- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Codec Preferance
by Bharat Lalcheta 06 Jan '12

06 Jan '12

> Can you please provide a simple example to use different codec in both leg > using dialplan > > Regards, > > Bharat Lalcheta > > On Fri, Jan 6, 2012 at 9:10 PM, < > freeswitch-users-request(a)lists.freeswitch.org> wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users(a)lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request(a)lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner(a)lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> Today's Topics: >> >> 1. Re: Codec Preferance (Kristian Kielhofner) >> 2. Re: Codec Preferance (curriegrad2004) >> 3. Re: FreeSWITCH-users Digest, Vol 67, Issue 51 (Bharat Lalcheta) >> >> >> ---------- Forwarded message ---------- >> From: Kristian Kielhofner <kris(a)kriskinc.com> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Date: Fri, 6 Jan 2012 10:24:01 -0500 >> Subject: Re: [Freeswitch-users] Codec Preferance >> inbound_codec_prefs >> outbound_codec_prefs >> >> These are Sofia profile configuration options. They are NOT valid >> options for the directory UNLESS you're setting them as variable to >> use in your dialplan later. >> >> If you want to control codes on a per-user basis you have to set late >> negotiation and use the dialplan. >> >> On Fri, Jan 6, 2012 at 8:18 AM, Bharat Lalcheta >> <bharatlalcheta(a)gmail.com> wrote: >> > Hiii, >> > >> > I am new to freeswitch. Prior to freeswitch i was using asterisk. >> > >> > I have 200 extensions working in my office and want to move all to >> > freeswitch from asterisk. >> > >> > In asterisk, i can give codec selection and preferance in sip.conf to >> all >> > extensions. In the same way i created 200 extensions under internal >> profile >> > in freeswitch. >> > >> > Follwing is one example.... >> > ----------------------------------------------------------------- >> > <include> >> > <user id="590"> >> > <params> >> > <param name="password" value="590"/> >> > <param name="vm-password" value=""/> >> > <param name="vm-enabled" value="true"/> >> > <param name="inbound_codec_prefs" value="PCMA,H264"/> >> > <param name="outbound_codec_prefs" value="PCMA,H264"/> >> > </params> >> > <variables> >> > <variable name="accountcode" value=""/> >> > <variable name="user_context" value="default"/> >> > <variable name="max-calls" value="2"/> >> > <variable name="bypass_media_after_bridge" value="no"/> >> > </variables> >> > </user> >> > </include> >> > ---------------------------------------------------------- >> > >> > Now when ever i called to 590 freeswitch sends all codecs to 590 sip >> phone >> > other than defined in 590.xml. It is seding codes which is mentioned in >> my >> > conf/sip_profiles/internal.xml and codec negotiation done on whatever >> codec >> > my sip phone having. >> > >> > I want to use different codecs for different extensions. >> > >> > Is it common behaviour of Freeswitch ? Should i override codec >> prerfrance in >> > my extension list from my internal profile or not ? >> > >> > If no, then is it that i have to create 200 profiles in freeswitch to >> solve >> > this problem ? >> > >> > Please guide me and provide solution for the same >> > >> > >> > Thanks in advance >> > >> > Bharat Lalcheta >> > >> > >> _________________________________________________________________________ >> > 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 >> > >> >> >> >> -- >> Kristian Kielhofner >> >> >> >> >> ---------- Forwarded message ---------- >> From: curriegrad2004 <curriegrad2004(a)gmail.com> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Date: Fri, 6 Jan 2012 07:37:06 -0800 >> Subject: Re: [Freeswitch-users] Codec Preferance >> >> In theory he could use those variable names but it would involve some >> extra work ;) >> On 2012-01-06 7:24 AM, "Kristian Kielhofner" <kris(a)kriskinc.com> wrote: >> >>> inbound_codec_prefs >>> outbound_codec_prefs >>> >>> These are Sofia profile configuration options. They are NOT valid >>> options for the directory UNLESS you're setting them as variable to >>> use in your dialplan later. >>> >>> If you want to control codes on a per-user basis you have to set late >>> negotiation and use the dialplan. >>> >>> On Fri, Jan 6, 2012 at 8:18 AM, Bharat Lalcheta >>> <bharatlalcheta(a)gmail.com> wrote: >>> > Hiii, >>> > >>> > I am new to freeswitch. Prior to freeswitch i was using asterisk. >>> > >>> > I have 200 extensions working in my office and want to move all to >>> > freeswitch from asterisk. >>> > >>> > In asterisk, i can give codec selection and preferance in sip.conf to >>> all >>> > extensions. In the same way i created 200 extensions under internal >>> profile >>> > in freeswitch. >>> > >>> > Follwing is one example.... >>> > ----------------------------------------------------------------- >>> > <include> >>> > <user id="590"> >>> > <params> >>> > <param name="password" value="590"/> >>> > <param name="vm-password" value=""/> >>> > <param name="vm-enabled" value="true"/> >>> > <param name="inbound_codec_prefs" value="PCMA,H264"/> >>> > <param name="outbound_codec_prefs" value="PCMA,H264"/> >>> > </params> >>> > <variables> >>> > <variable name="accountcode" value=""/> >>> > <variable name="user_context" value="default"/> >>> > <variable name="max-calls" value="2"/> >>> > <variable name="bypass_media_after_bridge" value="no"/> >>> > </variables> >>> > </user> >>> > </include> >>> > ---------------------------------------------------------- >>> > >>> > Now when ever i called to 590 freeswitch sends all codecs to 590 sip >>> phone >>> > other than defined in 590.xml. It is seding codes which is mentioned >>> in my >>> > conf/sip_profiles/internal.xml and codec negotiation done on whatever >>> codec >>> > my sip phone having. >>> > >>> > I want to use different codecs for different extensions. >>> > >>> > Is it common behaviour of Freeswitch ? Should i override codec >>> prerfrance in >>> > my extension list from my internal profile or not ? >>> > >>> > If no, then is it that i have to create 200 profiles in freeswitch to >>> solve >>> > this problem ? >>> > >>> > Please guide me and provide solution for the same >>> > >>> > >>> > Thanks in advance >>> > >>> > Bharat Lalcheta >>> > >>> > >>> _________________________________________________________________________ >>> > 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 >>> > >>> >>> >>> >>> -- >>> Kristian Kielhofner >>> >>> _________________________________________________________________________ >>> 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 >>> >> >> >> ---------- Forwarded message ---------- >> From: Bharat Lalcheta <bharatlalcheta(a)gmail.com> >> To: freeswitch-users(a)lists.freeswitch.org >> Cc: >> Date: Fri, 6 Jan 2012 21:10:01 +0530 >> Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 67, Issue 51 >> can you please explain in details what you want to tell ? >> >> >> >> >>> ---------- Forwarded message ---------- >>> From: curriegrad2004 <curriegrad2004(a)gmail.com> >>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >>> Cc: >>> Date: Fri, 6 Jan 2012 07:11:52 -0800 >>> Subject: Re: [Freeswitch-users] Codec Preferance >>> >>> I highly would recommend that you change the name of those codecs to >>> something else because you might be making matters worse later down the >>> road >>> On 2012-01-06 5:21 AM, "Bharat Lalcheta" <bharatlalcheta(a)gmail.com> >>> wrote: >>> >>>> Hiii, >>>> >>>> I am new to freeswitch. Prior to freeswitch i was using asterisk. >>>> >>>> I have 200 extensions working in my office and want to move all to >>>> freeswitch from asterisk. >>>> >>>> In asterisk, i can give codec selection and preferance in sip.conf to >>>> all extensions. In the same way i created 200 extensions under internal >>>> profile in freeswitch. >>>> >>>> Follwing is one example.... >>>> ----------------------------------------------------------------- >>>> <include> >>>> <user id="590"> >>>> <params> >>>> <param name="password" value="590"/> >>>> <param name="vm-password" value=""/> >>>> <param name="vm-enabled" value="true"/> >>>> <param name="inbound_codec_prefs" value="PCMA,H264"/> >>>> <param name="outbound_codec_prefs" value="PCMA,H264"/> >>>> </params> >>>> <variables> >>>> <variable name="accountcode" value=""/> >>>> <variable name="user_context" value="default"/> >>>> <variable name="max-calls" value="2"/> >>>> <variable name="bypass_media_after_bridge" value="no"/> >>>> </variables> >>>> </user> >>>> </include> >>>> ---------------------------------------------------------- >>>> >>>> Now when ever i called to 590 freeswitch sends all codecs to 590 sip >>>> phone other than defined in 590.xml. It is seding codes which is mentioned >>>> in my conf/sip_profiles/internal.xml and codec negotiation done on whatever >>>> codec my sip phone having. >>>> >>>> I want to use different codecs for different extensions. >>>> >>>> Is it common behaviour of Freeswitch ? Should i override codec >>>> prerfrance in my extension list from my internal profile or not ? >>>> >>>> If no, then is it that i have to create 200 profiles in freeswitch to >>>> solve this problem ? >>>> >>>> Please guide me and provide solution for the same >>>> >>>> >>>> Thanks in advance >>>> >>>> Bharat Lalcheta >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>>> >>>> >>> >>> ---------- Forwarded message ---------- >>> From: Peter Olsson <peter.olsson(a)visionutveckling.se> >>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >>> Cc: >>> Date: Fri, 6 Jan 2012 15:21:00 +0000 >>> Subject: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR >>> Are you still using ignore_early_media=true - this must be set for this >>> to work correctly. >>> >>> You will see a EXECUTE log line when FS executes the application, with >>> ignore_early_media enabled it shouldn't execute until the call has been >>> answered. I just tried it myself, and it works as expected. >>> >>> Example "originate {ignore_early_media=true}sofia/internal/number@host&park()" >>> >>> Park application is only executed after the call was answered. >>> >>> /Peter >>> >>> ________________________________________ >>> Från: freeswitch-users-bounces(a)lists.freeswitch.org [ >>> freeswitch-users-bounces(a)lists.freeswitch.org] för Oliver Schenk [ >>> olimonkey(a)gmail.com] >>> Skickat: den 6 januari 2012 12:04 >>> Till: FreeSWITCH Users Help >>> Ämne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR >>> >>> Because I'm using an FXO card with voice, I added something to my >>> CISCO conf. Many others had the same thing: >>> >>> >>> voice-port 0/3/0 >>> ... >>> supervisory disconnect dualtone mid-call >>> supervisory answer dualtone <---- ADDED THIS ONE >>> ... >>> >>> >>> >>> Once I added this, the FS output now just showed the following while >>> the phone was ringing: >>> >>> 2012-01-05 16:19:31.644440 [NOTICE] switch_channel.c:816 New Channel >>> sofia/internal/109212xxxx(a)192.168.x.x >>> [69e3f13d-1e2a-409e-97a4-b5526ea6e4ec] >>> 2012-01-05 16:19:35.124882 [INFO] sofia.c:740 >>> sofia/internal/109212xxxx(a)192.168.x.x Update Callee ID to "Outbound >>> Call" <109212xxxx> >>> 2012-01-05 16:19:35.126883 [NOTICE] sofia_glue.c:3793 Pre-Answer >>> sofia/internal/109212xxxx(a)192.168.x.x! >>> >>> >>> Where as previous it would show the above and also show the following: >>> >>> 2012-01-05 16:19:35.127883 [INFO] switch_channel.c:2456 >>> sofia/internal/109212xxxx(a)192.168.x.x Flipping CID from "" >>> <0000000000> to "Outbound Call" <109212xxxx> >>> 2012-01-05 16:19:35.137384 [INFO] sofia.c:740 >>> sofia/internal/109212xxxx(a)192.168.x.x Update Callee ID to "109212xxxx" >>> <1092122856> >>> 2012-01-05 16:19:35.138384 [NOTICE] sofia.c:5296 Channel >>> [sofia/internal/109212xxxx(a)192.168.x.x] has been answered >>> >>> >>> >>> BUT, the IVR still started playing even before I pick up the phone. >>> Hmmmm.....so why is FS still starting the managed application when the >>> call has not been answered yet. Are we all sure that the managed >>> application should not be executed until the call "has been answered" >>> shows up in the log file? >>> >>> >>> Will have to keep testing on monday as I don't have access to the >>> CISCO from where i am now. I'll have to see whether the CISCO changes >>> had any impact on the times at which the SIP messages are sent back >>> and forth. Especially the 200 OK message. >>> >>> >>> Thanks again for help, maybe getting somewhere now...... >>> >>> Oliver >>> >>> >>> >>> >>> On Fri, Jan 6, 2012 at 4:20 PM, Peter Olsson >>> <peter.olsson(a)visionutveckling.se> wrote: >>> > If it sends 200 OK right after 183, this IS the problem. >>> > >>> > 200 OK means that the call was answered, it should not be sent until >>> the call was actually picked up in the remote end. When 200 OK arrives to >>> FS it will execute your app, and you will start playing the files. >>> > >>> > Seems to me there is something broken in the Cisco. >>> > >>> > /Peter >>> > >>> > ________________________________________ >>> > Från: freeswitch-users-bounces(a)lists.freeswitch.org [ >>> freeswitch-users-bounces(a)lists.freeswitch.org] för Oliver Schenk [ >>> olimonkey(a)gmail.com] >>> > Skickat: den 6 januari 2012 06:55 >>> > Till: FreeSWITCH Users Help >>> > Ämne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR >>> > >>> > I've tried looking at disable-early-media configuration command, but >>> > that didn't work and I doubt that has anything to do with the CISCO >>> > sending a 200 OK right after a 183 SESSION PROGRESS. >>> > >>> > >>> > >>> > >>> > On Fri, Jan 6, 2012 at 9:20 AM, Brian West <brian(a)freeswitch.org> >>> wrote: >>> >> Thats what the 183 is.. 180 vs 183 are kinda sketchy in some >>> devices.. 180 >>> >> is usually RINGING (generate ringback locally) while a 183 has >>> media... aka >>> >> early media and usually provides ringback inband. >>> >> >>> >> /b >>> >> >>> >> On Jan 5, 2012, at 7:13 PM, Oliver Schenk wrote: >>> >> >>> >> Shouldn't there be a 180 RINGING somewhere in there? >>> >> >>> >> >>> >> >>> >> >>> >> On Fri, Jan 6, 2012 at 8:25 AM, Oliver Schenk <olimonkey(a)gmail.com> >>> wrote: >>> >> >>> >> I just noticed something else, if I don't pick up the phone at all. >>> >> >>> >> The IVR just keeps playing until the menu timeout kicks in. >>> >> >>> >> >>> >> So here is a CISCO SIP log: >>> >> >>> >> http://pastebin.com/Y9sYkuxi >>> >> >>> >> >>> >> The FS server is 192.168.x.50 and the CISCO is 192.168.x.1. >>> >> >>> >> I hope the CISCO log is readable, it's a bit long because I just did >>> >> >>> >> "debug ccsip all". >>> >> >>> >> >>> >> >>> >> >>> >> In this test I didn't bother picking up the phone at all, but I can >>> >> >>> >> see that FS answered anyway and the IVR kept playing until it timed >>> >> >>> >> out. >>> >> >>> >> I'm not an expert, but here is what I picked out of it: >>> >> >>> >> >>> >> At 00:08:10 we get a >>> >> >>> >> Received: "INVITE sip:109212xxxx@192.168.x.1 SIP/2.0" >>> >> >>> >> >>> >> the further down at the same timestamp we get >>> >> >>> >> Sent: "SIP/2.0 100 Trying" >>> >> >>> >> >>> >> At 00:08:13 we get a >>> >> >>> >> Sent: "SIP/2.0 183 Session Progress" >>> >> >>> >> >>> >> At 00:18:13 we get a >>> >> >>> >> Sent: "SIP/2.0 200 OK" >>> >> >>> >> >>> >> Then at the same timestamp we get: >>> >> >>> >> Received: "ACK sip:109212xxxx@192.168.x.1:5060 SIP/2.0" >>> >> >>> >> >>> >> >>> >> >>> >> Once the IVR times out at 00:09:16 we get >>> >> >>> >> Received: "BYE sip:109212xxxx@192.168.x.1:5060 SIP/2.0" >>> >> >>> >> >>> >> And then the reply right after >>> >> >>> >> Sent: "SIP/2.0 200 OK" >>> >> >>> >> >>> >> >>> >> >>> >> So I think you were right, the CISCO is sending back an "OK" 3 seconds >>> >> >>> >> after the "INVITE" is received. >>> >> >>> >> >>> >> >>> >> >>> >> The part that is beyond my field of expertise so far is WHY? >>> >> >>> >> >>> >> >>> >> >>> >> Thanks, >>> >> >>> >> >>> >> >>> >> Oliver >>> >> >>> >> >>> >> >>> >> >>> >> On Fri, Jan 6, 2012 at 8:04 AM, Oliver Schenk <olimonkey(a)gmail.com> >>> wrote: >>> >> >>> >> By the way: >>> >> >>> >> >>> >> I tried {ignore_early_media=true} as well, but as I think we >>> >> >>> >> determined, my problem is probably with the CISCO telling FS that the >>> >> >>> >> call has been answered when really it hasn't yet. >>> >> >>> >> >>> >> >>> >> >>> >> On Fri, Jan 6, 2012 at 8:01 AM, Oliver Schenk <olimonkey(a)gmail.com> >>> wrote: >>> >> >>> >> Thanks for the help so far. >>> >> >>> >> >>> >> >>> >> Here is a pastebin of FreeSWITCH output: >>> >> >>> >> http://pastebin.com/i6Qgc7ws >>> >> >>> >> >>> >> Notice how the "has been answered" log message comes immediately >>> >> >>> >> (within a few milliseconds) after the call was originated. I think >>> >> >>> >> this would suggest that the CISCO is immediately sending a 200 OK, as >>> >> >>> >> you suggested. I also turned on CISCO debugging, but I'm just trying >>> >> >>> >> to figure out how to get the information regarding SIP messages back >>> >> >>> >> to Freeswitch. I'll run the test again and see if I can get some >>> >> >>> >> useful CISCO debug. >>> >> >>> >> >>> >> Which "debug ccsip" commands are relevant to what I want for the CISCO >>> >> >>> >> SIP debugging? >>> >> >>> >> >>> >> >>> >> Thanks! >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> 2012/1/6 Gustavo Mársico <gustavomarsico(a)gmail.com>: >>> >> >>> >> I think I've a similar problem related to callcenter app. When I made >>> an >>> >> originate like this: >>> >> >>> >> >>> >> originate loopback/2500/default/XML &bridge(user/2001) >>> >> >>> >> >>> >> 2500 is an extension that leads to a callcenter application. In this >>> case, >>> >> we dial first to the queue and when an agent answered we call to the >>> >> customer. As far as I know >>> >> >>> >> When the A-leg reaches to the queue, without selecting an agent, the >>> call is >>> >> automatically sent to the B-leg. As far as I see, there is a >>> pre-answer >>> >> method that fs needs to send the media to A-leg. >>> >> >>> >> In order to try to avoid this, I tried using ignore_early_media=true >>> as part >>> >> of the originate in A-leg and/or B-leg, with no luck. >>> >> >>> >> >>> >> originate {ignore_early_media=true}loopback/2500/default/XML >>> >> &bridge({ignore_early_media=true}user/2001) >>> >> >>> >> >>> >> Dialplan: loopback/2500-b Regex (PASS) [CallCenter_Click2Call] >>> >> destination_number(2500) =~ /^(2500)$/ break=on-false >>> >> >>> >> Dialplan: loopback/2500-b Action set(ignore_early_media=true) >>> >> >>> >> Dialplan: loopback/2500-b Action callcenter(click2call) >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:154 >>> >> (loopback/2500-b) State Change CS_ROUTING -> CS_EXECUTE >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:1180 Send >>> signal >>> >> loopback/2500-b [BREAK] >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >>> >> CHANNEL KILL >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:410 >>> >> (loopback/2500-b) State ROUTING going to sleep >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:362 >>> >> (loopback/2500-b) Running State Change CS_EXECUTE >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:417 >>> >> (loopback/2500-b) State EXECUTE >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:375 loopback/2500-b >>> >> CHANNEL EXECUTE >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:192 >>> >> loopback/2500-b Standard EXECUTE >>> >> >>> >> EXECUTE loopback/2500-b set(open=true) >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 >>> loopback/2500-b SET >>> >> [open]=[true] >>> >> >>> >> EXECUTE loopback/2500-b >>> >> >>> hash(insert/10.8.0.70-spymap/0000000000/fef7d864-b3c7-407c-a6e1-94386642bfbb) >>> >> >>> >> EXECUTE loopback/2500-b >>> hash(insert/10.8.0.70-last_dial/0000000000/2500) >>> >> >>> >> EXECUTE loopback/2500-b >>> >> >>> hash(insert/10.8.0.70-last_dial/global/fef7d864-b3c7-407c-a6e1-94386642bfbb) >>> >> >>> >> EXECUTE loopback/2500-b set(RFC2822_DATE=Thu, 05 Jan 2012 13:36:08 >>> -0300) >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 >>> loopback/2500-b SET >>> >> [RFC2822_DATE]=[Thu, 05 Jan 2012 13:36:08 -0300] >>> >> >>> >> EXECUTE loopback/2500-b set(ignore_early_media=true) >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 >>> loopback/2500-b SET >>> >> [ignore_early_media]=[true] >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:2133 >>> Application >>> >> callcenter Requires media! pre_answering channel loopback/2500-b >>> >> >>> >> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:760 Pre-Answer >>> >> loopback/2500-a! >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 >>> (loopback/2500-a) >>> >> Callstate Change RINGING -> EARLY >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send >>> signal >>> >> loopback/2500-b [BREAK] >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >>> >> CHANNEL KILL >>> >> >>> >> 2012-01-05 13:36:08.541517 [NOTICE] switch_core_session.c:2135 >>> Pre-Answer >>> >> loopback/2500-b! >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 >>> (loopback/2500-b) >>> >> Callstate Change RINGING -> EARLY >>> >> >>> >> EXECUTE loopback/2500-b callcenter(click2call) >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 >>> (loopback/2500-a) >>> >> Callstate Change EARLY -> ACTIVE >>> >> >>> >> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:755 Channel >>> >> [loopback/2500-a] has been answered >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send >>> signal >>> >> loopback/2500-b [BREAK] >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >>> >> CHANNEL KILL >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_ivr_originate.c:3266 >>> Originate >>> >> Resulted in Success: [loopback/2500-a] >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 >>> (loopback/2500-b) >>> >> Callstate Change EARLY -> ACTIVE >>> >> >>> >> 2012-01-05 13:36:08.541517 [INFO] switch_channel.c:2708 >>> loopback/2500-a >>> >> Flipping CID from "" <0000000000> to "Outbound Call" <XML> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> On Jan 5, 2012, at 4:17 AM, Oliver Schenk wrote: >>> >> >>> >> >>> >> Also, maybe I should be doing something like this: >>> >> >>> >> >>> >> sofia/gateway/mygatewayname/1091234567 '&managed(ivrAppName)' >>> >> >>> >> >>> >> instead of: >>> >> >>> >> >>> >> sofia/internal/1091234567(a)192.168.x.x '&managed(ivrAppName)' >>> >> >>> >> >>> >> >>> >> but, I don't really have the CISCO configured as a gateway, nor do I >>> >> >>> >> know how really...probably not on the right track there. >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> On Thu, Jan 5, 2012 at 3:06 PM, Oliver Schenk <olimonkey(a)gmail.com> >>> wrote: >>> >> >>> >> *bump* >>> >> >>> >> >>> >> >>> >> So I think maybe the way I'm doing the originate is the problem? In my >>> >> >>> >> call string I'm creating a connection directly from the CISCO >>> >> >>> >> (192.168.x.x) to the managed application, which may be why it starts >>> >> >>> >> playing straight away? >>> >> >>> >> >>> >> Maybe I should be originating a call first and then only once I know >>> >> >>> >> the other side has picked up will I bridge the call to the IVR managed >>> >> >>> >> application. >>> >> >>> >> >>> >> Problem is I dunno how to tell whether the other person has picked up >>> >> >>> >> (or even if the cisco is going to tell me) and I don't know how to do >>> >> >>> >> things to a call once it has been established. >>> >> >>> >> >>> >> >>> >> I'm currently reading the Dialplan wiki page, hoping to get something >>> >> >>> >> out of it there. >>> >> >>> >> >>> >> >>> >> Cheers >>> >> >>> >> >>> >> Oliver >>> >> >>> >> >>> >> >>> >> On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk <olimonkey(a)gmail.com> >>> wrote: >>> >> >>> >> I've been battling while creating an IVR using FreeSWITCH mod_managed >>> >> >>> >> and connecting through a CISCO 2811. Most things now work quite well, >>> >> >>> >> but I am having a few issues with the way the system answers calls (or >>> >> >>> >> doesn't answer calls...). >>> >> >>> >> >>> >> I have FreeSWITCH running as a windows service on Windows Server 2008, >>> >> >>> >> which is connected via LAN to a CISCO 2811 with a 4 port FXO card, >>> >> >>> >> which is then connected to a POTS phone line. >>> >> >>> >> >>> >> >>> >> Take the following scenario: >>> >> >>> >> >>> >> 1. Managed .NET application creates a call string and uses ESL to talk >>> >> >>> >> to freeswitch and originate a call: >>> >> >>> >> >>> >> string callstring = >>> >> >>> >> >>> "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567(a)192.168.x.x >>> >> >>> >> '&managed(ivrAppName)'"; >>> >> >>> >> eslConnection.API("originate", callstring); >>> >> >>> >> >>> >> where 192.168.x.x is the CISCO IP. >>> >> >>> >> >>> >> 2. The CISCO sees that the phone number (1091234567) starts with a 1 >>> >> >>> >> so it uses FXO port 1 and strips the 1 and uses the remaining phone >>> >> >>> >> number (091234567) to make the call. >>> >> >>> >> >>> >> 3. My phone rings, I pick up and I can hear my IVR playing. >>> >> >>> >> >>> >> >>> >> >>> >> These are my current problems: >>> >> >>> >> >>> >> - IVR starts playing before I even pick up the phone. This means that >>> >> >>> >> if the system calls a mobile phone and the person doesn't pick up, the >>> >> >>> >> IVR will start playing and eventually the mobile phone will divert to >>> >> >>> >> voice mail. Obviously I then get a missed call and an sms saying I >>> >> >>> >> have a new voice mail, which is annoying. Instead I would like it to >>> >> >>> >> KNOW that no one has picked up, but I don't know how to do this. >>> >> >>> >> Somehow the CISCO needs to be able to tell FreeSWITCH that the call >>> >> >>> >> has not yet been answered. For some reason however as soon as the >>> >> >>> >> CISCO starts calling FreeSWITCH thinks the call is already connected. >>> >> >>> >> It doesn't know that the CISCO is actually still ringing. Maybe I'm >>> >> >>> >> doing originate the wrong way or something ... >>> >> >>> >> >>> >> - The phone only rings for about 10 seconds before hanging up. I've >>> >> >>> >> tried "call_timeout", "bridge_answer_timeout". I've also tried setting >>> >> >>> >> CISCO "ring number". Nothing works, my phone still only rings for >>> >> >>> >> about 10 seconds. I don't know if this is a FreeSWITCH issue or a >>> >> >>> >> CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just >>> >> >>> >> starts playing even if no one answers the phone. >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> CISCO Config for relevant FXO port: >>> >> >>> >> >>> >> voice service voip >>> >> >>> >> allow-connections h323 to h323 >>> >> >>> >> allow-connections h323 to sip >>> >> >>> >> allow-connections sip to h323 >>> >> >>> >> allow-connections sip to sip >>> >> >>> >> no supplementary-service h450.2 >>> >> >>> >> no supplementary-service h450.3 >>> >> >>> >> supplementary-service h450.12 >>> >> >>> >> no supplementary-service sip moved-temporarily >>> >> >>> >> no supplementary-service sip refer >>> >> >>> >> fax protocol cisco >>> >> >>> >> sip >>> >> >>> >> registrar server expires max 3600 min 3600 >>> >> >>> >> no update-callerid >>> >> >>> >> no call service stop >>> >> >>> >> >>> >> voice-port 0/3/2 >>> >> >>> >> output attenuation -3 >>> >> >>> >> no comfort-noise >>> >> >>> >> cptone AU >>> >> >>> >> impedance complex1 >>> >> >>> >> caller-id enable >>> >> >>> >> ! >>> >> >>> >> dial-peer voice 100 pots >>> >> >>> >> preference 1 >>> >> >>> >> destination-pattern 1T >>> >> >>> >> port 0/3/2 >>> >> >>> >> ! >>> >> >>> >> >>> >> >>> >> >>> >> Many Thanks, >>> >> >>> >> >>> >> Oliver >>> >> >>> >> >>> >> >>> _________________________________________________________________________ >>> >> >>> >> Professional FreeSWITCH Consulting Services: >>> >> >>> >> consulting(a)freeswitch.org >>> >> >>> >> http://www.freeswitchsolutions.com >>> >> >>> >> >>> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> >> >>> >> http://www.cudatel.com >>> >> >>> >> >>> >> Official FreeSWITCH Sites >>> >> >>> >> http://www.freeswitch.org >>> >> >>> >> http://wiki.freeswitch.org >>> >> >>> >> http://www.cluecon.com >>> >> >>> >> >>> >> FreeSWITCH-users mailing list >>> >> >>> >> FreeSWITCH-users(a)lists.freeswitch.org >>> >> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> >>> >> http://www.freeswitch.org >>> >> >>> >> >>> >> >>> >> >>> _________________________________________________________________________ >>> >> >>> >> Professional FreeSWITCH Consulting Services: >>> >> >>> >> consulting(a)freeswitch.org >>> >> >>> >> http://www.freeswitchsolutions.com >>> >> >>> >> >>> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> >> >>> >> http://www.cudatel.com >>> >> >>> >> >>> >> Official FreeSWITCH Sites >>> >> >>> >> http://www.freeswitch.org >>> >> >>> >> http://wiki.freeswitch.org >>> >> >>> >> http://www.cluecon.com >>> >> >>> >> >>> >> FreeSWITCH-users mailing list >>> >> >>> >> FreeSWITCH-users(a)lists.freeswitch.org >>> >> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> >>> >> http://www.freeswitch.org >>> >> >>> >> >>> >> >>> _________________________________________________________________________ >>> >> Professional FreeSWITCH Consulting Services: >>> >> consulting(a)freeswitch.org >>> >> http://www.freeswitchsolutions.com >>> >> >>> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> >> http://www.cudatel.com >>> >> >>> >> Official FreeSWITCH Sites >>> >> http://www.freeswitch.org >>> >> http://wiki.freeswitch.org >>> >> http://www.cluecon.com >>> >> >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users(a)lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> >> >>> >> >>> >> -- >>> >> Brian West >>> >> FreeSWITCH Solutions, LLC >>> >> Phone: +1 (918) 420-9266 >>> >> Fax: +1 (918) 420-9267 >>> >> brian(a)freeswitch.org >>> >> http://www.freeswitch.org >>> >> >>> >> >>> >> >>> _________________________________________________________________________ >>> >> Professional FreeSWITCH Consulting Services: >>> >> consulting(a)freeswitch.org >>> >> http://www.freeswitchsolutions.com >>> >> >>> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> >> http://www.cudatel.com >>> >> >>> >> Official FreeSWITCH Sites >>> >> http://www.freeswitch.org >>> >> http://wiki.freeswitch.org >>> >> http://www.cluecon.com >>> >> >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users(a)lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> >> >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting(a)freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> > http://www.cudatel.com >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users(a)lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> > >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting(a)freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> > http://www.cudatel.com >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users(a)lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users(a)lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> !DSPAM:4f06d49b32762089563979! >>> >>> >>> >>> >>> _______________________________________________ >>> 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 >>> >>> >> >> >> -- >> Bharat Lalcheta >> >> _______________________________________________ >> 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 >> >> > > > -- > Bharat Lalcheta > -- Bharat Lalcheta

