March 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Hang Up Cause is Blank in Dial Plan
by Bob McCarthy 02 Mar '12

02 Mar '12

I am trying to play messages for failures on outbound Calls. When I try to use ${hangup_cause} after the failed bridge command it comes back as blank. what am I doing wrong ??? <extension name="Outbound_SwitchVox"> <condition regex="any"> <regex field="destination_number" expression="^(\d{10})$"/> <regex field="destination_number" expression="^2001$"/> <action application="set" data="call_timeout=20"/> <action application="set" data="hangup_after_bridge=false"/> <action application="bridge" data="sofia/external/${destination_number}@$${Switchvox}"/> <action application="set" data="test=${hangup_cause}"/> <action application="log" data="1 A-leg hangup cause: ${hangup_cause}"/> <action application="execute_extension" data="hangup_reason-${hangup_cause} XML features"/>"/> </condition> </extension> 2012-03-02 06:42:38.005915 [DEBUG] switch_channel.c:2850 (sofia/external/2001(a)192.168.1.195) Callstate Change RINGING -> HANGUP 2012-03-02 06:42:38.005915 [NOTICE] switch_ivr_originate.c:3183 Hangup sofia/external/2001(a)192.168.1.195 [CS_CONSUME_MEDIA] [NO_ANSWER] 2012-03-02 06:42:38.005915 [DEBUG] switch_channel.c:2873 Send signal sofia/external/2001(a)192.168.1.195 [KILL] 2012-03-02 06:42:38.005915 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/2001(a)192.168.1.195 [BREAK] 2012-03-02 06:42:38.005915 [INFO] mod_dptools.c:2922 Originate Failed. Cause: NO_ANSWER EXECUTE sofia/internal/CO999x1001.1(a)192.168.57.211 set(test=) 2012-03-02 06:42:38.005915 [DEBUG] mod_dptools.c:1281 sofia/internal/CO999x1001.1(a)192.168.57.211 SET [test]=[UNDEF] EXECUTE sofia/internal/CO999x1001.1(a)192.168.57.211 log(1 A-leg hangup cause: ) 2012-03-02 06:42:38.005915 [ALERT] mod_dptools.c:1420 A-leg hangup cause: EXECUTE sofia/internal/CO999x1001.1(a)192.168.57.211 execute_extension(hangup_reason- XML features) 2012-03-02 06:42:38.005915 [INFO] mod_dialplan_xml.c:485 Processing Dispatch 2 <CO999x1001.1>->hangup_reason- in context features Dialplan: sofia/internal/CO999x1001.1(a)192.168.57.211 parsing [features->dx] continue=false

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bypass_media_after_bridge codec Negotiation Problem
by Paul Hazlett 02 Mar '12

02 Mar '12

I need some advice on how to handle a codec issue during a bridge setup with media bypass. Here is an example of what is going on. I use freeswitch to allow potential customers to try our service before signing up. They can call the PSTN for a limited number of minutes - then freeswitch cuts them off. When they dial a PSTN number, freeswitch first tells them how many remaining minutes they have, then it bidges to the PSTN and then jumps out of the media stream with bypass_media_after_bridge - this all works and the call is setup, but without audio because the codecs are not negotiated correctly and I cannot figure out how to make freeswitch cooperate. A concrete example; The caller supports codecs X and Y, with X being preferred (SILK, PCMU). Freeswitch supports codecs X and Y with X being preferred (SILK, PCMU). The PSTN gateway supports codecs Y and Z with Z being preferred (G729, PCMU). When freeswitch answers the inbound call using ESL pre_answer(), the codec in the 183 response is X (SILK). Freeswitch plays audio to the caller to tell them their remaining trial usage minutes and that works as expected. Next, freeswitch does a bridge to the PSTN, offering codecs X and Y (SILK, PCMU). The PSTN gateway replies 200OK with codec Y (PCMU) as it does not support codec X (SILK). Freeswitch then does media bypass, first sending a re-INVITE to the PSTN gateway and getting it pointed to the caller's IP/port. It again offers codecs X and Y and the PSTN again replies 200OK with codec Y (PCMU). Now for the problem (line 749 in the pastbin log); Freeswitch then re-INVITES the caller, but with codecs X and Y instead of just Y and since the caller prefers X it replies 200OK with codec X (SILK). The the call is set up. The caller and the PSTN gateway are bridged and talking to each other on the correct IP and ports - but with different codecs. So how can I get freeswitch to re-INVITE the A-leg with the same codes that the B-leg accepted? See the highlighted area at line 749: http://pastebin.freeswitch.org/18561 Also not that this was captured at our SIP load balancer, so you'll see all the messages passing on internal and external interfaces). Configuration info: FreeSWITCH Version 1.0.head git-b290936 2012-03-01 10-21-17 -0800 inbound-late-negotiation == false inbound-codec-negotiation == generous global_codec_prefs ==SILK,PCMU,H263,H264 outbound_codec_prefs == SILK,PCMU I am implementing this with ESL. Regards, Paul

