- FreeSWITCH-users - FreeSWITCH Mailing Lists

What exactly enable_soa = false changes?
by Victor Chukalovskiy 10 Mar '15

10 Mar '15

Good day, I find documentation regarding enable_soa param very scarce, so looking for some clarifications here. What exactly changes in FS behavior with enable_soa = false? For example, I observe that with soa disabled, FS receiving "183" with SDP on one leg substitutes it with plain "180 Ringing" on another leg. This is not correct, but I'm not sure if it's a bug or an expected outcome of disabling soa. Thanks!! -Victor

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Console Loglevel Query
by Ali Jibran 10 Mar '15

10 Mar '15

Newbie to freeswitch so I apologize if it sounds basic. I was wondering if there was any way to access freeswitch console from fs_cli? I know fs_cli is a debug console for FS but is there any way I can access the original freeswitch console? I hope I make sense. Like I start freeswitch in background. Then I access it through fs_cli. Can I access the background-ed FS? Also if I can't, can I get the same loglevel in fs_cli as in FS?

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Bridge event socket channels with early media
by sukitha jayasinghe 10 Mar '15

10 Mar '15

I have developed a event socket application to handle my business logic with greater channel control. When internal user make outbound call it calls socket application and connect with the server, then server make outbound call using xml_rpc api with socket application parameters like ["&socket({0}:{1} async full)"] and that channel also connect with the server.I want to bridge channels before answer in-order to A-leg to listen telco messages such as "user not in service area". What is the proper mechanism for achive this. Regards, Sukitha

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any handset recommendations that operate like polycom+cisco?
by Nathan Neulinger 10 Mar '15

10 Mar '15

Looking for any handset/desk phone recommendations that have the "every line key can be it's own SIP account, even on expansion modules" behavior. Yealink units for example only allow a set number of lines, and the rest don't operate as line keys, so even though you can have a 30 + button phone, it's not usable for a receptionist or secretary that may handle calls for 15 different people with their own direct lines. With the polycom and cisco (sccp) phones, you can define N different SIP accounts. Are there any other phones out there that operate this way that anyone can recommend? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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Re: [Freeswitch-users] issue terminating outbound calls in freeswitch
by David Montecillo 10 Mar '15

10 Mar '15

It's a hardware where you can insert multiple sims but Im just using the single version(http://www.dbltek.com/products/goip-1.html). We use it as a gsm gateway for outbound calls and freeswitch setup instructions are available in the freeswitch pagehttps://freeswitch.org/ confluence/display/FREESWITCH/Goip+HowTo Do you have an idea whats causing the termination problem?

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Chat: invalid profile
by David Wafula 09 Mar '15

09 Mar '15

Hello all, I used to have the following command working with default installation. chat sip|1000@xxx.xxx.xxx.xxx|1001@xxx.xxx.xxx.xxx|hello sopranos After binding the chatplan to xml curl, the command no longer works, i get : 2015-03-09 12:47:41.238102 [ERR] sofia_presence.c:200 Chat proto [global] from [1000(a)xxx.xxx.xxx.xxx] to [1000(a)xxx.xxx.xxx.xxx] hello sopranos Invalid Profile xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx is the same IP through. Am not sure where to start looking. Help. -- David Wafula

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FreeSWITCH Week in Review (Master Branch) February 28th-March 6th
by Ken Rice 09 Mar '15

09 Mar '15

New Post on freeswitch.org from Kathleen check it out at http://ift.tt/18vTmtt FreeSWITCH Week in Review (Master Branch) February 28th-March 6th Hello, again. This passed week in the FreeSWITCH master branch we had 18 commits. The features for this week include: updating Windows build to use flite-2.0.0-release and updating mod_mongo driver to 1.1.0 . Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support. New features that were added: FS-7149 Update Windows build to use flite-2.0.0-release FS-7346 Update mod_mongo driver to 1.1.0 The following bugs were squashed: FS-7339 Move creation of view sql statements for basic_calls and detailed_calls to happen after the creation of the tables so the creation works and won’t have to be run a second time FS-7342 Fixed a crash regression in mod_conference caused by FS-7230 FS-7340 Remove json-c dependency in favor of our own json code FS-7350 Add –enable-address-sanitizer configure flag to enable clang address sanitizer FS-6758 Revert a commit from FS-6758 fixing hold dropping calls on Skinny Cisco 7961G FS-7305 Fix for making embedded versions of FS start up and shutdown faster like in the case of tone2wav FS-5570 Patch to add “multi” parameter to group api command. When the “multi” parameter is present, the group command will return a list of group members delimited by :_: which allows for multiply-registered endpoints to participate in a group.

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Re: [Freeswitch-users] Freeswitch --> Faxing issue
by Martin Schöpfer (GMX) 09 Mar '15

09 Mar '15

Hello, He told me so! Thanks >Step 1. Does your provider support T.38? >>On Mar 7, 2015, at 5:47 PM, Martin Schöpfer (GMX) < <mailto:martinschoepfer@gmx.de> martinschoepfer(a)gmx.de> wrote: >> >>Hello, >> >>can anyone help me by my config for faxing with freeswitch. I have testet al functions of spandsp also to set the software Modems active and receive faxes with hylafax but nothing works. >> >>Thats the function I want to get >> >>inbound >> Sip-provider à Freeswitch à Cisco SPA3102 Line 1(T.28 enabled) à hp fax or another >>For now i´m getting the message no fax detected from my faxing maschine >>Outbound >>Hp fax or another à Cisco SPA3102 Line 1 à Freeswitch à Sip-provider . >>There I get the message phone busy always I know the other faxing maschine isn busy thats my second >> >>The softmodem are also always busy I cant check the function by cu l FS because it tells me line in use. >> >>Freeswitch Version is the latest master branch of 1.4

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mod_conference and outbound call status
by Sharath Kumar 09 Mar '15

09 Mar '15

Hi, I am using mod_conference to call out a bunch of users. I would like to know how does the original caller(ie the moderator who called the bridge) come to know of the status of these calls, say I want to display the status of whether or not the users accepted the invitation or not. Any ideas ? Thanks Sharath

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Incoming call - ring call group - then voicemail
by Richard Mace 09 Mar '15

09 Mar '15

Hi All, I am trying to get an incoming call to ring the group 1000 in the domain wph.co.uk and then go into the voicemail for 200. Is the following code close because it doesn't seem to work? <action application="set" data="call_timeout=20"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/wph.co.uk/group/1000"/> <action application="answer"/> <action application="voicemail" data="default wph.co.uk 200″/> Thanks in advance Richard

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