- FreeSWITCH-users - FreeSWITCH Mailing Lists

Cullis, Tim is out of the office.
by tim.cullis＠convergys.com 05 Sep '07

05 Sep '07

I will be out of the office starting 04/09/2007 and will not return until 12/09/2007. I will respond to your message when I return. -- "NOTICE: The information contained in this electronic mail transmission is intended by Convergys Corporation for the use of the named individual or entity to which it is directed and may contain information that is privileged or otherwise confidential. If you have received this electronic mail transmission in error, please delete it from your system without copying or forwarding it, and notify the sender of the error by reply email or by telephone (collect), so that the sender's address records can be corrected."

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Is Freeswitch a good option for dynamic routing?
by Chris Ziomkowski 03 Sep '07

03 Sep '07

I need to create a simple wholesale switching application that consists of receiving a call via H.323 or SIP, running a custom algorithm and lookup to translate the incoming DNIS into a new number, and the processing this new number through a standard A-Z routing table for termination. The 2 platforms for this application I am considering are Asterisk and Freeswitch. Asterisk has the advantage that I'm familiar with its structure and configuration, therefore I can modify it and make it work quickly. The downside is it is clumsy for wholesale switching and does not scale well. Freeswitch seems to be a better bet given that I would like to scale this application to several thousand simultaneous calls, however I am under a time crunch to get a test application up and running. Can someone suggest to me if Freeswitch would be a good platform for this? If so, where would be the best place to plug in my custom translation routine? Can Freeswitch route signalling only, e.g. RTP/media goes directly from gateway to gateway without passing my network? If so, how can that be configured? I apologize for asking without completely studying all the available documentation and code, but as I said I am under pressure and I was hoping some benevolent individual would point me in the right direction to expedite this. Thank you for any assistance. Best Regards, Chris Ziomkowski

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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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Need Consulting help.
by Evelio Oliva 24 Aug '07

24 Aug '07

Hello everyone, We need to deliver a complete Triple Play Carrier Class solution. Looking for experienced professionals Consulting and/or Full time to help us with the following solutions: 1- Solution for Voip services Vonage style.(Including Users web interface), Carrier wholesale services (Wholesale termination, routes, calling cards, resellers, etc) 2- IPTV. Service Delivery platform 4- Metro-Wireless WIFI 802.11/n. 5- Unix, Customized BSD distribution for Multimedia Set-top Box with Voip/Video integrated. We are looking for consultants and/or a full time team to help us deliver the solution on time using customized open source and/or OEM elements. English/Spanish a plus. Interested parties please contact us with at evelio(a)olatelco.net. Please let us know of your experience and capabilities Evelio Oliva Riggio Manager Ola Communications LLC ____________________________________________________________________________________ Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase. http://farechase.yahoo.com/

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