June 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

SIP 500 overlappting requests
by Phil Quesinberry 19 Jun '14

19 Jun '14

We're having intermittent problems receiving calls from a particular DID. The basic scenario is as follows: Carrier sends invite w/SDP We respond w/OK and SDP when call is answered Carrier sends another invite with incremented sequence number and updated SDP (aren't they supposed to ACK/respond before doing this?) We send SIP 500 - overlapping requests Carrier sends BYE - terminating the call Sometimes (rarely) the call completes, most of the time FS sends a 500 and doesn't get the call. This just started happening recently and does not occur with any other DIDs, including other DIDs from the same carrier. I updated to the latest stable GIT to see if that might solve the problem but I'm pretty sure this is on the carrier's end. They're trying to blame it on us since we're sending the 500 and suggested that we allow overlapping requests. I'm assuming that their end needs to acknowledge our answer OK/SDP before sending their own but if that's the problem then it raises another question in my mind: since the SIP transport is generally UDP, what if we don't receive their ACK before receiving their subsequent invite/SDP? This could certainly happen and would cause the call to be rejected if the above is true. I updated to the latest stable GIT last night to rule that out as a possible cause of the problem. Is there a way to get FS to allow a limited number of overlapping requests as a possible workaround? If someone thinks this looks like it could be an FS bug, I'll file a Jira. Here is a trace which shows this happening: http://pastebin.freeswitch.org/22769 Thanks, Phil Quesinberry Q Systems Engineering, Inc. Embedded Hardware/Software Development (410) 969-8002 http://www.qsystemsengineering.com <http://www.qsystemsengineering.com/>

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BLF no longer display actual registration status of phones
by Nandy Dagondon 19 Jun '14

19 Jun '14

We have a problem in BLF indicators - disconnected phones still show as registered (solid green) in the displays. It's an Atcom AT640P phone but it worked before. FS version 1.2.13 64-bit on CentOS 6 2.6.32. How can I trace the problem? ESL, databases, etc. Thanks for any help/hint. Note: Please ignore my earlier post re "BLF Troubleshooting" because this is the real issue. /Nandy

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Can not find hangup log in the freeswitch logs
by rockxsj 19 Jun '14

