[Freeswitch-users] mod_xml_ldap compile issue.
Hi, I am having an issue getting mod_xml_ldap to compile properly making all mod_xml_cdr making all mod_xml_ldap Creating mod_xml_ldap.la... /usr/bin/ld: /home/keithl/freeswitch/freeswitch.trunk/libs/openldap-2.4.19/libraries/liblutil/liblutil.a(sasl.o): relocation R_X86_64_32S against `.rodata' can not be used when making a shared object; recompile with -fPIC /home/keithl/freeswitch/freeswitch.trunk/libs/openldap-2.4.19/libraries/liblutil/liblutil.a: could not read symbols: Bad value collect2: ld returned 1 exit status cat: .libs/mod_xml_ldap.log: No such file or directory make[5]: *** [mod_xml_ldap.la] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_xml_ldap-all] Error 1 make[2]: *** [all-recursive] Error 1 I notice the openldap library has been bumped up to .19 - not sure if that may have anything to do with it. At revision 15995 on a 2.6.31-15-generic Ubuntu x86_64 GNU/Linux notebook. mod_ldap compiles OK, but mod_xml_ldap fails as per the above. What am I doing working here ? Best Regards Keith ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Adding headers to INFO messages for Advice of Charge on SNOM
Hi, I have tried maintaining charging information on a SNOM 300's display using 'display' - but found that the phone has some timer, whereby every 60 seconds it wipes out whatever happens to be on the display at that time and replaces is with the dialled number. So not a viable option as it impacts usability. Really annoying when the display was just updated with valuable information for the user and a split second later it gets replaced. [If somebody knows how to disable this behaviour - please do tell...] I see that SNOM supports a number of features for Advice of Charge. >From their Wiki: http://wiki.snom.com/Advice_of_charge_%28AOC%29_in_SIP Example of an SIP-Info Message: - INFO sip:b...@snom.com SIP/2.0 From: ;tag=5354n3 To: ;tag=33rfh3 CSeq: 23423 INFO Call-ID: 3452tw43dt354dm03 AOC: charging;state=active; charging-info=currency; currency=EUR; amount=2000; multiplier=0.001 Content-Length: 0 - So the question - Is there some method available today to add these additional 'new' headers to an INFO message I can send out to these phones? If not, I guess it's a matter of looking at enhancing the "case SWITCH_MESSAGE_INDICATE_DISPLAY" section in mod_sofia.c ? Best Regards Keith ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Calls dropping on SIP timer expiry due to UPDATE's being ignored.
Hi, Hope someone knows how I am able to get around this one. Here goes... Did an upgrade to trunk (from a July vintage build) last week and noticed calls out to a provider were now failing after about 30 seconds or so - post answer. Tried latest (15183) - same thing. Analysing, I see that I have multiple UPDATE messages now being sent to the provider, but no response being sent back to FS. So FS times out and eventually kills the call. Interestingly, it only drops the A-leg; the B-leg remains up till the B party hangs up. I cant recall seeing these UPDATE messages before... The intent of the UPDATE seems to be to send the callee name & number to the B-leg. If its the provider's sip stack that's broken w.r.t. handling UPDATE - is there any way to get around it by doing something in my config to ensure these UPDATE's are not 'triggered' ? Some traces below. Any suggestions welcomed... Best Regards Keith Pretoria, South Africa. -- send 1048 bytes to udp/[196.10.11.12]:5060 at 13:24:04.249269: INVITE sip:27835551...