December 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] valet_park help needed
by Nitin Tomer 05 Dec '12

05 Dec '12

Hi Brian, I've posted the contents of dialplan. Please tell me what I am doing wrong. Regards Nitin Brian West <brian(a)freeswitch.org> wrote: >If the caller is hearing the location your doing the transfer wrong! > >Sent from my iPhone > >On Dec 2, 2012, at 1:52 PM, Abaci <abaci64(a)gmail.com> wrote: > > you can use the valet_announce_slot channel variable to disable >annoucement of the parking slot. see >http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_valet_park#Channel_Var… > >On 11/29/2012 9:59 PM, Michael Collins wrote: > >As far as I know you can't suppress the announcement of the location to the >caller. > >For the event socket you have a lot of homework to do. I recommend: >FS Book<https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-tel…>, >chapter 9 >FS Cookbook <http://link.packtpub.com/nuIOlX>, chapter 4 >Wiki event socket (see link on left) > >For a really quick dive into what events look like: >launch fs_cli and type: >/log 0 >/events plain all > >You'll see EVERY event that the system throws. Try this to narrow it down >just to valet events: > >/filter Event-Class valet_parking::info > >I typed most of this off the top of my head, so standard >disclaimer<http://wiki.freeswitch.org/wiki/Reporting_Bugs#Standard_Disclaimer>applies. >Hope this helps you get started! >-MC > > >On Thu, Nov 29, 2012 at 6:09 PM, Nitin Tomer <ntomer(a)newgen.co.in> wrote: > >> Dear Michael, >> >> Thanks for your help. >> >> About thsis - "Using 'auto in' the system will announce the parking >> location. If you are sending a call in from an IVR then the caller will >> hear their park location. The only way to know where the call went would be >> to watch the event socket for relevant valet events. " >> >> Yes, right now the extension where call is parked, is announced to the >> caller. I don't want that to happen. I don't want it announced to caller, >> rather I want it retruned to me, so that I can store it in database. So >> that my agents can pick the call after seeing the extension where it is >> parked. >> >> Please tell me more details about how to watch the event socket for valet >> events. >> >> Regards >> >> Nitin >> >> On Friday, 30-11-2012 on 6:38 Michael Collins wrote: >> >> Answers inline... >> >> On Thu, Nov 29, 2012 at 2:15 AM, Nitin Tomer <ntomer(a)newgen.co.in> wrote: >> >>> Hi, >>> >>> >>> >>> I am using valet_park. I’ve configure a IVR menu of an extension, based >>> on user’s input call is forwarded to other extensions. >>> >>> >>> >>> Extension on which end-users will call – >>> >>> >>> >>> <extension name="ivr_nitin"> >>> <condition field="destination_number" expression="^5002$"> >>> <action application="answer"/> >>> <action application="sleep" data="500"/> >>> <action application="ivr" data="nitin_ivr"/> >>> </condition> >>> </extension> >>> >>> >>> >>> The IVR configuration XML is – >>> >>> >>> >>> <include> >>> <menu name="nitin_ivr" >>> greet-long="say:Welcome to Newgen General Insurance >>> Company. Press 1 for Changing Address, 2 for Changing Nominee or 3 for >>> Close Policy." >>> greet-short="say:Welcome to Newgen. Press 1 for Changing >>> Address, 2 for Changing Nominee or 3 for Close Policy." >>> invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" >>> exit-sound="voicemail/vm-goodbye.wav" >>> confirm-macro="" >>> confirm-key="" >>> tts-engine="flite" >>> tts-voice="slt" >>> confirm-attempts="3" >>> timeout="3000" >>> inter-digit-timeout="2000" >>> max-failures="3" >>> max-timeouts="3" >>> digit-len="4"> >>> >>> <entry action="menu-exec-app" digits="1" param="transfer >>> 450 XML default"/>  >>> <entry action="menu-exec-app" digits="2" param="transfer >>> 451 XML default"/>  >>> <entry action="menu-exec-app" digits="3" param="transfer >>> 452 XML default"/>  >>> >>> </menu> >>> </include> >>> >>> Once user presses “1”, call is forwarded to 450, for this extension >>> dialplan entry is – >>> >>> >>> >>> <extension> >>> <condition field="destination_number" expression="^(450)$"> >>> <action application="play_and_get_digits" data="10 16 3 3000 # >>> say:'Press your account number, followed by hash key' say:'Wrong Input' res >>> \d+" /> >>> <action application="phrase" data="spell,${res}"/> >>> <action application="db" data="insert/testapp/newcall1/${res}" >>> /> >>> <action application="set" data="caller_id_name=Change >>> Address Request" /> >>> <action application="set" data="call_timeout=60" /> >>> <action application="set" data="originate_timeout=60" /> >>> <action application="valet_park" data="my_lot auto in 8501 >>> 8599" /> >>> >>> </condition> >>> </extension> >>> >>> >>> >>> Here, the call is parked at any available extension between 8501 to 8599. >>> >>> >>> >>> Then I’ve set up an extension to pick up calls – >>> >>> >>> >>> <extension name="park-out"> >>> <condition field="destination_number" expression="^(85\d\d)$"> >>> <action application="answer"/> >>> <action application="valet_park" data="my_lot $1"/> >>> </condition> >>> </extension> >>> >>> >>> >>> I have a few questions – >>> >>> >>> >>> 1. Valet_park parks the call on any available extension between >>> 8501 to 8599 (<action application="valet_park" data="my_lot auto in 8501 >>> 8599" />). Is there any way to let me know on which extension the call have >>> been parked? >>> >> Using 'auto in' the system will announce the parking location. If you are >> sending a call in from an IVR then the caller will hear their park >> location. The only way to know where the call went would be to watch the >> event socket for relevant valet events. >> >>> 2. How can I get the number from which call was made in >>> extension 450. The idea is to use the caller number as key and entered >>> value as value while making entry in database (<action application="db" >>> data="insert/testapp/newcall1/${res}" />)? >>> >> Do you mean the caller id number? That's literally in channel variable >> ${caller_id_number} >> >>> 3. If two users call on extension 5002 (where IVR menu is >>> played), what will happen? Will the second user have to wait for first to >>> finish or whether both will be connected parallel? >>> >> Both can be in the IVR at the same time and they won't affect each other >> at all. >> >> -MC >> >>> >>> >>> Please help me out. >>> >>> >>> >>> Regards >>> >>> >>> >>> Nitin >>> >>> Disclaimer :- This e-mail and any attachment may contain confidential, >>> proprietary or legally privileged information. If you are not the original >>> intended recipient and have erroneously received this message, you are >>> prohibited from using, copying, altering or disclosing the content of this >>> message. Please delete it immediately and notify the sender. Newgen >>> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or >>> damage arising from the use of the information transmitted by this email >>> including damages from virus and further acknowledges that no binding >>> nature of the message shall be implied or assumed unless the sender does so >>> expressly with due authority of NSTL. >>> >>> >>> _________________________________________________________________________ >>> 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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> >> Disclaimer :- This e-mail and any attachment may contain confidential, >> proprietary or legally privileged information. If you are not the original >> intended recipient and have erroneously received this message, you are >> prohibited from using, copying, altering or disclosing the content of this >> message. Please delete it immediately and notify the sender. Newgen >> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or >> damage arising from the use of the information transmitted by this email >> including damages from virus and further acknowledges that no binding >> nature of the message shall be implied or assumed unless the sender does so >> expressly with due authority of NSTL. >> >> >> _________________________________________________________________________ >> 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 >> >> > > >-- >Michael S Collins >Twitter: @mercutioviz >http://www.FreeSWITCH.org >http://www.ClueCon.com >http://www.OSTAG.org > > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting >Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication >Serverhttp://www.cudatel.com > >Official FreeSWITCH >Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.… > >FreeSWITCH-users mailing >listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershtt… > > >_________________________________________________________________________ >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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having problem using java with esl library
by shaik bawajan 05 Dec '12

