- FreeSWITCH-users - FreeSWITCH Mailing Lists

Calling more than 1 variable in condition
by Ahmed Munir 05 Nov '09

05 Nov '09

Hi, In my dial plan I've created a variable named SIP_CALL, PSTN_CALL. If SIP_CALL = true, it dials out to sip call, when PSTN_CALL=true, it dials out to landline call, as I provide sample below; <condition field="${SIP_CALL}" expression="^true$"> <action application="log" data="INFO SIP CALL"/> <action application="bridge" data="sofia/${domain}/$1"/> </condition> <condition field="${PSTN_CALL}" expression="^true$"> <action application="log" data="INFO PSTN CALL"/> <action application="bridge" data="sofia/gateway/$1"/> </condition> The problem I'm facing is how can I apply condition when I've to call more than 1 variables? Like if there are no records in SIP numbering plan table and PSTN numbering plan table so it get the digits and dial out the to carrier (how to apply AND operation in condition?) i.e. <condition field="*${SIP_CALL} and ${PSTN_CALL}*" expression="^false$"> <action application="log" data="INFO GET DTMF"/> <action application="getdigits" data="values for getdigits"/> <action application="bridge" data="sofia/gateway/$1"/> </condition> Kindly advise for this issue. -- Regards, Ahmed Munir

2 1

FS hangup
by Lars Zeb 05 Nov '09

05 Nov '09

I just updated to v15372 from v15311. When calling into FreeSWITCH, it hangs up the call rather than going to voicemail (line 262 in pastebin). I don't know what might be causing this. Can anyone help? Thanks, Lars http://pastebin.freeswitch.org/11006 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux

2 4

Re: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media
by Humberto Quintana 05 Nov '09

05 Nov '09

>> -You could check the sofia debug for r15332 here: http://pastebin.freeswitch.org/11008 Phone/Devices: The caller is the DID provider's Switch. The callee (which also sends the REFER) is Asterisk 1.4.26. Testing with other devices(linksys SPA962,Grandstream GXV3000) gathers the same results. > I did not ask you to send me a ladder diagram. > I asked you to send me a console trace from FreeSWITCH using latest trunk > (1.0.4 does not help me) > > 1) start FreeSWITCH > 2) run the cli command: console loglevel debug > 3) run the cli command: sofia profile internal siptrace on > 4) reproduce your issue and put the trace on freeswitch pastebin > http://pastebin.freeswitch.org (login and pass are stated in the auth > dialog) > > Also please answer brian's question. What phones and/or sip devices are > involved in this call. > > > > On Wed, Nov 4, 2009 at 3:39 PM, Humberto Quintana wrote: > >> >> Thanks for your time, >> >> -The scenario is still the same: >> >> Always bypass media. >> Environment 100% NAT free :-) >> Call established from A to B through FS. Then... >> Blind transfer from B to C (Refer-to: C) >> RTP should go directly between A and C. >> >> >> -With 1.0.4 and 1.0.5pre3, FS actually INVITEs C after receiving the >> REFER-to:C, BUT there is no 2-way audio. Only RTP from C to A (due to the >> lack of reINVITE to A, after C answers). >> >> Please check SIP diagram here: >> >> http://provision.netcelerate.net/ngrep/blindxfer2009-11-04-v1.0.5pre3.html >> >> >> -What it's wrong with r15332 is there is not such call to C. For sure I >> know SIP is a protocol, may be my description was not clear but this SIP >> diagram speaks by itself ;-) >> >> http://provision.netcelerate.net/ngrep/blindxfer2009-11-04rev15332.html >> >> >> -You could check the sofia debug for r15332 here: >> http://pastebin.com/m6f2b3836 >> >> >> Best regards, >> >> Humberto >> >>> >>> I don't know what you are talking about anymore. >>> >>> The scenario I had tested is when a call is bridged in bypass_media=true >>> bridge >>> and you blind transfer that call back to the dialplan >>> >>> as soon as it hits the routing state it will resume media. >>> >>> >>> it has been confirmed to not work and confirmed to have been fixed >> several >>> time and if you are still having a problem you must have something >> blocking >>> some of your packets or something . >>> >>> You have to understand that sip is a protocol and your description is >>> completely non-standard. >>> Perhaps you should get a console trace and attach it to a jira. The trace >>> probably makes more sense to me. >>> >>> sofia profile internal siptrace on >>> console loglevel debug >>> >>> reproduce and attach the whole capture. >>> >>> >>> >>> On Tue, Nov 3, 2009 at 6:05 PM, Humberto Quintana wrote: >>> >>>> >>>> Hi, >>>> >>>> I tried r15332 and set in the sofia profile: >>>> >>>> a) bypass_media_after_bridge=true only >>>> b) bypass_media_after_bridge=true, param name="media-option" >>>> value="resume-media-on-hold"/> >>>> >>>> >>>> In both cases FS is hanging up the initial call (A to FS) after >> accepting >>>> the REFER to C: >>>> >>>> A <- reINVITE with FS' SDP <- FS >>>> A -> 200 -> FS >>>> A <- ACK <- FS >>>> A <- BYE <- FS >>>> >>>> The call to C is not even tried. >>>> >>>> I found this line is the logs that could give some idea: >>>> >>>> 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup >>>> sofia/external/514xxxxxx at a.b.c.d [CS_ROUTING] >> [RECOVERY_ON_TIMER_EXPIRE] >>>> after sending the ACK for the reINVITE >>>> >>>> >>>> Regards, >>>> >>>> >>>> Humberto >>>> >>>>>please try r15326 >>>>>I think i have it working. >>>>> >>>>>I recommend for optimal results you set bypass_media_after_bridge=true >>>>>either as a global or in your DP in place of bypass_media=true >>>>> >>>>> >>>>>On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana >>>> hotmail.com>wrote: >>>>> >>>>>> Hi Mike, >>>>>> >>>>>> I re-tried with trunk rev 15319 but I got almost the same behavior: >>>> There >>>>>> is now a reINVITE (with FS' SDP) going to A when the REFER is >> accepted. >>>> But >>>>>> still there is no reINVITE for A (with C's SDP) after the call from FS >>>> to C >>>>>> is established. >>>>>> >>>>>> Anyway, we decided for now to do a different implementation but if you >>>> want >>>>>> to explore more in this issue count me in ;-) >>>>>> >>>>>> >>>>>> Thank you very much! >>>>>> >>>>>> Humberto > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com _________________________________________________________________ Ready. Set. Get a great deal on Windows 7. See fantastic deals on Windows 7 now http://go.microsoft.com/?linkid=9691818

1 0

Java example
by dipen＠entvoice.com 05 Nov '09

05 Nov '09

Hi, Can you please paste me your sample java dialplan code that work for you ? ..coz m also facing the same problem. My mod_java is loaded properly. Also /usr/lib/jvm/java-1.5.0-gcj-4.3-1.5.0.0/jre/lib/i386/client/libjvm.so and freeswitch.jar in java.conf.xml is specified properly. I have written a java code to print HIIIIIIIIII on the console but its not printing. Level mentioned is INFO. on FS console it just prints EXECUTE sofia/internal/1004@192.168.1.144:5061 java(testing.class) I am attaching my java code herewith. Can u just tell me where more i should do the modification to get my dialplan work. Waiting for your kind reply. Thanks & Regards, Dipen Velani On Fri 19/12/08 4:09 AM , kriko <kristjan.ugrin(a)gmail.com> wrote: > Seems like my dialplan was a bit problematic, it works now. > Thanks. > On Thu, 18 Dec 2008 15:19:22 +0100, Anthony Minessale > wrote: > > did you turn up your console log level high enough to see it? The > default > > level is "INFO" > > > > > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > -- > kriko > _______________________________________________ > Freeswitch-users mailing list > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ---- Msg sent via @Mail - http://atmail.com/

3 4

Filtering a particular event.
by Nagalenoj H. 05 Nov '09

05 Nov '09

Hi, I've tried to filter the events like below to filter a particular event. 1) register for all events 2) filter for one unique-id 3) filter only one/more events(ex: DTMF & CHANNEL_EXECUTE) So, I want to receive only these events for the specific unique-id. But, I am receiving other events too. I'm using perl ESL outbound. Is it possible to do like this?! -- Regards, Nagalenoj H.