@196.10.11.12 SIP/2.0 Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bKyQepDXQ5H8g5m Max-Forwards: 67 From: "Keith PhoneADSL" ;tag=Upa3NvXpBB1eF To: Call-ID: d821359d-38e7-122d-a38e-002264cc9b93 CSeq: 121947386 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, UPDATE, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 247 X-Actually-Support: UPDATE Remote-Party-ID: "Keith PhoneADSL" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1256118582 1256118583 IN IP4 10.17.10.10 s=FreeSWITCH c=IN IP4 10.17.10.10 t=0 0 m=audio 12862 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 2009-10-21 15:24:04.248448 [DEBUG] sofia.c:3493 Channel sofia/vvrf/2783555 entering state [calling][0] recv 601 bytes from udp/[196.10.11.12]:5060 at 13:24:04.307690: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bKyQepDXQ5H8g5m From: "Keith PhoneADSL" ;tag=Upa3NvXpBB1eF To: ;tag=GR52RWG346-34 Call-ID: d821359d-38e7-122d-a38e-002264cc9b93 CSeq: 121947386 INVITE Contact: "vprov C5CM" User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M Allow-Events: talk Allow-Events: refer Content-Disposition: session X-Actually-Support: UPDATE Remote-Party-ID: "Keith PhoneADSL" ;party=calling;screen=yes;privacy=off Content-Length: 0 recv 879 bytes from udp/[196.10.11.12]:5060 at 13:24:08.508162: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.17.10.10;rport;branch=z9hG4bKyQepDXQ5H8g5m From: "Keith PhoneADSL" ;tag=Upa3NvXpBB1eF To: ;tag=GR52RWG346-34 Call-ID: d821359d-38e7-122d-a38e-002264cc9b93 CSeq: 121947386 INVITE Contact: "vprov C5CM" User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15183M Allow-Events: talk Allow-Events: refer Content-Disposition: session X-Actually-Support: UPDATE Remote-Party-ID: "Keith PhoneADSL" ;party=calling;screen=yes;privacy=off Content-Type: application/sdp Content-Length: 233 v=0 o=Clarent 152602 152603 IN IP4 196.10.11.15 s=Clarent C5CM c=IN IP4 196.10.11.15 t=0 0 m=audio 5230 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=ptime:20 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2009-10-21 15:24:08.507506 [DEBUG] sofia.c:3493 Channel sofia/vvrf/2783555 entering state [proceeding][183] 2009-10-21 15:24:08.507506 [DEBUG] sofia.c:3500 Remote SDP: v=0 o=Clarent 152602 152603 IN IP4 196.10.11.15 s=Clarent C5CM c=IN IP4 196.10.11.15 t=0 0 m=audio 5230 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2009-10-21 15:24:08.507506 [DEBUG] sofia_glue.c:3144 Audio Codec Compare [G729:18:8000:20]/[G729:18:8000:20] 2009-10-21 15:24:08.508561 [DEBUG] sofia_glue.c:2102 Set Codec sofia/vvrf/2783555 G729/8000 20 ms 160 samples 2009-10-21 15:24:08.508561 [DEBUG] sofia_glue.c:3104 Set 2833 dtmf payload to 101 2009-10-21 15:24:08.508561 [DEBUG] sofia_glue.c:2336 AUDIO RTP [sofia/vvrf/2783555] 10.17.10.10 port 12862 -> 196.10.11.15 port 5230 codec:
Re: [Freeswitch-users] Ringback when running G729 codec
Hi, I am testing a trunk version from the weekend and have moved configs over from a box pre 1.04. Looks like Rev: 15011, but the box was built by someone else - so not 100% certain of exact number. With : I am not getting ringback after the 'pre_answer'. Was testing with G722 and G729 on the inbound leg. I wonder if it's related to this? Best Regards Keith On Fri, 2009-09-25 at 13:30 -0500, Anthony Minessale wrote: > fixed in latest trunk, > please test > thank you > > On Fri, Sep 25, 2009 at 6:17 AM, Hound Dog wrote: > Hi, > > very happy with freeswitch as a PBX/softswitch/SBC system its > working solidly for a few weeks now - just great > > > I have a question regarding ringback tones - custom or regular > - I cant get freeswitch to send ringback using G729 > > I used the following settings ( it will just play one of the > IVR prompts as ringback (filename > ivr-to_repeat_these_options) - I took it from the G729 > encoded files package , it has PCMA , G729 G723 extensions ) > > > expression="^4420885767(0\d)$"> > > data="ringback=/usr/local/freeswitch/sounds/en/us/callie/raw_files/ivr/ivr-to_repeat_these_options"/> > data="instant_ringback=true"/> > > > > > when I call with G711 enabled , it plays the file no problems > - see log > > 2009-09-25 11:29:58.641361 [DEBUG] sofia.c:3289 Channel > sofia/external/442078562...@80.80.80.80 entering state > [early][183] > 2009-09-25 11:29:58.641361 [DEBUG] switch_core_session.c:630 > Send signal sofia/external/442078562...@80.80.80.80 [BREAK] > 2009-09-25 11:29:58.641361 [DEBUG] switch_ivr_originate.c:1718 > Raw Codec Activation Success l...