September 2019 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FYI: DEB packages compiling trick
by Giovanni Maruzzelli 12 Sep '19

12 Sep '19

Hello fellow FreeSWITCHers, I just added to documentation in confluence, but I repeat here, for the defense of your hair: ***after installing cowbuilder*** and before building packages, you must, from command line: # echo "ALLOWUNTRUSTED=yes" >> /etc/pbuilderrc because nonstandard packages from freeswitch repo are not trusted by pbuilder. -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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Verto call waiting
by Jose Figueroa 12 Sep '19

12 Sep '19

Hello, I'm trying to implement a verto call waiting (2nd call). I mean when I'm in a call and there's a second call to the same extension I need that verto client can take the second call and leave the first call on waiting. But I cannot complete this, I see on old mail https://lists.freeswitch.org/pipermail/freeswitch-users/2016-May/120422.html which says that on the same video tag dynamically is set. Do you have any examples of this? Any help would be appreciated. Thank you. Jose Figueroa

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ETA 1.10.2? you cannot use DB with mod_fifo and mod_directory in 1.10.1
by Giovanni Maruzzelli 10 Sep '19

10 Sep '19

Hello fellow FreeSWITCHers, there is already an ETA for 1.10.2? Atm, with packages from 1.10.1, you cannot use DB in mod_fifo and mod_directory, due to a bug fixed in master... ( https://freeswitch.org/jira/browse/FS-11995 ) Obviously, if you compile from git master, is already fixed, thanks to Andrey Volk -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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Removing RPID header from 183/200 answer
by Igor Olhovskiy 10 Sep '19

10 Sep '19

Hi! I have a strange issue. Phone dials to Freeswitch with this INVITE INVITE sip:012345678@mypbx.net;user=phone SIP/2.0 ... Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change .... But than 200 OK from FS comes with Remote-Party-ID header SIP/2.0 200 OK .... User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces ... Remote-Party-ID: "012345678" <sip:12345678@mypbx.net>;party=calling;privacy=off;screen=no (https://link.getmailspring.com/link/171DA887-7BD6-4B11-A4A5-491E6BB82DFF@ge…) Main problem - in this header 0 is stripped from number and phone updates called number to without leading 0 and saves in a call list incorrect. (One of ideas why 0 is stripped - regular expression for this destination number is ^0([1-9]\d{4,30})$) My question is - how to suppress adding RPID header in FS answers? FS: 1.8.7

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caller ID updation on consultative transfers
by Gaurang Gohil 10 Sep '19

10 Sep '19

how can i achieve the below scenario using freeswitch when extension A calls extension B and extension B forward the call to external number, caller ID that is presented to the external number should be extension A. A(1000)(callerID=2323) -> B(2000)(callerID=12345) B transfer(blind) the call to XX929. XX929 show callerID as 2323 A(1000)(callerID=2323) -> B(2000)(callerID=12345) B transfer(consultative) the call to XX929. XX929 show callerID as 12345 caller ID updation is working for "blind transfer" but not for "consultative transfers" -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.

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SERVER DOWN - HELP
by Sean Devoy 09 Sep '19

