January 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Newbie question. Where to install Freeswitch?
by Brett Davidson 31 Jan '12

31 Jan '12

I am about to build myself a new PfSense box at home as I want Gigabit wirespeed routing between two subnets and then found FreeSwitch. It appears you can install FreeSwitch on pfsense and that would make sense from a consolidation and power usage factor but it doesn't seem advisable to put your Telephony app on your firewall. That a real concern or am I being a little too paranoid (I work in IT). ;-) I understand that FreeSwitch could run on an Alix 2D3 (my current PfSense machine) but am wondering if it's a little underpowered. (AMD Geode@500MHz with 256Mb RAM). I don't see much documented about resources and where to place FreeSwitch. Any recommendations? Cheers in advance, Brett.

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Newbie questions - Long time to connect
by Charles Bujold 31 Jan '12

31 Jan '12

Installing a new Freeswitch - Ubuntu (AMD Quad FX, 8 GB RAM, Intel Network card ) and I am getting several issues that hopefully somebody can point me in the correct direction. 1) I'm Using VoiceMeUp (Canada) as provider and I keep getting Gateway being Registered, Fail Wait and Trying after the first call is received or outgoing. If I do not use the line it shows Registered. After every call it seems to take several minutes before I can make another call. 2) I cannot seem to be able to make more than 1 call at a time although the service can support 5 calls simultaneously. 3) When making a call I see the connection appear in Freeswitch but it can take over a minute before I hear a ring . Some calls work and other do not. 4) When calling myself using the outside number I can see the call originating and then being received again (almost instantaneous) by Freeswitch but it still takes close to a minute before it starts ringing and when I pick up it hangs up immediately. ( I presume that a time out occurred. My question is what could cause this type of slow response to connect or answer a call. Is there a timer setting that needs to be adjusted? I have a first server similar in configuration and it works properly. I presume that I missed configured something but can't seem to figure out what? Thanks CJB

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Remote Extension
by Bob Coleman 31 Jan '12

31 Jan '12

Hi, I have a remote extension( Cisco SPA504G ) that connects to our FreeSWITCH server across the internet. I am using a typical double nat scenario as per wiki and the remote extension registers properly against the right profile. The remote extension via a dialplan can dial any of the extensions that are local to the FS boxes network successfully with no problems However when we dial out to the remote extension we get a "Cannot locate any authentication on credentials to complete an authentication request for realm '<our external domain>' I have tried using both the default force register options and also setting up another directory realm as well. When looking at the log you see the originate attempting to connect to the appropriate ip/port of the external phone, but after it "entering state [calling]" it fails with the above error. Just to test I have installed the 3cx softphone client on the remote network, and it can be dialled with no problems. The difference being that the 3cx seems to register a port in the dynamic range (eg 52789) where as the cisco phone uses the 5090 port Has anyone struck that issue before Thanks Bob

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ACLs / changing to which IPs FS binds to
by georg＠riseup.net 31 Jan '12

31 Jan '12

Hi all, I've got a server running FS with five nets associated. There are just two, from where I receive calls and my phones are registering. I would like to exclude all the nets by default from being allowed to contact / register at FS, and only allow - one net 172.251.X.XXX - one net 192.168.X.XXX I tried achieving this trough acl.conf, however, had no success. I disabled NAT at startup trough -nonat. 'sofia status profile internal' is showing me a public ip of my server next to "Pres Hosts" (but also one ip out of the mentioned 192.168.X.XXX net, which is fine). In internal.xml, I set rtp-ip and sip-ip to this (correct) ip. I think my main mistake is that I don't understand how things are handled in acl.conf. So far it looks like this: <configuration name="acl.conf" description="Network Lists"> <network-lists> <list name="localnet.auto" default="deny"> <node type="allow" cidr="192.168.X.X/24"/> <node type="deny" cidr="Public IP/29"/> </list> <list name="domains" default="deny"> <node type="allow" domain="192.168.X.XX"/> <node type="allow" cidr="192.168.X.X/24"/> </list> </network-lists> </configuration> Thanks in advance, Georg

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Re: [Freeswitch-users] Outbound Caller ID on anoymous calls
by bennygeorge 31 Jan '12

