April 2019 - FreeSWITCH-users - FreeSWITCH Mailing Lists

how to connect FS to openSIPS
by Mundkowsky, Robert 25 Apr '19

25 Apr '19

I have openSIPS configured as a load balancer behind FreeSWITCH. I setup a sip profile for openSIPS in /etc/freeswitch/sip_profiles/internal/engagement.xml <include> <gateway name="engagement"> <param name="realm" value="XXXX"/> <param name="username" value="1000"/> <param name="from-user" value="1000"/> <param name="from-domain" value="freeswitch.internal.org:5060"/> <param name="password" value="1000"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="ping" value="25"/> </gateway> </include> Then setup a diaplan that should call openSIPS: ... <action application="conference_set_auto_outcall" data="{sip_from_display=${client_session_id}}sofia/gateway/engagement/${as_ext}${destination_number}"/> ... When call is made, I don't see any errors in the FreeSWITCH log and nothing is seen in the openSIPS log. I suspect I have an error in the sip profile. ________________________________ 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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Caller ID NAme switches to "mod_sofia"
by Sean Devoy 23 Apr '19

23 Apr '19

Hi, My customer has Aastra 67xx phones. I have two problems relating to caller id. 1 - The setup on the phone appears to take president over my setup for that extension in the XML (for caller id and caller name). 2 - While the correct number and extension appear from interoffice phones, as soon as it gets answered the display changes to "mod_sofia". Any advise will be greatly appreciated. Thanks, Sean

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dial extension after call is setup
by Joli Martinez 21 Apr '19

21 Apr '19

I need to dial an extension within an IVR. So my freeswitch application will test PBX extensions. The application will dial an IVR (Could be running any PBX vendor), when the PBX picks up it will dial an extension and run several tests including testing two way audio, DTMF and other tests. I got my test scenarios working with a normal numbers, but now I need to extend it to dial extensions within an IVR. Most of this is written in Lua I have two problems with this setup. 1. How can I dial an extension after the call is setup. Tried the "send_dtmf" but it didnt work or maybe I just dont know how to use it. I have done "send_dtmf <extension>" and nothing happens. What would be the best way to dial an extension after the call has been answered? 2. Once I get this part working. How can I wait for the extension to answer before starting to execute my test script. I was thinking of doing "wait_for_silence 200 15 10 5000" then execute the rest of my script. What do you think? Is there a better way?

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[Best way to get calls per client]
by Bilal Abbasi 19 Apr '19

19 Apr '19

Hi Users, I am just building a platform, where i need to show "live connected" calls per account/client. I know that it can be done via events, so to fire event at start of call and at the end of the call to get the count of calls for each client. But i wanted to know if somebody has more better idea, or if somebody has done similar solution in a better way. BR Abbasi

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Anyone have any actual technical details of the new robocall verification stuff being turned up by various large providers?
by Nathan Neulinger 18 Apr '19

18 Apr '19

All of the articles I've seen are just the dumbed down public info with an occasional technical buzzword thrown in. https://tech.slashdot.org/story/19/04/17/1659243/t-mobile-comcast-turn-on-c… -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 341-6679 System Administrator - Architect (573) 612-1412 System and Desktop Infrastructure Team Manager

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Too many stacked extensions fixes
by Grant Bagdasarian 18 Apr '19

18 Apr '19

Hi, When will a new minor version of Freeswitch be released (1.8.6)? And will this include the following changes? * https://freeswitch.org/stash/projects/FS/repos/freeswitch/commits/c8e9eeaec… * https://freeswitch.org/stash/projects/FS/repos/freeswitch/commits/1a5ce0ebe… * https://freeswitch.org/stash/projects/FS/repos/freeswitch/commits/d86c71569… * https://freeswitch.org/stash/projects/FS/repos/freeswitch/commits/c40af7732… I want to get these patches in production asap, since we cannot reproduce the stack errors on our test environment. Thanks and regards, Grant

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Mod native file supports file_string ?
by sagar malam 18 Apr '19

18 Apr '19

Does mod native file support file_string ? I am not able to get it work.Any help would be appreciated. -- Thanks, Sagar

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freeswitch group-call/call-center behavior to unregistered users
by Leo Wolf 18 Apr '19

18 Apr '19

Hi, I am trying using freeswitch as sip servers for mobile client. And I have a question. If we call a group call to freeswitch, say group 100 with user 1001,1002,1003. And at this time, 1001 & 1002 is online but 1003 is not online. I want a behavior that call to group 100, 1001 & 1002 will ring, and after I notify user 1003 registering to freeswitch, 1003 will ring too and can pickup the group call just like 1003 is online while you making the group call. Is that possible? Or is there any way to achieve it? using call-center? Many Thanks, Leo

