December 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to bridge a call to an extension without the display updating?
by Michael Avers 13 Mar '18

13 Mar '18

Hello, I'm trying to create a random dialplan extension (say 5050) that when dialed into by a registered extension would bridge the call to another registered user (say 3000), but not change the display or P-Asserted-Identity. Right now on the bridge it sends to the calling extension *P-Asserted-Identity: "5050" <sip:3000@sip.domain>.* I tried setting all the possible values for sip_cid_type and setting ignore_display_updates to false, but it still sends the P-Asserted update as shown above. The bridge itself I tried both with user/3000(a)sip.domain and ${sofia_contact(566055(a)rsi.solidphone.net)} - same result. This is a problem because A. the display on the phone changes to "3000" and more importantly, B. if the calling user later hits redial it will dial 3000 instead of 5050. Any way to achieve what I'm trying to do? Thanks Mike

3 2

Set b-leg CallerID using from in a 300 Multiple Choices reply
by Joseph Waite 12 Feb '18

12 Feb '18

Hi Guys I am trying to use the From field in a 300 Multiple Choices message to set the outbound CallerID in the re-directed b-leg So call comes into FreeSwitch, gets sent out to a SIP Redirect server which replies with a 300 Multiple choices. I need to pull the number from the From field of this 300 Response and use it to set the From field, raid & Paid fields in the re-directed invite. Any help would be very much appreciated. Regards

3 5

FS weird issue with TLS/SRTP and xml_curl on windows
by Bipin Patel 26 Jan '18

26 Jan '18

hi, i have two instances of FS running on a windows server, one is used for routing to carriers and that uses xml files and that works fine, the second instance is set up to allow clients to register to it using tls and srtp and users are authenticated using xml_curl which calls a php script which inturn sends the directory users details on a register from a client. The problem is every few hours or so FS stops accepting new clients unless i restart the service, i check the php script and the webserver and those are running just fine so no idea whats causing FS to stop calling the script or something else. -- Regards, Bipin ------------------------------------------------------------------------

3 5

Received an unsupported RTCP packet version 3
by Mikhail Demekhov 16 Jan '18

16 Jan '18

Hello! I have FreeSWITCH Version 1.6.12. After connecting partner FS-1.7 I had a lot of warning: [WARNING] switch_rtp.c:6207 Received an unsupported RTCP packet version 3 How to fix it? -- Regards, Mikhail Demekhov

