May 2010 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] FreeSWITCH Conference Call Starting Now!
by Michael Collins 26 May '10

26 May '10

C'mon over: http://wiki.freeswitch.org/wiki/FS_weekly_2010_05_26 -MC

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Re: [Freeswitch-dev] Call to registered phone behind NAT
by Oleg Khovayko 26 May '10

26 May '10

Hi, I little narrowed problem with NATed client. There is his registration: Call-ID: 41f120d4(a)192.168.1.146 User: 1012(a)192.168.1.5 Contact: "user" <sip:1012@192.168.1.146:5060;fs_nat=yes;fs_path=sip%3A1012%4091.207.244.1%3A5060> Agent: AcctonVoIP/2.5 Status: Registered(UDP-NAT)(unknown) EXP(2010-05-26 02:45:48) Host: olegh.ath.cx IP: 91.207.244.1 Port: 5060 Auth-User: 1012 Auth-Realm: olegh.ath.cx MWI-Account: 1012(a)192.168.1.5 When he calls me, conversation is OK. When I (from the local phone) call him (he is so far, and behind NAT), ring is OK, but when he picked up handset -- we both heard nothing. Looks like FS try to establish connection to IP address, assigned him behind his NAT: 192.168.1.146. FS sends ARP request to resolve this address: 6191 packets received by filter 0 packets dropped by kernel [root@olegh /usr/local/freeswitch/conf]# tcpdump host '(91.207.244.1 or 192.168.1.146)' tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on xl0, link-type EN10MB (Ethernet), capture size 96 bytes 00:49:27.911731 IP deskpro.khovayko.com.sip > 91.207.244.1.sip: SIP, length: 1339 00:49:28.299781 IP 91.207.244.1.sip > deskpro.khovayko.com.sip: SIP, length: 349 00:49:28.317358 IP 91.207.244.1.sip > deskpro.khovayko.com.sip: SIP, length: 403 00:49:40.420237 IP 91.207.244.1.sip > deskpro.khovayko.com.sip: SIP, length: 829 00:49:40.451060 IP deskpro.khovayko.com.sip > 91.207.244.1.sip: SIP, length: 368 00:49:40.691549 arp who-has 192.168.1.146 tell deskpro.khovayko.com 00:49:41.107754 arp who-has 192.168.1.146 tell deskpro.khovayko.com 00:49:42.105717 arp who-has 192.168.1.146 tell deskpro.khovayko.com 00:49:43.107377 arp who-has 192.168.1.146 tell deskpro.khovayko.com 00:49:44.106215 arp who-has 192.168.1.146 tell deskpro.khovayko.com 00:49:45.107216 arp who-has 192.168.1.146 tell deskpro.khovayko.com 00:49:46.105957 arp who-has 192.168.1.146 tell deskpro.khovayko.com 00:49:47.105975 arp who-has 192.168.1.146 tell deskpro.khovayko.com 00:49:48.105804 arp who-has 192.168.1.146 tell deskpro.khovayko.com 00:49:49.107058 arp who-has 192.168.1.146 tell deskpro.khovayko.com 00:49:50.107512 arp who-has 192.168.1.146 tell deskpro.khovayko.com 00:49:50.533533 IP deskpro.khovayko.com.sip > 91.207.244.1.sip: SIP, length: 626 00:49:50.794874 IP 91.207.244.1.sip > deskpro.khovayko.com.sip: SIP, length: 355 About SIP trace: I do not know, how to save trace to file from fs_cli. On the sccreen, there is too many text lines, they very fast gone... Brian West wrote: > You don't hav eto do this if you have upnp freeswitch will do this for you even for the media ports... I am not sure what your problem is grab the default config and use it... I know it works if you really do have upnp. > /b > > On May 24, 2010, at 8:01 PM, Oleg Khovayko wrote: > > >> It is behind UPNP, and I forwarded ports 5060-5060 (both TCP+UDP) to server. >> Also, I have forwarded ports 20000~30000, and for FS setup in the >> file autoload_configs/switch.conf.xml >> >> <param name="rtp-start-port" value="20000"/>  >> <param name="rtp-end-port" value="25000"/> >> >

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[Freeswitch-dev] FreeSWITCH Architecture
by Jan Berger 26 May '10

26 May '10

hi, I would like to address some changes to FreeSWITCH, but the challenge I face is that I am not sure where and how I should do this. The objective with this phase is to throw some ideas around and get feedback before I consolidate the architecture and move on coding. In an ordinary project it would be a few engineers that made a draft, made a presentation and evolved the draft through the discussions that followed until a majority accepted the draft. Jan

