April 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

auto-stack sizing on linux appears to break voicemail email notification
by Nathan Neulinger 17 Apr '13

17 Apr '13

It seems to cause the child processes to inherit the small stack size, which results in sendmail segfaulting. Haven't dug through this yet. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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Re: [Freeswitch-users] PRI channels suspended
by Akash Deep Verma 17 Apr '13

17 Apr '13

Hi Ashish, PFA, the compiled file which I used. Thanks & Regards, - Akash Deep Verma On Wed, Apr 17, 2013 at 1:14 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: PRI channels suspended (Ashish gautam) > 2. Re: Respect the request URI domain and To URI domain when > bridging (Jos? Luis Mill?n) > > > ---------- Forwarded message ---------- > From: Ashish gautam <ashish(a)nms.co.in> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 17 Apr 2013 12:33:23 +0530 > Subject: Re: [Freeswitch-users] PRI channels suspended > Hi Akash, > > I made these changes to the source and recompiled. Still the same this > happening. Channels 5 to 30 are suspended. > > Thanks. > > On Tue, Apr 16, 2013 at 4:39 PM, Akash Deep Verma <akashdeep.co(a)gmail.com>wrote: > >> Hi Ashish, >> >> This is a hack but you get what you need. >> >> This is because, whenever retry attempt is made 1 channel per pri get >> started, as there are 30 channels and retry attempts are 3 so the channels >> that freeswitch load is in between 3-6, so I increased retry attempt and >> complied it again. As I got rid of this error but it takes some time to >> load the pri channels (30*10 = 300seconds(5 seconds) max). >> >> Also, remember if you dial a call in between restarting phase you may >> face some error. To avoid this, dial call only after all pri channel get >> started. >> >> Here are the steps: >> >> In freeswitch source: >> Edit libs/freetdm/src/ftmod/ftmod_libpri/ftmod_libpri.h >> >> 1. Decrease timeout of channels. >> #define T316_TIMEOUT_MS_DEFAULT 30000 /* 30 sec */ >> >> change this to >> >> #define T316_TIMEOUT_MS_DEFAULT 10000 /* 10 sec */ >> >> 2. Increase attempts to 30. >> #define T316_ATTEMPT_LIMIT_DEFAULT 3 >> >> change this to >> >> #define T316_ATTEMPT_LIMIT_DEFAULT 30 >> >> 3. Increase maximum limit as according to default(set more than that of >> default) >> #define T316_ATTEMPT_LIMIT_MAX 10 >> >> change this to >> >> #define T316_ATTEMPT_LIMIT_MAX 35 >> >> Thanks & Regards, >> - >> Akash Deep Verma >> http://in.linkedin.com/in/akashdeep1 >> >> _________________________________________________________________________ >> 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: "José Luis Millán" <jmillan(a)aliax.net> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 17 Apr 2013 09:43:31 +0200 > Subject: Re: [Freeswitch-users] Respect the request URI domain and To URI > domain when bridging > Hi, > > > 2013/4/16 João Mesquita <jmesquita(a)freeswitch.org> > >> Ain't that what fs_path is for? I am not sure, maybe someone can chime-in >> here... >> > > Yes, 'fs_path' is the directive I was looking for. It specifies to SIP > proxy to which the message is being routed. > > Thank you. > > >> >> João Mesquita >> FreeSWITCH™ Solutions >> >> >> On Tue, Apr 16, 2013 at 5:10 PM, José Luis Millán <jmillan(a)aliax.net>wrote: >> >>> Hi, >>> >>> It was easy to define the sip_invite_full_to for legB given the >>> information exposed. I'm still unable to send the SIP request to a >>> different host that the one specified in the request URI. >>> >>> Can I send a request to a host that differs from the request URI domain >>> and To header URI domain of legB? If it is possible, any reference to the >>> doc will be much appreciated. >>> >>> Regards. >>> >>> >>> 2013/4/16 Michael Collins <msc(a)freeswitch.org> >>> >>>> For the most part I'd say yes. I think just having them on the wiki >>>> will be useful, even if it's just the self-explanatory part. >>>> -MC >>>> >>>> >>>> On Tue, Apr 16, 2013 at 11:56 AM, João Mesquita < >>>> jmesquita(a)freeswitch.org> wrote: >>>> >>>>> MC, will do, although they are pretty self-explanatory, ain't it? >>>>> >>>>> João Mesquita >>>>> FreeSWITCH™ Solutions >>>>> >>>>> >>>>> On Tue, Apr 16, 2013 at 2:29 PM, Michael Collins <msc(a)freeswitch.org>wrote: >>>>> >>>>>> J, >>>>>> >>>>>> Thanks for this information. It looks like none of these are on the >>>>>> wiki. Any volunteers out there willing to add these? Even if you don't know >>>>>> what they do you could add placeholders for them. See >>>>>> http://wiki.freeswitch.org/wiki/Variable_skeleton for details. >>>>>> >>>>>> Thanks, >>>>>> MC >>>>>> >>>>>> >>>>>> On Tue, Apr 16, 2013 at 9:32 AM, João Mesquita < >>>>>> jmesquita(a)freeswitch.org> wrote: >>>>>> >>>>>>> From sofia_glue.c: >>>>>>> >>>>>>> const char *invite_record_route = >>>>>>> switch_channel_get_variable(tech_pvt->channel, "sip_invite_record_route"); >>>>>>> const char *invite_route_uri = >>>>>>> switch_channel_get_variable(tech_pvt->channel, "sip_invite_route_uri"); >>>>>>> const char *invite_full_from = >>>>>>> switch_channel_get_variable(tech_pvt->channel, "sip_invite_full_from"); >>>>>>> const char *invite_full_to = >>>>>>> switch_channel_get_variable(tech_pvt->channel, "sip_invite_full_to"); >>>>>>> const char *handle_full_from = >>>>>>> switch_channel_get_variable(tech_pvt->channel, "sip_handle_full_from"); >>>>>>> const char *handle_full_to = >>>>>>> switch_channel_get_variable(tech_pvt->channel, "sip_handle_full_to"); >>>>>>> const char *force_full_from = >>>>>>> switch_channel_get_variable(tech_pvt->channel, "sip_force_full_from"); >>>>>>> const char *force_full_to = >>>>>>> switch_channel_get_variable(tech_pvt->channel, "sip_force_full_to"); >>>>>>> >>>>>>> >>>>>>> Glad to see you using FS José Luis! :-D Welcome. >>>>>>> >>>>>>> João Mesquita >>>>>>> >>>>>>> >>>>>>> On Tue, Apr 16, 2013 at 5:19 AM, José Luis Millán <jmillan(a)aliax.net >>>>>>> > wrote: >>>>>>> >>>>>>>> Let me fix an error: >>>>>>>> >>>>>>>> >>>>>>>> This is the message I would like to send to the IP 2.2.2.2 (my >>>>>>>> proxy) >>>>>>>> ``` >>>>>>>> INVITE sip:jmillan@1.2.3.4 SIP/2.0 Via: SIP/2.0/TCP >>>>>>>> bl23sl8lmku7.invalid;branch=z9hG4bK9420964 To: <sip:jmillan@1.2.3.4<sip%3Ad(a)1.2.3.4> >>>>>>>> >``` >>>>>>>> ``` >>>>>>>> >>>>>>>> Regards. >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> 2013/4/16 José Luis Millán <jmillan(a)aliax.net> >>>>>>>> >>>>>>>>> Hi all, >>>>>>>>> >>>>>>>>> Every call bridged though a specific 'gateway' does rewrite the >>>>>>>>> SIP request URI domain and To URI domain values to the gateway's 'proxy' >>>>>>>>> parameter value. >>>>>>>>> >>>>>>>>> I would like to respect such values and receive in my proxy the >>>>>>>>> SIP messages with their original request URI domain and To URI domain. >>>>>>>>> >>>>>>>>> This is, I would like to be able to send a SIP message with the >>>>>>>>> following request URI and To headers, to the IP address 2.2.2.2 >>>>>>>>> >>>>>>>>> ``` >>>>>>>>> INVITE sip:jmillan@1.2.3.4 SIP/2.0 Via: SIP/2.0/TCP >>>>>>>>> bl23sl8lmku7.invalid;branch=z9hG4bK9420964 To: <sip:d@1.2.3.4>``` >>>>>>>>> ``` >>>>>>>>> >>>>>>>>> Is it possible to handle this with FreeSwitch? >>>>>>>>> >>>>>>>>> I have followed the FS wiki but I've been unable to handle this. >>>>>>>>> >>>>>>>>> Thanks in advance. >>>>>>>>> >>>>>>>>> -- >>>>>>>>> José Luis Millán >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> José Luis Millán >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> 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 >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Michael S Collins >>>>>> Twitter: @mercutioviz >>>>>> http://www.FreeSWITCH.org >>>>>> http://www.ClueCon.com >>>>>> http://www.OSTAG.org >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting(a)freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users(a)lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting(a)freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>> http://www.cudatel.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users(a)lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.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 >>>> >>>> >>> >>> >>> -- >>> José Luis Millán >>> >>> _________________________________________________________________________ >>> 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 >> >> > > > -- > José Luis Millán > > _______________________________________________ > 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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sofia profile determination
by Jimmy Chang 17 Apr '13