2 1

Lose Race instead of missed call
by Peter Steinbach 15 Jan '18

15 Jan '18

Hello, we have some Problems with the right signalling of missed calls when calling multiple phones in parallel Here's the scenario: Phone no 49 is calling a group with 2170 and 3275 with the following dialstring <action application="bridge" data="{default_language=de,ignore_early_media=true,global_to_originate_1=true,caller_cc=49,callee_cc=49,routing_flags=INT-ONSYSTEM-RELIA0-QUAL0-T38NOLAST-CALLG-CALLG5003-UUID,call_timeout=60,originate_timeout=60,origination_caller_id_number=49,effective_caller_id_number=49,caller_uuid=7f49c581-cf47-429d-85a0-365d00ff031c,origination_uuid=59eee5e0-c0bc-0135-7dc9-00505600a1a5,sip_invite_domain=flex.mydomain.de,customer_id=261}user/2170@flex.mydomain.de:_:{default_language=de,ignore_early_media=true,global_to_originate_1=true,caller_cc=49,callee_cc=49,routing_flags=INT-ONSYSTEM-RELIA0-QUAL0-T38NOLAST-CALLG-CALLG5003-UUID,call_timeout=60,originate_timeout=60,origination_caller_id_number=49,effective_caller_id_number=49,caller_uuid=7f49c581-cf47-429d-85a0-365d00ff031c,origination_uuid=59f07af0-c0bc-0135-7dca-00505600a1a5,sip_invite_domain=flex.mydomain.de,customer_id=261}user/3275@flex.mydomain.de" /> Destination dialstrings are seperated by ":_:" ("Enterprise Origination"). We use curly brackets instead of "<" as we sometimes have to insert asserted identy tags into the dialstring. We checked 3 versions of Freeswitch for this * Version Feb 2016 shows missed calls on both phones. Even if one phone answers, the other phone one still shows a missed call (reason for upgrading to newer Freeswitch) * Version Aug 2017 never shows missed call, see logs and hangup message below * Version 10/Dec 2017(yesterday) never shows missed call, as above So for the 2 never Freeswitch Versions, here are the logs at hangup 2017-12-11 17:07:46.022450 [DEBUG] sofia.c:7283 Channel sofia/internal/49@flex.mydomain.de:5060 entering state [terminated][487] 2017-12-11 17:07:46.022450 [NOTICE] sofia.c:8474 Hangup sofia/internal/*49*@flex.mydomain.de:5060 [CS_EXECUTE] *[ORIGINATOR_CANCEL]* 2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:3627 Hangup sofia/internal/*2170*@94.xx.xxx.xx:42170 [CS_CONSUME_MEDIA] *[LOSE_RACE]* 2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3852 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:2866 Cannot create outgoing channel of type [user] cause: [LOSE_RACE] 2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3863 Originate Resulted in Error Cause: 502 [LOSE_RACE] 2017-12-11 17:07:46.042331 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/2170@94.xx.xxx.xx:42170) Running State Change CS_HANGUP (Cur 3 Tot 22) 2017-12-11 17:07:46.042331 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/2170@94.xx.xxx.xx:42170) Callstate Change RINGING -> HANGUP 2017-12-11 17:07:46.042331 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/2170@94.xx.xxx.xx:42170) State HANGUP 2017-12-11 17:07:46.042331 [DEBUG] mod_sofia.c:449 Channel sofia/internal/2170@94.xx.xxx.xx:42170 hanging up, cause: LOSE_RACE 2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:3627 Hangup sofia/internal/*3275*@94.xx.xxx.xx:43275 [CS_CONSUME_MEDIA] [*LOSE_RACE*] 2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3852 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:2866 Cannot create outgoing channel of type [user] cause: [LOSE_RACE] 2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3863 Originate Resulted in Error Cause: 502 [LOSE_RACE] Here is the Cancel message for one of the called phones: U 2017/12/11 17:07:46.045120 144.xx.xxx.xx:5060 -> 94.xx.xxx.xx:42170 *CANCEL sip:2170*@94.xx.xxx.xx:42170 SIP/2.0. Via: SIP/2.0/UDP 144.xx.xxx.xx;rport;branch=z9hG4bK07r3ce94cvjpp. Max-Forwards: 70. From: "Test" <sip:49@flex.mydomain.de>;tag=mQS1eFDpp1peS. To: <sip:2170@94.xx.xxx.xx:42170>. Call-ID: 4238b5a7-5930-1236-b2ab-00505600a1a5. CSeq: 116168167 CANCEL. *Reason: SIP;cause=200;text="Call completed elsewhere"*. Content-Length: 0. Any hints why this happens, or anyone has this scenario working? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

1 3

(no subject)
by Ali Haider 11 Jan '18

11 Jan '18

Hiii jeol https://freeswitch.org/confluence/pages/viewpage.action?pageId=1966780 I’m followed these steps on windows but after installation of visual studio 2013 which was may be required 182 error is found for missing files Please tell me which version of visual studio is installed to build soln and how to remove error Sent from my iPhone

3 6

error correction
by Ali Haider 06 Jan '18

06 Jan '18

Hiiii Everyone Anyone who can help me fix the error that are being reviewed in picture during build on visual studio 2015

