April 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re-2: FreeSwitch Complex IVR System
by Guido Kuth 16 Apr '09

16 Apr '09

Hi Dave, thanks for the answer. I am playing around with FS and Event Socket Library for .NET. I get pretty much to run with this, but the reason why I came from Asterisk to FS is that I cannot get DTMF in a bridged call. I thought that I get an Event as soon one dtmf digit is recognized. Unfortunately this isn't the case. If I use the default config files and map the keys with bind_meta_app the dtmf tones are recognized and the function behind the bound app is executed. Is this maybe a bug. I have read about mod_managed and that I should use it, but I haven't found anything about the usage of it. Any suggestions would help.... thanks...Guido -------- Original Message -------- Subject: Re: [Freeswitch-users] FreeSwitch Complex IVR System (16-Apr-2009 17:35) From: David Knell <dave(a)3c.co.uk> To: gk(a)exram.de > Hi Guido, > > My preferred way is to talk to FS through its event socket > interface. This allows you fully to control FS, whilst giving > you the power to write the code in whatever language and on > whatever platform you choose. > > The documentation starts here: > http://wiki.freeswitch.org/wiki/Mod_event_socket > > Cheers -- > > Dave > > > Hi @all > > > > I have a question about a project I want to realize with FreeSwitch. I > > want to do a complex IVR System which takes a call, do many things in > > a MSSQL DB, send some Informations to one or many Middleware Servers > > via TCP/IP, call one or more mobile phones, the first is able to take > > the call, it can be that he must be able to hear a prompt before he is > > actually connected to the first caller, then the conversation must be > > recorded automatically and during the conversation it must be possible > > for the called party to redirect the call by dtmf. I know that this is > > all possible, but I want to know which way is the best to do all this? > > > > > > _______________________________________________ > > 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

2 1

RE FS/Sofia not sending bye when FreeSWITCH nds the call?
by Peter Olsson 16 Apr '09

