- FreeSWITCH-users - FreeSWITCH Mailing Lists

Skypiax, Skype endpoint and trunk, robustness patch
by Giovanni Maruzzelli 28 Aug '09

28 Aug '09

Hi FreeSWITCHers, all the users of mod_skypiax are kindly requested to test the svn trunk 14519. It contains a lot of changes meant to add stability and robustness, toward a production environment. Let me know how your feelings, and please add to the Jira any possible bug/issue/etc. Thanks to you all, -giovanni Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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Questions about att_xfer
by Anatoliy Kounitskiy 28 Aug '09

28 Aug '09

Hello everybody! I have few questions about the att_xfer application. First, what i want to accomplish is: user A calls user B, after that user B makes attended transfer to user C. In the dialplan i have: <context name="vpbx"> <extension name="local_number"> ... <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="bind_meta_app" data="4 b s execute_extension::attented_xfer XML features"/> .... </condition> </extension> So when user B answers the call, he sends *4 and the extensions for the attended transfer is started - the usual - plays message and read the input dtmf: features.xml ... <extension name="attented_xfer"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^attented_xfer$"> <action application="info"/> <action application="read" data="3 4 ivr/ivr-enter_ext.wav attxfer_callthis 30000 #"/> <action application="set" data="call_timeout=15"/> <action application="att_xfer" data="user/${attxfer_callthis}@${domain_name}"/> </condition> </extension> ... To this problems everything is perfect. But here comes the questions, so if you can give some tips would be great. 1) when user B enters the extension number of C - the C's phone starts ringing in the tcpdump i can see that the phone is sending 180 ringing, BUT user B does not hear the ringing. 2) as mentioned in the http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer quote: "If the other leg is a voicemail or doesn't answered you can hangup that leg by pressing dtmf # (fixed in r14438) " It doesn't work. The option 0 is working even before C answering the phone - after he answers it's a threeway conference :) - i like this feature. I'm using FreeSWITCH Version 1.0.trunk (14633M) Also I tried to set call timeout to see if I can go back the user A, who is listening to MOH - no luck here. Probably I'm missing something. Tried to look in the source of att_xfer to understand why the feature i want is not working - but it seems my C/C++ skills are not so good, as i want :( . Thank you in advance, Anatoliy Kounitskiy

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timeout while waiting for CF_BRIDGED flag
by Tamas Cseke 28 Aug '09

28 Aug '09

Hello, we have a strange problem, sometimes calls are hung up with DESTINATION_OUT_OF_ORDER cause. I added some extra log lines, and it turned out that switch_ivr_bridge.c:140 a timeout occurs while waiting for CF_BRIDGED flag switch_channel_wait_for_flag(chan_b, CF_BRIDGED, SWITCH_TRUE, 10000, chan_a); returns SWITCH_STATUS_TIMEOUT As far as I understand the flag on B channel is set in another thread, so maybe it happens slow sometimes. We are using r14144. I know it is quiet old, but maybe this code hasn't changed since them Do you have any idea, what could cause this? Do you think it is a bug? Thanks in advance, Tamas

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Who is "callie"?
by Carlos S. Antunes 28 Aug '09

28 Aug '09

Hi! I searched the wiki but couldn't find the answer. Is "callie" a real woman one may ask to record additional sounds? Thanks! Carlos

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Authorizations when using DNS SRV bug?
by Carlos S. Antunes 28 Aug '09

28 Aug '09

Hello! I am using Callcentric for my tests and have observed what appears to me a possible bug in the way Freeswitch handles DNS SRV records. Callcentric uses DNS SRV records as a way to direct traffic to their SIP server. A 'srv' 'dig' of '_sip._udp.callcentric.com' returns: _sip._udp.callcentric.com. 10025 IN SRV 20 7 5080 alpha6.callcentric.com. _sip._udp.callcentric.com. 10025 IN SRV 20 7 5080 alpha7.callcentric.com. _sip._udp.callcentric.com. 10025 IN SRV 20 7 5080 alpha1.callcentric.com. _sip._udp.callcentric.com. 10025 IN SRV 20 7 5080 alpha3.callcentric.com. Based on this information, Freeswitch appears to correctly round robin all available IP addresses except in a particular situation: in the middle of authorizations. For example, in a registration, Freeswitch send a packet to alpha1.callcentric.com. Callcentric then challenges Freeswitch with a Proxy Authorization request. Freeswitch then sends the packet with the requested credentials but not necessarily to alpha1.callcentric.com! In many cases, instead of sticking to the 'challenging' server, Freeswitch round robins and sends the second packet to, one of the other servers. This continues for a little while and eventually, simply by luck, the second packet is sent to the 'challenging' Callcentric server. Shouldn't Freeswitch stick to the same server when challenged for credentials? Is this a bug? Is there a way to make Freeswitch behave differently? Thanks! Carlos Antunes Nowthor Corporation

