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Freeswitch failed to initiate outbound call using SIPs + SRTP (SRTP unprotect )
by Chhorm Chhatra 22 Oct '18

22 Oct '18

Hello, Currently, I faced a problem regarding SRTP outbound call to user (Leg B). The scenario is like this, - We set up our own root CA to an IP address (e.g 192.168.0.13) - We create a server certificate for freeswitch at 192.168.0.13 - Linphone is used as SIP client and is configured to trust our root CA by default. - Linphone A is configured to register to Freeswitch vis TLS + SRTP. (One leg call to server has both SIPs and SRTP – completely secure) - Linphone B is registered to Freeswitch via TLS + SRTP, and waiting for Linphone A to call to. (One leg call to server, e.g. 9196 (echo test), is completely secure with SRTP + SIPs) - Unfortunately, if A call to B, only A leg has SIPs + SRTP, but Leg B is not encrypted with SRTP and SIPs at all. This causes *SRTP unprotect failed with code 7 (auth check failed)**.* + Dialplan Configuration <action application="set" data="rtp_secure_media=true"/> <action application="export" data="rtp_secure_media=true"/> The dial-string is <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> + Directory Configruation: <param name="dial-string" value="{rtp_secure_media=${ regex(${sofia_contact(${dialed_user}@${dialed_domain}) }|transport=tls)},presence_id=${dialed_user}@${dialed_ domain}}${sofia_contact(${dialed_user}@${dialed_domain})}" /> My question is that, is there any configuration left that I have to set up in order to let freeswitch initiate an outbound call to Leg B correctly with SRTP and SIPs (tls)? Any help would be really appreciated. Thank you so much. Best Regard,

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VideoCall hangup always after 32 seconds to answer
by Jose David Jurado Alonso 20 Oct '18

20 Oct '18

Hi, I'ḿ get the next dialplan configuration in order to set the call max limit timeout on 120 seconds after answer: <extension name="Local_Extension"> <condition field="destination_number" expression="^(.*)$"> <action application="export" data="dialed_extension=$1"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="ignore_early_media=false"/> <action application="set" data="execute_on_answer=sched_hangup +120" /> <action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/> </condition> </extension> But the call always disconnect after 32 seconds... (calling linphone to linphone android clients). I tried always using "call_timeout", "originate_timeout" but the result is the same. What I really want to do is: + Set timeout of 30 seconds to hangup if NO answer (nobody responds). + Set a maximum total time of 120 seconds since the call is answered. Can someone help me with this configuration? FS version: FreeSWITCH Version 1.9.0-654-ed4920e~64bit (-654-ed4920e 64bit) Thanks! José David Jurado Alonso.

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Accepting an "optional" SRTP offer (crypto in RTP/AVP) and establishing SRTP
by William Simon 16 Oct '18

16 Oct '18

We are working with an endpoint that offers optional SRTP in the non-RFC-compliant way of an a=crypto attribute within RTP/AVP. I have told FreeSWITCH to allow this with rtp_allow_crypto_in_avp=true at the right place in the dialplan. Immediately after that in the dialplan I have to reject SRTP by using rtp_secure_media=forbidden, otherwise the call setup still fails. By setting the value to forbidden, the call does proceed unencrypted. We want FreeSWITCH to proceed with media encryption. Setting rtp_secure_media to any other value results in FreeSWITCH rejecting the offer like this: v=0 o=FreeSWITCH 1538660754 1538660755 IN IP4 192.168.100.104 s=FreeSWITCH c=IN IP4 192.168.100.104 t=0 0 m=audio 0 RTP/AVP 19 m=video 0 RTP/AVP 19 Is there anything else I can do to force SRTP in the answer? “The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.”

