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Re: [Freeswitch-users] Improving freeswitch init script for fedora/centos/redhat/amazon-linux
by Robert Mundkowsky 09 Nov '15

09 Nov '15

Not sure if systemd start and stop scripts work on Fedora, but the work nicely on Debian. I copied the ones from: https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/71445… Also I think the wiki site for Freeswitch is old. Confluence site is newer. Robert -------- Original message -------- From: Jun Sun <jsun(a)junsun.net> Date: 11/08/2015 8:46 PM (GMT-05:00) To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: [Freeswitch-users] Improving freeswitch init script for fedora/centos/redhat/amazon-linux I'm porting freeswitch to amazon linux running on an EC2 instance. And I have installed the following fedora init script to automatically start/stop freeswitch. https://wiki.freeswitch.org/wiki/Freeswitch_init However I notice that this script does not work reliably. Specifically restart does not always do the job. I looked in and did some experiments. It appears "freeswitch -stop" does not kill the daemon reliably. It is especially so when freeswitch is just starting. So I add some code to do the killing more persistently. I also added "status" command. It is tested on amazon linux. I'm sure it will work on fedora and centos. Hopefully it can be of help to others in the similar suite. Jun

2 1

freeswitch.org down
by Mario G 06 Nov '15

06 Nov '15

In case none noticed yet. Was testing and updating the wiki when it went down, other sites work but not freeswitch.org <http://freeswitch.org/> (multiple computers). I lost all my edits when I saved!

4 3

Delay in Verto Media
by Danny Gershman 06 Nov '15

06 Nov '15

I'm seeing an odd delay in the DTLS connecting from HANDSHAKE to SETUP. What could be causing this? This doesn't happen from inside the firewall. During this time there is obviously no media or feedback on the client-side. 2015-10-29 14:43:25.651206 [ALERT] mod_verto.c:604 WRITE 24.55.19.11:52010 [{ "jsonrpc": "2.0", "id": 147, "method": "verto.answer", "params": { "callID": "f1ab2564-2996-1d5b-2c91-5f4f193253c0", "sdp": "v=0\no=FreeSWITCH 1446113261 1446113262 IN IP4 45.37.10.14\ns=FreeSWITCH\nc=IN IP4 45.37.10.14\nt=0 0\na=msid-semantic: WMS AX8g7czxgdq67ZTBW13sTKfklsnSeCBF\nm=audio 30944 UDP/TLS/RTP/SAVPF 9 126\na=rtpmap:9 G722/8000\na=rtpmap:126 telephone-event/8000\na=silenceSupp:off - - - -\na=ptime:20\na=sendrecv\na=fingerprint:sha-256 81:99:80:23:BE:2D:3D:AF:9E:41:44:C3:63:E1:0C:4F:6B:CD:18:03:75:18:2C:D4:33:CE:3D:47:4F:7D:13:A4\na=setup:active\na=rtcp-mux\na=rtcp:30944 IN IP4 45.37.10.14\na=ice-ufrag:hwV071aiko1pQsU6\na=ice-pwd:TwkmyVWDNtFtJ9WRwIHtzICK\na=candidate:5333896332 1 udp 659136 45.37.10.14 30944 typ host generation 0\na=ssrc:3393214333 cname:qAMnDbLKe1GcQqDD\na=ssrc:3393214333 msid:AX8g7czxgdq67ZTBW13sTKfklsnSeCBF a0\na=ssrc:3393214333 mslabel:AX8g7czxgdq67ZTBW13sTKfklsnSeCBF\na=ssrc:3393214333 label:AX8g7czxgdq67ZTBW13sTKfklsnSeCBFa0\n" } }] 2015-10-29 14:43:25.711338 [ALERT] mod_verto.c:1359 READ 24.55.19.11:52010 [{ "jsonrpc": "2.0", "id": 147, "result": { "method": "verto.answer" } }] 2015-10-29 14:43:56.871339 [INFO] switch_rtp.c:3044 Changing audio DTLS state from HANDSHAKE to SETUP 2015-10-29 14:43:56.871339 [INFO] switch_rtp.c:2953 audio Fingerprint Verified. 2015-10-29 14:43:56.871339 [INFO] switch_rtp.c:3543 Activating Audio Secure RTP SEND 2015-10-29 14:43:56.871339 [DEBUG] switch_core_sqldb.c:2607 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2015-10-29 14:43:56.871339 [INFO] switch_rtp.c:3521 Activating Audio Secure RTP RECV 2015-10-29 14:43:56.871339 [INFO] switch_rtp.c:2993 Changing audio DTLS state from SETUP to READY 2015-10-29 14:43:56.871339 [DEBUG] switch_core_sqldb.c:2607 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2015-10-29 14:43:56.911151 [DEBUG] switch_rtp.c:1797 rtcp_stats_init: audio ssrc[1172837471] base_seq[1534