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[Freeswitch-dev] Call to registered phone behind NAT
by Oleg Khovayko 24 May '10

24 May '10

Hi, Please help. My FreeSWITCH has some mobile clients (Sipdroid and Nokia), keeps registration on the FS. See registration record samples: Call-ID: 571195090210(a)14.144.242.244 User: 1019(a)192.168.1.5 Contact: "user" <sip:1019@14.144.242.244:38592;transport=udp> Agent: Sipdroid/1.4.6 beta/T-Mobile G1 Status: Registered(UDP)(unknown) EXP(2010-05-19 16:38:41) Host: olegh.ath.cx IP: 208.54.35.47 Port: 30727 Auth-User: 1019 Auth-Realm: olegh.ath.cx MWI-Account: 1019(a)192.168.1.5 Call-ID: 30IpTq1FoIfJnxJC5ROP1HByEOig2n User: 1015(a)192.168.1.5 Contact: "user" <sip:1015@10.0.0.135;transport=TCP;fs_nat=yes;fs_path=sip%3A1015%4094.71.98.45%3A17476%3Btransport%3DTCP> Agent: Nokia RA-6 210.34.75 Status: Registered(TCP-NAT)(unknown) EXP(2010-05-19 15:57:27) Host: olegh.ath.cx IP: 94.71.98.45 Port: 17476 Auth-User: 1015 Auth-Realm: olegh.ath.cx MWI-Account: 1015(a)192.168.1.5 As you can see, for last samle (nokia) "Contact" contains line: sip:1015@10.0.0.135. In the real life, this phone NATed, and real IP (gateway's) is 94.71.98.45. Call from that phone is OK. But, when I try to call that phone from local, FS try to use wrong IP for connect: 2010-05-19 14:28:24.156706 [NOTICE] switch_channel.c:669 New Channel sofia/internal/sip:1015@10.0.0.135 [62b74c4f-7463-df11-ac63-005004c3cb7e] Of course, connection can not be established. Question: How to fix this problem? Or maybe, need to change FS, and always use REAL IP:PORT for establish outgoing call? Following - header of "sofia status profile internal": ================================================================================================= Name internal Domain Name N/A Auto-NAT true DBName sofia_reg_internal Pres Hosts Dialplan XML Context public Challenge Realm auto_from RTP-IP 192.168.1.5 Ext-RTP-IP 173.73.120.93 SIP-IP 192.168.1.5 Ext-SIP-IP 173.73.120.93 URL sip:mod_sofia@192.168.1.5:5060 BIND-URL sip:mod_sofia@192.168.1.5:5060 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM CODECS OUT G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 96 FAILED-CALLS-IN 15 CALLS-OUT 113 FAILED-CALLS-OUT 79 Registrations: =================================================================================================

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[Freeswitch-dev] Implement Javascript setGlobalVariable() optional third parameter
by David Swardstrom 24 May '10

24 May '10

I am developing FreeSwitch Javascript capability to emulate some functionality that is currently provided by a Dialogic based proprietary system running on Microsoft Windows, originally using ISDN interfaces and then ported to Sip based VoIP capability. This system uses external post-processing code that processes "event" type data for billing and other purposes and I need to provide equivalent event data. As part of this effort, I need a counter that is guarantied unique when there is a possibility that multiple calls may access the counter. The proprietary code, this unique counter is provided by using InterlockedIncrement(). Looking at the available JavaScript code, it appears that using getGlobalVariable() and setGlobalVariable() as described in the SetGlobalVar Wiki page would allow me to do this as long as I was willing to allow holes in the count. However, I decided to look at the code to see exactly what was being done to implement this. I found was that there was no code at all to implement the 3rd optional parameter. I would like to implement this capability but extend the capability to truly provide a unique counter. To accomplish this, support must be provided by the switch core. I would like to provide a new routine into switch_core.c: <Following is the text for switch_core.h> /*! \brief Conditionally add a global variable to the core \param varname the name of the variable \param value the value of the variable \param val2 the value of the variable to verify against \ If the global did not exist and val2=="", add global with value, return true \ If the global exists with the value of val2, replace it, return true \ If the global exists with a value other than val2, return false */ SWITCH_DECLARE(switch_bool_t) switch_core_set_var_conditional(_In_z_ const char *varname, _In_opt_z_ const char *value, _In_opt_z_ const char *val2); Then the code in mod_spidermonkey.c can be modified to use this new switch core capability. Note: I have implemented this in my local version and have tested it.

