September 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Cluecon Videos
by Giovanni Maruzzelli 29 Sep '10

29 Sep '10

Hi there, As Claudio Cavalera posted in another thread, would be really nice to have a torrent for downloading videos and presentations of Cluecon 2010 (a torrent for Cluecon 2009 is available). To whom is concerned: please, pleeease, pretty pleeeease! :) -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

4 3

FreeSwitch - mod_skypopen has no read codec or channel not ready
by Shamun toha md 28 Sep '10

28 Sep '10

Hello, This is another strange error i catch by screening it. Materials: =========== 1. Switch 1 (30 channels as main simulation server) 2. Switch 2 (10 channels connects with in/out audio) Step 2.a: 6 times originate skypopen/ANY/skypeswitch1 9999 3. Another PC running skype alone itself as a consumer test, to see what happens in reality traffic. Step 3.a: 1 times with real skype call skypeswitch1 and listen music quality 4. Medium Flood to switch 1 using remaining switch 2 channels Note: When 7 channels are busy for 2 hours. And Problem 1 gets started, the sound gets drop, and start latency, CPU gets overflow, and in the voice there is a wind coming from left (like a storm/cyclone started). Hardware status only for 7 active channels becomes as following: =============================================== top - 13:15:44 up 3 days, 16:12, 2 users, load average: 13.74, 10.06, 5.64 Tasks: 178 total, 3 running, 175 sleeping, 0 stopped, 0 zombie *Cpu(s): 70.4%us, 13.6%sy, 0.0%ni, 15.5%id, 0.2%wa, 0.0%hi, 0.3%si, 0.0%st* Mem: 8177580k total, 3371388k used, 4806192k free, 201836k buffers Swap: 2096472k total, 0k used, 2096472k free, 1283920k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 18719 root 18 0 434m 96m 6628 S 247.4 1.2 10:59.05 freeswitch 23094 root 15 0 93316 30m 3552 S 10.3 0.4 17:35.42 skype 22932 root 15 0 93568 30m 3556 S 9.0 0.4 20:43.74 skype 22706 root 15 0 92240 29m 3552 S 8.6 0.4 14:55.03 skype ..... Problems1 : Non stop error like this, caused by flooding: ======================================== 2010-09-28 12:54:45.878273 [ERR] switch_core_io.c:121 skypopen/interface14 has no read codec. 2010-09-28 12:54:45.881274 [ERR] switch_core_io.c:121 skypopen/interface14 has no read codec. 2010-09-28 12:54:45.883271 [ERR] switch_core_io.c:121 skypopen/interface14 has no read codec. 2010-09-28 12:54:45.886277 [ERR] switch_core_io.c:121 skypopen/interface14 has no read codec. 2010-09-28 12:54:45.888352 [ERR] switch_core_io.c:121 skypopen/interface14 has no read codec. 2010-09-28 12:54:45.891345 [ERR] switch_core_io.c:121 skypopen/interface14 has no read codec. AND 2010-09-28 13:19:04.272557 [ERR] switch_core_io.c:121 skypopen/interface22 has no read codec. 2010-09-28 13:19:04.274541 [DEBUG] switch_ivr_play_say.c:1476 done playing file 2010-09-28 13:19:04.274541 [DEBUG] switch_core_file.c:176 File /usr/local/freeswitch/sounds/music/8000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav sample rate 8000 doesn't match requested rate 16000 2010-09-28 13:19:04.274541 [ERR] mod_skypopen.c:713 rev git2svn-syncpoint-master-296-g25ddef5[(nil)|37 ][ERRORA 713 ][interface14][-1, 0, 0] channel not ready 2010-09-28 13:19:04.274541 [DEBUG] switch_ivr_play_say.c:1476 done playing file 2010-09-28 13:19:04.274541 [DEBUG] switch_ivr_play_say.c:1162 Codec Activated L16@16000hz 1 channels 20ms 2010-09-28 13:19:04.274541 [DEBUG] switch_core_file.c:176 File /usr/local/freeswitch/sounds/music/8000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav sample rate 8000 doesn't match requested rate 16000 2010-09-28 13:19:04.274541 [DEBUG] switch_ivr_play_say.c:1162 Codec Activated L16@16000hz 1 channels 20ms 2010-09-28 13:19:04.274541 [ERR] switch_core_io.c:121 skypopen/interface25 has no read codec. 