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Version 1.4.23 - Timeof day Ranges
by Jonathan Hunter 06 Nov '15

06 Nov '15

Hi Guys, I am currently testing following the guide at; https://freeswitch.org/confluence/display/FREESWITCH/Time+of+Day+and+Holida… Where it states; Blocks Comma–separated values allow discrete blocks of time to be specified. <condition time-of-day="08:00-12:00,13:00-17:00">  I am trying to use it with the comma-seperated values and its not matching. Im basically using the same as above; <condition wday="6"/> <condition time-of-day="08:00-12:00,13:00-17:00" break="on-true"> If I test without the comma, it matches for the range specified. Can I just ask is it included in the software version Im using? And to use does anything else have to be used/implemented for it to work? Thanks Jon

1 0

mod_avmd stops
by francesco allertsen 06 Nov '15

06 Nov '15

Hello everyone, I was testing the avmd mod, and I found a problem. The voicemail detection starts, but after few seconds (2/3) it stops for 4/5 seconds, and then "misses" the voicemail beep. I'm using this small script for testing the avmd in python, partly taken from an old email --- import sys from freeswitch import * def input_callback(session, what, obj): consoleLog("info", "Callback") return "TRUE" def handler(session, args): session.answer() callback = session.setInputCallback("input_callback") session.execute("avmd", "start") consoleLog("info", "AVMD start\n") session.execute("sleep", "20000") consoleLog("info", "AVMD status %s\n" % session.getVariable('avmd_detect')) session.execute("avmd", "stop") consoleLog("info", "AVMD stop\n") session.hangup() --- And you can see an example of the console log here --- 2015-11-05 16:44:04.609583 [DEBUG] mod_avmd.c:538 <<< AVMD v=0.014491 f=0.364007 463.468433Hz sma=0.332653 sqa=0.125149 >>> 2015-11-05 16:44:04.649600 [DEBUG] mod_avmd.c:538 <<< AVMD v=0.008135 f=0.300647 382.795802Hz sma=0.030065 sqa=0.009039 >>> 2015-11-05 16:44:04.669598 [DEBUG] mod_avmd.c:538 <<< AVMD v=0.013395 f=0.277465 353.279229Hz sma=0.057811 sqa=0.016738 >>> 2015-11-05 16:44:04.669598 [DEBUG] mod_avmd.c:538 <<< AVMD v=0.019046 f=0.322899 411.127973Hz sma=0.090101 sqa=0.027164 >>> 2015-11-05 16:44:04.689594 [DEBUG] mod_avmd.c:538 <<< AVMD v=0.022637 f=0.323557 411.965056Hz sma=0.122457 sqa=0.037633 >>> 2015-11-05 16:44:04.689594 [DEBUG] mod_avmd.c:538 <<< AVMD v=0.024591 f=0.336631 428.611912Hz sma=0.156120 sqa=0.048965 >>> 2015-11-05 16:44:04.709584 [DEBUG] mod_avmd.c:538 <<< AVMD v=0.023547 f=0.309434 393.983412Hz sma=0.187063 sqa=0.058540 >>> 2015-11-05 16:44:04.709584 [DEBUG] mod_avmd.c:538 <<< AVMD v=0.020918 f=0.326125 415.235059Hz sma=0.219676 sqa=0.069176 >>> 2015-11-05 16:44:04.729583 [DEBUG] mod_avmd.c:538 <<< AVMD v=0.015642 f=0.275036 350.186756Hz sma=0.247179 sqa=0.076740 >>> 2015-11-05 16:44:09.709590 [DEBUG] mod_avmd.c:538 <<< AVMD v=0.055515 f=0.785391 999.990286Hz sma=0.078539 sqa=0.061684 >>> 2015-11-05 