@8000hz 1 channel 20ms > 2009-09-25 11:29:58.641361 [DEBUG] switch_ivr_originate.c:1745 > Play Ringback File > > [/usr/local/freeswitch/sounds/en/us/callie/raw_files/ivr/ivr-to_repeat_these_options.PCMA] > 2009-09-25 11:29:58.641361 [INFO] mod_native_file.c:82 Opening > File > > [/usr/local/freeswitch/sounds/en/us/callie/raw_files/ivr/ivr-to_repeat_these_options.PCMA] > 8000hz > 2009-09-25 11:29:58.889369 [DEBUG] sofia.c:3289 Channel > sofia/internal/sip:1...@82.80.131.233:40505 entering state > [proceeding][180] > > > when I call with G729 only - I get silence , and freeswitch > only send the comfort noise packet and no RTP , see log > > 2009-09-25 11:28:57.437537 [DEBUG] sofia.c:3289 Channel > sofia/external/442078562...@80.80.80.80 entering state > [early][183] > 2009-09-25 11:28:57.437537 [DEBUG] switch_core_session.c:630 > Send signal sofia/external/442078562...@80.80.80.80 [BREAK] > 2009-09-25 11:28:57.685536 [DEBUG] sofia.c:3289 Channel > sofia/internal/sip:1...@82.80.131.233:40505 entering state > [proceeding][180] > 2009-09-25 11:28:57.685536 [NOTICE] sofia.c:3353 Ring-Ready > sofia/internal/sip:1...@82.80.131.233:40505! > > mod_native_file works well for me when used in applications > and plays G729 files no problem > > any ideas why is that happening , any suggestions on how to > resolve ? > > thanks > Ori > > > > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Async JS functions?
Hi Nicolas, I wonder if session.execute( "sched_broadcast"", "+1 /path/file.wav aleg") followed immediately by your 'new Session' and 'bridge' would do the trick ? Not sure if/how "sched_broadcast" functions when the call has not yet been bridged though... Let us know.. Best Regards Keith From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Nicolas Brenner Sent: 30 July 2009 20:31 To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Async JS functions? Hi, I have a small JS script that calls a phonenumber, when the call is answered it plays a wave file, then it calls a second phonenumber and bridges the calls. Is it possible to make wave-playing async, so that the second call is generated as soon as the first is picked up? Right now the wave file takes about 2 secs to play, but I need to extend that time, and I don't want to delay the second call. Thanks! Nicolas ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Distortion on approx first 200ms of G722prompts on DECT based CPE
Hi, [Thanks for the advice Anthony] I tried "send_silence_when_idle=true " and restarted, but did not notice any change/improvement. But I had limited time to test, so will need to test more thoroughly with this CPE. An additional test was to configure the following media path scenarios: A: Policom<->FS<->DECT CPE B: Policom<->DECT CPE (media not via FS) [The only change was to add ) In scenario A, whenever the Policom's VAD kicked in, it resulted in the first 200ms of the restarting audio being distorted in the DECT CPE. In scenario B, no problems when the Policom's VAD kicked in. I also addressed the issue to the CPE vendor yesterday, who responded today : "We have observed the RTP-stream and found the following : - the RTP-stream is completely interrupted in the pause between the announcements - no RTP-packets are send at all during these pauses - jitter buffer runs empty and our device will automatically mute the connection when no packets comes in - last RTP-packets before pauses don't contain the info about comfort-noise We would like to ask you the following : - please check if it's possible to send dummy-packets during pause instead of sending no packets at all " In scanning the wiki on the topic of CNG, I found at http://wiki.freeswitch.org/wiki/VAD_and_CNG : "In FreeSWITCH the CNG options select whether or not FreeSWITCH will generate CN RTP packets. suppress-cng sofia profile option and suppress_cng channel variable used to set of this setting. When both sides are supporting RFC3389 (they agree in SDP message exchange, rtpmap:13), FreeSWITCH will send CN packets. Note: Allowing CNG in FreeSWITCH does not mean it will generate