January 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Nokia 6300i
by info＠freeswitch.es 13 Jan '09

13 Jan '09

Hi, i try to register my nokia 6300i sip client to freeswitch. The sip client connect to freeswitch but don't work well. There is a problem with sip NOTIFY. This is the log: nta: resent NOTIFY (109814666) to udp/190.90.122.161:4301 tport_pend(0x70dc60): pending 0x7b3cd0 for udp/209.20.72.171:5060 (already 3) nta_outgoing_timer: 1/4 resent, 0/4 tout, 0/1 term, 0/5 free nta: timer set next to 1270 ms nta: timer E fired, retransmit NOTIFY (109814655) tport_release(0x70dc60): 0x2aaab0123b20 by 0x2aaab0123ed0 with (nil) tport_tsend(0x70dc60) tpn = udp/190.90.122.161:4301 tport_resolve addrinfo = 190.90.122.161:4301 tport_by_addrinfo(0x70dc60): not found by name udp/190.90.122.161:4301 tport_vsend returned 893 nta: resent NOTIFY (109814655) to udp/190.90.122.161:4301 tport_pend(0x70dc60): pending 0x2aaab0123b20 for udp/209.20.72.171:5060 (already 3) nta_outgoing_timer: 1/4 resent, 0/4 tout, 0/1 term, 0/5 free nta: timer set next to 184 ms nta: timer K fired, terminate NOTIFY (109814654) outgoing_reclaim_all((nil), (nil), 0x422d2e30) nta_outgoing_timer: 0/4 resent, 0/4 tout, 1/1 term, 1/5 free nta: timer set next to 420 ms nta: timer E fired, retransmit NOTIFY (109814665) tport_release(0x70dc60): 0x2aaabc002b40 by 0x2aaabc0102f0 with (nil) tport_tsend(0x70dc60) tpn = udp/190.90.122.161:4301 tport_resolve addrinfo = 190.90.122.161:4301 tport_by_addrinfo(0x70dc60): not found by name udp/190.90.122.161:4301 tport_vsend returned 893 nta: resent NOTIFY (109814665) to udp/190.90.122.161:4301 tport_pend(0x70dc60): pending 0x2aaabc002b40 for udp/209.20.72.171:5060 (already 3) nta_outgoing_timer: 1/4 resent, 0/4 tout, 0/0 term, 0/4 free nta: timer set next to 1810 ms I put this line in 1000.xml (../conf/directory/default)for the NAT purpose <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> On the freeswitch server all ports are opened With Zoiper client all work fine This is the NAT configuration on nokia sip client:  <characteristic type="APPLICATION"> <parm name="APPID" value="w902E"/> <parm name="NAME" value="NAT/FW Traversal settings"/> <characteristic type="NW"> <parm name="APPREF" value="NAT_FW_settings"/> <parm name="DOMAIN" value=""/> <parm name="STUNSRVADDR" value=""/> <parm name="STUNSRVPORT" value=""/> <parm name="NATREFRESHTCP" value="30"/> <parm name="NATREFRESHUDP" value="20"/> </characteristic> <characteristic type="NATIAP"> <parm name="TO-NAPID" value="INTERNET"/> <parm name="REFRESHTCP" value="1200"/> <parm name="REFRESHUDP" value="28"/> <parm name="STUNRETRANSMIT" value="250"/> </characteristic> </characteristic> Any idea? Thank's

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SBC configuration
by Ashley van Gerven 13 Jan '09

13 Jan '09

Hi, I am interested in testing Freeswitch acting as an SBC. Is it simply a matter of configuring the dialplan correctly, using RE's so that inbound calls are just forwarded to our internal PBX and outbound calls from the PBX are forwarded to the VOIP provider? Or do I need to create an application that specifcally creates a new call and then joins the inbound and outbound calls? I haven't been able to find info on the wiki or google re. SBC setup. thanks Ash

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Re ad application not working as intended
by ahgindia 13 Jan '09

13 Jan '09

Hi all, I want have DTMF input from user without answering the call (i.e. in session progress). I used Read application for this, but it is not working as it is intended to be. It is only accepting one DTMF from the user and the channel gets hang up. Am I missing something? Here is the dialplan that I used for this: <action application="read" data="0 2 $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav res 10000 #"/> <action application="phrase" data="spell,${res}"/> The phrase application is used to check user input. -- View this message in context: http://www.nabble.com/Read-application-not-working-as-intended-tp21386369p2… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] mod_opal first unsuccessful test
by Apostolos Pantsiopoulos 13 Jan '09