16:44:09.769592 [DEBUG] mod_avmd.c:538 <<< AVMD v=0.055515 f=0.785391 999.990286Hz sma=0.078539 sqa=0.061684 >>> 2015-11-05 16:44:09.809600 [DEBUG] mod_avmd.c:538 <<< AVMD v=0.220715 f=1.566012 1993.908596Hz sma=0.156601 sqa=0.245239 >>> 2015-11-05 16:44:09.829595 [DEBUG] mod_avmd.c:538 <<< AVMD v=0.251632 f=0.785391 999.990286Hz sma=0.235140 sqa=0.306923 >>> 2015-11-05 16:44:09.869585 [DEBUG] mod_avmd.c:538 <<< AVMD v=0.055515 f=0.785391 999.990286Hz sma=0.078539 sqa=0.061684 >>> 2015-11-05 16:44:09.889639 [DEBUG] mod_avmd.c:538 <<< AVMD v=0.098694 f=0.785391 999.990286Hz sma=0.157078 sqa=0.123368 >>> 2015-11-05 16:44:10.329624 [DEBUG] mod_avmd.c:538 <<< AVMD v=0.170815 f=1.377658 1754.088297Hz sma=0.137766 sqa=0.189794 >>> 2015-11-05 16:44:10.349620 [DEBUG] mod_avmd.c:538 <<< AVMD v=0.055515 f=0.785391 999.990286Hz sma=0.078539 sqa=0.061684 >>> 2015-11-05 16:44:10.349620 [DEBUG] mod_avmd.c:538 <<< AVMD v=0.166566 f=1.201495 1529.791460Hz sma=0.198689 sqa=0.206043 >>> 2015-11-05 16:44:10.389591 [DEBUG] mod_avmd.c:538 <<< AVMD v=0.043729 f=0.697050 887.512119Hz sma=0.069705 sqa=0.048588 >>> 2015-11-05 16:44:10.389591 [DEBUG] mod_avmd.c:538 <<< AVMD v=0.056064 f=0.455676 580.184924Hz sma=0.115273 sqa=0.069352 >>> 2015-11-05 16:44:10.409623 [DEBUG] mod_avmd.c:538 <<< AVMD v=0.091352 f=1.007482 1282.766487Hz sma=0.100748 sqa=0.101502 >>> 2015-11-05 16:44:10.429621 [DEBUG] mod_avmd.c:538 <<< AVMD v=0.171934 f=1.064772 1355.710235Hz sma=0.207225 sqa=0.214876 >>> --- As you can see in the timestamps, it stopped for 5 seconds, and during those 5 seconds probably there was the "beep" which was not caught. I have checked in the wiki, but I don't see any specific configuration about the avmd, so I don't know where to look for anymore, and online I couldn't find any solution. If you need more informations I'm happy to provide so. Thanks a lot. Best Regards, Francesco

3 2

MIKEY-PK support
by Sergey Safarov 06 Nov '15

06 Nov '15

Hi According this message <http://lists.freeswitch.org/pipermail/freeswitch-users/2008-January/029822.…> to support MIKEY key exchange is required library with compatible licence. Now I am not find MIKEY support in source code. Are is posible to use libsrtp <http://srtp.sourceforge.net/license.html> to implement MIKEY key exchange? Sergey

4 7

Talk -Notalk events
by Mitja Sosic 06 Nov '15

06 Nov '15

Hello everyone, we are trying to implement a module, which would monitor for the silence during the call. If silence detected inside some thresholds, it should disconnect the call. We have started with VAD settings () in sip profile, and setting the variables in the dialplan: <action application="export" data="fire_talk_events=true"/> <action application="export" data="fire_not_talk_events=true"/> We are listening to all events. It works but only for the B channel. No matter what we do we cannot produce talk/notalk events for the A leg. It seems like a sofia bug ... Somebody any experience on that ?? Thx! Mitja

