April 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re ceiving only RTP packets
by rajprithiv89 05 May '12

05 May '12

Hi, we are using csipsimple android voip application which uses Freeswitch as SIP/Media server. We established voip calls between two android clients(A, B) by enabling SRTP in both the sides. Now the problem observed here is, when we captured the packet flow in wireshark, the client which initiates the call(A) is sending SRTP packets and the client which receives the call sends only RTP packets and the FreeSwitch sends SRTP packets to both the clients. We have tested FreeSwitch with other application, there also we ended up in the same problem(one sided SRTP packets flow). Also we have tested the same CSipSimple application with Asterisk server. It works fine by sending SRTP packets from both the clients. When i captured the Wireshark packets SDP contains: >From A to Server: m= audio 4006 RTP/SAVP 98 97 99 >From Server to B m = audio 31562 RTP/SAVP 3 98 99 9 m = audio 31562 RTP/AVP 3 98 99 9 (sends two media line, one for SRTP and one for RTP) >From B to Server(Negotiation done) m = audio 4014 RTP/SAVP 3 96 m = audio 0 RTP/AVP 3 98 99 9 (This is to inform, B is not going to use RTP by putting the port number as 0) >From Server to A(Negotiation done) m = audio 32338 RTP/SAVP 3 96 But in the end, A is sending SRTP packets Server is sending SRTP packets B is sending only RTP packets (But in the negotiation RTP/AVP port was 0) -- View this message in context: http://old.nabble.com/Receiving-only-RTP-packets-tp33758451p33758451.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

3 2

Park and Page use case
by Ian McMaster 04 May '12

04 May '12

As this is my first question, please let me first praise the FreeSWITCH team for a fantastic product and all the efforts of the community. The two books really helped convert all my Asterisk AGI / AMI java code to FreeSWITCH. All IVR play/record/dtmf, etc is working perfectly with FreeSwitch now (and the IVR application is quite large). I came to FreeSwitch after limitations/frustrations with Asterisk. Happy to be aboard, and hope to contribute in time. Use case: Controlling an IVR caller using the event socket; caller chooses an option which will park the caller, and page out to a group of phones for someone to pick up the parked caller. If the park is answered the script exits, or if the park times out then an alternate action is taken. The valet_park command is working fine, and I can decipher through the ChannelExecuteComplete event whether the park was retrieved or not. I am struggling a bit with the paging to multiple polycom phones. - using something like 'originate {sip_auto_answer=true}user/1001,user/1002,... &playback(announce_page.PCMU) has limited success - my understanding is that the first phone to answer 'wins' and the other phones do not receive the playback. - I have better success with conference (a) Invite phones into the conference: conference page_group dial {sip_auto_answer=true}user/1001 ... etc (b) Play file: conference page_group play announce_page.PCMU (although I am having a codec problem here!) (c) End the conference: conference page_group hup all While the conference by nature has certain pre-stocked files (i.e. you are the first caller, and music on hold, etc), I believe I can silence them all with a custom conference. My only show stopper is that if a page goes out to a phone with two line buttons, and is already in use (on one line), then things go bad. The incoming conference request is knocking the existing call on hold, the page is playing out, and the user must resume the call with a soft button. Questions: 1) Is there a better approach to paging multiple phones that I have yet to try? 2) Is there a way to avoid paging a phone that already has a caller on it? 3) Is there conference variable to set the codec to allow the PCMU file to play cleanly (it does play cleanly using originate without setting any explicit codec). Thank you in advance, Ian.

