June 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Help with inbound calling
by Rick Ewart 22 Jun '15

22 Jun '15

Hi All. I am new to FreeSwitch (but not to VoIP in general) and have been beating my head against the wall for a number of hours now on this. I seem to be having an issue with what seems like it should be a SIMPLE thing to do. I have a carrier that I am registering my FS to, and I am able to make outgoing calls without an issue as well as calls between extensions. However when I make inbound calls I get an error about no route. I have researched everything I can find, tried 50 different variations on the same thing, and haven't found a solution. It seems like it has to be something simple I am missing. The config is pretty standard from the regular install. FreeSWITCH Version 1.4.15~64bit on CentOS for whatever that matters. Below is the error I am receiving on the console. I made things a bit generic but in this case I am calling the DID 18585551212 (I get 11 digits from the carrier). My FS is 123.123.12.31 and the upstream carrier is 123.123.12.30. No NAT here - both are public IPs and actually in the same subnet. As you can see below, I have ext 1000 registered to the system, and the carrier gateway is registered also. Finally, I provided my inbound DID dialplan. I am grateful for any help you can provide. Thanks. Rick [NOTICE] switch_channel.c:1055 New Channel sofia/external/18585551212@123.123.12.30:5060 [abd31162-1886-11e5-a031-c396b897fabe] [INFO] mod_dialplan_xml.c:635 Processing 18585551212 <18585551212>->18585559999 in context external [WARNING] mod_dialplan_xml.c:665 Context external not found [INFO] switch_core_state_machine.c:241 No Route, Aborting [NOTICE] switch_core_state_machine.c:242 Hangup sofia/external/18585551212@123.123.12.30:5060 [CS_ROUTING] [NO_ROUTE_DESTINATION] [NOTICE] switch_core_session.c:1633 Session 32 (sofia/external/18585551212@123.123.12.30:5060) Ended [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/18585551212@123.123.12.30:5060 [CS_DESTROY] dialplan/public/00_inbound_did.xml: <extension name="18585559999"> <condition field="destination_number" expression="18585559999"> <action application="set" data="domain_name=$${domain}"/> <action application="transfer" data="1000 XML default"/> </condition> </extension> Name Type Data State ========================================================================== ======================= external profile sip:mod_sofia@123.123.12.31:5080 RUNNING (0) external::xyzswitch gateway sip:ricktest@123.123.12.30 REGED atl.somecompany.net alias internal ALIASED internal profile sip:mod_sofia@123.123.12.31:5060 RUNNING (0) ========================================================================== ======================= Registrations: ========================================================================== ======================= Call-ID: fd6446250b3b4d489aad6bb3a5bf5603 User: 1000(a)atl.somecompany.net Contact: "Rick" <sip:1000@10.1.10.123:5060;ob;fs_nat=yes;fs_path=sip:1000@70.12.123.222%3A 5060;ob> Agent: MicroSIP/3.3.31 Status: Registered(UDP-NAT)(unknown) EXP(2015-06-21 22:52:03) EXPSECS(208) Ping-Status: Reachable Host: atl.somecompany.net IP: 70.12.123.222 Port: 5060 Auth-User: 1000 Auth-Realm: 123.123.12.31 MWI-Account: 1000(a)atl.somecompany.net Total items returned: 1 ========================================================================== =======================

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Reinvite handling question
by John Nash 22 Jun '15

22 Jun '15

I am using freeswitch in media bypass mode. I have a scenario like ... UA-------->Freeswitch---------------> Gateway My gateway sends ReInvite in every 12 minutes as keep alive and freeswitch forwards it to UA (Which I think is expected behavior as freeswitch is not handling Media) but the issue is UA does not support it. Is there any way I can "handle" reinvite at freeswitch in such a way that in case gateway sends same SDP freeswitch sends 200 OK to gateway without forwarding it and send forbidden in case different SDP is sent? John

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Extension preferred gateway using dial prefix
by Roman Dissauer 22 Jun '15

22 Jun '15

Hi all, I have multiple gateways which can be reached directly via prefixes. *101<destination number> -> gateway1 *102<destination number> -> gateway2 and so on… Now I want some extensions to always use a specific gateway. Is it possible to set a fixed dial prefix for extensions without using the dialplan? Thanks! Roman

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Blocking commands from mod_v8
by Martyn Davies 22 Jun '15

