April 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Question about incoming calls.
by Javier Aristizábal 05 Apr '10

05 Apr '10

Hi Folks. I have an incoming TFN into my FreeSWITCH box, and i have two trunks.. Is it possible to send the 50% of the calls to one trunk and the other 50% of the calls to the other trunk? Many thanks -- -.-- --- / ... --- -.-- Javier Aristizábal

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Automatic IP Routing Table Updates From Freeswitch?
by Jerry Richards 05 Apr '10

05 Apr '10

We are running Freeswitch on a CentOS 5.3 machine with two network ports connected to two different networks. Is there an example of Freeswitch dynamically routing SIP traffic (i.e. update CentOS IP Routing Table?) based on which sip_profile the extension is REGISTERing with. This would apply to registration and call routing (and RTP packets). Best Regards, Jerry

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How to Disable FS Session Timers?
by Jerry Richards 05 Apr '10

05 Apr '10

How do I disable session timers in Freeswitch? Currently, FS is including "Supported: timer" during call establishment. Is there an FS tag to disable this? Best Regards, Jerry

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No audio/dtmf from softphone behind NAT
by Fraser Redmond 05 Apr '10

05 Apr '10

I've got a FreeSwitch server up on Amazon EC2, ports wide open for my office external-IP, server iptables disabled, and changed the FreeSwitch ACL domains to "allow", so it's all wide open for now. In the office I'm trying to connect to the server from Bria/X-lite. I've entered a stun server (stun.freeswitch.org) and I can now call to the server, but not from the server. I read this page: http://wiki.freeswitch.org/wiki/Nat_stun_debug_irc which suggested adding this variable to the user config: http://wiki.freeswitch.org/wiki/Natted_Softphone_ATA#NDLB-connectile-dysfun… With that on I can now call to and from the server. However with or without that although I can hear audio from the server, audio to the server isn't arriving (doesn't appear in recordings), and dtmf doesn't get received either. When I hang up from the client, I see in the CLI that it gets that instruction, so it hasn't started the call and lost all contact with the softphone, it's receiving some instructions, but not the audio and dtmf. The problem is that both the server and client are each behind NAT, so either could be having the problem (on EC2 the auto-NAT doesn't work, so I've specified the external rtp and sip ip's.. I've also turned on aggressive-NAT in case that helps. Also I'm connecting to the server by a sub-domain (A-name) rather than IP.) I've got almost the same setup working fine on the internal network (same dialplan and directory, and all the config is the same if it can be), so it's got to be something to do with the NAT's. Any suggestions on what the problem might be, or how to find it? Cheers, Fraser

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Openzap extension can't use outside lines.
by Jim 05 Apr '10

05 Apr '10

I obviously need to set a <variable name="toll_allow" value="domestic,international,local"/> somewhere but I can't figure out what file to put it in. The examples all show it in the directory/default/xxxx.xml files but those appear to be sip only. In any event creating files in that directory for my extension did nothing to help the problem. The only places I have the extension mentioned is in the openzap.conf file and dialplan/default/00_incoming-1.xml. Adding a <variables ... element to the latter does nothing. How do I set that variable? Or where? Thanks, Jim.

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Re: [Freeswitch-users] domain-wise context
by Jayesh Nambiar 05 Apr '10

05 Apr '10

> > Hi, > Now i know, why user_context is not working for me. My calls are coming from Opensips and are not authenticated by Freeswitch. --- Jayesh > > ---------- Forwarded message ---------- > From: Ognjen Seslija <oseslija(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Mon, 5 Apr 2010 13:14:41 +0200 > Subject: Re: [Freeswitch-users] domain-wise context > user_context applies to authenticated users. That means if a user > successfully gets its INVITE authenticated either by IP or SIP auth method > its call will end up in ${user_context}. > > Ognjen > > On Mon, Apr 5, 2010 at 7:43 AM, Jayesh Nambiar <jayesh.voip(a)gmail.com>wrote: > >> Hi Brian, >> I've tried user_context variable. Does user_context apply to only >> registered users in freeswitch? >> Because in my case the users are registered on a different >> software(OSips). Also out of curiosity, does the OSips needs to be defined >> as a Gateway in FS for handling the calls in a standard fashion? >> >> Thanks, >> >> --- Jayesh >> >> >>> ---------- Forwarded message ---------- >>> From: Brian West <brian(a)freeswitch.org> >>> To: freeswitch-users(a)lists.freeswitch.org >>> Date: Sat, 3 Apr 2010 22:57:19 -0500 >>> Subject: Re: [Freeswitch-users] domain-wise context >>> see user_context variable on the user. >>> >>> /b >>> >>> On Apr 3, 2010, at 10:47 PM, Jayesh Nambiar wrote: >>> >>> > Hi, >>> > Sorry for sounding so impatient, the anxiety only grew because before >>> posting it to the list I spent a week on reading all the documentation >>> available on and around this topic FS site and mailing lists. I really >>> appreciate and am thankful for your suggestions. >>> > I'll try out the suggestions given by you. Is there a way that we can >>> compare the domain name in the dialplan using regular expressions. Is there >>> a value in condition tag that can be used to compare this, something like >>> "extension_number". >>> > Or should I store the domain value in some variable(something like >>> sip_h_<domain_name>) and compare that variable to take the call to a >>> different context? >>> > >>> > Thanks, >>> > >>> > --- Jayesh >>> >> >> _______________________________________________ >> 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 >> >> > > _______________________________________________ > 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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Re: [Freeswitch-users] domain-wise context
by Jayesh Nambiar 05 Apr '10

