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TimeOfDay for company phone
by mailinglist 15 Jan '10

15 Jan '10

Hi Guys Still working on my pfsense based freeswitch. So far it's actually working really good, thanks to all the help on the mailinglist and the excellent tutorials, thankyou very much to all! I'm trying to have one of the phonenumbers react on the TimeOfDay. I followed the guide in the wiki. In public I have set it up to dial the local extension 8203 when the company phone rings. So I put all this in the default.xml to have it process it: <extension name="Company-x8203"> <condition field="destination_number" expression="^8203$"> <action application="transfer" data="CompanyIn"/> </condition> </extension> <extension name="CompanyIn" continue="true"> <condition field="destination_number" expression="^CompanyIn$"/> <condition wday="2-6"/> <condition minute-of-day="540-1020"> <action application="set" data="Company_open=true"/> <action application="transfer" data="xfer-to-company"/> <anti-action application="set" data="Company_open=false"/> <anti-action application="transfer" data="xfer-to-company"/> </condition> </extension> <extension name="xfer-to-company"> <condition field="destination_number" expression="^xfer-to-company$"/> <condition field="${Company_open}" expression="^true$"> <action application="bridge" data="group/company@${domain_name}"/ ( 'mailto:group/company@$domain_name"/' )> <action application="answer"/> <action application="sleep" data="2000"/> <action application="voicemail" data="default ${domain_name} 1011"/> <anti-action application="voicemail" data="default ${domain_name} 1011"/> </condition> </extension> But it doesn't react on the time. I'm wondering if wday and minute-of-day is working on the pfsense package, as far as I know it's version 0.9.5? Any input on this?

1 0

FreeSWITCH Conference Call Starting Shortly!
by Michael Collins 15 Jan '10

15 Jan '10

The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2010_01_05 Come join us and let's talk about FreeSWITCH, VoIP, and all things telephonic! -Michael

1 0

sip trunk question: why call through external profile is challenged?
by Nikolay Kondratyev 15 Jan '10

