May 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

SIP info not relayed when call is on hold
by François-Xavier G. 13 May '20

13 May '20

Hello guys, I am trying to exchange SIP info message (dtmf signal) between Phone 1 and Freeswitch B as follows : Phone 1 <----> Freeswitch A <-----> Freeswitch B Everything works fine except when the call leg between Freeswitch A and B is on hold (for instance, the Freeswitch A has received an INVITE with "a=sendonly" from Freeswitch B). In this case, Freeswitch A correctly receive the SIP INFO message from Phone 1, but does not transfert it to Freeswitch B (as it does normally without having the call on hold). I do not see any errors or warnings in freeswitch logs, simply, freeswitch A is not sending the SIP info to freeswitch B. I'm using fs v1.10.2. Have any of you ever had this kind of needs or know how to solve it ? Regards, FX

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SIP info not relayed when call is on hold
by FX Geneste 13 May '20

13 May '20

Hello guys, I am trying to exchange SIP info message (dtmf signal) between Phone 1 and Freeswitch B as follows : Phone 1 <----> Freeswitch A <-----> Freeswitch B Everything works fine except when the call leg between Freeswitch A and B is on hold (for instance, the Freeswitch A has received an INVITE with "a=sendonly" from Freeswitch B). In this case, Freeswitch A correctly receive the SIP INFO message from Phone 1, but does not transfert it to Freeswitch B (as it does normally without call on hold). I do not see any errors or warnings in freeswitch logs, simply, freeswitch A is not sending the SIP info to freeswitch B. I'm using fs v1.10.2. Have any of you ever had this kind of needs or know how to solve it ? Regards, FX

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SIP info not relayed when call is on hold
by François-Xavier Geneste 13 May '20

13 May '20

Hello guys, I am trying to exchange SIP info message (dtmf signal) between Phone 1 and Freeswitch B as follows : Phone 1 <----> Freeswitch A <-----> Freeswitch B - - - > Everything works fine except when the call leg between Freeswitch A and B is on hold (for instance, the Freeswitch A has received an INVITE with "a=sendonly" from Freeswitch B). In this case, Freeswitch A correctly receive the SIP INFO message from Phone 1, but does not transfert it to Freeswitch B (as it does normally without call on hold). I do not see any errors or warnings in freeswitch logs, simply, freeswitch A is not sending the SIP info to freeswitch B. I'm using fs v1.10.2. Have any of you ever had this kind of needs or know how to solve it ? Regards, FX

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Acl to ignore mdns ice candidates generated by browsers
by Babak Yakhchali 13 May '20

13 May '20

Hi I'm trying to use sipjs with freeswitch but the problem is freeswitch is accepting candidates like this: a=candidate:2131708102 1 udp 2113937151 ad7f0d0b-7245-4c33-84af-fda3f2921d18.local 54385 typ host generation 0 network-cost 999 these are generated by mdns from browser and cause errors like AUDIO RTP REPORTS ERROR: [Remote Address Error!] and then INCOMPATIBLE_DESTINATION I tried wan_v4.auto as "apply-candidate-acl" but it is accepting those candidate too is there any builtin acl that I can use to make fs accept only public ipv4 addresses? thanks

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asr_handle codec = (null)
by Bernd Krueger-Knauber 12 May '20

12 May '20

Hi, I'm testing unimrcp but get empty rtp packets from fs to unimrcp server when I want detect speech. The support from unimrcp told me that with my unimrcp setup is everything fine. He pointed me to the following log entry: [INFO] mod_unimrcp.c:3134 asr_handle: name = unimrcp, codec = (null), rate = 8000, grammar = (null), param = uni2 codec = (null) is wrong. But I have no idea what I can do. Everything else is working in this fs installation. Also speech generation via unimrcp is working. I use FreeSwitch 1.8.7. Any hints?

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OPUS: Jitter Buffer for bridged calls to use FEC?
by Alexander Haugg 12 May '20

12 May '20

Hi, I try to find out the best practice to use the OPUS features. The scenario is: Client A -------> (JitterBuffer) FreeSwitch -----------------------> Client B <---------------------- FreeSwitch (JitterBuffer) <-------- Is this the correct way? In the documentation for OPUS the message is, JitterBuffer is needed to use FEC. But in the documentation for the JitterBuffer is written: If both sides of a bridge are RTP and both sides have a jb, its fairly useless. In fact if anything, it can worsen call quality. You should only run jitterbuffers at points of termination change of protocol. Examples, if FS was hosting a conference or IVR, if you are bridging the call to a phone for instance, you want to not use a jitterbuffer because you want to preserve the original timestamps so your phone can use its own jitterbuffer. I think, package lost is the start indicator for FEC and normally the RTP sequence number is the indicator for package lost, but the sequence number is valid for the own SSRC space only. That means, if I have lost packages from Client A to Freeswitch is this information on the other side to client B missing, correct? Now the question, what is the best practice handle this scenario? Thanks a lot! Alex

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send dtmf after bridge
by Joli Martinez 11 May '20

11 May '20

Hello, I am trying to dial an internal conference server automatically. I am able to dial the conference extension and if I dial the PIN number I get connected, but I need to be able to dial it automatically. I have tried to send the PIN number with "send dtmf" after the bridge and that did not work. I have tried to dial the extension with ",, " (pause) and that did not work either. Finally I tried to "queue_dtmf" before the bridge and it also failed. My goal is to have a dial plan so the user just dials the 3+ <conference PIN number> This will trigger the dial plan to dial the conference server and auto join the conference. So far I got it to dial the conference server, but can't figure out how to make it dial the pin after connection.

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Variable issue for setting conference record via API?
by Avi Marcus 11 May '20

11 May '20

I'm using javascript to trigger the conference API to record when the moderator joins. apiExecute("conference",argv[0] + " recording start $${recordings_dir}/conf-"+argv[0]+"/${strftime(%Y-%m-%d-%H-%M)}.mp3"); Those variables should be automatically expanded by FS, just like if it was used in the dialplan? But it seems they are being treated literally: 2020-05-07 19:25:36.609713 [ERR] mod_shout.c:893 Error opening $${recordings_dir}/conf-test/${strftime(%Y-%m-%d-%H-%M)}.mp3 2020-05-07 19:25:36.609713 [ERR] conference_record.c:277 Error Opening File [$${recordings_dir}/conf-test/${strftime(%Y-%m-%d-%H-%M)}.mp3] Ideas? Should I expand them myself? I was wondering about wrapping it in an eval, but that would probably have the same issue. So running an API to get the results of an eval first... Thanks, -Avi Marcus

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dtmf after bridge
by Joli Martinez 11 May '20

11 May '20

Hello, I am trying to dial an internal conference server automatically. I am able to dial the conference extension and if I dial the PIN number it works, but I need to be able to dial it automatically. I have tried to send the PIN number with "send dtmf" after the bridge and that did not work. I have tried to dial the extension with ",, " (pause) and that did not work either. Finally I tried to "queue_dtmf" before the bridge and it also failed. My goal is to have a dial plan so the user just dials the 3+ <conference PIN number> This will trigger the dial plan to dial the conference server and auto join the conference. So far I got it to dial the conference server, but can't figure out how to make it dial the pin after connection. Box1 where the users are connected is a FusionPBX server and the conference server is a plain freeswitch box.

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Rtcp
by David Villasmil 08 May '20

08 May '20

Hello guys, We have clients/vendor who exchange RTP directly to each other. But we'd like to get RTCP packets (and proxy them to the other side). Is this at all possible? Regards, -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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