August 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Success! and an issue.
by Christensen Tom 29 Aug '09

29 Aug '09

well, just made my first extension to extension call using freeswitch, so that was pretty painless. Huge fan of the config layout at least (having the extensions defined each in their own file is going to be so much nicer than asterisks huge jumble). However, I did run into one issue, not sure if its somehow related to the phones I'm using (softphone of unknown quality (weephone) on iphones)... I'm using the default config, when I dial 4000 for voicemail, it works, however at apparently random times during the menu navigation I get call waiting beeps and I have an incoming call from 4000. If I decline or leave this call unanswered, my call into voicemail gets disconnected. If I "accept" this call, it apparently disconnects immediately (IE I only have 1 active call), however I can then continue navigating the menu, etc... Very strange to me, I will be setting up a couple of polycom 501s on this system tomorrow and will report if I see similar issues. Anyway, so far I'm pleased with freeswitch, I've got a list of features I need to support before I can deploy it, so I'll be bugging people (sorry), but so far thanks! -Tom

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external sip profile
by e schmidbauer 29 Aug '09

29 Aug '09

I am using a fresh install of freeswitch trunk and I am unable to register a phone on port 5080. I set the the registrar of the phone to server.host.com:5080 but when i do a sofia status profile external, there are no registrations. when i do a sofia status profile internal, it shows the phone registered to port 5060. Is this a bug in freeswitch or is there some configuration I need to change?

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FreeSWITCH environmental variables
by Thangappan.M 29 Aug '09

29 Aug '09

Dear all, In the case of asterisk PBX. I can get all the information about the call from the environmental variable itself like Caller id ,called id, channel no. But in freeswitch where all the environmental variables are resides? How can I access all the variables? Please Help me? -- Regards, Thangappan.M

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Disabling Core Dumping
by Max Bridgewater 29 Aug '09

29 Aug '09

Hi, Whenever i stop freeswich, it creates a core dump. How can i disable that? thanks, Max.

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mod_limit and memcache
by Woody Dickson 29 Aug '09

29 Aug '09

Hello, I read something that talks about using memcache for mod_limit before. Is it something that is available now? If I have multiple instances of freeswitch that need to share the same limit status, it there any existing solution? If no existing solution is available, what is the best way to go about modifying mod_limit to accomplish limiting for multiple freeswitch servers together? Thanks, Woody

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Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
by Hristo Benev 29 Aug '09

29 Aug '09

Hello Rod, I did the change. Here is extract of console: ----------------------------------------------------------------------------------------- variable_continue_on_fail: [true] variable_sip_h_X-ROUTE: [LOOKUP] variable_export_vars: [sip_h_X-ROUTE] variable_signal_bond: [a66d089a-93ee-11de-ab7a-094dff0ede20] variable_sip_redirect_contact_0: [sip:France@PEER_01] variable_sip_redirected_to: [sip:France@PEER_01] variable_sip_redirect_contact_user_0: [France] <------------------------------------------- variable_sip_redirect_contact_host_0: [PEER_01] variable_sip_redirect_dialstring_0: [sofia/internal/sip:France@PEER_01] variable_sip_redirect_dialstring: [sofia/internal/sip:France@PEER_01] variable_proto_specific_hangup_cause: [sip:503] variable_sip_hangup_phrase: [DNS Error] variable_originate_disposition: [NORMAL_TEMPORARY_FAILURE] variable_ROUTE_GW: [France] variable_AREA: [France] variable_current_application: [info] ------------------------------------------------------------------------------------------ I have different value it is actually the description field as shown here: -------------------------------- /opt/kamailio/sbin/kamctl cr show cr carrier names +----+---------+ | id | carrier | +----+---------+ | 1 | default | +----+---------+ cr domain names +----+---------+ | id | domain | +----+---------+ | 1 | default | +----+---------+ cr routes +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+ | id | carrier | domain | scan_prefix | flags | mask | prob | strip | rewrite_host | rewrite_prefix | rewrite_suffix | description | +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+ | 1 | 1 | 1 | 1000 | 0 | 0 | 1 | 0 | PEER_01 | | | France | +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+ ----------------------------------------- And here is what I have in kamailio: ------------------------------------------------------------------------------------- ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } t_check_trans(); if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # LOOKUP ROUTE TABLE WHEN ASKED BY HEADER: X-ROUTE:LOOKUP if (is_method("INVITE") && $hdr(X-ROUTE)=="LOOKUP"){ if(!cr_route("default", "default", "$rU", "$rU", "call_id","$avp(s:route_desc)")){ #xlog("ROUTING FAILED: no route found for $rU"); sl_send_reply("604", "Unable to route this call"); exit; } else { xlog("LOOKUP FOUND: $rd $avp(s:route_desc)"); avp_pushto("$ru/username", "$avp(s:route_desc)"); sl_send_reply("302", "$rd"); exit; } } } ---------------------------------------------------------------------------------- Another question... In that part of FreeSwitch dialplan.xml ----------------------------------- <context name="ROUTING"> <extension name="PEER_01"> <condition field="${sip_h_X-ROUTE}" expression="PEER_01"> <action application="set" data="hangup_after_bridge=true"/> ----------------------------------- X-ROUTE maybe should be replaced with ROUTE_GW otherwise there is no way the regex is true. As for thanks - for sure by default they are also for the developers of both apps. I'm new in freeSwitch and/or Kamailio for now I'm still testing and learning so it will be nice to have something working to start with. >-------- Оригинално писмо -------- >От: rod >Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route >До: freeswitch-users(a)lists.freeswitch.org >Изпратено на: Петък, 2009, Август 28 09:54:00 EEST >Hello, > >the trace seems good. >If you check the answer from Kamailio, you'll see that Kamailio answers >with "302 PEER_01". > >As Michael Collins stated before, you can get the variable containing >"PEER_01", then this variable is stored in a custom variable. > >In your dialplan, may you please add: > >, just before the transfer line, eg: > > > > > > > > > > > > > >Using application Info, you'll see on the console (or CDR) the list of >variables used for this call. You should see the content of >"${sip_redirect_contact_user_0}" that should contain the value >"PEER_01". Please check this and let me know. > >For the new configuration file, no problem for sharing. But as I wrote >on the wiki page, I worked on this setup cause LCR module was not >available when I start working on FS, nor I'm a good programmer to write >a server side HTTP script (used by xml_curl) that could scale to my needs. > >I enhanced a bit this configuration with support for fallback routing, >almost realtime graph (every minutes using www.cacti.net and some >functions in FS like limit_hash) of number of concurrent calls per AREA, >PEER... > >Thanks for "the good tutorial", but don't forget the dev team who did >this great product ;-) > >rod. > > >Hristo Benev a écrit : >> 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" ;tag=978g69jZaFpBD. >> To: . >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c. >> CSeq: 119574771 INVITE. >> Contact: . >> 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" ;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" ;tag=978g69jZaFpBD. >> To: ;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" ;tag=978g69jZaFpBD. >> To: ;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 >> > >> >> _______________________________________________ >> 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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Hello, and stuff.
by Christensen Tom 29 Aug '09