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French sounds
by Jimmy Godbout 02 Mar '12

02 Mar '12

Hi, When will the new french sounds be available ? Thanks, Sipster ____________________________________________________________ FREE 3D MARINE AQUARIUM SCREENSAVER - Watch dolphins, sharks & orcas on your desktop! Check it out at http://www.inbox.com/marineaquarium

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get digits during entire session
by spyros papadopoulos 02 Mar '12

02 Mar '12

Hi, I am trying to figure out a way to save the digits pressed during the entire duration of a call. I am particularly interested in doing this after a call has been bridged to an external provider. I currently use a perl script to bridge connections accordingly, based on various database look ups. Are there any functions that could be used for this? Posibbly through ESL? I would like to save the result string in an db. thanks in advance, spyros

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Use of ESL and PHP for Call
by Shahzad Bhatti 02 Mar '12

02 Mar '12

Hi, I am a new user and want to create an application to play a message using mod_tts (cepstral) like we have a meeting on Monday at 9 are you available, if yes press 1 and no then press 2 then record the key pressed in a variable or store it in the database. but when i originate call the call hangup after speak, but i need to make call continue so the caller press the key and then call hangup. now what i do, can i make it simple or make an IVR for it. i searched alot but not manage to get the solution. so far i get the digit pressed using $digit = getHeader("DTMF-Digit"); but i need to control the call to park it after playing the message till called press the digit. my code is as: <?php require_once('ESL.php'); $host_name = 'localhost'; $port = '8021'; $password = 'ClueCon'; $esl = new ESLconnection($host_name,$port,$password); $e = $esl->sendRecv("api create_uuid"); $uuid = $e->getBody(); $user = '1500'; $cmd1 = "bgapi originate {origination_uuid=$uuid}user/$user &speak('cepstral|william|Hi, we have a meeting on Monday at nine, are you joining us, if yes then press one and if not press two')"; $e = $esl->sendRecv($cmd1); $e = $esl->sendRecv("events plain all"); $e = $esl->filter("Unique-ID",$uuid); while ($esl->connected()) { $e = $esl->recvEvent(); $result = $e->getType(); if($result=='CHANNEL_EXECUTE_COMPLETE') { // what i do here...! $esl->execute("&sleep",10000); } $digit = $e->getHeader("DTMF-Digit"); $state_no = $e->getHeader("Channel-State-Number"); $curr_uuid = $e->getHeader("Channel-Call-UUID"); /* $ans_state = $e->getHeader("Answer-State"); $state_no = $e->getHeader("Channel-State-Number"); $core_id = $e->getHeader("Core-UUID"); */ if($curr_uuid==$uuid && $digit!=NULL) { print "input number is " . $digit . "\n"; } $state = $e->getHeader("Channel-Call-State"); if ($state == 'HANGUP') { print "\n\nHangup Cause Number: " . $state_no; $esl->disconnect(); } } ?> please help me to solve the problem Regards Shahzad Bhatti

