February 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

making group of phones ring from the command line
by Johan Helsingius 19 Feb '13

19 Feb '13

Another really stupid one... I want to have an alarm process make a group of phones ring. There doesn't need to be a real connection, as nobody will pick up the connection, but as an alternative it could play a standard recording if answered. I have the group defined as a group in the directory but also as a context and number leading to a group call in the dialplan. So what I need is a command to fs_cli that makes a call to the group or dialplan extension. Haven't managed to figure out how to do it... Julf

4 14

mod_directory - Cannot locate domain
by Lloyd Aloysius 19 Feb '13

19 Feb '13

Hi All I have a strange problem with the mod_directory. I use xml_curl for all the applications. Only mod_directory giving this trouble. EXECUTE sofia/sipinterface_1/marv(a)mydomain.com directory(mydomain.com mydomain.com) 2013-02-16 21:43:04.601315 [DEBUG] mod_directory.c:518 [mydomain.com] rwlock 2013-02-16 21:43:04.621318 [ERR] switch_xml.c:1650 Error[[error near line 1]: unclosed tag <document>] 2013-02-16 21:43:04.621318 [WARNING] mod_directory.c:545 Cannot locate domain mydomain.com -- my direcotroy.conf xml curl output <document type="freeswitch/xml"> <section name="configuration" description="FreeSWITCH Configuration"> <configuration name="directory.conf" description="Directory"> <settings> <param name="odbc-dsn" value="freeswitch:user:password"/> </settings> <profiles> <profile name="mydomain.com"> <param name="digit-timeout" value="3000"/> <param name="max-menu-attempts" value="3"/> <param name="max-result" value="5"/> <param name="min-search-digits" value="3"/> <param name="new-search-key" value="3"/> <param name="next-key" value="6"/> <param name="prev-key" value="4"/> <param name="search-order" value="last_name"/> <param name="select-name-key" value="1"/> <param name="switch-order-key" value="*"/> <param name="terminator-key" value="#"/> </profile> </profiles> </configuration> </document> Any help is appreciated. Thanks Lloyd

4 4

outbound caller ID
by shayne.alone＠gmail.com 19 Feb '13

19 Feb '13

Hi all; how can I change the green "FREESWITCH" string on outbound calls? 09:07:57.171420 IP 2.22.222.20.5080 > 2.22.222.10.5060: SIP, length: 1056 E..<.9..@.@..-.E...d.....(V.INVITE sip:12123089@2.22.222.10 SIP/2.0 Via: SIP/2.0/UDP 2.22.222.20:5080;rport;branch=z9hG4bKDZQUXBgeDHZpg Max-Forwards: 69 From: "18009889888" <sip:FreeSWITCH@2.22.222.10>;tag=DK6NFKH46tFam To: <sip:12123089@2.22.222.10> Call-ID: 5b2d87f7-f4f9-1230-e4b8-00505681534d CSeq: 40290074 INVITE Contact: <sip:gw+ICTswOne@2.22.222.20:5080;transport=udp;gw=ICTswOne> User-Agent: FreeSWITCH-mod_sofia/1.2.6+git~20130104T154559Z~a4247651ca Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 203 X-FS-Support: update_display,send_info Remote-Party-ID: "18009889888" <sip:18009889888@2.22.222.10 >;party=calling;screen=yes;privacy=off -- Regards, Ali R. Taleghani