09 Sep '19

I am traveling and FreeSwitch died. I have no idea what to do. This is the latest distro I think. Here is what I can see, after trying reboot: root@debian-FreeSwitch:~# service freeswitch restart Job for freeswitch.service failed because the control process exited with error code. See "systemctl status freeswitch.service" and "journalctl -xe" for details. root@debian-FreeSwitch:~# systemctl status freeswitch.service ● freeswitch.service - freeswitch Loaded: loaded (/lib/systemd/system/freeswitch.service; enabled; vendor preset: enabled) Active: failed (Result: exit-code) since Sun 2019-09-08 23:01:42 UTC; 19s ago Process: 1532 ExecStart=/usr/bin/freeswitch -u freeswitch -g freeswitch -ncwait $DAEMON_OPTS (code=exited, status=1/FAILU Process: 1529 ExecStartPre=/bin/chown -R freeswitch:freeswitch /var/lib/freeswitch /var/log/freeswitch /etc/freeswitch /u Sep 08 23:01:42 debian-FreeSwitch systemd[1]: Failed to start freeswitch. Sep 08 23:01:42 debian-FreeSwitch systemd[1]: freeswitch.service: Unit entered failed state. Sep 08 23:01:42 debian-FreeSwitch systemd[1]: freeswitch.service: Failed with result 'exit-code'. Sep 08 23:01:42 debian-FreeSwitch systemd[1]: freeswitch.service: Service hold-off time over, scheduling restart. Sep 08 23:01:42 debian-FreeSwitch systemd[1]: Stopped freeswitch. Sep 08 23:01:42 debian-FreeSwitch systemd[1]: freeswitch.service: Start request repeated too quickly. Sep 08 23:01:42 debian-FreeSwitch systemd[1]: Failed to start freeswitch. Sep 08 23:01:42 debian-FreeSwitch systemd[1]: freeswitch.service: Unit entered failed state. Sep 08 23:01:42 debian-FreeSwitch systemd[1]: freeswitch.service: Failed with result 'exit-code'. lines 1-15/15 (END)...skipping... ● freeswitch.service - freeswitch Loaded: loaded (/lib/systemd/system/freeswitch.service; enabled; vendor preset: enabled) Active: failed (Result: exit-code) since Sun 2019-09-08 23:01:42 UTC; 19s ago Process: 1532 ExecStart=/usr/bin/freeswitch -u freeswitch -g freeswitch -ncwait $DAEMON_OPTS (code=exited, status=1/FAILURE Process: 1529 ExecStartPre=/bin/chown -R freeswitch:freeswitch /var/lib/freeswitch /var/log/freeswitch /etc/freeswitch /usr Sep 08 23:01:42 debian-FreeSwitch systemd[1]: Failed to start freeswitch. Sep 08 23:01:42 debian-FreeSwitch systemd[1]: freeswitch.service: Unit entered failed state. Sep 08 23:01:42 debian-FreeSwitch systemd[1]: freeswitch.service: Failed with result 'exit-code'. Sep 08 23:01:42 debian-FreeSwitch systemd[1]: freeswitch.service: Service hold-off time over, scheduling restart. Sep 08 23:01:42 debian-FreeSwitch systemd[1]: Stopped freeswitch. Sep 08 23:01:42 debian-FreeSwitch systemd[1]: freeswitch.service: Start request repeated too quickly. Sep 08 23:01:42 debian-FreeSwitch systemd[1]: Failed to start freeswitch. Sep 08 23:01:42 debian-FreeSwitch systemd[1]: freeswitch.service: Unit entered failed state. Sep 08 23:01:42 debian-FreeSwitch systemd[1]: freeswitch.service: Failed with result 'exit-code'. Thanks, Sean

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Freeswitch 1.10 conference choppy audio issue
by devang nathwani 06 Sep '19

06 Sep '19

Hello Team, OS: Debian 10.0 FS Version: FreeSWITCH (Version 1.10.1 -release-12-f9990221e6 64bit) The issue when we are adding agents in conference with external call, we start facing choppy audio sound. We are using g729 codec in pass through, vars.xml configured with global_codec_prefs and outbound_codec_prefs set to G729 We have tried setting rtp-autofix-timing to false, but it didn't help. inbound-late-negotiation is set to true. Here we are attaching the .txt of logs and also adding the pastebin link https://pastebin.freeswitch.org/view/b4deae7e Please let us know what configuration should be done to resolve this issue. Thanks in advance.

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Teens Direct Routing - contact and record-route headers are missing
by Tulipa Frigo 06 Sep '19

