November 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to get digitals and stop play when speak tts? Just like session:playAndGetDigits do
by Lei Tang 03 Nov '09

03 Nov '09

Hi all, I'm writing lua ivr scirpt, Does some known how to get digitals and stop play when speak tts? Just like session:playAndGetDigits do. Thanks lots! Best Regards! -- Lei.Tang lei.tlfly(a)gmail.com

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"Can not record session. Media not enabled on channel."
by Maciej Aniserowicz 03 Nov '09

03 Nov '09

Hello, I am using the same set of extensions for testing the system during development, they include XLite, Cisco sip phone and several extensions that just play some audio file. Sometimes, very rarely, this message "Can not record session. Media not enabled on channel." appears on FS console. Like I wrote before, I don't change any codec settings and always use the same set of devices/emulators. What can cause this message? Thanks, Maciej Aniserowicz -- View this message in context: http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media
by Humberto Quintana 03 Nov '09

03 Nov '09

Hi Mike, I re-tried with trunk rev 15319 but I got almost the same behavior: There is now a reINVITE (with FS' SDP) going to A when the REFER is accepted. But still there is no reINVITE for A (with C's SDP) after the call from FS to C is established. Anyway, we decided for now to do a different implementation but if you want to explore more in this issue count me in ;-) Thank you very much! Humberto >Please re-try with latest svn trunk. > >Mike > >On Nov 2, 2009, at 9:36 AM, Humberto Quintana wrote: > >> >> Thanks for you answers guys, >> >> I test the parameters you suggested >> but still no audio due to the lack of reINVITE. By the way I'm using >> 1.0.4 but I also tried 1.0.5pre3. >> >> One particular condition is that there is no on-hold before the >> Blind Transfer. >> >> Regards, >> >> Humberto >> >>> <param name="media-option" value="resume-media-on-hold"/> >>> <param name="media-option" value="bypass-media-after-att-xfer"/> _________________________________________________________________ Lots of fantastic Windows 7 offers, in one convenient place. Get the perfect deal for you now. http://go.microsoft.com/?linkid=9691633

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SIP provider with extern rtp server
by Ivan C Myrvold 02 Nov '09

02 Nov '09

I have used a SIP provider for more than a year. A few days ago, he said he was moving to a new server, and asked me to reconfigure. I did, and everything seemed to work fine, until I did an outgoing call to an external telephone. I found out I had no audio, in neither direction. Incoming calls was working fine. My provider said that the rtp is not going through the sip server, as it did earlier, but now through several other IP's. Do I have to do some special configuration to handle that? Ivan

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Freeswitch in signaling path only
by DJB 02 Nov '09

02 Nov '09

I am wondering why I cannot do as condition#2. For Lua in dialplan, when I have the followings: --WORKING-- (Condition#1) . . session:execute("set","bypass_media=true") session:execute("set","hangup_after_bridge=true") session:execute("set","continue_on_fail=true") . . session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.1|sofia/external/" .. called_num .. "@1.1.1.2") . . --NOT WORKING-- (Condition#2) Note: FS tries to be in media path and send re-invite. . . session:execute("set","bypass_media=true") session:execute("set","hangup_after_bridge=true") session:execute("set","continue_on_fail=true") . . session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.1") session:execute("bridge","sofia/external/" .. called_num .. "@1.1.1.2") . . Thank you, Dorn B.

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CDR CSV variables
by DJB 02 Nov '09

02 Nov '09

I wonder if I don't want to have b-leg in cdr csv, is there any variables that can give me the actual gateway ip address that is actually went out. For instance, if I have this in my dialplan: <action application="bridge" data="sofia/default/$1@192.168.1.4|sofia/default/$1@192.168.1.5"/> the only value that I can think of from cdr csv is to get remote_ip_last_arg, but it would contains the whole line of both ip addresses. Thank you, Dorn B.