15 Jan '10

Hi all! I'm brand new to FreeSwitch, but have some experience with SipX. Our company is evaluating FS. For test purposes I set up FS on a virtual machine (vmware esxi). I use CentOS. The FS version I use is 1.0.5-20100110-0400. I have a question regarding sip trunk between FS and SipX. I created the following GW in external profile: [freeswitch@freeswitch external]$ cat sipx-lab.xml | grep -v '<!--' <include> <gateway name="sipx4.lab.nstel.ru"> <param name="username" value="zxcv"/> <param name="password" value="2007"/> <param name="register" value="false"/> </gateway> </include> External.xml file is not modified after installation. I see this gateway via fs_cli: freeswitch@internal> sofia status gateway sipx4.lab.nstel.ru ============================================================================ ===================== Name sipx4.lab.nstel.ru Profile external Scheme Digest Realm sipx4.lab.nstel.ru Username zxcv Password yes >From <sip:zxcv@sipx4.lab.nstel.ru;transport=udp> Contact <sip:gw+sipx4.lab.nstel.ru@172.23.22.49:5080;transport=udp;gw=sipx4.lab.nste l.ru> Exten zxcv To sip:zxcv@sipx4.lab.nstel.ru Proxy sip:sipx4.lab.nstel.ru Context public Expires 3600 Freq 3600 Ping 0 PingFreq 0 PingState 0/0/0 State NOREG Status UP CallsIN 0 CallsOUT 0 ============================================================================ ===================== I created new FS extension 2853. I registers (xlite) and I can call it. Now I want to call FS user from sipx. >From the sipx side one can configure link to FS differently. There are two options: 1. Call FS directly trough sipxproxy (the core part of sipx works as sip proxy, not as b2bua) 2. Call trough embedded b2bua, named sipxbridge. When a call is going trough sipxbridge, it is successfully landed at FS extension 2853. But when a call is going from the sipxproxy, it is challenged with Status 407 "proxy authentication required', and then call fails. (I'm not sure if sipx should handle this challenge, but this is separate question for the sipx forum). At the default log level I see nothing in the freeswitch.log. So the question is why one call through external profile is being challenged while the other is not? I suspect that the reason is in the difference in the two Invite messages: Here is challenged Invite: INVITE sip:2853@fs.lab.nstel.ru:5080;transport=udp SIP/2.0 Record-Route: <sip:172.23.12.104:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EOTRiZjI3MjgtYWMxN zBjZTktMTNjNC0yZjZkZWMtNjNkNmUwN2UtMmY2ZGVj.900_ntap%2Aid%7EMjIyNjAtMQ%60%60 %213ba6d85f6946c4c6001bee1d3b54474f> From: "testphone3857"<sip:3857@lab.nstel.ru>;tag=94bf2728-ac170ce9-13c4-2f6dec-63d 6e07e-2f6dec To: <sip:2853@lab.nstel.ru> Call-Id: 94bed2a0-ac170ce9-13c4-2f6dec-68ffdb81-2f6dec(a)lab.nstel.ru Cseq: 2 INVITE Via: SIP/2.0/UDP 172.23.12.104;branch=z9hG4bK-sipXecs-000eaf48a36fe4029c7cde004a6f44424847 Via: SIP/2.0/TCP 172.23.12.104;branch=z9hG4bK-sipXecs-000bf3008cfa3f3e70470789c75232ba9499~ac 7fd729330fd563f83cacd941311e75;id=22260-1 Via: SIP/2.0/UDP 172.23.12.233:5060;branch=z9hG4bK-2f6dec-b9456412-7ad07784 Max-Forwards: 18 Supported: replaces User-Agent: LG-Nortel LIP 6812 v1.2.38sp SN/00405A18B634 Contact: <sip:3857@172.23.12.233:5060;x-sipX-nonat> Proxy-Authorization: Digest username="3857",realm="lab.nstel.ru",nonce="e2152722611af1bbe59f3a8eb31b8eb8 4b4db2ca",uri="sip:2853@lab.nstel.ru",response="a09c34fbd897783c9b9af50bd044 ccaa",algorithm=MD5 Content-Type: application/sdp Content-Length: 301 Date: Wed, 13 Jan 2010 11:47:22 GMT Expires: 60 X-Sipx-Handled: X172.23.12.104-81.211.30.104 v=0 o=LGEIPP 16246 16247 IN IP4 172.23.12.233 s=SIP Call c=IN IP4 172.23.12.233 t=0 0 m=audio 23008 RTP/AVP 0 8 18 4 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-11 a=sendrecv And here is successful Invite: INVITE sip:2853@fs.lab.nstel.ru;user=phone SIP/2.0 Call-ID: 94bec3b8-ac170ce9-13c4-2f6d47-2fee09d0-2f6d47(a)lab.nstel.ru.0 CSeq: 1 INVITE From: "testphone3857" <sip:3857@172.23.12.104>;tag=816640414244159961 To: <sip:2853@fs.lab.nstel.ru;user=phone> Via: SIP/2.0/UDP 172.23.12.104:5080;branch=z9hG4bK885dde2c8680f5315845cd3350b8b605373534 Max-Forwards: 70 User-Agent: sipXecs/4.0.2 sipXecs/sipxbridge (Linux) P-Asserted-Identity: <sip:1004@172.23.12.104> Contact: <sip:1004@172.23.12.104:5080;transport=udp> Route: <sip:172.23.22.49:5080;transport=udp;lr> Session-Expires: 1800;refresher=uac Allow: INVITE,BYE,ACK,CANCEL,OPTIONS Content-Type: application/sdp Content-Length: 315 v=0 o=sipxbridge 6640787141824452741 1 IN IP4 172.23.12.104 s=SIP Call c=IN IP4 172.23.12.104 t=0 0 m=audio 30248 RTP/AVP 0 8 18 4 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-11 a=sendrecv Can somebody please if it is a FS configuration problem or a software problem or is it a problem on the sipx side? Thanks in advance, Nikolay.