2 2

mod_flite compile error
by ram 19 Feb '13

19 Feb '13

Hi iam having bit old Centos on hosted Platform 2.6.18-348.1.1.el5PAE #1 SMP Tue Jan 22 17:06:35 EST 2013 i686 i686 i386 GNU/Linux iam trying to install Freeswitch GIT i get following error any advice making in main ... ../build/i386-linux-gnu/lib/libflite_cmu_us_rms.a(cmu_us_rms_cg.o):(. data.rel.ro+0x38): undefined reference to `cmu_us_rms_f0_carts' ../build/i386-linux-gnu/lib/libflite_cmu_us_slt.a(cmu_us_slt_cg.o):(. data.rel.ro+0x4c): undefined reference to `cmu_us_slt_spamf0_phrase_cart' collect2: ld returned 1 exit status make[8]: *** [../bin/flite] Error 1 make[7]: *** [build/i386-linux-gnu/obj//.make_build_dirs] Error 2 make[7]: Leaving directory `/usr/src/freeswitch/libs/flite-1.5.4-current' make[6]: *** [/usr/src/freeswitch/libs/flite-1.5.4-current/build/libs/libflite_cmu_us_awb.a] Error 2 make[6]: Leaving directory `/usr/src/freeswitch/src/mod/asr_tts/mod_flite' make[5]: *** [all] Error 1 make[5]: Leaving directory `/usr/src/freeswitch/src/mod/asr_tts/mod_flite' make[4]: *** [mod_flite-all] Error 1 make[4]: Leaving directory `/usr/src/freeswitch/src/mod' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/usr/src/freeswitch/src' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/src/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/freeswitch' make: *** [current] Error 2

3 3

FreeSWITCH Weekly News and Notes
by Michael Collins 19 Feb '13

19 Feb '13

Greetings all! Last week was rather interesting. Initially we had planned on doing a CudaTel <http://www.cudatel.com> demonstration on the weekly conference call. However, interest in Eliot Gable's mod_ha_cluster<http://wiki.freeswitch.org/wiki/Mod_ha_cluster>and the accompanying conversation<http://lists.freeswitch.org/pipermail/freeswitch-users/2013-February/092187…>was particularly intense. That being the case, on last week's conference call <http://wiki.freeswitch.org/wiki/FS_weekly_2013_02_13> we spent most of the time talking about HA in general and how we could build a FreeSWITCH HA system. We also invited everyone who is interested in the subject to call in to the public FreeSWITCH conference at 8PM EST on Tuesday evening (Feb 19) for the first HA conference call. (Eliot won't join until about 8:15PM.) If you have a vested interest in HA for FreeSWITCH then please join the conference call. For this week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_02_20>we will ask one of the participants on the Tuesday night call to give us a brief overview of the HA discussion. After that I will be doing a demonstration of the CudaTel to show off what the FreeSWITCH team has been working so hard to develop these past few years. We hope you enjoy it! For the DIY crowd you may enjoy this site <http://switchpi.org/> that Ken Rice set up. It's dedicated to doing cool things with FreeSWITCH and the Raspberry Pi <http://www.raspberrypi.org/>. Check it out! In ClueCon <http://www.cluecon.com>2013 news we are getting things all set. In the coming days we will be making announcements about new sponsors, registration and hotel information, and a call for speakers for this year's event. Don't forget to plan for a full 3 days, August 6th - 8th. Also, we tend to do a lot of fun stuff on the Monday before ClueCon, so it's a good idea to arrive on or before Monday August 7. Hope you have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

5 4

Creating a conference for an incoming call via socket interface
by lndspereira-fs＠yahoo.com 18 Feb '13

18 Feb '13

Hey all. I configured the FS to answer any incoming call and I have an application that keeps listen to Freeswitch events. I would like know if there is a way to create a conference and add an incoming call to this conference using the FS socket interface. Thanks in advance for your help. Leo Leonardo Nogueira de Sá Pereira Tel.: +55 19 3307-5589 Cel.: +55 19 9122-5943 Skype: leonardo_pereira_77