2010-09-28 13:19:04.274541 [DEBUG] switch_ivr_play_say.c:1476 done playing file 2010-09-28 13:19:04.274541 [DEBUG] switch_core_file.c:176 File /usr/local/freeswitch/sounds/music/8000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav sample rate 8000 doesn't match requested rate 16000 2010-09-28 13:19:04.274541 [DEBUG] switch_ivr_play_say.c:1162 Codec Activated L16@16000hz 1 channels 20ms 2010-09-28 13:19:04.276545 [ERR] switch_core_io.c:121 skypopen/interface22 has no read codec. 2010-09-28 13:19:04.276545 [DEBUG] switch_ivr_play_say.c:1476 done playing file 2010-09-28 13:19:04.276545 [DEBUG] switch_core_file.c:176 File /usr/local/freeswitch/sounds/music/8000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav sample rate 8000 doesn't match requested rate 16000 2010-09-28 13:19:04.276545 [DEBUG] switch_ivr_play_say.c:1162 Codec Activated L16@16000hz 1 channels 20ms 2010-09-28 13:19:04.276545 [ERR] switch_core_io.c:121 skypopen/interface25 has no read codec. 2010-09-28 13:19:04.276545 [DEBUG] switch_ivr_play_say.c:1476 done playing file 2010-09-28 13:19:04.278564 [DEBUG] switch_core_file.c:176 File /usr/local/freeswitch/sounds/music/8000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav sample rate 8000 doesn't match requested rate 16000 2010-09-28 13:19:04.278564 [DEBUG] switch_ivr_play_say.c:1162 Codec Activated L16@16000hz 1 channels 20ms 2010-09-28 13:19:04.278564 [ERR] mod_skypopen.c:713 rev git2svn-syncpoint-master-296-g25ddef5[(nil)|37 ][ERRORA 713 ][interface14][-1, 0, 0] channel not ready 2010-09-28 13:19:04.278564 [DEBUG] switch_ivr_play_say.c:1476 done playing file 2010-09-28 13:19:04.278564 [ERR] switch_core_io.c:121 skypopen/interface22 has no read codec. 2010-09-28 13:19:04.278564 [DEBUG] switch_ivr_play_say.c:1476 done playing file 2010-09-28 13:19:04.280571 [DEBUG] switch_core_file.c:176 File /usr/local/freeswitch/sounds/music/8000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav sample rate 8000 doesn't match requested rate 16000 2010-09-28 13:19:04.280571 [DEBUG] switch_ivr_play_say.c:1162 Codec Activated L16@16000hz 1 channels 20ms 2010-09-28 13:19:04.280571 [ERR] switch_core_io.c:121 skypopen/interface25 has no read codec. 2010-09-28 13:19:04.280571 [DEBUG] switch_ivr_play_say.c:1476 done playing file 2010-09-28 13:19:04.280571 [DEBUG] switch_core_file.c:176 File /usr/local/freeswitch/sounds/music/8000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav sample rate 8000 doesn't match requested rate 16000 2010-09-28 13:19:04.280571 [DEBUG] switch_ivr_play_say.c:1162 Codec Activated L16@16000hz 1 channels 20ms 2010-09-28 13:19:04.281800 [DEBUG] switch_core_file.c:176 File /usr/local/freeswitch/sounds/music/8000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav sample rate 8000 doesn't match requested rate 16000 2010-09-28 13:19:04.281800 [DEBUG] switch_ivr_play_say.c:1162 Codec Activated L16@16000hz 1 channels 20ms 2010-09-28 13:19:04.281800 [ERR] switch_core_io.c:121 skypopen/interface22 has no read codec. 2010-09-28 13:19:04.281800 [DEBUG] switch_ivr_play_say.c:1476 done playing file 2010-09-28 13:19:04.281800 [ERR] switch_core_io.c:121 skypopen/interface25 has no read codec. 