3 7

Re: [Freeswitch-users] multiple leg and multiple rtp
by Anthony Minessale 15 Jan '10

15 Jan '10

If you want to mix them it would cost double. On Jan 14, 2010 8:01 PM, "David Villasmil" <david.villasmil.work(a)gmail.com> wrote: Anthony, What about mixing the RTPs? what's the bounty for that? Cheers David On Fri, Jan 15, 2010 at 12:54 AM, Anthony Minessale <anthony.minessale(a)gmail.com> wrote: > {bridge_early_media=true} > in the dial string in place of ig...

2 1

Not able to capture the custom event
by Jingwei Yang 15 Jan '10

15 Jan '10

Hi Guys, I'm testing to see whether a custom event can be triggered and captured. Here's the extension: <extension name="test_ivr"> <condition field="destination_number" expression="^777$"> <action application="event" data="Event-Name=*CUSTOM* ,Event-Subclass=myevent::ACDnotify"/> <action application="transfer" data="9999 XML default" /> </condition> </extension> If I telnet and use "events plain all", I can get the event. However, if I use "events plain CUSTOM" or "event plain CUSTOM", it doesn't get captured. Is there anything wrong with the command? Regards, -Jingwei

2 2

No ringback.
by Harry Vangberg 14 Jan '10

14 Jan '10

Hello. I am controlling my calls with Freeswitcher via the event socket. When a call comes in I set the following channel variables: hangup_after_bridge=true ignore_early_media=true call_timeout=20 originate_continue_on_timeout=true instant_ringback=true ringback=%(2000,4000,440,480) transfer_ringback=%(2000,4000,440,480) Then I pre_answer and bridge to one or more endpoints. The trouble is, there's no ringback. This is with trunk@16305. For reference, here's excerpt from the log: 2010-01-14 22:47:14.540747 [NOTICE] switch_channel.c:642 New Channel sofia/external/31354228(a)129.142.224.250 [c27529c2-015e-11df-8d0d-c7205d2c96c7] 2010-01-14 22:47:14.540747 [DEBUG] sofia.c:3897 Channel sofia/external/31354228(a)129.142.224.250 entering state [received][100] 2010-01-14 22:47:14.540747 [DEBUG] sofia.c:3908 Remote SDP: v=0 o=root 22377 22377 IN IP4 129.142.224.250 s=session c=IN IP4 129.142.224.250 t=0 0 m=audio 14962 RTP/AVP 8 0 3 97 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 2010-01-14 22:47:14.540747 [DEBUG] sofia_glue.c:3344 Audio Codec Compare [PCMA:8:8000:20]/[PCMA:8:8000:20] 2010-01-14 22:47:14.540747 [DEBUG] sofia_glue.c:2171 Set Codec sofia/external/31354228(a)129.142.224.250 PCMA/8000 20 ms 160 samples 2010-01-14 22:47:14.540747 [DEBUG] sofia_glue.c:3300 Set 2833 dtmf payload to 101 2010-01-14 22:47:14.540747 [DEBUG] sofia.c:4056 (sofia/external/31354228(a)129.142.224.250) State Change CS_NEW -> CS_INIT 2010-01-14 22:47:14.540747 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/31354228(a)129.142.224.250 [BREAK] 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:314 (sofia/external/31354228(a)129.142.224.250) Running State Change CS_INIT 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:338 (sofia/external/31354228(a)129.142.224.250) State INIT 2010-01-14 22:47:14.540747 [DEBUG] mod_sofia.c:83 sofia/external/31354228(a)129.142.224.250 SOFIA INIT 2010-01-14 22:47:14.540747 [DEBUG] mod_sofia.c:111 (sofia/external/31354228(a)129.142.224.250) State Change CS_INIT -> CS_ROUTING 2010-01-14 22:47:14.540747 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/31354228(a)129.142.224.250 [BREAK] 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:338 (sofia/external/31354228(a)129.142.224.250) State INIT going to sleep 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:314 (sofia/external/31354228(a)129.142.224.250) Running State Change CS_ROUTING 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:341 (sofia/external/31354228(a)129.142.224.250) State ROUTING 2010-01-14 22:47:14.540747 [DEBUG] mod_sofia.c:132 sofia/external/31354228(a)129.142.224.250 SOFIA ROUTING 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:78 sofia/external/31354228(a)129.142.224.250 Standard ROUTING 