05 Dec '12

Hi, I am new to freeswitch and am using java app with esl library to make outbound calls and am getting the error is "Create additional event dispatch thread 2". I saw in the freeswitch logs is that - Event 0 Blocking and its keeps printing. Kindly let me know where am doing mistake and how to resolve it. Thanks in Advance, Bawajan

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Can not use Python ESL with Django and apache web server
by chaiyawut.so 05 Dec '12

05 Dec '12

I tried freeswitch python ESL library with Django and apache mod_wsgi today. I already compiled and installed pymod in Freeswitch source folder. My freeswitch server runs as root and my DJnago web server runs as www-data and mod_wsgi. When used command "ESLconnection("localhost", "8021", "ClueCon")" in my views.py to connect to freeswitch server, django froze permanently and it said waiting for the request... -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-not-use-Python-ESL-with-D… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] Explanation of rtp_audio_in / out fields in XML CDR's?
by Peter Olsson 05 Dec '12

05 Dec '12

1) I guess this is for how many packets that has been skipped for incoming and/or outgoing traffic. For instance, if you get the same packet twice with the same timestamp, this would increase the skipped_packet_count. 2) There is no reason that in and out is always the same. For instance, if you mute your phone, it might stop sending RTP frames, but you might still receive frames. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Mick Stevens Skickat: den 5 december 2012 11:54 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Explanation of rtp_audio_in / out fields in XML CDR's? Hey Team, Sorry to go on about this, but really keen to understand these RTP fields in the XML CDR's better... 1) At the risk of sounding too desperate... Please, please, please can someone explain to me what the <rtp_audio_in_skip_packet_count> & <rtp_audio_out_skip_packet_count> fields indicates? 2) Also, for example in the extract below, why are the rtp_audio_in & rtp_audio_out bytes / packet counts not the same? Is this a red herring based on some codec interworking issue or similar, an indication of packet loss or expected/normal behaviour? <rtp_audio_in_raw_bytes>436848</rtp_audio_in_raw_bytes> <rtp_audio_in_media_bytes>436536</rtp_audio_in_media_bytes> <rtp_audio_in_packet_count>2562</rtp_audio_in_packet_count> <rtp_audio_in_media_packet_count>2538</rtp_audio_in_media_packet_count> <rtp_audio_in_skip_packet_count>1904</rtp_audio_in_skip_packet_count> <rtp_audio_in_jb_packet_count>0</rtp_audio_in_jb_packet_count> <rtp_audio_in_dtmf_packet_count>0</rtp_audio_in_dtmf_packet_count> <rtp_audio_in_cng_packet_count>24</rtp_audio_in_cng_packet_count> <rtp_audio_in_flush_packet_count>0</rtp_audio_in_flush_packet_count> <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size> <rtp_audio_out_raw_bytes>624570</rtp_audio_out_raw_bytes> <rtp_audio_out_media_bytes>624360</rtp_audio_out_media_bytes> <rtp_audio_out_packet_count>3645</rtp_audio_out_packet_count> <rtp_audio_out_media_packet_count>3630</rtp_audio_out_media_packet_count> <rtp_audio_out_skip_packet_count>0</rtp_audio_out_skip_packet_count> <rtp_audio_out_dtmf_packet_count>0</rtp_audio_out_dtmf_packet_count> <rtp_audio_out_cng_packet_count>15</rtp_audio_out_cng_packet_count> Please help! [:) happy] Rgds, Mick Tel/SMS. +44(0)7967 594432 Fax. +44(0)7053 452429 Email/IM. mickstevens(a)yahoo.com Skype: mick_stevens www.facebook.com/mickstevens www.twitter.com/mickstevens ________________________________ From: Mick Stevens <mickstevens(a)yahoo.com<mailto:mickstevens@yahoo.com>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Sent: Thursday, 29 November 2012, 20:35 Subject: Re: [Freeswitch-users] Explanation of rtp_audio_in / out fields in XML CDR's? Hi Michael & Co, Many thanks for the prompt & positive response! Yes, more than happy to collate feedback from the global team & update the wiki accordingly (I'd like to be able to document what the field values between the > < indicate as well as just explain the field names if possible...) +anything else I can do to to contribute to the project...? Yes, now the wiki is back up I've managed to work out that cng_packet = comfort noise generation! Also from the wiki, possibly that the jb in rtp_audio_in_jb_packet_count & rtp_audio_in_largest_jb_size = jitter buffer? (apologies if everybody else already knows this!). To provide the background to my original enquiry, I'm trying to identify if the rtp_audio_in_skip_packet_count in the following XML CDR extract is indicative of packet loss, or perhaps more accurately noticeable audio loss (as the packets haven't been lost, just ignored/"skipped"?: If anybody knows please speak