- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] deflect question
by Madovsky 15 Jan '11

15 Jan '11

now I can see in log from a registered user deflect: EXECUTE sofia/internal/9999999999999@domain deflect(sip:conf_9999999999999@12.34.56.78) 2011-01-15 12:14:14.303271 [DEBUG] sofia.c:4604 Channel sofia/internal/9999999999999@domain entering state [terminated][401] 2011-01-15 12:14:14.303271 [DEBUG] switch_channel.c:2455 (sofia/internal/9999999999999@domain) Callstate Change RINGING -> HANGUP 2011-01-15 12:14:14.303271 [NOTICE] sofia.c:5244 Hangup sofia/internal/9999999999999@domain [CS_EXECUTE] [CALL_REJECTED] 2011-01-15 12:14:14.303271 [DEBUG] switch_channel.c:2471 Send signal sofia/internal/9999999999999@domain [KILL] 2011-01-15 12:14:14.303271 [DEBUG] switch_core_session.c:1083 Send signal sofia/internal/9999999999999@domain [BREAK] 2011-01-15 12:14:14.353430 [INFO] sofia_presence.c:785 IN START_PRESENCE_SQL (internal) 2011-01-15 12:14:14.353430 [ERR] sofia_presence.c:796 EVENT DUMP: ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Saturday, January 15, 2011 12:02 PM Subject: Re: [Freeswitch-users] deflect question Not sure that what I said below is the answer of the problem. I made other tests that show that if the caller is not registered user and call from external sip profile, deflect works flawessly. if I try to deflect from a registered user (internal.xml) to an external or internal clustered user it locks after execute deflect about 5 mn and hangup. ----- Original Message ----- From: Madovsky To: FreeSWITCH Users Help Sent: Wednesday, January 12, 2011 11:58 PM Subject: Re: [Freeswitch-users] deflect question no, but in fact I just release that my sip phone doesnt' support RFC 3515... ----- Original Message ----- From: Sam To: FreeSWITCH Users Help Sent: Wednesday, January 12, 2011 11:05 PM Subject: Re: [Freeswitch-users] deflect question it originates a call with refer message,when the call lands on the different server does it releases the first leg ? and the call bridges the leg b without the leg a on the first server ? C --> A --> B A-> server 1 B -> server 2 C-> call initiator After deflect from A to B will the call flow be C-->B with out A ? Regards Sam On Thu, Jan 13, 2011 at 2:38 AM, Madovsky <infos(a)madovsky.org> wrote: if I use <action application="deflect" data="sip:someone@otherFS.com"/> in the otherFS dialplan should I createan extension with "someone" condition only ? Thanks _______________________________________________ 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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Re: [Freeswitch-users] deflect question
by Madovsky 15 Jan '11

15 Jan '11

Not sure that what I said below is the answer of the problem. I made other tests that show that if the caller is not registered user and call from external sip profile, deflect works flawessly. if I try to deflect from a registered user (internal.xml) to an external or internal clustered user it locks after execute deflect about 5 mn and hangup. ----- Original Message ----- From: Madovsky To: FreeSWITCH Users Help Sent: Wednesday, January 12, 2011 11:58 PM Subject: Re: [Freeswitch-users] deflect question no, but in fact I just release that my sip phone doesnt' support RFC 3515... ----- Original Message ----- From: Sam To: FreeSWITCH Users Help Sent: Wednesday, January 12, 2011 11:05 PM Subject: Re: [Freeswitch-users] deflect question it originates a call with refer message,when the call lands on the different server does it releases the first leg ? and the call bridges the leg b without the leg a on the first server ? C --> A --> B A-> server 1 B -> server 2 C-> call initiator After deflect from A to B will the call flow be C-->B with out A ? Regards Sam On Thu, Jan 13, 2011 at 2:38 AM, Madovsky <infos(a)madovsky.org> wrote: if I use <action application="deflect" data="sip:someone@otherFS.com"/> in the otherFS dialplan should I createan extension with "someone" condition only ? Thanks _______________________________________________ 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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Newbie Dial Plan question
by Charles Bujold 15 Jan '11

