- FreeSWITCH-users - FreeSWITCH Mailing Lists

replace SIP from user and host part
by Tamer Higazi 27 Oct '19

27 Oct '19

Hi people, My ISP provider is very special and forces me to reset the user and host part of my external profile. At the current stage, FS sets the endpoint user ID as sip-from user, and the external ip-address as the sip-from host and the ISP gateway returns me a 404 not found. When I turn off the PBX and register to the gateway directly with a softphone, the softphone can place a call. sip-from user is the userID of the gateway login and host the proxy address of the gateway. How can I set a new sip-from-user and host prior bridging a call to the gateway? I found 2 channel variables. variable_sip_from_user - sip_from_user variable_sip_from_host - sip_from_host Is that the solution manipulating these variables? Am I capable manipulate those ? Would that be the solution ? I am a little bit new to FS. For a dialplan example I welcome thankfully any advise. wireshark dump: JITSI Softphone Call: Session Initiation Protocol (100) Status-Line: SIP/2.0 100 trying -- your call is important to us Status-Code: 100 [Resent Packet: False] [Request Frame: 2481] [Response Time (ms): 13] Message Header Call-ID: bb85334b160ce6cb26ab480fd2a13236@0:0:0:0:0:0:0:0 [Generated Call-ID: bb85334b160ce6cb26ab480fd2a13236@0:0:0:0:0:0:0:0] CSeq: 2 INVITE Sequence Number: 2 Method: INVITE From: "117xxxxxx" <sip:117xxxxxx@sip12.e-fon.ch>;tag=51a3e1d6 SIP Display info: "117xxxxxx" SIP from address: sip:117xxxxxx@sip12.e-fon.ch SIP from address User Part: 117xxxxxx SIP from address Host Part: sip12.e-fon.ch SIP from tag: 51a3e1d6 To: <sip:0764823425@sip12.e-fon.ch> SIP to address: sip:076xxxxxxx@sip12.e-fon.ch SIP to address User Part: 076xxxxxxx SIP to address Host Part: sip12.e-fon.ch Via: SIP/2.0/UDP 217.162.xxx.xxx:5060;branch=z9hG4bK-383638-321b4bfd0c24f1f48906ff6382c3b18b;rport=5060 Transport: UDP Sent-by Address: 217.162.xxx.xxx Sent-by port: 5060 Branch: z9hG4bK-383638-321b4bfd0c24f1f48906ff6382c3b18b RPort: 5060 Content-Length: 0 (successfull placing the call) FS Call: Session Initiation Protocol (404) Status-Line: SIP/2.0 404 Not found Status-Code: 404 [Resent Packet: False] [Request Frame: 12659] [Response Time (ms): 13] Message Header Via: SIP/2.0/UDP 217.162.xxx.xxx:5080;rport=5080;branch=z9hG4bKmNHFUDQav3tUj;received=217.162.xxx.xxx Transport: UDP Sent-by Address: 217.162.xxx.xxx Sent-by port: 5080 RPort: 5080 Branch: z9hG4bKmNHFUDQav3tUj Received: 217.162.xxx.xxx From: "1000" <sip:1000@217.162.xxx.xxx>;tag=gNe9eHQm44HBj SIP Display info: "1000" SIP from address: sip:1000@217.162.xxx.xxx SIP from address User Part: 1000 SIP from address Host Part: 217.162.xxx.xxx SIP from tag: gNe9eHQm44HBj To: <sip:076xxxxxxx@sip12.e-fon.ch>;tag=56dedfd3319cd1b9fedc1197b499329c.26ea SIP to address: sip:076xxxxxxx@sip12.e-fon.ch SIP to address User Part: 076xxxxxxx SIP to address Host Part: sip12.e-fon.ch SIP to tag: 56dedfd3319cd1b9fedc1197b499329c.26ea Call-ID: 84f02fe4-71ef-1238-b881-00012e808c2d [Generated Call-ID: 84f02fe4-71ef-1238-b881-00012e808c2d] CSeq: 11458424 INVITE Sequence Number: 11458424 Method: INVITE Content-Length: 0

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Re: [Freeswitch-users] Voicemail on AWS
by Per Modin 26 Oct '19