up! [:) happy] <rtp_audio_in_raw_bytes>1565066</rtp_audio_in_raw_bytes> <rtp_audio_in_media_bytes>1564856</rtp_audio_in_media_bytes> <rtp_audio_in_packet_count>9113</rtp_audio_in_packet_count> <rtp_audio_in_media_packet_count>9098</rtp_audio_in_media_packet_count> <rtp_audio_in_skip_packet_count>1609</rtp_audio_in_skip_packet_count> <rtp_audio_in_jb_packet_count>0</rtp_audio_in_jb_packet_count> <rtp_audio_in_dtmf_packet_count>0</rtp_audio_in_dtmf_packet_count> <rtp_audio_in_cng_packet_count>15</rtp_audio_in_cng_packet_count> <rtp_audio_in_flush_packet_count>0</rtp_audio_in_flush_packet_count> <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size> Thank you in anticipation of enlightenment! #[:x lovestruck]FreeSWITCH Rgds, Mick Tel/SMS. +44(0)7967 594432 Fax. +44(0)7053 452429 Email/IM. mickstevens(a)yahoo.com Skype: mick_stevens www.facebook.com/mickstevens www.twitter.com/mickstevens ________________________________ From: Michael Collins <msc(a)freeswitch.org<mailto:msc@freeswitch.org>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Sent: Thursday, 29 November 2012, 19:14 Subject: Re: [Freeswitch-users] Explanation of rtp_audio_in / out fields in XML CDR's? Mick, This data is definitely not on the wiki - or anywhere else that I can see. I think we can crowdsource this to get the info collected and then add it to the mod_xml_curl wiki page<http://wiki.freeswitch.org/wiki/Mod_xml_cdr>. For the record, here's a quick dump of the fields that I took from an XML CDR: <rtp_audio_in_raw_bytes>10664</rtp_audio_in_raw_bytes> <rtp_audio_in_media_bytes>8772</rtp_audio_in_media_bytes> <rtp_audio_in_packet_count>62</rtp_audio_in_packet_count> <rtp_audio_in_media_packet_count>51</rtp_audio_in_media_packet_count> <rtp_audio_in_skip_packet_count>18</rtp_audio_in_skip_packet_count> <rtp_audio_in_jb_packet_count>0</rtp_audio_in_jb_packet_count> <rtp_audio_in_dtmf_packet_count>0</rtp_audio_in_dtmf_packet_count> <rtp_audio_in_cng_packet_count>0</rtp_audio_in_cng_packet_count> <rtp_audio_in_flush_packet_count>11</rtp_audio_in_flush_packet_count> <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size> <rtp_audio_out_raw_bytes>11524</rtp_audio_out_raw_bytes> <rtp_audio_out_media_bytes>11524</rtp_audio_out_media_bytes> <rtp_audio_out_packet_count>67</rtp_audio_out_packet_count> <rtp_audio_out_media_packet_count>67</rtp_audio_out_media_packet_count> <rtp_audio_out_skip_packet_count>0</rtp_audio_out_skip_packet_count> <rtp_audio_out_dtmf_packet_count>0</rtp_audio_out_dtmf_packet_count> <rtp_audio_out_cng_packet_count>0</rtp_audio_out_cng_packet_count> I think raw_bytes, media_bytes, packet_count, and media_packet_count are self-explanatory. I think cng_packet_count is probably self-explanatory too. My question on dtmf_packet_count would be whether it's only for RFC2833 packets (I suspect yes, but would like confirmation). If anyone knows these please reply to this email and Mick and I will get them documented on the wiki (right Mick? ;) rtp_audio_in_skip_packet_count rtp_audio_out_skip_packet_count rtp_audio_in_jb_packet_count rtp_audio_in_flush_packet_count rtp_audio_in_largest_jb_size Thanks all! -MC On Wed, Nov 28, 2012 at 3:28 AM, Mick Stevens <mickstevens(a)yahoo.com<mailto:mickstevens@yahoo.com>> wrote: Hi Folks, I'm trying to use FS XML CDR's to diagnose historic audio problems. I think I can work out some of the rtp_audio_in / out fields (raw bytes & media bytes being nearly equal looks like a good sign) but am wondering about the skip, cng & flush fields for example? I have tried Googling this & can find evidence of other people having asked this question but not of the answer. I have also checked my FS 106 & Cookbook book's without success. The wiki appears to be down at the moment so my apologies if the answer lies there. I know how to do this in real time using wireshark etc but am interested in being able to do some analysis on historic problems reported by customers that aren't willing/able to replicate the problem in order for a protocol trace to be captured. Any help much appreciated! Rgds, Mick Tel/SMS. +44(0)7967 594432 Fax. +44(0)7053 452429 Email/IM. mickstevens(a)yahoo.com Skype: mick_stevens www.facebook.com/mickstevens www.twitter.com/mickstevens _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com<http://www.cudatel.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://wiki.freeswitch.org<http://wiki.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto: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<http://www.freeswitch.org/> -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org<http://www.freeswitch.org/> http://www.ClueCon.com<http://www.cluecon.com/> http://www.OSTAG.org<http://www.ostag.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com<http://www.cudatel.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://wiki.freeswitch.org<http://wiki.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto: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<http://www.freeswitch.org/> !DSPAM:50bf25f932762027524863!

