August 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

"Missing local host" error message while FS establishing early media
by Anthony Brodi 07 Aug '16

07 Aug '16

Hi, Currently my FS is set up that way that it does not sent 183 Session Progress but 180 Ringing at once. It makes the calling side sometimes confused whether the call is being processed or not. To change this, I use pre_answer() before bridge, but then run onto this issue - FS jump up with a AUDIO RTP REPORTS ERROR: [Missing local host] as you can see in the log below. Can anyone help me on this. Thanks. EXECUTE sofia/internal/1000(a)Registar_IP.X.X.X pre_answer() [INFO] mod_dptools.c:1327 Sending early media [DEBUG] switch_core_media.c:2507 Set Codec sofia/internal/1000(a)Registar_IP.X.X.X PCMU/0 0 ms 160 samples 64000 bits 1 channels [DEBUG] switch_core_codec.c:111 sofia/internal/1000(a)Registar_IP.X.X.X Original read codec set to PCMU:0 [DEBUG] switch_core_media.c:5244 PROXY AUDIO RTP [sofia/internal/1000(a)Registar_IP.X.X.X] Softphone_local_IP.X.X.X:5930->Softphone_local_IP.X.X.X:5930 codec: 0 ms: 20 [ERR] switch_core_media.c:5844 AUDIO RTP REPORTS ERROR: [Missing local host] freeswitch@internal> sofia status profile internal ================================================================================================= Name internal Domain Name N/A Auto-NAT false DBName sofia_reg_internal Pres Hosts FS_external_IP.x.x.x,FS_external_IP.x.x.x Dialplan XML Context public Challenge Realm auto_from RTP-IP FS_external_IP.x.x.x SIP-IP FS_external_IP.x.x.x URL sip:mod_sofia@FS_external_IP.x.x.x:5060 BIND-URL sip:mod_sofia@FS_external_IP.x.x.x :5060;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN PCMU, PCMA,G729,G723,GSM,G722,OPUS CODECS OUT PCMU, PCMA,G729,G723,GSM,G722,OPUS TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA true ZRTP-PASSTHRU true AGGRESSIVENAT true

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Group Call
by Shawn Wheeler 06 Aug '16

06 Aug '16

I have an odd situation. I am using the mad boss group call. The phones are set to auto answer. When I make the call, some phone will connect but others will not. However, I can call each phone individually and the phone will answer. Thoughts? Thank you in advance Shawn

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Conference Call Limits
by Deepika Yadav 05 Aug '16

05 Aug '16

Hi, Can anyone tell me, what is the maximum number of mobile phone numbers that can be connected in a conference call and how many connected members can give same DTMF input simultaneously. I have a requirement where I need to call 22 phone numbers in a conference call and receive DTMF input(Digit 1) from these callers at once. I tested with 17 callers, when all of these 17 members pressed 1, my esl program received only DTMF event from only 8 callers. Just wanted to confirm if there are any sort of constraints regarding conference calls. Regards, Deepika

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uuid_dump returns blank
by Don Hawkins 05 Aug '16

05 Aug '16

Using the web API uuid_dump returns a blank page (unless the uuid does not exist then it returns a error) http://192.00.22.22:8080/api/uuid_dump?0c35eaeb-4082-4e2c-9421-ddf614e188ab… uuid_dump woth same uuid is working fine from fs_cli Is this a bug?

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Interesting stats ..
by Muhammad Naseer Bhatti 05 Aug '16

05 Aug '16

I did not knew FreeSWITCH is that old but looks interesting .. on an old version where enable-use-system-time is enabled for some testing. freeswitch@internal> status *UP 93 years, 92 days, 0 hours, 46 minutes, 41 seconds, 34 milliseconds, 220 microseconds* FreeSWITCH (Version 1.4.26 64bit) is ready 0 session(s) since startup 0 session(s) - peak 0, last 5min 0 0 session(s) per Sec out of max 30, peak 0, last 5min 0 1000 session(s) max min idle cpu 0.00/97.93 Current Stack Size/Max 240K/8192K -- Sent with Airmail

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Which module to use?
by Don Hawkins 05 Aug '16

05 Aug '16

Very simple situation... While on a call I want to press * to post to an external URL, I want to post global variables, ie. UUID. I have this working with httapi: <execute application="bind_digit_action" data="httapi,*,exec:httapi, http://myurl.com,aleg,aleg\"></execute> Only problem is, httapi expects XML to be returned and I don't actually want to return anything, I just want to send the information to the URL and for FreeSwitch to go on about it's business. What module is best for this job?