any comfort noise into the media channel. In case one of the parties in bridge do not handle VAD and asynchronous RTP media, there should be an issue as the one might think hearing perfect silence and might think the connection has been dropped. Another example is when on one side is Asterisk or CallWeaver. For handling these endpoints, there has been added (r9543) a new channel variable: bridge_generate_comfort_noise which will generate fake audio" So the options here seem to be : a) Get FS to send CNG packet(s) before going into 'pauses'. From the vendor's analysis they are not seeing this when testing (FYI, these observations were made calling into vmail). Could this be because the CPE is perhaps not supporting RFC3389 - FS did not see the rtpmap:13 in the SDP ? b) Make sure FS keeps sending packets during pauses and silence. I am not clear on the difference between the 'send_silence_when_idle=true' and 'bridge_generate_comfort_noise=true' options. Ideally I would still want to leverage VAD, but then need the CNG messages to be forwarded in scenarios where I have media passing through FS on a call between two customers and when a customer is interacting with vmail (or other IVR type application). Any advise appreciated. Best Regards Keith From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 28 July 2009 02:06 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Distortion on approx first 200ms of G722prompts on DECT based CPE you can set the global var send_silence_when_idle=true in vars.xml On Mon, Jul 27, 2009 at 4:23 PM, Keith Laaks wrote: Hi All, I am testing a range of G722 capable DECT based CPE. With one range, I have noticed that the first 200ms or so of each separate prompt file being played back is played out distorted from the DECT handset. When having a normal conversation, the quality is excellent, but when accessing your vmail, all the individual audio files making up the menu choices exhibit the distortion, which is pretty annoying. The same unit using G729, alaw or ulaw works 100%. I wonder if anybody else has uncounted this issue? My guess at this point - There may be a short break in the RTP between the separate files being played out by FS that makes up any menu. During this time the DECT handset's AGC probably goes to MAX amplification (as its not receiving any input during the short break in RTP). Then, when the RTP returns at the start of the next file, the AGC boosts the audio into clipping zone and takes 200ms to dampen down back to normal good levels. Looks like in these devices the G722 encode/decode is actually done in the DECT handset and not the voip-base unit. Is there any parameter that can be set in FS to ensure that the RTP keeps flowing, sending 'silence' between prompts ? Would be interesting to validate the above 'guess'. Best Regards Keith ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http:/
[Freeswitch-users] Distortion on approx first 200ms of G722 prompts on DECT based CPE
Hi All, I am testing a range of G722 capable DECT based CPE. With one range, I have noticed that the first 200ms or so of each separate prompt file being played back is played out distorted from the DECT handset. When having a normal conversation, the quality is excellent, but when accessing your vmail, all the individual audio files making up the menu choices exhibit the distortion, which is pretty annoying. The same unit using G729, alaw or ulaw works 100%. I wonder if anybody else has uncounted this issue? My guess at this point - There may be a short break in the RTP between the separate files being played out by FS that makes up any menu. During this time the DECT handset's AGC probably goes to MAX amplification (as its not receiving any input during the short break in RTP). Then, when the RTP returns at the start of the next file, the AGC boosts the audio into clipping zone and takes 200ms to dampen down back to normal good levels. Looks like in these devices the G722 encode/decode is actually done in the DECT handset and not the voip-base unit. Is there any parameter that can be set in FS to ensure that the RTP keeps flowing, sending 'silence' between prompts ? Would be interesting to validate the above 'guess'. Best Regards Keith ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Freeswitch taking too long to start up