2010-01-14 22:47:14.540747 [INFO] mod_dialplan_xml.c:408 Processing 31354228->77344536 in context public Dialplan: sofia/external/31354228(a)129.142.224.250 parsing [public->ff-phone-sound-recorder] continue=false Dialplan: sofia/external/31354228(a)129.142.224.250 Regex (FAIL) [ff-phone-sound-recorder] destination_number(77344536) =~ /^77348624$/ break=on-false Dialplan: sofia/external/31354228(a)129.142.224.250 parsing [public->ff-ivr] continue=false Dialplan: sofia/external/31354228(a)129.142.224.250 Regex (PASS) [ff-ivr] destination_number(77344536) =~ /^(\d+)$/ break=on-false Dialplan: sofia/external/31354228(a)129.142.224.250 Action socket(127.0.0.1:8084 full) 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:122 (sofia/external/31354228(a)129.142.224.250) State Change CS_ROUTING -> CS_EXECUTE 2010-01-14 22:47:14.540747 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/31354228(a)129.142.224.250 [BREAK] 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:341 (sofia/external/31354228(a)129.142.224.250) State ROUTING going to sleep 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:314 (sofia/external/31354228(a)129.142.224.250) Running State Change CS_EXECUTE 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:348 (sofia/external/31354228(a)129.142.224.250) State EXECUTE 2010-01-14 22:47:14.540747 [DEBUG] mod_sofia.c:181 sofia/external/31354228(a)129.142.224.250 SOFIA EXECUTE 2010-01-14 22:47:14.540747 [DEBUG] switch_core_state_machine.c:159 sofia/external/31354228(a)129.142.224.250 Standard EXECUTE EXECUTE sofia/external/31354228(a)129.142.224.250 socket(127.0.0.1:8084 full) 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551 sofia/external/31354228(a)129.142.224.250 Command Execute set(hangup_after_bridge=true) EXECUTE sofia/external/31354228(a)129.142.224.250 set(hangup_after_bridge=true) 2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818 sofia/external/31354228(a)129.142.224.250 SET [hangup_after_bridge]=[true] 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551 sofia/external/31354228(a)129.142.224.250 Command Execute set(ignore_early_media=true) EXECUTE sofia/external/31354228(a)129.142.224.250 set(ignore_early_media=true) 2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818 sofia/external/31354228(a)129.142.224.250 SET [ignore_early_media]=[true] 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551 sofia/external/31354228(a)129.142.224.250 Command Execute set(call_timeout=20) EXECUTE sofia/external/31354228(a)129.142.224.250 set(call_timeout=20) 2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818 sofia/external/31354228(a)129.142.224.250 SET [call_timeout]=[20] 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551 sofia/external/31354228(a)129.142.224.250 Command Execute set(originate_continue_on_timeout=true) EXECUTE sofia/external/31354228(a)129.142.224.250 set(originate_continue_on_timeout=true) 2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818 sofia/external/31354228(a)129.142.224.250 SET [originate_continue_on_timeout]=[true] 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551 sofia/external/31354228(a)129.142.224.250 Command Execute set(instant_ringback=true) EXECUTE sofia/external/31354228(a)129.142.224.250 set(instant_ringback=true) 2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818 sofia/external/31354228(a)129.142.224.250 SET [instant_ringback]=[true] 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551 sofia/external/31354228(a)129.142.224.250 Command Execute set(ringback=%(2000,4000,440,480)) EXECUTE sofia/external/31354228(a)129.142.224.250 set(ringback=%(2000,4000,440,480)) 2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818 sofia/external/31354228(a)129.142.224.250 SET [ringback]=[%(2000,4000,440,480)] 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551 sofia/external/31354228(a)129.142.224.250 Command Execute set(transfer_ringback=%(2000,4000,440,480)) EXECUTE