- FreeSWITCH-users - FreeSWITCH Mailing Lists

Is there a way to deal with "false ring-back than 503" carriers?
by Victor Chukalovskiy 03 Feb '16

03 Feb '16

Hello, A downstream carrier would accept INVITE, send 183 w/SDP, and almost instantaneously respond "503". This kills the call not letting re-route to the next carrier. Without calling any specific names, this seems to become more common. Unfortunately, often the offending carrier is a few steps down the call path and I can't reach them with a cluebat. Is there a way to make FS not fail the call and try the next route if using LCR as per below? Note that we are in bypass_media mode: <action application="set" data="bypass_media=true"/> <action application="bridge" data="lcr/lcr_profile/${destination_number}"/> Or to keep it simple, forget LCR for a moment, can we make it go to gateway_2 in the following example if gw1 returns "183" followed by "503"? <action application="set" data="bypass_media=true"/> <action application="bridge" data="sofia/gateway/gw_1/${destination_number}|sofia/gateway/gw_2/${destination_number}"/> Many thanks, -Victor

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How do you access caller profile fields in the XML dialplan?
by vfclists . 03 Feb '16

03 Feb '16

What is the syntax for accessing caller profile fields in the dialplan, for instance if you want to set a channel variable to a caller profile field? Is there also a way of performing some string manipulation and extraction on the variables in the XML dialplan? This is what I am trying to achieve. The CDR from a service provider contains the CLI from a gateway on the customers premises, but it doesn't show which extension on the customer's premises the call came from. What I need is to be able to obtain the extension of the sip device and combine that with the customer gateway's CLI so that it shows in the CDR record. eg if the gateway's CLI is 2340 and the extension of the caller is 1001, corresponding to the UserId on a sipura, I want the CLI passed to the service provider to be 23401001. When I check the XML cdr in Freeswitch I see an XML value which in XPath would be accessed as /callflow/caller_profile/caller_id_number. All the information is therefore passed onto Freeswitch for use in the call, and what I need is to be able to access it and change the effective_caller_id_number before bridging the call. -- Frank Church ======================= http://devblog.brahmancreations.com

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Best practices for determining if originated call is answered by human or voicemail system?
by Bob Hartwig 03 Feb '16

03 Feb '16

I have a client that needs to reliably detect if their outbound calls are answered by a human or voicemail system, so that they can take different actions based on that determination. I looked at the AVMD module documentation at https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd , and it seems to indicate that this simply detects a beep, i.e. it does not use talking / silence heuristics into account to determine if the call is answered by a human or machine. Am I correct about that? How is this intended to be used, do I assume that the call is answered by a human until this module sends a avmd%3A%3Abeep event to me? How do others use this module or other techniques to determine human / machine answer to outbound calls with Freeswitch? Thanks! Bob

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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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Storm of NOTIFY from FS
by SamyGo 03 Feb '16

03 Feb '16

Hi All, Ive few FS servers which solely handle Parking lots and their states behind Kamailio SBC. It has all been running very good for about two years now. Just as of this morning all of those FS servers have been sending storm of NOTIFY messages to Kamailio servers. This in turn consumes all CPU of FS server and fs cli becomes just irresponsive. I'm investigating whether Kamailio have blocked FS servers via pike module and hence FS servers go crazy on resending NOTIFY ever more aggressively. So far it doesnt look like it. I just want to ask under what circumstances that even after restarting service FS it would pick on all the states and resume sending to SBC. Is there a cache/DB somehwere which causes new FS process to continue with NOTIFY storm ? Ive FS version 1.5 deployed. Thanks in advance for any help. Regards, Sammy

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FreeSWITCH in virtual environments
by Dan Edwards 02 Feb '16

