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Re: [Freeswitch-users] Fwd: Major deployment of outbound FAX on latest version of Freeswitch question
by Steve Underwood 06 Jun '11

06 Jun '11

On 05/12/2011 05:21 PM, Juan Antonio Ibañez Santorum wrote: > Hello Steve! > > I would like to ask you one question about fax system. You told > that using FS+mod_spandsp could be gotten successful faxing rates over > 99%. Looking at your experience, is 99% nearer to 100% or to 99%? > Which errors do you find usually on that 1%? > You can run tests all day in a lab without failure. The real test is how well things work against a wide diversity of machines in the real world. This http://www.soft-switch.org/spandsp-soft-fax-performance.html says something about our testing, and what is achievable. Its quite a lot of work to get real reliability numbers. You get a lot of call failures on a typical public FAX server due to things like wrong numbers, voice calls into a FAX port, people purposefully dropping half completed FAX calls, and buggy FAX machines doing weird things. Don't ignore the last one. There are famous make machines with serious bugs. You need to manually check the failed calls one by one to get real reliability figures. In the end there are always a few calls where you are never sure who's fault the failure is. All I can say with hand on heart is the number of unexplainable failures is well below 1% Regards, Steve

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Sangoma Transcoding
by Gustavo Espeche 05 Jun '11

05 Jun '11

Hi some one can do work sangoma d100 transcoding, i installed the card, load the sangoma codec into freeswitch, but i can't do that the freeswitch do a transcoding using sangoma card. I configure my profile <param name="inbound-late-negotiation" value="false"/> <param name="disable-transcoding" value="false"/> if some one configure with success this card i appreciate a lot any advice. Best Regards. -- Gustavo Espeche EasyIpCall S.R.L. www.easyipcall.com Bv Mitre 517 24° E Cordoba - Argentina Te: +54 - 351 - 4280633

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mod_skypopen is awesome issue with demo-ivr conference
by Henry Huang 05 Jun '11

05 Jun '11

Hi: The new interactive installation for mod_skypopen is a piece of cake. Made live much easier. I was so excited and tried it out right after the installation was done. But I found a few issues calling the demo-ivr from skyTpe client. 1. The screaming monkey does not scream anymore. 2. The conference that connects to the freeswitch.org doesn't respond to dtmf pressed on the skype client. Ever since the second client is connected to the conference, the conference tell the client it's muted. So I tried push 0 and any other key to see if I get to unmute. But no, non of the key press works. If anyone have previous experiences on these and fixed them, please do share your methods. Sound quality with the new virtual sound drive is very good. Thanks, Henry Huang US: +1(818)6885508 | 台灣(Taiwan): +886 933847619 Contact Me [image: LinkedIn]<http://www.linkedin.com/profile/view?id=4590380> [image: Facebook] <http://www.facebook.com/profile.php?id=595148342> [image: Twitter] <http://twitter.com/unicsolution> IM [image: Google Talk] red_rain_seven(a)gmail.com [image: Skype] unicsolution [image: MSN] b_ball_henry(a)hotmail.com

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Strage E1 PCI card
by Boris Kovalenko 04 Jun '11

04 Jun '11

Hello! I found strange PCI E1 card marked as DIALOGIC DM/IP0821A-E1-120. No drivers, no manuals. It is possible to get it working with FreeSwitch? -- Regards, Boris

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collectInput does not call callback function
by Kamen 04 Jun '11

04 Jun '11

Hi folks! I seem to have an odd problem, since it works for all others. From my java script I try to collect user input: session.collectInput( mycb, "digits", 7000, 0); where mycb is a regular function collecting user input: function mycb( session, type, obj, arg ) { console_log( "info", "collect digits \n" ); try { if ( type == "digits" ) { console_log( "info", "digit: "+obj.digit+"\n" ); if ( obj.digit == "#" ) { console_log( "info", "detected pound sign.\n" ); exit = true; return( false ); } dtmf.digits += obj.digit; if ( dtmf.digits.length >= digitmaxlength ) { exit = true; return( false ); } } } catch (e) { console_log( "err", e+"\n" ); } return( true ); } //end function mycb The very first line is a console info message, so I would know the function is actually called. It never happens. Right before collectInput call I have a message playing a prompt, which I can hear. So workflow definitely goes into the collectInput function, and then after 7 seconds it continues with dtmf.digits.length=0 even though I punched in the numbers, of course. Thats another proof the callback function never was called. Has anyone experienced this problem before? I run FS in Windows. Appreciate your help. Regards, Sergei Kamen -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/collectInput-does-not-call-ca… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Need help interpreting fs_cli debug messages
by Roman Gelfand 03 Jun '11