16 Apr '09

My current revision is r13015. I will do an update as soon as possible and see if that solves the issue. Thanks! //Peter ________________________________ Från: freeswitch-users-bounces(a)lists.freeswitch.org [freeswitch-users-bounces(a)lists.freeswitch.org] för Anthony Minessale [anthony.minessale(a)gmail.com] Skickat: den 15 april 2009 18:46 Till: freeswitch-users(a)lists.freeswitch.org Ämne: Re: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the call? This sounds familiar: What revision of the code is this? Can you confirm you have this problem with SVN trunk (r13034 at the time of this writing). On Wed, Apr 15, 2009 at 11:24 AM, Peter Olsson <peter.olsson(a)visionutveckling.se<mailto:peter.olsson@visionutveckling.se>> wrote: This is the full SIP-trace for the call. It’s not sending a BYE at all, and I can’t see one in Wireshark either. As you can see in the end there is a call to hangup_function(), but no SIP messages after that. When I manually hangup the phone I can see it sends BYE to FreeSWITCH (which is quite expected, since it thinks the call still exists), and FreeSWITCH just answers ”481 Call Does Not Exist” – which of course is also expected, since the call was dropped. recv 1255 bytes from udp/[192.168.94.53]:32769 at 16:17:57.853727: ------------------------------------------------------------------------ INVITE sip:2100@192.168.1.155:5060;lr SIP/2.0 Accept-Language: en Call-ID: 80948a675733de14449f79df00 CSeq: 1 INVITE From: "Peter Olsson" <sip:1002@sip.se:6001<http://sip:1002@sip.se:6001>>;tag=80948a675733de13449f79df00 Record-Route: <sip:192.168.94.53:5060;lr>,<sip:192.168.94.53:6001;lr;transport=tls> To: "2100" <sip:2100@192.168.94.53<mailto:sip%3A2100@192.168.94.53>> Via: SIP/2.0/UDP 192.168.94.53:5060;branch=z9hG4bK8383830303039393936541.0,SIP/2.0/TLS 192.168.94.53:6001;psrrposn=2;received=192.168.94.53;branch=z9hG4bK80948a675733de15449f79df00 Content-Length: 165 Content-Type: application/sdp Contact: "Peter Olsson" <sip:1002@192.168.94.53:6001;transport=tls> Max-Forwards: 67 User-Agent: Avaya CM/R015x.01.1.415.1 Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH Supported: 100rel,timer,replaces,join,histinfo Alert-Info: <cid:internal@invalid.unknown.domain>;avaya-cm-alert-type=internal Min-SE: 1200 Session-Expires: 1200;refresher=uac P-Asserted-Identity: "Peter Olsson" <sip:1002@sip.se:6001<http://sip:1002@sip.se:6001>> History-Info: <sip:2100@192.168.94.53<mailto:sip%3A2100@192.168.94.53>>;index=1,"2100" <sip:2100@192.168.94.53<mailto:sip%3A2100@192.168.94.53>>;index=1.1 v=0 o=- 1 1 IN IP4 192.168.94.53 s=- c=IN IP4 192.168.94.59 b=AS:64 t=0 0 m=audio 2062 RTP/AVP 8 127 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 ------------------------------------------------------------------------ send 541 bytes to udp/[192.168.94.53]:5060 at 16:17:57.854727: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.94.53:5060;branch=z9hG4bK8383830303039393936541.0,SIP/2.0/TLS 192.168.94.53:6001;psrrposn=2;received=192.168.94.53;branch=z9hG4bK80948a675733de15449f79df00 Record-Route: <sip:192.168.94.53:5060;lr> Record-Route: <sip:192.168.94.53:6001;lr;transport=tls> From: "Peter Olsson" <sip:1002@sip.se:6001<http://sip:1002@sip.se:6001>>;tag=80948a675733de13449f79df00 To: "2100" <sip:2100@192.168.94.53<mailto:sip%3A2100@192.168.94.53>> Call-ID: 80948a675733de14449f79df00 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Content-Length: 0 ------------------------------------------------------------------------ 2009-04-15 18:17:57 [NOTICE] switch_channel.c:597 switch_channel_set_name() NewChannel sofia/internal/1002@sip.se:6001<http://1002@sip.se:6001> [fa1c328e-bdfe-7d49-ab6f-dc9ec791c455] 2009-04-15 18:17:57 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing Peter Olsson->2100 in context public 2009-04-15 18:17:57 [NOTICE] mod_dptools.c:649 answer_function() Channel [sofia/internal/1002@sip.se:6001<http://1002@sip.se:6001>] has been answered send 1322 bytes to udp/[192.168.94.53]:5060 at 16:17:57.871727: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.94.53:5060;branch=z9hG4bK8383830303039393936541.0,SIP/2.0/TLS 192.168.94.53:6001;psrrposn=2;received=192.168.94.53;branch=z9hG4bK80948a675733de15449f79df00 Record-Route: <sip:192.168.94.53:5060;lr> Record-Route: <sip:192.168.94.53:6001;lr;transport=tls> From: "Peter Olsson" <sip:1002@sip.se:6001<http://sip:1002@sip.se:6001>>;tag=80948a675733de13449f79df00 To: "2100" <sip:2100@192.168.94.53<mailto:sip%3A2100@192.168.94.53>>;tag=Sv6KrDv9vQrer Call-ID: 80948a675733de14449f79df00 CSeq: 1 INVITE Contact: <sip:mod_sofia@192.168.1.155:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Require: timer Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 1200;refresher=uac Min-SE: 1200 Content-Type: application/sdp Content-Disposition: session Content-Length: 265 v=0 o=FreeSWITCH 484797194364394181 220756314446402535 IN IP4 192.168.1.155 s=FreeSWITCH c=IN IP4 192.168.1.155 t=0 0 m=audio 23574 RTP/AVP 8 127 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ recv 521 bytes from udp/[192.168.94.53]:32769 at 16:17:57.880727: ------------------------------------------------------------------------ ACK sip:mod_sofia@192.168.1.155:5060;transport=udp SIP/2.0 From: "Peter Olsson" <sip:1002@sip.se:6001<http://sip:1002@sip.se:6001>>;tag=80948a675733de13449f79df00 To: "2100" <sip:2100@192.168.94.53<mailto:sip%3A2100@192.168.94.53>>;tag=Sv6KrDv9vQrer Call-ID: 80948a675733de14449f79df00 CSeq: 1 ACK Max-Forwards: 69 Via: SIP/2.0/UDP 192.168.94.53:5060;branch=z9hG4bK8383830303039393936541.1,SIP/2.0/TLS 192.168.94.53:6001;psrrposn=1;branch=z9hG4bK80948a675733de16449f79df00 User-Agent: Avaya CM/R015x.01.1.415.1 Content-Length: 0 Record-Route: <sip:192.168.94.53:5060;lr> ------------------------------------------------------------------------ 2009-04-15 18:18:02 [NOTICE] mod_dptools.c:633 hangup_function() Hangup sofia/internal/1002@sip.se:6001<http://1002@sip.se:6001> [CS_EXECUTE] [NORMAL_CLEARING] 2009-04-15 18:18:02 [NOTICE] switch_core_session.c:1021 switch_core_session_thread() Session 5 (sofia/internal/1002@sip.se:6001<http://1002@sip.se:6001>) Ended 2009-04-15 18:18:02 [NOTICE] switch_core_session.c:1023 switch_core_session_thread() Close Channel sofia/internal/1002@sip.se:6001<http://1002@sip.se:6001> [CS_DESTROY] Från: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] För Anthony Minessale Skickat: den 15 april 2009 17:27 Till: freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Ämne: Re: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the call? type: sofia profile internal siptrace on at the cli and try again see if you cen see FS sending BYE to the wrong address. This can be caused by a false positive on the NAT detection or when you need NAT mode and you don't have it enabled. first edit the sofia profile in your config and comment out any line with the word nat in them On Wed, Apr 15, 2009 at 8:43 AM, Peter Olsson <peter.olsson(a)visionutveckling.se<mailto:peter.olsson@visionutveckling.se>> wrote: When I do a call from my Avaya SIP Server to FreeSWITCH. And then let FreeSWITCH do a hangup of the call, FreeSWITCH doesn’t seem to send a ”BYE” back to the Avaya PBX. I’ve narrowed it down to this simple example in the dialplan; <action application="answer"/> <action application="sleep" data="5000"/> <action application="hangup"/> In this case no BYE is issued, and the phone still thinks the call is alive. If you want to I could send the SIP headers as well for this scenario.. Regards, Peter Olsson _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org<mailto:Freeswitch-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com<mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888<http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org<mailto:Freeswitch-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com<mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888<http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 !DSPAM:49e611a532932126013823!

