October 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

inaccurate display of Call-info 'appearance-uri' in SIP NOTIFY
by Vik Killa 29 Oct '12

29 Oct '12

I have a semi-complex dial-plan that uses the 'transfer' dp tool to route calls through various contexts. For instance, if a caller dials '*1' it will eventual go to '5' (which is an arbitrary ID) in the 'route' context. What I've noticed is, when using SLA, the Polycom's display is updated with the '5', instead of displaying '*1' as it should. I do not have this problem when not using SLA. I looked through the SIP trace to see where the Polycom was getting this on the display and it's happening in the NOTIFY (which makes sense why I don't have this problem when not using SLA). If you look at this line in the NOTIFY, you'll see: Call-Info: <sip:21.local>;appearance-index=1;appearance-state=alerting;appearance-uri="\"5\" <sip:5@21.local>" Here is the full NOTIFY packet... NOTIFY sip:2000@XX.YY.ZZ.YYY:1411 SIP/2.0 Via: SIP/2.0/UDP XX.YY.ZZ.154;rport;branch=z9hG4bKQp5tXcrD1jj2j Max-Forwards: 70 From: <sip:2000@21.local>;tag=YN59QVIfZbRE To: "Ext. 2000" <sip:2000@21.local>;tag=307FCA01-A83BB3C Call-ID: 510f5fc5-b143c4b0-ecbece7f(a)192.168.99.78 CSeq: 388215190 NOTIFY Contact: <sip:2000@XX.YY.ZZ.154:5060> Expires: 0 Call-Info: <sip:21.local>;appearance-index=1;appearance-state=alerting;appearance-uri="\"5\" <sip:5@21.local>" Call-Info: <sip:21.local>;appearance-index=*;appearance-state=idle User-Agent: AMPSwitch Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: line-seize Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: terminated;reason=noresource Content-Length: 0 Is there anyway to set the 'appearance-uri' or have FS use the original SIP URI dialed here (*1) ??? Thanks in advance!

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Re: [Freeswitch-users] mod_xml_curl SOAP newbie question (Jeff Geddes)
by Jeff Geddes 29 Oct '12