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Re: [Freeswitch-users] Fwd: Many CS_REPORTING state Zombie session
by Anthony Minessale 02 Nov '09

02 Nov '09

We already concluded its your unacceptabe use of originate in hangup hook right? On Nov 1, 2009 7:45 PM, "Seven Du" <dujinfang(a)gmail.com> wrote: Just suspicious would be possible that happened on sqlite stage? I manually deleted the channels from sqlite and nothing bad happend. just FYI. ---------- Forwarded message ---------- From: Seven Du <dujinfang(a)gmail.com> Date: Sun, 1 Nov 2009 10:24:32 +0800 Subject: Fwd: [Freeswitch-users] Many CS_REPORTING state Zombie session To: Thank you Anthony. We are on r14696 and no non-standard mods loaded. we even unloaded mod_xml_cdr. But we are heavily using mod_erlang_event both inbound and outbound. I must be very careful if I upgrade to trunk and turn on rwlock debug because it's on production and the the problem not happening that much so would be hard to trace. But I will find time to test and report back. FYI, I also noticed that in some zombile channels, couple of INVITEs sent but never got a response. However, it sent out an ACK at last. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@list...

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Re: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media
by Humberto Quintana 02 Nov '09

02 Nov '09

Thanks for you answers guys, I test the parameters you suggested but still no audio due to the lack of reINVITE. By the way I'm using 1.0.4 but I also tried 1.0.5pre3. One particular condition is that there is no on-hold before the Blind Transfer. Regards, Humberto > <param name="media-option" value="resume-media-on-hold"/> > <param name="media-option" value="bypass-media-after-att-xfer"/> >> My scenario is as follows: >> >> inbound-bypass-media is set in the profile because we dont want FS handling >> the media. >> >> 1. A calls B >> 2. FS sends to B the A's SDP >> 3. B answers >> 4. FS sends to A the B's SDP >> 5. Media going directly between A and B >> 6. B REFERs the call to C (blind transfer with no reINVITE for Hold) >> 7. FS accepts(202) the REFER and sends the NOTIFY >> 7a. B and FS send the BYE >> 8. FS sends an INVITE to C with A's SDP >> 9. C answers >> 10. FS doesn't send a reINVITE to A to let it know about C's SDP >> >> >> Is that the expected FS behavior or is this a bug? _________________________________________________________________ Ready for a deal-of-a-lifetime? See fantastic offers on Windows 7, in one convenient place. http://go.microsoft.com/?linkid=9691634

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Re: [Freeswitch-users] Get error "415 Unsupported Media Type" whenreceiving call from softswitch
by Lei Tang 02 Nov '09