2 1

Meaning of originate_timeout
by Malay Thakershi 04 May '12

04 May '12

Hello, When doing an originate, I set originate_timeout=45 Does it mean the called phone will ring for maximum 45 seconds? Thanks. Malay

2 1

Receiving "early media" during call
by Pete Kelly 04 May '12

04 May '12

Hi I have a scenario where there is an established call (originated from a freeswitch bridge to a downstream UAC) The downstream UA then initiates a re-invite and freeswitch accepts and keeps sending the 200OK with SDP as normal. However the downstream UA is also sending a 183 with sdp to freeswitch at this time, and it looks like freeswitch is ignoring this media. Is it possible to get freeswitch to accept this 183 media during a reinvite transaction? Pete

2 1

can't build mod_gsmopen ?!
by Tamer Higazi 03 May '12

03 May '12

Hi people! I am on gentoo, and fetched through it's overlay with git the snapshot and tried to build FS with mod_gsmopen. During built process it stoped with this error: making all mod_gsmopen copying header files quiet_libtool: link: warning: `/usr/lib/gcc/x86_64-pc-linux-gnu/4.5.3/../../../../lib64/libodbc.la' seems to be moved Compiling gsmopen_protocol.cpp... In file included from gsmopen_protocol.cpp:1:0: gsmopen.h:121:26: schwerwiegender Fehler: ctb-0.16/ctb.h: Datei oder Verzeichnis nicht gefunden Kompilierung beendet. make[5]: *** [gsmopen_protocol.o] Fehler 1 make[4]: *** [all] Fehler 1 make[3]: *** [mod_gsmopen-all] Fehler 1 make[3]: *** Warte auf noch nicht beendete Prozesse... Any advises how to solve that?! Would be really cool to connect the cell through bluetooth and route it to my A100 board. Tamer

5 6

Help - SIP gateway issue with fs_cli but not with freeswtich console.
by Mahendra Bhegade 02 May '12

02 May '12

Hi , I was wondering if you can help me with the issue that I am facing. I have a lua script which dials out using a gateway and works fine from freeswitch console. If I execute the same script from fs_cli I get an error identifiying an error with gateway and the complains about the numbe being invalued I am using a flowroute gateway and sip end point is sofia/gateway/flowroute/15104496650 my entry for flowroute.xml is under conf/sip_profiles/external If some one can shed some light on the matter I would appreciate it. Mahendra 510-449-6650

2 2

mod_voicemail and non-numeric id
by Anton Kvashenkin 02 May '12

02 May '12

Hello list. I've decided to use non-numeric id's at users configuration. <user id="001565229c95" number-alias="109" mailbox="109"> But when the call goes to voicemail app <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> switch_ivr_play_say.c:244 Handle say:[001565229c95] (ru:ru) 2012-04-27 11:39:54.452208 [INFO] mod_say_ru.c:721 ru_say!!! 001565229c95! say_opt.gender=0 say_opt.cases=0 2012-04-27 11:39:54.532157 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16@16000hz 1 channels 20ms The mod_say pronounces user id. How to force to pronounce number-alias?

2 2

sofia_count_reg - return -1
by Lloyd Aloysius 01 May '12

01 May '12

Hi , Today I update the most recent version from git. Does sofia_count_reg broken? It always return -1 Thanks Lloyd

10 13

How to update a live system with very little downtime (How does 'make current' work?)
by Jason Moran 01 May '12

01 May '12

Sorry for such a basic question: 1. If I do a 'make current' while a system is running it will not have any effect until I restart FS, correct? 2. What if 'make current' fails? What if it fails and the server gets restarted, will FS work? 3. Is it more advisable to stop FS, do the 'make current', then start FS even though it will have a longer down time? I know it's best to keep FS up-to-date - but it is difficult when you only have 1 system available and it is expected to have 100% uptime.. Thanks, Jason