sofia/external/31354228(a)129.142.224.250 set(transfer_ringback=%(2000,4000,440,480)) 2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:818 sofia/external/31354228(a)129.142.224.250 SET [transfer_ringback]=[%(2000,4000,440,480)] 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551 sofia/external/31354228(a)129.142.224.250 Command Execute bind_meta_app(6 b ai transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML public) EXECUTE sofia/external/31354228(a)129.142.224.250 bind_meta_app(6 b ai transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML public) 2010-01-14 22:47:14.792781 [INFO] switch_ivr_async.c:2322 Bound B-Leg: 6 transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML public 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551 sofia/external/31354228(a)129.142.224.250 Command Execute bind_meta_app(7 b ai transfer::ff-transfer- XML public) EXECUTE sofia/external/31354228(a)129.142.224.250 bind_meta_app(7 b ai transfer::ff-transfer- XML public) 2010-01-14 22:47:14.792781 [INFO] switch_ivr_async.c:2322 Bound B-Leg: 7 transfer::ff-transfer- XML public 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551 sofia/external/31354228(a)129.142.224.250 Command Execute bind_meta_app(1 b ai transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML public) EXECUTE sofia/external/31354228(a)129.142.224.250 bind_meta_app(1 b ai transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML public) 2010-01-14 22:47:14.792781 [INFO] switch_ivr_async.c:2322 Bound B-Leg: 1 transfer::ff-transfer-sofia/gateway/gw1.fonet.dk/004546934488 XML public 2010-01-14 22:47:14.792781 [DEBUG] switch_ivr.c:551 sofia/external/31354228(a)129.142.224.250 Command Execute pre_answer() EXECUTE sofia/external/31354228(a)129.142.224.250 pre_answer() 2010-01-14 22:47:14.792781 [DEBUG] mod_dptools.c:736 sofia/external/31354228(a)129.142.224.250 receive message [PROGRESS] 2010-01-14 22:47:14.792781 [INFO] mod_dptools.c:736 Sending early media 2010-01-14 22:47:14.792781 [DEBUG] sofia_glue.c:2409 AUDIO RTP [sofia/external/31354228(a)129.142.224.250] 10.224.54.102 port 22822 -> 129.142.224.250 port 14962 codec: 8 ms: 20 2010-01-14 22:47:14.792781 [DEBUG] switch_rtp.c:1168 Starting timer [soft] 160 bytes per 20ms 2010-01-14 22:47:14.807902 [INFO] mod_sofia.c:1779 Ring SDP: v=0 o=FreeSWITCH 1263486412 1263486413 IN IP4 79.125.35.134 s=FreeSWITCH c=IN IP4 79.125.35.134 t=0 0 m=audio 22822 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2010-01-14 22:47:14.807902 [NOTICE] mod_sofia.c:1782 Pre-Answer sofia/external/31354228(a)129.142.224.250! 2010-01-14 22:47:14.807902 [DEBUG] switch_core_session.c:633 Send signal sofia/external/31354228(a)129.142.224.250 [BREAK] 2010-01-14 22:47:14.827576 [DEBUG] sofia.c:3892 Channel sofia/external/31354228(a)129.142.224.250 skipping state [early][183] 2010-01-14 22:47:14.827576 [DEBUG] switch_ivr.c:551 sofia/external/31354228(a)129.142.224.250 Command Execute bridge({sip_cid_type=rpid,origination_caller_id_name=77344536,origination_caller_id_number=77344536}sofia/gateway/gw1.fonet.dk/004546934488|loopback/ff-record-77344536) EXECUTE sofia/external/31354228(a)129.142.224.250 bridge({sip_cid_type=rpid,origination_caller_id_name=77344536,origination_caller_id_number=77344536}sofia/gateway/gw1.fonet.dk/004546934488|loopback/ff-record-77344536) 2010-01-14 22:47:14.827576 [DEBUG] switch_ivr_originate.c:1738 variable string 0 = [sip_cid_type=rpid] 2010-01-14 22:47:14.827576 [DEBUG] switch_ivr_originate.c:1738 variable string 1 = [origination_caller_id_name=77344536] 2010-01-14 22:47:14.827576 [DEBUG] switch_ivr_originate.c:1738 variable string 2 = [origination_caller_id_number=77344536] 2010-01-14 22:47:14.827576 [NOTICE] switch_channel.c:642 New Channel sofia/external/004546934488 [c2a1095c-015e-11df-8d0e-c7205d2c96c7] 2010-01-14 22:47:14.827576 [DEBUG] mod_sofia.c:3227 (sofia/external/004546934488) State Change CS_NEW -> CS_INIT 2010-01-14 22:47:14.827576 