03 Jun '11

I was wondering if somebody could help interpret the registration issue. +OK log level [7] freeswitch@internal> nua: nh_create_handle: entering nua: nua_handle_bind: entering 2011-06-02 13:35:18.401497 [NOTICE] sofia_reg.c:367 Registering ###########-callcentric.com nua: nua_register: entering nua(0x873dc88): sent signal r_register nua: nua_stack_set_params: entering soa_clone(static::0x86d2e30, 0x86fba08, 0x873dc88) called soa_set_params(static::0x8dd11008, ...) called soa_set_params(static::0x8dd11008, ...) called nua(0x873dc88): adding register usage nta_leg_tcreate(0x8dd16398) nta: selecting scheme sip tport_tsend(0x86afae0) tpn = */XX.XX.XXX.XXX:5060 tport_resolve addrinfo = XX.XX.XXX.XXX:5060 tport_by_addrinfo(0x86afae0): not found by name */XX.XX.XXX.XXX:5060 tport_vsend returned 639 nta: sent REGISTER (13181995) to */XX.XX.XXX.XXX:5060 tport_pend(0x86afae0): pending 0x8dd0a588 for udp/XX.XX.XXX.XXX:5080 (already 0) nta: timer set to 32000 ms nta: timer shortened to 1000 ms tport_wakeup_pri(0x86afae0): events IN tport_recv_event(0x86afae0) tport_recv_iovec(0x86afae0) msg 0x8dd08cf8 from (udp/XX.XX.XXX.XXX:5080) has 526 bytes, veclen = 1 tport_deliver(0x86afae0): msg 0x8dd08cf8 (526 bytes) from udp/XX.XX.XXX.XXX:5080/sip next=(nil) nta: received 407 Proxy Authentication Required for REGISTER (13181995) nta: 407 Proxy Authentication Required is going to a transaction nta_outgoing: RTT is 135.683 ms tport_release(0x86afae0): 0x8dd0a588 by 0x8dd009d8 with 0x8dd08cf8 auth_digest_challenge_get(): got 7 nta: outgoing_free(0x8dd009d8) nua: nua_application_event: entering nua: nua_authenticate: entering nua(0x873dc88): sent signal r_authenticate nua: nua_handle_magic: entering auth_digest_a1() has A1 = MD5(###########:callcentric.com:passwd) = 8dddd49339918097d09e22dd7b15c38b A2 = MD5(REGISTER:sip:XX.XX.XXX.XXX:5060) auth_response: 9f45cc31d0328fdf18a7e4d3b547e3ea = MD5(8dddd49339918097d09e22dd7b15c38b:dbcc6c9095c35c4be730ec1110f970e5:db5f9cddc4d0d297aa85cc660f20595a) (qop=NONE) nta: selecting scheme sip tport_tsend(0x86afae0) tpn = */XX.XX.XXX.XXX:5060 tport_resolve addrinfo = XX.XX.XXX.XXX:5060 tport_by_addrinfo(0x86afae0): not found by name */XX.XX.XXX.XXX:5060 tport_vsend returned 859 nta: sent REGISTER (13181996) to */XX.XX.XXX.XXX:5060 tport_pend(0x86afae0): pending 0x8dd08cf8 for udp/XX.XX.XXX.XXX:5080 (already 0) tport_wakeup_pri(0x86afae0): events IN tport_recv_event(0x86afae0) tport_recv_iovec(0x86afae0) msg 0x8dd1ae88 from (udp/XX.XX.XXX.XXX:5080) has 526 bytes, veclen = 1 tport_deliver(0x86afae0): msg 0x8dd1ae88 (526 bytes) from udp/XX.XX.XXX.XXX:5080/sip next=(nil) nta: received 407 Proxy Authentication Required for REGISTER (13181996) nta: 407 Proxy Authentication Required is going to a transaction nta_outgoing: RTT is 127.055 ms tport_release(0x86afae0): 0x8dd08cf8 by 0x8dd08a98 with 0x8dd1ae88 auth_digest_challenge_get(): got 7 auth_digest_a1() has A1 = MD5(###########:callcentric.com:passwd) = 8dddd49339918097d09e22dd7b15c38b A2 = MD5(REGISTER:sip:XX.XX.XXX.XXX:5060) auth_response: c0b3965ec27e2e3386277645ae14f324 = MD5(8dddd49339918097d09e22dd7b15c38b:c76385015930949803ca8b0946f81d0e:db5f9cddc4d0d297aa85cc660f20595a) (qop=NONE) nta: selecting scheme sip tport_tsend(0x86afae0) tpn = */XX.XX.XXX.XXX:5060 tport_resolve addrinfo = XX.XX.XXX.XXX:5060 tport_by_addrinfo(0x86afae0): not found by name */XX.XX.XXX.XXX:5060 tport_vsend returned 859 nta: sent REGISTER (13181997) to */XX.XX.XXX.XXX:5060 tport_pend(0x86afae0): pending 0x8dd19ea8 for udp/XX.XX.XXX.XXX:5080 (already 0) nta: outgoing_free(0x8dd08a98) nua: nua_application_event: entering nua: nua_handle_magic: entering tport_wakeup_pri(0x86afae0): events IN tport_recv_event(0x86afae0) tport_recv_iovec(0x86afae0) msg 0x8dd1ae88 from (udp/XX.XX.XXX.XXX:5080) has 526 bytes, veclen = 1 tport_deliver(0x86afae0): msg 0x8dd1ae88 (526 bytes) from udp/XX.XX.XXX.XXX:5080/sip next=(nil) nta: received 407 Proxy Authentication Required for REGISTER (13181997) nta: 407 Proxy Authentication Required is going to a transaction nta_outgoing: RTT is 139.175 ms tport_release(0x86afae0): 0x8dd19ea8 by 0x8dd18790 with 0x8dd1ae88 auth_digest_challenge_get(): got 7 auth_digest_a1() has A1 = MD5(###########:callcentric.com:passwd) = 8dddd49339918097d09e22dd7b15c38b A2 = MD5(REGISTER:sip:XX.XX.XXX.XXX:5060) auth_response: f014cc64fa858ec3615b2b1caab196a6 = MD5(8dddd49339918097d09e22dd7b15c38b:274bba14cbca0384c30f227da6c42c35:db5f9cddc4d0d297aa85cc660f20595a) (qop=NONE) nta: selecting scheme sip tport_tsend(0x86afae0) tpn = */XX.XX.XXX.XXX:5060 tport_resolve addrinfo = XX.XX.XXX.XXX:5060 tport_by_addrinfo(0x86afae0): not found by name */XX.XX.XXX.XXX:5060 tport_vsend returned 859 nta: sent REGISTER (13181998) to */XX.XX.XXX.XXX:5060 tport_pend(0x86afae0): pending 0x8dd18f80 for udp/XX.XX.XXX.XXX:5080 (already 0) nta: outgoing_free(0x8dd18790) nua: nua_application_event: entering nua: nua_handle_magic: entering tport_wakeup_pri(0x86afae0): events IN tport_recv_event(0x86afae0) tport_recv_iovec(0x86afae0) msg 0x8dd1ae88 from (udp/XX.XX.XXX.XXX:5080) has 526 bytes, veclen = 1 tport_deliver(0x86afae0): msg 0x8dd1ae88 (526 bytes) from udp/XX.XX.XXX.XXX:5080/sip next=(nil) nta: received 407 Proxy Authentication Required for REGISTER (13181998) nta: 407 Proxy Authentication Required is going to a transaction nta_outgoing: RTT is 114.044 ms tport_release(0x86afae0): 0x8dd18f80 by 0x8dd16e10 with 0x8dd1ae88 nta: outgoing_free(0x8dd16e10) nua(0x873dc88): removing register usage nta_leg_destroy(0x8dd16398) nua: nua_application_event: entering nua: nua_authenticate: entering nua(0x873dc88): sent signal r_authenticate nua: nua_handle_magic: entering nua: nua_application_event: entering nua: nua_handle_destroy: entering nua(0x873dc88): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_bind: entering nua: nua_handle_destroy: entering nta_leg_destroy((nil)) soa_destroy(static::0x8dd11008) called nta: timer not set 2011-06-02 13:36:19.875422 [WARNING] sofia_reg.c:401 Timeout Registering ###########-callcentric.com nua: nua_handle_destroy: entering 2011-06-02 13:36:20.881513 [WARNING] sofia_reg.c:425 ###########-callcentric.com Failed Registration [0], setting retry to 30 seconds. Thanks in advance