3 6

Re: [Freeswitch-users] how many simultaneous calls support in freeswitch
by Parveen Kumar Jain 16 Apr '09

16 Apr '09

> Hi, > > I need to develop an IVR application which makes an outbound calls and > then plays the some audio file for the user. For this I was trying to > evaluate Freeswitch under following criteria: > > - Does freeswitch have outbound calls support(is there any conf file file > avilabel where I just can put some series of no. and freeswitch just calls > those no. sequentially)? > - If yes, how many simultaneous calls are possible on a simple pentium-4 > using G711U as a codec(1 GB RAM, 2.2 GHz machine) machine?(I had checked the > switch.conf file and it says that by default it can support upto 1000 calls > , is it true for a small machine also)? In other terms is Freeswitch is > scalable if I need to add more users to call from here? > - Does freeswitch have the support of CDR(call data record) after > succesfull calls ? > > It will be great help if any of you can comment on these question and it > can save my several hours of testing before I can make my conclusion :) > > Best Regards, > Parveen Jain >

4 3

Re: [Freeswitch-users] Entire Wiki.FreeSwitch.org on Single PDF ?
by Will Boyce 16 Apr '09

16 Apr '09

It may be worth looking at http://www.mediawiki.org/wiki/Extension:Pdf_Export or http://www.mediawiki.org/wiki/Extension:Pdf_Book -- Regards, Will Boyce <mail(a)willboyce.com> tel: 07933 515 987 url: http://willboyce.com ----- "Michael Collins" <msc(a)freeswitch.org> wrote: | From: "Michael Collins" <msc(a)freeswitch.org> | To: freeswitch-users(a)lists.freeswitch.org | Sent: Wednesday, 15 April, 2009 18:12:26 GMT +00:00 GMT Britain, Ireland, Portugal | Subject: Re: [Freeswitch-users] Entire Wiki.FreeSwitch.org on Single PDF ? | | Unfortunately this isn't being maintained and Bret didn't give his script to any of us. If anyone out there is familiar with converting wiki pages to PDF and is willing to pick this up then by all means contact me off list and we'll discuss it. | | -MC | | | On Wed, Apr 15, 2009 at 12:22 AM, Mitul Limbani < mitul(a)enterux.com > wrote: | Hello there, | | In my previous encounter with FreeSwitch, I had found that Bret had posted on the Mailing List somewhere about availability of the entire FreeSwitch Wiki Documentation on a single PDF, this is useful coz at the offset apart from Wiki there is no other offline media to learn it. | | Is the same PDF available looking at the growth of Wiki pages and the updation. | | I look forward to hear from you guys, | | Thanks & Regards, | Mitul Limbani, | Founder & CEO, | Enterux Solutions, | The Enterprise Linux Company (TM), | www.enterux.com | +91-9820332422 | | Msg sent via Enterux Enterprise Email Server : http://www.enterux.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