15 Jan '11

Newbie question, I have a ht503(FXO) connected to freeswitch to make local calls (7digits) The HT503-FXO is registered as extension 510. If I dial 510 (Bria softphone) via freeswitch I get dial tone and can dial my local number. However I can't seem to get it working via my dial plan. I created a gateway to connect to the HT503 see below, and I created a local 7digit dial plan. When I try to connect I get the error "Network_out_of_Order". Can somebody point out my error. Thanks Charles Gateway: /sip_profiles/external <include> <gateway name="8580444"> <param name="username" value="510"/> <param name="password" value="510"/> <param name="proxy" value="192.168.10.82"/> <param name="expire-seconds" value="800"/> <param name="register" value="false"/> <param name="retry-seconds" value="60"/> <param name="context" value="public"/> </gateway> </include> Dial Plan: /dialplan/default <extension name="LocalCalls.7d" > <condition field="destination_number" expression="^\d{7}$"> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="bridge" data="sofia/gateway/8580444/$1"/> </condition> </extension>

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Confirming if conference is being recorded
by Richard Alam 14 Jan '11

14 Jan '11

Hi, I am planning on recording a conference. Is there are way to regularly check if the conference is being recorded aside from regularly checking the recorded file increasing in size? Can I tell the conference app to send out events to ESL regularly saying that it is recording? Thanks in advance. Richard -- --- BigBlueButton http://www.bigbluebutton.org http://code.google.com/p/bigbluebutton

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FS as sip proxy registration
by Madovsky 14 Jan '11

14 Jan '11

if I use FS as SIP proxy registrar, does the REFER method will go through the client transparently ? I'm facing problem of this method with this configuration Thanks

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SIP/Sofia create channel 503 error…
by Bit Rot 14 Jan '11