02 Nov '09

Hi all, The problem is solved. I ask the softswitch to send only sdp in INVITE message, then It works. I think sofia doesn't support multipart content currently. is it right? 2009/11/2 Lei Tang <lei.tlfly(a)gmail.com> > Hi Daniel. > Sure. pls email me to tlfly(a)hotmail.com. > > 2009/11/2 Zeng, Qinglan (Daniel) <qzeng(a)avaya.com> > > Lei, >> >> This is Daniel Zeng and I got your email address from the maillist of FS. >> I have personal interestings on FS and if possible can we have a talk on >> this? >> >> Thanks >> Daniel Zeng >> >> ------------------------------ >> *From:* Lei Tang [mailto:lei.tlfly@gmail.com] >> *Sent:* Monday, November 02, 2009 10:27 AM >> *To:* freeswitch-users(a)lists.freeswitch.org >> *Subject:* [Freeswitch-users] Get error "415 Unsupported Media Type" >> whenreceiving call from softswitch >> >> Hi all, I get a "415 Unsupported Media Type" when FS receiving call from >> a softswitch. I captured some packets, It seems that the softswitch use >> SIP-I protocol, does FS can handle SIP-I message? >> >> ===here is the invite messagefrom softswitch >> INVITE sip:xxxxx@xxxx:5060;user=phone SIP/2.0 >> Contact: <sip:xxxx@xxxx:5061> >> MIME-version: 1.0 >> Content-Type: multipart/mixed;boundary=Alcatel-boundary >> To: <sip:xxxx@xxxx:5060;user=phone> >> From: xxxx<sip:xxxx@xxxxx:5061;user=phone>;tag=73D332463135364195291201 >> P-Asserted-Identity: <sip:xxxx@xxxxx:5061;user=phone> >> Allow: >> INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,COMET,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY,MESSAGE >> Supported: 100rel,timer,replaces,diversion >> Expires: 155 >> Session-Expires: 1800 >> Min-SE: 90 >> Call-ID: 01FD034872814000000230A1@sip-3 >> Max-Forwards: 70 >> CSeq: 1 INVITE >> Timestamp: 10645 >> Via: SIP/2.0/UDP xxxxx:5061;branch=z9hG4bK8E1558EA4F1BE09A2BCB669331A9AC7E >> Content-Length: 542 >> >> --Alcatel-boundary >> Content-Type: application/sdp >> >> v=0 >> o=- 5 8 IN IP4 xxxxxx >> s=SDP Data >> c=IN IP4 xxxxx >> t=0 0 >> m=audio 10266 RTP/AVP 0 8 96 110 111 112 113 3 97 >> a=rtpmap:110 speex/8000/1 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:96 TELEPHONE-EVENT/8000 >> a=fmtp:110 mode=3 >> a=rtpmap:111 speex/8000/1 >> a=rtpmap:3 GSM/8000 >> a=ptime:30 >> a=fmtp:111 mode=2 >> a=rtpmap:112 speex/8000/1 >> >> --Alcatel-boundary >> Content-Type: application/ISUP;version=N/A >> >> .. .... >> ...0.D... >> ...V..0......1...n..p... >> --Alcatel-boundary-- >> >> =====response message from FS >> SIP/2.0 415 Unsupported Media Type >> Via: SIP/2.0/UDP xxxxx:5061;branch=z9hG4bK8E1558EA4F1BE09A2BCB669331A9AC7E >> From: xxxxx <sip:xxxxx@xxxx:5061;user=phone>;tag=73D332463135364195291201 >> To: <sip:xxxxx@xxxxx:5060;user=phone>;tag=m6XS44BKF4pSS >> Call-ID: 01FD034872814000000230A1@sip-3 >> CSeq: 1 INVITE >> Timestamp: 10645 0.000000 >> User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460 >> Accept: application/sdp >> Accept-Encoding: >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> NOTIFY, REFER, UPDATE, REGISTER, INFO >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, refer >> Content-Length: 0 >> -- >> Lei.Tang >> lei.tlfly(a)gmail.com >> > > > > -- > Lei.Tang > lei.tlfly(a)gmail.com > -- Lei.Tang lei.tlfly(a)gmail.com

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Freeswitch seems to doesn't reknow dial tone after the first call using OpenZAP (analog spans)
by Albano Daniele Salvatore - Lavoro 02 Nov '09

