June 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Concurrent RINGING calls limitation
by Kevin Mathy 17 Jun '14

17 Jun '14

Hi List, I'm trying to find something on freeswitch to make a "per-user" limitation of concurrent "RINGING" calls (or more generally, calls in PROCEEDING state). I've found the "max-proceeding" value I can set in switch.conf.xml, but it's a system's wide parameter, and I'd like more granularity... So, after some time spent on the mailing list and with my friend Google, it seems that someone asked for quite the same thing 2 years ago, but didn't get any answer : http://marc.info/?l=freeswitch-users&m=134375187528797 So, today, is there anything to achieve this ? Maybe something based on the "limit" module ? Thanks a lot, Kevin *Bien cordialement, Best Regards, **Kevin MATHY* | Ingénieur VoIP --

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 96, Issue 138
by Amin Dhamani Askhanuman 17 Jun '14

17 Jun '14

Thanks alot for your prompt reply. I will do myself. what configuration I need to change? Because provider is same for both inbound and outbound. On Tue, Jun 17, 2014 at 1:41 AM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Need to deploy 5 inbound and 5 outbound (Brian West) > > > ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 16 Jun 2014 13:40:53 -0500 > Subject: Re: [Freeswitch-users] Need to deploy 5 inbound and 5 outbound > should be no different than when you added the first one, Did you do that > or did you hire someone? Are they all different sip accounts or one sip > account with multiple DID's? Also one question at a time please. > -- > Brian West > brian(a)freeswitch.org > > > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > > > On Jun 16, 2014, at 9:54 AM, Amin Dhamani Askhanuman <amin(a)askhanuman.com> > wrote: > > Hi, > > Currently we are using freeswitch with 1 inbound and 1 outbound number now > we want to add 5 more inbound and outbound numbers to existing > configuration and call from 5 inbound and outbound will route to certain > extensions. Can you guide me how I can do that. > > > > _______________________________________________ > 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 > > -- Best regards, Amin Dhamani Amin Dhamani | System Administrator | +66 911167159 | Krits Tower, 1032/1-5, 14 Rama 4, Bangkok 10120 *www.askhanuman.co.th <http://www.askhanuman.co.th/>* | www.facebook.com/AskHanuman | amin(a)askhanuman.com This communication may contain information that is legally privileged, confidential or exempt from disclosure. If you are not the intended recipient, please note that any dissemination, distribution, or copying of this communication is strictly prohibited. Anyone who receives this message in error should notify the sender immediately by telephone or by return e-mail and delete this communication entirely from his or her computer. This communication may contain information that is legally privileged, confidential or exempt from disclosure. If you are not the intended recipient, please note that any dissemination, distribution, or copying of this communication is strictly prohibited. Anyone who receives this message in error should notify the sender immediately by telephone or by return e-mail and delete this communication entirely from his or her computer. ______________________________________________________________________________ -- This communication may contain information that is legally privileged, confidential or exempt from disclosure. If you are not the intended recipient, please note that any dissemination, distribution, or copying of this communication is strictly prohibited. Anyone who receives this message in error should notify the sender immediately by telephone or by return e-mail and delete this communication entirely from his or her computer. ______________________________________________________________________________

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Core Dump when Calling Extension from WebRTC/SIP.js
by Lyle Pratt 17 Jun '14

17 Jun '14

I'm trying to do some testing with WebRTC support using OnSIP's SIP.js library. However, whenever I try to call a FreeSwitch extension, I get a core dump on 1.4.beta. I'm able to connect the website AND register, but the moment I do an INVITE, it crashes. Has anyone else encountered this? Console Log: 2014-06-17 01:48:13.856557 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/1111(a)sip3.xyz.com [71d88070-f5c1-11e3-b884-638e8bd00f1c] 2014-06-17 01:48:13.856557 [DEBUG] switch_core_session.c:1052 Send signal sofia/internal/1111(a)sip3.xyz.com [BREAK] 2014-06-17 01:48:13.856557 [DEBUG] switch_core_session.c:1052 Send signal sofia/internal/1111(a)sip3.xyz.com [BREAK] 2014-06-17 01:48:13.856557 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1111(a)sip3.xyz.com) Running State Change CS_NEW 2014-06-17 01:48:13.856557 [DEBUG] sofia.c:8334 sofia/internal/1111(a)sip3.xyz.com receiving invite from 70.113.42.156:57858 version: 1.4.3 git 4904746 2014-05-19 17:38:31Z 64bit 2014-06-17 01:48:13.876638 [DEBUG] switch_core_session.c:1052 Send signal sofia/internal/1111(a)sip3.xyz.com [BREAK] 2014-06-17 01:48:13.876638 [DEBUG] sofia.c:2033 detaching session 71d88070-f5c1-11e3-b884-638e8bd00f1c 2014-06-17 01:48:13.876638 [DEBUG] switch_core_state_machine.c:486 (sofia/internal/1111(a)sip3.xyz.com) State NEW 2014-06-17 01:48:13.996579 [DEBUG] sofia.c:2138 Re-attaching to session 71d88070-f5c1-11e3-b884-638e8bd00f1c 2014-06-17 01:48:13.996579 [DEBUG] switch_core_session.c:1052 Send signal sofia/internal/1111(a)sip3.xyz.com [BREAK] 2014-06-17 01:48:13.996579 [DEBUG] switch_core_session.c:1052 Send signal sofia/internal/1111(a)sip3.xyz.com [BREAK] 2014-06-17 01:48:14.016556 [DEBUG] sofia.c:8334 sofia/internal/1111(a)sip3.xyz.com receiving invite from 70.113.42.156:57858 version: 1.4.3 git 4904746 2014-05-19 17:38:31Z 64bit Floating point exception (core dumped)