29 Aug '09

As a background, I ran an asterisk consulting company for about 3 years that I gave up on 2 years ago after repeatedly failing to achieve any sort of stability on any sort install over about 30 phones, I gave up. Maybe that was wrong, I am open to the possibility that I just didn't know enough and I was building things wrong, but I worked inside the asterisk code (which I feel is a hopeless mess), I implemented a few small custom features, anyway... I'm coming back into the VoIP space now, and I'm wondering what sort of issues can I expect in trying to pick up and learn freeswitch? From what I've read on the website, it appears to have a much more sane architecture. I've used Cisco, Broadsoft, and asterisk in the past. By far the least stable and worst general call quality was asterisk. I constantly contended with strange call quality issues in asterisk, lots of echo (even with hardware echo cancellation cards), lots of jitter, lots of call break up (even on small systems with 10-20 users, using QoS on the network, and in general doing everything I could to prioritize voice over anything else). When I used Cisco call manager and broadsoft, the voice quality issues were basically non-existant, as long as the network was running QoS echo, stutter, calls breaking up, just didn't happen. So, I guess my question is, does freeswitch show a marked improvement over asterisk in this department? As long as you configure QoS and have hardware echo cancellation does it actually work reliably? Thanks for any additional information about freeswitch you can provide as well. I am a software developer primarily by trade, but I do lots of consulting type work in the SME space and I've had a couple projects thrown to me that require some integration with a phone system, and I just can't in good conscience recommend asterisk anymore. -Tom

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alternative to java-asterisk
by tom 28 Aug '09

28 Aug '09

hi, with the java-asterisk bridge its possible to to AMI-stuff. is thi as well possible with freeswitch? if so, are there any tutorials, docu etc. out there? thx

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Centos 5.3 vs Ubuntu LTS
by Fernando Testa 28 Aug '09

28 Aug '09

Hi folks,What linux distro would you suggest to deploy FS with django: Centos 5.3 or Ubuntu LTS? Thanks

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newbie alert. Help with dialplan
by Julian Lyndon-Smith 28 Aug '09

28 Aug '09

I am a long-time asterisk user (2005), so I come from an unfortunate position of having to unlearn an awful lot of stuff in order to make freeswitch do the things I want ;( I *think* i've got my head around using mod_curl_xml (?) to read all the config stuff from my webserver. I *think* I've got my head around setting up the sip clients etc etc ... However, where I am really struggling is the dialplan. For the life of me I simply cannot seem to grasp the fs way - that's no disrespect to the fs way, but perhaps the failure of this old brain to change ! have pastebinned an example of show 1234546@inboundq at http://www.pastebin.ca/1544890. If possible, would someone be able to show me how to convert this dialplan to the fs way ? If I could be given a little foot-up I'm sure that I would be able to convert the rest of the dialplan ! Thanks in advance, and please go easy ! Julian

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