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/004546934488 [BREAK] 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:314 (sofia/external/004546934488) Running State Change CS_INIT 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:338 (sofia/external/004546934488) State INIT 2010-01-14 22:47:14.827576 [DEBUG] mod_sofia.c:83 sofia/external/004546934488 SOFIA INIT 2010-01-14 22:47:14.827576 [DEBUG] mod_sofia.c:111 (sofia/external/004546934488) State Change CS_INIT -> CS_ROUTING 2010-01-14 22:47:14.827576 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/004546934488 [BREAK] 2010-01-14 22:47:14.827576 [DEBUG] sofia.c:3897 Channel sofia/external/004546934488 entering state [calling][0] 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:338 (sofia/external/004546934488) State INIT going to sleep 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:314 (sofia/external/004546934488) Running State Change CS_ROUTING 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:341 (sofia/external/004546934488) State ROUTING 2010-01-14 22:47:14.827576 [DEBUG] mod_sofia.c:132 sofia/external/004546934488 SOFIA ROUTING 2010-01-14 22:47:14.827576 [DEBUG] switch_ivr_originate.c:66 (sofia/external/004546934488) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2010-01-14 22:47:14.827576 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/004546934488 [BREAK] 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:341 (sofia/external/004546934488) State ROUTING going to sleep 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:314 (sofia/external/004546934488) Running State Change CS_CONSUME_MEDIA 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:360 (sofia/external/004546934488) State CONSUME_MEDIA 2010-01-14 22:47:14.827576 [DEBUG] switch_core_state_machine.c:360 (sofia/external/004546934488) State CONSUME_MEDIA going to sleep 2010-01-14 22:47:14.876945 [DEBUG] switch_ivr_originate.c:1057 Raw Codec Activation Success L16@8000hz 1 channel 20ms 2010-01-14 22:47:14.876945 [DEBUG] switch_core_codec.c:112 sofia/external/31354228(a)129.142.224.250 Push codec L16:10 2010-01-14 22:47:14.876945 [DEBUG] switch_ivr_originate.c:1119 Play Ringback Tone [%(2000,4000,440,480)] 2010-01-14 22:47:14.876945 [DEBUG] switch_core_io.c:652 sofia/external/31354228(a)129.142.224.250 receive message [TRANSCODING_NECESSARY] 2010-01-14 22:47:14.876945 [DEBUG] sofia.c:3897 Channel sofia/external/004546934488 entering state [calling][0] 2010-01-14 22:47:14.912774 [DEBUG] switch_rtp.c:2022 Correct ip/port confirmed. 2010-01-14 22:47:15.668817 [INFO] sofia.c:514 Update Callee ID to "004546934488" <004546934488> 2010-01-14 22:47:15.668817 [DEBUG] sofia.c:3897 Channel sofia/external/004546934488 entering state [proceeding][180] 2010-01-14 22:47:15.668817 [NOTICE] sofia.c:3967 Ring-Ready sofia/external/004546934488! 2010-01-14 22:47:15.668817 [DEBUG] sofia.c:3897 Channel sofia/external/004546934488 entering state [proceeding][183] 2010-01-14 22:47:15.668817 [DEBUG] sofia.c:3908 Remote SDP: v=0 o=root 22377 22377 IN IP4 129.142.224.250 s=session c=IN IP4 129.142.224.250 t=0 0 m=audio 11676 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 2010-01-14 22:47:15.668817 [DEBUG] sofia_glue.c:3344 Audio Codec Compare [PCMA:8:8000:20]/[PCMA:8:8000:20] 2010-01-14 22:47:15.668817 [DEBUG] sofia_glue.c:2171 Set Codec sofia/external/004546934488 PCMA/8000 20 ms 160 samples 2010-01-14 22:47:15.668817 [DEBUG] sofia_glue.c:2409 AUDIO RTP [sofia/external/004546934488] 10.224.54.102 port 17204 -> 129.142.224.250 port 11676 codec: 8 ms: 20 2010-01-14 22:47:15.668817 [DEBUG] switch_rtp.c:1168 Starting timer [soft] 160 bytes per 20ms 2010-01-14 22:47:15.668817 [NOTICE] sofia_glue.c:2937 Pre-Answer sofia/external/004546934488! 2010-01-14 22:47:15.668817 [DEBUG] switch_channel.c:2053 Send signal sofia/external/31354228(a)129.142.224.250 [BREAK] -- Harry Vangberg <harry(a)vangberg.name> http://harry.vangberg.name