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[Freeswitch-dev] Problem in uuid_broadcast and uuid_displace
by Sathish Babu 21 May '10

21 May '10

Hi Brian West, I received one incoming and put an fifo (in queue) and playing music continuously using the below code. (Caller A) @con.execute("fifo", "callforce_fifo_#{@sa_park_num} out wait undef '#{music}'") In the another channel i am receiving the incoming call (Caller B) and i have bridged that call with the customer number (Caller C) using the below code. @con.execute("bridge", "sofia/gateway/junctionnetworks/#{@callout_number}") . Now Caller B and C are connected and i am transferring the Caller C to Caller A using the uuid_transfer and disconnecting the caller B @con.sendRecv("api uuid_transfer #{@gcallin_uuid} -bleg callforce_fifo_#{@gsa_id} XML public") After executing the above code Caller A and Caller C is connected and Caller B is disconnected. Caller A and Call C both can speak each other. But i need to play the message message to both Caller A and Caller C also i have the uuid of Caller A and Caller B. In order to play the message to the both the caller i tried using the uuid_broadcast and uuid_displace but the message is not played to any one of the caller. Also i could not found any error in the free switch log file. Help me to solve this issue. Tried using the Caller A UUID: @con.execute("uuid_broadcast", "#{Caller_A_UUID} playback::/opt/freeswitch/sounds/wel_lf1.wav bleg") - Not Working @con.execute("uuid_broadcast", "#{Caller_A_UUID} playback::/opt/freeswitch/sounds/wel_lf1.wav aleg") - Not Working @con.execute("uuid_broadcast", "#{Caller_A_UUID} playback::/opt/freeswitch/sounds/wel_lf1.wav both") - Not Working @con.execute("uuid_displace", "#{Caller_A_UUID} start /opt/freeswitch/sounds/wel_lf1.wav") - Not Working Tried using the Caller B UUID: @con.execute("uuid_broadcast", "#{Caller_C_UUID} playback::/opt/freeswitch/sounds/wel_lf1.wav bleg") - Not Working @con.execute("uuid_broadcast", "#{Caller_C_UUID} playback::/opt/freeswitch/sounds/wel_lf1.wav aleg") - Not Working @con.execute("uuid_broadcast", "#{Caller_C_UUID} playback::/opt/freeswitch/sounds/wel_lf1.wav both") - Not Working @con.execute("uuid_displace", "#{Caller_C_UUID} start /opt/freeswitch/sounds/wel_lf1.wav") - Not Working I need urgent help from any one to fix this issue. How to identify where is the problem? Thanks in advance Thanks and Regards, Sathishbabu

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[Freeswitch-dev] Get the uuid of bridge call
by Sathish Babu 21 May '10

21 May '10

Hi Brian West, I received one incoming and put an fifo (in queue) and playing music continuously using the below code. (Caller A) @con.execute("fifo", "callforce_fifo_#{@sa_park_num} out wait undef '#{music}'") In the another channel i am receiving the incoming call (Caller B) and i have bridged that call with the customer number (Caller C) using the below code. @con.execute("bridge", "sofia/gateway/junctionnetworks/#{@callout_number}") . Now Caller B and C are connected and i am transferring the Caller C to Caller A using the uuid_transfer and disconnecting the caller B @con.sendRecv("api uuid_transfer #{@gcallin_uuid} -bleg callforce_fifo_#{@gsa_id} XML public") After executing the above code Caller A and Caller C is connected and Caller B is disconnected. Caller A and Call C both can speak each other. But i need to play the message message to both Caller A and Caller C also i have the uuid of Caller A and Caller B. In order to play the message to the both the caller i tried using the uuid_broadcast and uuid_displace but the message is not played to any one of the caller. Also i could not found any error in the free switch log file. Help me to solve this issue. Tried using the Caller A UUID: @con.execute("uuid_broadcast", "#{Caller_A_UUID} playback::/opt/freeswitch/sounds/wel_lf1.wav bleg") - Not Working @con.execute("uuid_broadcast", "#{Caller_A_UUID} playback::/opt/freeswitch/sounds/wel_lf1.wav aleg") - Not Working @con.execute("uuid_broadcast", "#{Caller_A_UUID} playback::/opt/freeswitch/sounds/wel_lf1.wav both") - Not Working @con.execute("uuid_displace", "#{Caller_A_UUID} start /opt/freeswitch/sounds/wel_lf1.wav") - Not Working Tried using the Caller B UUID: @con.execute("uuid_broadcast", "#{Caller_B_UUID} playback::/opt/freeswitch/sounds/wel_lf1.wav bleg") - Not Working @con.execute("uuid_broadcast", "#{Caller_B_UUID} playback::/opt/freeswitch/sounds/wel_lf1.wav aleg") - Not Working @con.execute("uuid_broadcast", "#{Caller_B_UUID} playback::/opt/freeswitch/sounds/wel_lf1.wav both") - Not Working @con.execute("uuid_displace", "#{Caller_B_UUID} start /opt/freeswitch/sounds/wel_lf1.wav") - Not Working I need urgent help from any one to fix this issue. How to identify where is the problem? Thanks in advance Thanks and Regards, Sathishbabu