2010-09-28 13:19:04.281800 [DEBUG] switch_ivr_play_say.c:1476 done playing file 2010-09-28 13:19:04.281800 [DEBUG] switch_core_file.c:176 File /usr/local/freeswitch/sounds/music/8000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav sample rate 8000 doesn't match requested rate 16000 2010-09-28 13:19:04.281800 [DEBUG] switch_ivr_play_say.c:1162 Codec Activated L16@16000hz 1 channels 20ms Switch 1 - sk list ================================ freeswitch(a)a.com> sk list sk console is NOT yet assigned F ID Name IB (F/T) OB (F/T) State CallFlw UUID = ==== ======== ======= ======= ====== ============ ====== 1 [interface1] 1/3 0/0 IDLE CALL_IDLE 2 [interface2] 0/1 0/0 UP STATUS_INPROGRS e2f255cd-89ba-43e0-8d01-bf9579c98ffa 3 [interface3] 1/7 0/0 IDLE CALL_IDLE 4 [interface4] 0/1 0/0 PREANSW CALL_IDLE 1cc57d27-1db7-420e-8459-c27e588fd5b8 5 [interface5] 0/1 0/0 UP STATUS_INPROGRS 8d4719fe-8567-4c62-9224-8c9766aebc08 6 [interface6] 0/0 0/0 IDLE CALL_IDLE 7 [interface7] 0/1 0/0 IDLE CALL_IDLE 8 [interface8] 0/1 0/0 IDLE CALL_IDLE 9 [interface9] 0/2 0/0 IDLE CALL_IDLE 10 [interface10] 0/1 0/0 UP STATUS_INPROGRS 3ba37a2b-e1ec-436d-86be-76510ca7fe6a 11 [interface11] 0/0 0/0 IDLE CALL_IDLE 12 [interface12] 1/4 0/0 IDLE CALL_IDLE 13 [interface13] 1/8 0/0 IDLE CALL_IDLE 14 [interface14] 1/5 0/0 IDLE CALL_IDLE 15 [interface15] 0/0 0/0 IDLE CALL_IDLE 16 [interface16] 0/1 0/0 PREANSW CALL_IDLE 638935df-443d-4f02-af8d-40fb93eb21d4 17 [interface17] 0/2 0/0 IDLE CALL_IDLE 18 [interface18] 0/1 0/0 UP STATUS_INPROGRS 8aec831e-c410-4d85-aa5f-a84c3f364b4e 19 [interface19] 1/5 0/0 IDLE CALL_IDLE 20 [interface20] 0/1 0/0 IDLE CALL_IDLE 21 [interface21] 0/1 0/0 UP STATUS_INPROGRS cf26b813-8557-46e0-8dcc-e30bdecce7d9 22 [interface22] 6/9 0/0 IDLE CALL_IDLE 23 [interface23] 2/5 0/0 PREANSW CALL_IDLE a1fa5e30-5491-4941-b54c-1bec882cc4cd 24 [interface24] 1/4 0/0 IDLE CALL_IDLE 25 [interface25] 2/5 0/0 IDLE CALL_IDLE 26 [interface26] 2/5 0/0 UP STATUS_INPROGRS 9289a13e-bedc-43d1-92a2-8bec775188c6 27 [interface27] 0/1 0/0 IDLE CALL_IDLE 28 [interface28] 0/0 0/0 IDLE CALL_IDLE 29 [interface29] 0/1 0/0 UP STATUS_INPROGRS 32100a1d-38ac-4104-b5d7-35b5c4603a84 30 [interface30] 1/2 0/0 IDLE CALL_IDLE Total Interfaces: 30 IB Calls(Failed/Total): 20/78 OB Calls(Failed/Total): 0/0 Switch 2 - sk list ======================================== freeswitch(a)a.com> freeswitch(a)localhost.localdomain> sk list sk console is NOT yet assigned F ID Name IB (F/T) OB (F/T) State CallFlw UUID = ==== ======== ======= ======= ====== ============ ====== 1 [interface1] 0/0 5/8 UP STATUS_INPROGRS a7a35355-63de-4734-97a3-18dd97ee2f1e 2 [interface2] 0/0 16/17 IDLE CALL_IDLE * 3 [interface3] 0/0 0/5 UP STATUS_INPROGRS a9256794-fa7f-4fac-bb23-aa3553bf9bb8 4 [interface4] 0/0 0/3 UP STATUS_INPROGRS 3423612f-8db5-481c-9fe8-a79cdd1234df 5 [interface5] 0/1 1/3 UP STATUS_INPROGRS 8f0878f5-d830-417a-a9b0-17c6b46f9e56 6 [interface6] 0/0 7/8 UP STATUS_INPROGRS 60b0b495-3115-42dd-bb18-a8b3d8b71087 7 [interface7] 0/0 16/16 IDLE CALL_IDLE 8 [interface8] 0/0 1/3 UP STATUS_INPROGRS 9da0beb5-0442-4c78-a2ed-ddd3c0852560 9 [interface9] 0/0 16/16 IDLE CALL_IDLE 10 [interface10] 0/0 16/16 IDLE CALL_IDLE Total Interfaces: 10 IB Calls(Failed/Total): 0/1 OB Calls(Failed/Total): 78/95 freeswitch(a)localhost.localdomain> When ERR switch_core_io.c:121 skypopen/xxxxinterface has no read codec and ERR mod_skypopen.c:713 channel not ready gets started, whole active calls get affected and this problem gets started. Thanks for your guide, let me know your tips. Best Regards