6 5

No ringback tone for google voice calls
by Y. Aydogan 01 May '12

01 May '12

Hi, I am having some issues with ringback tones. I configured google voice and I don't hear any ringback tone for outgoing calls through google voice. Attached is a capture which shows an outgoing call. I replaced my destination number with 5555555555 in the logs for privacy. Thanks in advance. Yucel freeswitch@internal> 2012-04-29 12:49:51.258713 [DEBUG] sofia.c:6288 IP 192.168.0.101 Rejected by acl "net_list_5". Falling back to Digest auth. 2012-04-29 12:49:51.284881 [DEBUG] sofia.c:6288 IP 192.168.0.101 Rejected by acl "net_list_5". Falling back to Digest auth. 2012-04-29 12:49:51.289720 [NOTICE] switch_channel.c:784 New Channel sofia/sipinterface_1/yu...(a)192.168.0.112 [27a51939- dda7-42db-8210-3aa643ebc7e6] 2012-04-29 12:49:51.289720 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/yu...(a)192.168.0.112) Running State Change CS_NEW 2012-04-29 12:49:51.289720 [DEBUG] switch_core_state_machine.c:324 (sofia/sipinterface_1/yu...(a)192.168.0.112) State NEW 2012-04-29 12:49:51.305713 [DEBUG] sofia.c:4559 Channel sofia/ sipinterface_1/yu...(a)192.168.0.112 entering state [received][100] 2012-04-29 12:49:51.305713 [DEBUG] sofia.c:4570 Remote SDP: v=0 o=- 22714590 22714590 IN IP4 192.168.0.101 s=- c=IN IP4 192.168.0.101 t=0 0 m=audio 16416 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:30:64000]/[G7221:115:32000:20:48000] 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:30:64000]/[G7221:107:16000:20:32000] 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:30:64000]/[G722:9:8000:20:64000] 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:30:64000]/[PCMU:0:8000:20:64000] 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:30:64000]/[PCMA:8:8000:20:64000] 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:30:64000]/[GSM:3:8000:20:13200] 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4406 Substituting codec PCMU@30i@8000h 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:2721 Set Codec sofia/ sipinterface_1/yu...(a)192.168.0.112 PCMU/8000 30 ms 240 samples 64000 bits 2012-04-29 12:49:51.307729 [DEBUG] sofia_glue.c:4457 Set 2833 dtmf send/recv payload to 101 2012-04-29 12:49:51.307729 [DEBUG] sofia.c:4732 (sofia/sipinterface_1/ yu...(a)192.168.0.112) State Change CS_NEW -> CS_INIT 2012-04-29 12:49:51.307729 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/yu...(a)192.168.0.112 [BREAK] 2012-04-29 12:49:51.307729 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/yu...(a)192.168.0.112) Running State Change CS_INIT 2012-04-29 12:49:51.307729 [DEBUG] switch_core_state_machine.c:342 (sofia/sipinterface_1/yu...(a)192.168.0.112) State INIT 2012-04-29 12:49:51.307729 [DEBUG] mod_sofia.c:83 sofia/sipinterface_1/ yu...(a)192.168.0.112 SOFIA INIT 2012-04-29 12:49:51.307729 [DEBUG] mod_sofia.c:123 (sofia/ sipinterface_1/yu...(a)192.168.0.112) State Change CS_INIT -> CS_ROUTING 2012-04-29 12:49:51.307729 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/yu...(a)192.168.0.112 [BREAK] 2012-04-29 12:49:51.307729 [DEBUG] switch_core_state_machine.c:342 (sofia/sipinterface_1/yu...(a)192.168.0.112) State INIT going to sleep 2012-04-29 12:49:51.307729 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/yu...(a)192.168.0.112) Running State Change CS_ROUTING 2012-04-29 12:49:51.307729 [DEBUG] switch_channel.c:1615 (sofia/ sipinterface_1/yu...(a)192.168.0.112) Callstate Change DOWN -> RINGING 2012-04-29 12:49:51.307729 [DEBUG] switch_core_state_machine.c:345 (sofia/sipinterface_1/yu...