March 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

segfaults from mod_easyroute.
by Юрий Насида 16 Mar '16

16 Mar '16

Hi list, I would like to ask if somebody else had segfaults from mod_easyroute. I have JIRA ticket opened but really need additional advice to solve this issue because I am not C developer. https://freeswitch.org/jira/browse/FS-8937 At this moment I just got idea to re-write mod_easyroute using lua. Any advice - welcome. Thanks

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Spanish Say support in freeswitch
by Ashwin Rath 16 Mar '16

16 Mar '16

I am trying to build in Spanish say support . However from what i see there seem to be be a few issues. I first downloaded the freeswitch-sounds-es-mx-maria-44100.tar.gz file and unpacked it at the appropriate location. Then i got mod_say_es to load anf modified the XMLs in the lang folder to load the phrases / IVR sounds. I then tried to use Say in a lua script but i ran into lot of issues like missing file names . Examples being FS expecting "thousands.wav" but folder having "thousand.wav" instead. Is Spanish officially supported in FS ? if yes is there any known issue ? -- Ashwin Kumar Rath

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FusionPBX - connection issues
by Shlomi Schwartz 16 Mar '16

16 Mar '16

Hi All, I have installed FusionPBX, and when going to "active calls" menu I get *Connection to Event Socket failed.* I'm Installing FS and FusionPBX on a digitalocean virtual server Any clue? -- Shlomi Schwartz R&D Technical Leader T: +972-74-700-4511 <http://www.linkedin.com/company/164748> <http://twitter.com/liveperson> <http://www.facebook.com/LivePersonInc> We Create Meaningful Connections <http://info.liveperson.com/201602/liz-welch/liveperson-robert-locascio-on-d…> -- 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.

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Default Freeswitch configuration - Cannot call internal extensions
by Varsha Agarwal 16 Mar '16

16 Mar '16

Hi, I just installed Freeswitch on a CentOS 7.0 system and registered a phone on 1001. When I dial another extension (ex. 5000 for IVR), I see this in the logs freeswitch@leccnha2> 2016-03-15 12:55:41.216586 [NOTICE] switch_channel.c:1101 New Channel sofia/external/1001(a)172.18.19.71 [e56f4f6a-eae7-11e5-b3e9-335237ee0f3f] 2016-03-15 12:55:41.216586 [INFO] mod_dialplan_xml.c:637 Processing 1001 <1001>->5000 in context public 2016-03-15 12:55:41.216586 [NOTICE] switch_core_state_machine.c:315 sofia/external/1001(a)172.18.19.71 has executed the last dialplan instruction, hanging up. Why is it using "public context" for internal extensions? How can I make it use default? Thanks, Varsha

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SSL implementation in FreeSWITCH
by Emrah 16 Mar '16

16 Mar '16

Hi all, I’m writing to document where I’m at with my issues with FreeSWITCH and SSL / TLS and share my conclusions so far. I am hoping that this can give lieu to some further testing in different environments, and a proper fix if a bug is indeed confirmed. First, I am running FreeSWITCH 1.6.6 on a Debian 8. Vars.xml shows sip_tls_version=tlsv1,tlsv1.1,tlsv1.2. What I’ve observed is that in a sequence where client sens an invite to FS; FS responds with 407 proxy authorization required; client sends ack; Client sends the invite with the digest auth. The last packet can easily exceed the max segment size of a TCP segment, typically if the SDP advertises a bunch of codecs, or if the client uses SRTP and the SAVP contains many crypto suites. Now, when this occurs, the packets should be sent fragmented so they can fit in the MTU. It is then up to the receiving end to reassemble the segments and feed the complete packet to the application layer. What I’ve noticed is that a packet that is too large is simply never received by FreeSWITCH. Since this is systematically the case with every software and hardware client I’ve used, I am drawn to think that the issue lies in the SSL implementation of FreeSWITCH. In the event that for some reason my network or server OS configuration may be behind this, I would appreciate if someone would be willing to share some SIP credentials that can let me test TLS and SRTP. If getting to the bottom of this is of interest to any of you, I’d obviously be keen on handing out a couple of accounts. I hope this message can be the starting point of a fruitful resolution process. Thank you if you’ve read this up to here. Now hit reply and give me your 2 cents! :) Best, Emrah

