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krysa79 / asterisk-chan-dongle

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One way audio only when make call... #129

Open GoogleCodeExporter opened 9 years ago

GoogleCodeExporter commented 9 years ago 
Hello, first sorry for my low english.

Have test the modem E3131A with voice feature enabled:

-------------- Status -------------
  Device                  : dongle0
  State                   : Free
  Audio                   : /dev/ttyUSB1
  Data                    : /dev/ttyUSB2
  Voice                   : Yes
  SMS                     : Yes
  Manufacturer            : huawei
  Model                   : E3131A
  Firmware                : 21.157.41.00.314
  IMEI                    : 8654560107xxxxx
  IMSI                    : 724316428xxxxx
  GSM Registration Status : Registered, home network
  RSSI                    : 21, -112 dBm
  Mode                    : No Service
  Submode                 : No service
  Provider Name           : Oi
  Location area code      : "234B"
  Cell ID                 : "0CC7"
  Subscriber Number       : +55858780xxxx
  SMS Service Center      : 002B003500350030003
  Use UCS-2 encoding      : Yes
  USSD use 7 bit encoding : No
  USSD use UCS-2 decoding : Yes
  Tasks in queue          : 0
  Commands in queue       : 0
  Call Waiting            : Disabled
  Current device state    : start
  Desired device state    : start
  When change state       : now
  Calls/Channels          : 0
    Active                : 0
    Held                  : 0
    Dialing               : 0
    Alerting              : 0
    Incoming              : 0
    Waiting               : 0
    Releasing             : 0
    Initializing          : 0

Have test with windows and Mobile Partner (voice support): and make/receive 
call work fine.

Have tested chan_dongle with Debian stable and testing (weezly) + asterisk 11 
but the problem still:

when modem receive a call the audio is perfect in two way
BUT
when modem make a call get one way audio (i can feel but oter person not)

Thanks
Giovanni B.

Original issue reported on code.google.com by gio...@gmail.com on 4 May 2013 at 1:57

GoogleCodeExporter commented 9 years ago 
asterisk log:

E160 make a call (work fine):
...
[2013-05-04 13:48:50] VERBOSE[15956][C-00000008] app_dial.c:     -- Called 
Dongle/dongle1/86647881
[2013-05-04 13:48:50] DEBUG[15956][C-00000008] channel.c: Set channel 
Dongle/dongle1-0100000005 to read format slin
[2013-05-04 13:48:50] DEBUG[15956][C-00000008] channel.c: Set channel 
Dongle/dongle1-0100000005 to write format slin
[2013-05-04 13:48:50] DEBUG[15926] at_read.c: [dongle1] receive 6 byte, used 6, 
free 2042, read 0, write 6
...

E3131 make a call (one way audio problem)
...
[2013-05-04 13:52:24] VERBOSE[16042][C-0000000a] app_dial.c:     -- Called 
Dongle/dongle0/86647881
[2013-05-04 13:52:24] DEBUG[16042][C-0000000a] channel.c: Set channel 
Dongle/dongle0-0100000002 to read format slin
[2013-05-04 13:52:24] DEBUG[16038] at_read.c: [dongle0] receive 6 byte, used 6, 
free 2042, read 0, write 6
...
please note: channel.c do not set write channel

have any suggestion?

Original comment by gio...@gmail.com on 4 May 2013 at 5:34

GoogleCodeExporter commented 9 years ago 
I have try to install with asterisk 1.8.21.00 and chan_dongle Version 1.1, 
Revision 16

Same problem: one way voice call only when make a call.