19 Jun '14

Hi,I notice that the cdr is less than my expect.So I look up the logs of freeswitch,and I find some call did not get into CS_DESTROY state,the two legs of the call is below: fs.log.2014-05-28-00-00-01.1:4d57ac06-52fa-4ef5-956c-c1cc1db8ad40 2014-05-27 19:02:50.480500 [NOTICE] switch_channel.c:1053 New Channel sofia/internal/4000001234(a)sip.whoknowme.cn [4d57ac06-52fa-4ef5-956c-c1cc1db8ad40] fs.log.2014-05-28-00-00-01.1:4d57ac06-52fa-4ef5-956c-c1cc1db8ad40 2014-05-27 19:02:50.600479 [INFO] mod_dialplan_xml.c:558 Processing GY%B3%E0%C1%FA <4000001234>->018876590253 in context default fs.log.2014-05-28-00-00-01.1:4d57ac06-52fa-4ef5-956c-c1cc1db8ad40 2014-05-27 19:02:50.780497 [WARNING] switch_core_media.c:2548 NO candidate ACL defined, Defaulting to wan.auto fs.log.2014-05-28-00-00-01.1:4d57ac06-52fa-4ef5-956c-c1cc1db8ad40 2014-05-27 19:02:50.780497 [NOTICE] switch_core_media.c:2586 Save audio Candidate cid: 1 proto: udp type: host addr: 192.168.155.1:56436 fs.log.2014-05-28-00-00-01.1:4d57ac06-52fa-4ef5-956c-c1cc1db8ad40 2014-05-27 19:02:50.780497 [NOTICE] switch_core_media.c:2586 Save audio Candidate cid: 2 proto: udp type: host addr: 192.168.155.1:56436 fs.log.2014-05-28-00-00-01.1:4d57ac06-52fa-4ef5-956c-c1cc1db8ad40 2014-05-27 19:02:50.780497 [NOTICE] switch_core_media.c:2586 Save audio Candidate cid: 1 proto: udp type: host addr: 192.168.1.2:56437 fs.log.2014-05-28-00-00-01.1:4d57ac06-52fa-4ef5-956c-c1cc1db8ad40 2014-05-27 19:02:50.780497 [NOTICE] switch_core_media.c:2586 Save audio Candidate cid: 2 proto: udp type: host addr: 192.168.1.2:56437 fs.log.2014-05-28-00-00-01.1:4d57ac06-52fa-4ef5-956c-c1cc1db8ad40 2014-05-27 19:02:50.780497 [NOTICE] switch_core_media.c:2581 Choose audio Candidate cid: 1 proto: udp type: srflx addr: 120.32.104.13:9115 fs.log.2014-05-28-00-00-01.1:4d57ac06-52fa-4ef5-956c-c1cc1db8ad40 2014-05-27 19:02:50.780497 [NOTICE] switch_core_media.c:2581 Choose audio Candidate cid: 2 proto: udp type: srflx addr: 120.32.104.13:9115 fs.log.2014-05-28-00-00-01.1:4d57ac06-52fa-4ef5-956c-c1cc1db8ad40 2014-05-27 19:02:50.780497 [NOTICE] switch_core_media.c:2586 Save audio Candidate cid: 1 proto: udp type: srflx addr: 120.32.104.13:9117 fs.log.2014-05-28-00-00-01.1:4d57ac06-52fa-4ef5-956c-c1cc1db8ad40 2014-05-27 19:02:50.780497 [NOTICE] switch_core_media.c:2586 Save audio Candidate cid: 2 proto: udp type: srflx addr: 120.32.104.13:9117 fs.log.2014-05-28-00-00-01.1:4d57ac06-52fa-4ef5-956c-c1cc1db8ad40 2014-05-27 19:02:50.780497 [NOTICE] switch_core_media.c:2710 setting remote audio ice addr to 120.32.104.13:9115 based on candidate fs.log.2014-05-28-00-00-01.1:4d57ac06-52fa-4ef5-956c-c1cc1db8ad40 2014-05-27 19:02:50.780497 [NOTICE] switch_core_media.c:2730 setting remote rtcp audio addr to 120.32.104.13:9115 based on candidate fs.log.2014-05-28-00-00-01.1:4d57ac06-52fa-4ef5-956c-c1cc1db8ad40 2014-05-27 19:02:50.780497 [INFO] switch_core_media.c:5016 Activating Audio ICE fs.log.2014-05-28-00-00-01.1:4d57ac06-52fa-4ef5-956c-c1cc1db8ad40 2014-05-27 19:02:50.780497 [NOTICE] switch_rtp.c:3775 Activating RTP audio ICE: gdzTwX2PR4YWwRvr:Qr3qnaiYtrc9YirV 120.32.104.13:9115 fs.log.2014-05-28-00-00-01.1:4d57ac06-52fa-4ef5-956c-c1cc1db8ad40 2014-05-27 19:02:50.780497 [INFO] switch_core_media.c:5059 Activating RTCP PORT 9115 fs.log.2014-05-28-00-00-01.1:4d57ac06-52fa-4ef5-956c-c1cc1db8ad40 2014-05-27 19:02:50.780497 [INFO] switch_core_media.c:5067 Skipping RTCP ICE (Same as RTP) fs.log.2014-05-28-00-00-01.1:4d57ac06-52fa-4ef5-956c-c1cc1db8ad40 2014-05-27 19:02:50.830486 [INFO] switch_rtp.c:3151 Activating Audio Secure RTP SEND fs.log.2014-05-28-00-00-01.1:4d57ac06-52fa-4ef5-956c-c1cc1db8ad40 2014-05-27 19:02:50.840496 [INFO] switch_rtp.c:3129 Activating Audio Secure RTP RECV fs.log.2014-05-28-00-00-01.1:4d57ac06-52fa-4ef5-956c-c1cc1db8ad40 2014-05-27 19:02:50.840496 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/4000001234(a)sip.whoknowme.cn! fs.log.2014-05-28-00-00-01.1:4d57ac06-52fa-4ef5-956c-c1cc1db8ad40 2014-05-27 19:02:51.040498 [NOTICE] switch_rtp.c:1132 Auto Changing stun/rtp/dtls port from 120.32.104.13:9115 to 120.32.104.13:9117 fs.log.2014-05-28-00-00-01.1:4d57ac06-52fa-4ef5-956c-c1cc1db8ad40 2014-05-27 19:02:58.720594 [NOTICE] switch_ivr_bridge.c:496 Channel [sofia/internal/4000001234(a)sip.whoknowme.cn] has been answered fs.log.2014-05-28-00-00-01.1:2127f5f1-fa2d-469d-ad0a-8df709d484ec 2014-05-27 19:02:50.860494 [NOTICE] switch_channel.c:1053 New Channel sofia/internal/018876590253 [2127f5f1-fa2d-469d-ad0a-8df709d484ec] fs.log.2014-05-28-00-00-01.1:2127f5f1-fa2d-469d-ad0a-8df709d484ec 2014-05-27 19:02:51.570502 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/018876590253! fs.log.2014-05-28-00-00-01.1:2127f5f1-fa2d-469d-ad0a-8df709d484ec 2014-05-27 19:02:58.700582 [NOTICE] sofia.c:6955 Channel [sofia/internal/018876590253] has been answered fs.log.2014-05-28-00-00-01.1:2127f5f1-fa2d-469d-ad0a-8df709d484ec 2014-05-27 19:04:32.330784 [WARNING] sofia.c:8096 IGNORE INFO DTMF(6) (This channel was not configured to use INFO DTMF!) Is this may be a bug?