17 Apr '13

Hi, I have defined two SIP profile, external and internal. The external profile listen port 5080, and the internal profile listen port 5060. When a user in the same domain with FS dials telephone number using port 5080, FS determines to use internal profile. I want to use external profile when user dials port 5080 no matter where he or she is. Is it possible and what to do? Thanks. Jimmy

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Re: [Freeswitch-users] PRI channels suspended
by Akash Deep Verma 17 Apr '13

17 Apr '13

Hi Ashish, This is a hack but you get what you need. This is because, whenever retry attempt is made 1 channel per pri get started, as there are 30 channels and retry attempts are 3 so the channels that freeswitch load is in between 3-6, so I increased retry attempt and complied it again. As I got rid of this error but it takes some time to load the pri channels (30*10 = 300seconds(5 seconds) max). Also, remember if you dial a call in between restarting phase you may face some error. To avoid this, dial call only after all pri channel get started. Here are the steps: In freeswitch source: Edit libs/freetdm/src/ftmod/ftmod_libpri/ftmod_libpri.h 1. Decrease timeout of channels. #define T316_TIMEOUT_MS_DEFAULT 30000 /* 30 sec */ change this to #define T316_TIMEOUT_MS_DEFAULT 10000 /* 10 sec */ 2. Increase attempts to 30. #define T316_ATTEMPT_LIMIT_DEFAULT 3 change this to #define T316_ATTEMPT_LIMIT_DEFAULT 30 3. Increase maximum limit as according to default(set more than that of default) #define T316_ATTEMPT_LIMIT_MAX 10 change this to #define T316_ATTEMPT_LIMIT_MAX 35 Thanks & Regards, - Akash Deep Verma http://in.linkedin.com/in/akashdeep1

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plan break
by Jimmy Chang 17 Apr '13

17 Apr '13

Hi, I have a chatplan like this: <?xml version="1.0" encoding="utf-8"?> <include> <context name="default"> <extension name="testnumber"> <condition field="to" expression="^testnumber"/> <condition field="${cond(${_body} < 10 ? YES : NO)}" expression="^YES$"> <action application="reply" data="OK"/> </condition> </extension> <extension name="demo"> <condition field="to" expression="^(.*)$"> <action application="send" /> </condition> </extension> </context> </include> No matter what result the {cond(${_body} is, I don't want the plan flow go down to demo extension. What should I do? Thanks. Jimmy

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Testing WebRTC with JsSIP and OverSIP
by Henry Huang 17 Apr '13