2 4

NAT settings
by George Cooper 18 Feb '13

18 Feb '13

Hi, I just started trying to work with freeswitch using the FS 1.06 book( although I am running FS ver 1.2.6+git.) , FS cookbook,, and wiki. I and have a question regarding NAT traversal. I have been stuck trying to get phones to register and be able to call each other using ext 1000 to 1002 for example. I couldn't get the phones to register until I read an archive regarding polycom NAT issues. The freeswitch server I'm connecting to is hosted on a remote public IP. I'm using a polycom 335, grandstream xpx 2020, and an x-lite client on a Local 192 network. The below settings seem to have fixed my registration issue. Is there a reason that I would NOT set the parameter settings in sip_profiles/internal.xml to true:  <param name="NDLB-force-rport" value="true"/> and to add the variable "sip-force-contact" for each user in directory/default/1000.xml - 1019.xml ? <include> <user id="1000"> <params> <param name="password" value="$${default_password}"/> <param name="vm-password" value="1000"/> </params> <variables> <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1000"/> <variable name="effective_caller_id_number" value="1000"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> Sorry if this has been gone over, but I'm not sure if there is something I'm missing because the wiki ( http://wiki.freeswitch.org/wiki/NAT ) says "NAT just works" and the internal.xml file says ""... Thanks for any clarity you can provide, Geo

3 2

FW: In-band and out band call issues
by Archana Venugopan 18 Feb '13

18 Feb '13

Can anyone please help me on this peculiar issue. Regards, Archana From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Archana Venugopan Sent: 15 February 2013 16:15 To: FreeSWITCH Users Help Subject: [Freeswitch-users] In-band and out band call issues Hi, Am facing with a typical issue. In my phone settings, if i enable 'Force RFC2833 Out-of-Band DTMF' then my voicemail is not working properly( If I record any greeting or VM, only part of the message is being recorded) but offnet calls for banks or shops is working. If 'Force RFC2833 Out-of-Band DTMF' is disabled my voicemail is working properly but When I call offnet numbers like Banks or shops, then my key press is not being recognised i.e no DTMF detection. My internal .xml and external settings for DTMF is <param name="dtmf-type" value="rfc2833"/> <param name="rfc2833-pt" value="101"/> <param name="dtmf-duration" value="2000"/> Please let me know what needs to be changed so that both voicemail and offnet numbers work properly. Regards, Archana