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OT: what carrier do you recommend me?
by Luis Daniel Lucio Quiroz 05 Dec '12

05 Dec '12

Helo every one, In your experience what carrier do you recommend me to route calls, I looking carriers for next countries: - China - USA/CA - Mexico I am using now Flowroute, but it fails sometimes and it is a little expensive. Regards, LD

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Call for assistance: splicing/editing MP4 files
by Michael Collins 05 Dec '12

05 Dec '12

Hi folks! If you have the time, equipment, and bandwidth to assist with some video/mp4 editing please contact me off list. Thanks! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Re: [Freeswitch-users] Doubts over G729 passthru for 50ms ptimes
by Ben 04 Dec '12

04 Dec '12

I tried setting the {passthru_ptime_mismatch=true} variable in the dialplan, but no joy unfortunately. I'm also fairly sure that the codecs are matched on both call legs, so shouldn't be having that sort of issue. I'm going to gather up some evidence and console logs/traces etc so that I can put something together for a bug report on Jira. Is there anything else that someone can suggest I try before it goes on Jira? Is anyone able to re-produce the same problem? Admittedly, I'm not on the latest GIT, but I'm on a recent (last couple of months) stable release. Thanks in advance for any assistance! Regards, Ben From: Ben N [mailto:ben122uk@gmail.com] Sent: 29 November 2012 09:58 To: clive engelberg Subject: Re: [Freeswitch-users] Doubts over G729 passthru for 50ms ptimes Hi Brian, yes I've set G729@50i on both the external and internal sofia profiles statically, and confirmed the allowed codecs by running "sofia status profile profilename". I can also confirm in the FS console that codec comparisons for both the a-leg, and b-leg match instantly to G729@50i. Clive, thanks for the pointer, although in my case I don't think there's a mis-match as each side of the call supports sending/receiving G729 50ms ptimes, and is negotiated correctly in the SDP. However I'm still willing to give it a try as I need to try everything! I'll let you know what happens. Cheers, Ben On Thu, Nov 29, 2012 at 8:08 AM, clive engelberg <clive18(a)webmail.co.za> wrote: Hi I have had limited success with : <action application="bridge" data="{passthru_ptime_mismatch=true}sofia/gateway/icall/$1(a)1.2.3.4"/> Sometimes the sound was garbled, sometimes worked fine, not sure why it was sometimes garbled, so I dont use it anymore, as complaints are not what I want :) Hope this helps. regards Clive On Tue, 27 Nov 2012 11:06:18 +0000 Ben N wrote Hi All, I'm wondering if anyone out there is using G729 with 50ms ptime through Freeswitch, without proxy media? I am unable to get this to work, even in a very basic lab with just two clients and an up to date FS server on the same LAN. I can get G729 with 20ms working, but 50ms causes Freeswitch to hang the call. It might be that mod_g729 and mod_com_g729 simply don't support it, so if anyone in the know can tell me that would be great! Further info can be supplied, I believe I have tried pretty much all options.... Cheers, Ben _____ South Africa premier free email service - webmail.co.za <http://www.webmail.co.za/> <http://b.wm.co.za/click.pwm?cid=20053491&loc=N-MT&seq=50b717ec> <http://b.wm.co.za/click.pwm?cid=20053491&loc=N-MT&seq=50b717ec>

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mod_sndfile can't find files
by Jakob 04 Dec '12

