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SRTP and multiple endpoints
by Denis Papes 19 Feb '19

19 Feb '19

Hi, I have FreeSWITCH server with two softphones and one Polycom VVX 310 phone. All endpoints use SRTP. One softphone and Polycom phone are registered as same user. When I make a call to that user, softphone and Polycom both ring as expected. Problem is that Polycom does early media, so it sends encryption key in 183 message and FreeSWITCH does not update key if I answer on softphone and call breaks after several "SRTP audio unprotect failed with code 7 (auth check failed) 83 bytes 10 errors" errors. Does anyone know of any solution for that? Call to Polycom ---------------------------------------------------- ------------------------------------------------------------------------ recv 1268 bytes from udp/[XXX.XXX.XXX.XXX]:5060 at 13:23:24.192990: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress P-Asserted-Identity: "User Name" <sip:User.Name@domain.tld>,<tel:+11111111111;ext=1111> Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5062;received=XXX.XXX.XXX.XXX;rport=5062;branch=z9hG4bKH13aN0H9550Ue From: "Extension 1000" <sip:1000@fs01.domain.tld>;tag=vvZQ9m8H03Npp To: "User Name" <sip:User.Name@domain.tld>;epid=0004f2853296;tag=BB7B7F4F-B75734BA CSeq: 737508 INVITE Call-ID: 5eaff2fd-aeec-1237-6aaa-00155d018038 Contact: <sip:User.Name@domain.tld;opaque=user:epid:eji6PlZNF1Sy2o_kLXZ3sQAA;gruu> Record-Route: <sip:ucs.domain.tld:5061;transport=tls;ms-fe=server3.domain.tld;opaque=state:T;lr>, <sip:XXX.XXX.XXX.XXX:5061;transport=tls;r2=on;lr>, <sip:XXX.XXX.XXX.XXX;r2=on;lr> User-Agent: Polycom/5.9.0.9373 PolycomVVX-VVX_310-UA/5.9.0.9373 Accept-Language: en Content-Type: application/sdp Content-Length: 354 v=0 o=- 1550582604 1550582604 IN IP4 YYY.YYY.YYY.YYY s=Polycom IP Phone c=IN IP4 YYY.YYY.YYY.YYY t=0 0 a=sendrecv m=audio 5380 RTP/SAVP 102 101 a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:GKS3Pyolq2FNrLsVljrZWO3ziKdaRG+9G8mXDn2F|2^31|1:1 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=24000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=rtcp:5381 ------------------------------------------------------------------------ 2019-02-19 13:23:23.731999 [INFO] sofia.c:1356 sofia/internal/User.Name(a)domain.tld Update Callee ID to "User Name" <User.Name> 2019-02-19 13:23:23.731999 [DEBUG] sofia.c:7291 Channel sofia/internal/User.Name(a)domain.tld entering state [proceeding][183] 2019-02-19 13:23:23.731999 [DEBUG] sofia.c:7301 Remote SDP: v=0 o=- 1550582604 1550582604 IN IP4 YYY.YYY.YYY.YYY s=Polycom IP Phone c=IN IP4 YYY.YYY.YYY.YYY t=0 0 a=sendrecv m=audio 5380 RTP/SAVP 102 101 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=24000 a=rtpmap:101 telephone-event/8000 a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:GKS3Pyolq2FNrLsVljrZWO3ziKdaRG+9G8mXDn2F|2^31|1:1 a=rtcp:5381 2019-02-19 13:23:23.731999 [NOTICE] sofia.c:7304 Pre-Answer sofia/internal/User.Name(a)domain.tld! 2019-02-19 13:23:23.731999 [DEBUG] switch_channel.c:3482 (sofia/internal/User.Name(a)domain.tld) Callstate Change RINGING -> EARLY 2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:1804 looking for crypto suite [AEAD_AES_256_GCM_8]alias=[] in [5 AES_CM_128_HMAC_SHA1_80 inline:GKS3Pyolq2FNrLsVljrZWO3ziKdaRG+9G8mXDn2F|2^31|1:1] 2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:1804 