1 0

conference from a sip provider
by Antony King 16 Apr '09

16 Apr '09

I'm just getting started with freeswitch; I'd like to create a public phone number that a small number of people can dial into to join a conference. I've got calls and the conference rooms working internally, and I've got a link to my sipgate account which directs to extension 1000 . The configuration is virtually unchanged from the default. I created sip_profiles/external/sipgate.xml using the default template in that folder and put my details in. However, if I change sip_profiles/external/sipgate.xml to point to 3001, ie <param name="extension" value="3001"/> then when the external call comes in, freeswitch does this 3 times then stops: 2009-04-15 16:05:30 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/external/0XXXXXXXXX(a)sipgate.co.uk [e56ac315-62e8-40b8-aa5a-bcc5dbd841ee] 2009-04-15 16:05:30 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 0XXXXXXXXX->3001 in context public 2009-04-15 16:05:30 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup sofia/external/0XXXXXXX(a)sipgate.co.uk [CS_EXECUTE] [NORMAL_CLEARING] 2009-04-15 16:05:30 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 1 (sofia/external/0XXXXXXX(a)sipgate.co.uk) Ended 2009-04-15 16:05:30 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/external/0XXXXXXX(a)sipgate.co.uk [CS_HANGUP] Presumably there's some difference between calls coming in via a gateway and localy generated calls; could someone give me some pointers as to how to get it to accept the call ?

3 3

[ANN] Spice Telephony 0.3 - an open source FreeSWITCH/Erlang callcenter
by Andrew Thompson 16 Apr '09

16 Apr '09

Well, it's been a few months since I mentioned this project last here, so here's an update over my last announcement (see http://lists.freeswitch.org/pipermail/freeswitch-users/2009-January/010048.… ) Things have improved a *lot* since the last time I mentioned it: * Support for inbound calls (somehow last time that wasn't even finished) * Support for outbound calls * Support for ringing agents on a softphone or ringing agents on a POTS line * A web interface for managing the system (managing agents, skills, queues, etc) * A web interface for agents to manage their state (go released, go idle, indicate when they're done with wrapup, etc) Along with a boatload of minor features and even more bugfixes. This release is actually something you can play with without knowing Erlang, we've even included a boot script to setup and run everything for you. So, if you're interested in a distributed, fault tolerant callcenter platform built on top of FreeSWITCH I invite you to check out http://wiki.opencsm.org/wiki/index.php/Spice_Telephony , the rest of the opencsm.org site and especially the other telephony pages on the wiki and, if you're interested, please give it a try and let us know what you think. Just remember that it's still very much a work in progress (although I hope to be able to give a cool demo at ClueCon in August :) ). Andrew Thompson - opencsm.org

2 1

Re: [Freeswitch-users] RE FS/Sofia not sending bye when FreeSWITCH nds the call?
by Peter Olsson 16 Apr '09

16 Apr '09

It's standard port 5060 UDP at both ends. I'll enable tracing on the external profile as well - but I'm quite sure it's not used at all in this case. I'm on my way to the lab now, so I'll soon get back with my results after using latest SVN revision (my 'old' revision was just about 2 days old though). Peter ________________________________ Från: Brian West <brian(a)freeswitch.org> Skickat: den 15 april 2009 23:29 Till: freeswitch-users(a)lists.freeswitch.org <freeswitch-users(a)lists.freeswitch.org> Ämne: Re: [Freeswitch-users] RE FS/Sofia not sending bye when FreeSWITCH nds the call? What port are you hitting? Make sure you turn sip tracing on external and internal just in case you're using either or both. /b On Apr 15, 2009, at 4:12 PM, Peter Olsson wrote: I've built using latest trunk now, but I won't be able to test again until tomorrow - I'll get back to you after that. Just to make the scenario a bit more clear; The Avaya CM has an internal SIP-trunk over tls, to an Avaya SES Server (SIP Enablement Services), this one talks UDP to FreeSWITCH. Could this be something that causes the problem? I also tried to dial into the dialplan, answer the call, and then try to deflect the call using REFER. This didn't create any SIP messages either (and nothing happened with the call), so it seems there might be a bigger issue than just BYE. Peter Brian West brian(a)freeswitch.org<mailto:brian@freeswitch.org> -- Meet us at ClueCon! http://www.cluecon.com<http://www.cluecon.com/> !DSPAM:49e651b332933023977319!