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Hostname vs IP in SDP
by Colin Morelli 05 Aug '16

05 Aug '16

Hey all, So it looks like WebRTC has issues dealing with hostnames in the SDP instead of IP addresses. Particularly, when including a hostname in the SDP there's no audio in either direction. This is the case in both the latest Firefox + Chrome browsers. This is particularly problematic for applications that are WebRTC + mobile, now that Apple is enforcing applications to support IPv6-only networks. Using hostnames in the record-route and SDP addresses meets the requirement since the OS will synthesize an IPv6 address when a DNS query is performed, but according to the documentation they strongly urge applications to avoid using address literals in their applications. Has anyone had success with methods to switch between advertising a hostname vs IP address in the SDP? Or other methods for solving this problem? Best, Colin

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Press 1 to callback
by Kamil Nigmatullin 05 Aug '16

05 Aug '16

Hi. Does anyone know how can I do th following: When client arrives at FS I put him into the callcenter queue. And at the time he wait I want client to be able to press 1. After pressing FS should played mp3 and hanhup. This event somehow shuld be fired in lua or something else so that our operator could call client back. Thanks -- Kamil Nigmatullin Skype: kamil.nigmatullin

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CURL from Lua Script
by Don Hawkins 05 Aug '16

05 Aug '16

root@debian:/usr/share/freeswitch/scripts# vi test.lua api = freeswitch.API(); api:execute("curl", "http://myurl.com/test3.php post name1=value1") Snatched the code straight from the the example shown at the URL below. FS does execute the script but never reaches my URL with the POST request. Did I miss something? https://freeswitch.org/confluence/display/FREESWITCH/mod_curl

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Re: [Freeswitch-users] Faxing transcoding issues.
by Andrew Cassidy 05 Aug '16