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Troubleshooting FreeSwitch Configuration
by Roman Gelfand 03 Jun '11

03 Jun '11

I am running into trouble trying to register to ITSP. The log file shows ... 2011-06-01 21:58:10.950068 [WARNING] sofia_reg.c:401 Timeout Registering 1XXXXXXXXXX-callcentric.com 2011-06-01 21:58:11.957417 [WARNING] sofia_reg.c:425 1XXXXXXXXXX-callcentric.com Failed Registration [0], setting retry to 30 seconds. 2011-06-01 21:58:42.046331 [NOTICE] sofia_reg.c:367 Registering 1XXXXXXXXXX-callcentric.com When challenged by 407 message from ITSP, I get 05:03:12.857: Missing contact header in REGISTER response. Does anyone know what is happening here? or pointers how to troubleshoot this? Thanks in advance

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Codec enforcment from dialplan?
by Dmitry Saratsky 03 Jun '11

03 Jun '11

Hi All, I have some codec-prefs in my FS configured globally for all profiles (In my case: "global_codec_prefs=G729,PCMU,PCMA,GSM"/>) In addition, I need to provide for some specific incoming channels other codec order (PCMU,PCMA) that destined to FS itself. I can identify those channels by destination number of incoming call (fax service) in my dialplan. I can enforce needed codec from dialplan successfully for my outgoing channels by using (absolute_codec_string='PCMU,PCMA'), but I'm failing to do the same for incoming channels. So now I'm enforcing all incoming calls from my provider to different from global order (<param name="inbound-codec-prefs" value="PCMU,PCMA"/>), instead of doing this for fax calls only. Any ideas? Relevant part of profile config: <profile name="providers"> <gateways> <X-PRE-PROCESS cmd="include" data="providers/*.xml"/> </gateways> <settings> <param name="debug" value="0"/> <param name="shutdown-on-fail" value="true"/> <param name="auth-calls" value="false"/> <param name="sip-trace" value="no"/> <param name="disable-register" value="true"/> <param name="rfc2833-pt" value="101"/> <param name="dialplan" value="XML"/> <param name="context" value="providers"/> <param name="apply-inbound-acl" value="013SONUS-IPS"/> <param name="hold-music" value="$${hold_music}"/> <param name="stun-enabled" value="false"/> <param name="stun-auto-disable" value="true"/> <param name="manage-presence" value="false"/> <param name="nonce-ttl" value="60"/> <param name="dtmf-type" value="rfc2833"/> <param name="dtmf-duration" value="2000"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-timeout-sec" value="50"/> <param name="rtp-hold-timeout-sec" value="300"/> <param name="sip-port" value="$${sip_port}"/> <param name="rtp-ip" value="$${rtp_ip}"/> <param name="sip-ip" value="$${ext_ip}"/>  <param name="inbound-codec-prefs" value="PCMU,PCMA,G729"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="inbound-late-negotiation" value="false"/> <param name="inbound-codec-negotiation" value="greedy"/> <param name="disable-transcoding" value="false"/> </settings> </profile>

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FIFO not working properly
by Joao Leme 03 Jun '11

03 Jun '11

FIFO is not ringing any extension while first call out is not hang up. Even with five different agents to answer the queue only one (the first to answer) get to take the calls, one at a time. Why? It used to work just fine before. What may be causing this behavior? Thanks, John

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Vestec ASR: 3 New Language Models
by Kashif Kahn 03 Jun '11

03 Jun '11

Dear All, We have launched three new acoustic models for use with Vestec ASR: (1) Australian English, (2) Chinese Mandarin, and (3) Indian English. Vestec offers the best deal around for enabling sophisticated speech recognition with command-and-control type IVR applications. Our ASR supports SRGS grammar format, can be integrated using MRCP (v1 & v2) interface, offers high recognition accuracy, and costs a fraction of conventional speech engines. We offer the ASR in two versions according to vocabulary size: 500 items vocabulary and 2,500 items vocabulary, per recognition. A starter kit is available for $25 from Vestec webstore: http://www.vestec.com/products Please do not hesitate to contact me with any questions. We are looking forward to being of service. Regards, -Kashif -- Kashif Kahn VP Business Development Vestec Inc Waterloo, ON Canada phone: +1 519 885-7615

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