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 67, Issue 52
by Bharat Lalcheta 06 Jan '12

06 Jan '12

Can you please provide a simple example to use different codec in both leg using dialplan Regards, Bharat Lalcheta On Fri, Jan 6, 2012 at 9:10 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Codec Preferance (Kristian Kielhofner) > 2. Re: Codec Preferance (curriegrad2004) > 3. Re: FreeSWITCH-users Digest, Vol 67, Issue 51 (Bharat Lalcheta) > > > ---------- Forwarded message ---------- > From: Kristian Kielhofner <kris(a)kriskinc.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Fri, 6 Jan 2012 10:24:01 -0500 > Subject: Re: [Freeswitch-users] Codec Preferance > inbound_codec_prefs > outbound_codec_prefs > > These are Sofia profile configuration options. They are NOT valid > options for the directory UNLESS you're setting them as variable to > use in your dialplan later. > > If you want to control codes on a per-user basis you have to set late > negotiation and use the dialplan. > > On Fri, Jan 6, 2012 at 8:18 AM, Bharat Lalcheta > <bharatlalcheta(a)gmail.com> wrote: > > Hiii, > > > > I am new to freeswitch. Prior to freeswitch i was using asterisk. > > > > I have 200 extensions working in my office and want to move all to > > freeswitch from asterisk. > > > > In asterisk, i can give codec selection and preferance in sip.conf to all > > extensions. In the same way i created 200 extensions under internal > profile > > in freeswitch. > > > > Follwing is one example.... > > ----------------------------------------------------------------- > > <include> > > <user id="590"> > > <params> > > <param name="password" value="590"/> > > <param name="vm-password" value=""/> > > <param name="vm-enabled" value="true"/> > > <param name="inbound_codec_prefs" value="PCMA,H264"/> > > <param name="outbound_codec_prefs" value="PCMA,H264"/> > > </params> > > <variables> > > <variable name="accountcode" value=""/> > > <variable name="user_context" value="default"/> > > <variable name="max-calls" value="2"/> > > <variable name="bypass_media_after_bridge" value="no"/> > > </variables> > > </user> > > </include> > > ---------------------------------------------------------- > > > > Now when ever i called to 590 freeswitch sends all codecs to 590 sip > phone > > other than defined in 590.xml. It is seding codes which is mentioned in > my > > conf/sip_profiles/internal.xml and codec negotiation done on whatever > codec > > my sip phone having. > > > > I want to use different codecs for different extensions. > > > > Is it common behaviour of Freeswitch ? Should i override codec > prerfrance in > > my extension list from my internal profile or not ? > > > > If no, then is it that i have to create 200 profiles in freeswitch to > solve > > this problem ? > > > > Please guide me and provide solution for the same > > > > > > Thanks in advance > > > > Bharat Lalcheta > > > > _________________________________________________________________________ > > 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 > > > > > > -- > Kristian Kielhofner > > > > > ---------- Forwarded message ---------- > From: curriegrad2004 <curriegrad2004(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Fri, 6 Jan 2012 07:37:06 -0800 > Subject: Re: [Freeswitch-users] Codec Preferance > > In theory he could use those variable names but it would involve some > extra work ;) > On 2012-01-06 7:24 AM, "Kristian Kielhofner" <kris(a)kriskinc.com> wrote: > >> inbound_codec_prefs >> outbound_codec_prefs >> >> These are Sofia profile configuration options. They are NOT valid >> options for the directory UNLESS you're setting them as variable to >> use in your dialplan later. >> >> If you want to control codes on a per-user basis you have to set late >> negotiation and use the dialplan. >> >> On Fri, Jan 6, 2012 at 8:18 AM, Bharat Lalcheta >> <bharatlalcheta(a)gmail.com> wrote: >> > Hiii, >> > >> > I am new to freeswitch. Prior to freeswitch i was using asterisk. >> > >> > I have 200 extensions working in my office and want to move all to >> > freeswitch from asterisk. >> > >> > In asterisk, i can give codec selection and preferance in sip.conf to >> all >> > extensions. In the same way i created 200 extensions under internal >> profile >> > in freeswitch. >> > >> > Follwing is one example.... >> > ----------------------------------------------------------------- >> > <include> >> > <user id="590"> >> > <params> >> > <param name="password" value="590"/> >> > <param name="vm-password" value=""/> >> > <param name="vm-enabled" value="true"/> >> > <param name="inbound_codec_prefs" value="PCMA,H264"/> >> > <param name="outbound_codec_prefs" value="PCMA,H264"/> >> > </params> >> > <variables> >> > <variable name="accountcode" value=""/> >> > <variable name="user_context" value="default"/> >> > <variable name="max-calls" value="2"/> >> > <variable name="bypass_media_after_bridge" value="no"/> >> > </variables> >> > </user> >> > </include> >> > ---------------------------------------------------------- >> > >> > Now when ever i called to 590 freeswitch sends all codecs to 590 sip >> phone >> > other than defined in 590.xml. It is seding codes which is mentioned in >> my >> > conf/sip_profiles/internal.xml and codec negotiation done on whatever >> codec >> > my sip phone having. >> > >> > I want to use different codecs for different extensions. >> > >> > Is it common behaviour of Freeswitch ? Should i override codec >> prerfrance in >> > my extension list from my internal profile or not ? >> > >> > If no, then is it that i have to create 200 profiles in freeswitch to >> solve >> > this problem ? >> > >> > Please guide me and provide solution for the same >> > >> > >> > Thanks in advance >> > >> > Bharat Lalcheta >> > >> > >> _________________________________________________________________________ >> > 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 >> > >> >> >> >> -- >> Kristian Kielhofner >> >> _________________________________________________________________________ >> 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 >> > > > ---------- Forwarded message ---------- > From: Bharat Lalcheta <bharatlalcheta(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Fri, 6 Jan 2012 21:10:01 +0530 > Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 67, Issue 51 > can you please explain in details what you want to tell ? > > > > >> ---------- Forwarded message ---------- >> From: curriegrad2004 <curriegrad2004(a)gmail.com> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Date: Fri, 6 Jan 2012 07:11:52 -0800 >> Subject: Re: [Freeswitch-users] Codec Preferance >> >> I highly would recommend that you change the name of those codecs to >> something else because you might be making matters worse later down the >> road >> On 2012-01-06 5:21 AM, "Bharat Lalcheta" <bharatlalcheta(a)gmail.com> >> wrote: >> >>> Hiii, >>> >>> I am new to freeswitch. Prior to freeswitch i was using asterisk. >>> >>> I have 200 extensions working in my office and want to move all to >>> freeswitch from asterisk. >>> >>> In asterisk, i can give codec selection and preferance in sip.conf to >>> all extensions. In the same way i created 200 extensions under internal >>> profile in freeswitch. >>> >>> Follwing is one example.... >>> ----------------------------------------------------------------- >>> <include> >>> <user id="590"> >>> <params> >>> <param name="password" value="590"/> >>> <param name="vm-password" value=""/> >>> <param name="vm-enabled" value="true"/> >>> <param name="inbound_codec_prefs" value="PCMA,H264"/> >>> <param name="outbound_codec_prefs" value="PCMA,H264"/> >>> </params> >>> <variables> >>> <variable name="accountcode" value=""/> >>> <variable name="user_context" value="default"/> >>> <variable name="max-calls" value="2"/> >>> <variable name="bypass_media_after_bridge" value="no"/> >>> </variables> >>> </user> >>> </include> >>> ---------------------------------------------------------- >>> >>> Now when ever i called to 590 freeswitch sends all codecs to 590 sip >>> phone other than defined in 590.xml. It is seding codes which is mentioned >>> in my conf/sip_profiles/internal.xml and codec negotiation done on whatever >>> codec my sip phone having. >>> >>> I want to use different codecs for different extensions. >>> >>> Is it common behaviour of Freeswitch ? Should i override codec >>> prerfrance in my extension list from my internal profile or not ? >>> >>> If no, then is it that i have to create 200 profiles in freeswitch to >>> solve this problem ? >>> >>> Please guide me and provide solution for the same >>> >>> >>> Thanks in advance >>> >>> Bharat Lalcheta >>> >>> _________________________________________________________________________ >>> 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 >>> >>> >> >> ---------- Forwarded message ---------- >> From: Peter Olsson <peter.olsson(a)visionutveckling.se> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Date: Fri, 6 Jan 2012 15:21:00 +0000 >> Subject: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR >> Are you still using ignore_early_media=true - this must be set for this >> to work correctly. >> >> You will see a EXECUTE log line when FS executes the application, with >> ignore_early_media enabled it shouldn't execute until the call has been >> answered. I just tried it myself, and it works as expected. >> >> Example "originate {ignore_early_media=true}sofia/internal/number@host&park()" >> >> Park application is only executed after the call was answered. >> >> /Peter >> >> ________________________________________ >> Från: freeswitch-users-bounces(a)lists.freeswitch.org [ >> freeswitch-users-bounces(a)lists.freeswitch.org] för Oliver Schenk [ >> olimonkey(a)gmail.com] >> Skickat: den 6 januari 2012 12:04 >> Till: FreeSWITCH Users Help >> Ämne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR >> >> Because I'm using an FXO card with voice, I added something to my >> CISCO conf. Many others had the same thing: >> >> >> voice-port 0/3/0 >> ... >> supervisory disconnect dualtone mid-call >> supervisory answer dualtone <---- ADDED THIS ONE >> ... >> >> >> >> Once I added this, the FS output now just showed the following while >> the phone was ringing: >> >> 2012-01-05 16:19:31.644440 [NOTICE] switch_channel.c:816 New Channel >> sofia/internal/109212xxxx(a)192.168.x.x >> [69e3f13d-1e2a-409e-97a4-b5526ea6e4ec] >> 2012-01-05 16:19:35.124882 [INFO] sofia.c:740 >> sofia/internal/109212xxxx(a)192.168.x.x Update Callee ID to "Outbound >> Call" <109212xxxx> >> 2012-01-05 16:19:35.126883 [NOTICE] sofia_glue.c:3793 Pre-Answer >> sofia/internal/109212xxxx(a)192.168.x.x! >> >> >> Where as previous it would show the above and also show the following: >> >> 2012-01-05 16:19:35.127883 [INFO] switch_channel.c:2456 >> sofia/internal/109212xxxx(a)192.168.x.x Flipping CID from "" >> <0000000000> to "Outbound Call" <109212xxxx> >> 2012-01-05 16:19:35.137384 [INFO] sofia.c:740 >> sofia/internal/109212xxxx(a)192.168.x.x Update Callee ID to "109212xxxx" >> <1092122856> >> 2012-01-05 16:19:35.138384 [NOTICE] sofia.c:5296 Channel >> [sofia/internal/109212xxxx(a)192.168.x.x] has been answered >> >> >> >> BUT, the IVR still started playing even before I pick up the phone. >> Hmmmm.....so why is FS still starting the managed application when the >> call has not been answered yet. Are we all sure that the managed >> application should not be executed until the call "has been answered" >> shows up in the log file? >> >> >> Will have to keep testing on monday as I don't have access to the >> CISCO from where i am now. I'll have to see whether the CISCO changes >> had any impact on the times at which the SIP messages are sent back >> and forth. Especially the 200 OK message. >> >> >> Thanks again for help, maybe getting somewhere now...... >> >> Oliver >> >> >> >> >> On Fri, Jan 6, 2012 at 4:20 PM, Peter Olsson >> <peter.olsson(a)visionutveckling.se> wrote: >> > If it sends 200 OK right after 183, this IS the problem. >> > >> > 200 OK means that the call was answered, it should not be sent until >> the call was actually picked up in the remote end. When 200 OK arrives to >> FS it will execute your app, and you will start playing the files. >> > >> > Seems to me there is something broken in the Cisco. >> > >> > /Peter >> > >> > ________________________________________ >> > Från: freeswitch-users-bounces(a)lists.freeswitch.org [ >> freeswitch-users-bounces(a)lists.freeswitch.org] för Oliver Schenk [ >> olimonkey(a)gmail.com] >> > Skickat: den 6 januari 2012 06:55 >> > Till: FreeSWITCH Users Help >> > Ämne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR >> > >> > I've tried looking at disable-early-media configuration command, but >> > that didn't work and I doubt that has anything to do with the CISCO >> > sending a 200 OK right after a 183 SESSION PROGRESS. >> > >> > >> > >> > >> > On Fri, Jan 6, 2012 at 9:20 AM, Brian West <brian(a)freeswitch.org> >> wrote: >> >> Thats what the 183 is.. 180 vs 183 are kinda sketchy in some devices.. >> 180 >> >> is usually RINGING (generate ringback locally) while a 183 has >> media... aka >> >> early media and usually provides ringback inband. >> >> >> >> /b >> >> >> >> On Jan 5, 2012, at 7:13 PM, Oliver Schenk wrote: >> >> >> >> Shouldn't there be a 180 RINGING somewhere in there? >> >> >> >> >> >> >> >> >> >> On Fri, Jan 6, 2012 at 8:25 AM, Oliver Schenk <olimonkey(a)gmail.com> >> wrote: >> >> >> >> I just noticed something else, if I don't pick up the phone at all. >> >> >> >> The IVR just keeps playing until the menu timeout kicks in. >> >> >> >> >> >> So here is a CISCO SIP log: >> >> >> >> http://pastebin.com/Y9sYkuxi >> >> >> >> >> >> The FS server is 192.168.x.50 and the CISCO is 192.168.x.1. >> >> >> >> I hope the CISCO log is readable, it's a bit long because I just did >> >> >> >> "debug ccsip all". >> >> >> >> >> >> >> >> >> >> In this test I didn't bother picking up the phone at all, but I can >> >> >> >> see that FS answered anyway and the IVR kept playing until it timed >> >> >> >> out. >> >> >> >> I'm not an expert, but here is what I picked out of it: >> >> >> >> >> >> At 00:08:10 we get a >> >> >> >> Received: "INVITE sip:109212xxxx@192.168.x.1 SIP/2.0" >> >> >> >> >> >> the further down at the same timestamp we get >> >> >> >> Sent: "SIP/2.0 100 Trying" >> >> >> >> >> >> At 00:08:13 we get a >> >> >> >> Sent: "SIP/2.0 183 Session Progress" >> >> >> >> >> >> At 00:18:13 we get a >> >> >> >> Sent: "SIP/2.0 200 