1 0

Re: [Freeswitch-users] New FreeSWITCH monitoring dashboard - Open Source
by apappas 06 Nov '15

06 Nov '15

Hi Guillermo,Can you help me reproduce a call in your way?BTW I also connect through ESL so is not the cdrs the issue but the messages generated in every call you make. Right now I get for two legged calls, 2 answers and 2 hangups. With these messages I track the state of the call. Regards,Alex Sent from my Samsung device -------- Original message -------- From: Guillermo Ruiz Camauer <grcamauer(a)gmail.com> Date: 11/5/2015 22:07 (GMT+02:00) To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] New FreeSWITCH monitoring dashboard - Open Source Alex, I have an external process that connects to FS through ESL and dials outbound calls and plays IVR prompts (phone surveys). They are all one legged calls (FS dialing out). I think that the problem is that my external process handles CDRs, not FS. Regards, Guillermo On Thu, Nov 5, 2015 at 3:32 PM, Alex Pappas <apappas(a)vieras.eu> wrote: Hi Guillermo, Can you provide me an example on how you use FS? Ideally you could send me the events generated from a call, otherwise you could describe the scenario of the call in words so I can reproduce.. Cheers,Alex On Thu, Nov 5, 2015 at 6:12 PM, Guillermo Ruiz Camauer <grcamauer(a)gmail.com> wrote: Alex, Thanks for getting back. Please let me know if you ever get it to work for one legged calls, as that is all we do! I didn't think you could extract any RTP quality measure from the heartbeat. Probably need to do some packet sniffing for that. Regards, Guillermo On Wed, Nov 4, 2015 at 4:22 AM, Alex Pappas <apappas(a)vieras.eu> wrote: Hi Guillermo, First of all thank you for your feedback! Concerning 1, ASR, ACD and RTP Quality should work only for completed calls. If your where getting empty table that means that the calls are not terminated normally.That also goes to point 2 where you are testing with one leg calls (IVR). We have tested the system with calls that are coming inside FS and they are terminated from an other Provider. Concerning 3, We are aware of this and in the next releases we will introduce the customer and the provider by IP, prefix or both. Concerning 4,5 we are getting these numbers from the FS Heartbeat. maybe we have to reconcider these two.. We are also thinking to add functionality for extra FS boxes in the near future. Concerning the connection to port 8021, you had problem because it was hardcoded from us :(. I will change this and let you know.. Thank you for your feedback, Alex On Tue, Nov 3, 2015 at 12:05 AM, Guillermo Ruiz Camauer <grcamauer(a)gmail.com> wrote: Alex / Nathan, Telneting to port 5000 works OK. It seems that the program is not accepting the PORT and always going to port 8021. I have changed my configuration on the FS side from port 5000 to port 8021 and now it connects! I have placed several hundred calls and have the following comments / questions: 1) View Details works for Concurrent Calls and for Failed Calls, but NOT for ASR, ACD or RTP Quality. The latter 3 always return Invalid JSON Response. I have inspected using browser tools (Chrome) and the response is EMPTY in all 3 cases. 2) Concurrent Calls seems to be an ever increasing number. It never goes back down, even when the box is idle. Note that when the box is idle, DETAILS returns 0 records which is OK, but the number displayed in the Summary is always non-zero, and increasing. If it matters, all my calls are one legged outbound calls (Outbound IVR). 3) Because my SIP provider requires me to prepend an account number before all dial strings, your software thinks all my calls are to BRUNEI... There should be some way to strip the first n digits so that it reads the actual dialed number. 4) Sessions/Sec was always stuck at 120. No matter what the box was doing. 