January 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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

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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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Skypopen Debian Jessie install issues
by Jason Bedward 20 Feb '16

20 Feb '16

Hi, I have tried to install this on Debian Jessie but had some issues with dependencies which are not matching what the install guide says. When I got to the CLI and load the module I get the error below. I have libjpeg62-turbo & libjpeg62-turbo-dev installed which I think it the current version for Debian. Please can someone help guide me with this version of Debian. Thanks +OK log level 9 [9] +OK console log level set to DEBUG freeswitch@192.168.2.233@internal> load mod_skypopen +OK Reloading XML -ERR [module load file routine returned an error] 2015-12-15 12:01:55.996236 [INFO] mod_enum.c:880 ENUM Reloaded 2015-12-15 12:01:55.996236 [INFO] switch_time.c:1415 Timezone reloaded 1781 definitions 2015-12-15 12:01:55.996236 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_skypopen.so **/usr/local/freeswitch/mod/mod_skypopen.so: undefined symbol: jpeg_resync_to_restart**

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extra header account code is not written to cdr if cancel is received a few ms after invite
by Panagiotis Skoulikaritis 16 Feb '16

16 Feb '16

Dear all I have an implementation FreeSWITCH as a sort of SBC, it is used to send the calls to the terminating carriers and do topology hiding, nothing fancy. Also I gather cdrs from the FreeSWITCH. In order to distinguish each customer on the FS cdrs I send an extra header containing the accountcode. I have noticed that if the call is canceled immediately on the same sec, the account code is not written on the cdr. To be more precise the cancel is send a few milliseconds after it has received the invite, and before the FreeSWITCH has sent the call to the terminating carrier (I'm using Homer Sipcapture to capture all the traces and I don't see an attempt being made at the terminating carrier) also I don't see a b-leg cdr. FreeSWITCH is writing both a-leg and b-leg cdrs in csv format. The dialplan that I use is simple <extension name="carrier-1"> <condition field="network_addr" expression="^(^xx\.xx\.xx\.xx|^yy\.yy\.yy\.yy)$"/> <condition field="destination_number" expression="^2#(\d+)$"> <action application="set" data="call_direction=outgoing"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="sip_cid_in_1xx=false"/> <action application="set" data="inherit_codec=true"/> <action application="set" data="accountcode=${sip_h_X-ACCOUNTCODE}"/> <action application="unset" data="sip_h_X-ACCOUNTCODE"/> <action application="set" data="execute_on_answer=sched_hangup +21600 alloted_timeout"/> <action application="bridge" data="sofia/external/4567$1@zz.zz.zz.zz:5070"/> </condition> </extension> any idea how I can make sure that the account code will always be written ? Best Regards Panagiotis

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Mod_callcenter ODBC vs File
by Jai Rangi 11 Feb '16

11 Feb '16

Hello, FS version: FreeSWITCH Version 1.4.18+git~20150312T185523Z~4eed221b69~64bit (git 4eed221 2015-03-12 18:55:23Z 64bit) I found a very strange behavior on mod_call center when used with odbc. This was easy to reproduce when the agents are busy and new calls come in. New member goes in abandoned state immediately and stay in same state forever. Even if previous call is complete and Agent is in Waiting state. Infact some times new members go in abandoned state immediately even if the agents are available. Everything works perfect once I changed the mod_callcenter.conf.xml config to file only mode, new calls go in Waiting state and call is connected to agent as soon as the agent is free. Any idea why the new member goes in abandoned state randomly in place of being in waiting state? Can it be due to too small timeout in call center module as it query the DB every time? Can that be changed? Anyone else had smiler problem, any experience or any recommendations will be much appreciated. Thank you, -Jai

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mod_esf with polycom
by Peter Hartmann 05 Feb '16

05 Feb '16

Hey, how do you use mod_esf with Polycom? The source suggests that it's supported, but the arguments in the documentation seem specific to Linksys. From using the feature directly from a handset, I imagine we have to pass multicast-address, multicast-port and group-number. I've tried every combination and permutation with no luck. The reason I'm doing this is because if the emergency group number is used, the announcements will be at full volume regardless of the hands-free volume setting. And also I'm looking to do this in FS because I'm bridging in another PA system also. Thanks, Peter Hartmann Hartmann Computer Consulting http://blog.hartmanncomputer.com (212)203-8870 If I can't explain it to you in plain language, that means I don't understand it.