Hi, Try starting using the -nonat switch. Best Regards Keith From: Muhammad Shahzad [mailto:shaherya...@googlemail.com] Sent: 02 June 2009 14:39 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch taking too long to start up Yes, this resolves the problem. Thank you. On Tue, Jun 2, 2009 at 5:27 PM, dujinfang wrote: Actually Brain mentioned that you can comment out switch_nat_init(); in switch_core.c On Jun 2, 2009, at 7:08 PM, Nik Middleton wrote: As I understand it, a new ‘feature’ was added over the weekend to resolve NAT. If you’re firewall is not allowing ICMP then FS waits until it times out. At this time there is no option to disable it. Regards From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Muhammad Shahzad Sent: 02 June 2009 11:40 To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Freeswitch taking too long to start up Hi, I have just upgraded Freeswitch from svn revision 12432 to 13544. I am using 32bit CentOS 5.3, "make current" command completes successfully without any errors but when i start freeswitch it take considerable time (roughly 90 - 120 seconds) to start up. During this time no message is display on console. Once successfully started, it works fine. However, this initial delay is really annoying. Is there anyway to reduce/remove this delay? Thank you. -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Javascript, Hanguphooks, CDRs and User Variables.
Hi,
Here is what I am trying to accomplish:
//--- completecall.js trigged via api_hangup_hook
use("CURL");
const loglevel='notice';
var uuid = request.getHeader("Core-UUID");
var billmsec = request.getHeader("billmsec");
var urlrequest = "UUID=" + uuid + "&billmsec=" + billmsec;
function reply_callback(string, arg) {
string = string.substring(string.search("API")+3);
string = string.substring(0,string.search("<"));
var splits = string.split("~");
var i = 0;
var length = splits.length;
for (i=0; i < length; i++) {
var fv=splits[i].split("=");
console_log(loglevel, "setting: " + fv[0] + " = " + fv[1] + "\n");
//session.setVariable(fv[0],fv[1]); // Cant use set here as the
session is dead by now - the call has been terminated
}
return true;
}
var curl = new CURL();
console_log(loglevel,"-- completecall.js ->" + url + "?" + urlrequest +
"\n");
var env = request.dumpENV("text"); // debug
console_log("ENV:\n", env + "\n"); // debug
curl.run("POST", "http://127.0.0.1:10502/rate";, urlrequest,
reply_callback, "CBrate\n");
//returns string like APIcharge=10.20
exit();
So the trick is that I am accessing an external system via CURL where
the call rate is calculated and returned (based on the call duration and
uuid).
Now I need to use a 'setVariable' to get it back as one of the
parameters that the cdr module can write out for me.
Note that in the above, the external system will return 'charge=value'.
I need to set the variable 'charge' to the value in 'value'.
Then using the config below, 'charge' can be written out as one of the
cdr fields.
"${accountcode}","${billsec}","${charge}","${sip_req_user}","
${hangup_cause}","${uuid}","${bleg_uuid}"
Best Regards
Keith
From: Anthony Minessale [mailto:anthony.miness...@gmail.com]
Sent: 31 March 2009 00:13
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Javascript, Hanguphooks,CDRs and User
Variables.
in your script called via api_hangup_hook:
var env = request.dumpENV("text");
consoleLog("info", env);
all those vars are there for you, you can get the individually with
var hval = request.getHeader("some_header");
2009/3/30 Keith Laaks
Hi,
I have an application where my Javascript hanguphook code calculates a
value (e.g. the cost of the call which can only be calculated post
hangup) and I need to have that value appear as a field in the cdrs.
