750c73c’
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 0 from 0xb6753708 to 0xb750c8e8
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 3 from 0xb6753708 to 0xb750c8e8
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 8 from 0xb6753708 to 0xb750c8e8
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0xb6753708 to 0xb750c8e8
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: res_rtp_asterisk.c:3814 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance ‘0xb750c73c’
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:10347 process_sdp: We’re settling with these formats: (gsm|ulaw|alaw)
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:10354 process_sdp: We have an owner, now see if we need to change this call
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: format_pref.c:339 ast_codec_choose: Could not find preferred codec - Going for the best codec
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:6352 update_call_counter: Updating call counter for outgoing call
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:15774 build_route: build_route: Contact hop: sip:200@192.168.15.200:5070
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting ‘192.168.15.200:5070’ into…
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: …host ‘192.168.15.200’ and port ‘5070’.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting ‘192.168.15.200:5070’ into…
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: …host ‘192.168.15.200’ and port ‘5070’.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: chan_sip.c:3587 __sip_xmit: Trying to put ‘ACK sip:200’ onto UDP socket destined for 192.168.15.200:5070
– SIP/200-00000000 answered Motif/xxxxxx-3a32
– Sending DTMF ‘1’ to the calling party.
[Dec 8 12:03:34] DEBUG[3517]: devicestate.c:342 _ast_device_state: No provider found, checking channel drivers for SIP - 200
[Dec 8 12:03:34] DEBUG[3517]: chan_sip.c:28948 sip_devicestate: Checking device state for peer 200
[Dec 8 12:03:34] DEBUG[3517]: devicestate.c:460 do_state_change: Changing state for SIP/200 - state 1 (Not in use)
[Dec 8 12:03:34] DEBUG[3517]: devicestate.c:440 devstate_event: device ‘SIP/200’ state ‘1’
[Dec 8 12:03:34] DEBUG[3556]: app_queue.c:1751 handle_statechange: Device ‘SIP/200’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Dec 8 12:03:34] DEBUG[3528][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: res_rtp_asterisk.c:3502 ast_rtp_read: 0xb7511690 – start learning mode pass with addr = 192.168.15.200:8000
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: res_rtp_asterisk.c:1605 rtp_learning_rtp_seq_update: 0xb7511690 – probation = 4, seq = 6844
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: res_rtp_asterisk.c:3508 ast_rtp_read: 0xb7511690 – Condition for learning hasn’t exited, so reject the frame.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: res_rtp_asterisk.c:3502 ast_rtp_read: 0xb7511690 – start learning mode pass with addr = 192.168.15.200:8000
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: res_rtp_asterisk.c:1605 rtp_learning_rtp_seq_update: 0xb7511690 – probation = 3, seq = 6845
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: res_rtp_asterisk.c:3508 ast_rtp_read: 0xb7511690 – Condition for learning hasn’t exited, so reject the frame.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: res_rtp_asterisk.c:3502 ast_rtp_read: 0xb7511690 – start learning mode pass with addr = 192.168.15.200:8000
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: res_rtp_asterisk.c:1605 rtp_learning_rtp_seq_update: 0xb7511690 – probation = 2, seq = 6846
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: res_rtp_asterisk.c:3508 ast_rtp_read: 0xb7511690 – Condition for learning hasn’t exited, so reject the frame.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: res_rtp_asterisk.c:3502 ast_rtp_read: 0xb7511690 – start learning mode pass with addr = 192.168.15.200:8000
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: res_rtp_asterisk.c:1605 rtp_learning_rtp_seq_update: 0xb7511690 – probation = 1, seq = 6847
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: res_rtp_asterisk.c:3512 ast_rtp_read: 0xb7511690 – Probation Ended. Set strict_rtp_state to STRICT_RTP_CLOSED with address 192.168.15.200:8000
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:34] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:35] DEBUG[3564][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:35] DEBUG[3517]: devicestate.c:342 _ast_device_state: No provider found, checking channel drivers for Motif - xxxxxx
[Dec 8 12:03:35] DEBUG[3563][C-00000000]: features.c:4387 ast_bridge_call: bridge answer set, chan answer set
[Dec 8 12:03:35] DEBUG[3563][C-00000000]: features.c:4229 clear_dialed_interfaces: Removing dialed interfaces datastore on SIP/200-00000000 since we’re bridging