Complete log:
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:145 at_response_ok: [dongle0] 
AT sent successfully
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:145 at_response_ok: [dongle0] 
ATZ sent successfully
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:145 at_response_ok: [dongle0] 
ATE sent successfully
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:1834 at_response: [dongle0] 
Got AT_CGMI data (manufacturer info)
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:145 at_response_ok: [dongle0] 
AT+CGMI sent successfully
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:1838 at_response: [dongle0] 
Got AT_CGMM data (model info)
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:145 at_response_ok: [dongle0] 
AT+CGMM sent successfully
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:1842 at_response: [dongle0] 
Got AT+CGMR data (firmware info)
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:145 at_response_ok: [dongle0] 
AT+CGMR sent successfully
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:145 at_response_ok: [dongle0] 
AT+CMEE sent successfully
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:1846 at_response: [dongle0] 
Got AT+CGSN data (IMEI number)
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:145 at_response_ok: [dongle0] 
AT+CGSN sent successfully
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:1850 at_response: [dongle0] 
Got AT+CIMI data (IMSI number)
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:145 at_response_ok: [dongle0] 
AT+CIMI sent successfully
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:145 at_response_ok: [dongle0] 
AT+CPIN? sent successfully
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:149 at_response_ok: [dongle0] 
Operator select parameters set
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:153 at_response_ok: [dongle0] 
registration info enabled
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:157 at_response_ok: [dongle0] 
registration query sent
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:161 at_response_ok: [dongle0] 
Subscriber phone number query successed
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:165 at_response_ok: [dongle0] 
Dongle has voice support
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:777 at_response_csca: 
[dongle0] CSCA: +550310000010
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:145 at_response_ok: [dongle0] 
AT+CSCA sent successfully
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:175 at_response_ok: [dongle0] 
Supplementary Service Notification enabled successful
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:180 at_response_ok: [dongle0] 
SMS operation mode set to PDU
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:184 at_response_ok: [dongle0] 
UCS-2 text encoding enabled
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:190 at_response_ok: [dongle0] 
SMS storage location is established
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:194 at_response_ok: [dongle0] 
SMS new message indication enabled
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:195 at_response_ok: [dongle0] 
Dongle has sms support
    -- [dongle0] Dongle initialized and ready
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:283 at_response_ok: [dongle0] 
Got signal strength result
[2013-05-05 20:48:07] DEBUG[12119]: at_response.c:271 at_response_ok: [dongle0] 
Provider query successfully
[2013-05-05 20:48:17] DEBUG[12119]: at_response.c:145 at_response_ok: [dongle0] 
AT sent successfully
[2013-05-05 20:48:27] DEBUG[12119]: at_response.c:145 at_response_ok: [dongle0] 
AT sent successfully
[2013-05-05 20:48:30] DEBUG[12021]: acl.c:736 ast_ouraddrfor: For destination 
'10.1.1.2', our source address is '10.1.1.4'.
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:3836 ast_sip_ouraddrfor: Setting 
SIP_TRANSPORT_UDP with address 10.1.1.4:5060
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:8195 sip_alloc: Allocating new 
SIP dialog for MzNlOGRlY2ExOGM4NzYwODAyMTY0YzBkNzFkN2ZhZWQ - INVITE (No RTP)
[2013-05-05 20:48:30] DEBUG[12021]: sip/reqresp_parser.c:1603 
