February 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

SRTP breaks my TLS session
by Emrah 02 Mar '16

02 Mar '16

Hello list, I thought I had solved this issue by reducing my codec list to a minimum, but it still persists, unfortunately. This was to reduce the TLS packet size. Anytime I enable SRTP on my phones, outgoing calls will randomly fail. The problem goes away when I disable SRTP. I only work over TLS. Incoming calls work reliably with or without SRTP. How do you suggest debugging this? I tried setting up a fresh instance of FS but the issue persists. Now. It should be noted that calls fail sensibly more often when I have more than one account registered on the same server with the same device. Any suggestion is welcome. Have you experienced this? I’m running FreeSWITCH Version 1.6.5 on Debian. My phone in this case is a Yealink SIP-T46G running firmware 28.80.0.95. E

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all-reg-options-ping and tls issue
by Seeliger, Adam 02 Mar '16

02 Mar '16

Hi all, I have a problem, when I enable TLS and register a phone using TLS on Port 5061. FreeSWITCH still tries to "ping" the phone using Port 5060 using UDP, which is ignored by the phone. Moments later FreeSWITCH deletes the registration, because "unregister-on-options-fail" is set to "true". I already figured out, that you can set "all-reg-options-ping" to "udp-only", but this would completely disable this feature for TLS. Is there any way to ping TLS registered using TLS? Thanks in advance - Adam

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Unable to compile mod_sndfile on CentOS 7
by Vladislav Ivanov 01 Mar '16

01 Mar '16

Hey guys, I'm unable to compile mod_sndfile even though I have installed all the devel packages. # yum install libsndfile-devel Package libsndfile-devel-1.0.25-10.el7.x86_64 already installed and latest version # make mod_sndfile make[1]: Entering directory `/opt/freeswitch' make libfreeswitch.la make[2]: Entering directory `/opt/freeswitch' make[2]: Leaving directory `/opt/freeswitch' make[1]: Leaving directory `/opt/freeswitch' make[1]: Entering directory `/opt/freeswitch/src/mod' make[2]: Entering directory `/opt/freeswitch/src/mod' making all mod_sndfile make[3]: Entering directory `/opt/freeswitch/src/mod/formats/mod_sndfile' Makefile:888: *** You must install libsndfile-dev to build mod_sndfile. Stop. make[3]: Leaving directory `/opt/freeswitch/src/mod/formats/mod_sndfile' make[2]: *** [mod_sndfile-all] Error 1 make[2]: Leaving directory `/opt/freeswitch/src/mod' make[1]: *** [mod_sndfile] Error 2 make[1]: Leaving directory `/opt/freeswitch/src/mod' make: *** [mod_sndfile] Error 2 Any tips? Best Regards.

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Debugging Perl code
by ik 01 Mar '16

01 Mar '16

Hello, I have Perl code that does new freeswitch::Session. When that happens, freeswitch crashes. It's an old freeswitch v1.4.26. I execute it from now using the cli either by perl <file.pl> or perlrun. How can I debug the cause of such crash? Thank you Ido

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segfaults because odbc mysql driver
by Юрий Насида 01 Mar '16

01 Mar '16

Hi guys, As far as I know there is one already known problems with FS segfaults that happen because of odbc mysql driver. I have latest recommended setup with debian 8.3 + FS 1.6. Somebody mentioned that version of unixODBC should be higher than 2.3 to solve this but... I have unixODBC 2.3.1 and it doesn't help. Everybody use pgsql ? :) Please advice if somebody use mysql with FS 1.6 and still happy. I also have backtrace file here - https://freeswitch.org/jira/browse/FS-8857 and will be glad any ideas. Thanks.

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originate and valet_park
by Gregor Nanger 29 Feb '16

29 Feb '16

Is it possible to originate call and after answer park it in valet? I tried something like: originate user/1000 &valet_park mylot 5555 Extension rings and after pickup call is disconnected. Please for hint how to correctly format command or how to achive such scenario in different way. Best regards, Gregor

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File not found
by Vishal Sharma 29 Feb '16

29 Feb '16

Hi, I am using FS 1.6.5, when FS tries to play a file (/doesnotexist/srv/sounds/automation/media-sec.mp3) which doesn't exist on file system, it takes 20 seconds for it to respond that file is not present and dead air is played during this time. On FS 1.4 and 1.2, it used to respond instantly. Am i missing some setting ... Regards, Vishal Sharma -- SuperReceptionist is now available on Android mobiles. Track your business on the go with call analytics, recordings, insights and more: Download the app here <https://play.google.com/store/apps/details?id=com.superreceptionist>

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Recording each leg of a conference call separately
by Delvin Varghese 29 Feb '16

29 Feb '16

Hi all, I am familiar with the record and record_session commands in freeswitch to initiate recordings, and also using ESL.. But is there a way to record ONLY the speech from one device at a time in separate recordings? The way i am envisioning is that if there are 9 different devices involved in a conference call, that at the end of it I would be able to access 9 recordings (each containing only the outbound voice from that device)… Is this possible? What are people’s thoughts? Delvin

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where mod_com_g729 ?
by Юрий Насида 29 Feb '16

29 Feb '16

Hi list I have installed FS 1.6.6 via apt-get method (all modules) and don't see mod_com_g729 anywhere. Should I install it from source ? Please advice. # apt-cache search freeswitch | grep 729 freeswitch-mod-g729 - mod_g729 for FreeSWITCH freeswitch-mod-g729-dbg - mod_g729 for FreeSWITCH (debug)

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Call dropping after 32 seconds
by Rutu Patel 29 Feb '16

29 Feb '16

Hello, Having issue of call dropping after 32 seconds, here are the details- x.x.x.174: opensips server x.x.x.166: freeswitch server x.x.x.3: another opensips server which is registered as gateway on above freeswitch server x.x.x.6: freeswitch server x.x.x.47: server through which the user is registered I am trying to call from xxxx9 to xxxxxxx29858 xxxxxxx00181 is caller-id name and caller-id number Call flow is like this: registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server) Outbound-proxy is set to x.x.x.174 in Gateway configuration. 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 2) Call hit the freeswitch server x.x.x.166 3) After '180 Ringing' and '183 Session Progress' packet sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the gateway x.x.x.3 But after 32 seconds call is dropped, Within 32 seconds audio is ok from both end so it should not be the RTP issue. Here I have attached the file with sip logs, you can observer from the file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped. What is wrong here? Any help would be appreciated here. Here is the file with sip logs -- Thanks, Rutu Patel <http://www.inextrix.com>

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