OK" >> >> >> >> >> >> Then at the same timestamp we get: >> >> >> >> Received: "ACK sip:109212xxxx@192.168.x.1:5060 SIP/2.0" >> >> >> >> >> >> >> >> >> >> Once the IVR times out at 00:09:16 we get >> >> >> >> Received: "BYE sip:109212xxxx@192.168.x.1:5060 SIP/2.0" >> >> >> >> >> >> And then the reply right after >> >> >> >> Sent: "SIP/2.0 200 OK" >> >> >> >> >> >> >> >> >> >> So I think you were right, the CISCO is sending back an "OK" 3 seconds >> >> >> >> after the "INVITE" is received. >> >> >> >> >> >> >> >> >> >> The part that is beyond my field of expertise so far is WHY? >> >> >> >> >> >> >> >> >> >> Thanks, >> >> >> >> >> >> >> >> Oliver >> >> >> >> >> >> >> >> >> >> On Fri, Jan 6, 2012 at 8:04 AM, Oliver Schenk <olimonkey(a)gmail.com> >> wrote: >> >> >> >> By the way: >> >> >> >> >> >> I tried {ignore_early_media=true} as well, but as I think we >> >> >> >> determined, my problem is probably with the CISCO telling FS that the >> >> >> >> call has been answered when really it hasn't yet. >> >> >> >> >> >> >> >> >> >> On Fri, Jan 6, 2012 at 8:01 AM, Oliver Schenk <olimonkey(a)gmail.com> >> wrote: >> >> >> >> Thanks for the help so far. >> >> >> >> >> >> >> >> Here is a pastebin of FreeSWITCH output: >> >> >> >> http://pastebin.com/i6Qgc7ws >> >> >> >> >> >> Notice how the "has been answered" log message comes immediately >> >> >> >> (within a few milliseconds) after the call was originated. I think >> >> >> >> this would suggest that the CISCO is immediately sending a 200 OK, as >> >> >> >> you suggested. I also turned on CISCO debugging, but I'm just trying >> >> >> >> to figure out how to get the information regarding SIP messages back >> >> >> >> to Freeswitch. I'll run the test again and see if I can get some >> >> >> >> useful CISCO debug. >> >> >> >> >> >> Which "debug ccsip" commands are relevant to what I want for the CISCO >> >> >> >> SIP debugging? >> >> >> >> >> >> >> >> Thanks! >> >> >> >> >> >> >> >> >> >> >> >> 2012/1/6 Gustavo Mársico <gustavomarsico(a)gmail.com>: >> >> >> >> I think I've a similar problem related to callcenter app. When I made >> an >> >> originate like this: >> >> >> >> >> >> originate loopback/2500/default/XML &bridge(user/2001) >> >> >> >> >> >> 2500 is an extension that leads to a callcenter application. In this >> case, >> >> we dial first to the queue and when an agent answered we call to the >> >> customer. As far as I know >> >> >> >> When the A-leg reaches to the queue, without selecting an agent, the >> call is >> >> automatically sent to the B-leg. As far as I see, there is a pre-answer >> >> method that fs needs to send the media to A-leg. >> >> >> >> In order to try to avoid this, I tried using ignore_early_media=true >> as part >> >> of the originate in A-leg and/or B-leg, with no luck. >> >> >> >> >> >> originate {ignore_early_media=true}loopback/2500/default/XML >> >> &bridge({ignore_early_media=true}user/2001) >> >> >> >> >> >> Dialplan: loopback/2500-b Regex (PASS) [CallCenter_Click2Call] >> >> destination_number(2500) =~ /^(2500)$/ break=on-false >> >> >> >> Dialplan: loopback/2500-b Action set(ignore_early_media=true) >> >> >> >> Dialplan: loopback/2500-b Action callcenter(click2call) >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:154 >> >> (loopback/2500-b) State Change CS_ROUTING -> CS_EXECUTE >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:1180 Send >> signal >> >> loopback/2500-b [BREAK] >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >> >> CHANNEL KILL >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:410 >> >> (loopback/2500-b) State ROUTING going to sleep >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:362 >> >> (loopback/2500-b) Running State Change CS_EXECUTE >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:417 >> >> (loopback/2500-b) State EXECUTE >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:375 loopback/2500-b >> >> CHANNEL EXECUTE >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:192 >> >> loopback/2500-b Standard EXECUTE >> >> >> >> EXECUTE loopback/2500-b set(open=true) >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b >> SET >> >> [open]=[true] >> >> >> >> EXECUTE loopback/2500-b >> >> >> hash(insert/10.8.0.70-spymap/0000000000/fef7d864-b3c7-407c-a6e1-94386642bfbb) >> >> >> >> EXECUTE loopback/2500-b >> hash(insert/10.8.0.70-last_dial/0000000000/2500) >> >> >> >> EXECUTE loopback/2500-b >> >> >> hash(insert/10.8.0.70-last_dial/global/fef7d864-b3c7-407c-a6e1-94386642bfbb) >> >> >> >> EXECUTE loopback/2500-b set(RFC2822_DATE=Thu, 05 Jan 2012 13:36:08 >> -0300) >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b >> SET >> >> [RFC2822_DATE]=[Thu, 05 Jan 2012 13:36:08 -0300] >> >> >> >> EXECUTE loopback/2500-b set(ignore_early_media=true) >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b >> SET >> >> [ignore_early_media]=[true] >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:2133 >> Application >> >> callcenter Requires media! pre_answering channel loopback/2500-b >> >> >> >> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:760 Pre-Answer >> >> loopback/2500-a! >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 >> (loopback/2500-a) >> >> Callstate Change RINGING -> EARLY >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send >> signal >> >> loopback/2500-b [BREAK] >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >> >> CHANNEL KILL >> >> >> >> 2012-01-05 13:36:08.541517 [NOTICE] switch_core_session.c:2135 >> Pre-Answer >> >> loopback/2500-b! >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 >> (loopback/2500-b) >> >> Callstate Change RINGING -> EARLY >> >> >> >> EXECUTE loopback/2500-b callcenter(click2call) >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 >> (loopback/2500-a) >> >> Callstate Change EARLY -> ACTIVE >> >> >> >> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:755 Channel >> >> [loopback/2500-a] has been answered >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send >> signal >> >> loopback/2500-b [BREAK] >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >> >> CHANNEL KILL >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_ivr_originate.c:3266 >> Originate >> >> Resulted in Success: [loopback/2500-a] >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 >> (loopback/2500-b) >> >> Callstate Change EARLY -> ACTIVE >> >> >> >> 2012-01-05 13:36:08.541517 [INFO] switch_channel.c:2708 >> loopback/2500-a >> >> Flipping CID from "" <0000000000> to "Outbound Call" <XML> >> >> >> >> >> >> >> >> >> >> >> >> On Jan 5, 2012, at 4:17 AM, Oliver Schenk wrote: >> >> >> >> >> >> Also, maybe I should be doing something like this: >> >> >> >> >> >> sofia/gateway/mygatewayname/1091234567 '&managed(ivrAppName)' >> >> >> >> >> >> instead of: >> >> >> >> >> >> sofia/internal/1091234567(a)192.168.x.x '&managed(ivrAppName)' >> >> >> >> >> >> >> >> but, I don't really have the CISCO configured as a gateway, nor do I >> >> >> >> know how really...probably not on the right track there. >> >> >> >> >> >> >> >> >> >> >> >> On Thu, Jan 5, 2012 at 3:06 PM, Oliver Schenk <olimonkey(a)gmail.com> >> wrote: >> >> >> >> *bump* >> >> >> >> >> >> >> >> So I think maybe the way I'm doing the originate is the problem? In my >> >> >> >> call string I'm creating a connection directly from the CISCO >> >> >> >> (192.168.x.x) to the managed application, which may be why it starts >> >> >> >> playing straight away? >> >> >> >> >> >> Maybe I should be originating a call first and then only once I know >> >> >> >> the other side has picked up will I bridge the call to the IVR managed >> >> >> >> application. >> >> >> >> >> >> Problem is I dunno how to tell whether the other person has picked up >> >> >> >> (or even if the cisco is going to tell me) and I don't know how to do >> >> >> >> things to a call once it has been established. >> >> >> >> >> >> >> >> I'm currently reading the Dialplan wiki page, hoping to get something >> >> >> >> out of it there. >> >> >> >> >> >> >> >> Cheers >> >> >> >> >> >> Oliver >> >> >> >> >> >> >> >> On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk <olimonkey(a)gmail.com> >> wrote: >> >> >> >> I've been battling while creating an IVR using FreeSWITCH mod_managed >> >> >> >> and connecting through a CISCO 2811. Most things now work quite well, >> >> >> >> but I am having a few issues with the way the system answers calls (or >> >> >> >> doesn't answer calls...). >> >> >> >> >> >> I have FreeSWITCH running as a windows service on Windows Server 2008, >> >> >> >> which is connected via LAN to a CISCO 2811 with a 4 port FXO card, >> >> >> >> which is then connected to a POTS phone line. >> >> >> >> >> >> >> >> Take the following scenario: >> >> >> >> >> >> 1. Managed .NET application creates a call string and uses ESL to talk >> >> >> >> to freeswitch and originate a call: >> >> >> >> >> >> string callstring = >> >> >> >> >> "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567(a)192.168.x.x >> >> >> >> '&managed(ivrAppName)'"; >> >> >> >> eslConnection.API("originate", callstring); >> >> >> >> >> >> where 192.168.x.x is the CISCO IP. >> >> >> >> >> >> 2. The CISCO sees that the phone number (1091234567) starts with a 1 >> >> >> >> so it uses FXO port 1 and strips the 1 and uses the remaining phone >> >> >> >> number (091234567) to make the call. >> >> >> >> >> >> 3. My phone rings, I pick up and I can hear my IVR playing. >> >> >> >> >> >> >> >> >> >> These are my current problems: >> >> >> >> >> >> - IVR starts playing before I even pick up the phone. This means that >> >> >> >> if the system calls a mobile phone and the person doesn't pick up, the >> >> >> >> IVR will start playing and eventually the mobile phone will divert to >> >> >> >> voice mail. Obviously I then get a missed call and an sms saying I >> >> >> >> have a new voice mail, which is annoying. Instead I would like it to >> >> >> >> KNOW that no one has picked up, but I don't know how to do this. >> >> >> >> Somehow the CISCO needs to be able to tell FreeSWITCH that the call >> >> >> >> has not yet been answered. For some reason however as soon as the >> >> >> >> CISCO starts calling FreeSWITCH thinks the call is already connected. >> >> >> >> It doesn't know that the CISCO is actually still ringing. Maybe I'm >> >> >> >> doing originate the wrong way or something ... >> >> >> >> >> >> - The phone only rings for about 10 seconds before hanging up. I've >> >> >> >> tried "call_timeout", "bridge_answer_timeout". I've also tried setting >> >> >> >> CISCO "ring number". Nothing works, my phone still only rings for >> >> >> >> about 10 seconds. I don't know if this is a FreeSWITCH issue or a >> >> >> >> CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just >> >> >> >> starts playing even if no one answers the phone. >> >> >> >> >> >> >> >> >> >> >> >> >> >> CISCO Config for relevant FXO port: >> >> >> >> >> >> voice service voip >> >> >> >> allow-connections h323 to h323 >> >> >> >> allow-connections h323 to sip >> >> >> >> allow-connections sip to h323 >> >> >> >> allow-connections sip to sip >> >> >> >> no supplementary-service h450.2 >> >> >> >> no supplementary-service h450.3 >> >> >> >> supplementary-service h450.12 >> >> >> >> no supplementary-service sip moved-temporarily >> >> >> >> no supplementary-service sip refer >> >> >> >> fax protocol cisco >> >> >> >> sip >> >> >> >> registrar server expires max 3600 min 3600 >> >> >> >> no update-callerid >> >> >> >> no call service stop >> >> >> >> >> >> voice-port 0/3/2 >> >> >> >> output attenuation -3 >> >> >> >> no comfort-noise >> >> >> >> cptone AU >> >> >> >> impedance complex1 >> >> >> >> caller-id enable >> >> >> >> ! >> >> >> >> dial-peer voice 100 pots >> >> >> >> preference 1 >> >> >> >> destination-pattern 1T >> >> >> >> port 0/3/2 >> >> >> >> ! >> >> >> >> >> >> >> >> >> >> Many Thanks, >> >> >> >> >> >> Oliver >> >> >> >> >> >> >> _________________________________________________________________________ >> >> >> >> Professional FreeSWITCH Consulting Services: >> >> >> >> consulting(a)freeswitch.org >> >> >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> >> >> >> http://www.cudatel.com >> >> >> >> >> >> Official FreeSWITCH Sites >> >> >> >> http://www.freeswitch.org >> >> >> >> http://wiki.freeswitch.org >> >> >> >> http://www.cluecon.com >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> >> >> FreeSWITCH-users(a)lists.freeswitch.org >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> >> http://www.freeswitch.org >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> >> >> >> Professional FreeSWITCH Consulting Services: >> >> >> >> consulting(a)freeswitch.org >> >> >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> >> >> >> http://www.cudatel.com >> >> >> >> >> >> Official FreeSWITCH Sites >> >> >> >> http://www.freeswitch.org >> >> >> >> http://wiki.freeswitch.org >> >> >> >> http://www.cluecon.com >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> >> >> FreeSWITCH-users(a)lists.freeswitch.org >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> >> http://www.freeswitch.org >> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting(a)freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> >> http://www.cudatel.com >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users(a)lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> >> -- >> >> Brian West >> >> FreeSWITCH Solutions, LLC >> >> Phone: +1 (918) 420-9266 >> >> Fax: +1 (918) 420-9267 >> >> brian(a)freeswitch.org >> >> http://www.freeswitch.org >> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting(a)freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> >> http://www.cudatel.com >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users(a)lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting(a)freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> > http://www.cudatel.com >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users(a)lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting(a)freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> > http://www.cudatel.com >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users(a)lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> !DSPAM:4f06d49b32762089563979! >> >> >> >> >> _______________________________________________ >> 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 >> >> > > > -- > Bharat Lalcheta > > _______________________________________________ > 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 > > -- Bharat Lalcheta