04 Dec '12

Hi, I did a clean install from git master and keep getting file not found in the logs. This was noticed in late nov on a fedora 17 git install but is still present in a recent (as of today dec 4:th) debian squeeze git master system. This was tested using sipdroid on the same lan. Relevant part of the log follows: 2012-11-25 21:10:35.053024 [DEBUG] switch_core_session.c:830 Send signal sofia/internal/1050(a)172.16.10.9 [BREAK] 2012-11-25 21:10:35.053024 [DEBUG] switch_channel.c:3380 (sofia/internal/1050(a)172.16.10.9) Callstate Change RINGING -> ACTIVE 2012-11-25 21:10:35.053024 [DEBUG] switch_core_session.c:976 Send signal sofia/internal/1050(a)172.16.10.9 [BREAK] 2012-11-25 21:10:35.053024 [NOTICE] mod_dptools.c:1176 Channel [sofia/internal/1050(a)172.16.10.9] has been answered 2012-11-25 21:10:35.053024 [DEBUG] sofia.c:5607 Channel sofia/internal/1050(a)172.16.10.9 entering state [completed][200] EXECUTE sofia/internal/1050(a)172.16.10.9 sleep(1000) 2012-11-25 21:10:35.113025 [DEBUG] switch_core_session.c:976 Send signal sofia/internal/1050(a)172.16.10.9 [BREAK] 2012-11-25 21:10:35.113025 [DEBUG] switch_core_session.c:976 Send signal sofia/internal/1050(a)172.16.10.9 [BREAK] 2012-11-25 21:10:35.113025 [DEBUG] switch_core_session.c:976 Send signal sofia/internal/1050(a)172.16.10.9 [BREAK] 2012-11-25 21:10:35.133027 [DEBUG] sofia.c:5607 Channel sofia/internal/1050(a)172.16.10.9 entering state [ready][200] EXECUTE sofia/internal/1050(a)172.16.10.9 voicemail(check default) 2012-11-25 21:10:36.053026 [DEBUG] switch_ivr_play_say.c:67 No language specified - Using [en] 2012-11-25 21:10:36.253026 [DEBUG] switch_ivr_play_say.c:67 No language specified - Using [en] 2012-11-25 21:10:36.253026 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-enter_id.wav] (en:en) 2012-11-25 21:10:36.253026 [ERR] mod_sndfile.c:198 Error Opening File [/usr/share/freeswitch/sounds/en/us/callie/voicemail/vm-enter_$ 2012-11-25 21:10:36.353027 [DEBUG] switch_ivr_play_say.c:67 No language specified - Using [en] 2012-11-25 21:10:36.353027 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-goodbye.wav] (en:en) 2012-11-25 21:10:36.353027 [ERR] mod_sndfile.c:198 Error Opening File [/usr/share/freeswitch/sounds/en/us/callie/voicemail/vm-goodby$ 2012-11-25 21:10:36.453048 [NOTICE] switch_core_state_machine.c:262 sofia/internal/1050(a)172.16.10.9 has executed the last dialplan i$ 2012-11-25 21:10:36.453048 [DEBUG] switch_channel.c:2979 (sofia/internal/1050(a)172.16.10.9) Callstate Change ACTIVE -> HANGUP Best regards Jakob Sundberg

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CPS
by Ержан Туле пов 04 Dec '12

04 Dec '12

Hi, Could anyone explain me what impact does CPS config param has on FS? Say it's set to 30, does that mean that FS box won't allow creation of more than 30 call per sec? Or FS just log that in a log? Regards, Y.

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Freeswitch is crashing when httapi playback application is executed
by Srigo 04 Dec '12

04 Dec '12

Hello, I'm using Httapi for dynamic application service. The problem, I'm getting is when im using httapi with the application "playback", freeswitch is crashing. Seems the crashed is caused by the parameter of "file" in playback. Thats mean when i'm getting a file from a remote server (http://xxxxxx.com/test.wav) its crashing but when using a local wav file everthing is fine. here is the httapi xml: <document type="xml/freeswitch-httapi"> <params/> <variables> <sou>someval</sou> </variables> <work> <execute application="flush_dtmf"/> <execute application="set" data="${http_prefetch(http://xxxxxxx.com/test.wav)}"></execute> <execute application="info"/> <playback name="user_entry" file="http://xxxxxxx.com/test.wav" loops="3" digit-timeout="1000" input-timeout="1000"> <bind strip="#">~\d+</bind> </playback> </work> </document> Here the log before the freeswitch crash: freeswitch: src/switch_event.c:710: switch_event_get_header_ptr: Assertion `event' failed. I try to use http_cache to try to resolve the prob: <playback name="user_entry" file="${http_get(http://xxxxxxx.com/test.wav)}" loops="3" digit-timeout="1000" input-timeout="1000"> <bind strip="#">~\d+</bind> </playback> but i'm getting the following error in the log: 2012-11-23 19:06:10.015936 [ERR] switch_core_file.c:112 Unknown file Format [$] Finally i try the following idea: - set a variable (<execute application="set" data="play=${http_get(http://xxxxxxx.com/test.wav)}"></execute> then use in playback parameter the variable ${play} but the variable is not parsed! 2012-11-23 17:30:09.855933 [ERR] mod_native_file.c:74 Error opening /usr/share/freeswitch/sounds/en/us/callie/${play}.PCMA Could anyone please tell me where i did the mistake ? thx a lot -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-is-crashing-when-h… Sent from the freeswitch-users mailing list archive at Nabble.com.

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