looking for crypto suite [AEAD_AES_128_GCM_8]alias=[] in [5 AES_CM_128_HMAC_SHA1_80 inline:GKS3Pyolq2FNrLsVljrZWO3ziKdaRG+9G8mXDn2F|2^31|1:1] 2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:1804 looking for crypto suite [AES_256_CM_HMAC_SHA1_80]alias=[AES_CM_256_HMAC_SHA1_80] in [5 AES_CM_128_HMAC_SHA1_80 inline:GKS3Pyolq2FNrLsVljrZWO3ziKdaRG+9G8mXDn2F|2^31|1:1] 2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:1804 looking for crypto suite [AES_192_CM_HMAC_SHA1_80]alias=[AES_CM_192_HMAC_SHA1_80] in [5 AES_CM_128_HMAC_SHA1_80 inline:GKS3Pyolq2FNrLsVljrZWO3ziKdaRG+9G8mXDn2F|2^31|1:1] 2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:1804 looking for crypto suite [AES_CM_128_HMAC_SHA1_80]alias=[] in [5 AES_CM_128_HMAC_SHA1_80 inline:GKS3Pyolq2FNrLsVljrZWO3ziKdaRG+9G8mXDn2F|2^31|1:1] 2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:1813 Found suite AES_CM_128_HMAC_SHA1_80 2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:1883 Set Remote Key [5 AES_CM_128_HMAC_SHA1_80 inline:GKS3Pyolq2FNrLsVljrZWO3ziKdaRG+9G8mXDn2F|2^31|1:1] 2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:5478 Audio Codec Compare [G7221:102:16000:20:24000:1]/[G7221:107:16000:20:24000:1] 2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:5533 Audio Codec Compare [G7221:107:16000:20:24000:1] ++++ is saved as a match 2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:5478 Audio Codec Compare [G7221:102:16000:20:24000:1]/[G722:9:8000:20:64000:1] 2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:5478 Audio Codec Compare [G7221:102:16000:20:24000:1]/[PCMU:0:8000:20:64000:1] 2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:5478 Audio Codec Compare [G7221:102:16000:20:24000:1]/[PCMA:8:8000:20:64000:1] 2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:5394 Set telephone-event payload to 101@8000 2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/User.Name(a)domain.tld G7221/16000 20 ms 320 samples 24000 bits 1 channels 2019-02-19 13:23:23.731999 [DEBUG] switch_core_codec.c:111 sofia/internal/User.Name(a)domain.tld Original read codec set to G7221:107 2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:5795 sofia/internal/User.Name(a)domain.tld Set 2833 dtmf send payload to 101 recv payload to 101 2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:8511 AUDIO RTP [sofia/internal/User.Name(a)domain.tld] XXX.XXX.XXX.XXX port 26804 -> YYY.YYY.YYY.YYY port 5380 codec: 102 ms: 20 2019-02-19 13:23:23.731999 [DEBUG] switch_rtp.c:4300 Starting timer [soft] 320 bytes per 20ms 2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:8731 Activating RTCP PORT 5381 2019-02-19 13:23:23.731999 [DEBUG] switch_rtp.c:4696 RTCP send rate is: 500 and packet rate is: 20000 Remote Port: 5381 2019-02-19 13:23:23.731999 [DEBUG] switch_rtp.c:2572 Setting RTCP remote addr to YYY.YYY.YYY.YYY:5381 2 2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:8815 sofia/internal/User.Name(a)domain.tld Set 2833 dtmf send payload to 101 2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:8822 sofia/internal/User.Name(a)domain.tld Set 2833 dtmf receive payload to 101 2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:8845 sofia/internal/User.Name(a)domain.tld Set rtp dtmf delay to 40 2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:1495 LIFETIME found in |2^31|1:1, base 2 