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Multi gateways in profile
by Mickael Hubert 16 Oct '18

16 Oct '18

Hi guys, I want to identify source of inbounds calls. In my external profile I have two gateways: - One with REGISTER method - One without REGISTER method (IP to IP), only iptables filtering <gateways> <gateway name="GW1"> <param name="from-domain" value="**********"/> <param name="caller-id-in-from" value="true"/> <param name="proxy" value="1.1.1.1"/> <param name="register" value="true"/> <param name="username" value="**************"/> <param name="password" value="****************"/> <param name="expire-seconds" value="3600"/> <param name="retry-seconds" value="30"/> <param name="ping" value="25"/> </gateway> <gateway name="GW2"> <param name="from-domain" value="*********"/> <param name="caller-id-in-from" value="true"/> <param name="proxy" value="2.2.2.2"/> <param name="register" value="false"/> <param name="retry-seconds" value="30"/> <param name="ping" value="25"/> </gateway> </gateways> In my dialplan I have this conf: <condition field="${sip_gateway}" expression="GW1" break="never"> <action application="set" data="number_dest=${destination_number}" /> <action application="execute_extension" data="to-private XML from-gw1"/> </condition> <condition field="${sip_gateway}" expression="GW2" break="never"> <action application="set" data="number_dest=${destination_number}" /> <action application="execute_extension" data="to-private XML from-gw2"/> </condition> When a call arrives throught GW1 (REGISTERED), ${sip_gateway} is populate with "GW1" perfect !. But when another call arrives throught GW2 (not REGISTERED), ${sip_gateway} is empty. Please, how can I identify the source gateway where calls arrive ? maybe it's not a good way to do that ? thanks in advance

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[Missing local host] in proxy media mode
by Marc Quore 16 Oct '18

16 Oct '18

Hi, All I've got FreeSwitch 1.8.2 running in media proxy mode: <param name="inbound-proxy-media" value="true"/> <param name="inbound-late-negotiation" value="true"/> Dialplan is configured to bridge incoming calls to remote gateway and most of calls completing successfully, however some of them fail with INCOMPATIBLE_DESTINATION cause. For failed calls everything goes normally until bridged connection gets early media from callee: [NOTICE] sofia.c:7304 Pre-Answer sofia/external/2222(a)2.2.2.2! [INFO] switch_ivr_originate.c:3747 Sending early media [DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111(a)1.1.1.1 PCMU/0 0 ms 960 samples 384000 bits 1 channels [DEBUG] switch_core_codec.c:111 sofia/internal/1111(a)1.1.1.1 Original read codec set to PCMU:0 [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/ 1111(a)1.1.1.1] 1.1.1.1:18792->1.1.1.1:18792 codec: 0 ms: 20 [ERR] switch_core_media.c:9510 AUDIO RTP REPORTS ERROR: *[Missing local host]* [NOTICE] switch_core_media.c:9511 Hangup sofia/internal/1111(a)1.1.1.1 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] Here's a dump for this call: [image: image.png] Healthy call on the same machine: [DEBUG] switch_core_media.c:3781 Set Codec sofia/vpn/2222(a)2.2.2.2 PROXY/0 0 ms 160 samples 0 bits 1 channels [DEBUG] switch_core_codec.c:111 sofia/internal/2222(a)2.2.2.2 Original read codec set to PROXY:0 [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/ 2222(a)2.2.2.2] 2.2.2.2:60818->2.2.2.2:60818 codec: 0 ms: 20 [DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111(a)1.1.1.1 PROXY/0 0 ms 160 samples 0 bits 1 channels [DEBUG] switch_core_codec.c:111 sofia/external/1111(a)1.1.1.1 Original read codec set to PROXY:0 [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/external/ 1111(a)1.1.1.1] 1.1.1.1:31412->1.1.1.1:31412 codec: 0 ms: 20 So it looks like proxified media connection cannot be established for some calls. It shouldn't be a codec problem as call parties are configured with common codecs, FS is bond to local interface which is running with no errors. But I have no idea what "local host is missing" in my case so I'll be appreciated for any hint. FreeSWITCH Version 1.8.2+git~20180926T175525Z~a98a958ac3~64bit (git a98a958 2018-09-26 17:55:25Z 64bit) CentOS Linux release 7.5.1804 (Core) Best, Marc

