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Perl equivalent dialplan
by mehroz 05 Nov '13

05 Nov '13

I can get the registration status of a user in XML dialplan with : <condition field="${sofia_contact(internal/${destination_number}(a)192.168.1.5)}" expression="error/user_not_registered"> ... and route the call accordingly. I am writing a perl script and calling it in XML with <action application="perl" data="/usr/local/src/test.pl/> and want to get the same registration status from the same CHANNEL VARIABLE. I can get CHANNEL VARIABLE using $session->getVariable("variable"); but cannot get ${sofia_contact(internal/${destination_number}(a)192.168.1.5)} value with this procedure. I believe this is not actually a CHANNEL VARIABLE , but I cant find ant other way of retrieving this information within my script. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Perl-equivalent-dialplan-tp75… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] Scalability and Bridge
by DanB 05 Nov '13

05 Nov '13

REFER is another good idea. The only issue is that you will need to trust the originator handling it properly vs establishing the legs yourself and reinvite which could be more on your control. Cheers, DanB On 05.11.2013 16:02, freeswitch-users-request(a)lists.freeswitch.org wrote: > A REFER (transfer) would be more appropriate in this scenario.

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Re: [Freeswitch-users] Raspberry Pi, mod_gsmopen GSM dongle
by Sergey Zhuravlov 05 Nov '13

05 Nov '13

Hi! Not so bad FS I still work with the module gsm_open, sometimes it is not clear why the device is heavily loaded. load average: 4.07, 3.99, 3.96 Now I want to watch CPU module with or without gsm_open. I want to collect data using sar and show to the public. The sound quality is not very much, but like a normal. use Raspbian Linux raspberrypi 3.6.11+ #538 PREEMPT Fri Aug 30 20:42:08 BST 2013 armv6l GNU/Linux For I am not afraid that the long compile, just to work with dignity;-) From: Johannes Stads <stadsj(a)yahoo.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Sun, 3 Nov 2013 11:39:49 -0800 (PST) > Subject: Re: [Freeswitch-users] Raspberry Pi, mod_gsmopen GSM dongle > Sergey I tend to agree with the statement in your first post, it's a shame > that Asterisk works better > in your setup. But i think there the problem is not in Freeswitch. The > problem is > in gsmopen module, in its features and support. > I made a question about gsmopen back in september, and did not receive any > answer. Gsmopen is in > a beta state, we should not require any stable operation of FS with > gsmopen module in a production > environment. > I have a long experience with Asterisk, and decided to give a try to > Freeswitch to see its superior > performance. I started using a linux virtual machine. Then tried it also > with BeagleBone Black which is > just a bit more powerful beast than PI. All compiled well, including the > latest version of FS. > Building of a mod_gsmopen was a bit of a pain, but finally worked too. > What linux flavor were you using with FS on PI? > In my setup I used bare minimum config, and again can confirm that FS is a > superior platform with > plenty of advanced features, though its concept is more difficult for > understanding initially. > Although I was able to send SMS and make calls out of the box, gsmopen > worked highly unstable and > sound quality was not good, to say the least. > > To prove my concept, today i took a fresh SD card to put Debian Wheezy on > it and installed Asterisk > with chan_dongle. The platform used was BBB. > What can i say, the experience was different. To compare, FS build time on > BBB/Debian/SD card was > such: > ./configure (about 1h) > make (about 2h) > Asterisk was up and running in about 30 min. > chan_dongle was built smoothly without a need to tweak libraries. > > In my FS setup gsmopen did not show a claimed 'auto discovery' feature and > some exercise with > UDEV rules was required to make sure that modems are loaded in right order. > In chan_dongle it is nicely done out of the box. > > Overall, chan_dongle is much more advanced piece of software, I want to > use this opportunity to thank its > author. If your goal is calling to mobiles, you have to consider > chan_dongle against gsmopen, > and not FS against *. > > Cheers, > Jo > > ------------------------------ > -------------- > On Wed, 10/30/13, Sergey Zhuravlov <4orbit(a)gmail.com> wrote: > > Subject: Re: [Freeswitch-users] Raspberry Pi, mod_gsmopen GSM dongle > To: freeswitch-users(a)lists.freeswitch.org > Date: Wednesday, October 30, 2013, 4:16 AM > > I am glad to > tell you that: > 1) mod_gsmopen and e1550 work on Raspberry Pi (incoming, > outgoing SMS in both directions). > > 2) and the stable version FS and the latest FS compile on Raspberry Pi, > but it's a long time. > > 3) The latest version does not run on my raspberries. > > Need advice on optimizing the FS for raspberry Pi, as low-productivity > device. > > > Can any startup options? > Maybe someone knows what modules you can disable resource-intensive. > > Something else? > > > Does somebody > have a positive experience with > Raspberry > Pi, mod_gsmopen GSM > dongle ??? > > I use Raspberry Pi > B and e1550. > > It's a shame that > with an asterisk all > works fine -- RasPBX. http://www.raspberry-asterisk.org/ > > > > But I prefer the > FreeSWITCH ;-) > On the > basis of experience, I thought > that PS will work best > for this modest > hardware faster and > require fewer resources. > > > > But it turns out it is not. > Asterisk with a web > server, MySQL database, and > other, less load > than FS. > > > In this connection, other > parameters may be used > starting FS to save > resources? > > > And the main > question! Silence an > incoming call on 5000 > (IVR) as well as an > outgoing in gsm > network. Those are not > sound. > > > > -- > WBR, Sergey > > GTALK/JABBER:4orbit@gmail.com > > > > -- > WBR, Sergey > > GTALK/JABBER:4orbit@gmail.com > > > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.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 > > > > -- WBR, Sergey GTALK/JABBER:4orbit@gmail.com -- WBR, Sergey GTALK/JABBER:4orbit@gmail.com

