March 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

opensips+freeswitch on different amazon ec2 servers: 488 error
by Стас Тельнов 14 Mar '16

14 Mar '16

Hello. I have had difficulties at control of the opensips and freeswitch connective. The task is: to organize voip-conference between mobile application users through our opensips server or between mobile application users with addition of a call to the city line. Sip and freeswitch servers are on amazon ec2 each. Physical servers are different. Firstly, I test the opportunity to call to freeswitch and to receive the answer. From the part of opensips everything works, as far as I can judge, correctly. A problem appears at a call with use of freeswitch. In particular for tests I call on number 5000 through Zoiper . In reply I receive the mistake: 488 INCOMPATIBLE_DESTINATION Here is the text of freeswitch tracking data: EXECUTE sofia/external/8@SIP_DOMAIN set(rtp_secure_media=true) 2016-03-14 02:54:02.731306 [DEBUG] mod_dptools.c:1498 SET sofia/external/8@SIP_DOMAIN [rtp_secure_media]=[true] EXECUTE sofia/external/8@SIP_DOMAIN hash(insert/172.31.22.124-spymap/8/89574eb2-e9b1-11e5-97c9-1f13b1bd8007) EXECUTE sofia/external/8@SIP_DOMAINhash (insert/172.31.22.124-last_dial/8/5000) EXECUTE sofia/external/8@SIP_DOMAIN hash(insert/172.31.22.124-last_dial/global/89574eb2-e9b1-11e5-97c9-1f13b1bd8007) EXECUTE sofia/external/8@SIP_DOMAIN export(RFC2822_DATE=Mon, 14 Mar 2016 02:54:02 -0400) 2016-03-14 02:54:02.731306 [DEBUG] switch_channel.c:1293 EXPORT (export_vars) [RFC2822_DATE]=[Mon, 14 Mar 2016 02:54:02 -0400] EXECUTE sofia/external/8@SIP_DOMAIN answer() 2016-03-14 02:54:02.731306 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [GSM:3:8000:20:13200:1]/[G722:9:8000:20:64000:1] 2016-03-14 02:54:02.731306 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [GSM:3:8000:20:13200:1]/[PCMU:0:8000:20:64000:1] 2016-03-14 02:54:02.731306 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [GSM:3:8000:20:13200:1]/[PCMA:8:8000:20:64000:1] 2016-03-14 02:54:02.731306 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [GSM:3:8000:20:13200:1]/[GSM:3:8000:20:13200:1] 2016-03-14 02:54:02.731306 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [GSM:3:8000:20:13200:1] ++++ is saved as a match 2016-03-14 02:54:02.731306 [WARNING] switch_core_media.c:4234 Crypto not negotiated but required. 2016-03-14 02:54:02.731306 [DEBUG] switch_core_media.c:4492 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2016-03-14 02:54:02.731306 [NOTICE] switch_channel.c:3806 Hangup sofia/external/8@SIP_DOMAIN [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] 2016-03-14 02:54:02.731306 [DEBUG] switch_core_session.c:2796 sofia/external/8@SIP_DOMAIN skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2016-03-14 02:54:02.731306 [DEBUG] switch_core_state_machine.c:539 (sofia/external/8@SIP_DOMAIN) State EXECUTE going to sleep 