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Re: [Freeswitch-users] How to make a call back
by Michael S Collins 28 Aug '09

28 Aug '09

Sent from my iPhone On Aug 27, 2009, at 10:01 PM, lakshmanan ganapathy <lakindia89(a)gmail.com> wrote: > No. In the dial plan I said, application="perl" data="The perl > script". > I also checked $session->execute("bridge","user/1010"). This is > working fine. > But originate is not working as I expected. > I think you might be confusing Dialplan apps with API commands. The $session object represents an existing channel and therefore it uses Dialplan apps. Originate is an API, that is, it is a command that you type at the CLI. You need an API object to use originate from a script: my $api = new $freeswitch::API(); my $res = $api->executeString("originate user/1010 &bridge(user/1000); What kind of application are you developing? I'm curious why you need an originate to create a whole new call. -MC > On Thu, Aug 27, 2009 at 9:46 PM, Michael Collins > <msc(a)freeswitch.org> wrote: > > > On Wed, Aug 26, 2009 at 9:38 PM, lakshmanan <lakindia89(a)gmail.com> > wrote: > > When I give the following from the command line it calls to 1010 > extension > and once answered, it calls to 1000 and bridge the connection. > originate user/1010 &bridge(user/1000) > But I want to do this in perl. So I have given as follows > $session->originate($session,"user/1010 &bridge user/ > 1000"); > But it is not working. It says "user/1010 &bridge user/1000 is invalid > user". > How to do this in perl. pls help. > > Are you calling this perl script from the CLI? If so you won't have > the $session object because a channel does not exist for a simple > API call. > -MC > > > _______________________________________________ > 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] freeswitch as SBC and kamailio - no route
by Hristo Benev 28 Aug '09

28 Aug '09

I assume you asked for port 5062 since I do not have any traffic on 5060 (I have one IP and my internal sip port is 5090 and external 5080). If you need additional info I'll provide it. Here is trace: ngrep -d any -nn -i '1000' port 5062 -W byline interface: any filter: (ip or ip6) and ( port 5062 ) match: 1000 # U 10.10.10.10:5090 -> 127.0.0.1:5062 INVITE sip:1000@127.0.0.1:5062 SIP/2.0. Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj. Max-Forwards: 69. From: "Extension 1001" <sip:1001@10.10.10.10>;tag=978g69jZaFpBD. To: <sip:1000@127.0.0.1:5062>. Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c. CSeq: 119574771 INVITE. Contact: <sip:mod_sofia@10.10.10.10:5090>. User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 429. X-ROUTE: LOOKUP. Remote-Party-ID: "Extension 1001" <sip:1001@10.10.10.10>;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1251355692 1251355693 IN IP4 10.10.10.10. s=FreeSWITCH. c=IN IP4 10.10.10.10. t=0 0. m=audio 30522 RTP/AVP 0 115 107 9 8 3 101 13. a=rtpmap:0 PCMU/8000. a=rtpmap:115 G7221/32000. a=fmtp:115 bitrate=48000. a=rtpmap:107 G7221/16000. a=fmtp:107 bitrate=32000. a=rtpmap:9 G722/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:13 CN/8000. a=ptime:20. # U 127.0.0.1:5062 -> 10.10.10.10:5090 SIP/2.0 302 PEER_01. Via: SIP/2.0/UDP 10.10.10.10:5090;rport=5090;branch=z9hG4bKtrDyU1US5X5tj. From: "Extension 1001" <sip:1001@10.10.10.10>;tag=978g69jZaFpBD. To: <sip:1000@127.0.0.1:5062>;tag=458fb4012080e656b6742c09466dabcd.31c8. Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c. CSeq: 119574771 INVITE. Contact: sip:France@PEER_01. Server: Kamailio (1.5.2-notls (i386/linux)). Content-Length: 0. . # U 10.10.10.10:5090 -> 127.0.0.1:5062 ACK sip:1000@127.0.0.1:5062 SIP/2.0. Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj. Max-Forwards: 69. From: "Extension 1001" <sip:1001@10.10.10.10>;tag=978g69jZaFpBD. To: <sip:1000@127.0.0.1:5062>;tag=458fb4012080e656b6742c09466dabcd.31c8. Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c. CSeq: 119574771 ACK. Content-Length: 0. . >-------- Оригинално писмо -------- >От: Hristo Benev >Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route >До: freeswitch-users(a)lists.freeswitch.org >Изпратено на: Четвъртък, 2009, Август 27 14:58:53 EEST > >Bojnour, > >I'll send a trace ASAP. > >What I see is that SIP header does not get updated -> regex is not true then it does not go to the peer. (I assume that is coming from kamailio config) > >I'm really interested to see the updates of the project. > >Thank you for the good tutorial. > >Hristo > > > >-------- Оригинално писмо -------- > >От: rod > >Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route > >До: freeswitch-users(a)lists.freeswitch.org > >Изпратено на: Четвъртък, 2009, Август 27 09:41:05 EEST > > >Hi Hristo, > > > >I'm the author of this setup and wiki page. I did a lot of modifications > >on this setup (alternative routing if failure essentially) but don't > >have too much time to update the wiki. > > > >May you please send me an ngrep trace when you call 1000: > > > >ngrep -d any -nn -i '1000' port 5060 -W byline > > > >I will check what's happening. > >Do you have an entry for 1000 in your mysql database ? > > > >regards, > >rod > > > >Hristo Benev a écrit : > >> > >> It seems that the problem is on kamailio configuration. > >> Will ask on their list. > >> > >> But to test i try to connect to my asterisk server and i receive 407 proxy authentication required. > >> > >> I have it setup as friend in asterisk, but still ??? > >> > >> Any ideas? > >> > >> Thanks, > >> > >> > >> >-------- Оригинално писмо -------- > >> >От: Hristo Benev > >> >Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route > >> >До: freeswitch-users(a)lists.freeswitch.org > >> >Изпратено на: Сряда, 2009, Август 26 22:02:13 EEST > >> > >> > I think that the problem is here: > >> >------------------------- > >> >2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING > >> >Dialplan: sofia/internal/1001(a)209.71.254.33 parsing [ROUTING->PEER_01] continue=false > >> >Dialplan: sofia/internal/1001(a)209.71.254.33 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false > >> >2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No Route, Aborting > >> >-------------------------- > >> > > >> >Actually Regex FAIL > >> > > >> >I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be PEER_01 for success? > >> >Here is my default.xml: > >> >---------------- > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> >-------------------------- > >> > > >> > > >> > > >> > >-------- Оригинално писмо -------- > >> > >От: Brian West > >> > >Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route > >> > >До: freeswitch-users(a)lists.freeswitch.org > >> > >Изпратено на: Сряда, 2009, Август 26 19:47:37 EEST > >> > > >> > >We do not blindly follow 302's as that is a dangerous thing to do. > >> >You have to process all 302's in the dialplan. > >> >Set this on your sofia profile > >> >You can set these variables sip_redirect_profile, > >> >sip_redirect_context, > >> >sip_redirect_dialplan, > >> >When a redirect happens you get these variables - sip_redirect_contact_%d, > >> >sip_redirected_to, > >> >sip_redirect_contact_user_%d, > >> >sip_redirect_contact_host_%d, > >> >sip_redirect_contact_params_%d, > >> >sip_redirect_dialstring_%d, > >> >sip_redirect_dialstring, > >> >sip_redirected_byThen its up to you to process the redirect in your dialplan, If you don't set the > >> >sip_redirect_context then it'll default to redirected context and XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not routeWhere to look for problems? > >> > > > >> > > >> >_______________________________________________ > >> >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 >