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Can't connect to 800 or toll calls
by Sean Devoy 18 Apr '19

18 Apr '19

Hi, I have recently created a new server with the latest repo (1.8) Most things are working great. Extension to Extension works, even across NAT on both ends. I can call out to local numbers like my cell phone. However, when I try to call 800 numbers or calls outside my local area codes it fails. The logs look identical to the old server. The call goes through the same gateway. I never get any audio back and 15 seconds later the log returns "no answer" and the call fails. If it matters the gateway is through Vitelity. I have opened a ticket there but no response yet. Log excerpt: [DEBUG] switch_core_state_machine.c:40 sofia/external/13522298979 Standard INIT [DEBUG] switch_core_state_machine.c:48 (sofia/external/13522298979) State Change CS_INIT -> CS_ROUTING [DEBUG] switch_core_state_machine.c:627 (sofia/external/13522298979) State INIT going to sleep [DEBUG] switch_core_state_machine.c:584 (sofia/external/13522298979) Running State Change CS_ROUTING (Cur 2 Tot 2) [DEBUG] switch_core_state_machine.c:643 (sofia/external/13522298979) State ROUTING [DEBUG] mod_sofia.c:154 sofia/external/13522298979 SOFIA ROUTING [DEBUG] switch_ivr_originate.c:67 (sofia/external/13522298979) State Change CS_ROUTING -> CS_CONSUME_MEDIA [DEBUG] switch_core_state_machine.c:643 (sofia/external/13522298979) State ROUTING going to sleep [DEBUG] switch_core_state_machine.c:584 (sofia/external/13522298979) Running State Change CS_CONSUME_MEDIA (Cur 2 Tot 2) [DEBUG] sofia.c:7291 Channel sofia/external/13522298979 entering state [calling][0] [DEBUG] switch_core_state_machine.c:662 (sofia/external/13522298979) State CONSUME_MEDIA [DEBUG] switch_core_state_machine.c:662 (sofia/external/13522298979) State CONSUME_MEDIA going to sleep [NOTICE] switch_ivr_originate.c:3705 Hangup sofia/external/13522298979 [CS_CONSUME_MEDIA] [NO_ANSWER] [DEBUG] switch_core_state_machine.c:584 (sofia/external/13522298979) Running State Change CS_HANGUP (Cur 2 Tot 2) [DEBUG] switch_core_state_machine.c:847 (sofia/external/13522298979) Callstate Change DOWN -> HANGUP [DEBUG] switch_core_state_machine.c:849 (sofia/external/13522298979) State HANGUP [DEBUG] mod_sofia.c:449 Channel sofia/external/13522298979 hanging up, cause: NO_ANSWER [DEBUG] mod_sofia.c:514 Sending CANCEL to sofia/external/13522298979 [DEBUG] switch_core_state_machine.c:60 sofia/external/13522298979 Standard HANGUP, cause: NO_ANSWER [DEBUG] switch_core_state_machine.c:849 (sofia/external/13522298979) State HANGUP going to sleep [DEBUG] switch_core_state_machine.c:619 (sofia/external/13522298979) State Change CS_HANGUP -> CS_REPORTING [DEBUG] switch_core_state_machine.c:584 (sofia/external/13522298979) Running State Change CS_REPORTING (Cur 2 Tot 2) [INFO] mod_dptools.c:3518 Originate Failed. Cause: NO_ANSWER Thanks, Sean

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WebRTC (jsSIP) + Kamailio + Sofia -
by Marcin Kowalczyk 17 Apr '19