parse_sip_options: Begin: parsing SIP "Supported: replaces"
[2013-05-05 20:48:30] DEBUG[12021]: sip/reqresp_parser.c:1619 
parse_sip_options: Found SIP option: -replaces-
[2013-05-05 20:48:30] DEBUG[12021]: sip/reqresp_parser.c:1627 
parse_sip_options: Matched SIP option: replaces
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:3684 __sip_xmit: Trying to put 
'SIP/2.0 401' onto UDP socket destined for 10.1.1.2:9638
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:4368 __sip_ack: Stopping 
retransmission on 'MzNlOGRlY2ExOGM4NzYwODAyMTY0YzBkNzFkN2ZhZWQ' of Response 1: 
Match Found
[2013-05-05 20:48:30] DEBUG[12021]: rtp_engine.c:350 ast_rtp_instance_new: 
Using engine 'asterisk' for RTP instance '0xb5c01ce8'
[2013-05-05 20:48:30] DEBUG[12021]: res_rtp_asterisk.c:558 ast_rtp_new: 
Allocated port 17348 for RTP instance '0xb5c01ce8'
[2013-05-05 20:48:30] DEBUG[12021]: rtp_engine.c:359 ast_rtp_instance_new: RTP 
instance '0xb5c01ce8' is setup and ready to go
[2013-05-05 20:48:30] DEBUG[12021]: res_rtp_asterisk.c:2541 ast_rtp_prop_set: 
Setup RTCP on RTP instance '0xb5c01ce8'
  == Using SIP RTP CoS mark 5
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:5471 do_setnat: Setting NAT on 
RTP to Off
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:9313 process_sdp: Processing 
session-level SDP v=0... UNSUPPORTED OR FAILED.
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:9313 process_sdp: Processing 
session-level SDP o=- 13012271318452696 1 IN IP4 10.1.1.2... OK.
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:9313 process_sdp: Processing 
session-level SDP s=Bria 3 release 3.5.1 stamp 69738... UNSUPPORTED OR FAILED.
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:9313 process_sdp: Processing 
session-level SDP c=IN IP4 10.1.1.2... OK.
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:9313 process_sdp: Processing 
session-level SDP b=AS:2064... UNSUPPORTED OR FAILED.
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:9313 process_sdp: Processing 
session-level SDP t=0 0... UNSUPPORTED OR FAILED.
[2013-05-05 20:48:30] DEBUG[12021]: rtp_engine.c:541 
ast_rtp_codecs_payloads_set_m_type: Setting payload 9 based on m type on 
0xb5e1e340
[2013-05-05 20:48:30] DEBUG[12021]: rtp_engine.c:541 
ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 
0xb5e1e340
[2013-05-05 20:48:30] DEBUG[12021]: rtp_engine.c:541 
ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 
0xb5e1e340
[2013-05-05 20:48:30] DEBUG[12021]: rtp_engine.c:541 
ast_rtp_codecs_payloads_set_m_type: Setting payload 18 based on m type on 
0xb5e1e340
[2013-05-05 20:48:30] DEBUG[12021]: rtp_engine.c:541 
ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 
0xb5e1e340
[2013-05-05 20:48:30] DEBUG[12021]: rtp_engine.c:608 
ast_rtp_codecs_payloads_unset: Unsetting payload 123 on 0xb5e1e340
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:9585 process_sdp: Processing 
media-level (audio) SDP a=rtpmap:123 SILK/16000... UNSUPPORTED OR FAILED.
[2013-05-05 20:48:30] DEBUG[12021]: rtp_engine.c:608 
ast_rtp_codecs_payloads_unset: Unsetting payload 122 on 0xb5e1e340
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:9585 process_sdp: Processing 
media-level (audio) SDP a=rtpmap:122 SILK/8000... UNSUPPORTED OR FAILED.
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:9585 process_sdp: Processing 
media-level (audio) SDP a=rtpmap:18 G729/8000... OK.
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:9585 process_sdp: Processing 
media-level (audio) SDP a=fmtp:18 annexb=yes... UNSUPPORTED OR FAILED.
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:9585 process_sdp: Processing 
media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:9585 process_sdp: Processing 
media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED.
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:9585 process_sdp: Processing 
media-level (audio) SDP a=sendrecv... OK.
[2013-05-05 20:48:30] DEBUG[12021]: rtp_engine.c:644 
ast_rtp_codecs_payload_formats: Incorporating payload 0 on 0xb5e1e340
[2013-05-05 20:48:30] DEBUG[12021]: rtp_engine.c:644 
ast_rtp_codecs_payload_formats: Incorporating payload 8 on 0xb5e1e340