2 1

Blind transfer dropped calls
by g 18 Feb '13

18 Feb '13

Hi everyone. I'm not absolute begineer with freeswitch, but pretty new after long experience with asterisk. I'm facing now with an installation (1.3.13b+git~20130213) of FS with an issue I can't solve (the same issue was present in 1.0head previous installation), so I invoke your help The scenario is simple: Grandstream (updated firmware) Ip phones connected to FS, gateway to SIP VoIP provider via trunk. Everything works great. Internal phones can talk each oters, calls can came in and go out to PSTN without problem, but ... When a phone call came and an internal phone picks it up, it is impossible to blind tranfert that call to onother internal phone. I mean that the phone i try to transfert the call rings, then it picks up the call and the call is dropped. I suspect is a codec problem, but I can't figure why. All the phones are configured the same way. Here below the log af the process, from external call incoming until the drop. I can see two suspicious steps, where a "no read codec" error and an "incopatible_destination" message appears. I think I made a trivial error, but I really can't understand where. Can please anyone point me in the right direction to solve it? Many thanks g ____ call log taken from the cli - some data is ### masked ### ___ 2013-01-25 17:15:04.770166 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_1/201(a)192.168.35.80 [BREAK] 2013-01-25 17:15:04.770166 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_1/201(a)192.168.35.80 [BREAK] 2013-01-25 17:15:04.770166 [DEBUG] sofia.c:5370 Channel sofia/sipinterface_1/201(a)192.168.35.80 entering state [received][100] 2013-01-25 17:15:04.770166 [DEBUG] sofia.c:5381 Remote SDP: v=0 o=201 8000 8004 IN IP4 192.168.35.114 s=SIP Call c=IN IP4 192.168.35.114 t=0 0 m=audio 5086 RTP/AVP 18 0 8 2 97 9 3 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=sendonly a=ptime:20 2013-01-25 17:15:04.770166 [DEBUG] sofia.c:5810 Passing SDP to other leg. v=0 o=201 8000 8004 IN IP4 192.168.35.114 s=SIP Call c=IN IP4 192.168.35.114 t=0 0 m=audio 5086 RTP/AVP 18 0 8 2 97 9 3 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=sendonly a=ptime:20 2013-01-25 17:15:04.770166 [DEBUG] switch_channel.c:1554 (sofia/sipinterface_1/201(a)192.168.35.80) Callstate Change ACTIVE -> HELD 2013-01-25 17:15:04.770166 [DEBUG] switch_core_session.c:790 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:04.770166 [DEBUG] mod_sofia.c:1812 sofia/sipinterface_3/347******** ### masked ### Sending media re-direct: v=0 o=201 8000 8004 IN IP4 192.168.35.114 s=SIP Call c=IN IP4 192.168.35.114 t=0 0 m=audio 5086 RTP/AVP 18 0 8 2 97 9 3 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=sendonly a=ptime:20 2013-01-25 17:15:04.770166 [DEBUG] switch_core_session.c:728 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:04.770166 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:04.770166 [DEBUG] sofia.c:5370 Channel sofia/sipinterface_3/347******** ### masked ### entering state [calling][0] 2013-01-25 17:15:04.810163 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:04.810163 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:04.830180 [DEBUG] sofia.c:4953 Passing 200 OK to other leg 2013-01-25 17:15:04.830180 [DEBUG] switch_core_session.c:790 Send signal sofia/sipinterface_1/201(a)192.168.35.80 [BREAK] 2013-01-25 17:15:04.830180 [DEBUG] sofia.c:5370 Channel sofia/sipinterface_3/347******** ### masked ### entering state [completing] [200] 2013-01-25 17:15:04.830180 [DEBUG] sofia.c:5381 Remote SDP: v=0 o=root 1558328831 1558328833 IN IP4 xxx.xxx.xxx.xxx s=VOIP ### provider masked ### c=IN IP4 xxx.xxx.xxx.xxx #### masked ### t=0 0 m=audio 46518 RTP/AVP 18 3 97 8 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=recvonly a=ptime:20 2013-01-25 17:15:04.830180 [DEBUG] mod_sofia.c:2363 Responding with 200 [OK] 2013-01-25 17:15:04.830180 [DEBUG] switch_core_session.c:728 Send signal sofia/sipinterface_1/201(a)192.168.35.80 [BREAK] 2013-01-25 17:15:04.830180 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:04.830180 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:04.830180 [DEBUG] sofia.c:5370 Channel sofia/sipinterface_3/347******** ### masked ### entering state [ready][200] 2013-01-25 17:15:04.830180 [DEBUG] switch_core_session.c:790 Send signal sofia/sipinterface_1/201(a)192.168.35.80 [BREAK] 2013-01-25 17:15:04.830180 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_1/201(a)192.168.35.80 [BREAK] 2013-01-25 17:15:04.830180 [DEBUG] sofia.c:5370 Channel sofia/sipinterface_1/201(a)192.168.35.80 entering state [completed][200] 2013-01-25 17:15:04.910172 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_1/201(a)192.168.35.80 [BREAK] 2013-01-25 17:15:04.910172 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_1/201(a)192.168.35.80 [BREAK] 2013-01-25 17:15:04.910172 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_1/201(a)192.168.35.80 [BREAK] 2013-01-25 17:15:04.910172 [DEBUG] sofia.c:5370 Channel sofia/sipinterface_1/201(a)192.168.35.80 entering state [ready][200] 2013-01-25 17:15:09.270174 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_1/201(a)192.168.35.80 [BREAK] 2013-01-25 17:15:09.270174 [DEBUG] sofia.c:6356 Process REFER to [203(a)192.168.35.80] 2013-01-25 17:15:09.270174 [DEBUG] switch_channel.c:2842 (sofia/sipinterface_1/201(a)192.168.35.80) Callstate Change HELD -> HANGUP 2013-01-25 17:15:09.270174 [NOTICE] switch_ivr.c:1696 Hangup sofia/sipinterface_1/201(a)192.168.35.80 [CS_HIBERNATE] [BLIND_TRANSFER] 2013-01-25 17:15:09.270174 [DEBUG] switch_channel.c:2865 Send signal sofia/sipinterface_1/201(a)192.168.35.80 [KILL] 2013-01-25 17:15:09.270174 [DEBUG] switch_ivr_bridge.c:929 sofia/sipinterface_1/201(a)192.168.35.80 skip receive message [UNBRIDGE] (channel is hungup already) 2013-01-25 17:15:09.270174 [DEBUG] switch_core_session.c:1179 Send signal sofia/sipinterface_1/201(a)192.168.35.80 [BREAK] 2013-01-25 17:15:09.270174 [DEBUG] switch_ivr.c:1705 (sofia/sipinterface_3/347******** ### masked ###) State Change CS_HIBERNATE -> CS_ROUTING 2013-01-25 17:15:09.270174 [DEBUG] switch_core_session.c:728 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:09.270174 [DEBUG] switch_core_session.c:1179 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:09.270174 [DEBUG] switch_core_session.c:728 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:09.270174 [NOTICE] switch_ivr.c:1711 Transfer sofia/sipinterface_3/347******** ### masked ### to XML[203@context_1] 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/201(a)192.168.35.80) Running State Change CS_HANGUP 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_3/347******** ### masked ###) Running State Change CS_ROUTING 2013-01-25 17:15:09.270174 [DEBUG] switch_channel.c:1880 (sofia/sipinterface_3/347******** ### masked ###) Callstate Change ACTIVE -> RINGING 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:410 (sofia/sipinterface_3/347******** ### masked ###) State ROUTING 2013-01-25 17:15:09.270174 [DEBUG] mod_sofia.c:148 sofia/sipinterface_3/347******** ### masked ### SOFIA ROUTING 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:104 sofia/sipinterface_3/347******** ### masked ### Standard ROUTING 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:602 (sofia/sipinterface_1/201(a)192.168.35.80) State HANGUP 2013-01-25 17:15:09.270174 [DEBUG] mod_sofia.c:465 Channel sofia/sipinterface_1/201(a)192.168.35.80 hanging up, cause: BLIND_TRANSFER 2013-01-25 17:15:09.270174 [DEBUG] mod_sofia.c:509 Sending BYE to sofia/sipinterface_1/201(a)192.168.35.80 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:47 sofia/sipinterface_1/201(a)192.168.35.80 Standard HANGUP, cause: BLIND_TRANSFER 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:602 (sofia/sipinterface_1/201(a)192.168.35.80) State HANGUP going to sleep 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:393 (sofia/sipinterface_1/201(a)192.168.35.80) State Change CS_HANGUP -> CS_REPORTING 2013-01-25 17:15:09.270174 [DEBUG] switch_core_session.c:1179 Send signal sofia/sipinterface_1/201(a)192.168.35.80 [BREAK] 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/201(a)192.168.35.80) Running State Change CS_REPORTING 2013-01-25 17:15:09.270174 [DEBUG] switch_core_state_machine.c:662 (sofia/sipinterface_1/201(a)192.168.35.80) State REPORTING 2013-01-25 17:15:09.270174 [DEBUG] switch_nat.c:510 mapped public port 29868 protocol UDP to localport 29868 2013-01-25 17:15:09.290165 [DEBUG] switch_nat.c:510 mapped public port 29869 protocol UDP to localport 29869 2013-01-25 17:15:09.290165 [DEBUG] switch_core_session.c:728 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:09.290165 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:09.330130 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:09.330130 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:09.330130 [DEBUG] switch_core_session.c:874 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:09.730204 [DEBUG] switch_core_state_machine.c:79 sofia/sipinterface_1/201(a)192.168.35.80 Standard REPORTING, cause: BLIND_TRANSFER 2013-01-25 17:15:09.730204 [DEBUG] switch_core_state_machine.c:662 (sofia/sipinterface_1/201(a)192.168.35.80) State REPORTING going to sleep 2013-01-25 17:15:09.730204 [DEBUG] switch_core_state_machine.c:387 (sofia/sipinterface_1/201(a)192.168.35.80) State Change CS_REPORTING -> CS_DESTROY 2013-01-25 17:15:09.730204 [DEBUG] switch_core_session.c:1179 Send signal sofia/sipinterface_1/201(a)192.168.35.80 [BREAK] 2013-01-25 17:15:09.730204 [DEBUG] switch_core_session.c:1379 Session 77 (sofia/sipinterface_1/201(a)192.168.35.80) Locked, Waiting on external entities 2013-01-25 17:15:09.730204 [NOTICE] switch_core_session.c:1397 Session 77 (sofia/sipinterface_1/201(a)192.168.35.80) Ended 2013-01-25 17:15:09.730204 [NOTICE] switch_core_session.c:1399 Close Channel sofia/sipinterface_1/201(a)192.168.35.80 [CS_DESTROY] 2013-01-25 17:15:09.730204 [DEBUG] switch_core_state_machine.c:491 (sofia/sipinterface_1/201(a)192.168.35.80) Callstate Change HANGUP -> DOWN 2013-01-25 17:15:09.730204 [DEBUG] switch_core_state_machine.c:494 (sofia/sipinterface_1/201(a)192.168.35.80) Running State Change CS_DESTROY 2013-01-25 17:15:09.730204 [DEBUG] switch_core_state_machine.c:504 (sofia/sipinterface_1/201(a)192.168.35.80) State DESTROY 2013-01-25 17:15:09.730204 [DEBUG] mod_sofia.c:370 sofia/sipinterface_1/201(a)192.168.35.80 SOFIA DESTROY 2013-01-25 17:15:09.730204 [DEBUG] switch_core_state_machine.c:86 sofia/sipinterface_1/201(a)192.168.35.80 Standard DESTROY 2013-01-25 17:15:09.730204 [DEBUG] switch_core_state_machine.c:504 (sofia/sipinterface_1/201(a)192.168.35.80) State DESTROY going to sleep 2013-01-25 17:15:41.650174 [ERR] switch_core_io.c:131 sofia/sipinterface_3/347******** ### masked ### has no read codec. 2013-01-25 17:15:41.650174 [DEBUG] switch_channel.c:2842 (sofia/sipinterface_3/347******** ### masked ###) Callstate Change RINGING -> HANGUP 2013-01-25 17:15:41.650174 [NOTICE] switch_core_io.c:132 Hangup sofia/sipinterface_3/347******** ### masked ### [CS_ROUTING] [INCOMPATIBLE_DESTINATION] 2013-01-25 17:15:41.650174 [DEBUG] switch_channel.c:2865 Send signal sofia/sipinterface_3/347******** ### masked ### [KILL] 2013-01-25 17:15:41.650174 [DEBUG] switch_core_session.c:1179 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:41.650174 [INFO] mod_dialplan_xml.c:481 Processing Gianbattista Gallus <0707966538>->203 in context context_1 Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing [context_1->conditioning_callerid] continue=true Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (FAIL) [conditioning_callerid] ${internal_caller_id_number}() =~ /^.