2 1

Skypiax load error
by 大泥人 05 Nov '09

05 Nov '09

Hi All, I once meet the Skypiax load error issue and some guys infomed me that there is no configuration file for Skypiax. When I follow these intructions -> http://wiki.freeswitch.org/wiki/Skypiax#Config_files_location_and_script_to… I still have some difficulties unstanding this: ." So, go and copy src\mod\endpoints\mod_skypiax\configs/skypiax.conf.xml to Debug\conf\autoload_configs." I did not find such directories in my freeswitch folder. Did not understand what "src" means, I checked the freeswitch folder and did not find such a folder named"src". There is a folder named"mod" under freeswitch while look into "mod" folder there are only some DLL files and can not find endpoints and etc. 2.You'll probably build the "Debug" version I just build this from the precompiled binaries and then launched FS. I'm not sure what I launched is in debug mode or not. If anyone can offer some help that really be appriciated. Thanks Daniel Zeng From: freeswitch-users-request(a)lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 41, Issue 40 To: freeswitch-users(a)lists.freeswitch.org Date: Thu, 5 Nov 2009 06:57:44 -0800 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..." --附转发的邮件-- From: anthony.minessale(a)gmail.com To: freeswitch-users(a)lists.freeswitch.org Date: Thu, 5 Nov 2009 08:13:46 -0600 Subject: Re: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media I did not ask you to send me a ladder diagram. I asked you to send me a console trace from FreeSWITCH using latest trunk (1.0.4 does not help me) 1) start FreeSWITCH 2) run the cli command: console loglevel debug 3) run the cli command: sofia profile internal siptrace on 4) reproduce your issue and put the trace on freeswitch pastebin http://pastebin.freeswitch.org (login and pass are stated in the auth dialog) Also please answer brian's question. What phones and/or sip devices are involved in this call. On Wed, Nov 4, 2009 at 3:39 PM, Humberto Quintana <hjqlopez(a)hotmail.com> wrote: Thanks for your time, -The scenario is still the same: Always bypass media. Environment 100% NAT free :-) Call established from A to B through FS. Then... Blind transfer from B to C (Refer-to: C) RTP should go directly between A and C. -With 1.0.4 and 1.0.5pre3, FS actually INVITEs C after receiving the REFER-to:C, BUT there is no 2-way audio. Only RTP from C to A (due to the lack of reINVITE to A, after C answers). Please check SIP diagram here: http://provision.netcelerate.net/ngrep/blindxfer2009-11-04-v1.0.5pre3.html -What it's wrong with r15332 is there is not such call to C. For sure I know SIP is a protocol, may be my description was not clear but this SIP diagram speaks by itself ;-) http://provision.netcelerate.net/ngrep/blindxfer2009-11-04rev15332.html -You could check the sofia debug for r15332 here: http://pastebin.com/m6f2b3836 Best regards, Humberto > > I don't know what you are talking about anymore. > > The scenario I had tested is when a call is bridged in bypass_media=true > bridge > and you blind transfer that call back to the dialplan > > as soon as it hits the routing state it will resume media. > > > it has been confirmed to not work and confirmed to have been fixed several > time and if you are still having a problem you must have something blocking > some of your packets or something . > > You have to understand that sip is a protocol and your description is > completely non-standard. > Perhaps you should get a console trace and attach it to a jira. The trace > probably makes more sense to me. > > sofia profile internal siptrace on > console loglevel debug > > reproduce and attach the whole capture. > > > > On Tue, Nov 3, 2009 at 6:05 PM, Humberto Quintana wrote: > >> >> Hi, >> >> I tried r15332 and set in the sofia profile: >> >> a) bypass_media_after_bridge=true only >> b) bypass_media_after_bridge=true, param name="media-option" >> value="resume-media-on-hold"/> >> >> >> In both cases FS is hanging up the initial call (A to FS) after accepting >> the REFER to C: >> >> A <- reINVITE with FS' SDP <- FS >> A -> 200 -> FS >> A <- ACK <- FS >> A <- BYE <- FS >> >> The call to C is not even tried. >> >> I found this line is the logs that could give some idea: >> >> 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup >> sofia/external/514xxxxxx at a.b.c.d [CS_ROUTING] [RECOVERY_ON_TIMER_EXPIRE] >> after sending the ACK for the reINVITE >> >> >> Regards, >> >> >> Humberto >> >>>please try r15326 >>>I think i have it working. >>> >>>I recommend for optimal results you set bypass_media_after_bridge=true >>>either as a global or in your DP in place of bypass_media=true >>> >>> >>>On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana >> hotmail.com>wrote: >>> >>>> Hi Mike, >>>> >>>> I re-tried with trunk rev 15319 but I got almost the same behavior: >> There >>>> is now a reINVITE (with FS' SDP) going to A when the REFER is accepted. >> But >>>> still there is no reINVITE for A (with C's SDP) after the call from FS >> to C >>>> is established. >>>> >>>> Anyway, we decided for now to do a different implementation but if you >> want >>>> to explore more in this issue count me in ;-) >>>> >>>> >>>> Thank you very much! >>>> >>>> Humberto >> >> >> _________________________________________________________________ >> Windows Live: Friends get your Flickr, Yelp, and Digg updates when they >> e-mail you. >> http://go.microsoft.com/?linkid=9691817 >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > _________________________________________________________________ > Ready. Set. Get a great deal on Windows 7. See fantastic deals on Windows 7 now > http://go.microsoft.com/?linkid=9691818 _________________________________________________________________ Windows Live: Make it easier for your friends to see what you’re up to on Facebook. http://go.microsoft.com/?linkid=9691816 _______________________________________________ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 --附转发的邮件-- From: rob4manhere(a)gmail.com To: freeswitch-users(a)lists.freeswitch.org Date: Thu, 5 Nov 2009 08:52:05 -0600 Subject: Re: [Freeswitch-users] Setting up Conference with Moderator Hi UK, From what I've done and read, the caller-controls (in conference.conf.xml) can be modified to almost anything you can think of, BUT, they are mapped 1-to-1 to a conference- ie you can't map a caller control just for those with the moderator flag. So unless you want everyone able to mute/kick everyone then you can't do it. The wiki seems to indicate this as well: "Be aware that the caller-controls are applied across the entire conference. You cannot enter one member of the conference using caller- controls ABC and then enter a second member using caller-controls XYZ." http://wiki.freeswitch.org/wiki/Mod_conference I think this might be a limitation of mod_conference. Perhaps one of the pros can chime in if I'm off-base or there's some nifty way to accomplish this. Cheers, Rob On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote: > Any ideas on the below...has anyone implemented the below: > > Once I have the Moderator and Participants logged on, how do I > invoke the moderator previlidges, LIk esay muting everyone/someone > or kicking someone out of the Conf and the like? > > > > -----Original Message----- > From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org > ] On Behalf Of Ujjval Karihaloo > Sent: Monday, November 02, 2009 12:52 PM > To: freeswitch-users(a)lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > Rob: > > Once I have the Moderator and Participants logged on, how do I > invoke the moderator previlidges, LIk esay muting everyone/someone > or kicking someone out of the Conf and the like? > > > > -----Original Message----- > From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org > ] On Behalf Of Rob Forman > Sent: Friday, October 30, 2009 9:34 AM > To: freeswitch-users(a)lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > Hm, strange. I haven't seen that before. Can you pastebin your logs > at debug level? > > On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: > >> It's strange... a tcpdump tells me that there is no DTMF from my >> provider when using IVR, but when I call into a TN that goes >> directly into the Conference App, I see DTMF from the provider. >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org >> ] On Behalf Of Rob Forman >> Sent: Friday, October 30, 2009 7:23 AM >> To: freeswitch-users(a)lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> I've never had any problem with that. Is your logging at debug level >> so you can see the RECV DTFM in the log/fs_cli? Are you calling from >> a SIP phone on the pbx, or via a PSTN provider? Maybe your provider >> isn't passing them through. >> >> Make sure your logging is turned up then try something simpler, like >> calling the echo application, and see if DTFM comes through. >> >> Rob >> >> On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote: >> >>> Rob: >>> >>> For some reason, I don't see the DTMF appear on the fs_CLI when >>> using the below configuration....so it basically timesout. >>> >>> UK >>> >>> >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org >>> ] On Behalf Of Ujjval Karihaloo >>> Sent: Monday, October 26, 2009 9:21 AM >>> To: freeswitch-users(a)lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> Thx a lot Rob, reading the wiki your way or using IVR seems >>> correct.. >>> =============== >>> The wiki also says that the wait-mod might be "used in conjunction >>> with an IVR where the moderators are authenticated with an extra >>> pass- >>> code", which is what I did. I guess that's why I didn't understand >>> the point of the +pin. >>> ====================== >>> >>> I will try it out. >>> >>> Again thx a lot for your help. Will keep everyone posted. >>> >>> ________________________________________ >>> From: freeswitch-users-bounces(a)lists.freeswitch.org [freeswitch-users-bounces(a)lists.freeswitch.org >>> ] On Behalf Of Rob Forman [rob4manhere(a)gmail.com] >>> Sent: Friday, October 23, 2009 12:22 PM >>> To: freeswitch-users(a)lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> I just re-tested with the pin in my dial plan: >>> >>> <action application="conference" data="conference 123456@default >>> +flags{}+1234" /> >>> >>> And it doesn't challenge me for the pin. I just drop right in. I >>> figured this is how it was intended, since the wiki says the pin is >>> set initially and only challenged in later attempts [by future >>> callers]: >>> >>> "The first time a conference name (confname) is used, it will be >>> created on demand, and the pin will be set to what ever is specified >>> at that time: the pin in the data string if specified, or if not, >>> the >>> "pin" setting in the conference profile, and if that is also >>> unspecified, then there is no pin protection. Any later attempt to >>> join the conference must specify the same pin number, if one existed >>> when it was created. " >>> >>> >>> The wiki also says that the wait-mod might be "used in conjunction >>> with an IVR where the moderators are authenticated with an extra >>> pass- >>> code", which is what I did. I guess that's why I didn't understand >>> the point of the +pin. >>> >>> I'm sure there's a scenario where its used and useful, the wiki just >>> doesn't explain it. >>> >>> Rob >>> >>> On Oct 23, 2009, at 12:43 PM, Brian West wrote: >>> >>>> Well first off you're not defining a pine here... >>>> >>>> confname@profilename+flags{mute|deaf|waste|moderator}+[conference >>>> pin >>>> number] >>>> >>>> That might be why its not asking for a pin. >>>> >>>> /b >>>> >>>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote: >>>> >>>>> <entry action="menu-exec-app" digits="1" param="conference >>>>> 123456@default+flags{}" /> >>>> >>>> >>>> _______________________________________________ >>>> 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 >>> >>> >>> _______________________________________________ >>> 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 >>> >>> _______________________________________________ >>> 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 >>> >>> _______________________________________________ >>> 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 >> >> >> _______________________________________________ >> 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 >> >> _______________________________________________ >> 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 > > > _______________________________________________ > 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 > > _______________________________________________ > 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 > > _______________________________________________ > 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 --附转发的邮件-- From: rob4manhere(a)gmail.com To: freeswitch-users(a)lists.freeswitch.org Date: Thu, 5 Nov 2009 08:57:08 -0600 Subject: [Freeswitch-users] Wideband / HD phones Hey all, Looking at buying some high def phones. Any recommendations (preferably based on experience) for hardware based on product quality, standards compliance, features integration with Freeswitch, etc? Thank you! Rob Forman _________________________________________________________________ 约会说不清地方？来试试微软地图最新msn互动功能！ http://ditu.live.com/?form=TL&swm=1