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loopback/XYZ doesn't contain variables from channel
by Michael Nielsen 04 Feb '16

04 Feb '16

I've got directory with subscribers containing a variable called "country_code". This variable can be used in regular dial plans and works perfectly. However, I've got a conference call dial plan where I use conference_set_auto_outcall. I'mm calling loopback/XYZ to get my other dial plans into play. Everything works, except my subscriber variable "country_code" doesn't get recognised after the loopback/. How can I fix this kind of issue?

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mod_potaudio eror on opening device
by Pete Kay 03 Feb '16

03 Feb '16

Hi <freeswitch(a)vps57327.vps.ovh.ca> I am not able to get pa play to work. could someone please kindly help me out? I am getting the following ERROR messages: <freeswitch(a)vps57327.vps.ovh.ca> freeswitch(a)vps57327.vps.ovh.ca> pa play /tmp/running.wav Expression 'SetApproximateSampleRate( pcm, hwParams, sr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1293 Expression 'PaAlsaStreamComponent_InitialConfigure( &self->capture, inParams, self->primeBuffers, hwParamsCapture, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1869 Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1993 2016-01-24 07:02:32.591002 [ERR] mod_portaudio.c:2428 Error opening audio device retrying Expression 'SetApproximateSampleRate( pcm, hwParams, sr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1293 Expression 'PaAlsaStreamComponent_InitialConfigure( &self->capture, inParams, self->primeBuffers, hwParamsCapture, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1869 Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1993 Failed to engage audio device 2016-01-24 07:02:34.090957 [ERR] mod_portaudio.c:2435 Can't open audio device freeswitch(a)vps57327.vps.ovh.ca> pa outdev #4 outdev set to 4 freeswitch(a)vps57327.vps.ovh.ca> pa indev #5 indev set to 5 freeswitch(a)vps57327.vps.ovh.ca> pa play /tmp/running.wav Expression 'SetApproximateSampleRate( pcm, hwParams, sr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1293 Expression 'PaAlsaStreamComponent_InitialConfigure( &self->playback, outParams, self->primeBuffers, hwParamsPlayback, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1872 Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1993 2016-01-24 07:02:51.991013 [ERR] mod_portaudio.c:2428 Error opening audio device retrying Expression 'SetApproximateSampleRate( pcm, hwParams, sr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1293 Expression 'PaAlsaStreamComponent_InitialConfigure( &self->playback, outParams, self->primeBuffers, hwParamsPlayback, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1872 Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1993 Failed to engage audio device 2016-01-24 07:02:53.491018 [ERR] mod_portaudio.c:2435 Can't open audio device freeswitch(a)vps57327.vps.ovh.ca>

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Freeswitch as Registrar for SIP Trunk
by Marc S 03 Feb '16

03 Feb '16

Hello, i'm discovering FS. I hav read a lot about users and gateways. I would like to FS act as registrar for authenticated SIP trunking. - Customers IPBX would register with login/password to Freeswitch. - Incoming call would be routed to these SIP trunks in dialplan XML. directory/users does not seem to be the solution because in dialplan, destination DID can't be defined, only user id : <action application="bridge" data="user/myuserid"/> gateway seems to be designed for SIP trunking to remote SIP gateway, not for FS to act as registrar. Is it possible to FS to act as authenticated SIP trunking registrar ? Thanks a lot, Marc

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FreeSWITCH vs Asterisk on Reddit...
by Brian West 03 Feb '16

03 Feb '16

Interesting thread... https://www.reddit.com/r/VOIP/comments/3wy9h8/freeswitch_vs_asterisk/ Everyone should check it out. Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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