17 Apr '19

Hi, For some time I'm struggling with missing audio between jsSIP ( https://tryit.jssip.net/ ) and FreeSwitch (Sofia). On Kamailio side I'm doing WSS <-> SIP/udp gateway and works like a charm together with RTPEngine. I've deciced to remove RTPEngine from setup and leave only kamailio in front of FreeSwitch as WSS-SIP gateway. I'm working on latest master (8df9ac43) Sofia profile settings: <param name="stun-enabled" value="true"/> <param name="apply-candidate-acl" value="localnet.auto"/> <param name="ext-rtp-ip" value="54.154.222.32"/> <param name="rtp-ip" value="$${local_ip_v4}"/> I originate call from Freeswitch to Kamailio with media_webrtc=true,rtcp_mux=1 On browser side I can see following Invite INVITE sip:fc6inuoq@p1ir3vbttphp.invalid;transport=ws SIP/2.0 Via: SIP/2.0/WS qa1-rtc.example.com:443 ;branch=z9hG4bKd523.0e53395c81c97731505d9a419e74277f.0 Max-Forwards: 69 From: <sip:9005551212@rd9liaa7lu.example.com>;tag=DrZQygUQ7tg3S To: <sip:calling-1@rd9liaa7lu.example.com> Call-ID: 6c33377c-6142-11e9-9a3a-b966c9f42136 CSeq: 3209884 INVITE Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 969 Remote-Party-ID: <sip:9005551212@10.23.253.61 >;party=calling;screen=yes;privacy=off P-FS-IP: 10.23.253.61:3010 P-Asserted-Identity: <sip:9005551212@rd9liaa7lu.example.com> Contact: <sip:btpsh-5cb74b85-302b-e@qa1-rtc.example.com:443;transport=ws> v=0 o=FreeSWITCH 1555495382 1555495383 IN IP4 54.154.222.32 s=FreeSWITCH c=IN IP4 54.154.222.32 t=0 0 a=msid-semantic: WMS x4bcMHQEW0SfVL62y1GNNQP1dDUnlcJI m=audio 31970 RTP/SAVPF 102 a=rtpmap:102 opus/48000/2 a=fmtp:102 useinbandfec=1; maxaveragebitrate=14400; maxplaybackrate=8000; ptime=20; minptime=10; maxptime=40 a=fingerprint:sha-256 8F:B3:EB:3D:C8:8C:9E:45:EC:EF:F7:E4:9F:DB:CC:DC:BC:E9:B7:08:20:DD:51:23:A0:07:C9:39:BF:2B:CF:9B a=setup:actpass a=rtcp-mux a=rtcp:31970 IN IP4 54.154.222.32 a=ssrc:3636203136 cname:6LLIjrc5iUa3v6VU a=ssrc:3636203136 msid:x4bcMHQEW0SfVL62y1GNNQP1dDUnlcJI a0 a=ssrc:3636203136 mslabel:x4bcMHQEW0SfVL62y1GNNQP1dDUnlcJI a=ssrc:3636203136 label:x4bcMHQEW0SfVL62y1GNNQP1dDUnlcJIa0 a=ice-ufrag:YsxB8dtcGdpSgZqc a=ice-pwd:B1KKjNGsyGlUD7Esp2To5dqC a=candidate:3449784969 1 udp 659136 54.154.222.32 31970 typ host generation 0 a=candidate:3449784969 2 udp 659136 54.154.222.32 31970 typ host generation 0 a=ptime:20 But jssip fails call with tryit-jssip.js:8 JsSIP:Transport sending message:SIP/2.0 408 RTP Timeout Via: SIP/2.0/WS qa1-rtc.crazy-calls.com:443 ;branch=z9hG4bKd523.0e53395c81c97731505d9a419e74277f.0 To: <sip:calling-1@rd9liaa7lu.crazycall.com>;tag=02uof1d2c1 From: <sip:9005551212@rd9liaa7lu.crazycall.com>;tag=DrZQygUQ7tg3S Call-ID: 6c33377c-6142-11e9-9a3a-b966c9f42136 CSeq: 3209884 INVITE Supported: timer,ice,replaces,outbound Content-Length: 0 +0ms while I can see UDP traffic hitting server from both IP's (running with VPN connected) 18:56:05.897282 IP ZZ.VV.199.228.51931 > 10.23.253.61.31970: UDP, length 108 18:56:05.997047 IP XX.YY.99.114.62790 > 10.23.253.61.31970: UDP, length 108 18:56:06.095205 IP ZZ.VV.199.228.51931 > 10.23.253.61.31970: UDP, length 108 18:56:06.229668 IP XX.YY.99.114.62790 > 10.23.253.61.31970: UDP, length 108 18:56:06.322814 IP ZZ.VV.199.228.51931 > 10.23.253.61.31970: UDP, length 108 18:56:06.423745 IP XX.YY.99.114.62790 > 10.23.253.61.31970: UDP, length 108 18:56:06.544214 IP ZZ.VV.199.228.51931 > 10.23.253.61.31970: UDP, length 108 When I put RTPEngine in a middle works perrfectly. Only difference I can see in SDP that in such case there is NO duplicated line with same candidate in SDP a=candidate:3449784969 1 udp 659136 54.154.222.32 31970 typ host generation 0 a=candidate:3449784969 2 udp 659136 54.154.222.32 31970 typ host generation 0 I've tried to put instead of fixed ip into ext-rtp-ip stun:stun:google.... but did not change anything. Do you have idea what might be wrong here?

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