[Dec 8 12:03:35] DEBUG[3563][C-00000000]: res_rtp_asterisk.c:2085 ast_rtp_update_source: Setting the marker bit due to a source update
[Dec 8 12:03:35] DEBUG[3563][C-00000000]: res_rtp_asterisk.c:2085 ast_rtp_update_source: Setting the marker bit due to a source update
– Locally bridging Motif/xxxxxx-3a32 and SIP/200-00000000
[Dec 8 12:03:35] DEBUG[3517]: devicestate.c:460 do_state_change: Changing state for Motif/xxxxxx - state 2 (In use)
[Dec 8 12:03:35] DEBUG[3517]: devicestate.c:440 devstate_event: device ‘Motif/xxxxxx’ state ‘2’
[Dec 8 12:03:35] DEBUG[3556]: app_queue.c:1751 handle_statechange: Device ‘Motif/xxxxxx’ changed to state ‘2’ (In use) but we don’t care because they’re not a member of any queue.
[Dec 8 12:03:35] DEBUG[3523]: res_xmpp.c:3508 xmpp_client_receive: XML parsing successful
[Dec 8 12:03:36] DEBUG[3523][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:36] DEBUG[3523][C-00000000]: chan_motif.c:2374 jingle_action_session_terminate: Hanging up channel ‘Motif/xxxxxx-3a32’ due to session terminate message with cause ‘16’
[Dec 8 12:03:36] DEBUG[3523][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:36] DEBUG[3523]: res_xmpp.c:3508 xmpp_client_receive: XML parsing successful
[Dec 8 12:03:36] DEBUG[3563][C-00000000]: rtp_engine.c:1029 local_bridge_loop: rtp-engine-local-bridge: Ooh, got a hangup
[Dec 8 12:03:36] DEBUG[3563][C-00000000]: channel.c:7917 ast_channel_bridge: Returning from native bridge, channels: Motif/xxxxxx-3a32, SIP/200-00000000
[Dec 8 12:03:36] DEBUG[3563][C-00000000]: channel.c:2830 ast_hangup: Hanging up channel ‘SIP/200-00000000’
[Dec 8 12:03:36] DEBUG[3563][C-00000000]: chan_sip.c:6732 sip_hangup: Hangup call SIP/200-00000000, SIP callid 5714f99d46e148ef1f5008f82e570a72@192.168.15.208:5060
[Dec 8 12:03:36] DEBUG[3563][C-00000000]: res_rtp_asterisk.c:3893 ast_rtp_remote_address_set: Setting RTCP address on RTP instance ‘0xb750c73c’
[Dec 8 12:03:36] DEBUG[3563][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting ‘192.168.15.200:5070’ into…
[Dec 8 12:03:36] DEBUG[3563][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: …host ‘192.168.15.200’ and port ‘5070’.
[Dec 8 12:03:36] DEBUG[3563][C-00000000]: chan_sip.c:3587 __sip_xmit: Trying to put ‘BYE sip:200’ onto UDP socket destined for 192.168.15.200:5070
[Dec 8 12:03:36] DEBUG[3517]: devicestate.c:342 _ast_device_state: No provider found, checking channel drivers for SIP - 200
[Dec 8 12:03:36] DEBUG[3517]: chan_sip.c:28948 sip_devicestate: Checking device state for peer 200
[Dec 8 12:03:36] DEBUG[3517]: devicestate.c:460 do_state_change: Changing state for SIP/200 - state 1 (Not in use)
[Dec 8 12:03:36] DEBUG[3517]: devicestate.c:440 devstate_event: device ‘SIP/200’ state ‘1’
[Dec 8 12:03:36] DEBUG[3556]: app_queue.c:1751 handle_statechange: Device ‘SIP/200’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Dec 8 12:03:36] DEBUG[3563][C-00000000]: app_dial.c:3096 dial_exec_full: Exiting with DIALSTATUS=ANSWER.
[Dec 8 12:03:36] DEBUG[3563][C-00000000]: pbx.c:6090 __ast_pbx_run: Spawn extension (incoming-11111,s,2) exited non-zero on ‘Motif/xxxxxx-3a32’
== Spawn extension (incoming-11111, s, 2) exited non-zero on ‘Motif/xxxxxx-3a32’
[Dec 8 12:03:36] DEBUG[3563][C-00000000]: channel.c:2651 ast_softhangup_nolock: Soft-Hanging up channel ‘Motif/xxxxxx-3a32’
[Dec 8 12:03:36] DEBUG[3563][C-00000000]: channel.c:2830 ast_hangup: Hanging up channel ‘Motif/xxxxxx-3a32’
[Dec 8 12:03:36] DEBUG[3563][C-00000000]: rtp_engine.c:226 instance_destructor: Destroyed RTP instance ‘0xb732b164’
[Dec 8 12:03:36] DEBUG[3528]: chan_sip.c:8798 find_call: = Looking for Call ID: 5714f99d46e148ef1f5008f82e570a72@192.168.15.208:5060 (Checking To) --From tag as70de1a28 --To-tag 8087
[Dec 8 12:03:36] DEBUG[3528][C-00000000]: logger.c:1315 ast_callid_threadassoc_add: CALL_ID [C-00000000] bound to thread.
[Dec 8 12:03:36] DEBUG[3528][C-00000000]: chan_sip.c:4279 __sip_ack: Stopping retransmission on ‘5714f99d46e148ef1f5008f82e570a72@192.168.15.208:5060’ of Request 103: Match Found
[Dec 8 12:03:36] DEBUG[3528][C-00000000]: logger.c:1337 ast_callid_threadassoc_remove: Call_ID [C-00000000] being removed from thread.
[Dec 8 12:03:36] DEBUG[3528]: chan_sip.c:6500 sip_destroy: Destroying SIP dialog 5714f99d46e148ef1f5008f82e570a72@192.168.15.208:5060
[Dec 8 12:03:36] DEBUG[3528]: rtp_engine.c:226 instance_destructor: Destroyed RTP instance ‘0xb750c73c’
[Dec 8 12:03:36] DEBUG[3517]: devicestate.c:342 _ast_device_state: No provider found, checking channel drivers for Motif - xxxxxx
[Dec 8 12:03:36] DEBUG[3517]: devicestate.c:460 do_state_change: Changing state for Motif/xxxxxx - state 0 (Unknown)
[Dec 8 12:03:36] DEBUG[3517]: devicestate.c:440 devstate_event: device ‘Motif/xxxxxx’ state ‘0’
[Dec 8 12:03:36] DEBUG[3556]: app_queue.c:1751 handle_statechange: Device ‘Motif/xxxxxx’ changed to state ‘0’ (Unknown) but we don’t care because they’re not a member of any queue.