02 Nov '09

Hi to all, this is my first message here! I'm trying to setup a freeswitch box using 1.0.4 version compiled from sources, ubuntu 8.04.03 lts with lastest updates and lastest zaptel modules/tools compiled from sources. As hardware i'm using an OpenVox A800P. My problem is the following: if i do a call trought openzap it works, but when i hang down and try a second call it doesn't work! Looking with ztmonitor and hearing recorded audio file it seems that freeswitch do the job on the first outgoing call but, on the second, it does nothing because it doesn't reknow dial tone (but it is there). If i call the freeswitch mananged number (i can answer or not, nothing change) i can do another call, but, in any case, the successive call doesn't start. Looking with ztmonitor 1 -v i see that before the first call i get the special dial tone (a continue 425 frequency to state that the line is free to dial sended by the provider) but after the first call is done, before the second, i didn't see any special dial tone ... that appears for less than a sec when freeswitch starts to compose the number. Here logs for first call ------ 2009-10-28 21:38:39.616112 [INFO] ozmod_zt.c:636 Setting echo cancel to 64 taps for 1:1 2009-10-28 21:38:39.616112 [DEBUG] ozmod_analog.c:450 Executing state handler on 1:1 for DIALING 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/xyz) Running State Change CS_INIT 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:481 (OpenZAP/1:1/xyz) State INIT 2009-10-28 21:38:39.616112 [DEBUG] mod_openzap.c:368 (OpenZAP/1:1/xyz) State Change CS_INIT -> CS_ROUTING 2009-10-28 21:38:39.616112 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/xyz [BREAK] 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:481 (OpenZAP/1:1/xyz) State INIT going to sleep 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/xyz) Running State Change CS_ROUTING 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:484 (OpenZAP/1:1/xyz) State ROUTING 2009-10-28 21:38:39.616112 [DEBUG] mod_openzap.c:391 OpenZAP/1:1/xyz CHANNEL ROUTING 2009-10-28 21:38:39.616112 [DEBUG] switch_ivr_originate.c:63 (OpenZAP/1:1/xyz) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-10-28 21:38:39.616112 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/xyz [BREAK] 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:484 (OpenZAP/1:1/xyz) State ROUTING going to sleep 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/xyz) Running State Change CS_CONSUME_MEDIA 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:503 (OpenZAP/1:1/xyz) State CONSUME_MEDIA 2009-10-28 21:38:40.475897 [DEBUG] ozmod_analog.c:655 Detected tone DIAL on 1:1 2009-10-28 21:38:40.475897 [DEBUG] mod_openzap.c:1216 got FXO sig 1:1 [TONE_DETECTED] ------ Here logs for second call ------ 2009-10-28 21:39:07.105916 [INFO] ozmod_zt.c:636 Setting echo cancel to 64 taps for 1:1 2009-10-28 21:39:07.105916 [DEBUG] ozmod_analog.c:450 Executing state handler on 1:1 for DIALING 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/xyz) Running State Change CS_INIT 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:481 (OpenZAP/1:1/xyz) State INIT 2009-10-28 21:39:07.118644 [DEBUG] mod_openzap.c:368 (OpenZAP/1:1/xyz) State Change CS_INIT -> CS_ROUTING 2009-10-28 21:39:07.118644 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/xyz [BREAK] 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:481 (OpenZAP/1:1/xyz) State INIT going to sleep 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/xyz) Running State Change CS_ROUTING 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:484 (OpenZAP/1:1/xyz) State ROUTING 2009-10-28 21:39:07.118644 [DEBUG] mod_openzap.c:391 OpenZAP/1:1/xyz CHANNEL ROUTING 2009-10-28 21:39:07.118644 [DEBUG] switch_ivr_originate.c:63 (OpenZAP/1:1/xyz) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-10-28 21:39:07.118644 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/xyz [BREAK] 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:484 (OpenZAP/1:1/xyz) State ROUTING going to sleep 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/xyz) Running State Change CS_CONSUME_MEDIA 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:503 (OpenZAP/1:1/xyz) State CONSUME_MEDIA ------ Following my configs! Really thanks for the help, Best Regards, Daniele --- ZAPTEL.CONF --- # Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: OPVXA1200/0 "OpenVox A1200P/A800P Board 1" (MASTER) fxsks=1 fxsks=2 fxsks=3 fxsks=4 fxoks=5 # Global data loadzone = it defaultzone = it --- ZT.CONF --- [defaults] codec_ms => 20 wink_ms => 150 flash_ms => 750 echo_cancel_level => 64 rxgain => 0.0 txgain => 0.0 --- OPENZAP.CONF --- [span zt] number => xyz fxo-channel => 1 number => xyz fxs-channel => 2 number => xyz fxs-channel => 3 number => xyz fxs-channel => 4 [span zt] fxs-channel => 5 --- TONES.CONF --- [it] generate-dial => %(200, 200, 425);%(600, 1000, 425) detect-dial => 425 generate-ring => %(1000, 4000, 425) detect-ring => 425 generate-busy => %(500, 500, 425) detect-busy => 425 generate-callwaiting-sas => %(400,100,425);%(250,100,425);%(150,0,425); detect-callwaiting-sas => 425 --- OPENZAP.CONF.XML --- <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/> <param name="hold-music" value="$${moh_uri}"/> </settings> <analog_spans> <span id="1"> <param name="tonegroup" value="it"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/> </span> <span id="2"> <param name="tonegroup" value="it"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/> </span> </analog_spans> </configuration>

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