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Diversion header
by Chris Young 19 Jun '14

19 Jun '14

Hi all, We have a FreeSWITCH server connected to a Cisco UC560 which is using CUE for voicemail. We are experiencing problems with calls which are redirected to the voicemail system because the INVITE sent by FreeSWITCH does not contain the Diversion header present in the preceding 302 response. Is there something specific I need to configure to get this included? I've tried to set up manual redirection on the SIP profile, as per the wiki, so that I could export the header using sip_h_Diversion but it doesn't seem to be working - the call does not enter the 'redirected' context and continues to be processed internally. Is there another way of achieving this? Thanks, Chris Chris Young Software Engineer [cid:image737b08.PNG@6a440e0d.419a719a] t: +44 118 943 9249 e: Chris.Young(a)enghouse.com<mailto:Chris.Young@enghouse.com> w: www.enghouseinteractive.co.uk<http://www.enghouseinteractive.co.uk> [cid:image72f27b.PNG@7bde397e.49b56865]

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Multiple SIP-registrations and leg-specific variables.
by Alex Lake 19 Jun '14

19 Jun '14

Got a situation where an extension has two phones registered, but my dialplan is a little too simplistic to handle: i.e my bridge string has something like this in it: [tenant_id=0024,b_ext=201,accept_mode=Direct,leg_timeout=15]${sofia_contact(0024312)}' But ${sofia_contact(0024312)} expands to "sofia/internal/sip:0024307@84.92.90.149:59031;transport=udp,sofia/internal/sip:0024307@37.157.38.162:51910" so the leg-specific parameters only affect the first registration. I suppose I could write something in Lua to work it all out properly, but it might be more elegant if there was something like this: [tenant_id=0024,b_ext=201,accept_mode=Direct,leg_timeout=15](${sofia_contact(0024312)}) Where the brackets mean to apply the [] parameters to anything that comes out of the expression. Is this available?!

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freeswitch on wifi
by Shlomi Agiv 19 Jun '14

19 Jun '14

We intend to put freeswitch on an arm powered portable device, which will connect using a usb gsm stick and will serve as a wifi/gsm router so, a call will connect two endpoints - one to the sip client through wifi, the other through mod_gsmopen to the gsm. How will the voice quality be? which voice codec is most recommended?

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Dangerous Demos Coming To ClueCon 2014!!
by Ken Rice 18 Jun '14

18 Jun '14

DANGEROUS DEMOS are coming to ClueCon! Our friends over at Truphone are at it again! Come to ClueCon to see the other Dangerous Demo, and while you are there show off your Dangerous Demo! Running for the first time in it¹s new and improved format, DANGEROUS DEMOS offers developers and hackers a chance to show off examples of their work in front of a live ClueCon audience! Competitors are invited to present a demonstration of something interesting that they have built. * Maximum demo duration will be 4 minutes * Bonus marks awarded for demos which > * Include FreeSWITCH > * Contain elements of uncertainty and/or danger > * Entertain and amuse the audience > * Present clear revenue generating potential > * Incorporate external parties/services > * Operate at multiple levels * Prizes: There will be at least THREE prizes (donated by Truphone), with the prospect of a FOURTH should special conditions be met. * Scoring will include ³Audience Participation² utilising a new realtime voting system (which would qualify as Dangerous Demo in it¹s own right) * Entries, containing a one or two sentence describing broadly what the demo may contain need to be sent to James Body (james.body(a)truphone.com) prior to the competition. Register for ClueCon today @ https://cluecon.com/

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questions about addons
by Jason Holden 18 Jun '14

18 Jun '14

Does Anyone have a recommendation for the following. 1. external speaker for paging with freeswitch, only transmit is required. 2. an application that may be loaded on a computer providing an operator full call routing capabilities. Thanks.

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DNS name as the public IP respond 403
by Shahzad Bhatti 18 Jun '14

18 Jun '14

Hi Brian, what do you mean by provider please explain a bit about provider is it about route provider or something else . Regards Shahzad Bhatti > ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 18 Jun 2014 08:21:44 -0500 > Subject: Re: [Freeswitch-users] DNS name as the public IP respond 403 > Again you need to talk to your provider they are the ones responding with > 403. > -- > Brian West > brian(a)freeswitch.org > > > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > On Jun 17, 2014, at 1:08 PM, Shahzad Bhatti <shahzad.bhatti(a)g-r-v.com> > wrote: > > here i post 2 sip traces first for successive registration with same >> credentials but using static ip >> > with headings as > > on line 40 > Sip Trace of Successive Call using Public IP > and error sip trace on line 288 > SIP TRACE OF USING DNS CALL FOR REGISTER (having problem) > > any further information is required to trace the issue i will happy to > send also > > regards > > Shahzad Bhatti > > > > _______________________________________________ > 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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No Ringback on few calls
by Nagalenoj H. 18 Jun '14

18 Jun '14

Hi, I am facing an inconsistent behavior in getting Ringback for calls. When I call to few numbers, I am not getting ringback tone. I did capture the packets and checked the difference between the call which I get RBT and failed case. I couldn't figure out what is stopping Freeswitch to provide '183 session progress' to the application after receiving '180 ringing' from the server on the other side. I checked the other server and it is sending the tone to FS, it is in FS machine the tone it not forwarded due to some reason. In Freeswitch log, there is no log for "Sending early media" in case of failure case. I've attached the packet capture for the calls. It has both working and non-working case. Please help me in debugging this issue.

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