January 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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

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No response to registration requests with high port numbers?
by Alex Lake 15 Jan '18

15 Jan '18

Hi, Got a problem with a handset that doesn't seem to be registering on our Freeswitch server. The diagnostics from a member of Cisco support said... "At packet 422 we see the phone "0_4147057716(a)192.168.40.26" send a registration request to 5.9.36.113:5060 from 175.159.77.82:32060 - We do not see a reply to this from the SIP provider. We responses to requests from 0_359240637(a)192.168.40.26 <mailto:0_359240637@192.168.40.26> with source port 5060, which suggests to me that the SIP provider is not replying to the high UDP port requests. As the MX is a NAT device, subsequent SIP requests will always have a high UDP port." I'm not sure where we might configure acceptable ranges of SIP source or if it's something else. Thanks for any suggestions... Alex

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Problem With TLS + Contact Header - Bug?
by Geoff Mina 15 Jan '18

15 Jan '18

Greetings, I have a pretty odd issue that I have no idea how to resolve. We have a Sofia profile that is TLS only. It listens on 5061. This is with an upstream vendor and we want them to be able to send inbound calls to extensions without register. Our FS instance is sending back "transport=udp" and port 5060 in the Contact header of the 200 OK. Even though the invite is for TLS and the sip_profile is tls-only=true. I have attached a SIP trace and a copy of our external.xml sofia profile. I am thinking this is a Freeswitch bug - or maybe just an oddly documented "feature". Any help is really appreciated. In the sip-trace attached, you can see an INVITE followed by a Trying and then an OK. The OK has Contact information for transport=udp and port 5060, which is invalid, so the ACK never gets back to us and we retransmit the 200 OK until it expires. Thanks, Geoff

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Fresh i386 install failing to download packages
by LinMin 15 Jan '18

15 Jan '18

Hi all, Just trying to install on Debian 8 32bit and getting the following messages in apt: Package 'freeswitch-all' has no installation candidate Unable to locate package freeswitch-meta-all System is a Via C7 256mb ram 16gb ssd. Other packages are installing fine. Apt is updated. Doing 'apt-cache search freeswitch' shows some packages, but not all. Have I missed something here? Cheers, Francis

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Freeswitch as sip phone and call transfer
by Mehdi Shirazi 14 Jan '18

14 Jan '18

Hi Is it possible to use Freeswitch as sip softphone and use call transfer ? please note this is not same when Freeswitch connected to provider with sip trunk and makes call transfer.when Freeswitch registers in provider as sip phone , when he receives a call ,Transfer calls via provider network and in this case Freeswitch is not in signaling or media path. My goal is dynamically transfer sip phone calls based on time and caller id via provider network.There is no siptrunk just one sip phone account. thanks

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Freeswitch as sip phone and call transfer
by Mehdi Shirazi 14 Jan '18

14 Jan '18

HiIs it possible to use Freeswitch as sip softphone and use call transfer ? please note this is not same when Freeswitch connected to provider with sip trunk and makes call transfer.when Freeswitch registers in provider as sip phone , when he receives a call ,Transfer calls via provider network and in this case Freeswitch is not in signaling or media path.My goal is dynamically transfer sip phone calls based on time and caller id via provider network.There is no siptrunk just one sip phone account. thanks M.Shirazi

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ESL - Event Variables
by Gregor Nanger 14 Jan '18

14 Jan '18

Hi Guys! I am using ESL. On some events like ChannelState there are only few variables sent to ESL client. Is there a way to for some variables, that I created in channel, to force them to be sent on each event? Thank you for advice, Gregor

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[mod http cache][zero byte file]
by Bilal Abbasi 12 Jan '18

12 Jan '18

Hi Users, I am using mod http cache in freeswitch to get sound file from amazon S3, today i noticed a file downloaded with zero bytes, i tried alot to replicate the issue but could not able to get that. I was just intrested to know if someone get across this issue? Or some one using http cache in production. P.S: my system is in dev phase right now, and i am curious to know is it safe to use http cache in prod. Regards Abbasi

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Re: [Freeswitch-users] verto, private ip, stun, and Chrome
by Mundkowsky, Robert 11 Jan '18

11 Jan '18

Well, we found out commenting out the ACLs in the verto.conf.xml worked:  Seems the client's private IP was one from our LAN, so FreeSWITCH thought it was a valid request from our LAN. But I wonder if STUN should have fixed this? Robert Mundkowsky From: Mundkowsky, Robert Sent: Thursday, January 11, 2018 5:25 PM To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: verto, private ip, stun, and Chrome Hi, We are using a Verto javascript that is maybe a little out of date and most recent version of Chrome. We are seeing that client's private IP address is being sent to FreeSWITCH rather than client's public IP. We have "iceServers: true" which should tell verto to use the default google STUN server. I am guessing the below code that uses "navigator.mozGetUserMedia" likely is causing the problems since getusermedia has been tweaked a lot by Chrome and Firefox (per https://developer.mozilla.org/en-US/docs/Web/API/Navigator/getUserMedia ). window.moz = !!navigator.mozGetUserMedia; function RTCPeerConnection(options) { var gathering = false, done = false; var w = window, PeerConnection = w.mozRTCPeerConnection || w.webkitRTCPeerConnection, SessionDescription = w.mozRTCSessionDescription || w.RTCSessionDescription, IceCandidate = w.mozRTCIceCandidate || w.RTCIceCandidate; var STUN = { url: !moz ? 'stun:stun.l.google.com:19302' : 'stun:23.21.150.121' }; robert ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________

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verto, private ip, stun, and Chrome
by Mundkowsky, Robert 11 Jan '18

11 Jan '18

Hi, We are using a Verto javascript that is maybe a little out of date and most recent version of Chrome. We are seeing that client's private IP address is being sent to FreeSWITCH rather than client's public IP. We have "iceServers: true" which should tell verto to use the default google STUN server. I am guessing the below code that uses "navigator.mozGetUserMedia" likely is causing the problems since getusermedia has been tweaked a lot by Chrome and Firefox (per https://developer.mozilla.org/en-US/docs/Web/API/Navigator/getUserMedia ). window.moz = !!navigator.mozGetUserMedia; function RTCPeerConnection(options) { var gathering = false, done = false; var w = window, PeerConnection = w.mozRTCPeerConnection || w.webkitRTCPeerConnection, SessionDescription = w.mozRTCSessionDescription || w.RTCSessionDescription, IceCandidate = w.mozRTCIceCandidate || w.RTCIceCandidate; var STUN = { url: !moz ? 'stun:stun.l.google.com:19302' : 'stun:23.21.150.121' }; robert ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________

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