14 Jan '11

Hey y'all, We have FreeSWITCH and Microsoft Speech Server 2007 running fine on separate boxes. However when we try to consolidate onto a single box as per instructions for inbound DID calls in Brian Campbell’s most excellent blog: http://gotspeech.net/blogs/verbalinput/archive/2010/01/08/speech-server-200… We receive a 503 error – terminate and SIP returns Error Cause: 41 [NORMAL_TEMPORARY_FAILURE]. The first leg from the provider to FreeSwitch works fine and the dialplan is parsed correctly, also outbound calls through FreeSWITCH work. The problem occurs when the second leg of the call is trying to create a channel from FreeSWITCH to the Speech Server. Here is the trace: EXECUTE sofia/external/106308_1(a)74.54.54.178 bridge(sofia/internal/2146130149@127.0.0.1:5060;transport=tcp) 2011-01-13 22:23:27.387971 [NOTICE] switch_channel.c:784 New Channel sofia/internal/2146130149@127.0.0.1:5060 [1df88e9c-16e6-415a-ac1b-83e29d25b3b0] 2011-01-13 22:23:27.387971 [DEBUG] mod_sofia.c:3976 (sofia/internal/2146130149@127.0.0.1:5060) State Change CS_NEW -> CS_INIT 2011-01-13 22:23:27.387971 [DEBUG] switch_core_session.c:1064 Send signal sofia/internal/2146130149@127.0.0.1:5060 [BREAK] 2011-01-13 22:23:27.387971 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/2146130149@127.0.0.1:5060) Running State Change CS_INIT 2011-01-13 22:23:27.387971 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/2146130149@127.0.0.1:5060) State INIT 2011-01-13 22:23:27.387971 [DEBUG] mod_sofia.c:86 sofia/internal/2146130149@127.0.0.1:5060 SOFIA INIT 2011-01-13 22:23:27.388971 [DEBUG] mod_sofia.c:126 (sofia/internal/2146130149@127.0.0.1:5060) State Change CS_INIT -> CS_ROUTING 2011-01-13 22:23:27.388971 [DEBUG] switch_core_session.c:1064 Send signal sofia/internal/2146130149@127.0.0.1:5060 [BREAK] 2011-01-13 22:23:27.388971 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/2146130149@127.0.0.1:5060) State INIT going to sleep 2011-01-13 22:23:27.388971 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/2146130149@127.0.0.1:5060) Running State Change CS_ROUTING 2011-01-13 22:23:27.388971 [DEBUG] switch_channel.c:1615 (sofia/internal/2146130149@127.0.0.1:5060) Callstate Change DOWN -> RINGING 2011-01-13 22:23:27.388971 [DEBUG] switch_core_state_machine.c:359 (sofia/internal/2146130149@127.0.0.1:5060) State ROUTING 2011-01-13 22:23:27.388971 [DEBUG] mod_sofia.c:149 sofia/internal/2146130149@127.0.0.1:5060 SOFIA ROUTING 2011-01-13 22:23:27.388971 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/2146130149@127.0.0.1:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2011-01-13 22:23:27.388971 [DEBUG] switch_core_session.c:1064 Send signal sofia/internal/2146130149@127.0.0.1:5060 [BREAK] 2011-01-13 22:23:27.388971 [DEBUG] switch_core_state_machine.c:359 (sofia/internal/2146130149@127.0.0.1:5060) State ROUTING going to sleep 2011-01-13 22:23:27.388971 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/2146130149@127.0.0.1:5060) Running State Change CS_CONSUME_MEDIA 2011-01-13 22:23:27.388971 [DEBUG] switch_core_state_machine.c:378 (sofia/internal/2146130149@127.0.0.1:5060) State CONSUME_MEDIA 2011-01-13 22:23:27.388971 [DEBUG] switch_core_state_machine.c:378 (sofia/internal/2146130149@127.0.0.1:5060) State CONSUME_MEDIA going to sleep 2011-01-13 22:23:27.389971 [DEBUG] sofia.c:4604 Channel sofia/internal/2146130149@127.0.0.1:5060 entering state [calling][0] 2011-01-13 22:23:27.389971 [DEBUG] sofia.c:4604 Channel sofia/internal/2146130149@127.0.0.1:5060 entering state [terminated][503] 2011-01-13 22:23:27.389971 [DEBUG] switch_channel.c:2455 (sofia/internal/2146130149@127.0.0.1:5060) Callstate Change RINGING -> HANGUP 2011-01-13 22:23:27.389971 [NOTICE] sofia.c:5244 Hangup sofia/internal/2146130149@127.0.0.1:5060 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2011-01-13 22:23:27.389971 [DEBUG] switch_channel.c:2471 Send signal sofia/internal/2146130149@127.0.0.1:5060 [KILL] The Speech Server is listening at TCP 0.0.0.0:5060 and Sofia Internal at TCP IP Addy of box also on port 5060. Any help or insight would be much appreciated. Cheers. bit

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How to set sip_callee_id_name in a conference
by Fabio Bigliardi 14 Jan '11

14 Jan '11

Hi all, I would like to configure a dialplan so that a tag with the name of the conference appears in the calling phone's display. Thanks a lot for your support. Best regards, F. Bigliardi

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CDR Frontend Report/GUI
by Joshua Foshee 14 Jan '11

14 Jan '11

I have a client that is wanting a polished commercial CDR GUI frontend. They have been looking at Fonality and trixbox pro reports and want something similar. I have attached address to some screen shots below. Does anyone have any suggestions on commercial or open source? If not I might be looking to put out a bounty if we can't find one that has all the features the client is looking for. Thanks, Josh http://cdnso.pbxtra.fonality.com/images/screenshots/800/ss-callcenter-ag entreport.jpg http://cdnso.pbxtra.fonality.com/images/screenshots/800/ss-powerfulrepor ting-calldistribution.jpg http://cdnso.pbxtra.fonality.com/images/screenshots/800/ss_powerful_repo rting.jpg http://web2.fonality.com/images/call_dist-snapshot.gif http://web2.fonality.com/images/call_dist-month.gif http://web2.fonality.com/images/call_dist-day.gif http://web2.fonality.com/images/call_dist-weekday.gif http://web2.fonality.com/images/call_dist-hour.gif http://web2.fonality.com/images/abandoned-detail-(as-admin).gif