+$/ break=on-false Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing [context_1->postroute_global] continue=true Dialplan: sofia/sipinterface_3/347******** ### masked ### Absolute Condition [postroute_global] Dialplan: sofia/sipinterface_3/347******** ### masked ### Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/sipinterface_3/347******** ### masked ### Action hash(insert/${domain_name}- last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/sipinterface_3/347******** ### masked ### Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/sipinterface_3/347******** ### masked ### Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing [context_1->main_trunk_1_pattern_5] continue=true Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (FAIL) [main_trunk_1_pattern_5] destination_number(203) =~ /^0(\d+)$/ break=on-false Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing [context_1->main_number_2] continue=true Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (FAIL) [main_number_2] destination_number(203) =~ /^999$/ break=on-false Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing [context_1->main_number_4] continue=true Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (FAIL) [main_number_4] destination_number(203) =~ /^201$/ break=on-false Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing [context_1->main_number_5] continue=true Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (FAIL) [main_number_5] destination_number(203) =~ /^202$/ break=on-false Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing [context_1->main_number_6] continue=true Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (PASS) [main_number_6] destination_number(203) =~ /^203$/ break=on-false Dialplan: sofia/sipinterface_3/347******** ### masked ### Action set(hangup_after_bridge=true) Dialplan: sofia/sipinterface_3/347******** ### masked ### Action set(continue_on_fail=true) Dialplan: sofia/sipinterface_3/347******** ### masked ### Action set(vm_auto_play=false) Dialplan: sofia/sipinterface_3/347******** ### masked ### Action set(call_timeout=20) Dialplan: sofia/sipinterface_3/347******** ### masked ### Action set(ringback=${us-ring}) Dialplan: sofia/sipinterface_3/347******** ### masked ### Action set(transfer_ringback=${us-ring}) Dialplan: sofia/sipinterface_3/347******** ### masked ### Action export(sip_callee_id_name=Vanna Cuccu) Dialplan: sofia/sipinterface_3/347******** ### masked ### Action export(sip_callee_id_number=203) Dialplan: sofia/sipinterface_3/347******** ### masked ### Action bridge(user/203(a)192.168.35.80) Dialplan: sofia/sipinterface_3/347******** ### masked ### Action answer() Dialplan: sofia/sipinterface_3/347******** ### masked ### Action voicemail(default voicemail_1 203) Dialplan: sofia/sipinterface_3/347******** ### masked ### Action hangup() Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing [context_1->main_number_7] continue=true Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (FAIL) [main_number_7] destination_number(203) =~ /^204$/ break=on-false Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing [context_1->main_number_8] continue=true Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (FAIL) [main_number_8] destination_number(203) =~ /^200$/ break=on-false Dialplan: sofia/sipinterface_3/347******** ### masked ### parsing [context_1- >catchall] continue=false Dialplan: sofia/sipinterface_3/347******** ### masked ### Regex (PASS) [catchall] destination_number(203) =~ /^(\d+)$/ break=on-false Dialplan: sofia/sipinterface_3/347******** ### masked ### Action playback(misc/invalid_extension.wav) 2013-01-25 17:15:41.670347 [INFO] switch_channel.c:2704 sofia/sipinterface_3/347******** ### masked ### Flipping CID from "Gianbattista Gallus" <0707966538> to "Outbound Call" <347******** ### masked ###> 2013-01-25 17:15:41.670347 [DEBUG] switch_core_state_machine.c:410 (sofia/sipinterface_3/347******** ### masked ###) State ROUTING going to sleep 2013-01-25 17:15:41.670347 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_3/347******** ### masked ###) Running State Change CS_HANGUP 2013-01-25 17:15:41.670347 [DEBUG] switch_core_state_machine.c:602 (sofia/sipinterface_3/347******** ### masked ###) State HANGUP 2013-01-25 17:15:41.670347 [DEBUG] mod_sofia.c:465 