3 2

Newbie trying to setup Cisco 7940 phones
by mkitchin.public＠gmail.com 05 Nov '09

05 Nov '09

I'm working on an alternative to a $120,000 Cisco phone system that my company is looking at. I got Freeswitch installed on CentOS last week using the Quick and Dirty instructions. That part was painless. We had a few 7940s laying around. After some wrestling with it, I got the latest SIP firmware installed and what I hoped was a functional config (attached). X-Lite phones can call each other no problem. 7940s can call X-Lite no problem. Anytime I try and call a 7940, it goes straight to voicemail. I attached a log file that shows the activity when trying to call a7940 from X-Lite. X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is nshplpbx1.unix/10.85.0.53. Everything is on the same LAN. Different subnets, but no firewalls. I didn't see anything that said posting attachments was frowned upon. I apologize if it isn't appropriate. I'm guessing this is something simple and I'm just clueless on how to diagnose the issue. I'm not tied to using this model for good, but it is what we had laying around. Any help would be greatly appreciated. Next step is configuring it to talk to Verizon VOIP over a DS3. Thanks, Matthew Kitchin dsi> sh conf ------ Current *FLASH* Configuration ------ Platform : Cisco Systems, Inc. IP Phone CP-7940G Elapsed Time: 01:01:06 dhcp_server : Disabled my_ip_addr : 10.86.11.50 subnet_mask : 255.255.0.0 defaultgw : 10.86.0.1 dyn_dns_addr_1 : 0.0.0.0 dyn_dns_addr_2 : 0.0.0.0 dns_addr : 10.85.0.11 dns_backup_1: 10.85.0.10 primary_tftp_addr : 10.86.10.58 dyn_tftp_addr : 0.0.0.0 my_mac_addr : 0012:7f98:eaa9 domain_name : dsi-corp.net my_name : SIP00127F98EAA9 Status Flags : 12300001 image_version : "P003-8-12-00" FirmLoadID : "PC030301" DSPLoadID : "PS03AT38" network_media_type : Auto network_port2_type : Hub/Switch dscpForAudio : 184 phone_label : "Matthew Kitchin" tftp_cfg_dir : "" phone_password : ********** phone_prompt : "dsi" language : english sntp_mode : Unicast sntp_server : 10.85.0.10 time_zone : CST dst_offset : 01/00 dst_start_month : March dst_start_day : 0 dst_start_day_of_week : Sunday dst_start_week_of_month : 8 dst_start_time : 02/00 dst_stop_month : October dst_stop_day : 0 dst_stop_day_of_week : Sunday dst_stop_week_of_month : 8 dst_stop_time : 02/00 dst_auto_adjust : 1 time_format_24hr : 1 date_format : M/D/Y nat_enable : 0 nat_address : UNPROVISIONED voip_control_port : 5060 start_media_port : 16384 end_media_port : 32766 sync : "1" xml_card_dir : "" xml_card_file : "CARD.XML" telnet_level : 1 services_url : "" directory_url : "" logo_url : "" http_proxy_addr : UNPROVISIONED http_proxy_port : 80 garp_enable : 0 enable_vad : 0 dial_template : "dialplan" callerid_blocking : 0 anonymous_call_block : 0 autocomplete : 0 messages_uri : "" dnd_control : 2 preferred_codec : g711ulaw dtmf_outofband : avt_always dtmf_avt_payload : 101 dtmf_db_level : 3 dtmf_inband : 0 call_manager1_addr : "UNPROVISIONED" call_manager2_addr : "UNPROVISIONED" call_manager3_addr : "UNPROVISIONED" call_manager1_sip_port : 5060 call_manager2_sip_port : 5060 call_manager3_sip_port : 5060 call_manager5_addr : "UNPROVISIONED" call_manager5_sip_port : 5060 call_manager4_addr : "UNPROVISIONED" call_manager4_sip_port : 0 line1_name : "1008" line2_name : "1001" line1_authname : "1008" line2_authname : "1001" line1_password : ********** line2_password : ********** line1_shortname : "1008" line2_shortname : "1001" line1_displayname : "1008" line2_displayname : "UNPROVISIONED" line1_contact : "UNPROVISIONED" line2_contact : "UNPROVISIONED" proxy1_address : "nshplpbx1.unix" proxy2_address : "nshplpbx1.unix" proxy1_port : 5060 proxy2_port : 5060 sip_retx : 10 sip_invite_retx : 6 timer_t1 : 500 timer_t2 : 4000 timer_invite_expires : 180 timer_register_expires : 3600 proxy_register : 1 proxy_backup : "" proxy_emergency : "" proxy_backup_port : 5060 proxy_emergency_port : 5060 outbound_proxy : UNPROVISIONED outbound_proxy_port : 5060 nat_received_processing : 0 mwi_status : 0 call_waiting : 1 user_info : phone cnf_join_enable : 0 remote_party_id : 1 semi_attended_transfer : 1 transfer_onhook_enabled : 0 call_hold_ringback : 3 stutter_msg_waiting : 0 cfwd_url : "" call_stats : 0 auto_answer : 0 local_cfwd_enable : 1 timer_register_delta : 5 sip_max_forwards : 70 rfc_2543_hold : 0 version_stamp : "" timer_keepalive_expires : 120 connection_monitor_duration : 120 encrypt_key : ********** 2009-11-03 15:39:50.762957 [DEBUG] sofia.c:5067 0 acls to check for proxy 2009-11-03 15:39:50.762957 [DEBUG] sofia.c:5085 network ip is a proxy [0] 2009-11-03 15:39:50.762957 [DEBUG] sofia.c:5113 IP 10.86.10.58 Rejected by acl "domains". Falling back to Digest auth. 2009-11-03 15:39:50.780028 [DEBUG] sofia.c:5067 0 acls to check for proxy 2009-11-03 15:39:50.780028 [DEBUG] sofia.c:5085 network ip is a proxy [0] 2009-11-03 15:39:50.780028 [DEBUG] sofia.c:5113 IP 10.86.10.58 Rejected by acl "domains". Falling back to Digest auth. 2009-11-03 15:39:50.797901 [NOTICE] switch_channel.c:613 New Channel sofia/internal/1000(a)nshplpbx1.unix [8c133ad4-67cf-4ffa-8655-56ffa0e3933d] 2009-11-03 15:39:50.799061 [DEBUG] sofia.c:5812 Setting NAT mode based on nat.auto 2009-11-03 15:39:50.799061 [DEBUG] sofia.c:3588 Channel sofia/internal/1000(a)nshplpbx1.unix entering state [received][100] 2009-11-03 15:39:50.799061 [DEBUG] sofia.c:3599 Remote SDP: v=0 o=- 8 2 IN IP4 10.86.10.58 s=CounterPath X-Lite 3.0 c=IN IP4 10.86.10.58 t=0 0 m=audio 37250 RTP/AVP 107 0 8 101 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=alt:1 1 : cwKnUvoi NsQWtT4P 10.86.10.58 37250 2009-11-03 15:39:50.799061 [DEBUG] sofia_glue.c:3160 Audio Codec Compare [BV32:107:16000:0]/[G7221:115:32000:20] 2009-11-03 15:39:50.799061 [DEBUG] sofia_glue.c:3160 Audio Codec Compare [BV32:107:16000:0]/[G7221:107:16000:20] 2009-11-03 15:39:50.799061 [DEBUG] sofia_glue.c:3160 Audio Codec Compare [BV32:107:16000:0]/[G722:9:8000:20] 2009-11-03 15:39:50.799061 [DEBUG] sofia_glue.c:3160 Audio Codec Compare [BV32:107:16000:0]/[PCMU:0:8000:20] 2009-11-03 15:39:50.799061 [DEBUG] sofia_glue.c:3160 Audio Codec Compare [BV32:107:16000:0]/[PCMA:8:8000:20] 2009-11-03 15:39:50.799061 [DEBUG] sofia_glue.c:3160 Audio Codec