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Simple dialplan extension
by Milos Jovanovic 02 Mar '12

02 Mar '12

Hello. I need a simple dialplan extension that will do the following: 1) answer the call 2) play a sound file in loop (unlimited loops) 3) hangup the call after X seconds (number of seconds should be defined in extension) What's the simplest way to do that? Thanks in advance, Milos

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FreeSWITCH Cookbook Is Published!
by Michael Collins 01 Mar '12

01 Mar '12

Hey all, Just wanted to mention a little story I wrote up on the main FreeSWITCH website: http://www.freeswitch.org/node/381 Packt has officially published the the cookbook, so get your copy today! -Michael

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Re: [Freeswitch-users] 401 Unauthorized in iptel external gateway
by Stephen Dame 01 Mar '12

01 Mar '12

Sent from my U.S. Cellular® Android phone Vitalie Colosov <vetali100(a)gmail.com> wrote: >Yes I was not sure if you was able to call from registered endpoint thru FS >and also directly, or only directly. >So FS does not create correct #3 query...? >Probably as was suggested above, someone can investigate directly at your >server. >Or if anybody has better ideas... > > >2012/2/26 Anita Hall <anita.hall(a)simmortel.com> > >> Vitalie is right. I have seen similar logs. >> >> I am currently using localphone and rapidvox as SIP provider and both >> register Ok. >> >> This is my sip_profile >> http://wiki.freeswitch.org/wiki/Provider_Configuration:_Localphone.com >> >> I could look it up for you if you can give remote access to your machine >> and watch over screen or byobu. >> >> Alternatively, you could give me your account name and password. I will >> check if it works from my installation. I promise not to make any calls and >> you can change the password later :) >> >> regards, >> Anita >> >> >> >> >> On Sun, Feb 26, 2012 at 11:34 AM, Denis Gasparin < >> denis.gasparin(a)edistar.com> wrote: >> >>> Sorry, i did not exposed well the problem... >>> >>> I was able to call iptel account directly without the gateway (so without >>> registering to iptel). >>> >>> When i setup the gateway i see sofia trying continuosly to register to >>> iptel without success. >>> I see many 401 messages like this: >>> >>> 1. FS->IPTEL Register >>> 2. IPTEL->FS Unauthorized >>> 3. FS->IPTEL Register >>> 4. IPTEL->FS Unauthorized >>> 5. FS->IPTEL Register >>> ... >>> >>> After a while sofia stops trying to register and retries after 30 seconds >>> and so on. >>> >>> Thank you for your help, >>> Denid >>> >>> Il giorno 26/feb/2012, alle ore 02:20, Vitalie Colosov < >>> vetali100(a)gmail.com> ha scritto: >>> >>> Usually SIP registration goes like this: >>> >>> 1. FS->IPEL Register >>> >>> 2. IPTEL->FS Unauthorized >>> >>> 3. FS->IPTEL Register (yes, again, with additional credentials) >>> 4. IPTEL->FS OK >>> >>> You say that you was able to call iptel successfully, so FS is registered. >>> You just did not observed and did not provide SIP messages #3 and #4. >>> >>> >>> >>> 2012/2/25 Denis Gasparin <denis.gasparin(a)edistar.com> >>> >>>> Hi to all. >>>> >>>> Today I installed freeswitch following the guide in Freeswitch book. >>>> I made successfully all the tests in the book... well all but one. >>>> >>>> When I tried to setup an external gateway on my iptel account, the >>>> registration process failed with 401 sip message. >>>> >>>> This is the sip trace: >>>> >>>> freeswitch@dhcppc7> send 647 bytes to udp/[217.9.36.145]:5060 at >>>> 17:59:20.965613: >>>> >>>> ------------------------------------------------------------------------ >>>> REGISTER sip:sip.iptel.org;transport=udp SIP/2.0 >>>> Via: SIP/2.0/UDP MY_PUBLIC_IP:5080;rport;branch=z9hG4bKXNc7jeQB7BXrp >>>> Max-Forwards: 70 >>>> From: <sip:MY_ACCOUNT@iptel.org;transport=udp>;tag=XQ58BvHmj15eD >>>> To: <sip:MY_ACCOUNT@iptel.org;transport=udp> >>>> Call-ID: 8c2823cd-573d-4cc2-8e01-cba340ae890f >>>> CSeq: 24760296 REGISTER >>>> Contact: <sip:gw+iptel@MY_PUBLIC_IP:5080;transport=udp;gw=iptel> >>>> Expires: 3600 >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1e6c4be 2012-02-24 >>>> 14-15-32 -0600 >>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>>> REGISTER, REFER, NOTIFY >>>> Supported: timer, precondition, path, replaces >>>> Content-Length: 0 >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> recv 710 bytes from udp/[217.9.36.145]:5060 at 17:59:21.070238: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 401 Unauthorized >>>> Via: SIP/2.0/UDP 192.168.1.9:5080 >>>> :5080;rport=19198;branch=z9hG4bKXNc7jeQB7BXrp >>>> From: <sip:MY_ACCOUNT@iptel.org;transport=udp>;tag=XQ58BvHmj15eD >>>> To: <sip:MY_ACCOUNT@iptel.org >>>> ;transport=udp>;tag=f18024bbca64dd60f33bcd92c390ee0f-5904 >>>> Call-ID: 8c2823cd-573d-4cc2-8e01-cba340ae890f >>>> CSeq: 24760296 REGISTER >>>> Expires: 600 >>>> Min-Expires: 240 >>>> WWW-Authenticate: Digest realm="iptel.org", >>>> nonce="T0ki/U9JIsE+F7QdNo/qIktZaDvpS0sn" >>>> Server: ser (3.3.0-dev0 (i386/linux)) >>>> Content-Length: 0 >>>> Warning: 392 217.9.36.145:5060 "Noisy feedback tells: pid=863 >>>> req_src_ip=MY_PUBLIC_IP req_src_port=19198 in_uri=sip:sip.iptel.org;transport=udp >>>> out_uri=sip:sip.iptel.org;transport=udp via_cnt==1" >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> >>>> I'm sure that user account and password are valid. >>>> I also tested (successfully) a call from my extension to my >>>> account(a)iptel.org. >>>> Registration to iptel.org using a sip phone with the same user works >>>> just fine. >>>> >>>> What's wrong? >>>> >>>> Thank you in advance for your help, >>>> >>>> Denis >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting(a)freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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