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Peculiar behavior in FreeSWITCH when media streams for SRTP and RTP are offered in the same SDP
by Goutham BG 14 Jan '11

14 Jan '11

Hi, I am observing a peculiar behavior in FreeSWITCH-1.0.7 with SRTP. I have the following entry in my dialplan XML file: <extension name="IVR"> <condition field="destination_number" expression="^IVR$"> <action application="set" data="sip_secure_media=true"/> <action application="bridge" data="loopback/app=socket: 47.152.232.156:8084 async full"/> </condition> </extension> A Polycom SoundPoint IP 550 configured in "SRTP best effort" mode dials into this extension and is connected to the IVR. In this mode, the phone offers two media streams in the SDP of INVITE; 1st one for SRTP and the 2nd one for RTP. But the problem is that the media is established in SRTP in one way and RTP in the other way. The phone offers the following SDP in the INVITE message: v=0 o=- 1167766638 1167766638 IN IP4 47.152.232.149 s=Polycom IP Phone c=IN IP4 47.152.232.149 t=0 0 a=sendrecv m=audio 5040 RTP/SAVP 9 0 8 18 127 a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:svWhF9Uh0VonfhtmuvRLM4B9S6+ZEicc3hDd3dAQ a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 m=audio 5040 RTP/AVP 9 0 8 18 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 As we can see, there are two "m=" lines in the SDP of the offer; the 1st one for SRTP and the other one for RTP. FreeSWITCH-1.0.7 answers the call by sending 200OK with the following SDP: v=0 o=FreeSWITCH 1294817064 1294817065 IN IP4 47.152.232.156 s=FreeSWITCH c=IN IP4 47.152.232.156 t=0 0 m=audio 11552 RTP/SAVP 9 127 a=rtpmap:9 G722/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=silenceSupp:off - - - - a=ptime:20 a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:z5/m/v2U0negVFjfkCsXR/sNAo9fhBv+dspBIbtx m=audio 0 RTP/AVP 19 As we can see above, FreeSWITCH accepts the SRTP stream and rejects the RTP stream (by sending port as 0) in the SDP. The Polycom phone sends the media in SRTP as expected. But, FreeSWITCH sends the media in RTP to the phone even though it accepted SRTP in the answer (200OK). Please let me know if this is a bug or am I missing something here? The debug log of the call coming into FreeSWITCH is pasted here http://pastebin.freeswitch.org/14998 . If required I can also send the wireshark traces of this scenario captured in the system where FreeSITCH is running. Thanks Goutham B G

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Freeswitch & SNMP ( again , but with a difference )
by jay binks 14 Jan '11

14 Jan '11

has anyone done any work on getting SNMP monitoring of freeswitch ?? from looking at some other emails about this, it seems that there is not a heap of desire ( from developers point of view ) to have a FS SNMP module. Ive also noticed most requests for this type of monitoring are directed to ESL. so my thought, has anyone done any work on building a script / module for NET-SNMP ( http://net-snmp.sourceforge.net/ ) that uses ESL to fetch its data. so we would run snmpd on our FS boxes, with the FS-ESL module ( or script ) loaded into snmpd. from there SNMP requests would be handled by snmpd and the data would then be pulled over ESL directly from freeswitch. this seems like a decent solution that would not risk stability of FS by introducing yet another module. so I guess my question here is, has anyone done something like this ? what do you all think of this idea ? is there anyone who would like to contribute to this idea or offer other suggestions. oh yea, and why SNMP.. I duno... SNMP is just so easy to integrate into other monitoring systems. -- Sincerely Jay

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