1 0

I can not receive inbound calls thru FreeSWITCH
by winday 16 Apr '09

16 Apr '09

I'm a fresh user of FreeSWITCH. I just configed the basic things. Right now, all outbound calls are ok. I can call any number I want. But when I use my own mobile phone to call the phone number(I got from my sip provider) bind with my sip account, I can not receive the call. I configed that the inbound calls are all forwarded to extension 1001. But I got the following message: 2009-04-15 22:57:42 [NOTICE] switch_channel.c:592 switch_channel_set_name() New Channel sofia/external/3521111111(a)8.17.37.23 [5a5e70c2-2a32-11de-865a-c350fb010f87] 2009-04-15 22:57:42 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing Cell Phone FL->winday in context public 2009-04-15 22:57:42 [NOTICE] switch_core_state_machine.c:172 switch_core_standard_on_execute() Hangup sofia/external/3521111111(a)8.17.37.23 [CS_EXECUTE] [NORMAL_CLEARING] 2009-04-15 22:57:42 [NOTICE] switch_core_session.c:1019 switch_core_session_thread() Session 1 (sofia/external/3528713122(a)8.17.37.23) Ended 2009-04-15 22:57:42 [NOTICE] switch_core_session.c:1021 switch_core_session_thread() Close Channel sofia/external/3528713122(a)8.17.37.23 [CS_DONE] I don't know why. The public.xml configration is: <extension name="*****">  <condition field="destination_number" expression="*************">  <action application="transfer" data="$1 XML default"/> </condition> </extension> The default.xml configration is: <extension name="Inbound"> <condition field="destination_number" expression="^***********$"> <action application="ring_ready" /> <action application="set" data="call_timeout=20"/> <action application="bridge" data="sofia/sip/1001@my ip"/> <action application="javascript" data="/usr/local/freeswitch/scripts/answermachine.js"/> </condition> </extension> -- View this message in context: http://www.nabble.com/I-can-not-receive-inbound-calls-thru-FreeSWITCH-tp230… Sent from the Freeswitch-users mailing list archive at Nabble.com.

3 4

My softphone with extension 1001 doesn't ring when receive inbound calls
by Jing Qin 16 Apr '09

16 Apr '09

I'm a fresh user of FreeSWITCH. I just configed the basic things. Right now, all outbound calls are ok. I can call any number I want. But when I use my own mobile phone to call the phone number(I got from my sip provider) bind with my sip account, I can not receive the call. My softphone on my another computer doesn't ring at all. I configed that the inbound calls are all forwarded to extension 1001. And my softphone on my another computer is configed to use extension 1001. But I got the following message on my freeswitch server: 2009-04-15 22:57:42 [NOTICE] switch_channel.c:592 switch_channel_set_name() New Channel sofia/external/3521111111(a)8.17.37.23 [5a5e70c2-2a32-11de-865a-c350fb010f87] 2009-04-15 22:57:42 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing Cell Phone FL->winday in context public 2009-04-15 22:57:42 [NOTICE] switch_core_state_machine.c:172 switch_core_standard_on_execute() Hangup sofia/external/3521111111(a)8.17.37.23 [CS_EXECUTE] [NORMAL_CLEARING] 2009-04-15 22:57:42 [NOTICE] switch_core_session.c:1019 switch_core_session_thread() Session 1 (sofia/external/3528713122(a)8.17.37.23) Ended 2009-04-15 22:57:42 [NOTICE] switch_core_session.c:1021 switch_core_session_thread() Close Channel sofia/external/3528713122(a)8.17.37.23 [CS_DONE] I don't know why. The public.xml configration is: <extension name="*****">  <condition field="destination_number" expression="*************">  <action application="transfer" data="$1 XML default"/> </condition> </extension> The default.xml configration is: <extension name="Inbound"> <condition field="destination_number" expression="^***********$"> <action application="ring_ready" /> <action application="set" data="call_timeout=20"/> <action application="bridge" data="sofia/sip/1001@my ip"/> <action application="javascript" data="/usr/local/freeswitch/scripts/answermachine.js"/> </condition> </extension>

2 1

sbc/proxy setup
by Loren Salsgiver 16 Apr '09

16 Apr '09

Advice/Help needed... I'm looking to use freeswitch to sbc or possibly proxy connections to my calix ONTs. Currently the ONTs have public ips that talk to a Metaswitch. But the sip clients occassionally get a "sip" "sip" ring only message. They appear to be an exploit attempt to gain the passwords to make free calls. I want to setup the sip clients on invalid addresses and pass all the traffic thru freeswitch to avoid outside contact directly to the sip clients. Is this more sbc or proxy? Can freeswith pass the user auth from the sip clients directly to the Metaswitch or do I have to have individual accounts setup on the freeswith to do this? Details on where to set this up and how would be helpful. Thanks for the newbie help. Loren

1 0