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Re: [Freeswitch-users] Cisco media gateway and DTMF
by Dan 14 Jan '10

14 Jan '10

So I did figure it out, it ended up being the CISCO_SKIP_MARK_BIT_2833 issue mentioned on the RTP troubleshooting wiki page. I had incorrectly added this to my dialplan rather then to the external sip profile where it belonged. Thanks Dan- ----- Original Message ----- From: "Dan" <freeswitch-users(a)digitaldan.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, January 13, 2010 1:56:16 PM Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF Thanks, I will try and turn up the debugging on it. I have a lua script that is calling session:getDigits(20,"#",15000) . When I watch it under fs_cli I'm getting this output 2010-01-13 10:44:15.750329 [NOTICE] switch_cpp.cpp:1129 main: Getting digits 2010-01-13 10:44:17.586676 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 0:400 2010-01-13 10:44:17.790705 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 5:400 2010-01-13 10:44:17.910737 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 1:400 2010-01-13 10:44:18.114779 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 7:400 2010-01-13 10:44:18.234812 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 2:400 2010-01-13 10:44:18.462832 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 8:400 2010-01-13 10:44:18.582852 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF 9:400 2010-01-13 10:44:23.311740 [DEBUG] switch_rtp.c:2393 RTP RECV DTMF #:400 2010-01-13 10:44:23.311740 [DEBUG] switch_cpp.cpp:735 getDigits dtmf_buf: 0517289 It will drop any one or more numbers from any part of that sequence. Thanks Dan (sorry for the duped message) ----- Original Message ----- From: "Michael Jerris" <mike(a)jerris.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, January 13, 2010 1:41:51 PM Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF What are you doing with these digits on the freeswitch side, what is it thinking is passed, and do you see anything in debug logs. You also could uncomment the #define DEBUG_2833 from switch_rtp.c for more info. Mike On Jan 13, 2010, at 3:12 PM, Dan wrote: Thanks for your reply, Wireshark shows all of the digits coming through cleanly, in the case below (also on pastebin) its 0951*72789, freeswitch does not seem to be processing one or more of the rtp events. Is there some specific debugging I should turn on that might give me some more info on why freeswitch is not processing some of these events? http://pastebin.freeswitch.org/11780 449 12.280148 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 450 12.285108 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 451 12.290082 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 452 12.335129 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 453 12.340124 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 454 12.345124 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Zero 0 (end) 455 12.365162 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21862, Time=359229277, Mark 456 12.380167 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21863, Time=359229605 461 12.385248 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 462 12.385253 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 465 12.390163 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 466 12.435211 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 467 12.440201 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 468 12.445205 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 469 12.480251 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21870, Time=359230405 470 12.485275 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 471 12.485284 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 472 12.490251 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 475 12.535277 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 476 12.540287 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 477 12.545287 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Five 5 (end) 478 12.580337 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21877, Time=359231197 484 12.585337 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 485 12.585343 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 486 12.590362 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 492 12.635393 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 493 12.640368 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 494 12.645369 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF One 1 (end) 495 12.665406 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21884, Time=359231677, Mark 496 12.680431 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21885, Time=359232005 497 12.680438 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 502 12.685408 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 506 12.690412 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * 507 12.735450 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 508 12.740453 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 509 12.745469 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Star * (end) 511 12.780526 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21892, Time=359232805 512 12.780534 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 513 12.785502 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 514 12.790492 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 515 12.835544 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 516 12.840537 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 517 12.845537 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 523 12.880582 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21899, Time=359233605 524 12.885593 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 525 12.885599 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 526 12.890580 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 528 12.935618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 529 12.940618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 530 12.945618 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Two 2 (end) 531 12.965656 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21906, Time=359234077, Mark 532 12.980659 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21907, Time=359234413 538 12.985690 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 539 12.985695 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 540 12.990657 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 541 13.035703 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 542 13.040701 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 543 13.045700 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Seven 7 (end) 544 13.080762 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21914, Time=359235205 545 13.080768 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 546 13.085736 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 547 13.090798 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 549 13.135809 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 550 13.140780 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 551 13.145793 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Eight 8 (end) 554 13.180827 192.168.3.21 192.168.6.130 RTP PT=ITU-T G.711 PCMU, SSRC=0x1D050315, Seq=21921, Time=359236005 558 13.185847 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 559 13.185854 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 560 13.190821 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 566 13.235872 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 567 13.240864 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) 568 13.245872 192.168.3.21 192.168.6.130 RTP EVENT Payload type=RTP Event, DTMF Nine 9 (end) ----- Original Message ----- From: "Brian West" < brian(a)freeswitch.org > To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, January 13, 2010 12:39:52 PM Subject: Re: [Freeswitch-users] Cisco media gateway and DTMF Cisco gateways can detect DTMF at random if you have kids screaming or loud noises in the background... very unreliable. /b On Jan 13, 2010, at 1:36 PM, Dan wrote: Sorry, I should have made that more clear, this is using 2833. As I mentioned tcpdump shows all the digits coming through as DTMF events on the RTP stream (and all the correct DTMF digits are coming through). _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Announcement: FreeSWITCH Event Socket Library Supports TCL
by Michael Collins 14 Jan '10