17 Apr '13

Hi, Like the rest of the world, I am crazy about getting webrtc to work. Since FS does not yet do websocket. I have used JsSIP on Chrome and OverSIP as the sip proxy for websocket. I was able to call from Chrome with extension 1001 to another regular extension 1004 on FS with 2 way audio and dtmf working fine. The media is transporting directly between Chrome and FS. That is awesome. But when I try to call ext. 1001(JsSIP on Chrome) back from 1004. I am getting the following error from the javascript console, but I believe it's just reporting what the Chrome is not happy about. And if I try to pick up the call on Chrome, it would exit that session and kept the calling ringing. 1. JsSIP | RTC SESSION | unable to set local description jssip-devel.js:3313 <http://jssip.net/download/jssip-devel.js> 1. Failed to parse SessionDescription. m=audio 1 RTP/SAVPF Expects at least 4 fields. I have checked specifically about the SAVPF field I am sending from FS, and it has 4 field if I am not mistaken: INVITE sip:id2uaa1s@ehg4bjsdusqf.invalid;transport=ws SIP/2.0 Record-Route: <sip:08b1bd9007@72.1.46.750ddc3f-2196-1231-13a0-5254004b917b :10080;transport=ws;lr;ovid=f96ad63d> Record-Route: <sip:_NzIuMS40Ni4xMjJfNTA2MA--@oversip-public-ip :5060;transport=udp;lr;ovid=f96ad63d> Via: SIP/2.0/WS oversip-public-ip:10080;branch=z9hG4bK2b52650d8dfffc0bec5baf62029018cbb31b2e6c;rport Via: SIP/2.0/UDP freeswich-public-ip;branch=z9hG4bKme6eNg3jBvHpm;received=freeswich-public-ip;rport=5060 Max-Forwards: 10 From: "Extension 1004" <sip:1004@freeswich-public-ip>;tag=2SNgp6mXyc0Nr To: <sip:id2uaa1s@ehg4bjsdusqf.invalid;transport=ws;ov-ob=08b1bd9007> Call-ID: 750ddc3f-2196-1231-13a0-5254004b917b CSeq: 42742737 INVITE Contact: <sip:mod_sofia@freeswich-public-ip:5060> User-Agent: FreeSWITCH-mod_sofia/1.5.1b+git~20130408T160028Z~d50d43f788 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 886 X-Fs-Support: update_display,send_info Remote-Party-Id: "Extension 1004" <sip:1004@freeswich-public-ip >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1366127964 1366127965 IN IP4 freeswich-public-ip s=FreeSWITCH c=IN IP4 freeswich-public-ip t=0 0 a=msid-semantic: WMS XbFAafgwqma0yucP2swUgWG29xewft33 *m=audio 29638 RTP/SAVPF 0 8 3 13* a=fingerprint:sha-256 3A:E4:E2:49:05:98:07:F6:A4:50:EA:8E:E8:04:C7:85:9E:AF:9E:42:E0:42:D2:17:96:EA:59:B6:FA:17:C4:24 a=rtcp-mux a=rtcp:29638 IN IP4 freeswich-public-ip a=ssrc:3312676386 cname:vzuvEASUNTLceOS7 a=ssrc:3312676386 msid:XbFAafgwqma0yucP2swUgWG29xewft33 a0 a=ssrc:3312676386 mslabel:XbFAafgwqma0yucP2swUgWG29xewft33 a=ssrc:3312676386 label:XbFAafgwqma0yucP2swUgWG29xewft33a0 a=ice-ufrag:6eSd1o51L9nRthLO a=ice-pwd:G5abwTWaDycHcUF9 a=candidate:7283942946 1 udp 659136 freeswich-public-ip 29638 typ host generation 0 a=candidate:7283942946 2 udp 659134 freeswich-public-ip 29638 typ host generation 0 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:p0HhxqXHnO/K86u3Qu1StqefUKtbbGVKA+Y8KSj1 a=ptime:20 At this point, I don't know how to troubleshoot further. I am posting this here to see if there is anyone running into similar issue and have a solution to this? Thanks, Henry

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Javadoc for esl.jar
by Marwan Idriss 16 Apr '13

16 Apr '13

Greeting, Is their any Javadoc for esl.jar ? Regards Marwan

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Re: [Freeswitch-users] IVR, bridge, ring back problem
by andpe 16 Apr '13