exp 31 2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:1502 MKI found in |2^31|1:1, id 1 size 1 2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:1495 LIFETIME found in |2^31|1:1, base 2 exp 31 2019-02-19 13:23:23.731999 [DEBUG] switch_core_media.c:1502 MKI found in |2^31|1:1, id 1 size 1 2019-02-19 13:23:23.731999 [INFO] switch_rtp.c:4104 Activating audio Secure RTP SEND (with MKI) 2019-02-19 13:23:23.731999 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_80 2019-02-19 13:23:23.731999 [INFO] switch_rtp.c:4082 Activating audio Secure RTP RECV (with MKI) 2019-02-19 13:23:23.731999 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_80 Call to Softphone ---------------------------------------------------- recv 1860 bytes from udp/[XXX.XXX.XXX.XXX]:5060 at 13:23:25.884591: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5062;received=XXX.XXX.XXX.XXX;rport=5062;branch=z9hG4bKH13aN0H9550Ue Content-Length: 428 P-Asserted-Identity: <sip:User.Name@domain.tld>, <tel:+11111111111;ext=1111> From: "Extension 1000" <sip:1000@fs01.domain.tld>;tag=vvZQ9m8H03Npp To: <sip:User.Name@domain.tld>;epid=111ee812f0;tag=1f0cd7c868 Call-ID: 5eaff2fd-aeec-1237-6aaa-00155d018038 CSeq: 737508 INVITE Record-Route: <sip:ucs.domain.tld:5061;transport=tls;ms-fe=server3.domain.tld;opaque=state:T;lr> Record-Route: <sip:XXX.XXX.XXX.XXX:5061;transport=tls;r2=on;lr> Record-Route: <sip:XXX.XXX.XXX.XXX;r2=on;lr> Contact: <sip:User.Name@domain.tld;opaque=user:epid:Xq6LHtQWXFWBMxOY6SkY-wAA;gruu> User-Agent: UCCAPI/15.0.5093.1000 OC/15.0.5111.1000 (Skype for Business) Supported: histinfo Supported: ms-safe-transfer Supported: ms-dialog-route-set-update Allow: INVITE, BYE, ACK, CANCEL, INFO, UPDATE, REFER, NOTIFY, BENOTIFY, OPTIONS Session-Expires: 720;refresher=uac ms-endpoint-location-data: NetworkScope;ms-media-location-type=Intranet Supported: ms-bypass Supported: replaces Content-Type: application/sdp ms-application-via: ms-udc.cdr%3D70847bfc9ab1b8a46ead5d92c6ad493a%3A6%3Bconvhist%3D0%3A6;ms-pool=ucs.domain.tld;ms-application=http%3A%2F%2Fwww.microsoft.com%2FLCS%2FUdcAgent;ms-server=server3.domain.tld v=0 o=- 0 1 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ s=FreeSWITCH c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ b=CT:99980 t=0 0 m=audio 5407 RTP/SAVP 102 9 0 8 101 a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:Pb70Q8VAa0lf1GNvKwjxRjUsK3pu1L28ga4pdn7s|2^31|1:1 a=maxptime:200 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=24000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-mux a=ptime:20 ------------------------------------------------------------------------ 2019-02-19 13:23:25.431998 [INFO] sofia.c:1356 sofia/internal/User.Name(a)domain.tld Update Callee ID to "User.Name" <User.Name> 2019-02-19 13:23:25.431998 [DEBUG] sofia.c:7291 Channel sofia/internal/User.Name(a)domain.tld entering state [completing][200] 2019-02-19 13:23:25.431998 [DEBUG] sofia.c:7301 Remote SDP: v=0 o=- 0 1 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ s=FreeSWITCH c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ b=CT:99980 t=0 0 m=audio 5407 RTP/SAVP 102 9 0 8 101 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=24000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:Pb70Q8VAa0lf1GNvKwjxRjUsK3pu1L28ga4pdn7s|2^31|1:1 a=maxptime:200 a=rtcp-mux a=ptime:20 Thanks, Denis