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[Freeswitch-dev] Problem in uuid_broadcast and uuid_displace
by Sathish Babu 21 May '10

21 May '10

Hi Brian West, I received one incoming and put an fifo (in queue) and playing music continuously using the below code. (Caller A) @con.execute("fifo", "callforce_fifo_#{@sa_park_num} out wait undef '#{music}'") In the another channel i am receiving the incoming call (Caller B) and i have bridged that call with the customer number (Caller C) using the below code. @con.execute("bridge", "sofia/gateway/junctionnetworks/#{@callout_number}") . Now Caller B and C are connected and i am transferring the Caller C to Caller A using the uuid_transfer and disconnecting the caller B @con.sendRecv("api uuid_transfer #{@gcallin_uuid} -bleg callforce_fifo_#{@gsa_id} XML public") After executing the above code Caller A and Caller C is connected and Caller B is disconnected. Caller A and Call C both can speak each other. But i need to play the message message to both Caller A and Caller C also i have the uuid of Caller A and Caller B. In order to play the message to the both the caller i tried using the uuid_broadcast and uuid_displace but the message is not played to any one of the caller. Also i could not found any error in the free switch log file. Help me to solve this issue. Tried using the Caller A UUID: @con.execute("uuid_broadcast", "#{Caller_A_UUID} playback::/opt/freeswitch/sounds/wel_lf1.wav bleg") - Not Working @con.execute("uuid_broadcast", "#{Caller_A_UUID} playback::/opt/freeswitch/sounds/wel_lf1.wav aleg") - Not Working @con.execute("uuid_broadcast", "#{Caller_A_UUID} playback::/opt/freeswitch/sounds/wel_lf1.wav both") - Not Working @con.execute("uuid_displace", "#{Caller_A_UUID} start /opt/freeswitch/sounds/wel_lf1.wav") - Not Working Tried using the Caller B UUID: @con.execute("uuid_broadcast", "#{Caller_B_UUID} playback::/opt/freeswitch/sounds/wel_lf1.wav bleg") - Not Working @con.execute("uuid_broadcast", "#{Caller_B_UUID} playback::/opt/freeswitch/sounds/wel_lf1.wav aleg") - Not Working @con.execute("uuid_broadcast", "#{Caller_B_UUID} playback::/opt/freeswitch/sounds/wel_lf1.wav both") - Not Working @con.execute("uuid_displace", "#{Caller_B_UUID} start /opt/freeswitch/sounds/wel_lf1.wav") - Not Working I need urgent help from any one to fix this issue. How to identify where is the problem? Thanks in advance Thanks and Regards, Sathishbabu

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[Freeswitch-dev] Get the uuid of bridge call
by Sathish Babu 20 May '10

20 May '10

Hi Brian West* *, Thanks for your kind reply. But i could not able to get the uuid of the bridged call i am using the below code but Other-Leg-Unique-ID returns empty. @con.execute("uuid_dump", "#{@uuid}") #Where @uuid is the uuid of the inbound call transfer_info = @con.getInfo(); transfer_uuid = transfer_info.getHeader("unique-id") other_leg_uuid = transfer_info.getHeader("Other-Leg-Unique-ID") @logger.debug "The transfer uuid is #{transfer_uuid} and other leg uuid is #{other_leg_uuid}" while executing the above code other_leg_uuid is empty. Please help me to solve this issue. Thanks and Regards, Sathishbabu

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[Freeswitch-dev] FreeSWITCH Training!
by Michael Collins 20 May '10

20 May '10

We are please to let everyone know that official FreeSWITCH training is now available! More information available here: http://www.freeswitch.org/node/259 Thanks to Darren Schreiber for organizing the first formal FreeSWITCH training course. -Michael

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