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How to update the caller id when bridging?
by Andrew Clayton 16 Oct '18

16 Oct '18

So this is a question based around the FreeSWITCH C API. I have two scenarios involving two endpoints, A & B. Scenario 1 FS calls B using switch_ivr_originate(), it then calls A (say a Linphone client) and creates a new switch_caller_profile_t for it using switch_caller_profile_new() and sets the ->caller_id_number field to e.g 1234 to indicate a call from B and then does a switch_ivr_originate(). A rings and Linphone shows the call coming from 1234@... OK, that's all fine. The calls are bridged together with switch_ivr_uuid_bridge(A, B) Scenario 2 FS calls A (again lets say a Linphone client) using switch_ivr_originate(), A rings and shows the call coming from 0000...@..., that's fine, I didn't set any CID. Now A is parked and is off-hook. FS then calls B, A & B are then bridged together with switch_ivr_uuid_bridge(A, B), now of course Linphone still shows the call as from 0000...@... I'd like it show for example 1234@... like above. Now I've tried setting numerous fields in the switch_caller_profile_t structure for A just before doing the bridge, but to no avail. So does what I'm trying to do make sense? Is it even possible? Cheers, Andrew

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Re- DTMF 2833 issue
by Andrew Keil 14 Oct '18

14 Oct '18

To FreeSWITCH Users, I have just noticed when making calls internationally to the UK from Australia that the following SDP sometimes happens when the call lands on FreeSWITCH: d5f14e47-2156-4236-be16-4eef1b7e9d3d 2018-10-04 06:59:28.819816 [DEBUG] sofia.c:7301 Remote SDP: d5f14e47-2156-4236-be16-4eef1b7e9d3d v=0 d5f14e47-2156-4236-be16-4eef1b7e9d3d o=sbc-uk-mr-dh05a 451499 139728 IN IP4 XXX.XXX.XXX.XXX d5f14e47-2156-4236-be16-4eef1b7e9d3d s=sip call d5f14e47-2156-4236-be16-4eef1b7e9d3d c=IN IP4 XXX.XXX.XXX.XXX d5f14e47-2156-4236-be16-4eef1b7e9d3d t=0 0 d5f14e47-2156-4236-be16-4eef1b7e9d3d m=audio 62754 RTP/AVP 8 d5f14e47-2156-4236-be16-4eef1b7e9d3d a=rtpmap:8 PCMA/8000 d5f14e47-2156-4236-be16-4eef1b7e9d3d a=ptime:20 This then creates the following message around call answer: d5f14e47-2156-4236-be16-4eef1b7e9d3d 2018-10-04 06:59:28.839799 [DEBUG] switch_core_media.c:5766 No 2833 in SDP. Liberal DTMF mode adding 101 as telephone-event. >From this point on DTMF is NOT detected by the FreeSWITCH IVR service. I assume I would then need to use mod_dptools: start_dtmf to detect the DTMF tones within the channel. Which I am yet to test. However my main question is whether there is a channel variable that would enable me to check whether 2833 support is not inside the SDP? Regards, Andrew Keil

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Freeswitch is not handling 180 after 200 OK
by Mody, Darshan (Darshan) 14 Oct '18

14 Oct '18

Hi We are running UDP traffic. During heavy load FS receives sometimes receives 180 response after 200 OK already received on the same call-id (dialog). However FS does not discard the same and starts processing 180 even though it has already received 200 OK on the same call-id (dialog). Freeswitch should be discarding the 180 response after it gets 200 OK response from far-end. In our case the FS core state machine even though it gets 200 OK waits for 200 Ok again. Thanks Darshan

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Is it possible add new item in sip message header?
by 薛光宏 13 Oct '18