26 Oct '19

On 2019-09-23 20:10 (CEST), H Yavari via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> wrote: > Date: Sun, 22 Sep 2019 09:32:56 +0000 (UTC) > From: H Yavari <hyavari(a)rocketmail.com> > To: "freeswitch-users(a)lists.freeswitch.org" > <freeswitch-users(a)lists.freeswitch.org> > Subject: Voicemail on AWS > X-Mailer: WebService/1.1.14303 YMailNorrin Mozilla/5.0 (Windows NT 10.0; > Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/77.0.3865.75 > Safari/537.36 > > I am looking for some hints how we can make an voicemail server for > our company users to have their own voicemail box, then we save all > voice files on AWS S3. I see mod_http_cache, but afraid of it's only > solution or there is better one. Hi, we used to run on S3, where we mounted the recordings with s3fs-fuse[1] which we ran successfully on 1.6.20. HOWEVER, when we updated to 1.8.5 we ran into problems with the wave files, the media stream was not saved and we ended up with empty media containers, with only some meta data of around 250 bytes ended up being saved. This issue affected about half of all recordings, and we found no correlation in regards to codecs used etc. If this was an issue with configuration, s3fs-fuse or Freeswitch we didn't investigate. We solved this by moving the storage from S3 to EFS, and are currently running it in production without issues in that regard. [1]: https://github.com/s3fs-fuse/s3fs-fuse -- Per Modin Software Engineer Working Group Two m: +46724452037 w: https://wgtwo.com e: per(a)wgtwo.com

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ICE/DTLS handshake
by Mirko Brankovic 25 Oct '19

25 Oct '19

HI, Has anyone experienced DTLS handshake takes 5s to get to SETUP state: > 2017-09-21 11:40:40.021178 [INFO] switch_rtp.c:3515 Changing audio DTLS > state from OFF to HANDSHAKE > 2017-09-21 11:40:45.319457 [INFO] switch_rtp.c:3172 Changing audio DTLS > state from HANDSHAKE to SETUP In network dump i see that answering side is not sending STUN for this 5s and then suddenly answers last 5 STUNs from A side. Has anyone encountered this kind of problem ? I have a pcap if necessary... Thanks, -- Regards, Mirko

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uuid_record to .raw crash
by Callum Guy 25 Oct '19

25 Oct '19

Easily reproducible issue, simply try and record to a file with the .raw extension. This causes FreeSWITCH to immediately crash: uuid_record 01386fbc-b552-487e-a5af-4bd1661a97f0 start /tmp/file.raw Using: FreeSWITCH Version 1.10.1-release-12-f9990221e6~64bit (-release-12-f9990221e6 64bit) OS: Debian 10 Buster (4.19.0-6-amd64) Apologies, no experience capturing a core dump and have failed to do so from the notes on https://freeswitch.org/confluence/display/FREESWITCH/Debugging, wanted to report the issue. -- *0333 332 0000 | www.x-on.co.uk <http://www.x-on.co.uk> | ** <https://www.linkedin.com/company/x-on> <https://www.facebook.com/XonTel> <https://twitter.com/xonuk> * X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales. Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. The information in this e-mail is confidential and for use by the addressee(s) only. If you are not the intended recipient, please notify X-on immediately on +44(0)333 332 0000 and delete the message from your computer. If you are not a named addressee you must not use, disclose, disseminate, distribute, copy, print or reply to this email. Views or opinions expressed by an individual within this email may not necessarily reflect the views of X-on or its associated companies. Although X-on routinely screens for viruses, addressees should scan this email and any attachments for viruses. X-on makes no representation or warranty as to the absence of viruses in this email or any attachments.

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Re: [Freeswitch-users] webrtc DTLS1.2 support in Freeswitch
by Mike Jerris 25 Oct '19

25 Oct '19

Newest release defaults to 1.2 as long as you use an openssl that supports it (note that this is not the case on some older distros but should be on at least buster). That was not the case in previous releases… You can check release notes to confirm version but I think that change was in 1.10.0. The commit note and release notes will have more details on this. > On Oct 22, 2019, at 7:55 AM, kaiduan xie via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> wrote: > > > From: kaiduan xie <kaiduanx(a)yahoo.ca> > Subject: webrtc DTLS1.2 support in Freeswitch > Date: October 22, 2019 at 7:54:37 AM MDT > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > Hi, > > Does Freeswitch support DTLS 1.2 for webrtc? If yes, which version supports DTLS 1.2? Google will drop DTLS1.0 support in M81 per > > https://bugs.chromium.org/p/webrtc/issues/detail?id=10261 <https://bugs.chromium.org/p/webrtc/issues/detail?id=10261> > > Thanks, > > /Kaiduan > >

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Date/time of call when listening to voicemail
by Peter Steinbach 25 Oct '19

25 Oct '19

Hello, when listening to voicemail, it only tells me the date of the recording but not date/time. Example * Message number 1 * 18:10 And I would like to hear * Message number 1 * Wed 23/10 at 18:10 How can I solve this? When I look into my german VM phases, I see for 'voicemail_say_date' <action function="say" data="$1" method="pronounced" type="current_date_time"/> So I expect it should tell me date and time. -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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WS SETUP FAILED
by Mundkowsky, Robert 24 Oct '19

