August 2007 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] mfcr2
by Mike Murdock 11 Apr '08

11 Apr '08

What is mfcr2 used to interface with? Michael Murdock VP Operations & Development Switchmaxx Product Line 125 N. Market. Suite 1520 Wichita KS. 67202 (316) 932-2693 -----Original Message----- From: Brian West [mailto:brian.west@mac.com] Sent: Tuesday, January 30, 2007 11:46 AM Eastern Standard Time To: Yeruult[ICN]; freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] mfcr2 Do you happen to know of an open source libmfcr2 that is license compatible with FreeSWITCH? /b On Jan 29, 2007, at 7:10 AM, Yeruult[ICN] wrote: > Hello all, > > > > Does Freeswitch supports mfcr2? I know mfcr2 is old protocol but > that doesn't mean ppl don't use it. As I know there is still many > guys from everywhere. As I understood asterisk v1.4 becoming > complicated for Steve Underwood's unicall library. Is there any > other open sources which supports mfcr2? > > > > -- > > Best regards, > > Yeruult[ICN] mailto:yeruult@icn-mn.net > > _______________________________________________ > 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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trying to compile freeswitch
by Mark Quitoriano 25 Sep '07

25 Sep '07

Hi list, im new to freeswitch and im trying to compile freeswitch which i got from the SVN. Im using centos 4.5 and im having a problem with libtool. The program needs 1.5.14 version but centos 4.x latest version is only 1.5.6, what should i do with this? Does it mean i need to upgrade to centos 5.xjust to make this works? is there any centos users that made this work with 4.x branch? Thanks!

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Re: [Freeswitch-users] 2 SIP Lines, Difficulty Originating Call
by Anthony Minessale 07 Sep '07

07 Sep '07

You should choose one or the other not both. One is inband ringing generated by FreeSWITCH and the other is just the sip message telling the phone to indicate ringinig. Here is a description I wrote on an earlier posting here that someone may want to stick in the wiki. ---snip ring_ready is a dialplan application that sends the protocol specific message to indicate ringing. In the case of SIP a 180 Ringing. ringback is a channel variable you can set to artificially generate a tone or play an audio file to an originating channel while it waits to be connected to another call. pre_answer is a dialplan application that will send the protocol specific message to indicate early media. In the case of SIP a 183 Progress So, if you call into freeswitch with sip and the first entry in your dialplan is ring_ready followed by a bridge to some other destination when it hits ring_ready it will send "180 Ringing" back to your phone so it can generate the ringing sound. If instead you use the set application to set ringback to a tone spec or audio file followed by a call to pre_answer to establish an early media connection followed by a call to bridge to another dest, then the core will generate this audio locally and send it back to your phone. Again, this occurs during early media meaning the call has not been answered but the SIP has negotiated a media path in advance for this type of pre-answer audio indication. SIP has no promise that early media must be supported so some switches and devices opt to not support it meaning you may not be able to hear any audio until the call is officially answered which would keep you from hearing the artificial ringback at all. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Tamas Cseke <cstomi.levlist(a)gmail.com> To: chris(a)maxpowersoft.com; freeswitch-users(a)lists.freeswitch.org Sent: Monday, August 27, 2007 2:34:22 AM Subject: Re: [Freeswitch-users] 2 SIP Lines, Difficulty Originating Call Chris Danielson írta: > If any one can help me, I am having two issues. Basically, while my > javascript is executing a single inbound SIP call the script > conditionally gets to a point where the end-user can originate a call to > an external phone number. Keep in mind that I have two sip lines > defined as gateways supplied by my carrier vonics.net. Upon calling the > originate method, the call is actually placed and the destination phone > actually begins to ring. At this point I still cannot hear the ringing > on the original session that was originated by my inbound call. Also, > the originate method times out regardless of whether or not the > destination "phone" number was answered or not. > > So my two issues are: > 1) Why does the originate method start a call, the destination phone > called actually rings, but always times out, regardless of whether or > not the phone was answered? > 2) When calling the originate method, I never hear the phone ringing on > my inbound connection. > You should call ring_ready app, if you want to hear the ringing afaik. /* set ringback tone */ session.setVariable("ringback", "%(2000, 4000, 440.0, 480.0)"); session.execute("ring_ready", ""); I found these pages, maybe you can find more about it on wiki. http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready Hope this help! Regards, Tamas _______________________________________________ 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 ____________________________________________________________________________________ Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. http://smallbusiness.yahoo.com/webhosting