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 67, Issue 51
by Bharat Lalcheta 06 Jan '12

06 Jan '12

can you please explain in details what you want to tell ? > ---------- Forwarded message ---------- > From: curriegrad2004 <curriegrad2004(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Fri, 6 Jan 2012 07:11:52 -0800 > Subject: Re: [Freeswitch-users] Codec Preferance > > I highly would recommend that you change the name of those codecs to > something else because you might be making matters worse later down the > road > On 2012-01-06 5:21 AM, "Bharat Lalcheta" <bharatlalcheta(a)gmail.com> wrote: > >> Hiii, >> >> I am new to freeswitch. Prior to freeswitch i was using asterisk. >> >> I have 200 extensions working in my office and want to move all to >> freeswitch from asterisk. >> >> In asterisk, i can give codec selection and preferance in sip.conf to all >> extensions. In the same way i created 200 extensions under internal profile >> in freeswitch. >> >> Follwing is one example.... >> ----------------------------------------------------------------- >> <include> >> <user id="590"> >> <params> >> <param name="password" value="590"/> >> <param name="vm-password" value=""/> >> <param name="vm-enabled" value="true"/> >> <param name="inbound_codec_prefs" value="PCMA,H264"/> >> <param name="outbound_codec_prefs" value="PCMA,H264"/> >> </params> >> <variables> >> <variable name="accountcode" value=""/> >> <variable name="user_context" value="default"/> >> <variable name="max-calls" value="2"/> >> <variable name="bypass_media_after_bridge" value="no"/> >> </variables> >> </user> >> </include> >> ---------------------------------------------------------- >> >> Now when ever i called to 590 freeswitch sends all codecs to 590 sip >> phone other than defined in 590.xml. It is seding codes which is mentioned >> in my conf/sip_profiles/internal.xml and codec negotiation done on whatever >> codec my sip phone having. >> >> I want to use different codecs for different extensions. >> >> Is it common behaviour of Freeswitch ? Should i override codec prerfrance >> in my extension list from my internal profile or not ? >> >> If no, then is it that i have to create 200 profiles in freeswitch to >> solve this problem ? >> >> Please guide me and provide solution for the same >> >> >> Thanks in advance >> >> Bharat Lalcheta >> >> _________________________________________________________________________ >> 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 >> >> > > ---------- Forwarded message ---------- > From: Peter Olsson <peter.olsson(a)visionutveckling.se> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Fri, 6 Jan 2012 15:21:00 +0000 > Subject: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR > Are you still using ignore_early_media=true - this must be set for this to > work correctly. > > You will see a EXECUTE log line when FS executes the application, with > ignore_early_media enabled it shouldn't execute until the call has been > answered. I just tried it myself, and it works as expected. > > Example "originate {ignore_early_media=true}sofia/internal/number@host&park()" > > Park application is only executed after the call was answered. > > /Peter > > ________________________________________ > Från: freeswitch-users-bounces(a)lists.freeswitch.org [ > freeswitch-users-bounces(a)lists.freeswitch.org] för Oliver Schenk [ > olimonkey(a)gmail.com] > Skickat: den 6 januari 2012 12:04 > Till: FreeSWITCH Users Help > Ämne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR > > Because I'm using an FXO card with voice, I added something to my > CISCO conf. Many others had the same thing: > > > voice-port 0/3/0 > ... > supervisory disconnect dualtone mid-call > supervisory answer dualtone <---- ADDED THIS ONE > ... > > > > Once I added this, the FS output now just showed the following while > the phone was ringing: > > 2012-01-05 16:19:31.644440 [NOTICE] switch_channel.c:816 New Channel > sofia/internal/109212xxxx(a)192.168.x.x > [69e3f13d-1e2a-409e-97a4-b5526ea6e4ec] > 2012-01-05 16:19:35.124882 [INFO] sofia.c:740 > sofia/internal/109212xxxx(a)192.168.x.x Update Callee ID to "Outbound > Call" <109212xxxx> > 2012-01-05 16:19:35.126883 [NOTICE] sofia_glue.c:3793 Pre-Answer > sofia/internal/109212xxxx(a)192.168.x.x! > > > Where as previous it would show the above and also show the following: > > 2012-01-05 16:19:35.127883 [INFO] switch_channel.c:2456 > sofia/internal/109212xxxx(a)192.168.x.x Flipping CID from "" > <0000000000> to "Outbound Call" <109212xxxx> > 2012-01-05 16:19:35.137384 [INFO] sofia.c:740 > sofia/internal/109212xxxx(a)192.168.x.x Update Callee ID to "109212xxxx" > <1092122856> > 2012-01-05 16:19:35.138384 [NOTICE] sofia.c:5296 Channel > [sofia/internal/109212xxxx(a)192.168.x.x] has been answered > > > > BUT, the IVR still started playing even before I pick up the phone. > Hmmmm.....so why is FS still starting the managed application when the > call has not been answered yet. Are we all sure that the managed > application should not be executed until the call "has been answered" > shows up in the log file? > > > Will have to keep testing on monday as I don't have access to the > CISCO from where i am now. I'll have to see whether the CISCO changes > had any impact on the times at which the SIP messages are sent back > and forth. Especially the 200 OK message. > > > Thanks again for help, maybe getting somewhere now...... > > Oliver > > > > > On Fri, Jan 6, 2012 at 4:20 PM, Peter Olsson > <peter.olsson(a)visionutveckling.se> wrote: > > If it sends 200 OK right after 183, this IS the problem. > > > > 200 OK means that the call was answered, it should not be sent until the > call was actually picked up in the remote end. When 200 OK arrives to FS it > will execute your app, and you will start playing the files. > > > > Seems to me there is something broken in the Cisco. > > > > /Peter > > > > ________________________________________ > > Från: freeswitch-users-bounces(a)lists.freeswitch.org [ > freeswitch-users-bounces(a)lists.freeswitch.org] för Oliver Schenk [ > olimonkey(a)gmail.com] > > Skickat: den 6 januari 2012 06:55 > > Till: FreeSWITCH Users Help > > Ämne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR > > > > I've tried looking at disable-early-media configuration command, but > > that didn't work and I doubt that has anything to do with the CISCO > > sending a 200 OK right after a 183 SESSION PROGRESS. > > > > > > > > > > On Fri, Jan 6, 2012 at 9:20 AM, Brian West <brian(a)freeswitch.org> wrote: > >> Thats what the 183 is.. 180 vs 183 are kinda sketchy in some devices.. > 180 > >> is usually RINGING (generate ringback locally) while a 183 has media... > aka > >> early media and usually provides ringback inband. > >> > >> /b > >> > >> On Jan 5, 2012, at 7:13 PM, Oliver Schenk wrote: > >> > >> Shouldn't there be a 180 RINGING somewhere in there? > >> > >> > >> > >> > >> On Fri, Jan 6, 2012 at 8:25 AM, Oliver Schenk <olimonkey(a)gmail.com> > wrote: > >> > >> I just noticed something else, if I don't pick up the phone at all. > >> > >> The IVR just keeps playing until the menu timeout kicks in. > >> > >> > >> So here is a CISCO SIP log: > >> > >> http://pastebin.com/Y9sYkuxi > >> > >> > >> The FS server is 192.168.x.50 and the CISCO is 192.168.x.1. > >> > >> I hope the CISCO log is readable, it's a bit long because I just did > >> > >> "debug ccsip all". > >> > >> > >> > >> > >> In this test I didn't bother picking up the phone at all, but I can > >> > >> see that FS answered anyway and the IVR kept playing until it timed > >> > >> out. > >> > >> I'm not an expert, but here is what I picked out of it: > >> > >> > >> At 00:08:10 we get a > >> > >> Received: "INVITE sip:109212xxxx@192.168.x.1 SIP/2.0" > >> > >> > >> the further down at the same timestamp we get > >> > >> Sent: "SIP/2.0 100 Trying" > >> > >> > >> At 00:08:13 we get a > >> > >> Sent: "SIP/2.0 183 Session Progress" > >> > >> > >> At 00:18:13 we get a > >> > >> Sent: "SIP/2.0 200 OK" > >> > >> > >> Then at the same timestamp we get: > >> > >> Received: "ACK sip:109212xxxx@192.168.x.1:5060 SIP/2.0" > >> > >> > >> > >> > >> Once the IVR times out at 00:09:16 we get > >> > >> Received: "BYE sip:109212xxxx@192.168.x.1:5060 SIP/2.0" > >> > >> > >> And then the reply right after > >> > >> Sent: "SIP/2.0 200 OK" > >> > >> > >> > >> > >> So I think you were right, the CISCO is sending back an "OK" 3 seconds > >> > >> after the "INVITE" is received. > >> > >> > >> > >> > >> The part that is beyond my field of expertise so far is WHY? > >> > >> > >> > >> > >> Thanks, > >> > >> > >> > >> Oliver > >> > >> > >> > >> > >> On Fri, Jan 6, 2012 at 8:04 AM, Oliver Schenk <olimonkey(a)gmail.com> > wrote: > >> > >> By the way: > >> > >> > >> I tried {ignore_early_media=true} as well, but as I think we > >> > >> determined, my problem is probably with the CISCO telling FS that the > >> > >> call has been answered when really it hasn't yet. > >> > >> > >> > >> > >> On Fri, Jan 6, 2012 at 8:01 AM, Oliver Schenk <olimonkey(a)gmail.com> > wrote: > >> > >> Thanks for the help so far. > >> > >> > >> > >> Here is a pastebin of FreeSWITCH output: > >> > >> http://pastebin.com/i6Qgc7ws > >> > >> > >> Notice how the "has been answered" log message comes immediately > >> > >> (within a few milliseconds) after the call was originated. I think > >> > >> this would suggest that the CISCO is immediately sending a 200 OK, as > >> > >> you suggested. I also turned on CISCO debugging, but I'm just trying > >> > >> to figure out how to get the information regarding SIP messages back > >> > >> to Freeswitch. I'll run the test again and see if I can get some > >> > >> useful CISCO debug. > >> > >> > >> Which "debug ccsip" commands are relevant to what I want for the CISCO > >> > >> SIP debugging? > >> > >> > >> > >> Thanks! > >> > >> > >> > >> > >> > >> 2012/1/6 Gustavo Mársico <gustavomarsico(a)gmail.com>: > >> > >> I think I've a similar problem related to callcenter app. When I made an > >> originate like this: > >> > >> > >> originate loopback/2500/default/XML &bridge(user/2001) > >> > >> > >> 2500 is an extension that leads to a callcenter application. In this > case, > >> we dial first to the queue and when an agent answered we call to the > >> customer. As far as I know > >> > >> When the A-leg reaches to the queue, without selecting an agent, the > call is > >> automatically sent to the B-leg. As far as I see, there is a pre-answer > >> method that fs needs to send the media to A-leg. > >> > >> In order to try to avoid this, I tried using ignore_early_media=true as > part > >> of the originate in A-leg and/or B-leg, with no luck. > >> > >> > >> originate {ignore_early_media=true}loopback/2500/default/XML > >> &bridge({ignore_early_media=true}user/2001) > >> > >> > >> Dialplan: loopback/2500-b Regex (PASS) [CallCenter_Click2Call] > >> destination_number(2500) =~ /^(2500)$/ break=on-false > >> > >> Dialplan: loopback/2500-b Action