December 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH 1.0.5 is (almost) here!
by Michael Collins 08 Dec '09

08 Dec '09

Greetings, The FreeSWITCH developers have uploaded the latest and greatest FreeSWITCH 1.0.5 pre-release version. Please check out the release announcement<http://www.freeswitch.org/node/220>. Let's all get updated as soon as possible. Also, please report bugs right away and follow up when the developers need further information. We have had to close out some bugs due to lack of information from the one reporting. Of course, those running SVN trunk are asked to do a "make current" as soon as reasonably possible. The devs love it when you are on the latest trunk. :) Thanks again for all of your help! Let's keep up the good work and we'll have 1.0.5 available in no time. -Michael

2 1

100% CPU Usage when Asterisk 1.6.0.19 Registers
by Spencer Thomason 08 Dec '09

08 Dec '09

Hi I'm trying to use Freeswitch as a SBC for a few Asterisk boxes. The Asterisk boxes are individual hosted PBXs but they are configured with identical software. This a x86_64 CentOS 5.4 system. I've tried 1.0.4 and the latest svn with the same results. Basically Freeswitch registers with outbound providers and I can send and receive test calls. Then without warning, i.e. the Asterisk boxes are all idle and there are no calls, the Freeswitch process starts using 100% of the cpu.

3 4

presense. freeswitch vs. spa962+932
by Vladimir Elizarov 08 Dec '09

08 Dec '09

Hello, I tune in the presence freeswitch and linksys spa962 932 I had a few qusteions: 1) if $ PROXY specified domain name and not ip the phone records. But all the buttons on spa932 blinking orange indicating that no subscriptions. phone logs like this: Call-ID: 76e0f816-9617ab46(a)192.168.0.100 User: 100(a)192.168.50.10 Contact: "user" <sip:100@192.168.0.100:1024;fs_nat=yes;fs_path=sip:100@192.168.0.100:1024> Agent: Linksys/SPA962-6.1.3 (a) Status: Registered (UDP-NAT) (unknown) EXP (2009-12-08 21:26:14) Host: pbx0.test.lan IP: 192.168.0.100 Port: 1024 Auth-User: 100 Auth-Realm: pbx0.test.lan MWI-Account: 100(a)192.168.50.10 while the phone is not a nat. spa932 shows that subscriptions present. 2) how to see that now there is a basis of presence of fs_cli? 3) Can I configure two fs a mutually shared presence? This is done using <param name="presence-hosts" value="$${domain}"/>? Thabks.

1 0

Skype SIP Beta
by Chris Fowler 08 Dec '09

08 Dec '09

Skype have opened their beta program up to all comers. http://www.skype.com/business/products/pbx-systems/sip/support-faqs/#padded… Three lines in a sip_profile make FreeSWITCH talk nicely; but using the PCMU codec. Any progress on SILK native support? Last I saw was discussion back in September with Brian lamenting that Skype was hard to work with on this. I know I could use mod_skypiax; but having a native solution would be one less IT headache. Thx, Chris.

8 13

Mutual Registration of servers
by Samuel Abekah-Mensah 08 Dec '09

08 Dec '09

Pardon me if this has been addressed already. How does one go about having in the simplest instance 2 servers registering with each other on startup whereby the users registering would be able to call each other. The 2 servers are in different domains. Thanks.

3 3

Force presence status manually
by Peter P GMX 08 Dec '09

08 Dec '09

Hello, is there a way to manually force a presence status update? In our scenario we have a Freeswitch cluster. As phones sometimes register on one and one time on another machine via the load balancer, we cannot dial via user/exten. Instead we dial each phone by it's register string via xml-curl. That way -when a phone is called - other phones who subscribed to this phone, do not receive a message to update their presence status. Is there a way to force the pesence status of a phone manually in the dialplan? We may then set the status before bridging and then reset it with a hangup hook. Best regards Peter

2 1

Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP
by Jerry Richards 08 Dec '09

