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Freeswitch 1.6.19 and T.38 / INCOMPATIBLE_DESTINATION
by Dominique Jeannerod 27 Apr '18

27 Apr '18

Hello, I currently use Freeswitch 1.6.8-1 in production, and have some ISDN gateways sending SDP including T.38 for voice calls (not faxes). Calls are handled without problems by freeswitch. I have another platform, with freeswitch 1.6.19 and same configuration. However, voice calls received with SDP including t.38 are rejected with 488 INCOMPATIBLE_DESTINATION : ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG] switch_core_media.c:4028 sofia/internal_1/33183817938@gv-ics-prd-301 T38 REFUSE on request ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [NOTICE] sofia.c:7566 Hangup sofia/internal_1/33183817938@gv-ics-prd-301 [CS_NEW] [INCOMPATIBLE_DESTINATION] I didn't find any valid configuration to make those calls accepted, except configuring proxy_media, which I don't want to use. Is this behaviour a "normal" behaviour ? Could it be a bug ? This is a very big issue for us ... Here is the log extract : recv 1391 bytes from udp/[10.9.30.152]:5060 at 23:47:39.443941: ------------------------------------------------------------------------ INVITE sip:0170993660@10.9.30.188:5060 SIP/2.0 Record-Route: <sip:10.9.30.152;lr=on> Record-Route: <sip:10.3.1.220;lr=on> Via: SIP/2.0/UDP 10.9.30.152;branch=z9hG4bK51d.d7e922c4.0 Via: SIP/2.0/UDP 10.3.1.220;branch=z9hG4bK51d.ba6f34d1.0 Via: SIP/2.0/UDP 10.3.1.231;branch=z9hG4bKac1339541470 Max-Forwards: 68 From: <sip:33183817938@gv-ics-prd-301>;tag=1c1339531455 To: <sip:0170993660@10.3.1.220;user=phone> Call-ID: 13395229682542018214736(a)10.3.1.231 CSeq: 1 INVITE Contact: <sip:33183817938@10.3.1.231:5060> Supported: em,100rel,timer,replaces,path,resource-priority,sdp-anat Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Remote-Party-ID: <sip:33183817938@10.3.1.220 >;party=calling;privacy=off;screen=yes;screen-ind=3;npi=1;ton=1 Remote-Party-ID: <sip:0170993660@10.3.1.220>;party=called;npi=1;ton=2 User-Agent: Audiocodes-Sip-Gateway-Mediant 3000/v.6.00A.056 Content-Type: application/sdp Content-Length: 417 v=0 o=AudiocodesGW 1339507326 1339507042 IN IP4 10.3.1.231 s=Phone-Call c=IN IP4 10.3.1.231 t=0 0 m=audio 23970 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv m=image 23972 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:33600 a=T38FaxMaxBuffer:3000 a=T38FaxMaxDatagram:560 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy ------------------------------------------------------------------------ send 477 bytes to udp/[10.9.30.152]:5060 at 23:47:39.444372: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.9.30.152;branch=z9hG4bK51d.d7e922c4.0 Via: SIP/2.0/UDP 10.3.1.220;branch=z9hG4bK51d.ba6f34d1.0 Via: SIP/2.0/UDP 10.3.1.231;branch=z9hG4bKac1339541470 Record-Route: <sip:10.9.30.152;lr=on> Record-Route: <sip:10.3.1.220;lr=on> From: <sip:33183817938@gv-ics-prd-301>;tag=1c1339531455 To: <sip:0170993660@10.3.1.220;user=phone> Call-ID: 13395229682542018214736(a)10.3.1.231 CSeq: 1 INVITE User-Agent: FreeSWITCH Content-Length: 0 ------------------------------------------------------------------------ ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [NOTICE] switch_channel.c:1104 New Channel sofia/internal_1/33183817938@gv-ics-prd-301 [ba17b1eb-961d-43fe-b820-cb469a8f292d] ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG] switch_core_state_machine.c:584 (sofia/internal_1/33183817938@gv-ics-prd-301) Running State Change CS_NEW (Cur 1 Tot 69) ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG] sofia.c:9874 sofia/internal_1/33183817938@gv-ics-prd-301 receiving invite from 10.9.30.152:5060 version: 1.6.19 64bit ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG] sofia.c:7085 Channel sofia/internal_1/33183817938@gv-ics-prd-301 entering state [received][100] ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG] sofia.c:7095 Remote SDP: ba17b1eb-961d-43fe-b820-cb469a8f292d v=0^M ba17b1eb-961d-43fe-b820-cb469a8f292d o=AudiocodesGW 1339507326 1339507042 IN IP4 10.3.1.231^M ba17b1eb-961d-43fe-b820-cb469a8f292d s=Phone-Call^M ba17b1eb-961d-43fe-b820-cb469a8f292d c=IN IP4 10.3.1.231^M ba17b1eb-961d-43fe-b820-cb469a8f292d