5) RTP Quality was always 100. Aside from these problems, the program seems very responsive and well thought out. It will be a great tool to monitor several FS boxes. Regards, Guillermo On Mon, Nov 2, 2015 at 4:34 PM, Alex Pappas <apappas(a)vieras.eu> wrote: You could also try to telnet your FS instance from a different box to port 5000. If that works or not could tell us a lot. On Mon, Nov 2, 2015 at 9:14 PM, Alex Pappas <apappas(a)vieras.eu> wrote: Hi Guillermo,Can you open port 8021 just for testing purposes? That could eliminate some possible confusion. Thank you,Alex On Mon, Nov 2, 2015 at 6:31 PM, Guillermo Ruiz Camauer <grcamauer(a)gmail.com> wrote: Nathan, I think that it is not taking the port number. I have opened ESL on port 5000, but the program complains that it cannot connect to port 8021. As far as I can tell, port 8021 has something to do with remote hash from mod_hash. Does that module need to be configured? Regards, Guillermo On Sat, Oct 31, 2015 at 5:12 AM, apappas <apappas(a)gnostix.gr> wrote: Nathan thanks for the reply,Probably the problem is that the connection is not open, but still you are receiving the wrong error. I will have a look.. Let us know if you were able to connect. Regards,Alex Sent from my Samsung device -------- Original message -------- From: Nathan Neulinger <nneul(a)mst.edu> Date: 10/31/2015 03:05 (GMT+02:00) To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] New FreeSWITCH monitoring dashboard - Open Source Make sure you've enabled remote ESL socket connections. Normally it's bound to localhost only. autoload_configs/event_socket.conf.xml -> listen-ip -- Nathan On 10/30/2015 04:12 PM, Guillermo Ruiz Camauer wrote: > Thank you for publishing this! I have looked at the on-line demo and decided to give it a try. > > I have installed on a server and I am trying to connect to another server which has FS installed. > In the SETUP-CONNECTIONS page I am putting the IP address of my FS server (local LAN), 5000 for port and the ESL > password. When I try to login I get an error about not being able to connect to port 8021. What am I missing? > > Guillermo > > On Fri, Oct 30, 2015 at 4:20 PM, Alex Pappas <apappas(a)vieras.eu <mailto:apappas@vieras.eu>> wrote: > > For a start do you think you can use the current version? Is in any help for your installations? > We need to get an overall feedback so we understand what needs to be developed first. We will create a priority list > with your features. > > Guys our joy is this software to be useful and be used by you, so please let us know about the above. > > Cheers, > Alex > > > On Fri, Oct 30, 2015 at 12:17 AM, Sergey Safarov <s.safarov(a)gmail.com <mailto:s.safarov@gmail.com>> wrote: > > Very interest idea. > For me will be helpful: > > 1. multi tenant support; > 2. callcenter statistics > 1. query length, average query wait time, query received calls, query answered calls, query abandoned calls; > 2. agent statistic like talk time, answered calls, non answered calls; > 3. oAuth support > > > On Fri, Oct 30, 2015 at 12:04 AM, Alex Pappas <apappas(a)vieras.eu <mailto:apappas@vieras.eu>> wrote: > > Hey guys, > > We would love some feedback. What kind of features do you need to see in real time? How can we improve? > > Many thanks, > Alex > > On Thu, Oct 29, 2015 at 9:42 PM, servtelar <servtelar(a)gmail.com <mailto:servtelar@gmail.com>> wrote: > > Niiice! Congrats! > > >> On Oct 29, 2015, at 2:59 PM, apappas <apappas(a)gnostix.gr <mailto:apappas@gnostix.gr>> wrote: >> >> >> >> Dear all, >> >> For the past 4 months I've been working with two other colleagues (in Cc) on an OpenSource project to >> monitor the FreeSWITCH Softswitch in real-time via ESL events. >> >> Our background is in Telecoms so that is where we're coming from with the dashboard: You can see live >> call numbers, and you can click and drill down to see call details. We also provide KPIs like ASR, >> ACD, Dialcodes matching etc. - functionalities that will help anyone using FreeSWITCH as a SBC or >> using FreeSWITCH to pass traffic that's originated from a sip proxy like Kamailio or OpenSips, and >> also to help users understand what is happening in their FreeSWITCH instances in real-time. >> >> Here's the git account <https://github.com/gnostix/freeswitch-monitoring>, and the live instance of >> the dashboard is here <http://fs-moni.cloudapp.net:8080/>. The credentials are *admin/admin* >> >> I hope you will find it useful and we are awaiting for your feedback! >> >> Thanks for taking the time to read this rather long email! >> >> >> >> Kind Regards >> >> Alex >> _________________________________________________________________________ >> 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 > > > _________________________________________________________________________ > 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 > > > > > -- > Alexandros Pappas | Business Development | Vieras > Phone: +30 6937630910 <tel:%2B30%206937630910> > Email: apappas(a)vieras.eu <mailto:apappas@vieras.eu> > Skype: alex.pappas1 > Twitter: @rebel_alx > > > >