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Linksys SPA942
by Jonas Gauffin 30 Aug '07

30 Aug '07

Hello Anyone got a SPA942? Or similiar phone. Under the sip settings I can specify a Proxy. The ip to my proxy (for internal testing) is 192.168.1.102. But the proxy address are also used when registering (i.e. the phone wants to register with freeswitch as jonas(a)192.168.1.102) I want it to register as jonas(a)mydomain.com. Anyone know if that is possible? Thanks, Jonas

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Astricon Meetup
by Brian West 28 Aug '07

28 Aug '07

Everyone, I will be attending Astricon in Phoenix and would like to have a little get together to discuss Open Source Telephony and the challenges we as developers and system integrators face. Exchange ideas and go over some use cases and see how we can all work together to improve our understanding of the dynamics of how everything works together. * Scaleability * Reusability of code * Standards (VoiceXML, MRCP and more) If anyone is interested please email me off list and we'll plan on having a meeting of minds. Thanks, Brian West FreeSWITCH.org

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FreeSwitch New Users.. Question
by ram 27 Aug '07

27 Aug '07

Hi I was going the documents of Freeswitch i have not found the real document what iam looking for. I have already setup openser + mediaproxy + radius + mysql ( modular setup) is the Freeswitch replace above setup ? Like call routing, media handling, b2b , cdr ? if yes, can some one point to right URL to read and deployments what is user experince maximum concurrent calls achived. what hardware is the best to achive ? more than 1000 concurrent calls with free switch. suggestions are welcome ram

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$100 bounty for VoIP client on iPhone
by Jason Garland 27 Aug '07

27 Aug '07

I saw this on the iPhone forums... http://hackint0sh.org/forum/showthread.php?t=2211

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2 SIP Lines, Difficulty Originating Call
by Chris Danielson 27 Aug '07