(a)192.168.0.112) State ROUTING 2012-04-29 12:49:51.307729 [DEBUG] mod_sofia.c:146 sofia/ sipinterface_1/yu...(a)192.168.0.112 SOFIA ROUTING 2012-04-29 12:49:51.307729 [DEBUG] switch_core_state_machine.c:77 sofia/sipinterface_1/yu...(a)192.168.0.112 Standard ROUTING 2012-04-29 12:49:51.307729 [INFO] mod_dialplan_xml.c:331 Processing Yucel Aydogan <yucel>->5555555555 in context context_1 Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 parsing [context_1- >dingaling_2_pattern_1] continue=true Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 Regex (PASS) [dingaling_2_pattern_1] destination_number(5555555555) =~ /^1{0,1} ([2-9][0-8][0-9][2-9][0-9]{6})$/ break=on-false Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 Action set(prepend=1) Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 Action set(hangup_after_bridge=true) Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 Action bridge(dingaling/GVOICE/+${prepend}5555555...(a)voice.google.com) Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 parsing [context_1- >conditioning_callerid] continue=true Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 Regex (PASS) [conditioning_callerid] ${internal_caller_id_number}(8309) =~ /^.+$/ break=on-false Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 Action set(effective_caller_id_name=${internal_caller_id_name}) Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 Action set(effective_caller_id_number=${internal_caller_id_number}) Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 parsing [context_1- >postroute_global] continue=true Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 Absolute Condition [postroute_global] Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 Action hash(insert/$ {domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 Action hash(insert/$ {domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 Action hash(insert/$ {domain_name}-last_dial/global/${uuid}) Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 parsing [context_1- >preanswer_gtalk] continue=true Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 Regex (FAIL) [preanswer_gtalk] source(mod_sofia) =~ /^mod_dingaling$/ break=on- false Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 parsing [context_1- >main_number_9] continue=true Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 Regex (FAIL) [main_number_9] destination_number(5555555555) =~ /^978nxxxxxx$/ break=on-false Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 parsing [context_1- >main_number_8] continue=true Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 Regex (FAIL) [main_number_8] destination_number(5555555555) =~ /^6000$/ break=on- false Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 parsing [context_1- >main_number_7] continue=true Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 Regex (FAIL) [main_number_7] destination_number(5555555555) =~ /^1039$/ break=on- false Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 parsing [context_1- >main_number_2] continue=true Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 Regex (FAIL) [main_number_2] destination_number(5555555555) =~ /^8309$/ break=on- false Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 parsing [context_1- >main_number_3] continue=true Dialplan: sofia/sipinterface_1/yu...(a)192.168.0.112 Regex (FAIL) [main_number_3] destination_number(5555555555) =~ /^8310$/ break=on- false 2012-04-29 12:49:51.309717 [DEBUG] switch_core_state_machine.c:119 (sofia/sipinterface_1/yu...