June 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] mod_gsmopen: are USB dongles the only option?
by Giovanni Maruzzelli 12 Jun '16

12 Jun '16

Ciao Stanislav, the whole idea of gsmopen is to use cheap and ready available radio interfaces, that maybe you can get off ebay or surplus. Also, they are undistinguishable from "normally used" dongles, and that is a plus in a lot of countries. I don't know about pci modems that exposes an audio interface to the operating system. Some soldering and demuxing will be probably needed. But at that point, I would go with Samgoma or Khomp cards... Also because for doing it well, a lot of engineering details are involved, from shielding to low pass filtering, to magnetic interferences, etc Anyway, I like those things too! :) -giovanni sent from mobile cell: +39 347 266 56 18 Giovanni Maruzzelli OpenTelecom.IT Il 05/Giu/2016 16:06, "Stanislav Sinyagin" <ssinyagin(a)gmail.com> ha scritto: does anyone know a GSM/UMTS/LTE modem that has mini-PCIe interface and is able to deliver voice signal that can be passed to mod_gsmopen? The only one that has any kind of voice signal that I found, is Huawei MU609 and the like. But they deliver PCM voice as a raw signal on mini-PCIe pins which are marked as "reserved" in the interface specifications, so normally the boards don't have any wiring for those pins. I just don't like the idea of an USB dongle, and there are many hardware platforms with mini-PCIe interfaces. They work perfectly for mobile data, but I couldn't find any option for voice. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Dynamic Group Call
by Mititelu Stefan 12 Jun '16

12 Jun '16

Hi, I'm a Freeswitch newbee and I'm trying to set up dynamic group calls. Freeswitch will receive a header with "Group-Users:user1,user2.." and what I want to do is to "transfer" to a dinamycally created context that will look like: <action application="conference_set_auto_outcall" data="user1"/> <action application="conference_set_auto_outcall" data="user2"/> Is this possible? Can you give me some references? >From what I see in some examples one can transfer only to existent XML context: <action application="transfer" data="1105 XML default"/> Thanks, Stefan

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Dynamically change conference moh-sound
by Oleg Stolyar 11 Jun '16

11 Jun '16

Hi guys, I need to play different hold music (local stream) to conferences depending on where the first conference participant is calling from. I have a couple dozen various moh streams. I'd rather not have to create a couple dozen almost identical conference profiles. Is there a way to dynamically set moh-sound on a conference in the channel before creating the conference?

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Enable nat-options-ping per user
by Volodymyr Fedorov 10 Jun '16

10 Jun '16

Hi Community, I have a question: according to wiki: https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal parameter <param name="nat-options-ping" value="true"/> may be assigned to profile. But may it be assigned to particular user ? I tried to do this like that: <user id="pbxuser"> <params> <param name="password" value="PwgEn123"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1000"/> <variable name="user_context" value="mypbx"/> <variable name="callgroup" value="techsupport"/> <variable name="sip-force-expires" value="180"/> <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> <variable name="nat-options-ping" value="true"/> </variables> </user> But freeswitch does not send option to registered endpoint. Thanks ! -- Best regards, Volodymyr

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play_and_get_digits issue
by Colin Morelli 10 Jun '16

10 Jun '16

Hello, I'm trying to use play_and_get_digits with mod_ssml to speak an SSML string while listening for DTMF tones. However, I seem to be having an issue with how to escape the playback string. If I execute: <action application="play_and_get_digits" data="0 1 1 5000 # silence_stream://5000 silence_stream://250 digits" /> Everything works great, but if I instead do <action application="play_and_get_digits" data="0 1 1 5000 # ssml://<speak>Hello there</speak> silence_stream://250 digits" /> I hear the audio and FS collects the digits, but it doesn't assign the dialplan variable. I can only assume that this is a result of the space in the "Hello there" string, but I can't figure out how to properly escape it (wrapping in single quotes doesn't seem to work, setting a channel variable and using that doesn't seem to work). Am I missing something obvious here? Tried to search the documentation for escaping dialplan variables but didn't find anything. Thanks, Colin

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Enterprise originate, SIP 302, sip_redirect_profile
by royj＠yandex.ru 10 Jun '16

10 Jun '16

Hi, all The goal is call multiple destinations simultaneously. And when receiving SIP 302 from some of destination, the another calls must not be interrupted. There is enterprise originate functionality for that. But we can not use 'sip_redirect_profile', 'sip_redirect_context' channel variables to route to redirection number in a dialplan when enterprise originate in action. I mean there is no effect of setting that variables. When non enterprise originate (',' as separator) 'sip_redirect_profile', 'sip_redirect_context' channel variables work as expected, but calls to another destinations end. Is there any way to use 'sip_redirect_profile', 'sip_redirect_context' in the enterprise originate?

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Activating MWI from a Mitel 3300
by Ralph Willett 10 Jun '16

10 Jun '16

I have a freeswitch connecting to a Mitel 3300 via sip trunks. We are using the Mitel embedded voice mail. I'm trying to figure out how to light the MWI light on sets running off of the freeswitch. Is it possible to light a MWI across the SIP trunks? Does anyone have suggestions on how to proceed to figure out how to do this? Dryaquaman

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RFC 2833 payload type mismatch issue on bridged call
by Dave Horton 10 Jun '16

10 Jun '16

I have a client sending an INVITE to FS with RFC 2833 offered as payload type 96. My freeswitch device profile has "rfc2833-pt" set to 101 and "inbound-late-negotiation” set to true. The FS then bridges this call to a gateway, sending an INVITE offering pt 101 for 2833. The 200 OK on the B leg comes back (with pt 101), and FS then sends a 200 OK to the original INVITE with pt 101. So, to summarize, the client has offered 101 for 2833 and the FS responds with an SDP with 96. At that point, DTMF does not work (DTMF generated by the client not recognized by the far end gateway). Is there a configuration setting that would tell FS to use the RDC 2833 payload type from the A leg when bridging a call? Interestingly, I am only seeing this problem on servers I have upgraded (to 1.6.8) — older servers do bridge the B leg offering pt 96 if that came in on the A leg. (Note, though, that the older servers are also not configured to do inbound-late-negotiation).

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Safe SIP URI Formatting
by Vladislav Ivanov 10 Jun '16

10 Jun '16

Hello, Is it possible to somehow alter the logic of Safe URI Formating to not transform '#' sign to '%23'? Maybe I can somehow escape the hash sign to not transform? There is some equipments that cant do back-formating and reject calls with "%23" signs with them. Best Regards

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System global mode variable
by Pete Ashdown 10 Jun '16

10 Jun '16

What is the simplest way to have a conditional branch based on a global variable in a dialplan? Actually using a global variable and xml is giving me mixed results. Do I need to use Lua or another scripting language? This is what I have, which doesn't give the results desired. The "speak" actions are merely debug placeholders. I know there are better ways to do conditional voice to text. <condition regex="any" field="${cond(${mode} == 'open' ? YES : NO)}" expression="^YES$"> <action application="speak" data="open"/> </condition> <condition regex="any" field="${cond(${mode} == 'tech only' ? YES : NO)}" expression="^YES$"> <action application="speak" data="teck only"/> </condition> <condition regex="any" field="${cond(${mode} == 'event' ? YES : NO)}" expression="^YES$"> <action application="speak" data="event"/> </condition>

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