14 Jan '10

Just to let everyone know, we've added TCL support for ESL. More info here: http://www.freeswitch.org/node/228 Let us know if you do anything interesting with it and we'll add it to the wiki. -Michael

1 0

Fwd: Re: playback and play_and_get_digits strange misunderstanding
by François Legal 14 Jan '10

14 Jan '10

I did with FreeSWITCH Version 1.0.5-20100112-0400 (hacked) shall I download another one (I don't have svn). François On Thu, 14 Jan 2010 12:20:31 -0500, Michael Jerris wrote: Could you try this with latest trunk as well for comparison, I just changed the way the sounds base directory is set and I want to make sure I did not mess anything up. Mike On Jan 14, 2010, at 3:23 AM, François Legal wrote: So I could (kind of) solve this by myself. There is in vars.xml the variable ${sound_prefix}. I did set it properly to my french sound path and then it worked. However, for the sake of discussion, I did try this with 1.0.5pre8 and the result was different : with the extension FS was trying to play the file ${FREESWITCH_PATH}/sounds/en/us/callie/misc/ringing_disabled.wav with the extension FS was trying to load ${FREESWITCH_PATH}/sounds/fr/fr/julie/misc/ringing_disabled.wav Now with the latest snapshot, with either one of the 2 mentionned extensions, FS tries to play the file in ${sound_prefix}/filepath/${codec_bit_rate}/filename whereas 1.0.5pre8 did not add the codec_bit_rate in the path but took care of the default_language variable. François On Wed, 13 Jan 2010 16:32:16 +0100, François Legal wrote: Hello, trying to make so dialplan extensions that use the playback and play_and_get_digits applications, but I'm having trouble with the file name specification. The files I want to play are in the french language (fr/fr/julie as configured in lang/fr/fr.xml) My extension is as follows : The channel is using a bit rate of 8000 Hz, so by the set default_language=fr I would expect freeswitch to playback the file at ${FREESWITCH_PATH}/sounds/fr/fr/julie/misc/8000/ringing_disabled.wav whereas it tries to playback the file at ${FREESWITCH_PATH}/sounds/en/us/callie/misc/8000/ringing_disabled.wav I have the same with play_and_get_digits application. What am I doing wrong ? François _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org [1] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Links: ------ [1] mailto:FreeSWITCH-users@lists.freeswitch.org

1 0

multiple leg and multiple rtp
by David Villasmil 14 Jan '10

14 Jan '10

hello is it possible to bridge multiple b-legs and provide all audio (progress) until there is an answer on one channel? thanks guys David

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