Compare [BV32:107:16000:0]/[GSM:3:8000:20] 2009-11-03 15:39:50.799061 [DEBUG] sofia_glue.c:3160 Audio Codec Compare [PCMU:0:8000:0]/[G7221:115:32000:20] 2009-11-03 15:39:50.799061 [DEBUG] sofia_glue.c:3160 Audio Codec Compare [PCMU:0:8000:0]/[G7221:107:16000:20] 2009-11-03 15:39:50.799061 [DEBUG] sofia_glue.c:3160 Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20] 2009-11-03 15:39:50.799061 [DEBUG] sofia_glue.c:3160 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-11-03 15:39:50.799061 [DEBUG] sofia_glue.c:2102 Set Codec sofia/internal/1000(a)nshplpbx1.unix PCMU/8000 20 ms 160 samples 2009-11-03 15:39:50.799061 [DEBUG] sofia_glue.c:3120 Set 2833 dtmf payload to 101 2009-11-03 15:39:50.799061 [DEBUG] sofia.c:3744 (sofia/internal/1000(a)nshplpbx1.unix) State Change CS_NEW -> CS_INIT 2009-11-03 15:39:50.799061 [DEBUG] switch_core_session.c:1003 Send signal sofia/internal/1000(a)nshplpbx1.unix [BREAK] 2009-11-03 15:39:50.799061 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/1000(a)nshplpbx1.unix) Running State Change CS_INIT 2009-11-03 15:39:50.799061 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/1000(a)nshplpbx1.unix) State INIT 2009-11-03 15:39:50.799061 [DEBUG] mod_sofia.c:83 sofia/internal/1000(a)nshplpbx1.unix SOFIA INIT 2009-11-03 15:39:50.799061 [DEBUG] mod_sofia.c:111 (sofia/internal/1000(a)nshplpbx1.unix) State Change CS_INIT -> CS_ROUTING 2009-11-03 15:39:50.799061 [DEBUG] switch_core_session.c:1003 Send signal sofia/internal/1000(a)nshplpbx1.unix [BREAK] 2009-11-03 15:39:50.800189 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/1000(a)nshplpbx1.unix) State INIT going to sleep 2009-11-03 15:39:50.800189 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/1000(a)nshplpbx1.unix) Running State Change CS_ROUTING 2009-11-03 15:39:50.800189 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/1000(a)nshplpbx1.unix) State ROUTING 2009-11-03 15:39:50.800189 [DEBUG] mod_sofia.c:130 sofia/internal/1000(a)nshplpbx1.unix SOFIA ROUTING 2009-11-03 15:39:50.800189 [DEBUG] switch_core_state_machine.c:78 sofia/internal/1000(a)nshplpbx1.unix Standard ROUTING 2009-11-03 15:39:50.800189 [INFO] mod_dialplan_xml.c:397 Processing 1000->1008 in context default Dialplan: sofia/internal/1000(a)nshplpbx1.unix parsing [default->unloop] continue=false Dialplan: sofia/internal/1000(a)nshplpbx1.unix Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1000(a)nshplpbx1.unix Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/1000(a)nshplpbx1.unix parsing [default->tod_example] continue=true Dialplan: day of week[3] =~ 2-6 (PASS) Dialplan: hour[15] =~ 9-18 (PASS) Dialplan: sofia/internal/1000(a)nshplpbx1.unix Date/Time Match (PASS) [tod_example] break=on-false Dialplan: sofia/internal/1000(a)nshplpbx1.unix Action set(open=true) Dialplan: sofia/internal/1000(a)nshplpbx1.unix parsing [default->global-intercept] continue=false Dialplan: sofia/internal/1000(a)nshplpbx1.unix Regex (FAIL) [global-intercept] destination_number(1008) =~ /^886$/ break=on-false Dialplan: sofia/internal/1000(a)nshplpbx1.unix parsing [default->group-intercept] continue=false Dialplan: sofia/internal/1000(a)nshplpbx1.unix Regex (FAIL) [group-intercept] destination_number(1008) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/1000(a)nshplpbx1.unix parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/1000(a)nshplpbx1.unix Regex (FAIL) [intercept-ext] destination_number(1008) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/1000(a)nshplpbx1.unix parsing [default->redial] continue=false Dialplan: sofia/internal/1000(a)nshplpbx1.unix Regex (FAIL) [redial] destination_number(1008) =~ /^870$/ break=on-false Dialplan: sofia/internal/1000(a)nshplpbx1.unix parsing [default->global] continue=true Dialplan: sofia/internal/1000(a)nshplpbx1.unix Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/1000(a)nshplpbx1.unix Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: sofia/internal/1000(a)nshplpbx1.unix Absolute Condition [global] Dialplan: sofia/internal/1000(a)nshplpbx1.unix Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/1000(a)nshplpbx1.unix Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/1000(a)nshplpbx1.unix Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/1000(a)nshplpbx1.unix parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/1000(a)nshplpbx1.unix Regex (FAIL) [snom-demo-2] destination_number(1008) =~ /^9001$/ break=on-false Dialplan: sofia/internal/1000(a)nshplpbx1.unix parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/1000(a)nshplpbx1.unix Regex (FAIL) [snom-demo-1] destination_number(1008) =~ /^9000$/ break=on-false Dialplan: sofia/internal/1000(a)nshplpbx1.unix parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1000(a)nshplpbx1.unix Regex (FAIL) [eavesdrop] destination_number(1008) =~ /^88(.*)$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/1000(a)nshplpbx1.unix parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1000(a)nshplpbx1.unix Regex (FAIL) [eavesdrop] destination_number(1008) =~ /^779$/ break=on-false Dialplan: sofia/internal/1000(a)nshplpbx1.unix parsing [default->call_return] continue=false Dialplan: sofia/internal/1000(a)nshplpbx1.unix Regex (FAIL) [call_return] destination_number(1008) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/1000(a)nshplpbx1.unix parsing [default->del-group] continue=false Dialplan: sofia/internal/1000(a)nshplpbx1.unix Regex (FAIL) [del-group] destination_number(1008) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/1000(a)nshplpbx1.unix parsing [default->add-group] continue=false Dialplan: sofia/internal/1000(a)nshplpbx1.unix Regex (FAIL) [add-group] destination_number(1008) =~ /^81(\d{2})$/ break=on-false Dialplan: sofia/internal/1000(a)nshplpbx1.unix parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/1000(a)nshplpbx1.unix Regex (FAIL) [call-group-simo] destination_number(1008) =~ /^82(\d{2})$/ break=on-false Dialplan: sofia/internal/1000(a)nshplpbx1.unix parsing [default->call-group-order] continue=false Dialplan: sofia/internal/1000(a)nshplpbx1.unix Regex (FAIL) [call-group-order] destination_number(1008) =~ /^83(\d{2})$/ break=on-false Dialplan: sofia/internal/1000(a)nshplpbx1.unix parsing [default->extension-intercom] continue=false