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Is there an exmple of "backup provider" ?
by Pierre Couderc 30 May '18

30 May '18

I have a provider not reliable but low cost so I need a backup provider in case of failure. Is there a typical configuration in this case ? Where ? Thanks in advance PC

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Can FS avoid sending re-INVITE to client? Or bridge on its behalf?
by M Yudkowsky 30 May '18

30 May '18

Comrades! I'd like to find out of if FS, which is making calls on behalf of a client, can intercept timer-based re-INVITEs and respond to them without sending the INVITE to the client first. Alternatively, can FS be made smart enough to bridge "in the cloud" on behalf a client, even if the client uses internal RTP port numbers? Details: I've got an internal server (IS) that uses FS as a firewall. The IS asks for leg A, then leg B, and then bridges leg A and leg B "in the cloud." FS (working in media bypass mode) handles this without any problems. For many non-NANP and some NANP calls, some SIP providers insist on a re-INVITE every 5 to 15 minutes. Unfortunately, the IS has a bug: if you send it a re-INVITE -- which is what FS does at this point -- the IS will respond correctly and continue the call in the cloud. But at the same time, the IS will open an internal RTP port (which is not used for anything) and then fail 60s later when no RTP reaches that port. At this point, the IS sends a BYE for one of the legs, requests the other leg back from the cloud, and continues to handle the remaining leg. Note that the IS in the course of normal processing does break the cloud conference, talk to each individual leg, bridges them together again or each with some other new leg... it's quite a dance of INVITEs and re-INVITEs that the IS sends. But inbound re-INVITEs related to timers is the problem at the moment. Two possible solutions: Solution A. Is there a way for FS to be made smart enough and aggressive enough to act on a routine, timer-related re-INVITE? If so, this would bypass the bug in our IS. Solution B. When FS is set to bypass RTP media, the bridge happens up in the cloud. In normal operating mode, without bypass or proxy of RTP set, this is what happens: When the IS attempts to bridge the calls it sends the bridging re-INVITEs to FS, and those re-INVITEs are for _internal_ ports that FS offered to the IS. FS bridges these internal ports but (and perhaps I'm reading Wireshark wrong) does not attempt to bridge the external ports and take the media into the cloud. The bridge is done on FS itself. A possible solution, then, would be to configure FS to determine that the media can be bridged in the cloud and have it do so. -- Moshe Yudkowsky Disaggregate Corporation 2952 W Fargo Chicago, IL 60645 USA +1 773 764 8727 speech(a)pobox.com http://www.Disaggregate.com http://www.PebbleAndAvalanche.com

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No Resend 183 with Ring Groups
by Ximo Peral 30 May '18

30 May '18

Hi! I have a bridge with ring groups in the dialplan, using ":_:". When I receive 183 from my provider, FS does not resend. I'm "solving" this with ring_ready, but I do not like this option. Do you have any idea why the 183 does not propagate?

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Call monitoring
by Francesco Facco de Lagarda 30 May '18

30 May '18

I am writing a "call monitor", something like an operator panel. I am capturing events via esl in php, filtering down the data and then sending them to a javascript UI via websocket. My problem is that I am totally bewildered by the amount of events that fs generates, and I'd like to filter it down to a bare minimum. New outgoing call (with uuid, extension and dialed number) Call answererd from other party Line busy Call hung up from either party. Can anyone give me a pointer to which events I really need to monitor, and which fields inside? I will obviously share finished code!

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Gateway
by Hossein Gholizadeh 30 May '18

30 May '18

Hello all I need to send fax from a fax machine which is out of our network to our freeswitch i need to register my gateway in freeswitch . my gateway configs is on HX4E which is on 192.168.30.32 . here i added a xml file in /conf/sip-profiles/external which content is like this: ------------------------------------------------------------------------------------------------------- <include> <gateway name="provider"> <param name="realm" value="192.168.30.32"/> <param name="proxy" value="192.168.30.211:5060"/> <param name="register" value="true"/> <param name="caller-id-in-from" value="true"/> <param name="username" value="*Etraabb2018*"/> <param name="password" value="*ET2018@)!**"/> </gateway> </include> ------------------------------------------------------------------------------------------------------ but after running freeswitch it cannot be registered and says : ---------------------------------------------------------------------------------------------------------- *2018-05-30 09:47:58.634332 [NOTICE] sofia_reg.c:448 Registering providerA* *2018-05-30 09:47:58.634332 [NOTICE] sofia_reg.c:448 Registering provider* *2018-05-30 09:47:58.655309 [WARNING] sofia_reg.c:2906 Can't find user [Etraab2018(a)192.168.30.211 <Etraab2018(a)192.168.30.211>] from 192.168.30.211* *You must define a domain called '192.168.30.211' in your directory and add a user with the id="Etraab2018" attribute* *and you must configure your device to use the proper domain in it's authentication credentials.* *2018-05-30 09:47:58.675323 [ERR] sofia_reg.c:2447 provider Failed Registration with status Forbidden [403]. failure #8* *2018-05-30 09:47:58.684466 [ERR] sofia_reg.c:2447 providerA Failed Registration with status Method Not Allowed [405]. failure #8* *2018-05-30 09:47:59.634936 [WARNING] sofia_reg.c:505 providerA Failed Registration [405], setting retry to 240 seconds.* *2018-05-30 09:47:59.644552 [WARNING] sofia_reg.c:505 provider Failed Registration [403], setting retry to 240 seconds.* ---------------------------------------------------------------------------------------------------------- can anyone help me?

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Outgoing calls through freetdm drop with cause code 102=recovery_timer_expire
by Babak Yakhchali 30 May '18

30 May '18

Hi I connected freeswitch to tda600 using a pri line and mod_freetdm. incomming calls from tda600 work without any problems but outgoing calls to tda600 drop with cause 102 after a few seconds after answering. I pasted the logs from console on pastebin: https://pastebin.freeswitch.org/view/5da22227

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No send 183 with Ring Groups
by Joaquin Peral 30 May '18

30 May '18

1 0

send fax
by Hossein Gholizadeh 30 May '18

30 May '18

hi all ... i already got the fax from a fax machine to my freeswitch gate way ok ... now i need to send this received fax file to other fax machine ... how can i do it?

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mod_lua
by Hossein Gholizadeh 30 May '18

30 May '18

hi ... after freeswitch version 1.4 , we don't need mod_lua?

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Mini IVR for caller inside of voicemail
by Preston Hagar 29 May '18

29 May '18

We've just finished replacing an older Mitel system with our new FreeSWITCH system. One feature of the old Mitel system we are missing that a few of our users have asked for worked like this: Person calls user's number (DID or internal). Caller gets user's voicemail. Either the users greeting could be recorded to say something like "To reach my cell phone, press 2, To reach my assistant, press 3, To leave me a message, wait for the beep". The caller could then decide to either press a key to be transferred to somewhere else, or just wait for the beep and leave a message. The user could setup their forwarding number(s) to prompt for through the voicemail menu prompts, it didn't require an admin or any other access. I know I could generally do this with an IVR setup, but it would be nice to have it like the old system where the user could add/remove the forwarding numbers at will and I wouldn't have to build out custom IVRs for multiple extensions. Does anyone have any pointers on how to achieve something like this? Is it even possible with mod_voicemail, or do I need to be looking into doing VM completely differently somehow? Thanks. Preston

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