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Re: [Freeswitch-users] Scalability and Bridge
by DanB 05 Nov '13

05 Nov '13

Valter, If the call is already established and you want to move it into a conference knowing the box where the conference is I guess you need own application logic which will call the bridge from the box where your call is active and do a reinvite then putting the party you want to bridge with the bridge itself. DanB

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Re: [Freeswitch-users] Scalability and Bridge
by Valter Cardoso 05 Nov '13

05 Nov '13

I think 302 Redirect only works at call establishment time. My problem is the call is already active in a FS and I need to bridge it with a call in another FS (or add it to a conference in another FS). For this, I think 302 will not help. Thanks, Valter

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481 Call Does Not Exist
by Miha 04 Nov '13

04 Nov '13

Hi, I need a little help with figuring out why FS sends "481". This is happing when call has been forward (302). All calls are relayed throught opensips as I am using it for registrations and load_balancing. http://pastebin.freeswitch.org/21555 tnx! miha

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Re: [Freeswitch-users] Scalability and Bridge
by Dan-Cristian Bogos 04 Nov '13

04 Nov '13

Ben, You are right about possible race here. One way I could sort of fix it is by checking at the establishment time for a reservation token in the database - redis in my case (valid for the probability time of the race). This tocken mechanism would be only used if there has been no prio conference active. I did not hit many race conditions so far but also not many concurrent setups in a conference to be honest. DanB Am 04.11.2013 11:06 schrieb <freeswitch-users-request(a)lists.freeswitch.org>: > 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. SIMPLE IM in UTF-16 (Alex Chan) > 2. Re: Raspberry Pi, mod_gsmopen GSM dongle (Jayanth Acharya) > 3. Re: 481 Call Does Not Exist (Miha) > 4. Re: Scalability and Bridge (DanB) > 5. Skype is likely to stop supporting desktop API by Dec. 2013 > (Henry Huang) > 6. Re: Scalability and Bridge (Ben Langfeld) > 7. Re: Execute_on_answer and ${strepoch} (Gregor Nanger) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 4 Nov 2013 09:46:32 +0800 > From: Alex Chan <alex(a)slackadmin.com> > Subject: [Freeswitch-users] SIMPLE IM in UTF-16 > To: FreeSWITCH-users(a)lists.freeswitch.org > Message-ID: > <CAG= > KnHW2BkP1k-PXGt4N8-t5DCH5Kye8bNtM+aqyauiMZtvtug(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi folks, > > I'm using FreeSWITCH as a SIP IM B2BUA. When the content-type was in > UTF-16BE, the message body will be dammed at where the zero byte 0x00 first > appears. Any ideas? > > Thanks in advance. > > Alex >

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Execute_on_answer and ${strepoch}
by Gregor Nanger 04 Nov '13

04 Nov '13

One funny "by design" catch. I want to set variable when channel is answered. execute_on_answer is perfect tool for this job. I have: <action application="set" data="execute_on_answer=set myvariable=${strepoch()}" /> Everything is working, except that ${strepoch()} is evaluated during dialplan parsing, not when on answer command is executed. So I get time when diaplan is processed, not when user pickups. Is there a notation to tell FS not to evaluate variable during dialplan parsing? Best regards, Gregor

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Re: [Freeswitch-users] Scalability and Bridge
by DanB 04 Nov '13

04 Nov '13

Hi Valter, There are multiple approaches to this issue. I personally go for the following: monitor active calls in a pool of FS boxes somewhere in a database (NoSQL preferred by me) and issue a 302 redirect from the FS box where the call hits initially towards the FS box where the conference was already established. If the conference is not active anywhere, start it in without redirect. Another scenario I know is using a SIP proxy (Kamailio/OpenSIPS) in front of a pool of FS boxes and monitor there where the conference was established directly proxying the call to the active one. Hope this helps! DanB

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Raspberry Pi, mod_gsmopen GSM dongle
by Sergey Zhuravlov 04 Nov '13

04 Nov '13

Hi! Does somebody have a positive experience with Raspberry Pi, mod_gsmopen GSM dongle ??? I use Raspberry Pi B and e1550. It's a shame that with an asterisk all works fine -- RasPBX. http://www.raspberry-asterisk.org/ But I prefer the FreeSWITCH ;-) On the basis of experience, I thought that PS will work best for this modest hardware faster and require fewer resources. But it turns out it is not. Asterisk with a web server, MySQL database, and other, less load than FS. In this connection, other parameters may be used starting FS to save resources? And the main question! Silence an incoming call on 5000 (IVR) as well as an outgoing in gsm network. Those are not sound. -- WBR, Sergey GTALK/JABBER:4orbit@gmail.com

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