1 0

FS No ACK for 200 OK from GW
by Salman 05 Nov '15

05 Nov '15

Hello all, I am having a situation where 200 OK from a GW arrives at FS and can be seen in FS CLI but no ACK is generated. Here in INVITE and 200 OK packets I am wondering what are the problematic headers I know contact is messy though. Shouldn't it at least generate an ACK ?. PUBLIC-IP:5070 = FS PUBLIC-IP or PUBLIC-IP:5060 = GW(Kamailio) -------------------------- INVITE -------From FS to GW-------------------- INVITE sip:81205999@public.domain.com SIP/2.0 Via: SIP/2.0/UDP PUBLIC-IP:5070;rport;branch=z9hG4bK4rSU85e8rjj6j Max-Forwards: 70 From: "UserA" <sip:16910067@PUBLIC-IP>;tag=BSNjv3XNrFU1m To: <sip:81205999@public.domain.com> Call-ID: 8fe2118a-fe89-1233-2781-069be24d4191 CSeq: 83037069 INVITE Contact: <sip:gw+pcall@PUBLIC-IP:5070;transport=udp;gw=pvcall> User-Agent: FreeSWITCH-mod_sofia/1.4.23+git~20151023T152155Z~9fcf3e0c9a~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 303 X-FS-Support: update_display,send_info Remote-Party-ID: "UserA" <sip:16910067@PUBLIC-IP >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1446729346 1446729347 IN IP4 PUBLIC-IP s=FreeSWITCH c=IN IP4 PUBLIC-IP t=0 0 m=audio 16920 RTP/AVP 102 0 8 9 101 13 a=rtpmap:102 SPEEX/16000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ---------------------------------- 200 OK -------From GW to FS---------------------------- SIP/2.0 200 OK Via: SIP/2.0/UDP PUBLIC-IP:5070;received=PUBLIC-IP;rport=5070;branch=z9hG4bK4rSU85e8rjj6j To: <sip:81205999@public.domain.com>;tag=shv30apv2f From: "UserA" <sip:16910067@PUBLIC-IP>;tag=BSNjv3XNrFU1m Call-ID: 8fe2118a-fe89-1233-2781-069be24d4191 CSeq: 83037069 INVITE Contact: <sip:k471p745@v3n0qcki9ah1.invalid ;alias=182.178.237.31~20760~6;transport=ws> X-Can-Renegotiate: true Supported: ice,replaces,outbound Content-Type: application/sdp Content-Length: 597 v=0 o=- 464531514895581299 2 IN IP4 PUBLIC-IP s=- t=0 0 a=msid-semantic: WMS QT4IzXsqqeNv2uaXsBBrmLBlUeHKFOYKSRMh m=audio 33416 RTP/AVP 0 8 9 101 13 c=IN IP4 PUBLIC-IP a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=rtpmap:13 CN/8000 a=ssrc:4081787962 cname:HWxDJwiTLUiGrbg/ a=ssrc:4081787962 msid:QT4IzXsqqeNv2uaXsBBrmLBlUeHKFOYKSRMh 3f289eda-ce8d-4219-ac04-1147e6a14fdb a=ssrc:4081787962 mslabel:QT4IzXsqqeNv2uaXsBBrmLBlUeHKFOYKSRMh a=ssrc:4081787962 label:3f289eda-ce8d-4219-ac04-1147e6a14fdb a=sendrecv a=rtcp:33417 -- Regards Salman

2 1

how to remove precondition in "Supported" sip heared.
by Юрий Насида 05 Nov '15

05 Nov '15

Hi there! I know that this bug is already fixed in new versions of FS https://freeswitch.org/jira/browse/FS-6221 But I would not like to update FS version now and looking for some workaround. Any advice ? Thanks.

2 1