29 Oct '12

I'm dealing with a third party interface which only supplies a SOAP response, so that's why I was wondering if that SOAP response has to be reformatted before sending the response back to FS. Is there a better way? thanks jeff On Mon, Oct 29, 2012 at 2:13 AM, < 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: Multiple Domains (Brian West) > 2. Re: sending notify to reset or restart cisco sip ip phones > (Brian West) > 3. Re: Using Freeswitch as a High CPS Outbound Dialer (Brian West) > 4. Re: anyone using t38faxing.com through ezcall? (Brian West) > 5. Re: mod_xml_curl SOAP newbie question (Brian West) > 6. Re: sending notify to reset or restart cisco sip ip phones > (Babak Yakhchali) > 7. Re: tx fax: call drops prematurely (Babak Yakhchali) > > > ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Sun, 28 Oct 2012 22:53:58 -0500 > Subject: Re: [Freeswitch-users] Multiple Domains > Yes, thats the entire purpose of that alias feature. While the > domain_exists doesn't use the alias feature to find the domain it actually > looks in the XML. > -- > Brian West > brian(a)freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9266 > UK: +44 20 3298 4900 > ISN: 410*543 > > > > > > On Oct 28, 2012, at 1:12 PM, Nick Vines <jnvines(a)gmail.com> wrote: > > > The domain tag in my sofia profile is: > > <domain name="all" alias="true" parse="true"/> > > > > If I change alias to false, I cannot run the sofia status profile > domainN reg commands, but domain_exists domainN.mine.com will still > return true. > > > > > > ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Sun, 28 Oct 2012 22:54:28 -0500 > Subject: Re: [Freeswitch-users] sending notify to reset or restart cisco > sip ip phones > You need to make sure you're on the latest code first. > -- > Brian West > brian(a)freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9266 > UK: +44 20 3298 4900 > ISN: 410*543 > > > > > > On Oct 27, 2012, at 5:21 PM, Babak Yakhchali <babak.freeswitch(a)gmail.com> > wrote: > > > Thanks. Finally I found the way I should send the restart request. I'm > using > > sofia profile internal check_sync 223(a)192.168.1.3 > > But nothing happens. I captured packets sent from freeswitch to my Cisco > Ip Communicator and attached them. it seems softphone is sending bad > request. > > > > > > > > ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Sun, 28 Oct 2012 22:55:51 -0500 > Subject: Re: [Freeswitch-users] Using Freeswitch as a High CPS Outbound > Dialer > Yah AGPL is the one that can reach over a socket and OWN YOU if I recall > correctly. You can't even build services that are web facing without > disclosure of all your code. > -- > Brian West > brian(a)freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9266 > UK: +44 20 3298 4900 > ISN: 410*543 > > > > > > On Oct 27, 2012, at 3:32 PM, Anthony Minessale < > anthony.minessale(a)gmail.com> wrote: > > > Eww AGPL..... > > > > > > ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Sun, 28 Oct 2012 22:56:37 -0500 > Subject: Re: [Freeswitch-users] anyone using t38faxing.com through ezcall? > I had but when they made changes recently I had stopped using them due to > not having time to trouble shoot yet another provider issue. I guess I > should revisit this! LOL > -- > Brian West > brian(a)freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9266 > UK: +44 20 3298 4900 > ISN: 410*543 > > > > > > On Oct 26, 2012, at 1:48 PM, Mitch Capper <mitch.capper(a)gmail.com> wrote: > > > I am curious if anyone actually uses t38faxing.com and their > > experience? I have used them as a backup but they seem to have multi > > day outages (current one is around two weeks now) periodically with > > little level of caring. I can't believe this is the actual service > > they provide but they claim the issues are with an upstream carrier so > > I would assume affecting all customers. > > > > ~Mitch > > > > > > ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Sun, 28 Oct 2012 22:57:41 -0500 > Subject: Re: [Freeswitch-users] mod_xml_curl SOAP newbie question > Bottom line is.. SOAP won't even factor in. > -- > Brian West > brian(a)freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9266 > UK: +44 20 3298 4900 > ISN: 410*543 > > > > > > On Oct 27, 2012, at 1:38 PM, Jeff Geddes <geddes.jeffrey(a)gmail.com> wrote: > > > this is not so much a question but rather I'd like to know if I'm on the > right track with this. > > > > I want to use mod_xml_curl to call a SOAP web service and return a > dialplan. > > > > Am I correct in saying that the SOAP response has to be re-formatted by > some other script/etc. on the web server before the response is sent back > to FS so it is in the correct XML dialplan format that FS requires? > > > > thanks > > jeff > > ________ > > > > > > ---------- Forwarded message ---------- > From: Babak Yakhchali <babak.freeswitch(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 29 Oct 2012 08:38:08 +0330 > Subject: Re: [Freeswitch-users] sending notify to reset or restart cisco > sip ip phones > I updated 2 days ago! I captured packets from a cucm and they look like fs > packets. but I donno why it is not working > > On Mon, Oct 29, 2012 at 7:24 AM, Brian West <brian(a)freeswitch.org> wrote: > >> You need to make sure you're on the latest code first. >> -- >> Brian West >> brian(a)freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH_Wire >> T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9266 >> UK: +44 20 3298 4900 >> ISN: 410*543 >> >> >> >> >> >> On Oct 27, 2012, at 5:21 PM, Babak Yakhchali <babak.freeswitch(a)gmail.com> >> wrote: >> >> > Thanks. Finally I found the way I should send the restart request. I'm >> using >> > sofia profile internal check_sync 223(a)192.168.1.3 >> > But nothing happens. I captured packets sent from freeswitch to my >> Cisco Ip Communicator and attached them. it seems softphone is sending bad >> request. >> > >> >> >> _________________________________________________________________________ >> 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: Babak Yakhchali <babak.freeswitch(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 29 Oct 2012 08:42:34 +0330 > Subject: Re: [Freeswitch-users] tx fax: call drops prematurely > Thanks Mitch > My set up is fs(g711) => cisco (e1 pri) => pstn. and I'm not using t38. If > you see my pastebin link it seems some timer is causing sofia to send bye > message and causes the call to drop. > > On Sun, Oct 28, 2012 at 11:35 PM, Mitch Capper <mitch.capper(a)gmail.com>wrote: > >> Hi Babak, >> Check the wiki for the suggested methods to try for faxing also >> provider can make a big difference. Let us know what you have tried >> /are doing. >> ~mitch >> >> On Sun, Oct 28, 2012 at 11:45 AM, Babak Yakhchali >> <babak.freeswitch(a)gmail.com> wrote: >> > Hi >> > Sending faxes results in dropped calls: "The call dropped prematurely" >> > http://pastebin.freeswitch.org/20133 >> > >> > >> _________________________________________________________________________ >> > 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 >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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Re: [Freeswitch-users] Open Source Question
by Chris B. Ware 29 Oct '12

29 Oct '12

Thank you for your answers. Anthony: I know the effort made by you and your team. Hope to earn some money from this project and support FS!

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Open Source Question
by Chris B. Ware 29 Oct '12

29 Oct '12

Hi all, Hope this question is not so OT. I've made a PBX, FS based, writing functions with XML and Lua, for a customer. FS and all its components are based on many open source licenses, so I can't sell this software. Here my question: can I sell the solution I've made? Can I sell my particular configuration? If the answer is 'No', can I mantain intellectual property of my work? I'd like to avoid that a customer copy 'my' files and sell the service. Thanks for your help.