16 Apr '13

Dziękuję bardzo, wszystko działa tak jak opisałeś. Pozdrawiam W dniu 2013-04-16 07:29:46 użytkownik Michael Collins <msc(a)freeswitch.org> napisał: I believe it's the space and not the colon. However, I don't personally recommend that you put everything into the IVR like that. Instead, try putting all those set apps and the bridge app into a separate XML dialplan extension and then use the transfer app to send the caller there. See the demo IVR in conf/ivr_menus/ for examples on how to do that. -MC On Fri, Apr 12, 2013 at 10:12 PM, Andrzej <andpe(a)poczta.onet.pl> wrote: I'll check on Monday. but: 1. Is this normal behavior IVR? 2. How it should be written: a) <entry action="menu-exec-app" digits="1" param="set ringback=$${uk-ring}"/> <entry action="menu-exec-app" digits="1" param="set instant_ringback=true"/> <entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/xxx"/> or b) <entry action="menu-exec-app" digits="1" param="set:ringback=$${uk-ring}"/> <entry action="menu-exec-app" digits="1" param="set:instant_ringback=true"/> <entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/xxx"/> look at the colon after "set". Regards, W dniu 12.04.2013 18:25, Paul Cupis pisze: > On Fri, Apr 12, 2013 at 02:07:45PM +0200, andpe wrote: >> I have a problem with the IVR in FreeSwitch and function bridge. >> >> How it works: >> customer calls IVR, IVR announses >> customer chooses button 1 >> number 1 uses function bridge to connect to, say, 1001. >> during the bridge establishes a connection user (calling party) can not hear the ring signal (silent in handset). >> when dialed user pick up handset then all works ok (we can talk). Also when dialed user reject the connection I hear busy signal. >> I have seen in logs Ring Ready not send to calling party. >> >> When user calls directly 1001 all works fine. > Do something like this before the bridge? > > <entry action="menu-exec-app" digits="1" param="set ringback=$${uk-ring}"/> > <entry action="menu-exec-app" digits="1" param="set instant_ringback=true"/> > > Regards, > > > _________________________________________________________________________ > 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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 82, Issue 133
by Akash Deep Verma 16 Apr '13

