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help in architecture
by Aqs Younas 11 Jun '15

11 Jun '15

Hi, Hope you all are having a good day. :) I need some suggestions in designing SIP trunking with freeswitch. We are required to register more than 10,000 users on multiple freeswitch instances sharing same database. How can we load balance registration on each freeswitch so, they have equal number of registrations. I know with mod_xml_curl i can have freeswitch share same database. But how can i load balance registrations? (inbound calls must be directed to that freeswitch with which user is registered) Or any suggestion how we can achieve this, so, that we can add more freeswitch in future. Thanks for your help.

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Adding Allow-Events header to 100 and 200 SIP messages
by SebA 11 Jun '15

11 Jun '15

Hi, I would like to test adding the following header to the 100 Trying and 200 OK messages: Allow-Events: telephone-event This is to try and interoperate with Cisco equipment that behaves this way. As it stands, perhaps as a result of not sending this header from our FreeSWITCH box, the Cisco is not offering RFC 2833 in the SIP ACK and therefore DTMF does not work (as they don't support SIP INFO). Is there any way of doing this in FreeSWITCH? Many thanks. Kind regards, SebA

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g729* offer > g729a answer
by Dmitry Saratsky 11 Jun '15

11 Jun '15

Hi All, Is it allowed to answer with G.729A (G.729, fmtp: annexb=no) to ANY G.729 combinations offer requests? in other words: If I receive offer request with: G.729, G.729A, G.729B, G.729AB May I answer with G.729A only? Thanks, Dmitry.

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Mono To Stereo
by Jon Lederman 11 Jun '15

11 Jun '15

Hi, We need to set up calls in FS where a mono signal coming into Freeswitch is split into a stereo signal. What is the easiest way to accomplish this? Thanks Sent from my iPhone

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mod_conference, endconf, execute_on_answer
by René Weiss 11 Jun '15

11 Jun '15

Hi I have setup an outbound conference which ends automaticly if the initiator leaves thanks to the "endconf" flag. Now I got an new requirement: The called participants shold not automaticly join the conference, but instead the system should first play an audio file like "this is a conference call, press 9 to enter", and only after pressing "9" they should be added to the conference. I solved this (mostly) with "execute_on_answer" and a lua script: --- repeat digits = session:playAndGetDigits(1, 1, 3, 3000, "#", "audio.wav", "", "[9]") until digits == "9" --- The only problem is now that when the initiator hangs up while this script is running the call/conference doesn't end. In my tests the call continued like nothing happend, and after pressing "9" the participant was still redirect to the conference where he was now the only member. Any ideas how to solve this? Thanks.

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Understanding mod_verto dial-string to activate verto to all local extensions
by Ricardo Hernandez 11 Jun '15

11 Jun '15

Hi, I'm trying to implement the changes explained in https://freeswitch.org/confluence/display/FREESWITCH/mod_verto, to activate it. The part I don't understand fully is the dial-string changes, for example in the choice 1: Usage: ${verto_contact ${dialed_user}@${dialed_domain}} My first try was to use the second option: <param name="dial-string" value= "{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(*/${dialed_user}@${dialed_domain})}:${verto_contact ${dialed_user}@${dialed_domain}}"/> What I try was to call a local extension that is logued using verto, but somehow is trying to look for it like if is a sofia user, of course it says that the user is not available. I want to understand this line fully, mostly because of the flow that traverse this dial-string. I'd also try the last option of saying that a specific user is a verto user: <params> <param name="dial-string" value="${verto_contact ${dialed_user}@ ${dialed_domain}}"/> </params> That works wonders, but I want my users to use SIP and Verto when one or other is available. Last thing I try is to use choice 2, to bridge in the dialplan: verto.rtc/ foo(a)bar.com In that case what I do was to use the local extensions dialplan and put the bridge after the sofia user/XXXX, something like a failover option. In this case works calling from SIP (sofia) to verto, but the other way it says the user is not available. The question then is, how can a have a user that can be called using sofia and verto, depending from where is registered/logued Thank You in advance

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Sofia recover without Floating-IP
by Sharath Kumar 11 Jun '15

11 Jun '15

Guys, I have FS1 and FS2 on say 2 different data centers and have different IPs. They do share the data via MariaDB, Now due to some limitation, I cannot use a floating-IP. Is it possible for me to "recover" calls from FS1 to FS2 when FS1 goes down and I issue "sofia recover from "fs2" ? i.e master-master sort of scenario. Also, can I recover conference calls using sofia recover ? Thank you, Sharath

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SIP Auxiliary Data and Notification
by Jon Lederman 11 Jun '15

11 Jun '15

Hi, I’d like to send some auxiliary data via SIP during call setup and have freeswitch provide notification to an external process when that call setup is occurring and provide the auxiliary data. What would be the best way to achieve this? In particular, what events should I subscribe to in order to track call setups occurring in freeswitch in such a way that I can grab the auxiliary SIP data? Thanks.

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IPv6 and FreeSWITCH
by Brian West 11 Jun '15

11 Jun '15

FreeSWITCHers, Keep an eye out, we now have quad A records published, you can access stash, jira, freeswitch.org, cluecon.com and freeswitch.com via IPv6. Report any issues you see to me please. Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> | Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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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improper conversation sequence
by Aakash 10 Jun '15

10 Jun '15

Hi All, Freeswitch Calls are recorded in mp3 format in my setup. I experience the improper conversation sequence on the recordings. Any advice and suggestions will be greatly appreciated Thank you, Aakash ----- Regards, Aakash -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/improper-conversation-sequenc… Sent from the freeswitch-users mailing list archive at Nabble.com.

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