Dialplan: sofia/internal/1000(a)nshplpbx1.unix Regex (FAIL) [extension-intercom] destination_number(1008) =~ /^8(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1000(a)nshplpbx1.unix parsing [default->Local_Extension] continue=false Dialplan: sofia/internal/1000(a)nshplpbx1.unix Regex (PASS) [Local_Extension] destination_number(1008) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1000(a)nshplpbx1.unix Action set(dialed_extension=1008) Dialplan: sofia/internal/1000(a)nshplpbx1.unix Action export(dialed_extension=1008) Dialplan: sofia/internal/1000(a)nshplpbx1.unix Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/internal/1000(a)nshplpbx1.unix Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav) Dialplan: sofia/internal/1000(a)nshplpbx1.unix Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: sofia/internal/1000(a)nshplpbx1.unix Action set(ringback=${us-ring}) Dialplan: sofia/internal/1000(a)nshplpbx1.unix Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/internal/1000(a)nshplpbx1.unix Action set(call_timeout=30) Dialplan: sofia/internal/1000(a)nshplpbx1.unix Action set(hangup_after_bridge=true) Dialplan: sofia/internal/1000(a)nshplpbx1.unix Action set(continue_on_fail=true) Dialplan: sofia/internal/1000(a)nshplpbx1.unix Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) Dialplan: sofia/internal/1000(a)nshplpbx1.unix Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: sofia/internal/1000(a)nshplpbx1.unix Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: sofia/internal/1000(a)nshplpbx1.unix Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/1000(a)nshplpbx1.unix Action bridge(user/${dialed_extension}@${domain_name}) Dialplan: sofia/internal/1000(a)nshplpbx1.unix Action answer() Dialplan: sofia/internal/1000(a)nshplpbx1.unix Action sleep(1000) Dialplan: sofia/internal/1000(a)nshplpbx1.unix Action voicemail(default ${domain_name} ${dialed_extension}) 2009-11-03 15:39:50.801914 [DEBUG] switch_core_state_machine.c:114 (sofia/internal/1000(a)nshplpbx1.unix) State Change CS_ROUTING -> CS_EXECUTE 2009-11-03 15:39:50.801914 [DEBUG] switch_core_session.c:1003 Send signal sofia/internal/1000(a)nshplpbx1.unix [BREAK] 2009-11-03 15:39:50.801914 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/1000(a)nshplpbx1.unix) State ROUTING going to sleep 2009-11-03 15:39:50.801914 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/1000(a)nshplpbx1.unix) Running State Change CS_EXECUTE 2009-11-03 15:39:50.801914 [DEBUG] switch_core_state_machine.c:340 (sofia/internal/1000(a)nshplpbx1.unix) State EXECUTE 2009-11-03 15:39:50.801914 [DEBUG] mod_sofia.c:173 sofia/internal/1000(a)nshplpbx1.unix SOFIA EXECUTE 2009-11-03 15:39:50.801914 [DEBUG] switch_core_state_machine.c:151 sofia/internal/1000(a)nshplpbx1.unix Standard EXECUTE EXECUTE sofia/internal/1000(a)nshplpbx1.unix set(open=true) 2009-11-03 15:39:50.803724 [DEBUG] mod_dptools.c:763 sofia/internal/1000(a)nshplpbx1.unix SET [open]=[true] EXECUTE sofia/internal/1000(a)nshplpbx1.unix hash(insert/10.85.0.53-spymap/1000/8c133ad4-67cf-4ffa-8655-56ffa0e3933d) EXECUTE sofia/internal/1000(a)nshplpbx1.unix hash(insert/10.85.0.53-last_dial/1000/1008) EXECUTE sofia/internal/1000(a)nshplpbx1.unix hash(insert/10.85.0.53-last_dial/global/8c133ad4-67cf-4ffa-8655-56ffa0e3933d) EXECUTE sofia/internal/1000(a)nshplpbx1.unix set(dialed_extension=1008) 2009-11-03 15:39:50.803724 [DEBUG] mod_dptools.c:763 sofia/internal/1000(a)nshplpbx1.unix SET [dialed_extension]=[1008] EXECUTE sofia/internal/1000(a)nshplpbx1.unix export(dialed_extension=1008) 2009-11-03 15:39:50.803724 [DEBUG] mod_dptools.c:846 EXPORT [dialed_extension]=[1008] EXECUTE sofia/internal/1000(a)nshplpbx1.unix bind_meta_app(1 b s execute_extension::dx XML features) 2009-11-03 15:39:50.804911 [INFO] switch_ivr_async.c:2151 Bound B-Leg: 1 execute_extension::dx XML features EXECUTE sofia/internal/1000(a)nshplpbx1.unix bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/1000.2009-11-03-15-39-50.wav) 2009-11-03 15:39:50.804911 [INFO] switch_ivr_async.c:2151 Bound B-Leg: 2 record_session::/usr/local/freeswitch/recordings/1000.2009-11-03-15-39-50.wav EXECUTE sofia/internal/1000(a)nshplpbx1.unix bind_meta_app(3 b s execute_extension::cf XML features) 2009-11-03 15:39:50.804911 [INFO] switch_ivr_async.c:2151 Bound B-Leg: 3 execute_extension::cf XML features EXECUTE sofia/internal/1000(a)nshplpbx1.unix set(ringback=%(2000,4000,440.0,480.0)) 2009-11-03 15:39:50.804911 [DEBUG] mod_dptools.c:763 sofia/internal/1000(a)nshplpbx1.unix SET [ringback]=[%(2000,4000,440.0,480.0)] EXECUTE sofia/internal/1000(a)nshplpbx1.unix set(transfer_ringback=local_stream://moh) 2009-11-03 15:39:50.804911 [DEBUG] mod_dptools.c:763 sofia/internal/1000(a)nshplpbx1.unix SET [transfer_ringback]=[local_stream://moh] EXECUTE sofia/internal/1000(a)nshplpbx1.unix set(call_timeout=30) 2009-11-03 15:39:50.804911 [DEBUG] mod_dptools.c:763 sofia/internal/1000(a)nshplpbx1.unix SET [call_timeout]=[30] EXECUTE sofia/internal/1000(a)nshplpbx1.unix set(hangup_after_bridge=true) 2009-11-03 15:39:50.806131 [DEBUG] mod_dptools.c:763 sofia/internal/1000(a)nshplpbx1.unix SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/1000(a)nshplpbx1.unix set(continue_on_fail=true) 2009-11-03 15:39:50.806131 [DEBUG] mod_dptools.c:763 sofia/internal/1000(a)nshplpbx1.unix SET [continue_on_fail]=[true] EXECUTE sofia/internal/1000(a)nshplpbx1.unix hash(insert/10.85.0.53-call_return/1008/1000) EXECUTE sofia/internal/1000(a)nshplpbx1.unix hash(insert/10.85.0.53-last_dial_ext/1008/8c133ad4-67cf-4ffa-8655-56ffa0e3933d) EXECUTE sofia/internal/1000(a)nshplpbx1.unix set(called_party_callgroup=techsupport) 2009-11-03 15:39:50.806131 [DEBUG] mod_dptools.c:763 sofia/internal/1000(a)nshplpbx1.unix SET [called_party_callgroup]=[techsupport] EXECUTE sofia/internal/1000(a)nshplpbx1.unix hash(insert/10.85.0.53-last_dial/techsupport/8c133ad4-67cf-4ffa-8655-56ffa0e3933d) EXECUTE sofia/internal/1000(a)nshplpbx1.unix bridge(user/1008(a)10.85.0.53) 2009-11-03 15:39:50.812162 [DEBUG] switch_ivr_originate.c:1333 variable string 0 = [presence_id=1008(a)10.85.0.53] 2009-11-03 15:39:50.812162 [NOTICE] switch_channel.c:613 New Channel sofia/internal/sip:1008@10.86.11.50:5060 [32661f42-b0a9-4179-867d-156b46ae71ce] 