31 Jan '12

Hello all we have a situation where we receive the caller id in from as per below: INVITE sip:123456789@1.1.1.1 SIP/2.0. From: "anonymous" <sip:1.2.3.4>; when this happens, freeswitch sends the call out and inserts the caller id number of 0000000000 I have changed the vars.xml to the following: <X-PRE-PROCESS cmd="set" data="outbound_caller_name=Anonymous"/> <X-PRE-PROCESS cmd="set" data="outbound_caller_id=anonymous(a)anonymous.invalid"/> however this doesn't work. What I'm trying to achieve is that if FS doesn't receive a good caller id, then it will format it with anonymous(a)anonymous.invalid This particular carrier is not using PAI. It would be fine if FS left the from number empty - i did try changing the vars.xml to the following: <X-PRE-PROCESS cmd="set" data="outbound_caller_id=_undef_"/> but that didn't work either any help greatly appreciated as always brgds -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/RE-Outbound-Caller-ID-on-anoy… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Installing with support for Digium card ... no freetdm in modules.conf?
by Kevin White 31 Jan '12

31 Jan '12

Hi all, I apologize for such a novice question, but I'm trying to get FreeSwitch installed correctly to support my Digium analog card, and then have Plivo run on top of it. According to FreeTDM site, it says to simply uncomment the mod_freetdm line in modules.conf of freeswitch. But I don't have that line in my modules.conf. I know I could just add it, but I'm not sure if the rest of the system actually has the required stuff in it to work that way. I look in the FreeSwitch 1.0.6 book and it has little to no actual information on doing anything with FreeTDM other than that it exists and replaces OpenZAP. No actual instructions to make it work. What's the "proper" way to get freeswitch installed and working with FreeTDM for digium analog card? Thanks! Kevin ________________________________ Kevin White Senior Software Engineer (801) 727-2362 office CentralLogic www.centrallogic.com<http://www.centrallogic.com>

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Music on hold
by Darcy 31 Jan '12

31 Jan '12

When I receive a call and place it on hold, the calling party receives MOH as specified in <X-PRE-PROCESS cmd="set" data= hold_music=/usr/local/freeswitch/sounds/music/8000/moh.WAV"/>. Also, I transfer this call to another extension, the caller receives MOH during the transfer, however, If I place an outbound call then place the party on hold, they just get silence. Am I missing a basic setting here. I also cannot transfer this call or park it. Using snom 3xx, 7xx and 8xx phones. Thanks, Darcy Primrose

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CAMA Stack
by Michael Gende 31 Jan '12

31 Jan '12

I'm just starting a search on this subject, but does a stack exist for FS implementing CAMA, or at least a subset of that protocol? Thanks in Advance for any info, an email address of someone working on this now, or anyone who has considered it. Regards, Mike G.

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Call FW, redirection
by Miha Zoubek 31 Jan '12

31 Jan '12

Hi, I need a little help about understanding 302 redirect. I have set on my phone (registered on FS) redirection on number, that is not registered on FS. I have made a condition in my external dialplan. IF the number is on FS that dial to this number, if the number is not on FS, than FS should redirect to default dialplan and call outside number. I have problems with redirecting (302) to default dialplan. I have put in sip_profiles/external this <param name="manual-redirect" value="true"/> and set in my public dialplan <anti-action application="set" data="sip_redirect_contex=default" />. How to put this variables for redirect BEFORE the redirect condition is hit. You can see in mypastebin what happens. http://pastebin.freeswitch.org/18257 Tahnks! -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o.

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unable to sending and receiving fax between two peer
by amit nakum 31 Jan '12

31 Jan '12

Dear Friends Pls Help me...... I am unable to send and receiving fax directly between two zoiper peer i.e 1001(zoiper peer) --->FS---->1002(zoiper peer) using spandsp module. My actual test scenario is,I dial 1002 from 1001 with selecting some.tiff file,and fax should be send to 1002.but this scenario is not working in zoiper. I also try another scenario.By dialing 555 and try to set some T.38 variable and then bridge the call to 1002 for sending fax. <extension name="555" continue=true> <condition field="destination_number" expression="^(555)$"> <action application="answer"/> <action application="set" data="fax_enable_t38=true"/> <action application="export" data="fax_enable_t38_request=true"/> <action application="set" data="execute_on_answer=t38_gateway self nocng"/> <action application="bridge" data="user/1001"/> </condition> </extension> but this scenario is also not working. pls help me resolved my problem soon. Thanks Amit Nakum

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