(a)192.168.0.112) State Change CS_ROUTING -> CS_EXECUTE 2012-04-29 12:49:51.309717 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/yu...(a)192.168.0.112 [BREAK] 2012-04-29 12:49:51.309717 [DEBUG] switch_core_state_machine.c:345 (sofia/sipinterface_1/yu...(a)192.168.0.112) State ROUTING going to sleep 2012-04-29 12:49:51.309717 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/yu...(a)192.168.0.112) Running State Change CS_EXECUTE 2012-04-29 12:49:51.309717 [DEBUG] switch_core_state_machine.c:352 (sofia/sipinterface_1/yu...(a)192.168.0.112) State EXECUTE 2012-04-29 12:49:51.309717 [DEBUG] mod_sofia.c:239 sofia/ sipinterface_1/yu...(a)192.168.0.112 SOFIA EXECUTE 2012-04-29 12:49:51.309717 [DEBUG] switch_core_state_machine.c:157 sofia/sipinterface_1/yu...(a)192.168.0.112 Standard EXECUTE EXECUTE sofia/sipinterface_1/yu...(a)192.168.0.112 set(prepend=1) 2012-04-29 12:49:51.309717 [DEBUG] mod_dptools.c:1028 sofia/ sipinterface_1/yu...(a)192.168.0.112 SET [prepend]=[1] EXECUTE sofia/sipinterface_1/yu...(a)192.168.0.112 set(hangup_after_bridge=true) 2012-04-29 12:49:51.309717 [DEBUG] mod_dptools.c:1028 sofia/ sipinterface_1/yu...(a)192.168.0.112 SET [hangup_after_bridge]=[true] EXECUTE sofia/sipinterface_1/yu...(a)192.168.0.112 bridge(dingaling/ GVOICE/+15555555...(a)voice.google.com) 2012-04-29 12:49:51.309717 [NOTICE] switch_channel.c:784 New Channel dingaling/GVOICE/+15555555...(a)voice.google.com [92032331-e330-4269- b1fe-062de435aa78] 2012-04-29 12:49:51.309717 [DEBUG] mod_dingaling.c:1818 (dingaling/ GVOICE/+15555555...(a)voice.google.com) State Change CS_NEW -> CS_INIT 2012-04-29 12:49:51.309717 [DEBUG] switch_core_session.c:1057 Send signal dingaling/GVOICE/+15555555...(a)voice.google.com [BREAK] 2012-04-29 12:49:51.309717 [DEBUG] mod_dingaling.c:1342 dingaling/ GVOICE/+15555555...(a)voice.google.com CHANNEL KILL 2012-04-29 12:49:51.311724 [DEBUG] switch_core_state_machine.c:318 (dingaling/GVOICE/+15555555...(a)voice.google.com) Running State Change CS_INIT 2012-04-29 12:49:51.311724 [DEBUG] switch_core_state_machine.c:342 (dingaling/GVOICE/+15555555...(a)voice.google.com) State INIT 2012-04-29 12:49:51.311724 [NOTICE] mod_dingaling.c:1104 Ring-Ready dingaling/GVOICE/+15555555...(a)voice.google.com! 2012-04-29 12:49:51.311724 [DEBUG] mod_dingaling.c:1057 Don't have my codec yet here's one 2012-04-29 12:49:51.311724 [DEBUG] mod_dingaling.c:1077 Send Describe [PCMU@8000] 2012-04-29 12:49:51.359973 [DEBUG] mod_dingaling.c:2935 using Existing session for 6160946247 2012-04-29 12:49:51.359973 [DEBUG] mod_dingaling.c:1077 Send Describe [PCMU@8000] 2012-04-29 12:49:51.544699 [DEBUG] mod_dingaling.c:2935 using Existing session for 6160946247 2012-04-29 12:49:51.544699 [DEBUG] mod_dingaling.c:1002 Send Candidate 192.168.0.112:28560 [3FFPxUKMcoG72pKD] 2012-04-29 12:49:51.777702 [DEBUG] mod_dingaling.c:2935 using Existing session for 6160946247 2012-04-29 12:49:51.777702 [DEBUG] mod_dingaling.c:3273 1 candidates 2012-04-29 12:49:51.777702 [DEBUG] mod_dingaling.c:3293 candidate 173.194.76.127:19305 PASS ACL wan.auto 2012-04-29 12:49:51.777702 [DEBUG] mod_dingaling.c:3345 Acceptable Candidate 173.194.76.127:19305 2012-04-29 12:50:04.139089 [DEBUG] mod_dingaling.c:2935 using Existing session for 6160946247 2012-04-29 12:50:04.139089 [DEBUG] mod_dingaling.c:3187 Already decided on a codec 2012-04-29 12:50:04.141096 [DEBUG] mod_dingaling.c:859 Set Read Codec to PCMU@8000 2012-04-29 