2016-03-14 02:54:02.731306 [DEBUG] switch_core_state_machine.c:473 (sofia/external/8@SIP_DOMAIN) Running State Change CS_HANGUP 2016-03-14 02:54:02.731306 [DEBUG] switch_core_state_machine.c:739 (sofia/external/8@SIP_DOMAIN) Callstate Change RINGING -> HANGUP 2016-03-14 02:54:02.731306 [DEBUG] switch_core_state_machine.c:741 (sofia/external/8@SIP_DOMAIN) State HANGUP 2016-03-14 02:54:02.731306 [DEBUG] mod_sofia.c:431 Channel sofia/external/8@SIP_DOMAIN hanging up, cause: INCOMPATIBLE_DESTINATION 2016-03-14 02:54:02.731306 [DEBUG] mod_sofia.c:568 Responding to INVITE with: 488 2016-03-14 02:54:02.731306 [DEBUG] switch_core_state_machine.c:60 sofia/external/8@SIP_DOMAIN Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2016-03-14 02:54:02.731306 [DEBUG] switch_core_state_machine.c:741 (sofia/external/8@SIP_DOMAIN) State HANGUP going to sleep 2016-03-14 02:54:02.731306 [DEBUG] switch_core_state_machine.c:508 (sofia/external/8@SIP_DOMAIN) State Change CS_HANGUP -> CS_REPORTING 2016-03-14 02:54:02.731306 [DEBUG] switch_core_state_machine.c:473 (sofia/external/8@SIP_DOMAIN) Running State Change CS_REPORTING 2016-03-14 02:54:02.731306 [DEBUG] switch_core_state_machine.c:827 (sofia/external/8@SIP_DOMAIN) State REPORTING 2016-03-14 02:54:02.731306 [DEBUG] switch_core_state_machine.c:104 sofia/external/8@SIP_DOMAIN Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2016-03-14 02:54:02.731306 [DEBUG] switch_core_state_machine.c:827 (sofia/external/8@SIP_DOMAIN) State REPORTING going to sleep 2016-03-14 02:54:02.731306 [DEBUG] switch_core_state_machine.c:499 (sofia/external/8@SIP_DOMAIN) State Change CS_REPORTING -> CS_DESTROY 2016-03-14 02:54:02.731306 [DEBUG] switch_core_session.c:1646 Session 1 (sofia/external/8@SIP_DOMAIN) Locked, Waiting on external entities 2016-03-14 02:54:02.731306 [NOTICE] switch_core_session.c:1664 Session 1 (sofia/external/8@SIP_DOMAIN) Ended 2016-03-14 02:54:02.731306 [NOTICE] switch_core_session.c:1668 Close Channel sofia/external/8@SIP_DOMAIN [CS_DESTROY] 2016-03-14 02:54:02.731306 [DEBUG] switch_core_state_machine.c:630 (sofia/external/8@SIP_DOMAIN) Running State Change CS_DESTROY 2016-03-14 02:54:02.731306 [DEBUG] switch_core_state_machine.c:640 (sofia/external/8@SIP_DOMAIN) State DESTROY 2016-03-14 02:54:02.731306 [DEBUG] mod_sofia.c:341 sofia/external/8@SIP_DOMAIN SOFIA DESTROY 2016-03-14 02:54:02.731306 [DEBUG] switch_core_state_machine.c:111 sofia/external/8@SIP_DOMAIN Standard DESTROY 2016-03-14 02:54:02.731306 [DEBUG] switch_core_state_machine.c:640 (sofia/external/8@SIP_DOMAIN) State DESTROY going to sleep Error in my_thread_global_end(): 1 threads didn't exit In such a case I use the following configuration of freeswitch: vars.xmp: http://pastebin.com/DijwsQ7j modules.conf.xml: http://pastebin.com/PjeNgVkh At the same time I use the following configuration of opensips: http://pastebin.com/49XpcGBj Whats i doing incorrect?