2009-11-03 15:39:50.812162 [DEBUG] mod_sofia.c:3061 (sofia/internal/sip:1008@10.86.11.50:5060) State Change CS_NEW -> CS_INIT 2009-11-03 15:39:50.812162 [DEBUG] switch_core_session.c:1003 Send signal sofia/internal/sip:1008@10.86.11.50:5060 [BREAK] 2009-11-03 15:39:50.812162 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/sip:1008@10.86.11.50:5060) Running State Change CS_INIT 2009-11-03 15:39:50.812162 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/sip:1008@10.86.11.50:5060) State INIT 2009-11-03 15:39:50.812162 [DEBUG] mod_sofia.c:83 sofia/internal/sip:1008@10.86.11.50:5060 SOFIA INIT 2009-11-03 15:39:50.812162 [DEBUG] sofia_glue.c:1761 sip:1008@10.86.11.50:51061;user=phone;transport=udp Setting proxy route to sofia/internal/sip:1008@10.86.11.50:5060 2009-11-03 15:39:50.814226 [DEBUG] mod_sofia.c:111 (sofia/internal/sip:1008@10.86.11.50:5060) State Change CS_INIT -> CS_ROUTING 2009-11-03 15:39:50.814226 [DEBUG] switch_core_session.c:1003 Send signal sofia/internal/sip:1008@10.86.11.50:5060 [BREAK] 2009-11-03 15:39:50.814226 [DEBUG] sofia.c:3588 Channel sofia/internal/sip:1008@10.86.11.50:5060 entering state [calling][0] 2009-11-03 15:39:50.814226 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/sip:1008@10.86.11.50:5060) State INIT going to sleep 2009-11-03 15:39:50.814226 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/sip:1008@10.86.11.50:5060) Running State Change CS_ROUTING 2009-11-03 15:39:50.814226 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/sip:1008@10.86.11.50:5060) State ROUTING 2009-11-03 15:39:50.814226 [DEBUG] mod_sofia.c:130 sofia/internal/sip:1008@10.86.11.50:5060 SOFIA ROUTING 2009-11-03 15:39:50.814226 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/sip:1008@10.86.11.50:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-11-03 15:39:50.814226 [DEBUG] switch_core_session.c:1003 Send signal sofia/internal/sip:1008@10.86.11.50:5060 [BREAK] 2009-11-03 15:39:50.814226 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/sip:1008@10.86.11.50:5060) State ROUTING going to sleep 2009-11-03 15:39:50.814226 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/sip:1008@10.86.11.50:5060) Running State Change CS_CONSUME_MEDIA 2009-11-03 15:39:50.814226 [DEBUG] switch_core_state_machine.c:352 (sofia/internal/sip:1008@10.86.11.50:5060) State CONSUME_MEDIA 2009-11-03 15:39:50.814226 [DEBUG] switch_core_state_machine.c:352 (sofia/internal/sip:1008@10.86.11.50:5060) State CONSUME_MEDIA going to sleep 2009-11-03 15:39:50.817484 [DEBUG] sofia.c:3588 Channel sofia/internal/sip:1008@10.86.11.50:5060 entering state [terminated][503] 2009-11-03 15:39:50.817484 [NOTICE] sofia.c:4187 Hangup sofia/internal/sip:1008@10.86.11.50:5060 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2009-11-03 15:39:50.817484 [DEBUG] switch_channel.c:1896 Send signal sofia/internal/sip:1008@10.86.11.50:5060 [KILL] 2009-11-03 15:39:50.817484 [DEBUG] switch_core_session.c:1003 Send signal sofia/internal/sip:1008@10.86.11.50:5060 [BREAK] 2009-11-03 15:39:50.817484 [DEBUG] switch_core_state_machine.c:451 thread mismatch skipping state handler. 2009-11-03 15:39:50.817484 [DEBUG] switch_ivr_originate.c:2543 Originate Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE] 2009-11-03 15:39:50.817484 [ERR] switch_ivr_originate.c:1829 Cannot create outgoing channel of type [user] cause: [NORMAL_TEMPORARY_FAILURE] 2009-11-03 15:39:50.817484 [DEBUG] switch_ivr_originate.c:2543 Originate Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE] 2009-11-03 15:39:50.817484 [INFO] mod_dptools.c:2290 Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2009-11-03 15:39:50.817484 [DEBUG] mod_dptools.c:2312 Continue on fail [true]: Cause: NORMAL_TEMPORARY_FAILURE EXECUTE sofia/internal/1000(a)nshplpbx1.unix answer() 2009-11-03 15:39:50.817484 [DEBUG] mod_dptools.c:653 sofia/internal/1000(a)nshplpbx1.unix receive message [ANSWER] 2009-11-03 15:39:50.817484 [DEBUG] sofia_glue.c:2336 AUDIO RTP [sofia/internal/1000(a)nshplpbx1.unix] 10.85.0.53 port 24712 -> 10.86.10.58 port 37250 codec: 0 ms: 20 2009-11-03 15:39:50.817484 [DEBUG] switch_rtp.c:1155 Starting timer [soft] 160 bytes per 20ms 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/sip:1008@10.86.11.50:5060) Running State Change CS_HANGUP 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/sip:1008@10.86.11.50:5060) State HANGUP 2009-11-03 15:39:50.818924 [DEBUG] mod_sofia.c:329 sofia/internal/sip:1008@10.86.11.50:5060 Overriding SIP cause 503 with 503 from the other leg 2009-11-03 15:39:50.818924 [DEBUG] mod_sofia.c:361 Channel sofia/internal/sip:1008@10.86.11.50:5060 hanging up, cause: NORMAL_TEMPORARY_FAILURE 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:46 sofia/internal/sip:1008@10.86.11.50:5060 Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/sip:1008@10.86.11.50:5060) State HANGUP going to sleep 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/sip:1008@10.86.11.50:5060) State Change CS_HANGUP -> CS_REPORTING 2009-11-03 15:39:50.818924 [DEBUG] switch_core_session.c:1003 Send signal sofia/internal/sip:1008@10.86.11.50:5060 [BREAK] 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/sip:1008@10.86.11.50:5060) Running State Change CS_REPORTING 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/sip:1008@10.86.11.50:5060) State REPORTING 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:53 sofia/internal/sip:1008@10.86.11.50:5060 Standard REPORTING, cause: NORMAL_TEMPORARY_FAILURE 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/sip:1008@10.86.11.50:5060) State REPORTING going to sleep 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:319 (sofia/internal/sip:1008@10.86.11.50:5060) State Change CS_REPORTING -> CS_DESTROY 2009-11-03 15:39:50.818924 [DEBUG] switch_core_session.c:1003 Send signal sofia/internal/sip:1008@10.86.11.50:5060 [BREAK] 2009-11-03 15:39:50.818924 [DEBUG] switch_core_session.c:1140 Session 32 (sofia/internal/sip:1008@10.86.11.50:5060) Locked, Waiting on external entities 2009-11-03 15:39:50.818924 [NOTICE] switch_core_session.c:1158 Session 32 (sofia/internal/sip:1008@10.86.11.50:5060) Ended 2009-11-03 15:39:50.818924 [NOTICE] switch_core_session.c:1160 Close Channel sofia/internal/sip:1008@10.86.11.50:5060 [CS_DESTROY] 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:1008@10.86.11.50:5060) Running State Change CS_DESTROY 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:426 (sofia/internal/sip:1008@10.86.11.50:5060) State DESTROY 2009-11-03 15:39:50.818924 [DEBUG] mod_sofia.c:278 sofia/internal/sip:1008@10.86.11.50:5060 SOFIA DESTROY 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:60 sofia/internal/sip:1008@10.86.11.50:5060 Standard DESTROY 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:426 (sofia/internal/sip:1008@10.86.11.50:5060) State DESTROY going to sleep 2009-11-03 15:39:50.820304 [DEBUG] mod_sofia.c:569 Local SDP sofia/internal/1000(a)nshplpbx1.unix: v=0 o=FreeSWITCH 1257259678 1257259679 IN IP4 10.85.0.53 s=FreeSWITCH c=IN IP4 10.85.0.53 t=0 0 m=audio 24712 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-11-03 15:39:50.821960 [DEBUG] sofia.c:3588 Channel sofia/internal/1000(a)nshplpbx1.unix entering state [completed][200] 2009-11-03 15:39:50.821960 [DEBUG] switch_core_session.c:664 Send signal sofia/internal/1000(a)nshplpbx1.unix [BREAK] 2009-11-03 15:39:50.821960 [NOTICE] mod_dptools.c:653 Channel [sofia/internal/1000(a)nshplpbx1.unix] has been answered 2009-11-03 15:39:50.821960 [DEBUG] switch_channel.c:182 sofia/internal/1000(a)nshplpbx1.unix receive message [AUDIO_SYNC] EXECUTE sofia/internal/1000(a)nshplpbx1.unix sleep(1000) 2009-11-03 15:39:50.821960 [DEBUG] switch_channel.c:182 sofia/internal/1000(a)nshplpbx1.unix receive message [AUDIO_SYNC] 2009-11-03 15:39:50.867612 [DEBUG] switch_rtp.c:1917 Correct ip/port confirmed. 2009-11-03 15:39:50.926503 [DEBUG] sofia.c:3588 Channel sofia/internal/1000(a)nshplpbx1.unix entering state [ready][200] EXECUTE sofia/internal/1000(a)nshplpbx1.unix voicemail(default 10.85.0.53 1008) 2009-11-03 15:39:51.840356 [DEBUG] mod_voicemail.c:799 [default] rwlock 2009-11-03 15:39:51.840356 [DEBUG] switch_channel.c:182 sofia/internal/1000(a)nshplpbx1.unix receive message [AUDIO_SYNC] 2009-11-03 15:39:51.959567 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-03 15:39:51.961607 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-person.wav] (en:en) 2009-11-03 15:39:51.961607 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-03 15:39:51.961607 [DEBUG] switch_core_io.c:660 sofia/internal/1000(a)nshplpbx1.unix receive message [TRANSCODING_NECESSARY] 2009-11-03 15:39:53.319706 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-03 15:39:53.439693 [DEBUG] switch_ivr_play_say.c:273 Handle say:[1008] (en:en) 2009-11-03 15:39:53.439693 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-03 15:39:53.439693 [DEBUG] switch_core_io.c:660 sofia/internal/1000(a)nshplpbx1.unix receive message [TRANSCODING_NECESSARY] 2009-11-03 15:39:53.899622 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-03 15:39:53.899622 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-03 15:39:53.899622 [DEBUG] switch_core_io.c:660 sofia/internal/1000(a)nshplpbx1.unix receive message [TRANSCODING_NECESSARY] 2009-11-03 15:39:54.127652 [NOTICE] sofia.c:328 Hangup sofia/internal/1000(a)nshplpbx1.unix [CS_EXECUTE] [NORMAL_CLEARING] 2009-11-03 15:39:54.127652 [DEBUG] switch_channel.c:1896 Send signal sofia/internal/1000(a)nshplpbx1.unix [KILL] 2009-11-03 15:39:54.127652 [DEBUG] switch_core_session.c:1003 Send signal sofia/internal/1000(a)nshplpbx1.unix [BREAK] 2009-11-03 15:39:54.127652 [DEBUG] switch_core_state_machine.c:451 thread mismatch skipping state handler. 2009-11-03 15:39:54.139706 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:340 (sofia/internal/1000(a)nshplpbx1.unix) State EXECUTE going to sleep 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/1000(a)nshplpbx1.unix) Running State Change CS_HANGUP 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/1000(a)nshplpbx1.unix) State HANGUP 2009-11-03 15:39:54.239929 [DEBUG] mod_sofia.c:329 sofia/internal/1000(a)nshplpbx1.unix Overriding SIP cause 480 with 503 from the other leg 2009-11-03 15:39:54.239929 [DEBUG] mod_sofia.c:361 Channel sofia/internal/1000(a)nshplpbx1.unix hanging up, cause: NORMAL_CLEARING 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1000(a)nshplpbx1.unix Standard HANGUP, cause: NORMAL_CLEARING 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/1000(a)nshplpbx1.unix) State HANGUP going to sleep 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1000(a)nshplpbx1.unix) State Change CS_HANGUP -> CS_REPORTING 2009-11-03 15:39:54.239929 [DEBUG] switch_core_session.c:1003 Send signal sofia/internal/1000(a)nshplpbx1.unix [BREAK] 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/1000(a)nshplpbx1.unix) Running State Change CS_REPORTING 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/1000(a)nshplpbx1.unix) State REPORTING 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1000(a)nshplpbx1.unix Standard REPORTING, cause: NORMAL_CLEARING 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/1000(a)nshplpbx1.unix) State REPORTING going to sleep 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:319 (sofia/internal/1000(a)nshplpbx1.unix) State Change CS_REPORTING -> CS_DESTROY 2009-11-03 15:39:54.239929 [DEBUG] switch_core_session.c:1003 Send signal sofia/internal/1000(a)nshplpbx1.unix [BREAK] 2009-11-03 15:39:54.239929 [DEBUG] switch_core_session.c:1140 Session 31 (sofia/internal/1000(a)nshplpbx1.unix) Locked, Waiting on external entities 2009-11-03 15:39:54.239929 [NOTICE] switch_core_session.c:1158 Session 31 (sofia/internal/1000(a)nshplpbx1.unix) Ended 2009-11-03 15:39:54.239929 [NOTICE] switch_core_session.c:1160 Close Channel sofia/internal/1000(a)nshplpbx1.unix [CS_DESTROY] 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1000(a)nshplpbx1.unix) Running State Change CS_DESTROY 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:426 (sofia/internal/1000(a)nshplpbx1.unix) State DESTROY 2009-11-03 15:39:54.239929 [DEBUG] mod_sofia.c:278 sofia/internal/1000(a)nshplpbx1.unix SOFIA DESTROY 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1000(a)nshplpbx1.unix Standard DESTROY 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:426 (sofia/internal/1000(a)nshplpbx1.unix) State DESTROY going to sleep ]0;root@NSHPLPBX1:/usr/local/freeswitch/log[root@NSHPLPBX1 log]#