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Problems with t38_gateway
by Steven Lam, KeenSystems B.V. 01 Mar '12

01 Mar '12

Hi all, I'm experimenting with the t38_gateway in mod_spandsp to build a gateway that fully supports "passive" T_38. My test config looks like this: <extension name="test"> <condition> <action application="set" data="fax_enable_t38=true"/> <action application="set" data="sip_execute_on_image=t38_gateway self nocng"/> <action application="bridge" data="{sip_execute_on_image='t38_gateway self nocng'}sofia/internal/1234@hostB"/> </condition> </extension> This works perfectly when: - the A-leg sends a T38 re-INVITE and the B-leg doesn't or - the A-leg doesn't send a T38 re-INVITE but the B-leg does or - nor the A-leg, nor the B-leg send a T38 re-INVITE This does _NOT_ work when both the A-leg and the B-leg send a T38 re-INVITE, in this case the call is hangup. A snipped from the logging on the cli after the second T38 re-INVITE is received shows: 2012-02-29 14:11:34.044702 [DEBUG] sofia.c:5526 Channel sofia/wan_out/+3188xxx(a)xxx.xxx.xxx.xxx entering state [received][100] 2012-02-29 14:11:34.044702 [DEBUG] sofia.c:5537 Remote SDP: v=0 o=keen 1348795719 1348795720 IN IP4 xxx.xxx.xxx.xxx s=Keen c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=image 4568 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:849 a=T38FaxUdpEC:t38UDPFEC EXECUTE sofia/wan_out/+3188xxx(a)xxx.xxx.xxx.xxx t38_gateway(self nocng) 2012-02-29 14:11:34.044702 [DEBUG] mod_spandsp_fax.c:1950 sofia/wan_in/anonymous(a)xxx.xxx.xxx.xxx starting gateway mode to sofia/wan_out/+3188xxx(a)xxx.xxx.xxx.xxx 2012-02-29 14:11:34.044702 [DEBUG] mod_spandsp_fax.c:1965 (sofia/wan_out/+3188xxx(a)xxx.xxx.xxx.xxx) State Change CS_CONSUME_MEDIA -> CS_RESET 2012-02-29 14:11:34.044702 [DEBUG] switch_core_session.c:1180 Send signal sofia/wan_out/+3188xxx(a)xxx.xxx.xxx.xxx [BREAK] 2012-02-29 14:11:34.044702 [DEBUG] mod_spandsp_fax.c:1968 (sofia/wan_in/anonymous(a)xxx.xxx.xxx.xxx) State Change CS_SOFT_EXECUTE -> CS_RESET 2012-02-29 14:11:34.044702 [DEBUG] switch_core_session.c:1180 Send signal sofia/wan_in/anonymous(a)xxx.xxx.xxx.xxx [BREAK] 2012-02-29 14:11:34.044702 [DEBUG] switch_core_session.c:729 Send signal sofia/wan_in/anonymous(a)xxx.xxx.xxx.xxx [BREAK] 2012-02-29 14:11:34.044702 [DEBUG] switch_channel.c:2848 (sofia/wan_out/+3188xxx(a)xxx.xxx.xxx.xxx) Callstate Change ACTIVE -> HANGUP 2012-02-29 14:11:34.044702 [DEBUG] switch_core_session.c:2285 sofia/wan_out/+3188xxx(a)xxx.xxx.xxx.xxx skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-02-29 14:11:34.044702 [NOTICE] mod_spandsp_fax.c:1663 Hangup sofia/wan_out/+3188xxx(a)xxx.xxx.xxx.xxx [CS_RESET] [NORMAL_CLEARING] 2012-02-29 14:11:34.044702 [DEBUG] switch_channel.c:2871 Send signal sofia/wan_out/+3188xxx(a)xxx.xxx.xxx.xxx [KILL] 2012-02-29 14:11:34.044702 [DEBUG] switch_core_session.c:1180 Send signal sofia/wan_out/+3188xxx(a)xxx.xxx.xxx.xxx [BREAK] 2012-02-29 14:11:34.044702 [DEBUG] switch_channel.c:2848 (sofia/wan_in/anonymous(a)xxx.xxx.xxx.xxx) Callstate Change ACTIVE -> HANGUP 2012-02-29 14:11:34.044702 [NOTICE] mod_spandsp_fax.c:1671 Hangup sofia/wan_in/anonymous(a)xxx.xxx.xxx.xxx [CS_RESET] [NORMAL_CLEARING] 2012-02-29 14:11:34.044702 [DEBUG] mod_spandsp_fax.c:1858 sofia/wan_out/+3188xxx(a)xxx.xxx.xxx.xxx skip receive message [UNBRIDGE] (channel is hungup already) Does anybody know what goes wrong here? Steven

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Segmentation fault
by a b 01 Mar '12

01 Mar '12

Hi, I just noticed that if I bridge a call to a user that exists in the directory but doesn't have the dial-string param set, FreeSWITCH will crash. this is happening on git from yesterday. can someone confirm if this is a bug or something in my setup? Thanks

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