Channel sofia/sipinterface_3/347******** ### masked ### hanging up, cause: INCOMPATIBLE_DESTINATION 2013-01-25 17:15:41.670347 [DEBUG] mod_sofia.c:509 Sending BYE to sofia/sipinterface_3/347******** ### masked ### 2013-01-25 17:15:41.670347 [DEBUG] switch_core_state_machine.c:47 sofia/sipinterface_3/347******** ### masked ### Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2013-01-25 17:15:41.670347 [DEBUG] switch_core_state_machine.c:602 (sofia/sipinterface_3/347******** ### masked ###) State HANGUP going to sleep 2013-01-25 17:15:41.670347 [DEBUG] switch_core_state_machine.c:393 (sofia/sipinterface_3/347******** ### masked ###) State Change CS_HANGUP -> CS_REPORTING 2013-01-25 17:15:41.670347 [DEBUG] switch_core_session.c:1179 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:41.670347 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_3/347******** ### masked ###) Running State Change CS_REPORTING 2013-01-25 17:15:41.670347 [DEBUG] switch_core_state_machine.c:662 (sofia/sipinterface_3/347******** ### masked ###) State REPORTING 2013-01-25 17:15:42.090203 [DEBUG] switch_core_state_machine.c:79 sofia/sipinterface_3/347******** ### masked ### Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2013-01-25 17:15:42.090203 [DEBUG] switch_core_state_machine.c:662 (sofia/sipinterface_3/347******** ### masked ###) State REPORTING going to sleep 2013-01-25 17:15:42.090203 [DEBUG] switch_core_state_machine.c:387 (sofia/sipinterface_3/347******** ### masked ###) State Change CS_REPORTING -> CS_DESTROY 2013-01-25 17:15:42.090203 [DEBUG] switch_core_session.c:1179 Send signal sofia/sipinterface_3/347******** ### masked ### [BREAK] 2013-01-25 17:15:42.090203 [DEBUG] switch_core_session.c:1379 Session 78 (sofia/sipinterface_3/347******** ### masked ###) Locked, Waiting on external entities 2013-01-25 17:15:42.090203 [NOTICE] switch_core_session.c:1397 Session 78 (sofia/sipinterface_3/347******** ### masked ###) Ended 2013-01-25 17:15:42.090203 [NOTICE] switch_core_session.c:1399 Close Channel sofia/sipinterface_3/347******** ### masked ### [CS_DESTROY] 2013-01-25 17:15:42.090203 [DEBUG] switch_core_state_machine.c:491 (sofia/sipinterface_3/347******** ### masked ###) Callstate Change HANGUP -> DOWN 2013-01-25 17:15:42.090203 [DEBUG] switch_core_state_machine.c:494 (sofia/sipinterface_3/347******** ### masked ###) Running State Change CS_DESTROY 2013-01-25 17:15:42.090203 [DEBUG] switch_core_state_machine.c:504 (sofia/sipinterface_3/347******** ### masked ###) State DESTROY 2013-01-25 17:15:42.090203 [DEBUG] mod_sofia.c:370 sofia/sipinterface_3/347******** ### masked ### SOFIA DESTROY 2013-01-25 17:15:42.110145 [DEBUG] switch_nat.c:570 unmapped public port 29868 protocol UDP to localport 29868 2013-01-25 17:15:42.130212 [DEBUG] switch_nat.c:570 unmapped public port 29869 protocol UDP to localport 29869 2013-01-25 17:15:42.130212 [DEBUG] switch_core_state_machine.c:86 sofia/sipinterface_3/347******** ### masked ### Standard DESTROY 2013-01-25 17:15:42.130212 [DEBUG] switch_core_state_machine.c:504 (sofia/sipinterface_3/347******** ### masked ###) State DESTROY going to sleep

2 1

Audio Issue: Digitized background beep when speaking
by Imran Moinuddin 18 Feb '13

18 Feb '13

Hi everyone, Recently we've been experiencing some unusual audio issues on our SIP gateway (running Freeswitch) where people on a conference are sometimes hearing periodic / digitized 'beeps' or 'clicks' during the call (mainly when someone's speaking). You can listen to a recorded clip from the call at: https://www.dropbox.com/s/aya3j8xkxtz0wx6/AudioSampleBeep.mp3 The caller was coming in via a landline through one of our carriers. Our FS server wasn't under any particular strain for resources at the time nor does it seem too widespread (it happens pretty non-deterministic from what I can tell) but thought I'd share in case anyone has heard anything similar in the past and if there are insights worth sharing on this. Any feedback would be awesome, Imran

2 1