12:50:04.141096 [DEBUG] mod_dingaling.c:874 Set Write Codec to PCMU@8000 2012-04-29 12:50:05.193038 [DEBUG] switch_nat.c:500 mapped public port 28560 protocol UDP to localport 28560 2012-04-29 12:50:05.193038 [DEBUG] mod_dingaling.c:886 SETUP RTP 192.168.0.112:28560 -> 173.194.76.127:19305 2012-04-29 12:50:05.193038 [DEBUG] switch_rtp.c:1426 Not using a timer 2012-04-29 12:50:05.195316 [DEBUG] switch_rtp.c:3570 Activate VAD codec PCMU 20ms 2012-04-29 12:50:05.195316 [DEBUG] switch_channel.c:2694 (dingaling/ GVOICE/+15555555...(a)voice.google.com) Callstate Change DOWN -> ACTIVE 2012-04-29 12:50:05.195316 [DEBUG] switch_channel.c:2706 Send signal sofia/sipinterface_1/yu...(a)192.168.0.112 [BREAK] 2012-04-29 12:50:05.195316 [NOTICE] mod_dingaling.c:1196 Channel [dingaling/GVOICE/+15555555...(a)voice.google.com] has been answered 2012-04-29 12:50:05.195316 [DEBUG] mod_dingaling.c:1199 (dingaling/ GVOICE/+15555555...(a)voice.google.com) State Change CS_INIT -> CS_ROUTING 2012-04-29 12:50:05.195316 [DEBUG] switch_core_session.c:1057 Send signal dingaling/GVOICE/+15555555...(a)voice.google.com [BREAK] 2012-04-29 12:50:05.195316 [DEBUG] mod_dingaling.c:1342 dingaling/ GVOICE/+15555555...(a)voice.google.com CHANNEL KILL 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:342 (dingaling/GVOICE/+15555555...(a)voice.google.com) State INIT going to sleep 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:318 (dingaling/GVOICE/+15555555...(a)voice.google.com) Running State Change CS_ROUTING 2012-04-29 12:50:05.195316 [DEBUG] switch_channel.c:1615 (dingaling/ GVOICE/+15555555...(a)voice.google.com) Callstate Change ACTIVE -> RINGING 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:345 (dingaling/GVOICE/+15555555...(a)voice.google.com) State ROUTING 2012-04-29 12:50:05.195316 [DEBUG] mod_dingaling.c:1213 dingaling/ GVOICE/+15555555...(a)voice.google.com CHANNEL ROUTING 2012-04-29 12:50:05.195316 [DEBUG] switch_ivr_originate.c:66 (dingaling/GVOICE/+15555555...(a)voice.google.com) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-04-29 12:50:05.195316 [DEBUG] switch_core_session.c:1057 Send signal dingaling/GVOICE/+15555555...(a)voice.google.com [BREAK] 2012-04-29 12:50:05.195316 [DEBUG] mod_dingaling.c:1342 dingaling/ GVOICE/+15555555...(a)voice.google.com CHANNEL KILL 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:345 (dingaling/GVOICE/+15555555...(a)voice.google.com) State ROUTING going to sleep 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:318 (dingaling/GVOICE/+15555555...(a)voice.google.com) Running State Change CS_CONSUME_MEDIA 2012-04-29 12:50:05.195316 [DEBUG] switch_channel.c:1617 (dingaling/ GVOICE/+15555555...(a)voice.google.com) Callstate Change RINGING -> ACTIVE 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:364 (dingaling/GVOICE/+15555555...(a)voice.google.com) State CONSUME_MEDIA 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:364 (dingaling/GVOICE/+15555555...(a)voice.google.com) State CONSUME_MEDIA going to sleep 2012-04-29 12:50:05.198143 [NOTICE] mod_sofia.c:2096 Ring-Ready sofia/ sipinterface_1/yu...(a)192.168.0.112! 2012-04-29 12:50:05.198143 [DEBUG] switch_core_session.c:676 Send signal sofia/sipinterface_1/yu...(a)192.168.0.112 [BREAK] 2012-04-29 12:50:05.198143 [NOTICE] switch_ivr_originate.c:472 Ring Ready sofia/sipinterface_1/yu...(a)192.168.0.112! 2012-04-29 12:50:05.198143 [DEBUG] sofia.c:4559 Channel sofia/ sipinterface_1/yu...