3 6

Connecting FreeSwitch to Genesis
by Shlomi Schwartz 14 Mar '16

14 Mar '16

Sorry if this is a stupid question: I've connected to FreeSwicth using mod_verto, now I would like to route the call to a Genesys via SIP. Is it the right approach? is there any documentation for it? The purpose is to be able to integrate my webRTC solution to many contact center platforms. -- This message may contain confidential and/or privileged information. If you are not the addressee or authorized to receive this on behalf of the addressee you must not use, copy, disclose or take action based on this message or any information herein. If you have received this message in error, please advise the sender immediately by reply email and delete this message. Thank you.

3 5

Call failed with mod_verto
by bhavik patel 14 Mar '16

14 Mar '16

Hi All, I have Freeswitch-1.6.5 running on Debian-8.2. I want to use mod_verto for webrtc and followed steps mentioned at https://freeswitch.org/confluence/display/FREESWITCH/mod_verto. But calls are not working. Outbound call fails with NORMAL_TEMPORARY_FAILURE. When make inbound call it rings at destination and drops as soon as it answer. Here i am attaching logs for both type of calls. Let me know if you require anything else from my end. Thanks

3 9

FS behind NAT
by Tao Zhang 14 Mar '16

14 Mar '16

Hi all, I have been tracking down why my FS verto video conf demo setup not working. Now I believe the reason is my FS server is behind a NAT. My previous posts may be misleading now, so I am creating a new thread. Some evidence I saw: 1) In the SDP dialog, I saw my FS server's internal IP got used, rather than its public IP, for example Dialog 6c62f631-469c-a2f0-e484-6e9cfcda7eedANSWER SDP v=0 o=FreeSWITCH 1444254094 1444254095 IN IP4 10.240.15.244 s=FreeSWITCH c=IN IP4 10.240.15.244 <============= t=0 0 2) Also, in wireshark , I saw my client is trying to send STUN packets to the FS server's internal IP, 450 18.252726000 192.168.1.7 10.240.15.244 STUN 158 Binding Request user: mlvLvmDzCphyQYz5:N3MZoB99aGiAZotr So now my questions are Will the verto video conf demo work when the FS server is behind NAT? If so, could someone help me with the necessary config changes and etc.? Thanks so much! Tao

3 6

Ask about the integration between xmpp and freeswitch
by atai＠unseen.is 14 Mar '16

14 Mar '16

Hello all, We tried to setup the FS and can call using SIP protocol, using different apps for ex. xLite, it works very well! And we tried to setup the FS with xmpp. If we have many xmpp users, and we want users to use SIP to call, so we tried to install the dingaling module, configured them, but when we create the xmpp users, there is no notification from xmpp to FS. Is there any full instruction of configure XMPP in FS and example? Thank you very much. Tai Duc.

1 0

mod_shout build problem
by Mi Ke 13 Mar '16

13 Mar '16

2 3

Freeswitch IVR: call drop on transfer to busy extensions
by Francesco Piccinin 13 Mar '16

13 Mar '16

Hi all, I'm experiencing an issue with IVR on production Freeswitch cluster. I'm going to explain the problem as clear as possible just to understand if anyone already get it and hopefully fix it. Phone A makes a call that match an IVR services: call is transfered to context services in xml services. Call is then answered, audio menù is prompted: phone A makes a selection (ex dtmf 2) and the call is trasfered to context default xml default in order to ring phone B. If phone B is in busy state the call is dropped. On cli logs you can observe this behaviour: Mod Sofia first hangup the channel with a USER BUSY cause and then it sends a BYE to Phone A (or to a gateway in case of external calls) including Q850 cause=17 user busy. Phone A or Gateway drop the call without busy tone. In this case I'm expecting a SIP 486 (user busy message) instead of a bye, is that correct? I'm using the app "limit" in order to count active calls on extensions and store them in system db (shared between freeswitch servers). Hope you can help me solving the problem. Thanks Regards Francesco

2 1

how to reply here in list to existing message and not open new thread?
by Dmitry Saratsky 13 Mar '16

13 Mar '16

test hope reply is working now... sorry if not...

1 0

how to reply here in list to existing message and not open new thread?
by Dmitry Saratsky 13 Mar '16

13 Mar '16

Hi, Probably it is most stupid question... But It's my 3rd time I'm trying to reply to existing message, but it is new thread opened here in lists... Please help...

1 0

Empty SDP in 200 OK on received re-invite while faxing with T.38
by Dmitry Saratsky 13 Mar '16

13 Mar '16

Yesterday I have updated to latest git to see if issue is still exists. Will update later

1 0