05 Apr '10

Hi Brian, I've tried user_context variable. Does user_context apply to only registered users in freeswitch? Because in my case the users are registered on a different software(OSips). Also out of curiosity, does the OSips needs to be defined as a Gateway in FS for handling the calls in a standard fashion? Thanks, --- Jayesh > ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: freeswitch-users(a)lists.freeswitch.org > Date: Sat, 3 Apr 2010 22:57:19 -0500 > Subject: Re: [Freeswitch-users] domain-wise context > see user_context variable on the user. > > /b > > On Apr 3, 2010, at 10:47 PM, Jayesh Nambiar wrote: > > > Hi, > > Sorry for sounding so impatient, the anxiety only grew because before > posting it to the list I spent a week on reading all the documentation > available on and around this topic FS site and mailing lists. I really > appreciate and am thankful for your suggestions. > > I'll try out the suggestions given by you. Is there a way that we can > compare the domain name in the dialplan using regular expressions. Is there > a value in condition tag that can be used to compare this, something like > "extension_number". > > Or should I store the domain value in some variable(something like > sip_h_<domain_name>) and compare that variable to take the call to a > different context? > > > > Thanks, > > > > --- Jayesh >

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Re: [Freeswitch-users] Some question about mod_fifo ??
by Nguyễn Mạnh Hùng 05 Apr '10

05 Apr '10

Hi Anthony, Can you discuss some details in how polycom or snom can do this and x-lite not. If can, I want to edit some open source soffphone like officeSIP to do this. Best Regards. Anthony Minessale [anthony.minessale(a)gmail.com] We already do it. X-Lite does not support it. If you try it with a phone like snom or polycom you will see it works just like that. 2010/4/3 Nguy�n Mạnh Hùng < hungngm(a)bkav.com.vn > Hi Seven Du. Thanks to yours suggetion. I have an ideal, it is: when the call between caller and agent is set, the caller_id is determined. So, i want to edit code to sent the agent information (the call_id and call_id_number) which will be displayed againt in the agent's softphone (as Xlite..) when the call is happening. I read some documents but i still can't determine: It's maybe yes or maybe to do this and where to do this. Can you give me some comments. Best Regard. Seven Du [ dujinfang(a)gmail.com ] ��As discussed in the list, it's not a freeswitch problem but a reality of life. Think about customer A and B calls in one after another, then if FreeSWITCH call agent X with caller id A and Y with caller id B, and angent Y answers before X, then 1) if bridge Y with A with the FIFO rule, then the caller id is wrong 2) if bridge Y with B, the caller id is right but it breaks the rule of FIFO - A should be served before B!! And what even worse is that if X never answer A then A never can be served which is really unfair!! Of course you don't want 1), and you don't need mod_fifo if you want behavior 2), you just need some dialplan trick or some simple Lua script I think. Also FreeSWITCH is designed to be easily extended with almost any languages so feel free to implement anything. 2010/3/31 Nguy�n Mạnh Hùng : > Hi Mike and Seven Du. > Thanks to yours help. > I known the mechanism of mod_fifo. >>> http://lists.freeswitch.org/pipermail/freeswitch-users/2009-November/050175… . > What a pity, It can't solve this problem. I can't use freeswitch for my call > center. > Hope new version can solve this !!! _______________________________________________ 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 _______________________________________________ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

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azdirectroute.com == SPAMMERS
by Brian West 05 Apr '10

05 Apr '10

Anyone else get spam from them? /b

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Freeswitch for Windows - command history too short?
by Frank Church 04 Apr '10

04 Apr '10

I am using svn 17048 for Windows and the command history is too short and doesn't persist between restarts. Are there some configuration settings to fix that? -- Frank Church ======================= http://devblog.brahmancreations.com

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