27 Aug '07

If any one can help me, I am having two issues. Basically, while my javascript is executing a single inbound SIP call the script conditionally gets to a point where the end-user can originate a call to an external phone number. Keep in mind that I have two sip lines defined as gateways supplied by my carrier vonics.net. Upon calling the originate method, the call is actually placed and the destination phone actually begins to ring. At this point I still cannot hear the ringing on the original session that was originated by my inbound call. Also, the originate method times out regardless of whether or not the destination "phone" number was answered or not. So my two issues are: 1) Why does the originate method start a call, the destination phone called actually rings, but always times out, regardless of whether or not the phone was answered? 2) When calling the originate method, I never hear the phone ringing on my inbound connection. Thank you in advance for any light you can shine on my issues. Kind Regards, Chris Danielson freeswitch@MaxPowerSoft001> version FreeSwitch Version 1.0.pre1 (5644M) (I just downloaded and compiled this version last week from subversion) ===================== MaxPowerSoft001:/usr/local/freeswitch/conf# uname -a Linux MaxPowerSoft001 2.6.18-4-amd64 #1 SMP Mon Mar 26 11:36:53 CEST 2007 x86_64 GNU/Linux ===================== /* SNIPPET of JavaScript that is supposed to bridge and inbound call with a newly created outbound call. */ var str = "sofia/gateway/line2/"+phone; var ns = new Session(); if (ns.originate(session, str, 30)) { //this actually places a call if (session.ready() && ns.ready()) { console_log("notice", "bridging call\n"); bridge(session, ns); } } else { console_log("notice", "originate failed.\n"); }  <gateways> <gateway name="sip.vonics.net"> <param name="username" value="1231234567"/> <param name="realm" value="sip.vonics.net:13412"/> <param name="password" value="xxxxxxxxxxxxxxx"/> <param name="extension" value="1231234567"/> <param name="expire-seconds" value="300"/> </gateway> <gateway name="line2"> <param name="username" value="1231234567"/> <param name="realm" value="sip.vonics.net"/> <param name="password" value="xxxxxxxxxxxxxxx"/> <param name="extension" value="2231234567"/> <param name="expire-seconds" value="300"/> <param name="register" value="false"/> </gateway> </gateways>  <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="13412"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${global_codec_prefs}"/> <param name="codec-ms" value="20"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="$${bind_server_ip}"/> <param name="sip-ip" value="$${bind_server_ip}"/> <param name="accept-blind-reg" value="true"/> <param name="nonce-ttl" value="60"/> <param name="ext-sip-ip" value="my.dyndns.org"/> </settings> LOG Output with export SOFIA_DEBUG=9 tport_vsend returned 1237 nta: resent INVITE (87950735) to udp/64.157.86.41:5060 tport_release(0x560f80): 0x5d03b0 by 0x5eac00 with (nil) tport_pend(0x560f80): pending 0x5d03b0 for udp/10.0.0.2:13412 (already 2) nta_outgoing_timer: 1/1 resent, 0/1 tout, 0/1 term, 0/4 free nta: timer set next to 1000 ms nta: timer A fired, retransmit INVITE (87950735) tport_tsend(0x560f80) tpn = udp/64.157.86.41:5060 tport_resolve addrinfo = 64.157.86.41:5060 tport_by_addrinfo(0x560f80): not found by name udp/64.157.86.41:5060 tport_vsend returned 1237 nta: resent INVITE (87950735) to udp/64.157.86.41:5060 tport_release(0x560f80): 0x5d03b0 by 0x5eac00 with (nil) tport_pend(0x560f80): pending 0x5d03b0 for udp/10.0.0.2:13412 (already 2) nta_outgoing_timer: 1/1 resent, 0/1 tout, 0/1 term, 0/4 free nta: timer set next to 2000 ms tport_wakeup_pri(0x560f80): events IN tport_recv_event(0x560f80) tport_recv_iovec(0x560f80) msg 0x5ebbb0 from (udp/10.0.0.2:13412) has 573 bytes, veclen = 1 tport_deliver(0x560f80): msg 0x5ebbb0 (573 bytes) from udp/64.157.86.41:13412/sip next=(nil) nta: received 100 trying -- your call is important to us for INVITE (87950735) nta: 100 trying -- your call is important to us is going to a transaction nta_outgoing: RTT is 1650.79 ms tport_release(0x560f80): 0x5d03b0 by 0x5eac00 with 0x5ebbb0 (preliminary) (SNIP) tport_wakeup_pri(0x560f80): events IN tport_recv_event(0x560f80) tport_recv_iovec(0x560f80) msg 0x5ebbb0 from (udp/10.0.0.2:13412) has 837 bytes, veclen = 1 tport_deliver(0x560f80): msg 0x5ebbb0 (837 bytes) from udp/64.157.86.41:13412/sip next=(nil) nta: received 180 Ringing for INVITE (87950735) nta: 180 Ringing was discarded nta: timer set next to 28460 ms (ABOVE REPEATS...) nta: 180 Ringing was discarded 2007-08-26 07:22:51 [NOTICE] switch_ivr_originate.c:805 switch_ivr_originate() Hangup sofia/sip.vonics.net/18001234567 [CS_HOLD] [NO_ANSWER] 2007-08-26 07:22:51 [DEBUG] switch_channel.c:1076 switch_channel_perform_hangup() Kill sofia/sip.vonics.net/18001234567 [KILL] 2007-08-26 07:22:51 [DEBUG] switch_core_session.c:647 switch_core_session_signal_state_change() Kill sofia/sip.vonics.net/18001234567 [BREAK] 2007-08-26 07:22:51 [DEBUG] switch_ivr_originate.c:878 switch_ivr_originate() Originate Cancelled by originator termination Cause: 19 [NO_ANSWER] 2007-08-26 07:22:51 [WARNING] mod_spidermonkey.c:2361 session_originate() Cannot Create Outgoing Channel! [sofia/gateway/line2/18001234567] 2007-08-26 07:22:51 [NOTICE] myjavascript.js:24 console_log() originate failed. nua: nua_cancel: entering nua(0x67bf60): signal r_cancel 2007-08-26 07:22:51 [DEBUG] switch_core_state_machine.c:347 switch_core_session_run() (sofia/sip.vonics.net/18001234567) State HANGUP 2007-08-26 07:22:51 [DEBUG] mod_sofia.c:217 sofia_on_hangup() Channel sofia/sip.vonics.net/18001234567 hanging up, cause: NO_ANSWER 2007-08-26 07:22:51 [DEBUG] mod_sofia.c:242 sofia_on_hangup() Sending CANCEL to sofia/sip.vonics.net/18001234567 2007-08-26 07:22:51 [DEBUG] switch_core_state_machine.c:45 switch_core_standard_on_hangup() Standard HANGUP sofia/sip.vonics.net/18001234567, cause: NO_ANSWER 2007-08-26 07:22:51 [DEBUG] switch_core_session.c:706 switch_core_session_thread() Session 6 (sofia/sip.vonics.net/18001234567) Locked, Waiting on external entities 2007-08-26 07:22:51 [INFO] switch_core_session.c:712 switch_core_session_thread() Session 6 (sofia/sip.vonics.net/18001234567) Ended 2007-08-26 07:22:51 [NOTICE] switch_core_session.c:714 switch_core_session_thread() Close Channel sofia/sip.vonics.net/18001234567 [CS_HANGUP] nua: nua_stack_set_params: entering soa_set_params(static::0x604d40, ...) called tport(0x560f80): EXPENSIVE unresolved udp/64.157.86.41:5060 tport_tsend(0x560f80) tpn = udp/64.157.86.41:5060 tport_resolve addrinfo = 64.157.86.41:5060 tport_by_addrinfo(0x560f80): not found by name udp/64.157.86.41:5060 tport_vsend returned 325 nta: sent CANCEL (87950735) to udp/64.157.86.41:5060 tport_pend(0x560f80): pending 0x5ec4d0 for udp/10.0.0.2:13412 (already 3) nta: timer shortened to 500 ms