As the 'session' object is no longer available for javascript logic post
hangup, I can't figure out how to 'set' a user variable post hangup,
such that it can be written to the cdr when the state changes from
CS_HANGUP -> CS_REPORTING. Maybe it's just not possible..? It would
be a pity to have to resort to writing out cdrs from the javascript
itself and duplicating what fs does so well already.
Any advice / suggestions would be appreciated.
Best Regards
Keith
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Anthony Minessale II
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[Freeswitch-users] Javascript, Hanguphooks, CDRs and User Variables.
Hi, I have an application where my Javascript hanguphook code calculates a value (e.g. the cost of the call which can only be calculated post hangup) and I need to have that value appear as a field in the cdrs. As the 'session' object is no longer available for javascript logic post hangup, I can't figure out how to 'set' a user variable post hangup, such that it can be written to the cdr when the state changes from CS_HANGUP -> CS_REPORTING. Maybe it's just not possible..? It would be a pity to have to resort to writing out cdrs from the javascript itself and duplicating what fs does so well already. Any advice / suggestions would be appreciated. Best Regards Keith ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Module Mod Native File - How to handle multiple rates under the same codec.
Hi,
To minimize/eliminate transcoding, I am using mod native file, with a
set of transcoded prompts with the appropriate set of file extensions.
Everything works as advertised when using the traditional codecs such as
pcma, g729, gsm.
But speex support is a bit of a challenge.
Freeswitch supports: sp...@8000h@20i, sp...@16000h@20i,
sp...@32000h@20i.
When I setup calls using these various codec flavors, I can see using
info (and of course detect by ear) that indeed the call is running at
the different 'rates'.
I see that regardless of which one I use, the "variable_read_codec" and
"variable_write_codec" remains "SPEEX", but depending on the flavor, a
"variable_read_rate" and "variable_write_rate" of either 8000,16000 or
32000.
But when I try play a file when in 16000 or 32000, I get:
2009-03-15 16:31:13 [INFO] mod_native_file.c:81 native_file_file_open()
Opening File
[/usr/local/freeswitch/sounds/en/us/callie/all/16000/SUCCESS.SPEEX]
8000hz
2009-03-15 16:31:13 [WARNING] switch_core_file.c:119
switch_core_perform_file_open() Sample rate doesn't match.
I created my SPEEX files using: speexenc -w
(note the -w option for 16kHz wideband)
So, even though the call is setup using a wideband 16kHz codec, it
appears that mod native file is expecting a 8kHz file for all the SPEEX
flavors.
What am I missing here? Is this module limited to 8KHz rates? I am on
1.0.trunk (12530M).
I have not yet looked at these codecs: g7...@16000h, g7...@32000h,
c...@32000h, c...@48000h, but as these also have multiple rates for the
same codec - I expect same issue.
I am using ${ variable_read_rate } in the filename path, so fs looks at
a set of files encoded with a matching sample rate.
But looks like it's always looking for a 8KHz file.
If you have had any experience with this, please let me have your
advice.
Thanks
Keith
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[Freeswitch-users] Core Dump on receiving a call from device with 'broken' G.722 codec.