1 1

webrtc with ecdsa
by GM phy 05 Jan '18

05 Jan '18

Hello, did anybody get webrtc working with a ECDSA signed cert? I have fs 1.6.19 compiled with openssl 1.0.2g and wss.pem file is -----BEGIN EC PARAMETERS----- BggqhkjOPQMBBw== -----END EC PARAMETERS----- -----BEGIN EC PRIVATE KEY----- MHcCAQEEIPNZ4YUGKQ2cySfnEp1RSOSDp9MTQ21xyISpBmBPRN1/oAoGCCqGSM49 AwEHoUQDQgAEnxYcss4/4yCzBl85psRqTLeQ0FnXqsoDCC/2lKMTsrS7mlPgCo7m +A4huOY2MtbwXtBdg2kqjYbcrkAk5zwa6Q== -----END EC PRIVATE KEY----- -----BEGIN CERTIFICATE----- MIIGGTCCBQGgAwIBAgIMAPaMiX8YXElRs/bJMA0GCSqGSIb3DQEBCwUAMEwxCzAJ BgNVBAYTAkJFMRkwFwYDVQQKExBHbG9iYWxTaWduIG52LXNhMSIwIAYDVQQDExlB bHBoYVNTTCBDQSAtIFNIQTI1NiAtIEcyMB4XDTE3MTAxNzEzNDY1MVoXDTE4MTAx ODEzNDY1MVowQTEhMB8GA1UECxMYRG9tYWluIENvbnRyb2wgVmFsaWRhdGVkMRww GgYDVQQDDBMqLmxhY29udGluZW50YWwuY29tMFkwEwYHKoZIzj0CAQYIKoZIzj0D AQcDQgAEnxYcss4/4yCzBl85psRqTLeQ0FnXqsoDCC/2lKMTsrS7mlPgCo7m+A4h uOY2MtbwXtBdg2kqjYbcrkAk5zwa6aOCA88wggPLMA4GA1UdDwEB/wQEAwIDiDCB iQYIKwYBBQUHAQEEfTB7MEIGCCsGAQUFBzAChjZodHRwOi8vc2VjdXJlMi5hbHBo .... when we try to connect (we are using sipjs and tried from chrome and firefox last versions) we get a handshake failure. any lead or help will be appreciated, thanks in advance.