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Originate working from fs_cli but not via ESL
by Gurteshwar S 29 Oct '12

29 Oct '12

Hi, I can't figure out what could be the problem here. If i run 'originate {ignore_early_media=true}freetdm/1/a/9999999999 &speak(flite|kal|hello)' from fs_cli it works as expected, the cellphone number is called and TTS message played. However, if I run the exact same command via ESL and watch the logs, the call never goes to the cell phone number and I get: "originate Resulted in Error Cause: 31 [NORMAL_UNSPECIFIED]" . I am using this library https://github.com/fiorix/eventsocket Any ideas why this happens? I tried the ESL command with api and bgapi, same result in both cases. Thanks

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G400P wit Freeswitch
by Kamminthang Nengzalam 29 Oct '12

29 Oct '12

Hi Team I'm trying to configure Freeswitch to work with my GSM G400P. Can you please let mi know the step by step procedure. I'm new to freeswitch. And thank you in advance for helping mi. Thanks, Kam

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how can reset hash, with out reloading mod_hash
by ram 29 Oct '12

29 Oct '12

Hi all I've tried hash_reset, but it did not work how can i reset hash_dump all L/MAX_CALLS/0/1/86400/1xxxxxx how can i reset this value in fs_cli ? any advise, thanks in advance. Ram

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Attended transfer fails when in bypass or proxy mode
by Sergei 29 Oct '12

29 Oct '12

Good day all. I've recently installed freeswitch in our office and in general everybody is happy. The only disappointing thing - is attended transfer. I've tried to make it working on Cisco7960 and on Grandstream GXP2000 - but with no luck. Could please, anybody respond, who successfully makes attended transfer using bypass mode? In sip profile I use following options: <param name="media-option" value="resume-media-on-hold"/> <param name="media-option" value="bypass-media-after-att-xfer"/> Scenario is the simplest one: A calls B, B transfers to C (with attended transfer - like putting A on hold, making outcall to C, bridging two calls together). What I observe on Cisco phone - is following: 1. A calls B. B answers, puts A on hold and calls C. C answers and talks to B. Up to this time - everything is fine. 2. B presses 'transfer', in order to bridge two calls together. But, instead, C hangs up, A stays on hold and B looks like he just put A on hold. If B right now presses hold button - he will appear in dialog with A again. Did anybody observe such behavior? What could be the advice in such situation? Thank you in advance, Siarhei.

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Mod_limit questions
by ivdreg ivdreg 29 Oct '12

29 Oct '12

Hi All, I have some questions regarding mod_limit. 1. I'm using 16674 in production but tested and with 17001. I try mod_limit with odbc/mysql and sqlite and have same problem: mod_limit using limit_hash never tries to write data to SQL. It works (counts sessions) on log screen but no data in MySQL/Sqlite. It seems that data is cached but cache never flushed to DB. Is anyone has this problem. My limits are: <action application="limit_hash" data="op_${sip_h_X-SRC-Cust} ${destination_number} 5 !INCOMING_CALL_BARRED" /> <action application="limit_hash" data="in_${sip_h_X-SRC-Cust} ${caller_id_number} 10/300 !OUTGOING_CALL_BARRED" /> and they are applied on inbound call profile in same extension. 2. I need to implement rule maximum of 50 calls per day/24h from the same CLID (A NUMBER) after that calls to be limited. I'm not sure but rule in example "Limit calls to 5 calls every 10 minutes" is about concurrent call but not calls processed as count. Any idea ? Thanks for any help

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sending notify to reset or restart cisco sip ip phones
by Babak Yakhchali 29 Oct '12

29 Oct '12

Hi Is there anyway to send a notify message to a cisco phone with sip firmware to reset or restart it like this link: http://www.freepbx.org/forum/freepbx/tips-and-tricks/cisco-9971-phone-confi… The following variables and functions should be included in sip_notify_custom.conf settings: [cisco-restart] Event=>service-control Subscription-State=>active Content-Type=>text/plain Content=>action=restart Content=>RegisterCallId={${SIPPEER(${PEERNAME},regcallid)}} Content=>ConfigVersionStamp={0000000000000000} Content=>DialplanVersionStamp={0000000000000000} Content=>SoftkeyVersionStamp={0000000000000000} [cisco-reset] Event=>service-control Subscription-State=>active Content-Type=>text/plain Content=>action=reset Content=>RegisterCallId={${SIPPEER(${PEERNAME},regcallid)}} Content=>ConfigVersionStamp={0000000000000000} Content=>DialplanVersionStamp={0000000000000000} Content=>SoftkeyVersionStamp={0000000000000000}

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