16 Apr '13

Hi Ashish, This is a hack but you can get what you need. This is because, whenever retry attempt is made 1 channel per pri get started, as there are 30 channels and retry attempts are 3 so the channels that freeswitch load is in between 3-6, so I increased retry attempt and complied it again. As I got rid of this error but it takes some time to load the pri channels (30*10 = 300seconds(5 seconds) max). Also, remember if you dial a call in between restarting phase you may face some error. To avoid this, dial call only after all pri channel get started. Here are the steps: In freeswitch source: Edit libs/freetdm/src/ftmod/ftmod_libpri/ftmod_libpri.h 1. Decrease timeout of channels. #define T316_TIMEOUT_MS_DEFAULT 30000 /* 30 sec */ change this to #define T316_TIMEOUT_MS_DEFAULT 10000 /* 10 sec */ 2. Increase attempts to 30. #define T316_ATTEMPT_LIMIT_DEFAULT 3 change this to #define T316_ATTEMPT_LIMIT_DEFAULT 30 3. Increase maximum limit as according to default(set more than that of default) #define T316_ATTEMPT_LIMIT_MAX 10 change this to #define T316_ATTEMPT_LIMIT_MAX 35 Thanks & Regards, - Akash Deep Verma http://in.linkedin.com/in/akashdeep1 On Tue, Apr 16, 2013 at 4:13 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: PRI channels suspended (Ashish gautam) > 2. Re: IVR, bridge, ring back problem (Paul Cupis) > 3. Re: Newbie question about inbound calls/NAT (yes I checked > wiki first:-) ) (Cal Leeming [Simplicity Media Ltd]) > > > ---------- Forwarded message ---------- > From: Ashish gautam <ashish(a)nms.co.in> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 16 Apr 2013 14:46:58 +0530 > Subject: Re: [Freeswitch-users] PRI channels suspended > 2013-04-16 14:41:04.214291 [ERR] ftmod_libpri.c:1950 [s1c30][1:30] -- T316 > timed out, channel reached restart attempt limit '3' and is suspended > > This is what happening on restarting/starting the FS for all the > channels(this is only for channel number 30). > > How to resolve it? Please help. > > -- > Regards. > > On Tue, Apr 16, 2013 at 2:46 PM, Ashish gautam <ashish(a)nms.co.in> wrote: > >> Sorry Michael mistakenly sent to you personally again. Sending to the >> list once more. >> >> >> On Tue, Apr 16, 2013 at 2:45 PM, Ashish gautam <ashish(a)nms.co.in> wrote: >> >>> 2013-04-16 14:41:04.214291 [ERR] ftmod_libpri.c:1950 [s1c30][1:30] -- >>> T316 timed out, channel reached restart attempt limit '3' and is suspended >>> >>> This is what happening on restarting/starting the FS for all the >>> channels(this is only for channel number 30). >>> >>> How to resolve it? Please help. >>> >>> -- >>> Regards. >>> >>> On Mon, Apr 15, 2013 at 9:51 PM, Michael Collins <msc(a)freeswitch.org>wrote: >>> >>>> Please only send this to the list, otherwise it is nearly impossible >>>> for me to reply in such a way that everyone on the list can join the >>>> conversation. >>>> -MC >>>> >>>> >>>> On Sun, Apr 14, 2013 at 11:29 PM, Ashish gautam <ashish(a)nms.co.in>wrote: >>>> >>>>> Hi, >>>>> >>>>> I am having another problem over here. When I run "ftdm dump 1 5" from >>>>> fs_cli I get my channel status as SUSPENDED. This happens for channel >>>>> number 5 to 30. Also when I try to make outgoing calls more than 4 in >>>>> number, four calls originate properly whereas for the rest it shows >>>>> NORMAL_CIRCUIT_CONGESTION. Here is the output: >>>>> >>>>> span_id: 1 >>>>> chan_id: 5 >>>>> physical_span_id: 1 >>>>> physical_chan_id: 5 >>>>> physical_status: ok >>>>> physical_status_red: 0 >>>>> physical_status_yellow: 0 >>>>> physical_status_rai: 0 >>>>> physical_status_blue: 0 >>>>> physical_status_ais: 0 >>>>> physical_status_general: 0 >>>>> signaling_status: UP >>>>> type: B >>>>> state: SUSPENDED >>>>> last_state: RESTART >>>>> txgain: 0.00 >>>>> rxgain: 0.00 >>>>> cid_date: >>>>> cid_name: >>>>> cid_num: >>>>> ani: >>>>> aniII: >>>>> dnis: >>>>> rdnis: >>>>> cause: NORMAL_UNSPECIFIED >>>>> session: (none) >>>>> >>>>> -- States -- -- Function -- -- >>>>> Location -- -- Time Offset -- >>>>> DOWN => RESTART [on_dchan_up] >>>>> [ftmod_libpri.c:2372] 0ms >>>>> RESTART => SUSPENDED [on_timeout_t316] >>>>> [ftmod_libpri.c:1952] 120008ms >>>>> >>>>> Time since last state change: 324790932ms >>>>> >>>>> Please help me getting out of this. >>>>> >>>>> Thanks. >>>>> >>>>> --Ashish >>>>> >>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> >>>> >>> >> > > > ---------- Forwarded message ---------- > From: Paul Cupis <paul(a)cupis.co.uk> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 16 Apr 2013 10:54:32 +0100 > Subject: Re: [Freeswitch-users] IVR, bridge, ring back problem > On Sat, Apr 13, 2013 at 07:12:58AM +0200, Andrzej wrote: > > I'll check on Monday. > > but: > > 1. Is this normal behavior IVR? > > This is normal behavior, yes - the call has already been answered, so the > caller is not going to be sent 180 Ringing or 183 Early media in order to > hear ringing during the bridge - you need to provide audio to fill the > silence. > > > 2. How it should be written: > > a. with a space, not a colon. > > Regards, > > > > > > ---------- Forwarded message ---------- > From: "Cal Leeming [Simplicity Media Ltd]" < > cal.leeming(a)simplicitymedialtd.co.uk> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 16 Apr 2013 11:43:15 +0100 > Subject: Re: [Freeswitch-users] Newbie question about inbound calls/NAT > (yes I checked wiki first:-) ) > Glad to hear you got it sorted, and thank you for posting back your > finding - hopefully this will help others in the future! > > Cal > > On Tue, Apr 16, 2013 at 9:29 AM, Jeff Bernhardt <jeff(a)askcornerstone.net>wrote: > >> You know, I actually just figured it out myself just a couple hours >> ago.... but I'm gonna give your method a shot too since I haven't played >> with ACLs yet and that would give me a chance to. >> >> I discovered that the problem was actually how I'd setup Vitelity to >> route to the server. I didn't realize this before, but in Vitelity, you can >> either route to the server IP address to allow IP based authentication OR >> you can route to the account login name that the trunk is being registered >> with. The former was giving errors since I didn't have that account setup, >> whereas the latter worked with no problem. I'm guessing this is because >> with the IP based auth method, Vitelity connects to port 5060 (can't change >> this), which is taken as Freeswitch's default internal profile port >> (requires authentication) instead of port 5080 (the public profile with >> doesn't require authentication?), and since there was no internal user for >> what Vitelity was sending essentially unsolicited on 5060, it got denied. >> As to why it works by routing to the registered account login name I'm not >> sure, but perhaps that's part of what registration does; it sends to the >> public profile by default...? >> >> Anyway, thanks for all the help! Great to see such an active and >> helpful list with something I'll need a lot of help with! >> >> ------------------------------ >> *From:* freeswitch-users-bounces(a)lists.freeswitch.org [ >> freeswitch-users-bounces(a)lists.freeswitch.org] on behalf of Michael >> Collins [msc(a)freeswitch.org] >> *Sent:* Monday, April 15, 2013 7:15 PM >> >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Newbie question about inbound >> calls/NAT (yes I checked wiki first:-) ) >> >> For inbound calls you can skip the authentication by adding the source >> IP address to the "domains" section of conf/autoload_configs/acl.conf.xml. >> This will prevent FreeSWITCH from challenging the inbound call if it comes >> from that IP address. This is more desirable than disabling call >> authentication altogether in your SIP profile. (i.e. you are allowing only >> a specific IP address to come in unauth'd as opposed to letting the whole >> world in!) >> >> Find out the source IP address for your Vitelity SIP traffic and add it >> to the acl.conf.xml file in the "domains" section. It will be something >> like this: >> >> <node type="allow" cidr="1.2.3.4/32"/> >> >> After you save that go to fs_cli and type "reloadacl". NOTE: simply doing >> a "reloadxml" is not sufficient to make the new ACL entry take effect. >> ("reloadxml" simply reloads the XML config; "reloadacl" does a reloadxml >> and then reloads the ACL entries.) >> >> Hope this helps. >> -MC >> >> On Mon, Apr 15, 2013 at 12:30 AM, Jeff Bernhardt <jeff(a)askcornerstone.net >> > wrote: >> >>> Thanks. I had a look at the links and changed "domain and even >>> "domain_name" in vars.xml to be the external ip, but the error now just >>> says it can's find user abcd_1.2.3.4(a)1.2.3.4 instead of @192.168.10.32. >>> >>> >>> And actually, I 'm mostly confused because I thought incoming calls >>> were treated as unauthenticated and allowed to pass through. Isn't >>> registering the trunk with the SIP provider enough to get incoming calls >>> accepted? >>> >>> The errors says "...you must configure your device to use the proper >>> domain in it's authentication credentials," but how is the SIP trunk >>> treated as a device? >>> >>> Like