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Freeswitch As SBC for sip_phones
by Mehdi Shirazi 19 Feb '19

19 Feb '19

Hi Thanks for suggestions, If we want to comply strictly with rfc 5853 in our SBC so we should have a B2BUA also int it. What is your recommendation? 1-Using Kamailio/Opensips for forwarding REGISTER/SUBSCRIBE/NOTIFY to external Registrar and using a Freeswitch for media+B2BUA ? 2-Using Kamailio/Opensips + rtpengine and just using Freeswitch as B2BUA in media bypass mode Regards. M.shirazi

2 1

Hold BUG
by Nathan Stratton 19 Feb '19

19 Feb '19

Larry calls Bob Bob puts Larry on hold this sends UPDATE to FreeSWITCH with setup:actpass media attribute 200 OK from FreeSWITCH responds correctly with setup:active everything works correctly Larry calls Bob Larry puts Bob on hold this send UPDATE to FreeSWITCH with setup:actpass media attribute 200 OK from FreeSWITCH responds incorrectly with setup:actpass this causes chrome to kill the call. The client is lastest JsSIP ><> nathan stratton

3 2

Called id name is not displayed - Cisco 7912G devices
by Panagiotis Voutskidis 18 Feb '19

18 Feb '19

I run a freeswitch pbx with about 10 cisco 7912G (with SIP firmware) and 2 linksys devices. When I call a local device from a cisco ip phone, we don't see the called device id name, during ringing and talking. The only thing displayed is the dialed number. Linksys devices display called user name as expected. Is it a problem with the specific device and firmware or there is something I could do to fix it? Is there anyone else using cisco 7912G phones that can help with the situation? Regards, Panos

1 0

How to find mistake in FS config?
by Yu Boot 15 Feb '19

15 Feb '19

I changed some things in internal.xml and now I got this: # fs_cli -x reloadxml +OK [[error near line 10374]: missing >] Searching in dialplan.xml at line 13074 gives nothing. How to find where's mistake?

4 3

missing allison/digits/40.wav audio file
by Brandon Youngdale 14 Feb '19

14 Feb '19

Hi, Is there a reason there is no '40.wav' for the allison recordings? I noticed on my install when I got an error that it was not in the digits directory, and when I looked it is not in the repo either. https://freeswitch.org/stash/projects/FS/repos/freeswitch-sounds/browse/en/… allison/digits/48000 Thank you, -- Brandon Youngdale

1 0

NORMAL_TEMPORARY_FAILURE Freeswitch Outbound via Gateway to PSTN
by Chhorm Chhatra 14 Feb '19

14 Feb '19

Hi, Currently, I faced the issue of getting NORMAL_TEMPORARY_FAILURE when FreeSWITCH is originating outbound calls to the PSTN via Dinstar GSM gateway device. I follow the guides from FreeSWITCH confluence and the inbound call is working fine. My Dinstar Gateway device version is UC2000-VE Business. Any help would be appreciated. Best regards, Chhorm Chhatra

2 2

Clean expired SIP registrations of FS internal database
by Jose David Jurado Alonso 13 Feb '19

13 Feb '19

Hi, I have seen that in the FS internal database the account SIP registrations are stored during a lot of time and it seems that they are not deleted never. Is there any mode or parameter to automatically delete these database records after X time, by expired time or number of records for example? To be better explained, I refer to the SIP profile records that appear when the command is executed: fs_cli> sofia status profile external reg Thank a lot. Regards, José David Jurado Alonso.

2 4

INCOMPATIBLE_DESTINATION
by Paul Muaddib 13 Feb '19

13 Feb '19

Hi, I have a rarely error where I get this failure message in the log file. I am bit wondering because this is suposed to be a normal voice call and not a fax. Best regards, Paul. fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 2018-08-20 14:04:54.056776 [INFO] switch_ivr_originate.c:1215 Sending early media fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 2018-08-20 14:04:54.056776 [ERR] mod_sofia.c:2343 CODEC NEGOTIATION ERROR. SDP: fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 v=0^M fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 o=- 20012711 20012711 IN IP4 217.0.15.67^M fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 s=on transit^M fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 c=IN IP4 X.X.X.X^M fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 t=0 0^M fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 m=audio 13274 RTP/AVP 8 118 126 101^M fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=rtpmap:8 PCMA/8000^M fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=rtpmap:118 PCMA/8000^M fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=rtpmap:126 CLEARMODE/8000^M fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=rtpmap:101 telephone-event/8000^M fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=fmtp:101 0-15^M fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=gpmd:118 vbd=yes^M fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=ptime:20^M fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 m=image 11232 udptl t38^M fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38FaxVersion:0^M fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38MaxBitRate:14400^M fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38FaxMaxBuffer:3000^M fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38FaxMaxDatagram:500^M fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38FaxRateManagement:transferredTCF^M fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38FaxUdpEC:t38UDPRedundancy^M fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 2018-08-20 14:04:54.056776 [NOTICE] switch_channel.c:3515 Hangup sofia/external/+ 49XXXXXXXXX(a)sip-trunk.telekom.de [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]

6 11

Count active calls on individual gateways
by Kieran Hackshall 13 Feb '19

13 Feb '19

Looking to get a count of active calls on a particular gateway for outbound calls.. Running "show calls" and "show channels" I cannot see anything that would allow me to determine what gateway is being used. Is there another way to do this? Cheers, Kieran

1 0