22 Jun '15

I'm running an application in javascript in mod_v8, but in a couple of cases I need to call external commands. Sometimes these block and never return, so I can see another freeswitch instance in the ps listing, and if I kill with -12, my javascript app will continue from where it got stuck. I've tried using 'system' and 'apiExecute' (in turn calling 'system'), and experience blocks with both. One example of a blocking command was 'soxi', but as a test I called to an executable that simply prints "0" and then exits. This also gets blocked. Freeswitch version is 1.4.12, and I'm running on Ubuntu 12.04. In my test environment this is running on a virtual machine, but I've also experienced the blocking problem on hardware with the same configuration. I would appreciate any tips you could give me. Regards, Martyn

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Hangup after first action in phrase
by Alex Yakovenko 22 Jun '15

22 Jun '15

Hi all. I'm running Freeswitch 1.4.19 and I have written a macro: <include> <macro name="saypin"> <input pattern="(.*)"> <match> <action function="execute" data="sleep(1000)"/> <action function="execute" data="sleep(1000)"/> <action function="execute" data="sleep(1000)"/> <action function="play-file" data="/usr/local/freeswitch/recordings/pre-code_text.wav"/> <action function="say" data="$1" method="en number iterated" type="number"/> <action function="play-file" data="/usr/local/freeswitch/recordings/once-again.wav"/> <action function="say" data="$1" method="en number iterated" type="number"/> </match> </input> </macro> </include> But it hangs after first action like "play-file" or "say" with NORMAL_CLEARING cause, though "execute" actions are going normal. In log I see following http://pastebin.com/DqJXhMWU What can cause this issue ? How I can debug this issue more deeply? -- WBR Alex Yakovenko

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Help required for integratrion of "Mod Nibblebill" in FreeSwitch
by Manish Talwar 22 Jun '15

22 Jun '15

Hi, We have integrated our IVR application with FreeSwitch by HTTAPI and make phone calls via DIAL HTTAPI command through Nexmo services. Its working fine now and we are able to make a phone call via Nexmo Gateways. But we need to integrate Billing of phone call in our IVR application now. I have heard about "Mod Nibblebill" in FreeSwich for controlling calls duration and making billing to a customer. If we can make billing process by "Mod Nibblebill" then please let me know some more useful information about "Mod Nibblebill" and how can we integrate it in FreeSwitch with Nexmo Gateway. Thanks, Regards, Manish Talwar

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telephone-event at multiple rates
by Dragos Oancea 22 Jun '15

22 Jun '15

Hi Can someone test this patch ? https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/274… It's Anthony's patch for "FS-7578" . I assume there are people out there who want DTMF (rfc2833) working with Opus , this patch fixes the issuewhere FS was answering (or offering) only telephone-event/8000 , and SIP clients would correctly refuse to send such rtpevents over an RTP stream with clock-rate 48 khz.This patch is obviously not only for Opus, is generic, for any codec.Thank you. Regards,Dragos

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how to signal ringing to incoming calls when user is not registered?
by wiebittewas 21 Jun '15

21 Jun '15

when a phone is offline freeswitch immediatly send an unavailable to the caller, but we want, that the caller think, it's ringing for a while, even if there's no voicemailbox available. campon does not work, because it takes the call. is there any possibility to wait a certain amount of time without answering or canceling the call so the caller just think the target-phone would be ringing? regards w.

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suggest some SIP hard phones (Support freeswitch).
by Shisheer Teli 21 Jun '15

21 Jun '15

Suggest some SIP hard phones (Support freeswitch). need to buy for my organisation -- Regards, Shisheer Teli

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Noise Detection
by Menachem Fish 21 Jun '15

21 Jun '15

hi everyone im trying to implement noise detection in freeswitch (as a house-monitoring system) id like to know when any noise occurs on the line so it can alert me by email etc. ive done the following settings SwitchExecute("set", "fire_talk_events=true", swchEvnt.ChannelCallUUID) SwitchExecute("set", "rtp_enable_vad_in=true", swchEvnt.ChannelCallUUID) (SwitchExecute just calls esl.Execute) i am getting "Talk" events in my log, but they do not seem to be related in any way to actual noise. maybe random, maybe when i press a dtmf (seems to be generating talk events) in any case, talking to the phone does not generate any talk events i am not setting vad=both or anything like that since i do not want the phone to disconnect or something upon silence. or did i get this wrong? any ideas? additional, id like it to generate events on any noise, not specifically human voice speaking *Menachem Fish* *Always at your service* *Machshevet - Your Data Your Way* *---*

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