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Freeswitch IVR: call drop on transfer to busy extensions
by Francesco Piccinin 15 Mar '16

15 Mar '16

Hello Giovanni, thanks for you quick answer. Please find below patebin link you requested: XML Dialplan file's extract: http://pastebin.com/dmHAKWxT Internal call fscli logs: http://pastebin.com/KfnaYN11 External call fscli logs http://pastebin.com/2Khp1MaX Regarding dialplan files, I'm going to explain here call flow: Default.xml 1.call hits extension call_local, lua script is run and channel vasr cType=SRV, service_type=IVR 2. extension apps in xml services is execute Services.xml lua script is run to retrieve ivr name 3. on selection (2 in the logs) call is transfer to 8910 XML calls Calls.xml 4. extension: call_inside is execute and call is hangup after <action application="limit" data="db ${domain_name} MAX_${cCalled} ${maxcallsB} !USER_BUSY"/> As you can see from the logs mod sofia hangup with user busy and then send a bye to external or internal channel: 2016-03-14 11:37:40.266607 [DEBUG] mod_sofia.c:506 Channel sofia/external/ 00403737752(a)10.200.24.10 hanging up, cause: USER_BUSY 2016-03-14 11:37:40.266607 [DEBUG] mod_sofia.c:558 Sending BYE to sofia/external/00403737752(a)10.200.24.10 2016-03-14 11:38:54.166606 [DEBUG] mod_sofia.c:506 Channel sofia/internal/ 5532(a)insiel.fvgvoipcoll.it hanging up, cause: USER_BUSY 2016-03-14 11:38:54.166606 [DEBUG] mod_sofia.c:558 Sending BYE to sofia/internal/5532(a)insiel.fvgvoipcoll.it I tried to removed the limit app and the calls then hint limit_exceeded extension in calls.xml but behaviour is the same. It seems to me that continue_on_fail where passing over IVR is not considered. If I call directly (without IVR) extension 8910 (while it is busy) behaviour for both internal or external call is correct and busy tone is received. Looking forward for a clue :-) Thanks Regards Francesco

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SILK codec
by Oleg Stolyar 15 Mar '16

15 Mar '16

Hi guys, does FreeSWITCH still support SILK?

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freeswitch with tickless linux kernel
by François 15 Mar '16

15 Mar '16

Is it still recommended to use custom 1000HZ kernels for FS servers or can we go tickless now? Any difference if FS is installed in baremetal vs virtualized? Thanks, François.

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socket_resume + bridge + att_xfer
by Luiz Fernando Softov 15 Mar '16

15 Mar '16

Hi, i'm using dialplan socket (outbound), until now it's ok. But sometimes if my daemon crash (i'm working on it) the call still active. Searching i found socket_resume "If this variable is set to true, the dialplan will resume execution with the next action after the call to the socket application. This can be used for example to allow you to do something intelligent in the dialplan if your IVR application gets killed in an unclean way. If there is a bridge active when the disconnect happens, it is killed. To do this from your application after the socket is already connected, issue the resume command." So i make <action application="set" data="socket_resume=true"/> <action application="socket" data="127.0.0.1:3012 async full"/> <action application="hangup"/> Now, when i make a bridge (A -> B), and make a att_xfer (B -> C), when B press 0 (tree_way), A hangup. If i remove socket_resume, it's OK (A+B+C), but if i kill my daemon (simulate a crash), the call still ACTIVE. Is there another way to do that? There is something I'm missing?

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FS, SS7 and Dialogic IMG2020 in the UK
by Gavin Henry 15 Mar '16

15 Mar '16

Hi all, We're looking to get a pair of Dialogic IMG2020 for some BT Ofcom regulated SS7 interconnects: http://www.dialogic.com/en/products/gateways/img/img2020.aspx Any issues known with FS behind these? Thanks, Gavin. -- http://www.surevoip.co.uk

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