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IPv6 Audit
by Brian West 16 Jun '14

16 Jun '14

We need the community to assist in modules that do not support IPv6, I've filed FS-6600 for everyone to comment on, Please do some research and help eradicate the last lingering modules that don't support IPv6. https://jira.freeswitch.org/browse/FS-6600 Thanks, -- Brian West brian(a)freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

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Missing modules message - FS Install with mod_freetdm
by bedgar＠vseinc.com 16 Jun '14

16 Jun '14

All, I am in the process of updating to the newest 1.2.stable build, commit 8e975d8fed7bc9ee39f5ddc9406ef36ec8f186e1, and have encountered a few messages at the end of "make install". I have not encountered these before. I am using RBS/CAS T1's and am sure that these will require updating. WARNING: installed module: ftmod_analog_em.so was not installed by this build. It is not present in modules.conf. WARNING: installed module: ftmod_analog.so was not installed by this build. It is not present in modules.conf. WARNING: installed module: ftmod_skel.so was not installed by this build. It is not present in modules.conf. WARNING: installed module: ftmod_wanpipe.so was not installed by this build. It is not present in modules.conf. WARNING: installed module: ftmod_zt.so was not installed by this build. It is not present in modules.conf. Can someone shed some light on this for me. Thank you, Brian

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mod_fifo documentation addition
by Alvaro Mari-Thompson 16 Jun '14

16 Jun '14

Hi all, Some of you may have some various posts by me requesting information on mod_fifo and mod_callcenter as we've been looking at introducing a very simple ring-all call queueing to our hunt groups. After delving through source code for mod_fifo I came across the outbound_per_cycle variable which determines how many concurrent members ring when a call comes in. This is something I've spent days and days trying to get working but couldn't figure out what the behaviour was. I can now calculate on the fly how many members a hunt group has, and set this variable on each fifo. I'd like to contribute to the documentation of mod_fifo by adding a small paragraph based on this variable. I tried registering on wiki.freeswitch.org but it tells me new accounts are locked as the documentation is now moving over to confluence.freeswitch.org, but that site doesn't have anything on mod_fifo yet. Will the documentation be moved across, or is it waiting for someone to write it all up? Regards, Alvaro --- Voicebooth Limited t: +44 (0)20 3475 3999 e: <mailto:info@voicebooth.com> info(a)voicebooth.com w: <http://www.voicebooth.com/> www.voicebooth.com Voicebooth specialises in Hosted Telephony Services and SIP Trunks. Voicebooth Limited is Registered in England and Wales with Registration Number 07758368.

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FreeTDM - any good for legacy CAS T1 protocols?
by Bob Hartwig 16 Jun '14

16 Jun '14

Hello all, I have a project coming up that will require me to interface Freeswitch to some rather arcane legacy T1 protocols. Specifically digital loop start, and station-side digital loop start. I'm concerned that the userland support for these may be incomplete or even non-existent in FreeTDM. I've successfully used FreeTDM to connect to PRI, but when I look for span configuration options for CAS T1, the only things that I see that may be applicable are described as "analog" options, and that doesn't sound quite right for loop start on T1 channels (but I could be wrong about that). If I were to use Asterisk for this, I would configure chan_dahdi.conf for " signalling=fxs_ls" and/or "signalling=fxo_ls". Should I pursue using FreeTDM for this, or would I be smarter to incorporate a simple Asterisk gateway to handle this legacy CAS T1 signaling? Thanks, Bob

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Re: [Freeswitch-users] Example of working with upstream distros...
by Brian West 16 Jun '14

16 Jun '14

Results https://github.com/joyent/pkgsrc/commit/df3a615de013424ac366e6e61e22031decd… -- Brian West brian(a)freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Jun 5, 2014, at 9:55 AM, Brian West <brian(a)freeswitch.org> wrote: > FreeSWITCHers, > I wanted to share a good example of working with an upstream distro on a problem that prevented building FreeSWITCH successfully without some LDFLAGS trickery. > > Unable to build on SmartOS, Patch attached to this jira https://jira.freeswitch.org/browse/FS-6375, The patch wasn't exactly correct so I did some digging. > > If you read the JIRA the links to the mailing list posts to smartos-discuss posts and responses, along with the root issue has been found and a really dirty work around till they update their pkgsrc to not include the 32bit path in 64bit packages so their .la files will work correctly. > > > -- > Brian West > brian(a)freeswitch.org > > <whmcslogo.png> > > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > >

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call rejection in simultaneous ringing
by Preeti Trivedi 16 Jun '14

16 Jun '14

Hi, I'm trying to ring my extension and mobile ring simultaneously and it worked. Now, I want to stop ringing the other phone when I reject the call from one phone, same as the other phone stops ringing when I answer in one phone. I tried few channel variables but I couldn't get that working. My dial string is as follows: dial_string = string.format("{sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@ ${dialed_domain}}sofia/internal/1%s%s%s(a)xx.xx.x.xx ,[leg_delay_start=%d]${sofia_contact(${dialed_user}@${dialed_domain})} Thanks & regards, Trivedi

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DNS name as the public IP respond 403
by Shahzad Bhatti 16 Jun '14

16 Jun '14

Hi, i am trying to use DNS Name and for that i start my Freeswitch Using -nonat and in sip profile create a new profile with the following changes <param name="sip-port" value="8080"/> and <param name="ext-rtp-ip" value="host:sip.mydomain-name.com"/> rest conf is same as internal and when i try to register i got 403 response while on internal profile it works fine i am using FreeSWITCH Version 1.5.8b+git~20131202T224806Z~e127ca7d40~64bit (git e127ca7 2013-12-02 22:48:06Z 64bit) on centos 64-bit reply me what else i am missing regards Shahzad Bhatti

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