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How to make a call back
by lakshmanan 28 Aug '09

28 Aug '09

When I give the following from the command line it calls to 1010 extension and once answered, it calls to 1000 and bridge the connection. originate user/1010 &bridge(user/1000) But I want to do this in perl. So I have given as follows $session->originate($session,"user/1010 &bridge user/1000"); But it is not working. It says "user/1010 &bridge user/1000 is invalid user". How to do this in perl. pls help. -- View this message in context: http://www.nabble.com/How-to-make-a-call-back-tp25166083p25166083.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Using a "Virtual" Extension
by Tina Martinez 27 Aug '09

27 Aug '09

2 1

Mixing mod_curl_xml dynamic dialplans and static ones
by Greg Thoen 27 Aug '09

27 Aug '09

I have mod_curl_xml working, but I also have several static dialplans in dialplan/public/ and I can't seem to get it to search those if my php generated xml page does not return a result. I know in the static public.xml there is this <extension name="unloop"> <condition field="${unroll_loops}" expression="^true$"/> <condition field="${sip_looped_call}" expression="^true$"> <action application="deflect" data="${destination_number}"/> </condition> </extension> Do I need my php page to generate something like that so it will continue looking in the static pages if it does not find a match in the xml_curl dynamic dialplan? I don't know if I have given enough data for someone to point me in the right direction... This is in the console when I have mod_xml_curl enabled: 2009-08-27 16:39:52 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing ->15854199896 in context public Dialplan: sofia/internal/585xxxxxxx(a)208.34.86.39 parsing [public- >curl_test] continue=false Dialplan: sofia/internal/585xxxxxxx(a)208.34.86.39 Regex (FAIL) [curl_test] destination_number(15854199896) =~ /^(18775844111)$/ break=on-false 2009-08-27 16:39:52 [INFO] switch_core_state_machine.c:136 switch_core_standard_on_routing() No Route, Aborting -- Greg

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