t=0 0^M ba17b1eb-961d-43fe-b820-cb469a8f292d m=audio 23970 RTP/AVP 8 101^M ba17b1eb-961d-43fe-b820-cb469a8f292d a=rtpmap:8 PCMA/8000^M ba17b1eb-961d-43fe-b820-cb469a8f292d a=rtpmap:101 telephone-event/8000^M ba17b1eb-961d-43fe-b820-cb469a8f292d a=fmtp:101 0-15^M ba17b1eb-961d-43fe-b820-cb469a8f292d a=ptime:20^M ba17b1eb-961d-43fe-b820-cb469a8f292d m=image 23972 udptl t38^M ba17b1eb-961d-43fe-b820-cb469a8f292d a=T38FaxVersion:0^M ba17b1eb-961d-43fe-b820-cb469a8f292d a=T38MaxBitRate:33600^M ba17b1eb-961d-43fe-b820-cb469a8f292d a=T38FaxMaxBuffer:3000^M ba17b1eb-961d-43fe-b820-cb469a8f292d a=T38FaxMaxDatagram:560^M ba17b1eb-961d-43fe-b820-cb469a8f292d a=T38FaxRateManagement:transferredTCF^M ba17b1eb-961d-43fe-b820-cb469a8f292d a=T38FaxUdpEC:t38UDPRedundancy^M ba17b1eb-961d-43fe-b820-cb469a8f292d ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG] switch_core_media.c:4365 Set telephone-event payload to 101@8000 ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG] switch_core_media.c:3061 Set Codec sofia/internal_1/33183817938@gv-ics-prd-301 PCMA/8000 20 ms 160 samples 64000 bits 1 channels ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG] switch_core_codec.c:111 sofia/internal_1/33183817938@gv-ics-prd-301 Original read codec set to PCMA:8 ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG] switch_core_media.c:4708 Set telephone-event payload to 101@8000 ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG] switch_core_media.c:4767 sofia/internal_1/33183817938@gv-ics-prd-301 Set 2833 dtmf send payload to 101 recv payload to 101 ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG] switch_core_media.c:4028 sofia/internal_1/33183817938@gv-ics-prd-301 T38 REFUSE on request ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [NOTICE] sofia.c:7566 Hangup sofia/internal_1/33183817938@gv-ics-prd-301 [CS_NEW] [INCOMPATIBLE_DESTINATION] ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG] switch_core_state_machine.c:603 (sofia/internal_1/33183817938@gv-ics-prd-301) State NEW ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG] switch_core_state_machine.c:584 (sofia/internal_1/33183817938@gv-ics-prd-301) Running State Change CS_HANGUP (Cur 1 Tot 69) ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG] switch_core_state_machine.c:850 (sofia/internal_1/33183817938@gv-ics-prd-301) Callstate Change DOWN -> HANGUP ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG] switch_core_state_machine.c:852 (sofia/internal_1/33183817938@gv-ics-prd-301) State HANGUP ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG] mod_sofia.c:438 Channel sofia/internal_1/33183817938@gv-ics-prd-301 hanging up, cause: INCOMPATIBLE_DESTINATION ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG] mod_sofia.c:577 Responding to INVITE with: 488 ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG] switch_core_state_machine.c:60 sofia/internal_1/33183817938@gv-ics-prd-301 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG] switch_core_state_machine.c:852 (sofia/internal_1/33183817938@gv-ics-prd-301) State HANGUP going to sleep ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG] switch_core_state_machine.c:619 (sofia/internal_1/33183817938@gv-ics-prd-301) State Change CS_HANGUP -> CS_REPORTING ba17b1eb-961d-43fe-b820-cb469a8f292d 2018-04-25 23:47:39.434491 [DEBUG] switch_core_state_machine.c:584 (sofia/internal_1/33183817938@gv-ics-prd-301) Running State Change CS_REPORTING (Cur 1 Tot 69) send 784 bytes to udp/[10.9.30.152]:5060 at 23:47:39.446918: ------------------------------------------------------------------------ SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 10.9.30.152;branch=z9hG4bK51d.d7e922c4.0 Via: SIP/2.0/UDP 10.3.1.220;branch=z9hG4bK51d.ba6f34d1.0 Via: SIP/2.0/UDP 10.3.1.231;branch=z9hG4bKac1339541470 Max-Forwards: 68 From: <sip:33183817938@gv-ics-prd-301>;tag=1c1339531455 To: <sip:0170993660@10.3.1.220;user=phone>;tag=9USNQNQDByZ6a Call-ID: 13395229682542018214736(a)10.3.1.231 CSeq: 1 INVITE User-Agent: FreeSWITCH Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 Remote-Party-ID: "0170993660" <sip:0170993660@10.3.1.220 >;party=calling;privacy=off;screen=no Thanks with anticipation D. Jeannerod