13 Oct '18

Hi All, We run a mobile app to *register* FreeSWITCH 1.6.20 via sip. This app will check value of server field in message header but the packet sent from FreeSWITCH doesn't have this field. We would like to know is it possible to add filed "Server" to message header by adjusting FreeSWITCH configuration? We tried this by adjusting dialplan ( https://freeswitch.org/confluence/display/FREESWITCH/Sofia+SIP+Stack) <action application="set"><![CDATA[sip_h_Server=xxxxxx]]></action> it works in invite stage, but can not work in register stage. The app checks the filed "Server" in register stage. Thanks in advance. -- Sincerely Albert Hsueh System Engineer Cloudpe Corporation 02-8712 5955 ext. 831 www.cloudpe.com

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Freeswitch 1.8.2 : Debian Package : sometimes crash when executing nolocal:execute_on_answer lua script.
by Julien Terrasson 12 Oct '18

12 Oct '18

I'm trying to use freeswitch as a transcription gateway : The goal is to allow a PSTN subscriber (A_party) to call another (B_party) through freeswitch : freeswitch being used to record and transcript the call. This is the scenario i'm trying to acheive : 1/ A_party call freeswitch through a PSTN gateway. 2/ freeswitch execute a lua script (IVRPrompt8.lua) that : * query the service database * Initialize few sessions variables * Play A_party a vocal prompt. * Attempt to bridge B_party. 3/ When B_party answer, a lua script is called : catchBAnswer2.lua This script is used to initialise several session variable and play legal warning to both party. <extension name="experimental_dialplan"> <condition field="destination_number" expression="^(\+33XXXXXXXXX)$">  <action inline="true" application="lua" data="IVRPrompt8.lua" />  <action application="export" data="RECORD_READ_ONLY=true"/> <action application="export" data="RECORD_STEREO=false"/>  <action application="set" data="ivr_menu_terminator=none"/> <action application="export" data="nolocal:execute_on_answer=lua catchBAnswer2.lua"/>  <action application="bridge" data="sofia/gateway/ippi_33XXXXXXXXX/${destination_number}"/> </condition> </extension> Sometimes the scenario works just fine, but most of the time it crash when calling catchBAnswer2.lua, as show the last console logs (DEBUG LEVEL): 2018-10-12 09:42:26.812449 [NOTICE] sofia.c:7304 Pre-Answer sofia/external5090/0665199963! 2018-10-12 09:42:26.812449 [DEBUG] switch_channel.c:3482 (sofia/external5090/0665199963) Callstate Change RINGING -> EARLY 2018-10-12 09:42:26.812449 [DEBUG] sofia.c:7291 Channel sofia/external5090/0665199963 entering state [ready][200] 2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:5478 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:5533 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:5394 Set telephone-event payload to 101@8000 2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:3781 Set Codec sofia/external5090/0665199963 PCMA/8000 20 ms 160 samples 64000 bits 1 channels 2018-10-12 09:42:26.812449 [DEBUG] switch_core_codec.c:111 sofia/external5090/0665199963 Original read codec set to PCMA:8 2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:5737 Set telephone-event payload to 101@8000 2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:5795 sofia/external5090/0665199963 Set 2833 dtmf send payload to 101 recv payload to 101 2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:8511 AUDIO RTP [sofia/external5090/0665199963] 94.23.42.139 port 29498 -> 194.169.214.61 port 59684 codec: 8 ms: 20 2018-10-12 09:42:26.812449 [DEBUG] switch_rtp.c:4300 Starting timer [soft] 160 bytes per 20ms 2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:8815 sofia/external5090/0665199963 Set 2833 dtmf send payload to 101 2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:8822 sofia/external5090/0665199963 Set 2833 dtmf receive payload to 101 2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:8845 sofia/external5090/0665199963 Set rtp dtmf delay to 40 2018-10-12 09:42:26.812449 [NOTICE] sofia.c:8429 Channel [sofia/external5090/0665199963] has been answered EXECUTE sofia/external5090/0665199963 lua(catchBAnswer2.lua) ** CORE DUMP ** I have no clues why this is happening, so i included the backtrack enclosed to have feedbacks (https://freeswitch.org/jira/browse/FS-11456) Can somebody have a look and give a more precise idea on what is making freeswitch to crash ? I can then setup monitoring and collect additional traces if needed (SIP, RTP, etc..). J. Terrasson

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