I said, I think I'm misunderstanding something fundamental about >>> this. >>> >>> Thanks you. >>> >>> ------------------------------ >>> *From:* freeswitch-users-bounces(a)lists.freeswitch.org [ >>> freeswitch-users-bounces(a)lists.freeswitch.org] on behalf of Cal Leeming >>> [Simplicity Media Ltd] [cal.leeming(a)simplicitymedialtd.co.uk] >>> *Sent:* Sunday, April 14, 2013 3:38 PM >>> *To:* FreeSWITCH Users Help >>> *Subject:* Re: [Freeswitch-users] Newbie question about inbound >>> calls/NAT (yes I checked wiki first:-) ) >>> >>> Hello Jeff, >>> >>> Here is a related thread; >>> >>> http://freeswitch-users.2379917.n2.nabble.com/You-must-define-a-domain-call… >>> >>> http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/00623… >>> >>> You need to configure your domain and username, but as you can see >>> here your gateway/users are sending the domain '192.168.10.32', but you >>> have not configured this domain in sofia.conf. >>> >>> If you are using multi-tenant then you'll need to put some more >>> thought into how you're going to approach this. >>> >>> If you are not using multi-tenant (i.e. one domain only) then you >>> could force the domain as discussed here; >>> >>> http://www.feedingedge.co.uk/blog/2010/08/03/techie-post-opensim-and-freesw… >>> >>> Hope this helps >>> >>> Cal >>> >>> On Sun, Apr 14, 2013 at 10:36 AM, Jeff Bernhardt < >>> jeff(a)askcornerstone.net> wrote: >>> >>>> Setting up my first test box with Freeswitch. I'm really excited >>>> about getting it up and running. >>>> >>>> I'm trying to get inbound calls working from external using a >>>> Vitelity trunk. Let's say my username from Vitelity is "abcd_efg" and >>>> password is "password," external IP is 1.2.3.4 and internal is >>>> 192.168.10.32. I get registered to Vitelity fine (their web panel shows >>>> I'm registered and fs_cli confirms). However, when I make a call inbound, I >>>> get: >>>> >>>> [WARNING] sofia_reg.c:2621 Can't find user [abcd_1.2.3.4(a)192.168.10.32] >>>> from <vitelity-inbound-trunk-ip> >>>> You must define a domain called '192.168.10.32' in your directory and >>>> add a user with the id="abcd_1.2.3.4" attribute >>>> and you must configure your device to use the proper domain in it's >>>> authentication credentials. >>>> >>>> I don't know if this is related, but to get around NAT, I have this >>>> in vars.xml: >>>> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=1.2.3.4"/> >>>> <X-PRE-PROCESS cmd="set" data="external_sip_ip=1.2.3.4"/> >>>> >>>> and this in both internal.xml and external.xml: >>>> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> >>>> <param name="ext-sip-ip" value="$${external_sip_ip}"/> >>>> >>>> I have a feeling I've misunderstood something fundamental about >>>> setting up Freeswitch for inbound access. I've put in the above configs as >>>> per what I've found in the wiki, but I'm not sure how to do what the >>>> warning says or if it really means I have something not setup right. What >>>> am I doing wrong? >>>> >>>> Thank you. >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.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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ringback and transfer_ringback ignored by FS
by Adam Raszynski 16 Apr '13

16 Apr '13

Hi Scenario: - SIP only - inbound call, hits dialplan - call answered, some IVR used - call bridged to selected department The problem is, that I would like to play music on hold while trying to bridge. FS seems to completely ignore custom ringback settings: <action application="set" data="ringback=myfile.wav"/> <action application="set" data="transfer_ringback=myfile.wav"/> <action application="bridge" data="{ignore_early_media=true}sofia/internal/x@y" /> tried also with: <action application="bridge" data="{ignore_early_media=true,transfer_ringback=myfile.wav,ringback=myfile.wav)}sofia/internal/x@y" /> and <action application="set" data="ringback=myfile.wav"/> <action application="set" data="transfer_ringback=myfile.wav"/> <action application="ring_ready" /> <action application="bridge" data="{ignore_early_media=true}sofia/internal/x@y" /> All with no success. I see in console that variables are set, but for some reason FS ignores them and there is no audio while bridging. At the same time I can play/stream sound file before bridge, but this operation is blocking and stream can not be run in background so it can't be used for MoH purpose. Any hints what should I check or how to achive nonblocking MoH?

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