3 3

Implementing emergency calls ... or call priority ?
by Peter Nerád 27 Apr '18

27 Apr '18

Whats are best practicies to implement emergency calls ? I can identify such a call by regexp but how to handle those calls ? Is there some build param to be set ? How I can set priority for call ? e.g. I have limited capacity for outgoing and incoming call to 10. I need to drop some calls with small priority to allow emergency call, but I have no idea where to start. Can somebody give me right direction for this ? Thank you very much PS: Google does not helped me :(

3 4

mod_say / switch_xml.c:2486 Invalid UTF-8 character to ampersand, skip it
by Thorsten Göllner 25 Apr '18

25 Apr '18

Hi, I am using Freeswitch 1.6.19~64bit on CentOs 7 (with FusionPBX 4.2.2) - working fine so far. But when calling a mailbox, I can hear the greeting but can not hear the destination number. I can find this macro is used: <macro name="voicemail_play_greeting"> <input pattern="^(.*)$"> <match> <action function="play-file" data="voicemail/vm-person.wav"/> <action function="say" data="$1" method="pronounced" type="name_spelled"/> <action function="play-file" data="voicemail/vm-not_available.wav"/> </match> </input> </macro> This is leading me to track it down to the function "say". So I configured a simple inbound destination which answers the call and just makes "say : de number pronounced 12345". When calling this number I get the following error: EXECUTE sofia/internal/12(a)mydomain.cloud say(de number pronounced 12345) 2018-04-25 15:07:09.732930 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f33b405a700 Connected. 2018-04-25 15:07:09.732930 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f33b405a700 released. 2018-04-25 15:07:09.732930 [WARNING] switch_xml.c:2486 Invalid UTF-8 character to ampersand, skip it 2018-04-25 15:07:09.732930 [WARNING] switch_xml.c:2486 Invalid UTF-8 character to ampersand, skip it 2018-04-25 15:07:09.732930 [WARNING] switch_xml.c:2486 Invalid UTF-8 character to ampersand, skip it 2018-04-25 15:07:09.732930 [WARNING] switch_xml.c:2486 Invalid UTF-8 character to ampersand, skip it 2018-04-25 15:07:09.732930 [ERR] switch_xml.c:3180 Can't find phrases tag. EXECUTE sofia/internal/12(a)mydomain.cloud hangup() I checked the source code but my coding skill is not good enough. I do not really know, where the problem is. Do you have any hint for me? Best regards, Thorsten

1 2

Linphone Softphone crashing freeswitch
by Jai Rangi 25 Apr '18

25 Apr '18

I have noticed Linphone client app can easily crash free-switch. Noticed few restarts right after. These logs d239a46b-2c2b-4ca9-9efc-1984b41e333f 2018-04-24 13:13:18.128573 [WARNING] switch_rtp.c:1092 missed 2 6c1e0ae1-dbe2-4630-8d8d-3cd20ba7e639 2018-04-24 13:13:18.348574 [ERR] switch_rtp.c:930 Invalid STUN/ICE packet received 28 bytes 6c1e0ae1-dbe2-4630-8d8d-3cd20ba7e639 2018-04-24 13:13:18.368567 [ERR] switch_rtp.c:930 Invalid STUN/ICE packet received 28 bytes d239a46b-2c2b-4ca9-9efc-1984b41e333f 2018-04-24 13:13:18.548572 [WARNING] switch_rtp.c:1092 missed 3 d239a46b-2c2b-4ca9-9efc-1984b41e333f 2018-04-24 13:13:18.668567 [WARNING] switch_rtp.c:1092 missed 1 de1df2ac-d136-4c24-9a4c-6a16f55a9422 2018-04-24 13:13:18.688574 [ALERT] switch_core_media.c:445 Looking for zrtp-hash de1df2ac-d136-4c24-9a4c-6a16f55a9422 2018-04-24 13:13:18.688574 [ALERT] switch_core_media.c:400 Deciding whether to pass zrtp-hash between legs de1df2ac-d136-4c24-9a4c-6a16f55a9422 2018-04-24 13:13:18.688574 [ALERT] switch_core_media.c:408 Found peer channel; propagating zrtp-hash if set de1df2ac-d136-4c24-9a4c-6a16f55a9422 2018-04-24 13:13:18.688574 [ALERT] switch_core_media.c:323 Deciding whether to pass zrtp-hash between a-leg and b-leg de1df2ac-d136-4c24-9a4c-6a16f55a9422 2018-04-24 13:13:18.688574 [ALERT] switch_core_media.c:323 Deciding whether to pass zrtp-hash between a-leg and b-leg And freeswitch restarts, de1df2ac-d136-4c24-9a4c-6a16f55a9422 2018-04-24 13:13:18.968579 [DEBUG] switch_rtp.c:7229 Correct audio ip/port confirmed. 2018-04-24 13:13:19.576345 [CONSOLE] switch_loadable_module.c:1540 Successfully Loaded [mod_logfile] 2018-04-24 13:13:19.576605 [CONSOLE] switch_loadable_module.c:1540 Successfully Loaded [mod_blacklist] 2018-04-24 13:13:19.576617 [NOTICE] switch_loadable_module.c:338 Adding API Function 'blacklist' 2018-04-24 13:13:19.577958 [CONSOLE] switch_loadable_module.c:1540 Successfully Loaded [mod_enum] 2018-04-24 13:13:19.577969 [NOTICE] switch_loadable_module.c:250 Adding Dialplan 'enum' 2018-04-24 13:13:19.577987 [NOTICE] switch_loadable_module.c:292 Adding Application 'enum' 2018-04-24 13:13:19.578009 [NOTICE] switch_loadable_module.c:338 Adding API Function 'enum' 2018-04-24 13:13:19.578029 [NOTICE] switch_loadable_module.c:338 Adding API Function 'enum_auto' Is there any setting to ignore ICE and STUN packets.