05 Aug '16

Sorry I wrote it while testing which is why the overall tone changes throughout. The without should should be without the execute_on_answer. I have a solution that works well enough, In that I have a documentable configuration that should work for all use cases on my particular implementation. If the user wants to use T38, they need to request it. On 5 August 2016 at 07:56, Andrew Cassidy <andee(a)bytz.co.uk> wrote: > Sorry I wrote it while testing which is why the overall tone changes > throughout. > > The without should should be without the execute_on_answer. > > I have a solution that works well enough, In that I have a documentable > configuration that should work for all use cases on my particular > implementation. If the user wants to use T38, they need to request it. > > On 4 Aug 2016 17:37, "Serge S. Yuriev" <me(a)nevian.org> wrote: > >> Hi, >> >> Your with and without t38 looks same for me - is this by intent? >> I'm a little puzzled with your mail - have you every success? >> >> On 04/08/16 17:47, Andrew Cassidy wrote: >> > Ok lets go into details a little more: >> > >> > User <--> FS <--> Provider >> > >> > When User sends a fax, provider detects tones and renegotiates with t38. >> > I can handle this case fine using sip_execute_on_image=t38_gateway >> > >> > Now, one way or another I also want the a-leg to be t38 if possible. Be >> > that via reinvite from FS or sepcifically requested by User. I can't >> > make this work at all. >> > >> > If FreeSWITCH renegotiates with User and User rejects T38, Freeswitch >> > does not reinvite with the original codecs. >> > >> > If User requests t38, fax negotiation fails in some cases, although that >> > may be irrelevant. >> > >> > So, here's my current configuration, that kind of works: >> > >> > <action application="export" data="fax_enable_t38=true" /> >> > <action application="export" data="sip_execute_on_image=t38_gateway >> self >> > nocng" /> >> > >> > To test the User end (in this case another FreeSWITCH box for testing): >> > >> > With T38: >> > >> > originate {execute_on_answer='t38_gateway >> > peer',fax_verbose=true,ignore_early_media=true,fax_ident=xxx >> x,fax_header='Test >> > Fax',absolute_codec_string=PCMA,origination_caller_id_number >> =xxxx}sofia/gateway/xxxx/xxxx >> > &txfax('/home/andrew/test_fax.tiff') >> > >> > Without T38: >> > >> > originate {execute_on_answer='t38_gateway >> > peer',fax_verbose=true,ignore_early_media=true,fax_ident=xxx >> x,fax_header='Test >> > Fax',absolute_codec_string=PCMA,origination_caller_id_number >> =xxxx}sofia/gateway/xxxx/xxxx >> > &txfax('/home/andrew/test_fax.tiff') >> > >> > However, the following fails: >> > >> > originate >> > {fax_enable_t38=true,fax_enable_t38_request=true,fax_verbose >> =true,ignore_early_media=true,fax_ident=xxxx,fax_header='Test >> > Fax',absolute_codec_string=PCMA,origination_caller_id_number >> =xxxx}sofia/gateway/xxxx/xxxx >> > &txfax('/home/andrew/test_fax.tiff') >> > >> > I think this configuration will work for now, the User end can decide >> > freely if they want to send t38 requests to FS or not. >> > >> > On 4 August 2016 at 14:39, Brian West <brian(a)freeswitch.org >> > <mailto:brian@freeswitch.org>> wrote: >> > >> > You also do NOT want pass thru set if you're doing this. >> > >> > On Thu, Aug 4, 2016 at 4:02 AM, Andrew Cassidy >> > <andrew(a)cassidywebservices.co.uk >> > <mailto:andrew@cassidywebservices.co.uk>> wrote: >> > >> > Hi all, >> > >> > Just realised the server I'm having issues with is an old wheezy >> > server running 1.4.26. I will test with latest and report back. >> > >> > Kind regards, >> > >> > On 4 August 2016 at 09:55, Andrew Cassidy >> > <andrew(a)cassidywebservices.co.uk >> > <mailto:andrew@cassidywebservices.co.uk>> wrote: >> > >> > Good morning all, >> > >> > I've been playing with faxing and t38 over the last few >> > days. I use a wholesale provider who detects and reinvites >> > for t38, and have implemented the same in my systems. >> > Passing t38 through unmodified is not an issue. >> > >> > The issue I do have, is in the event that the sender refuses >> > the reinvite, I'd like freeswitch to transcode. In theory, >> > this is set up correctly, but see this error message >> > repeated over and over and ultimately the fax fails: >> > >> > [DEBUG] mod_spandsp_fax.c:1756 >> > sofia/external/xxx(a)provider.com <mailto:xxx@provider.com> >> > Error decoding UDPTL (172 bytes) >> > >> > Every other scenario I have worked through is working fine. >> > >> > For reference, the fax-specific dialplan entries: >> > >> > <action application="export" data="fax_enable_t38=true"/> >> > <action application="export" >> > data="fax_enable_t38_request=true"/> >> > <action application="export" data="t38_passthru=true"/> >> > <action application="export" >> > data="nolocal:sip_execute_on_image=t38_gateway self" /> >> > <action application="set" >> > data="execute_on_answer=t38_gateway peer" /> >> > >> > Thanks in advance. >> > >> > -- >> > *Andrew Cassidy BSc (Hons) MBCS SSCA* >> > Managing Director >> > >> > 03303 880 960 >> > <tel:03303%20880%20960> andrew(a)cassidyweb.co.uk >> > <mailto:andrew@cassidywebservices.co.uk> >> www.cassidyweb.co.uk <http://www.cassidywebservices.co.uk> >> > >> > >> > >> > >> > -- >> > *Andrew Cassidy BSc (Hons) MBCS SSCA* >> > Managing Director >> > >> > 03303 880 960 <tel:03303%20880%20960> andrew(a)cassidyweb.co.uk >> > <mailto:andrew@cassidywebservices.co.uk> www.cassidyweb.co.uk >> > <http://www.cassidywebservices.co.uk> >> > >> > ___________________________________________________________ >> ______________ >> > Professional FreeSWITCH Consulting Services: >> > consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> >> > http://www.freeswitchsolutions.com >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://confluence.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users(a)lists.freeswitch.org >> > <mailto:FreeSWITCH-users@lists.freeswitch.org> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >> freeswitch-users >> > http://www.freeswitch.org >> > >> > >> > >> > >> > -- >> > >> > */Brian West/* >> > brian(a)freeswitch.org <mailto:brian@freeswitch.org> >> > >> > >> > */Twitter: @FreeSWITCH , @briankwest/* >> > http://www.freeswitchbook.com >> > http://www.freeswitchcookbook.com >> > https://www.gofundme.com/freeswitch_ubuntu >> > >> > Got Bugs? Report them here <https://freeswitch.org/jira>! | >> > Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> >> > >> > *T:*+19184209001 <tel:%2B19184209001> | *F:*+19184209002 >> > <tel:%2B19184209002> | *M:*+1918424WEST (9378) >> > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> > >> > >> > ___________________________________________________________ >> ______________ >> > Professional FreeSWITCH Consulting Services: >> > consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> >> > http://www.freeswitchsolutions.com >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://confluence.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users(a)lists.freeswitch.org >> > <mailto:FreeSWITCH-users@lists.freeswitch.org> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >> freeswitch-users >> > http://www.freeswitch.org >> > >> > >> > >> > >> > -- >> > *Andrew Cassidy BSc (Hons) MBCS SSCA* >> > Managing Director >> > >> > 03303 880 960 andrew(a)cassidyweb.co.uk >> > <mailto:andrew@cassidywebservices.co.uk> www.cassidyweb.co.uk >> > <http://www.cassidywebservices.co.uk> >> > >> > >> > ____________________________________________________________ >> _____________ >> > Professional FreeSWITCH Consulting Services: >> > consulting(a)freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://confluence.freeswitch.org >> > http://www.cluecon.com >> > >> > 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 >> > >> >> -- >> Serge S. Yuriev >> Lead VoIP engineer >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director 03303 880 960 andrew(a)cassidyweb.co.uk <andrew(a)cassidywebservices.co.uk> www.cassidyweb.co.uk <http://www.cassidywebservices.co.uk>

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