02 Feb '16

I am reviewing the Confluence Virtualization page and had some questions, in particular about VMWare. My company distributes some of its software as a VMWare image file and we were looking to distribute a new product using FS in the same manner. The products operate at a customer premise, on their VMWare infrastructure, not in a cloud environment. Since our customers already have VMWare, switching to a different VM infrastructure is going to hurt, so I am looking for options/alternatives. First, does anybody know if the virtual timing issues with VMWare have improved since this page was last updated in 2014? Is VMWare still not good enough? Is it possible to throw CPU & memory at this and make VMWare good enough, or is the virtual timing just not workable? On the virtualization page, there was a comment from 2010 that you might be happy with a High CPU Medium instance on AWS EC2. Certainly workload is a factor here, but I am trying to get my head around how big a machine to perform how small a workload. Is there a place where people talk about their experiences? Are there other VM platforms that might be acceptable? Any help or comment is appreciated. Thank you, Dan

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async, sendmsg, execute-app-name
by royj＠yandex.ru 02 Feb '16

02 Feb '16

Hi, all I am working on nodejs outbound socket application and faced that if send to FreeSWITCH message like: sendmsg execute-app-name: playback execute-app-arg: silence_stream://2000 call-command: execute when there is already no corresponding channel (a caller hung up by itself and from FreesSWITCH received CHANNEL_HANGUP_COMPLETE, text/disconnect-notice;linger) FreesSWITCH answers: Content-Type: command/reply Reply-Text: +OK Logic of library and application such, that we know that an application is finished, when received CHANNEL_EXECUTE_COMPLETE. There are cases when a caller hung up and exactly in this moment application not yet knows about it and sends message 'call-command: execute' and then there is javascript Promise in pending state, seems like forever. Is there any chance to understand in the answer that the channel does not exist?

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BLF Subscriptions sometimes don't send an initial Notify
by Stephen Thwaites 02 Feb '16

02 Feb '16

Hello, I have setup presence and in most cases it is working as expected. i.e. Subscription is sent to FS, FS returns Accepted then immediately FS sends the notify to the phone, thereafter all Notifies for ringing, pickup and hangup. Great. However in some cases a subscribe to an extension does subscribe, FS sends the accepted response but a Notify is not sent out at that point. However if I call the extension the Notify works perfectly. Any ideas of what could cause the initial Notify not to be sent after the Acceptance 202? Any help would be appreciated. Regards, Steve. Some info below: FS is configured as Multi-Tennant <param name="presence-proto-lookup" value="false"/> <param name="send-presence-on-register" value="true"/> <param name="manage-presence" value="true"/> <param name="presence-probe-on-register" value="true"/> <param name="manage-shared-appearance" value="true"/> <param name="dbname" value="freeswitch"/> <param name="presence-hosts" value="_DISABLE_"/> ** Multi-Tennant <param name="presence-privacy" value="false"/> SIP Trace for Subscription FS Receives this from the phone: SUBSCRIBE sip:203@xxx.mydomain.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.24:25060;branch=z9hG4bK991003231;rport From: <sip:201@xxx.mydomain.com>;tag=946578510 To: <sip:203@xxx.mydomain.com>;tag=nvqzSrr57RZg Call-ID: 968288340-25060-7(a)BJC.BGI.B.CE CSeq: 20050 SUBSCRIBE Contact: <sip:201@192.168.1.24:25060> X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2140 1.0.5.29 Expires: 480 Supported: replaces, path, timer, eventlist Event: dialog Accept: application/dialog-info+xml,multipart/related,application/rlmi+xml Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 FS Sends this back to the phone: SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.1.24:25060 ;branch=z9hG4bK991003231;rport=59364;received=x.x.x.x From: <sip:201@xxx.mydomain.com>;tag=946578510 To: <sip:203@xxx.mydomain.com>;tag=nvqzSrr57RZg Call-ID: 968288340-25060-7(a)BJC.BGI.B.CE CSeq: 20050 SUBSCRIBE Contact: <sip:203@y.y.y.y:5060> Expires: 480 User-Agent: Callstera VOIP PBX v1.20 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=480 Content-Length: 0

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