[size=150]can anyone help me to solve this issue ???[/size]

[size=200]LOOKS LIKE NO ONE WANTS TO TAKE THE TIME TO TRAVERSE THROUGH MASSIVE LOG FILES. OH WELL. ALSO, YOU’RE ON AN OLD VERSION.[/size]

And motif is somewhat of a minority channel driver.

[size=150][color=#0040FF]many many thanks for your reply malcommd (senior product manager) and david55
as you suggested i have updated my asterisk to version 11.3.0
and now gtalk web client calls are working fine as previus
but now situation is changed some diffrent i have collected some output
and i saw that now call is accepted by asterisk box but when file demo-echotest
file plyaback is starts then session terminate request recieved to the asterisk and call hangs up [/color]
[/size]

<— XMPP received from ‘muzamil8815’ —>
[color=#FF0000]
<------------->

<— XMPP sent to ‘muzamil’ —>

<------------->

<— XMPP sent to ‘muzamil’ —>

<------------->

-- Executing [shabbir92@from-trunk:2] Answer("Motif/shabbirabbasi92-dac4", "") in new stack

<— XMPP sent to ‘muzamil’ —>
<payload-type [color=#FF0000]xmlns=‘http://www.google.com/session/phone’ id=‘102’ name=‘iLBC’ channels=‘1’ clockrate=‘8000’/><payload-type xmlns=‘http://www.google.com/session/phone’ id=‘101’ name=‘telephone-event’ [/color]channels=‘1’ clockrate=‘8000’/>
<------------->

<— XMPP received from 'muzamil —>

<------------->

<— XMPP sent to ‘muzamil’ —>

<------------->
– Executing [shabbir92@from-trunk:3] Playback(“Motif/shabbir92-dac4”, “demo-echotest”) in new stack
– <Motif/shabbir92-dac4> Playing ‘demo-echotest.gsm’ (language ‘en’)

<— XMPP received from ‘muzamil’ —>

<------------->

<— XMPP received from ‘muzamil’ —>

<------------->

[color=#FF0000]<— XMPP received from ‘muzamil’ —>
[/color]
<------------->

<— XMPP sent to ‘muzamil’ —>

<------------->
== Spawn extension (from-trunk, shabbir92, 3) exited non-zero on ‘Motif/shabbir92-dac4’

[size=150][color=#0040FF]i have not terminated the call and i am unable to understand why sesion terminate request recieved
cna u like to put some light on this
[/color][/size]

[color=#0000FF][size=150]problum solved[/size][/color]
thanks to all
i have found and solved
i have edited chan_motif.c
and disabled the below lines by comment out
line 1165
line 1166
line 1167
line 1172
line 1210
line 1211
line 1212
and recompiled
now it is working