Hi, I am on fs 1.0.trunk (12530M) testing G.722 and found that when using a 'broken' configuration from a softphone configured for G.722, I get the warning on the cli: "We were told to use ptime 20 but what they meant to say was 820 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.. " when fs gets the invite, but then does a core dump when it tries to: Below are some of the traces and info output from before the core dump happens. I see this when I run gdb on the dumpfile. #0 0xb7e194d4 in switch_ivr_originate (session=0xb74640a8, bleg=0xb572b0b0, cause=0xb572b0ac, bridgeto=0xb74a1b18 "sofia/voxwan/8154%172.16.1.3", timelimit_sec=30, table=0xb7efcfc0, cid_name_override=0x0, cid_num_override=0x0, caller_profile_override=0x0, ovars=0x0, flags=) at src/switch_ivr_originate.c:1609 1609if (switch_core_codec_init(&write_codec, (gdb) frame 1 #1 0xb6df48f5 in ?? () from /usr/local/freeswitch/mod/mod_dptools.so I wonder if anybody else has seen this behavior? This happens when the destination phone is also G.722 capable (policom). If I change the "frame per packet" setting in the softphone to 2 - All works OK (but the default is 1 - so cant risk allowing G.722 if it's going to core dump fs if a user make a wrong configuration) Best Regards Keith * 2009-03-16 14:37:58 [NOTICE] switch_channel.c:592 switch_channel_set_name() New Channel sofia/sprof1/27879998...@196.99.88.77 [47e0c972-1227-11de-8b8e-1789e43c417d] <.> 2009-03-16 14:37:58 [INFO] mod_sofia.c:1310 sofia_receive_message() Asked to send early media by sofia/sprof1/27879998...@196.99.88.77 2009-03-16 14:37:58 [NOTICE] sofia_glue.c:2245 sofia_glue_tech_media() Pre-Answer sofia/sprof1/27879998...@196.99.88.77! 2009-03-16 14:37:58 [INFO] mod_sofia.c:1351 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1237190146 1237190147 IN IP4 196.99.88.77 s=FreeSWITCH c=IN IP4 196.99.88.77 t=0 0 m=audio 16932 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-03-16 14:37:58 [INFO] switch_rtp.c:1441 rtp_common_read() Auto Changing port from 172.16.0.63:29081 to 196.22.33.44:10634 2009-03-16 14:37:59 [NOTICE] checktalktime.js:1 console_log() -- checktalktime.js -- <.. In this js I do http call to collect maximum talktime allowed ..> 2009-03-16 14:37:59 [NOTICE] checktalktime.js:1 console_log() schedparms=+3600 tbhangupwarn XML hangupwarn 2009-03-16 14:37:59 [NOTICE] switch_ivr.c:1345 switch_ivr_session_transfer() Transfer sofia/sprof1/27879998...@196.99.88.77 to xml[27879998...@e164route] 2009-03-16 14:37:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing MeMe->27879998154 in context e164route 2009-03-16 14:37:59 [INFO] mod_dptools.c:945 info_function() CHANNEL_DATA: Event-Name: [CHANNEL_DATA] Core-UUID: [e57400ea-1223-11de-8b8e-1789e43c417d] FreeSWITCH-Hostname: [myfsbox] FreeSWITCH-IPv4: [196.99.88.77] FreeSWITCH-IPv6: [::1] Event-Date-Local: [2009-03-16 14:37:59] Event-Date-GMT: [Mon, 16 Mar 2009 12:37:59 GMT] Event-Date-Timestamp: [1237207079387869] Event-Calling-File: [mod_dptools.c] Event-Calling-Function: [info_function] Event-Calling-Line-Number: [941] Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [sofia/sprof1/27879998...@196.99.88.77] Unique-ID: [47e0c972-1227-11de-8b8e-1789e43c417d] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [early] Channel-Read-Codec-Name: [G722] Channel-Read-Codec-Rate: [16000] Channel-Write-Codec-Name: [G722] Channel-Write-Codec-Rate: [16000] Caller-Username: [27879998182] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [MeMe] Caller-Caller-ID-Number: [27879998182] Caller-Network-Addr: [196.22.33.44] Caller-Destination-Number: [27879998154] Caller-Unique-ID: [47e0c972-1227-11de-8b8e-1789e43c417d] Caller-Source: [mod_sofia] Caller-Context: [e164route] Caller-RDNIS: [27879998154] Caller-Channel-Name: [sofia/sprof1/27879998...@196.99.88.77] Caller-Profile-Index: [4] Caller-Profile-Created-Time: [1237207079387869] Caller-Channel-Created-Time: [1237207078659653] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [1237207078679638] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] variable_sip_received_ip: [196.22.33.44] variable_sip_received_port: [36745] variable_sip_via_protocol: [udp] variable_sip_author