3 2

About Codec Negotiation
by 王聡 04 Jan '18

04 Jan '18

Hey all, I had got an issue when I tried to bridge FS to Asterisk for a outgoing call. The late negotation had been set as true, but I got a codec error on Asterisk which showed: [Dec 29 16:16:52] NOTICE[2095][C-00000002]: chan_sip.c:10465 process_sdp: No compatible codecs, not accepting this offer! After this I checked the codec negotiation, like this: SDP message from softphone: 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:46.677190 [DEBUG] sofia.c:7094 Remote SDP:401e6cb7-3c4d-43ca-9bc9-a5055ad52485 v=0 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:46.677190 [DEBUG] sofia.c:7094 Remote SDP: 43e110bf-0dd2-4518-a5d8-3b3afd52f6de v=0 43e110bf-0dd2-4518-a5d8-3b3afd52f6de o=52bf0a8a88c515081c124a49cf6ce7bc 2651 2264 IN IP4 192.168.254.65 43e110bf-0dd2-4518-a5d8-3b3afd52f6de s=Talk 43e110bf-0dd2-4518-a5d8-3b3afd52f6de c=IN IP4 192.168.254.65 43e110bf-0dd2-4518-a5d8-3b3afd52f6de t=0 0 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics 43e110bf-0dd2-4518-a5d8-3b3afd52f6de m=audio 7076 RTP/AVP 96 97 98 99 0 8 101 100 102 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:96 opus/48000/2 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=fmtp:96 useinbandfec=1 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:97 SILK/16000 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:98 speex/16000 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=fmtp:98 vbr=on 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:99 speex/8000 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=fmtp:99 vbr=on 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:101 telephone-event/48000 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:100 telephone-event/16000 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:102 telephone-event/8000 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtcp-fb:* trr-int 5000 Negotiation steps: 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [opus:96:48000:20:0:1]/[PCMU:0:8000:20:64000:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [opus:96:48000:20:0:1]/[PCMA:8:8000:20:64000:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [opus:96:48000:20:0:1]/[GSM:3:8000:20:13200:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [opus:96:48000:20:0:1]/[SPEEX:99:8000:20:24600:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [opus:96:48000:20:0:1]/[SPEEX:99:16000:20:42200:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [SILK:97:16000:20:0:1]/[PCMU:0:8000:20:64000:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [SILK:97:16000:20:0:1]/[PCMA:8:8000:20:64000:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [SILK:97:16000:20:0:1]/[GSM:3:8000:20:13200:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [SILK:97:16000:20:0:1]/[SPEEX:99:8000:20:24600:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [SILK:97:16000:20:0:1]/[SPEEX:99:16000:20:42200:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:98:16000:20:0:1]/[PCMU:0:8000:20:64000:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:98:16000:20:0:1]/[PCMA:8:8000:20:64000:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:98:16000:20:0:1]/[GSM:3:8000:20:13200:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:98:16000:20:0:1]/[SPEEX:99:8000:20:24600:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:98:16000:20:0:1]/[SPEEX:99:16000:20:42200:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [SPEEX:99:16000:20:42200:1] ++++ is saved as a match 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:99:8000:20:0:1]/[PCMU:0:8000:20:64000:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:99:8000:20:0:1]/[PCMA:8:8000:20:64000:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:99:8000:20:0:1]/[GSM:3:8000:20:13200:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:99:8000:20:0:1]/[SPEEX:99:8000:20:24600:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [SPEEX:99:8000:20:24600:1] ++++ is saved as a match 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:99:8000:20:0:1]/[SPEEX:99:16000:20:42200:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[GSM:3:8000:20:13200:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[SPEEX:99:8000:20:24600:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[SPEEX:99:16000:20:42200:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[GSM:3:8000:20:13200:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[SPEEX:99:8000:20:24600:1] 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[SPEEX:99:16000:20:42200:1] Returned SDP message: 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] mod_sofia.c:2364 Ring SDP: 43e110bf-0dd2-4518-a5d8-3b3afd52f6de v=0 43e110bf-0dd2-4518-a5d8-3b3afd52f6de o=FreeSWITCH 1514482587 1514482588 IN IP4 172.16.25.25 43e110bf-0dd2-4518-a5d8-3b3afd52f6de s=FreeSWITCH 43e110bf-0dd2-4518-a5d8-3b3afd52f6de c=IN IP4 210.148.155.42 43e110bf-0dd2-4518-a5d8-3b3afd52f6de t=0 0 43e110bf-0dd2-4518-a5d8-3b3afd52f6de m=audio 50360 RTP/AVP 98 100 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:98 speex/16000 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:100 telephone-event/16000 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=fmtp:100 0-16 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=ptime:20 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=sendrecv I’m not clear what happened, but PCMU and PCMA codecs are gone even it passed the check. Next the SDP had sent to Asterisk: c5439be6-10e0-431b-b5e3-7904a14a37c0 Local SDP: c5439be6-10e0-431b-b5e3-7904a14a37c0 v=0 c5439be6-10e0-431b-b5e3-7904a14a37c0 o=FreeSWITCH 1514482871 1514482872 IN IP4 172.16.25.24 c5439be6-10e0-431b-b5e3-7904a14a37c0 s=FreeSWITCH c5439be6-10e0-431b-b5e3-7904a14a37c0 c=IN IP4 172.16.25.24 c5439be6-10e0-431b-b5e3-7904a14a37c0 t=0 0 c5439be6-10e0-431b-b5e3-7904a14a37c0 m=audio 50076 RTP/AVP 102 101 13 c5439be6-10e0-431b-b5e3-7904a14a37c0 a=rtpmap:102 SPEEX/16000 c5439be6-10e0-431b-b5e3-7904a14a37c0 a=rtpmap:101 telephone-event/16000 c5439be6-10e0-431b-b5e3-7904a14a37c0 a=fmtp:101 0-16 c5439be6-10e0-431b-b5e3-7904a14a37c0 a=rtpmap:13 CN/16000 c5439be6-10e0-431b-b5e3-7904a14a37c0 a=ptime:20 c5439be6-10e0-431b-b5e3-7904a14a37c0 a=sendrecv Asterisk can’t recognize these codecs, and it turned into a failure. So what happened in the codec negotiation, and how could I let PCMU enabled? PS. I had pasted some setting file and hope it helps. Vars.xml: … <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,GSM,speex@8000h@20i,speex@16000h@20i,VP8,H264,H263-1998"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM,speex@8000h@20i,speex@16000h@20i”/> … Sip_profiles/external.xml: … <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}”/> <param name="inbound-codec-negotiation" value=“generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="inbound-late-negotiation" value="true"/> <param name="inbound-zrtp-passthru" value="true"/>  … Regards.

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uuid_record options
by Madovsky 31 Dec '17

31 Dec '17

Hi Folks, is it possible to control the record settings with uuid_record like width,height,quality etc.? thanks and have a great new year.

1 0