06 Sep '19

Hi all, I'm trying to configure my freeswitch to send and receive calls from Teens Direct Routing . Then, I configured the SIP gateway /usr/local/freeswitch/sip_profiles/extern/gateway_teens.xml : <include> <gateway name="teens"> <param name="realm" value="sip.pstnhub.microsoft.com:5061"/> <param name="register" value="false"/> <param name="register-transport" value="tls"/> <param name="ping" value="25"/> </gateway> </include> The Teens Direct Routing only enable the sip trunk after receive and send SIP OPTIONS with success, in both direction. The OPTIONS FLOW description from Direct Routing Configure page, https://docs.microsoft.com/en-us/microsoftteams/direct-routing-configure: "The SBC sends a OPTIONS messages to Teens Direct Routing and receives a 200OK. When Direct Routing sees incoming OPTIONS, it will start sending outgoing SIP Options messages to the SBC FQDN configured in the Contact header field in the incoming OPTIONS message and the SBC reply with 200OK." The problem is that the freeswitch (SBC) is receiving the response FORBIDDEN , with reason : Q.850;cause=21;text="29175f0d-ebe8-43ea-836f-708b6d4b0d4f;Record-Route and Contact headers are missing" . Folowing is the OPTIONS msg sent from freeswitch to Teens Direct Rounting and the response: ------------------------------------------------------------------------ send 597 bytes to tls/[52.114.76.76]:5061 at 11:10:15.890560: ------------------------------------------------------------------------ OPTIONS sip:sip.pstnhub.microsoft.com:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 177.53.17.168:5081;rport;branch=z9hG4bKg5DyZ09c7Bv2r Max-Forwards: 70 From: <sip:sip.pstnhub.microsoft.com:5061>;tag=ge55jU7NBK1rr To: <sip:sip.pstnhub.microsoft.com:5061> Call-ID: 3f0f42b7-3a09-1238-5e8a-00155d11a606 CSeq: 8385268 OPTIONS User-Agent: FreeSWITCH-mod_sofia/1.8.6+git~20190606T190249Z~93b4c92e75~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 ------------------------------------------------------------------------ recv 545 bytes from tls/[52.114.76.76]:5061 at 11:10:16.070876: ------------------------------------------------------------------------ SIP/2.0 403 Forbidden FROM: <sip:sip.pstnhub.microsoft.com:5061>;tag=ge55jU7NBK1rr TO: <sip:sip.pstnhub.microsoft.com:5061> CSEQ: 8385268 OPTIONS CALL-ID: 3f0f42b7-3a09-1238-5e8a-00155d11a606 VIA: SIP/2.0/TLS 177.53.17.168:5081;branch=z9hG4bKg5DyZ09c7Bv2r;rport REASON: Q.850;cause=21;text="29175f0d-ebe8-43ea-836f-708b6d4b0d4f;Record-Route and Contact headers are missing" CONTENT-LENGTH: 0 ALLOW: INVITE ALLOW: ACK ALLOW: OPTIONS ALLOW: CANCEL ALLOW: BYE ALLOW: NOTIFY SERVER: Microsoft.PSTNHub.SIPProxy v.2019.7.4.9 i.EUNO.2 ------------------------------------------------------------------------ How can I add these headers into OPTIONS msg ? Thanks and Regards Tulipa Frigo

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How to add proxy realm to CLI originate command
by mario_fs 05 Sep '19

05 Sep '19

I am getting incorrect proxy authorization due to realm error. Trying many things nothing works. So far I have added sip_auth_realm and fs_path but the realm becomes my local IP address instead of the remote URL which is required: originate {sip_auth_username= 1234567890130,sip_auth_password=mypass,sip_auth_realm=callcentric.com}sofia/internal/1234567890130@callcentric.com;fs_path=sip:callcentric.com &echo According to ITSP, I need “callcentric.com <http://callcentric.com/>” instead of 10.12.13.7” in what I see in the INVITE message: Proxy-Authorization: Digest username="1234567890130", realm="10.12.13.7", nonce="1bb85bb0553333e87e0779e40444df4f", opaque="", algorithm=MD5, uri="sip: 1234567890130(a)callcentric.com", response="a269eaa24433332b49e8e5521002484a" Anyone know if this is even possible? I am using CLI to test then plan to use in bridge command in LUA. Thanks. Mario G

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Help with originate/bride command from FS to ITSP extension
by mario_fs 05 Sep '19

05 Sep '19

I am trying to get FS to make a call using a callcentric extension (not FS extension) without setting up the extension as an external gateway, I don’t want permanent since it’s only used to forward calls to cellphone. I am testing with the CLI originate command below, before I put it into a LUA script. originate {sip_auth_username=1234567890130,sip_auth_password=mypass,sip_auth_realm=callcentric.com}sofia/internal/1234567890130@callcentric.com;fs_path=sip:callcentric.com &echo Callcentric is returning: SIP/2.0 403 Incorrect Authentication v: SIP/2.0/UDP 10.11.12.7;rport=47865;branch=z9hG4bK08agtrXr3c94j;received=77.99.55.42 f: <sip:0000000000@10.11.12.7>;tag=mS3Q66862j8HK t: <sip:1234567890130@callcentric.com> i: 75dfa23c-49f2-4448-4cbd-5936885236fb CSeq: 9260033 INVITE l: 0 The username and password are correct, callcentric says: Looking at the INVITE 's "Proxy-Authorization" header we can see that the Username is correct as "1234567890130". However, the realm states "10.11.12.7" rather than "callcentric.com". If you would please ensure that the FreeSwitch configuration will utlize "callcentric.com" as the authorization realm. I looked at the wiki/google and tried the sip_auth_realm and fs_path to add the realm but no luck. O hope someone knows the magic option to set the realm. Thanks. Mario G

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