[2013-05-05 20:48:30] DEBUG[12021]: rtp_engine.c:644 
ast_rtp_codecs_payload_formats: Incorporating payload 9 on 0xb5e1e340
[2013-05-05 20:48:30] DEBUG[12021]: rtp_engine.c:644 
ast_rtp_codecs_payload_formats: Incorporating payload 18 on 0xb5e1e340
[2013-05-05 20:48:30] DEBUG[12021]: rtp_engine.c:644 
ast_rtp_codecs_payload_formats: Incorporating payload 101 on 0xb5e1e340
[2013-05-05 20:48:30] DEBUG[12021]: res_rtp_asterisk.c:2581 
ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xb5c01ce8'
[2013-05-05 20:48:30] DEBUG[12021]: rtp_engine.c:522 
ast_rtp_codecs_payloads_copy: Copying payload 0 from 0xb5e1e340 to 0xb5c01e94
[2013-05-05 20:48:30] DEBUG[12021]: rtp_engine.c:522 
ast_rtp_codecs_payloads_copy: Copying payload 8 from 0xb5e1e340 to 0xb5c01e94
[2013-05-05 20:48:30] DEBUG[12021]: rtp_engine.c:522 
ast_rtp_codecs_payloads_copy: Copying payload 9 from 0xb5e1e340 to 0xb5c01e94
[2013-05-05 20:48:30] DEBUG[12021]: rtp_engine.c:522 
ast_rtp_codecs_payloads_copy: Copying payload 18 from 0xb5e1e340 to 0xb5c01e94
[2013-05-05 20:48:30] DEBUG[12021]: rtp_engine.c:522 
ast_rtp_codecs_payloads_copy: Copying payload 101 from 0xb5e1e340 to 0xb5c01e94
[2013-05-05 20:48:30] DEBUG[12021]: res_rtp_asterisk.c:2507 ast_rtp_prop_set: 
Ignoring duplicate RTCP property on RTP instance '0xb5c01ce8'
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:9839 process_sdp: We're settling 
with these formats: 0x4 (ulaw)
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:23306 handle_request_invite: 
Checking SIP call limits for device 101
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:6297 update_call_counter: 
Updating call counter for incoming call
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:7483 sip_new: *** Our native 
formats are 0x4 (ulaw) 
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:7484 sip_new: *** Joint 
capabilities are 0x4 (ulaw) 
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:7485 sip_new: *** Our 
capabilities are 0x4 (ulaw) 
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:7486 sip_new: *** 
AST_CODEC_CHOOSE formats are 0x4 (ulaw) 
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:7516 sip_new: This channel will 
not be able to handle video.
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:14805 build_route: build_route: 
Contact hop: <sip:101@10.1.1.2:9638>
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:23610 handle_request_invite: 
SIP/101-00000002: New call is still down.... Trying... 
[2013-05-05 20:48:30] DEBUG[12021]: chan_sip.c:3684 __sip_xmit: Trying to put 
'SIP/2.0 100' onto UDP socket destined for 10.1.1.2:9638
[2013-05-05 20:48:30] DEBUG[12013]: chan_sip.c:27086 sip_devicestate: Checking 
device state for peer 101
[2013-05-05 20:48:30] DEBUG[12013]: devicestate.c:467 do_state_change: Changing 
state for SIP/101 - state 1 (Not in use)
[2013-05-05 20:48:30] DEBUG[12013]: devicestate.c:442 devstate_event: device 
'SIP/101' state '1'
[2013-05-05 20:48:30] DEBUG[12028]: app_queue.c:1544 handle_statechange: Device 
'SIP/101' changed to state '1' (Not in use) but we don't care because they're 
not a member of any queue.
[2013-05-05 20:48:30] DEBUG[12120]: pbx.c:4470 pbx_extension_helper: Launching 
'Dial'
    -- Executing [90@LocalSets:1] Dial("SIP/101-00000002", "Dongle/dongle0/86647881") in new stack
[2013-05-05 20:48:30] DEBUG[12120]: cpvt.c:70 cpvt_alloc: [dongle0] create cpvt 
for call_idx 1 dir 0 state 'initialize'
[2013-05-05 20:48:30] DEBUG[12120]: rtp_engine.c:1433 
ast_rtp_instance_early_bridge_make_compatible: Can't find native functions for 
channel 'Dongle/dongle0-0100000001'
[2013-05-05 20:48:30] DEBUG[12120]: rtp_engine.c:1494 
ast_rtp_instance_early_bridge_make_compatible: Seeded SDP of 
'Dongle/dongle0-0100000001' with that of 'SIP/101-00000002'
[2013-05-05 20:48:30] DEBUG[12120]: channel.c:6232 
ast_channel_inherit_variables: Not copying variable DIALEDTIME.
[2013-05-05 20:48:30] DEBUG[12120]: channel.c:6232 
ast_channel_inherit_variables: Not copying variable ANSWEREDTIME.
[2013-05-05 20:48:30] DEBUG[12120]: channel.c:6232 
ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME.
[2013-05-05 20:48:30] DEBUG[12120]: channel.c:6232 
ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER.
[2013-05-05 20:48:30] DEBUG[12120]: channel.c:6232 
ast_channel_inherit_variables: Not copying variable DIALSTATUS.
[2013-05-05 20:48:30] DEBUG[12120]: channel.c:6232 
ast_channel_inherit_variables: Not copying variable SIPCALLID.
[2013-05-05 20:48:30] DEBUG[12120]: channel.c:6232 
ast_channel_inherit_variables: Not copying variable SIPDOMAIN.
[2013-05-05 20:48:30] DEBUG[12120]: channel.c:6232 
ast_channel_inherit_variables: Not copying variable SIPURI.
[2013-05-05 20:48:30] DEBUG[12120]: channel.c:212 channel_call: [dongle0] 
Calling dongle0/86647881 on Dongle/dongle0-0100000001
[2013-05-05 20:48:30] DEBUG[12120]: helpers.c:55 get_at_clir_value: [dongle0] 
callingpres: Presentation Allowed, Passed Screen
[2013-05-05 20:48:30] DEBUG[12119]: at_response.c:145 at_response_ok: [dongle0] 
AT+CLIR sent successfully
    -- Called Dongle/dongle0/86647881
[2013-05-05 20:48:30] DEBUG[12120]: channel.c:5254 set_format: Set channel 
Dongle/dongle0-0100000001 to read format slin
[2013-05-05 20:48:30] DEBUG[12120]: channel.c:5254 set_format: Set channel 
SIP/101-00000002 to write format slin
[2013-05-05 20:48:30] DEBUG[12120]: channel.c:5254 set_format: Set channel 
SIP/101-00000002 to read format slin
[2013-05-05 20:48:30] DEBUG[12120]: channel.c:5254 set_format: Set channel 
Dongle/dongle0-0100000001 to write format slin
[2013-05-05 20:48:30] DEBUG[12119]: at_response.c:221 at_response_ok: [dongle0] 
ATD sent successfully for call id 1
[2013-05-05 20:48:30] DEBUG[12119]: at_response.c:635 at_response_orig: 
[dongle0] ORIG Received call_index: 1 call_type 0
[2013-05-05 20:48:30] DEBUG[12119]: channel.c:1011 change_channel_state: 
[dongle0] call idx 1 mpty 0, change state from 'initialize' to 'dialing' has 
channel
[2013-05-05 20:48:30] DEBUG[12013]: channel.c:926 channel_devicestate: Checking 
device state for device dongle0
[2013-05-05 20:48:30] DEBUG[12013]: devicestate.c:467 do_state_change: Changing 
state for Dongle/dongle0 - state 2 (In use)
[2013-05-05 20:48:30] DEBUG[12013]: devicestate.c:442 devstate_event: device 
'Dongle/dongle0' state '2'
[2013-05-05 20:48:30] DEBUG[12028]: app_queue.c:1544 handle_statechange: Device 
'Dongle/dongle0' changed to state '2' (In use) but we don't care because 
they're not a member of any queue.
    -- Dongle/dongle0-0100000001 is making progress passing it to SIP/101-00000002
[2013-05-05 20:48:30] DEBUG[12120]: chan_sip.c:12255 
transmit_response_with_sdp: Setting framing from config on incoming call
[2013-05-05 20:48:30] DEBUG[12120]: chan_sip.c:11852 add_sdp: ** Our 
capability: 0x4 (ulaw) Video flag: True Text flag: True
[2013-05-05 20:48:30] DEBUG[12120]: chan_sip.c:11853 add_sdp: ** Our prefcodec: 
0x0 (nothing) 
[2013-05-05 20:48:30] DEBUG[12120]: chan_sip.c:11962 add_sdp: -- Done with 
adding codecs to SDP
[2013-05-05 20:48:30] DEBUG[12120]: chan_sip.c:12148 add_sdp: Done building 
SDP. Settling with this capability: 0x4 (ulaw)
[2013-05-05 20:48:30] DEBUG[12120]: chan_sip.c:3684 __sip_xmit: Trying to put 
'SIP/2.0 183' onto UDP socket destined for 10.1.1.2:9638
[2013-05-05 20:48:30] DEBUG[12119]: at_response.c:904 at_response_clcc: 
[dongle0] CLCC callidx 1 dir 0 state 2 mode 0 mpty 0 number 86647881 type 129
[2013-05-05 20:48:30] DEBUG[12119]: at_response.c:145 at_response_ok: [dongle0] 
AT+CLCC sent successfully
[2013-05-05 20:48:30] DEBUG[12119]: at_response.c:231 at_response_ok: [dongle0] 
AT^DDSETEX sent successfully
[2013-05-05 20:48:30] DEBUG[12119]: at_response.c:904 at_response_clcc: 
[dongle0] CLCC callidx 1 dir 0 state 2 mode 0 mpty 0 number 86647881 type 129
[2013-05-05 20:48:30] DEBUG[12119]: at_response.c:145 at_response_ok: [dongle0] 
AT+CLCC sent successfully
[2013-05-05 20:48:30] DEBUG[12119]: at_response.c:290 at_response_ok: [dongle0] 
Volume level synchronized
[2013-05-05 20:48:30] DEBUG[12120]: res_rtp_asterisk.c:1808 ast_rtcp_read: Got 
RTCP report of 56 bytes
[2013-05-05 20:48:33] DEBUG[12120]: res_rtp_asterisk.c:1808 ast_rtcp_read: Got 
RTCP report of 56 bytes
[2013-05-05 20:48:33] DEBUG[12021]: chan_sip.c:3373 sip_alreadygone: Setting 
SIP_ALREADYGONE on dialog MzNlOGRlY2ExOGM4NzYwODAyMTY0YzBkNzFkN2ZhZWQ