2 13

Re: [Freeswitch-users] no audio in loopback mode
by Bernhard Suttner 28 Sep '10

28 Sep '10

Hi, I forget something to mention: If I set the loopback_bowout to "true" than I have media in both directions. Best regards, Bernhard ----- Original Message ----- From: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] To: 'FreeSWITCH Users Help' [mailto:freeswitch-users@lists.freeswitch.org] Sent: Tue, 28 Sep 2010 19:06:10 +0200 Subject: [Freeswitch-users] no audio in loopback mode > Hi, > > I am using the latest GIT version of freeswitch and I have the following > scenario: > > A ---SIP ---> FS ---loopback--> FS ----SIP---> B > > I am not using bypass_media, proxy_media and loopback_bowout is set to false > to get proper CDRs. The call setup is fine without a problem (ok, no session > progress audio). After the call is up, I have no audio. I need to use > loopback because for A the CDR as a outgoing call have to be written and for > B it is a incoming call (also CDR required). Is there maybe another way to > get that working without looback? I like the loopback method because its > "easy" to understand that there is a A-side of the call (outgoing) and a > B-side which is incoming. > > Could that be something operating system related, because it's a debian box > running on a virtual box? > > > Best regards, > Bernhard > > > _______________________________________________ > 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 >

2 2

setting channel variable sip_to_host or sip_to_uri for B leg
by Neill Wilkinson 28 Sep '10

28 Sep '10

All, trying to set the to field on the B leg of the call from from FS toward the phone: <action application="export" data="sip_to_host=aeonvista.co.uk"/> This seems to work - over-writing the From field in the B leg. <action application="export" data="sip_from_uri=${sip_from_uri}"/> For example in the dialplan I am doing: <extension name="vodafone_+441635698866"> <condition field="destination_number" expression="^(\+441635698866)$"> <action application="set" data="domain_name=aeonvista.co.uk"/> <action application="set" data="domain=aeonvista.co.uk"/> <action application="set" data="bypass_media=true"/> <action application="export" data="sip_from_uri=${sip_from_uri}"/> <action application="export" data="sip_to_host=aeonvista.co.uk"/> <action application="bridge" data="sofia/aeonvista.co.uk/+441635698866" /> </condition> </extension> The resulting SIP INVITE from FS to the B party looks like: INVITE sip:+441635698866@192.168.0.15:2048;line=3sllxdbk SIP/2.0 Via: SIP/2.0/UDP 192.168.0.56;rport;branch=z9hG4bK5D3HQUQUa4XpF Max-Forwards: 69 From: "+441635698863" <sip:+441635698863@aeonvista.co.uk<sip%3A%2B441635698863(a)aeonvista.co.uk> >;tag=Zy3U793m18aye To: <sip:+441635698866@192.168.0.15:2048;line=3sllxdbk> Call-ID: dc5311ea-45c1-122e-c586-000c290fc1a6 CSeq: 2509932 INVITE Contact: <sip:mod_sofia@192.168.0.56:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-svn-18355M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 330 X-Serialnumber: 0004132CB2E5 P-Key-Flags: keys="3" X-FS-Support: update_display Remote-Party-ID: "+441635698863" <sip:+441635698863@192.168.0.56<sip%3A%2B441635698863(a)192.168.0.56> >;party=calling;screen=yes;privacy=off <http://aeonvista.blogspot.com/> Aeonvista Ltd <http://aeonvista.blogspot.com/> *Opening Up New Ideas*

1 0

no audio in loopback mode
by Bernhard Suttner 28 Sep '10

28 Sep '10

Hi, I am using the latest GIT version of freeswitch and I have the following scenario: A ---SIP ---> FS ---loopback--> FS ----SIP---> B I am not using bypass_media, proxy_media and loopback_bowout is set to false to get proper CDRs. The call setup is fine without a problem (ok, no session progress audio). After the call is up, I have no audio. I need to use loopback because for A the CDR as a outgoing call have to be written and for B it is a incoming call (also CDR required). Is there maybe another way to get that working without looback? I like the loopback method because its "easy" to understand that there is a A-side of the call (outgoing) and a B-side which is incoming. Could that be something operating system related, because it's a debian box running on a virtual box? Best regards, Bernhard