(a)192.168.0.112 entering state [early][180] 2012-04-29 12:50:05.225311 [DEBUG] sofia_glue.c:2961 AUDIO RTP [sofia/ sipinterface_1/yu...(a)192.168.0.112] 192.168.0.112 port 19702 -> 192.168.0.101 port 16416 codec: 0 ms: 30 2012-04-29 12:50:05.225311 [DEBUG] switch_rtp.c:1418 Starting timer [soft] 240 bytes per 30ms 2012-04-29 12:50:05.227308 [DEBUG] sofia_glue.c:3179 Set 2833 dtmf send payload to 101 2012-04-29 12:50:05.227308 [DEBUG] sofia_glue.c:3184 Set 2833 dtmf receive payload to 101 2012-04-29 12:50:05.227308 [DEBUG] mod_sofia.c:680 Local SDP sofia/ sipinterface_1/yu...(a)192.168.0.112: v=0 o=FreeSWITCH 1335709303 1335709304 IN IP4 192.168.0.112 s=FreeSWITCH c=IN IP4 192.168.0.112 t=0 0 m=audio 19702 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:30 a=sendrecv 2012-04-29 12:50:05.227308 [DEBUG] switch_core_session.c:676 Send signal sofia/sipinterface_1/yu...(a)192.168.0.112 [BREAK] 2012-04-29 12:50:05.227308 [DEBUG] switch_channel.c:2694 (sofia/ sipinterface_1/yu...(a)192.168.0.112) Callstate Change RINGING -> ACTIVE 2012-04-29 12:50:05.227308 [NOTICE] switch_ivr_originate.c:3288 Channel [sofia/sipinterface_1/yu...(a)192.168.0.112] has been answered 2012-04-29 12:50:05.227308 [DEBUG] switch_ivr_originate.c:3333 Originate Resulted in Success: [dingaling/GVOICE/ +15555555...(a)voice.google.com] 2012-04-29 12:50:05.227308 [DEBUG] switch_core_session.c:676 Send signal dingaling/GVOICE/+15555555...(a)voice.google.com [BREAK] 2012-04-29 12:50:05.227308 [DEBUG] mod_dingaling.c:1342 dingaling/ GVOICE/+15555555...(a)voice.google.com CHANNEL KILL 2012-04-29 12:50:05.227308 [DEBUG] switch_core_session.c:676 Send signal sofia/sipinterface_1/yu...(a)192.168.0.112 [BREAK] 2012-04-29 12:50:05.227308 [DEBUG] switch_ivr_bridge.c:1226 (dingaling/ GVOICE/+15555555...(a)voice.google.com) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2012-04-29 12:50:05.227308 [DEBUG] switch_core_session.c:1057 Send signal dingaling/GVOICE/+15555555...(a)voice.google.com [BREAK] 2012-04-29 12:50:05.227308 [DEBUG] mod_dingaling.c:1342 dingaling/ GVOICE/+15555555...(a)voice.google.com CHANNEL KILL 2012-04-29 12:50:05.230264 [DEBUG] switch_core_state_machine.c:318 (dingaling/GVOICE/+15555555...(a)voice.google.com) Running State Change CS_EXCHANGE_MEDIA 2012-04-29 12:50:05.230264 [DEBUG] switch_core_state_machine.c:355 (dingaling/GVOICE/+15555555...(a)voice.google.com) State EXCHANGE_MEDIA 2012-04-29 12:50:05.230264 [DEBUG] mod_dingaling.c:1350 CHANNEL LOOPBACK 2012-04-29 12:50:05.230264 [DEBUG] sofia.c:4559 Channel sofia/ sipinterface_1/yu...(a)192.168.0.112 entering state [completed][200] 2012-04-29 12:50:05.242302 [DEBUG] sofia.c:4559 Channel sofia/ sipinterface_1/yu...(a)192.168.0.112 entering state [ready][200] 2012-04-29 12:50:05.244336 [DEBUG] switch_rtp.c:2544 Correct ip/port confirmed. 2012-04-29 12:50:05.258274 [DEBUG] switch_core_io.c:970 Engaging Write Buffer at 480 bytes to accommodate 320->480 2012-04-29 12:50:05.267313 [DEBUG] switch_core_session.c:738 Send signal dingaling/GVOICE/+15555555...(a)voice.google.com [BREAK] 2012-04-29 12:50:05.267313 [DEBUG] mod_dingaling.c:1342 dingaling/ GVOICE/+15555555...(a)voice.google.com CHANNEL KILL 2012-04-29 12:50:05.267313 [DEBUG] switch_core_session.c:738 Send signal sofia/sipinterface_1/yu...(a)192.168.0.112 [BREAK] 2012-04-29 12:50:05.358039 [DEBUG] switch_rtp.c:2544 Correct ip/port confirmed. 2012-04-29 12:50:05.358039 [DEBUG] switch_core_io.c:970 Engaging Write Buffer at 320 bytes to accommodate 480->320

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