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install requirements
by Josip Djuricic 24 Aug '07

24 Aug '07

Hi all, it would be nice if someone would wrote all of the prerequisite package's for *nix install of freeswitch, I'll probbably write it myself when I figure out all of the prerequisites (but it will take some time, cause I have no time atm). It's not heavy to install freeswitch but it would definetely make things easier for new folks that will sure come. J.

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Using FreeSwitch with OpenSbc as Proxy
by Yogesh Chaturvedi 24 Aug '07

24 Aug '07

Hi, I am configuring setting in that prototype freeswitch is running behind the sip-proxy and i am using OpenSbc as sip Proxy. My sip service Provider is Gafachi : I am able to register with gafachi for inbound/outbond call if i am not using OpenSbc but face problem when i give the proxy as ip on which OpenSbc Is running: Gateway setting i am using is that: <gateways> <gateway name="192.168.96.9"> /// account username *required* /// <param name="username" value="UserName"/> /// auth realm: *optional* same as gateway name, if blank /// <param name="realm" value="gafachi"/>  /// account password *required* /// <param name="password" value="password"/> /// extension for inbound calls: *optional* same as username, if blank /// <param name="extension" value="1020"/> /// proxy host: *optional* same as realm, if blank /// <param name="proxy" value="192.168.96.115"/> //Ip address on which OpenSbc running: /// expire in seconds: *optional* 3600, if blank /// <param name="expire-seconds" value="600"/> </gateway> </gateways> As far as i know realm create the address for where request should go. and in my case it should be : username@gafachi and proxy set from where request go i am giving it as machine ip where OpenSBC is running: Problem is that when i give proxy as 192.168.96.115: It make request as UserName(a)192.168.96.115 instead of UserName@gafachi Can you guide me where i am wrong : Thanks in advance Yogesh

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