set(ignore_early_media=true) > >> > >> Dialplan: loopback/2500-b Action callcenter(click2call) > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:154 > >> (loopback/2500-b) State Change CS_ROUTING -> CS_EXECUTE > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:1180 Send > signal > >> loopback/2500-b [BREAK] > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b > >> CHANNEL KILL > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:410 > >> (loopback/2500-b) State ROUTING going to sleep > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:362 > >> (loopback/2500-b) Running State Change CS_EXECUTE > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:417 > >> (loopback/2500-b) State EXECUTE > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:375 loopback/2500-b > >> CHANNEL EXECUTE > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:192 > >> loopback/2500-b Standard EXECUTE > >> > >> EXECUTE loopback/2500-b set(open=true) > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b > SET > >> [open]=[true] > >> > >> EXECUTE loopback/2500-b > >> > hash(insert/10.8.0.70-spymap/0000000000/fef7d864-b3c7-407c-a6e1-94386642bfbb) > >> > >> EXECUTE loopback/2500-b hash(insert/10.8.0.70-last_dial/0000000000/2500) > >> > >> EXECUTE loopback/2500-b > >> > hash(insert/10.8.0.70-last_dial/global/fef7d864-b3c7-407c-a6e1-94386642bfbb) > >> > >> EXECUTE loopback/2500-b set(RFC2822_DATE=Thu, 05 Jan 2012 13:36:08 > -0300) > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b > SET > >> [RFC2822_DATE]=[Thu, 05 Jan 2012 13:36:08 -0300] > >> > >> EXECUTE loopback/2500-b set(ignore_early_media=true) > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b > SET > >> [ignore_early_media]=[true] > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:2133 > Application > >> callcenter Requires media! pre_answering channel loopback/2500-b > >> > >> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:760 Pre-Answer > >> loopback/2500-a! > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 > (loopback/2500-a) > >> Callstate Change RINGING -> EARLY > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send > signal > >> loopback/2500-b [BREAK] > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b > >> CHANNEL KILL > >> > >> 2012-01-05 13:36:08.541517 [NOTICE] switch_core_session.c:2135 > Pre-Answer > >> loopback/2500-b! > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 > (loopback/2500-b) > >> Callstate Change RINGING -> EARLY > >> > >> EXECUTE loopback/2500-b callcenter(click2call) > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 > (loopback/2500-a) > >> Callstate Change EARLY -> ACTIVE > >> > >> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:755 Channel > >> [loopback/2500-a] has been answered > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send > signal > >> loopback/2500-b [BREAK] > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b > >> CHANNEL KILL > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_ivr_originate.c:3266 > Originate > >> Resulted in Success: [loopback/2500-a] > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 > (loopback/2500-b) > >> Callstate Change EARLY -> ACTIVE > >> > >> 2012-01-05 13:36:08.541517 [INFO] switch_channel.c:2708 loopback/2500-a > >> Flipping CID from "" <0000000000> to "Outbound Call" <XML> > >> > >> > >> > >> > >> > >> On Jan 5, 2012, at 4:17 AM, Oliver Schenk wrote: > >> > >> > >> Also, maybe I should be doing something like this: > >> > >> > >> sofia/gateway/mygatewayname/1091234567 '&managed(ivrAppName)' > >> > >> > >> instead of: > >> > >> > >> sofia/internal/1091234567(a)192.168.x.x '&managed(ivrAppName)' > >> > >> > >> > >> but, I don't really have the CISCO configured as a gateway, nor do I > >> > >> know how really...probably not on the right track there. > >> > >> > >> > >> > >> > >> On Thu, Jan 5, 2012 at 3:06 PM, Oliver Schenk <olimonkey(a)gmail.com> > wrote: > >> > >> *bump* > >> > >> > >> > >> So I think maybe the way I'm doing the originate is the problem? In my > >> > >> call string I'm creating a connection directly from the CISCO > >> > >> (192.168.x.x) to the managed application, which may be why it starts > >> > >> playing straight away? > >> > >> > >> Maybe I should be originating a call first and then only once I know > >> > >> the other side has picked up will I bridge the call to the IVR managed > >> > >> application. > >> > >> > >> Problem is I dunno how to tell whether the other person has picked up > >> > >> (or even if the cisco is going to tell me) and I don't know how to do > >> > >> things to a call once it has been established. > >> > >> > >> > >> I'm currently reading the Dialplan wiki page, hoping to get something > >> > >> out of it there. > >> > >> > >> > >> Cheers > >> > >> > >> Oliver > >> > >> > >> > >> On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk <olimonkey(a)gmail.com> > wrote: > >> > >> I've been battling while creating an IVR using FreeSWITCH mod_managed > >> > >> and connecting through a CISCO 2811. Most things now work quite well, > >> > >> but I am having a few issues with the way the system answers calls (or > >> > >> doesn't answer calls...). > >> > >> > >> I have FreeSWITCH running as a windows service on Windows Server 2008, > >> > >> which is connected via LAN to a CISCO 2811 with a 4 port FXO card, > >> > >> which is then connected to a POTS phone line. > >> > >> > >> > >> Take the following scenario: > >> > >> > >> 1. Managed .NET application creates a call string and uses ESL to talk > >> > >> to freeswitch and originate a call: > >> > >> > >> string callstring = > >> > >> > "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567(a)192.168.x.x > >> > >> '&managed(ivrAppName)'"; > >> > >> eslConnection.API("originate", callstring); > >> > >> > >> where 192.168.x.x is the CISCO IP. > >> > >> > >> 2. The CISCO sees that the phone number (1091234567) starts with a 1 > >> > >> so it uses FXO port 1 and strips the 1 and uses the remaining phone > >> > >> number (091234567) to make the call. > >> > >> > >> 3. My phone rings, I pick up and I can hear my IVR playing. > >> > >> > >> > >> > >> These are my current problems: > >> > >> > >> - IVR starts playing before I even pick up the phone. This means that > >> > >> if the system calls a mobile phone and the person doesn't pick up, the > >> > >> IVR will start playing and eventually the mobile phone will divert to > >> > >> voice mail. Obviously I then get a missed call and an sms saying I > >> > >> have a new voice mail, which is annoying. Instead I would like it to > >> > >> KNOW that no one has picked up, but I don't know how to do this. > >> > >> Somehow the CISCO needs to be able to tell FreeSWITCH that the call > >> > >> has not yet been answered. For some reason however as soon as the > >> > >> CISCO starts calling FreeSWITCH thinks the call is already connected. > >> > >> It doesn't know that the CISCO is actually still ringing. Maybe I'm > >> > >> doing originate the wrong way or something ... > >> > >> > >> - The phone only rings for about 10 seconds before hanging up. I've > >> > >> tried "call_timeout", "bridge_answer_timeout". I've also tried setting > >> > >> CISCO "ring number". Nothing works, my phone still only rings for > >> > >> about 10 seconds. I don't know if this is a FreeSWITCH issue or a > >> > >> CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just > >> > >> starts playing even if no one answers the phone. > >> > >> > >> > >> > >> > >> > >> CISCO Config for relevant FXO port: > >> > >> > >> voice service voip > >> > >> allow-connections h323 to h323 > >> > >> allow-connections h323 to sip > >> > >> allow-connections sip to h323 > >> > >> allow-connections sip to sip > >> > >> no supplementary-service h450.2 > >> > >> no supplementary-service h450.3 > >> > >> supplementary-service h450.12 > >> > >> no supplementary-service sip moved-temporarily > >> > >> no supplementary-service sip refer > >> > >> fax protocol cisco > >> > >> sip > >> > >> registrar server expires max 3600 min 3600 > >> > >> no update-callerid > >> > >> no call service stop > >> > >> > >> voice-port 0/3/2 > >> > >> output attenuation -3 > >> > >> no comfort-noise > >> > >> cptone AU > >> > >> impedance complex1 > >> > >> caller-id enable > >> > >> ! > >> > >> dial-peer voice 100 pots > >> > >> preference 1 > >> > >> destination-pattern 1T > >> > >> port 0/3/2 > >> > >> ! > >> > >> > >> > >> > >> Many Thanks, > >> > >> > >> Oliver > >> > >> > >> > _________________________________________________________________________ > >> > >> Professional FreeSWITCH Consulting Services: > >> > >> consulting(a)freeswitch.org > >> > >> http://www.freeswitchsolutions.com > >> > >> > >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server > >> > >> http://www.cudatel.com > >> > >> > >> Official FreeSWITCH Sites > >> > >> http://www.freeswitch.org > >> > >> http://wiki.freeswitch.org > >> > >> http://www.cluecon.com > >> > >> > >> FreeSWITCH-users mailing list > >> > >> FreeSWITCH-users(a)lists.freeswitch.org > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >> http://www.freeswitch.org > >> > >> > >> > >> > _________________________________________________________________________ > >> > >> Professional FreeSWITCH Consulting Services: > >> > >> consulting(a)freeswitch.org > >> > >> http://www.freeswitchsolutions.com > >> > >> > >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server > >> > >> http://www.cudatel.com > >> > >> > >> Official FreeSWITCH Sites > >> > >> http://www.freeswitch.org > >> > >> http://wiki.freeswitch.org > >> > >> http://www.cluecon.com > >> > >> > >> FreeSWITCH-users mailing list > >> > >> FreeSWITCH-users(a)lists.freeswitch.org > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >> http://www.freeswitch.org > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting(a)freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server > >> http://www.cudatel.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users(a)lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> -- > >> Brian West > >> FreeSWITCH Solutions, LLC > >> Phone: +1 (918) 420-9266 > >> Fax: +1 (918) 420-9267 > >> brian(a)freeswitch.org > >> http://www.freeswitch.org > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting(a)freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server > >> http://www.cudatel.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users(a)lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting(a)freeswitch.org > > http://www.freeswitchsolutions.com > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > > http://www.cudatel.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting(a)freeswitch.org > > http://www.freeswitchsolutions.com > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > > http://www.cudatel.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4f06d49b32762089563979! > > > > > _______________________________________________ > 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 > > -- Bharat Lalcheta