[2013-05-05 20:48:33] DEBUG[12021]: res_rtp_asterisk.c:2581 
ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xb5c01ce8'
[2013-05-05 20:48:33] DEBUG[12021]: chan_sip.c:3684 __sip_xmit: Trying to put 
'SIP/2.0 487' onto UDP socket destined for 10.1.1.2:9638
[2013-05-05 20:48:33] DEBUG[12021]: chan_sip.c:3684 __sip_xmit: Trying to put 
'SIP/2.0 200' onto UDP socket destined for 10.1.1.2:9638
[2013-05-05 20:48:33] DEBUG[12021]: chan_sip.c:4368 __sip_ack: Stopping 
retransmission on 'MzNlOGRlY2ExOGM4NzYwODAyMTY0YzBkNzFkN2ZhZWQ' of Response 2: 
Match Found
[2013-05-05 20:48:33] DEBUG[12120]: channel.c:2884 ast_hangup: Hanging up 
channel 'Dongle/dongle0-0100000001'
[2013-05-05 20:48:33] DEBUG[12120]: channel.c:341 channel_hangup: [dongle0] 
Hanging up call idx 1 need hangup 1
[2013-05-05 20:48:33] DEBUG[12013]: channel.c:926 channel_devicestate: Checking 
device state for device dongle0
[2013-05-05 20:48:33] DEBUG[12013]: devicestate.c:467 do_state_change: Changing 
state for Dongle/dongle0 - state 2 (In use)
[2013-05-05 20:48:33] DEBUG[12013]: devicestate.c:442 devstate_event: device 
'Dongle/dongle0' state '2'
[2013-05-05 20:48:33] DEBUG[12028]: app_queue.c:1544 handle_statechange: Device 
'Dongle/dongle0' changed to state '2' (In use) but we don't care because 
they're not a member of any queue.
[2013-05-05 20:48:33] DEBUG[12013]: channel.c:926 channel_devicestate: Checking 
device state for device dongle0
[2013-05-05 20:48:33] DEBUG[12013]: devicestate.c:467 do_state_change: Changing 
state for Dongle/dongle0 - state 2 (In use)
[2013-05-05 20:48:33] DEBUG[12013]: devicestate.c:442 devstate_event: device 
'Dongle/dongle0' state '2'
[2013-05-05 20:48:33] DEBUG[12028]: app_queue.c:1544 handle_statechange: Device 
'Dongle/dongle0' changed to state '2' (In use) but we don't care because 
they're not a member of any queue.
[2013-05-05 20:48:33] DEBUG[12120]: app_dial.c:3029 dial_exec_full: Exiting 
with DIALSTATUS=CANCEL.
[2013-05-05 20:48:33] DEBUG[12120]: pbx.c:5290 __ast_pbx_run: Spawn extension 
(LocalSets,90,1) exited non-zero on 'SIP/101-00000002'
  == Spawn extension (LocalSets, 90, 1) exited non-zero on 'SIP/101-00000002'
[2013-05-05 20:48:33] DEBUG[12120]: channel.c:2735 ast_softhangup_nolock: 
Soft-Hanging up channel 'SIP/101-00000002'
[2013-05-05 20:48:33] DEBUG[12120]: channel.c:2884 ast_hangup: Hanging up 
channel 'SIP/101-00000002'
[2013-05-05 20:48:33] DEBUG[12120]: chan_sip.c:6679 sip_hangup: Hangup call 
SIP/101-00000002, SIP callid MzNlOGRlY2ExOGM4NzYwODAyMTY0YzBkNzFkN2ZhZWQ
[2013-05-05 20:48:33] DEBUG[12120]: res_rtp_asterisk.c:2581 
ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xb5c01ce8'
[2013-05-05 20:48:33] DEBUG[12013]: chan_sip.c:27086 sip_devicestate: Checking 
device state for peer 101
[2013-05-05 20:48:33] DEBUG[12013]: devicestate.c:467 do_state_change: Changing 
state for SIP/101 - state 1 (Not in use)
[2013-05-05 20:48:33] DEBUG[12013]: devicestate.c:442 devstate_event: device 
'SIP/101' state '1'
[2013-05-05 20:48:33] DEBUG[12028]: app_queue.c:1544 handle_statechange: Device 
'SIP/101' changed to state '1' (Not in use) but we don't care because they're 
not a member of any queue.
[2013-05-05 20:48:33] DEBUG[12119]: at_response.c:1856 at_response: [dongle0] 
Ignoring unknown result: ''
[2013-05-05 20:48:33] DEBUG[12119]: at_response.c:243 at_response_ok: [dongle0] 
Successful hangup for call idx 1
[2013-05-05 20:48:33] DEBUG[12119]: at_response.c:145 at_response_ok: [dongle0] 
AT+CLCC sent successfully
[2013-05-05 20:48:34] DEBUG[12021]: chan_sip.c:6445 sip_destroy: Destroying SIP 
dialog MzNlOGRlY2ExOGM4NzYwODAyMTY0YzBkNzFkN2ZhZWQ
[2013-05-05 20:48:34] DEBUG[12021]: rtp_engine.c:298 instance_destructor: 
Destroyed RTP instance '0xb5c01ce8'
[2013-05-05 20:48:43] DEBUG[12119]: at_response.c:145 at_response_ok: [dongle0] 
AT sent successfully
[2013-05-05 20:48:53] DEBUG[12119]: at_response.c:145 at_response_ok: [dongle0] 
AT sent successfully
[2013-05-05 20:49:04] DEBUG[12119]: at_response.c:145 at_response_ok: [dongle0] 
AT sent successfully
[2013-05-05 20:49:14] DEBUG[12119]: at_response.c:145 at_response_ok: [dongle0] 
AT sent successfully