24 Oct '19

Hi, I am getting "WS SETUP FAILED" when trying to use Verto. Is there anyway to get more detailed information about what is wrong? Note the logs are already set to debug level. 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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where repo for FS 1.8 + Debian 9 ?
by Yuriy Nasida 24 Oct '19

24 Oct '19

Hi there, I know that currently we should use Freeswitch 1.10 + Debian 10 but I really need FS 1.8 + Debian 9. Looks like that repo is removed. Please advice if I am not correct and stil able to use it. Thanks

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Re: [Freeswitch-users] Multiple History Info header
by Giri 24 Oct '19

24 Oct '19

*I am trying to export sip_h_Diversion header twice but not working, only one header is sent out and not two. * *If I use export with “,” then I get the following results: *>* Diversion: <sip:19082302337 at 216.1.185.103 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>;reason=user-busy;counter=3;privacy=off,<sip:13216938059 at 216.1.185.103 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>;reason=no-answer;counter=1;privacy=off *> >* <extension name="public_did"> *>* <condition field="destination_number" expression="^9999(\d{10})$"> *> >* <action application="set" data="ocn=18005551212"/> *>* <action application="set" data="rdn=18005552323"/> *> >* <action application="export" data="sip_h_Diversion =<sip:${ocn}@${network_addr}>;reason=user-busy;counter=3;privacy=off, <sip:${rdn}@${network_addr}>;reason=no-answer;counter=1;privacy=off "/> *> >* <action application="set" data="hangup_after_bridge=true" /> *>* <action application="set" data="continue_on_fail=true"/> *> >* <action application="transfer" data="1001 XML default"/> *>* </condition> *>* </extension>* With Best Regards, Girish Dharmaraj On Sun, Oct 6, 2019 at 12:47 PM Giri <girish.dharmaraj(a)gmail.com> wrote: > *I am trying to export sip_h_Diversion header twice but not working, only one header is sent out and not two. * > > *If I use export with “,” then I get the following results: > *>* Diversion: <sip:19082302337 at 216.1.185.103 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>;reason=user-busy;counter=3;privacy=off,<sip:13216938059 at 216.1.185.103 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>;reason=no-answer;counter=1;privacy=off > *> >* <extension name="public_did"> > *>* <condition field="destination_number" expression="^9999(\d{10})$"> > *> >* <action application="set" data="ocn=18005551212"/> > *>* <action application="set" data="rdn=18005552323"/> > *> >* <action application="export" data="sip_h_Diversion =<sip:${ocn}@${network_addr}>;reason=user-busy;counter=3;privacy=off, <sip:${rdn}@${network_addr}>;reason=no-answer;counter=1;privacy=off "/> > *> >* <action application="set" data="hangup_after_bridge=true" /> > *>* <action application="set" data="continue_on_fail=true"/> > *> >* <action application="transfer" data="1001 XML default"/> > *>* </condition> > *>* </extension>* > > > With Best Regards, > Girish Dharmaraj > > > > <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> Virus-free. > www.avg.com > <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> > <#m_6076873152175307791_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2> > > On Sat, Oct 5, 2019 at 12:41 AM Giri <girish.dharmaraj(a)gmail.com> wrote: > >> when I add a multiple header with comma separated only one header is >> added, is there anything wrong with my syntax ? >> >> sip_h_History-Info=<sip:${CF_caller_id_number}@198.18.181.6;user=phone>;index=1;ms-retarget-reason=forwarding, >> <sip:${callee_id_number}@198.18.181.6;user=phone>;index=1.1; >> >> The Result is showing in one header. >> >> >> With Best Regards, >> Girish Dharmaraj >> >

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Cancel X-Reason between servers
by Jon Schøpzinsky 24 Oct '19

24 Oct '19

Hi, We have the following setup, with a queue server, running our own queue software, a main switch and a SBC. Queue FreeSWITCH -> Master FreeSWITCH -> OpenSIPS -> UA When a queue dials multiple UA’s, an one of them answers, the rest is correctly sent a CANCEL with the LOOSE_RACE as X-Reason header. This is sent from the Queue FreeSWITCH to the Master FreeSWITCH, but when the Master FreeSWITCH sends this on to the OpenSIPS, the X-Reason header gets changed to ORIGINATOR_CANCEL. I found the code that does this, and it seems to be this block in switch_ivr_originate: https://github.com/signalwire/freeswitch/blob/f9990221e6094886066ec2bf96856… <https://github.com/signalwire/freeswitch/blob/f9990221e6094886066ec2bf96856…> This always sets the reason to be ORIGINATOR_CANCEL, no matter what was set by the upstream server. I know FreeSWITCH is not supposed behave like a proxy, and even OpenSIPS overwrites CANCEL reasons in its standard configuration, but would there be interest in us implementing proxying functionality, using a configuration item or channel variable, enabling this reason to be proxied through? Kind Regards Jon Schøpzinsky

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