1 0

Bridge ignores call_timeout when using :_: separators
by Mario 28 Sep '10

28 Sep '10

OK folks, been battling this for several hours with nothing found here or on web searches. I hope I am missing something and would greatly appreciate any help. I am in the home stretch and this is the last big problem (I hope). Note this problem ONLY occurs with bridging when ALL three are true: 1. Use :_: between the destinations 2. One destination is a shared extension (multiple-registrations value="contact) 3. One destination is an outside phone (cell in this case). I am bridging to internal and an external cell phone. I want to regain control to go to IVR when there is no answer. I need to use the :_: separator because the internal extensions are set for multiple logins, the same extension # is used on multiple phones. This is because I have 2 shared mailboxes and the SPA962 voicemail button is tied to the user ID not the mailbox setting. So the only way for the button on multiple phones to get the same mailbox is to use shared extensions. I am using the :_: separator because if you use a comma, only one of the internal extensions rings because of the multiple logons. This is known and I found searching that :_: is needed when bridging to a number or group that uses multiple logons. Everything works fine with the exception that the call_timeout does not work, the call keeps ringing if no answer (or the cell phone goes to voicemail, not what I want). I would greatly appreciate any help since I think I have tried many things to get around it but nothing seems to work. I have tried 1 extension, etc. Thanks! Mario This rings all internal extensions (right) but no timeout: <action application="bridge" data="group/home@${domain_name}:_:sofia/gateway/uid1/19161234567:_:sofia/gateway/uid1/19162223333"/> This rings only 1 internal extension (wrong) but does timeout: <action application="bridge" data="group/home@${domain_name},sofia/gateway/uid1/19161234567,sofia/gateway/uid1/19162223333"/> Changing group to a single user ID (shared) has same results. This is the complete extension, many extra things are there to try to get it to work: <extension name="mvphone"> <condition field="destination_number" expression="^2920$">  <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>  <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/>  <action application="set" data="call_timeout=15"/> <action application="bridge" data="group/home@${domain_name}"/>  <action application="playback" data="/usr/local/freeswitch/conf/oursounds/we_trying_please_hold.wav"/>  <action application="set" data="ringback=$${hold_music}"/>  <action application="set" data="ignore_early_media=true"/> <action application="set" data="bridge_answer_timeout=10"/> <action application="ring_ready"/>  <action application="set" data="call_timeout=15"/>  <action application="bridge" data="group/home@${domain_name}:_:sofia/gateway/uid1/19161234567:_:sofia/gateway/uid1/19162223333"/>     <action application="answer"/> <action application="voicemail" data="default ${domain_name} 100"/> --> </condition> </extension>

3 4

Using Limit
by Muhammad Arfeen 28 Sep '10

28 Sep '10

hi im new to freeswitch. By looking at wiki, i dont see many examples using limit with API. I just want to limit the no. of calls on a particular DID number when calls comes to freeswitch. Before doing any other operation, I want to set / get the no. of channels on the number. Do I still need to using XML to set the limit before forwarding to desired SIP address ? or we need to use API ? Regards, Arfeen ___________________________ http://www.facebook.com/arfeen http://pk.linkedin.com/in/arfeen

2 1

Combining two dialplans
by mazilo 28 Sep '10

28 Sep '10

I have the following working outbound dialplan for http://gizmo5.com Gizmo5 that is capable of processing certain patterns of dialed numbers, i.e. 1747XXXXXXX and 0088351007XXXXXXX. <condition field="destination_number" expression="(1747\d{7}|0088351007\d{7})$"> Also, I would like to add the following to the above Gizmo5 outbound dialplan so that it will be able to strip off some leading digits, i.e. 010, of the dialed numbers before processing them. <condition field="destination_number" expression="^010(\d+)$"> Is it possible to combine the above two dialplans into a single dialplan in a single line? If not, how can one go about writing a dialplan that can filter all of the above? ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Combining-two-dialplans-tp557… Sent from the freeswitch-users mailing list archive at Nabble.com.

2 1

playback terminator
by Tihomir Culjaga 28 Sep '10

28 Sep '10

is there any playback terminator for phrase macros? e.g. when you do: <action application="phrase" data="saymoney , 2.31"/> or <action application="phrase" data="timeleft , 33"/> I'd like to make FS stop playing on any pressed digits... something like: <action application="set" data="playback_terminators=#*0123456789"/> but seems not to be working ... any ideas ?

3 4

continuous beep tone for tone_stream
by Pete Kay 28 Sep '10

28 Sep '10

Hi, How can I create a tone that will give out a continuous nonstop beep sound? Thanks, Pete

2 1