Original comment by gio...@gmail.com on 5 May 2013 at 11:56

GoogleCodeExporter commented 9 years ago 
currently official branch not supported asterisk 10 and 11 :)))

Original comment by bg_...@mail.ru on 15 May 2013 at 7:40

GoogleCodeExporter commented 9 years ago 
Hi, understand

only for information i have use with succes E160, E153, E156 and E160 on 
Asterisk 11 with 
https://github.com/jstasiak/asterisk-chan-dongle/archive/asterisk11.zip

You suggest is use asterisk 1.8 and official branch?

Thanks

Original comment by gio...@gmail.com on 15 May 2013 at 8:05

GoogleCodeExporter commented 9 years ago 
i develop & test for 1.6 1.8 

as you can see original chan_dongle not compiled for 10 11, right ?

Original comment by bg_...@mail.ru on 30 Jul 2013 at 7:44

GoogleCodeExporter commented 9 years ago 
I think I have found what is the problem with Huawei E3131 and chan_dongle. 
Recently I have compiled asterisk 1.8.32 and chan_dongle Version 1.1, Revision 
45.
One way audio problem during outgoing calls is because asterisk never get 
ANSWER state from chan_dongle and always has pre-answer (early media) state. 
May be it is incorrect processing AT response code from E3131?
I see the same problem is in Giovanni's logs. 
http://code.google.com/p/asterisk-chan-dongle/issues/detail?id=129#c2

--
Evgeniy Reshetnyak

Original comment by evgeshka...@gmail.com on 14 Mar 2015 at 9:31