6 8

problem with failover routes for LCR A-Z scenario
by Hound Dog 05 Nov '09

05 Nov '09

I have a general question regrading MOD_LCR and the way it chooses main and failover routes ( backups ) it came out a little long , sorry for that :) I found that it difficult/impossible to make LCR use only carriers that I choose scenario is as follows , taking the UK as example for a destination ( prices are not real , just an example ) I have 2 carriers offering routes to the UK , landline and mobile my buying prices Destination carrier1 Price carrier2 price 44 (all UK) $0.01 $0.01 447 (UK mobile) $0.15 $0.19 my selling prices Destination price 44 (all UK) $0.015 447 (UKmobile) $0.17 so for UK landline both carrier 1 and carrier 2 are good for me , so I use them and be profitable for UK mobile I can ** only ** make a profit if I use carrier 1 ( if I use carrier2 I actually lose money on every calls since I sell the call for 17 cents but buy for 19 cents so I LOSE 2 cents a minute) translating it to MOD_LCR information digits rate carrier_id ( other columns ignored ) 44 0.01 1 44 0.01 2 447 0.015 1 this looks good : 44 prefix will be shared between carrier 1 and 2 447 prefix will only go to carreir 1 so it fits perfectly - BUT testing this I get - API CALL [lcr(447965404547)] output: | Digit Match | Carrier | Rate | Codec | CID Regexp | Dialstring | | 447 | carr1 | 0.15 | G711 | | [lcr_carrier=carr1,lcr_rate=1.00000,absolute_codec_string=G729]sofia/external/447965404547(a)10.10.10.1 | | 44 | carr2 | 0.01 | G711 | | [lcr_carrier=carr2,lcr_rate=1.00000,absolute_codec_string=G729]sofia/external/447965404547(a)10.10.10.2 | Notice the lcr engine is using carrier2 to route the call as backup for carrier1 , because it has coverage of that range ( 44 covers 447xxxx ) - it all makes sense ** BUT ** carrier2 should not be used for 447 range , I will lose money on each call I send there , and I actually prefer calls to fail so far I didnt find a solution for that , so if there is one I love bo pointed there I did think it over a little and came up with 2 options that could be used , and I am also planning to code them and propose patch to maintainers , I would love to get comments on those ( in case there are no existing solution ) option 1 - setting some routes as last option , add another param to the LCR table called last_route , when hitting a route with last_route=1, stop processing additional routes and return your routing decision so far so in our case the route entry with 44 to carrier1 will have last_route=1 , the other 44 routes will have last_route=0 to allow for failover option 2 - don't allow shorter prefixes , once a prefix match was found with a N digits length , do not accept less digits prefix matches. in other words dont failover from a finer route to a wider route. it will need to be a global option and I will be quite simple to use, but will require entering mutiple entries of the same length prefix for each carrier you would like to use its intutive and relatively simple to manage , but requires more lcr entries to get you where you want

2 4

Users hanged up for unknown reason
by Maciej Aniserowicz 05 Nov '09

05 Nov '09

Hi, I have a strange problem. I control FS with commands sent by tcp in response to events published via tcp. I do something like: 1) call 1st user 2) call 2nd user 3) 1st and 2nd talk 4) call another user 5) 1st and another talk etc... Sometimes (quite regularly) users are hanged up (with cause NORMAL_CLEARING) even if they do not hangup manually. I pasted one such scenario in pastebin (http://pastebin.freeswitch.org/10955) it includes logs from commands sent by me and events received from FS. Could someone take a look and see what am I doing wrong? The scenario includes 3 users 1st user (Unique-ID: f076261a-4537-40f2-b46d-933141320314) is supposed to be connected all the time but gets diconnected 2nd user (Unique-ID: ebdfb398-ec82-4760-9f79-81364e0f37b6) is supposed to talk for a few seconds and get killed 3rd user (Unique-ID: d5cd839e-793c-4b3c-adda-327841672a5f) is supposed to work like 2nd user All of them are simulated by dialplan extensions (using answer and playback tools), but the same thing happends for xlite or cisco phone. Maciej Aniserowicz -- View this message in context: http://n2.nabble.com/Users-hanged-up-for-unknown-reason-tp3937601p3937601.h… Sent from the freeswitch-users mailing list archive at Nabble.com.

2 2

Re: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media
by Humberto Quintana 05 Nov '09

05 Nov '09

Thanks for your time, -The scenario is still the same: Always bypass media. Environment 100% NAT free :-) Call established from A to B through FS. Then... Blind transfer from B to C (Refer-to: C) RTP should go directly between A and C. -With 1.0.4 and 1.0.5pre3, FS actually INVITEs C after receiving the REFER-to:C, BUT there is no 2-way audio. Only RTP from C to A (due to the lack of reINVITE to A, after C answers). Please check SIP diagram here: http://provision.netcelerate.net/ngrep/blindxfer2009-11-04-v1.0.5pre3.html -What it's wrong with r15332 is there is not such call to C. For sure I know SIP is a protocol, may be my description was not clear but this SIP diagram speaks by itself ;-) http://provision.netcelerate.net/ngrep/blindxfer2009-11-04rev15332.html -You could check the sofia debug for r15332 here: http://pastebin.com/m6f2b3836 Best regards, Humberto > > I don't know what you are talking about anymore. > > The scenario I had tested is when a call is bridged in bypass_media=true > bridge > and you blind transfer that call back to the dialplan > > as soon as it hits the routing state it will resume media. > > > it has been confirmed to not work and confirmed to have been fixed several > time and if you are still having a problem you must have something blocking > some of your packets or something . > > You have to understand that sip is a protocol and your description is > completely non-standard. > Perhaps you should get a console trace and attach it to a jira. The trace > probably makes more sense to me. > > sofia profile internal siptrace on > console loglevel debug > > reproduce and attach the whole capture. > > > > On Tue, Nov 3, 2009 at 6:05 PM, Humberto Quintana wrote: > >> >> Hi, >> >> I tried r15332 and set in the sofia profile: >> >> a) bypass_media_after_bridge=true only >> b) bypass_media_after_bridge=true, param name="media-option" >> value="resume-media-on-hold"/> >> >> >> In both cases FS is hanging up the initial call (A to FS) after accepting >> the REFER to C: >> >> A <- reINVITE with FS' SDP <- FS >> A -> 200 -> FS >> A <- ACK <- FS >> A <- BYE <- FS >> >> The call to C is not even tried. >> >> I found this line is the logs that could give some idea: >> >> 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup >> sofia/external/514xxxxxx at a.b.c.d [CS_ROUTING] [RECOVERY_ON_TIMER_EXPIRE] >> after sending the ACK for the reINVITE >> >> >> Regards, >> >> >> Humberto >> >>>please try r15326 >>>I think i have it working. >>> >>>I recommend for optimal results you set bypass_media_after_bridge=true >>>either as a global or in your DP in place of bypass_media=true >>> >>> >>>On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana >> hotmail.com>wrote: >>> >>>> Hi Mike, >>>> >>>> I re-tried with trunk rev 15319 but I got almost the same behavior: >> There >>>> is now a reINVITE (with FS' SDP) going to A when the REFER is accepted. >> But >>>> still there is no reINVITE for A (with C's SDP) after the call from FS >> to C >>>> is established. >>>> >>>> Anyway, we decided for now to do a different implementation but if you >> want >>>> to explore more in this issue count me in ;-) >>>> >>>> >>>> Thank you very much! >>>> >>>> Humberto >> >> >> _________________________________________________________________ >> Windows Live: Friends get your Flickr, Yelp, and Digg updates when they >> e-mail you. >> http://go.microsoft.com/?linkid=9691817 >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > _________________________________________________________________ > Ready. Set. Get a great deal on Windows 7. See fantastic deals on Windows 7 now > http://go.microsoft.com/?linkid=9691818 _________________________________________________________________ Windows Live: Make it easier for your friends to see what you’re up to on Facebook. http://go.microsoft.com/?linkid=9691816

3 2