2 1

Implementing customer callback with mod_callcenter
by Michael Avers 25 Apr '18

25 Apr '18

Hello, Looking for some best practices.. I'm trying to let users exit a mod_callcenter queue to be placed in a callback queue that will (ideally) place them first in line to be connected to an agent when one becomes available. One way I had in mind is to disconnect the caller once we saved the fact that they requested a callback (application logic, doesn't matter here). Then, originate a call that plays in a loop "press 1 to connect" that is added to the relevant mod_callcenter queue with a base score that places it first in line to be picked up as the first priority. When the call is picked up the an agent, they hear the "press 1" playback, and once they actually press 1, we execute a lua script that calls the user that requested the callback and bridge agent and user. In theory this makes perfect sense . Any thoughts? Is there a simpler solution I am overlooking? Thanks Mike

1 0

Voicemail via web interface
by John 24 Apr '18

24 Apr '18

Hello, Can you point me to any additional information about the voice mail via web interface? I have it up and running; however if you click the play button there is no playback, if you click download it will play in MS media player. Thanks John

5 5

Question about large confeerence
by Abaci B 24 Apr '18

24 Apr '18

Hi, I'm trying to set up a large audio conference of over 1000 users where only 1 or 2 users are talking and the rest are just listening, but after 600-700 users the CPU usage gets too high and the audio gets choppy, I understand that mod_conference is not optimized for my use case but was wondering if someone has any suggestions on what can be done to optimize for my use case. I was also thinking that maybe there is a better alternative for what I look for, such as having just the talkers on the conference and the listeners should listen to it via mod_local_stream or maybe use something like eavesdrop for the listeners but not sure if and how much performance I would gain. Thanks for any help or input

8 18

How to add headers to an outgoing REGISTER
by Lars Paulsen 24 Apr '18

24 Apr '18

Hello, I have read the documentation under https://wiki.freeswitch.org/wiki/Sofia-SIP#Adding_Request_Headers and as I understood it is possible to add Headers to SIP requests simply by changing the configuration. But the documentation refers to the dialing plan. I tried to add headers to an outgoing REGISTER but failed. Because the documentation refers to the dialplan and changing headers for existing calls, I am not sure if adding headers to a REGISTER request is possible or not. Can anyone please tell me if it is possible or not and if yes, how to formulate the correct "<action application="set" .>" and "<condition>" sections in the configuration. Thanks in advance for all answers. Best Regards, Lars

5 7

Memory Leak in 1.6.20
by Jai Rangi 24 Apr '18

24 Apr '18

Not 100 % sure, but I noticed memory leak in version 1.6.20, anyone else noticed same? Just want to check out before I create a bug report. My quick test was on aws compute t2.medium with 4 GB RAM. Downgrading back to 1.6.16 seems to be working fine. Thank you, -Jai

1 0

RTP reception before sending any RTP packet
by Mickael Hubert 23 Apr '18

23 Apr '18

Hi list, I'd like to know if there's a way to tell FS to wait for RTP reception before sending any RTP packet ? We're facing a problem regarding early media with a PBX, NATed behind a dumb router : actually, if FS first sends RTP to the PBX, without waiting for RTP coming from the PBX, there's no audio. In some cases (related to the provider we use behind FS to establish call to the PSTN) FS isn't the 'first-shooter' of the RTP flow, and everything works fine, but when FS is the first to send RTP, it doesn't work... Because we can't change anything at our providers configs, we'd like to know if FS can "delay" RTP sending until it receives some RTP from the calling party. The schematic is following : PBX ===> FS ===> ...voice backbone... ===> Providers (FS is acting as an SBC) Thanks for your help,

2 3