08 Dec '09

When I got the latest trunk the make gets an error. Should I perhaps disable the mod_amr? making all mod_amr make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. Stop The method I used to get the latest trunk follows: svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch Best Regards, Jerry _____ From: Jerry Richards [mailto:jerry.richards@teotech.com] Sent: Monday, December 07, 2009 7:44 AM To: 'Michael Jerris'; 'freeswitch-users(a)lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. Best Regards, Jerry _____ From: Michael Jerris [mailto:mike@jerris.com] Sent: Saturday, December 05, 2009 7:30 PM To: Jerry Richards Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Jerry- Any update on this? Mike On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org <http://pastebin.freeswitch.org/> On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards <jerry.richards(a)teotech.com> wrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing <enable-3pcc> to "true" and also "proxy", but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry _______________________________________________ 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 <http://www.freeswitch.org/> -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 _______________________________________________ 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

3 2

Debugging reeswitch (especially TLS)
by Yehavi Bourvine 08 Dec '09

08 Dec '09

Hello, I have some black hole understading how to debug Freeswitch. In fs_cli I do "sofia debug all 7" and indeed get a lot of debugging messages on the console; however, the logfiles get only Critical messages. Where do I define which messages go to the logfile? And in a related topic: I've set a Polycom to use TLS with Freeswitch. I see it contacts FS on TCP port 5061, do some exchange, and then quits and does not use TLS. How do I debug TLS from FS side? Thanks! __Yehavi:

1 0

OpenZap issues with incoming and outgoing calls
by Jingwei Yang 08 Dec '09

08 Dec '09

Hello All, I have a Digium TDM400P pci card with two FXO ports installed on my linux box. I've connected an external telephone line to the first FXO port. But I can't either make outgoing calls or receive incoming ones. Here are my setups, please let me know where goes wrong. * /etc/zaptel.conf* loadzone = sg defaultzone=sg fxsks=1,2 */usr/local/freeswitch/conf/zt.conf* remains unchanged [defaults] codec_ms => 20 wink_ms => 150 flash_ms => 750 echo_cancel_level => 64 rxgain => 0.0 txgain => 0.0 */usr/local/freeswitch/conf/openzap.conf* [span zt] name => OpenZAP number => 1 fxo-channel => 1 [span zt] name => OpenZAP number => 2 fxo-channel => 2 */usr/local/freeswitch/conf/autoload_configs/openzap.conf.xml* <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/> </settings>  <analog_spans> <span id="1"> <param name="tonegroup" value="sg"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> <span id="2"> <param name="tonegroup" value="sg"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="1"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </analog_spans> </configuration> I defined an extension in dialplan/default.xml to receive bridge incoming calls to my skype instance. Frankly speaking, I'm not sure whether this definition is correct. How should I define the expression? When I dial the telephone number, the FS console has no response and I hear nother but busy tones. <extension name="incoming_fxo"> <condition field="destination_number" expression="^(1)$"> <action application="bridge" data="skypiax/ANY/my_skype_account"/> </condition> </extension> For outgoing calls, I tried something like this: originate openzap/1/1/xxxxxxxx &echo, while "xxxxxxxx" is my handphone number. Again, my handphone has no response. Hopefully I've explained my situation clearly. Please kindly enlighten where the problem might be. Thanks, -Jingwei p.s. here is the outgoing log trace for your reference. freeswitch(a)localhost.localdomain> originate openzap/1/1/xxxxxxxx &echo 2009-12-03 17:21:04.664276 [INFO] ozmod_zt.c:636 Setting echo cancel to 64 taps for 1:1 2009-12-03 17:21:04.664276 [DEBUG] mod_openzap.c:366 Set codec PCMU 20ms 2009-12-03 17:21:04.665278 [DEBUG] mod_openzap.c:1191 Connect outbound channel OpenZAP/1:1/xxxxxxxx 2009-12-03 17:21:04.665278 [NOTICE] switch_channel.c:613 New Channel OpenZAP/1:1/xxxxxxxx [6f843194-18ce-4525-862f-f5f4e96db5eb] 2009-12-03 17:21:04.665278 [DEBUG] mod_openzap.c:1203 (OpenZAP/1:1/xxxxxxxx) State Change CS_NEW -> CS_INIT 2009-12-03 17:21:04.665278 [DEBUG] switch_core_session.c:999 Send signal OpenZAP/1:1/xxxxxxxx [BREAK] 2009-12-03 17:21:04.665278 [DEBUG] ozmod_analog.c:59 Changing state on 1:1 from DOWN to DIALING 2009-12-03 17:21:04.665278 [DEBUG] ozmod_analog.c:279 ANALOG CHANNEL thread starting. 2009-12-03 17:21:04.665278 [INFO] ozmod_zt.c:636 Setting echo cancel to 64 taps for 1:1 2009-12-03 17:21:04.665278 [DEBUG] ozmod_analog.c:450 Executing state handler on 1:1 for DIALING 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/xxxxxxxx) Running State Change CS_INIT 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/1:1/xxxxxxxx) State INIT 2009-12-03 17:21:04.665278 [DEBUG] mod_openzap.c:390 (OpenZAP/1:1/xxxxxxxx) State Change CS_INIT -> CS_ROUTING 2009-12-03 17:21:04.665278 [DEBUG] switch_core_session.c:999 Send signal OpenZAP/1:1/xxxxxxxx [BREAK] 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/1:1/xxxxxxxx) State INIT going to sleep 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/xxxxxxxx) Running State Change CS_ROUTING 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/1:1/xxxxxxxx) State ROUTING 2009-12-03 17:21:04.665278 [DEBUG] mod_openzap.c:413 OpenZAP/1:1/xxxxxxxx CHANNEL ROUTING 2009-12-03 17:21:04.665278 [DEBUG] switch_ivr_originate.c:66 (OpenZAP/1:1/xxxxxxxx) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-12-03 17:21:04.665278 [DEBUG] switch_core_session.c:999 Send signal OpenZAP/1:1/xxxxxxxx [BREAK] 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/1:1/xxxxxxxx) State ROUTING going to sleep 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/xxxxxxxx) Running State Change CS_CONSUME_MEDIA 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/1:1/xxxxxxxx) State CONSUME_MEDIA 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/1:1/xxxxxxxx) State CONSUME_MEDIA going to sleep 2009-12-03 17:21:34.114940 [DEBUG] skypiax_protocol.c:141 rev 15765M[(nil)|37 ][DEBUG_SKYPE 141 ][skypiax8 ][-1, 0, 0] READING: |||USER amanda8884 PHONE_HOME ||| 2009-12-03 17:21:34.114940 [DEBUG] skypiax_protocol.c:141 rev 15765M[(nil)|37 ][DEBUG_SKYPE 141 ][skypiax8 ][-1, 0, 0] READING: |||USER amanda8884 PHONE_OFFICE ||| 2009-12-03 17:21:34.114940 [DEBUG] skypiax_protocol.c:141 rev 15765M[(nil)|37 ][DEBUG_SKYPE 141 ][skypiax8 ][-1, 0, 0] READING: |||USER amanda8884 PHONE_MOBILE ||| 2009-12-03 17:21:34.684709 [DEBUG] ozmod_analog.c:340 Changing state on 1:1 from DIALING to BUSY 2009-12-03 17:21:34.704705 [DEBUG] ozmod_analog.c:450 Executing state handler on 1:1 for BUSY 2009-12-03 17:21:34.704705 [DEBUG] ozmod_analog.c:579 Changing state on 1:1 from BUSY to DOWN 2009-12-03 17:21:34.724706 [DEBUG] ozmod_analog.c:450 Executing state handler on 1:1 for DOWN 2009-12-03 17:21:34.724706 [DEBUG] mod_openzap.c:1334 got FXO sig 1:1 [STOP] 2009-12-03 17:21:34.724706 [NOTICE] mod_openzap.c:1352 Hangup OpenZAP/1:1/xxxxxxxx [CS_CONSUME_MEDIA] [NORMAL_CIRCUIT_CONGESTION] 2009-12-03 17:21:34.724706 [DEBUG] switch_channel.c:1912 Send signal OpenZAP/1:1/xxxxxxxx [KILL] 2009-12-03 17:21:34.724706 [DEBUG] switch_core_session.c:999 Send signal OpenZAP/1:1/xxxxxxxx [BREAK] 2009-12-03 17:21:34.724706 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/xxxxxxxx) Running State Change CS_HANGUP 2009-12-03 17:21:34.724706 [DEBUG] switch_core_state_machine.c:459 thread mismatch skipping state handler. 2009-12-03 17:21:34.724706 [DEBUG] zap_io.c:1234 channel done 1:1 2009-12-03 17:21:34.724706 [DEBUG] ozmod_analog.c:766 ANALOG CHANNEL 1:1 thread ended. 2009-12-03 17:21:34.724706 [DEBUG] switch_core_state_machine.c:486 (OpenZAP/1:1/xxxxxxxx) State HANGUP API CALL [originate(openzap/1/1/xxxxxxxx &echo)] output: -ERR NORMAL_CIRCUIT_CONGESTION 2009-12-03 17:21:34.724706 [DEBUG] switch_ivr_originate.c:2988 Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION] 2009-12-03 17:21:34.725762 [WARNING] mod_openzap.c:474 VETO Changing state on 1:1 from DOWN to HANGUP 2009-12-03 17:21:34.725762 [DEBUG] mod_openzap.c:510 OpenZAP/1:1/xxxxxxxx CHANNEL HANGUP 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:46 OpenZAP/1:1/xxxxxxxx Standard HANGUP, cause: NORMAL_CIRCUIT_CONGESTION 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:486 (OpenZAP/1:1/xxxxxxxx) State HANGUP going to sleep 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:333 (OpenZAP/1:1/xxxxxxxx) State Change CS_HANGUP -> CS_REPORTING 2009-12-03 17:21:34.725762 [DEBUG] switch_core_session.c:999 Send signal OpenZAP/1:1/xxxxxxxx [BREAK] 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/xxxxxxxx) Running State Change CS_REPORTING 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:577 (OpenZAP/1:1/xxxxxxxx) State REPORTING 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:53 OpenZAP/1:1/xxxxxxxx Standard REPORTING, cause: NORMAL_CIRCUIT_CONGESTION 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:577 (OpenZAP/1:1/xxxxxxxx) State REPORTING going to sleep 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:327 (OpenZAP/1:1/xxxxxxxx) State Change CS_REPORTING -> CS_DESTROY 2009-12-03 17:21:34.725762 [DEBUG] switch_core_session.c:999 Send signal OpenZAP/1:1/xxxxxxxx [BREAK] 2009-12-03 17:21:34.725762 [DEBUG] switch_core_session.c:1136 Session 1 (OpenZAP/1:1/xxxxxxxx) Locked, Waiting on external entities 2009-12-03 17:21:34.725762 [NOTICE] switch_core_session.c:1154 Session 1 (OpenZAP/1:1/xxxxxxxx) Ended 2009-12-03 17:21:34.725762 [NOTICE] switch_core_session.c:1156 Close Channel OpenZAP/1:1/xxxxxxxx [CS_DESTROY] 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:423 (OpenZAP/1:1/xxxxxxxx) Running State Change CS_DESTROY 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:434 (OpenZAP/1:1/xxxxxxxx) State DESTROY freeswitch(a)localhost.localdomain> 2009-12-03 17:21:34.726741 [DEBUG] switch_core_state_machine.c:60 OpenZAP/1:1/xxxxxxxx Standard DESTROY 2009-12-03 17:21:34.726741 [DEBUG] switch_core_state_machine.c:434 (OpenZAP/1:1/xxxxxxxx) State DESTROY going to sleep

1 1

Translating DTMF from RFC2833 to INFO
by Yehavi Bourvine 08 Dec '09

08 Dec '09

Hello, I have Polycom phones which send only RFC-2833 (or inband which I dislike) and they should go out to the PSTN via a Cisco gateway. The Cisco gateway has some bug and accepts only INFO. I did a few tests: - Some of the phones are on different profile than the Cisco. On their profile I set 'dtmf-type=rfc2833' and on the Cisco's profile I set 'dtmf-type=info' and Freeswitch did the translation. All works ok... - Some of the phones are on the same profile as the Cisco, so I must set dtmf-type to rfc2833; it works with internal applications (like voicemail) but does not work through the Cisco as it misinterprets the rfc2833 Is there a way to set some variable (or a parameter to the bridge application) to do the translation? Thanks! __Yehavi:

4 7