13 Jan '09

Hi, Yes, openPhone is working with my soundcard. I am using it every day for testing purposes. I use the 1.8.1 version. Is there a newer version that uses OPAL? I didn't know that. Where can I get it from? Robert Jongbloed wrote: > > Hi guys, > > > > I was using the OpenPhone that you build with OPAL for my testing. So > that is identical (I think) to you. > > > > I have not (yet) do any third party client testing. > > > > Two ALERTING messages are fine, perfectly legal and OPAL can handle it. > > > > You say you can see the RTP packets flowing so that implies that the > mod_opal is actually working, so let's look somewhere else. Have you > confirmed that OpenPhone is using the sound card correctly? Made a > call between two machines JUST using OpenPhone for example? > > > > > > Robert Jongbloed > > OPAL/OpenH323/PTLib Architect and Co-founder. > > > > *From:* Anthony Minessale [mailto:anthony.minessale@gmail.com] > *Sent:* Tuesday, 13 January 2009 6:20 AM > *To:* Robert Jongbloed > *Subject:* Fwd: [Freeswitch-users] mod_opal first unsuccessful test > > > > Heh, > what client are you using in your tests that are working? > > ---------- Forwarded message ---------- > From: *Apostolos Pantsiopoulos* <regs(a)kinetix.gr <mailto:regs@kinetix.gr>> > Date: Mon, Jan 12, 2009 at 9:51 AM > Subject: [Freeswitch-users] mod_opal first unsuccessful test > To: freeswitch-users(a)lists.freeswitch.org > <mailto:freeswitch-users@lists.freeswitch.org> > > > Hi, > > I successfully compiled mod_opal using the latest svn for both opal > and ptlib as Brian suggested. > > When I try to establish a call using h323 from my openphone client > I get no audio although I can see RTP packets in both directions when I am > doing a capture. Some details : > > I am using the 11094 revision of the FS trunk. > I am using the PCMU codec. > I tried dialing the default IVR (5000) and other testing extensions > (freeswitch conference, echo test etc.) > I tried using fast start on and off , h245 tunneling on and off, h245 in > SETUP on and off. > > In my captures I have also noticed a strange behavior : FS sends to > my client 2 "alerting" packets > for no apparent reason. Could this be a cause of the problem? > > Had anyone any success with mod_opal lately? If yes, could you > please reply quoting your config > options (both on FS and on your client)? > > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs(a)kinetix.gr <mailto:regs@kinetix.gr> > ------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > <mailto:Freeswitch-users@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/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com > <mailto:MSN%3Aanthony_minessale@hotmail.com> > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > <mailto:PAYPAL%3Aanthony.minessale@gmail.com> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > <mailto:sip%3A888@conference.freeswitch.org> > iax:guest@conference.freeswitch.org/888 > <http://iax:guest@conference.freeswitch.org/888> > googletalk:conf+888@conference.freeswitch.org > <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> > pstn:213-799-1400 > -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs(a)kinetix.gr -------------------------------------------

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mod_java under MS Windows - is it possible?
by sellsky＠mail.ru 13 Jan '09

13 Jan '09

Hello, I'm new to FreeSwitch, and now I'm trying to set it up. The problem is, I want to use mod_java under MS Windows. I uncomment this line for mod_java in modules.conf.xml, and I got an error "Error Loading module W:\FreeSWITCH\mod\mod_java.dll". Yes, there is no such dll. Next I tried to compile FreeSwitch myself, and I found there is no MSVC project file for mod_java. I sure can't deal with it myself. Is it possible to make mod_java.dll? -- Best regards, sellsky mailto:sellsky@mail.ru

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recommend PSTN / SIP trunking?
by Miles Keaton 13 Jan '09

13 Jan '09

I'm using Junction Networks (http://pstn.junctionnetworks.com/) for SIP trunking of phone numbers, and I'm happy with them. Their servers are in NYC. Now I'm trying to find a similar company whose servers are based in LA or San Diego. Any recommendations?

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outbound call, new comer
by Will Smith 12 Jan '09

12 Jan '09

Hi, I am first time FS user, so it is a bit confused with all the setup. For inbound calls, I tried to add a voicepulse.xml in the sip_profiles/external with the following codes: <include> <gateway name="voicepulse"> <param name="username" value="3334445555"/> <param name="realm" value="my_sip_provider.com"/> <param name="password" value="3334445555"/> <param name="proxy" value="my_sip_provider.com"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> </gateway> <gateway name="voicepulse-backup"> <param name="username" value="3334445555"/> <param name="realm" value="my_sip_provider.com"/> <param name="password" value="3334445555"/> <param name="extension" value="3334445555"/> <param name="proxy" value="my_sip_provider.com"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> </gateway> </include> ------------ and in the conf/dialplan/default.xml file I added:  <extension name="Long Distance - voicepulse"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <action application="set" data="effective_caller_id_number=12223334444"/>   <action application="bridge" data="sofia/gateway/voicepulse/$1"/> </condition> </extension> ------------------ For inbound, I added <extension name="Voicepulse">  <condition field="destination_number" expression="3334445555">  <action application="transfer" data="1001 XML default"/> </condition> </extension> ---------- And, if I dial 3334445555 from a softphone registered with my_sip_provider, I got the message to the voice mail of 1001 - the 1001 extension does not ring. And if from 1001, I dial some real number like 18188892345, I got the error: Invalid Gateway ... Cannot create outgoing channel of type [fosia] cause: [Invalid_number_format] ... Would someone please give me some help to set this up. I am a bit confused with these. Thank you

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mod_opal first unsuccessful test
by Apostolos Pantsiopoulos 12 Jan '09

12 Jan '09

Hi, I successfully compiled mod_opal using the latest svn for both opal and ptlib as Brian suggested. When I try to establish a call using h323 from my openphone client I get no audio although I can see RTP packets in both directions when I am doing a capture. Some details : I am using the 11094 revision of the FS trunk. I am using the PCMU codec. I tried dialing the default IVR (5000) and other testing extensions (freeswitch conference, echo test etc.) I tried using fast start on and off , h245 tunneling on and off, h245 in SETUP on and off. In my captures I have also noticed a strange behavior : FS sends to my client 2 "alerting" packets for no apparent reason. Could this be a cause of the problem? Had anyone any success with mod_opal lately? If yes, could you please reply quoting your config options (both on FS and on your client)? -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs(a)kinetix.gr -------------------------------------------

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how many telephones can I drive from a linksys SPA-2100 FXS?
by freeswitch＠davidnicol.otherinbox.com 12 Jan '09

12 Jan '09

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Re: [Freeswitch-users] how many telephones can I drive from a linksys SPA-2100 FXS?
by freeswitch＠davidnicol.otherinbox.com 12 Jan '09

12 Jan '09

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