1 0

Codec Preferance
by Bharat Lalcheta 06 Jan '12

06 Jan '12

Hiii, I am new to freeswitch. Prior to freeswitch i was using asterisk. I have 200 extensions working in my office and want to move all to freeswitch from asterisk. In asterisk, i can give codec selection and preferance in sip.conf to all extensions. In the same way i created 200 extensions under internal profile in freeswitch. Follwing is one example.... ----------------------------------------------------------------- <include> <user id="590"> <params> <param name="password" value="590"/> <param name="vm-password" value=""/> <param name="vm-enabled" value="true"/> <param name="inbound_codec_prefs" value="PCMA,H264"/> <param name="outbound_codec_prefs" value="PCMA,H264"/> </params> <variables> <variable name="accountcode" value=""/> <variable name="user_context" value="default"/> <variable name="max-calls" value="2"/> <variable name="bypass_media_after_bridge" value="no"/> </variables> </user> </include> ---------------------------------------------------------- Now when ever i called to 590 freeswitch sends all codecs to 590 sip phone other than defined in 590.xml. It is seding codes which is mentioned in my conf/sip_profiles/internal.xml and codec negotiation done on whatever codec my sip phone having. I want to use different codecs for different extensions. Is it common behaviour of Freeswitch ? Should i override codec prerfrance in my extension list from my internal profile or not ? If no, then is it that i have to create 200 profiles in freeswitch to solve this problem ? Please guide me and provide solution for the same Thanks in advance Bharat Lalcheta

4 4

Help on RTMP
by RuiQiang Huang 06 Jan '12

06 Jan '12

Hi, I'm trying to add RTMP to my freeswitch server. I compiled mod_rtmp.so, mod_rtmp.la and put it under freeswitch/mod/. My freeswitch version is FreeSWITCH version: 1.0.head (git-c8c94f0 2011-08-19 11-52-40 -0500) My rtmp.conf.xml is <configuration name="rtmp.conf" description="RTMP Endpoint"> <profiles> <profile name="default"> <settings> <param name="bind-address" value="0.0.0.0:1935" /> <param name="context" value="public" /> <param name="dialplan" value="XML" />  <param name="auth-calls" value="true" />  <param name="buffer-len" value="50" />  <param name="chunksize" value="512" /> </settings> </profile> </profiles> </configuration> However, I didn't see freeswitch listening on 1935 port and in fs_cli when I run freeswitch > rtmp status It's showing unknown command. My question is, is there anything else needed to get RTMP working? How to know which modules are loaded? Thanks.

3 4

Codec selection
by Bharat Lalcheta 06 Jan '12

06 Jan '12

Hiii, I am new to freeswitch. Prior to freeswitch i was using asterisk. I have 200 extensions working in my office and want to move all to freeswitch from asterisk. In asterisk, i can give codec selection and preferance in sip.conf to all extensions. In the same way i created 200 extensions under internal profile in freeswitch. Follwing is one example.... ----------------------------------------------------------------- <include> <user id="590"> <params> <param name="password" value="590"/> <param name="vm-password" value=""/> <param name="vm-enabled" value="true"/> <param name="inbound_codec_prefs" value="PCMA,H264"/> <param name="outbound_codec_prefs" value="PCMA,H264"/> </params> <variables> <variable name="accountcode" value=""/> <variable name="user_context" value="default"/> <variable name="max-calls" value="2"/> <variable name="bypass_media_after_bridge" value="no"/> </variables> </user> </include> ---------------------------------------------------------- Now when ever i called to 590 freeswitch sends all codecs to 590 sip phone other than defined in 590.xml. It is seding codes which is mentioned in my conf/sip_profiles/internal.xml and codec negotiation done on whatever codec my sip phone having. I want to use different codecs for different extensions. Is it common behaviour of Freeswitch ? Should i override codec prerfrance in my extension list from my internal profile or not ? If no, then is it that i have to create 200 profiles in freeswitch to solve this problem ? Please guide me and provide solution for the same Thanks in advance -- Bharat Lalcheta

1 0

ring tone is heard for 2 sec before busy tone on Lync client
by Manjiri Deshpande 06 Jan '12

06 Jan '12

Hi, I am using freeswitch 107 on windows platform I am working for a product where lync client can call a phone registred with Lync server. When Phone is set to DND mode and lync client calls that phone,I could hear ring tone for abt 2 sec and then busy tone comes. The behvaiour should be ,only busy tone should be heard. I removed "ringback" and "ignore_early_media" from dialplan. Please find dial plan , freeswitch logs and Wireshark capture attached with this mail. From wireshark it is clear that no 180 or 183 messages are flowing.But still from FS logs it could be figure out that,FS is entering into "RINGING" mode. Thanks, Manjiri Deshpande

1 0

Session timeout
by Peter Steinbach 06 Jan '12

06 Jan '12

I have enabled session timers in my sofia internal profile the following way: <param name="enable-timer" value="true"/> <param name="session-timeout" value="120"/> However I do not see any Re-Invite or Update packets sent from Freeswitch to my User Agents. What am I doing wrong? Peter -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

3 2

Re: [Freeswitch-users] Encrypted clients to unencrypted destination ... possible ?
by Bob Smith 05 Jan '12

05 Jan '12

Hello Peter, I will have another go at this over the next couple of days and let you know ! Bob >Hello Bob, > >sure, this does work. I have a couple of phones working with TLS and >SRTP and they all communicate to the outside world through Freeswitch >via SIP and RTP. You have to ensure though that Freeswitch is not is >proxy or bypass media mode. > >Best regards >Peter

1 0

Encrypted clients to unencrypted destination ... possible ?
by Bob Smith 05 Jan '12

05 Jan '12

Hello List, To save me much head-banging and time-wasting, could someone please tell me whether it is possible to configure Freeswitch so that an endpoint can register over TLS, but can route out to and receive calls from the big bad unencrypted outside world ? I've got a basic DDI config at the moment that I can receive calls on fine when endpoint registers as unencrypted. But the moment I turn on TLS, the DDI no longer rings. I've followed the instructions here http://wiki.freeswitch.org/wiki/SIP_TLS to enable TLS (only enabling TLS on the internal Sofia obviously). By the way, it would be great if someone could update those instructions with more detail as to how to get things working with a commercial certificate, as I had no luck making it work, so there's obviously a trick that's missing, and had to resort to a self signed one. Thanks Bob

2 1