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Re: [Freeswitch-users] rtp timeout yet
by Madovsky 23 Mar '10

23 Mar '10

to be sure I'm silly I downgraded from trunk 17001 and rtp timeout disappeared. Regards ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, March 22, 2010 3:20 PM Subject: Re: [Freeswitch-users] rtp timeout yet so when I use a public IP to register a phone, there s a rtp timeout from line 853 of switch_rtp.c I looked at the code line 853 below : ------------------------------------------------------- old_sock = rtp_session->sock_input; rtp_session->sock_input = new_sock; new_sock = NULL; if (switch_test_flag(rtp_session, SWITCH_RTP_FLAG_USE_TIMER) || switch_test_flag(rtp_session, SWITCH_RTP_FLAG_NOBLOCK)) { switch_socket_opt_set(rtp_session->sock_input, SWITCH_SO_NONBLOCK, TRUE); switch_set_flag_locked(rtp_session, SWITCH_RTP_FLAG_NOBLOCK); } switch_socket_create_pollset(&rtp_session->read_pollfd, rtp_session->sock_input, SWITCH_POLLIN | SWITCH_POLLERR, rtp_session->pool); status = SWITCH_STATUS_SUCCESS; *err = "Success"; switch_set_flag_locked(rtp_session, SWITCH_RTP_FLAG_IO); done: if (new_sock) { switch_socket_close(new_sock); } if (old_sock) { switch_socket_close(old_sock); } if (rtp_session->ready != 1) { WRITE_DEC(rtp_session); READ_DEC(rtp_session); } return status; ----------------------------------- it seems that old_sock is the actual sock that it needs to use since there is no auto changing to do ? So does it mean maybe the old_sock is yet existing because there is no NAT ? Thanks ----- Original Message ----- From: "Madovsky" <infos(a)madovsky.org> To: <freeswitch-users(a)lists.freeswitch.org> Sent: Monday, March 22, 2010 2:31 AM Subject: Re: [Freeswitch-users] rtp timeout yet >I deactivated iptables, no router, no nat, > only 2 public IP, 2 softphones connected on the same FS which is on a public > IP to. > default config, result -> rtp timeout > crazy that's not working since I can make 2 softphones behind a firewall > work.... > > ok that's enough for me tonight ! ;) > > ----- Original Message ----- > From: "Madovsky" <infos(a)madovsky.org> > To: <freeswitch-users(a)lists.freeswitch.org> > Sent: Monday, March 22, 2010 1:55 AM > Subject: Re: [Freeswitch-users] rtp timeout yet > > >>I replace public ip by these fake one... >> >> ----- Original Message ----- >> From: "Jason White" <jason(a)jasonjgw.net> >> To: <freeswitch-users(a)lists.freeswitch.org> >> Sent: Monday, March 22, 2010 1:49 AM >> Subject: Re: [Freeswitch-users] rtp timeout yet >> >> >>> Madovsky <infos(a)madovsky.org> wrote: >>>> result with tcpdump -ennqti br0 -T rtp -vvv dst port 5060 >>>> >>>> 10.10.10.132 = softphone1 registered from public IP >>>> >>>> 11.11.11.= softphone2 registered from public IP >>>> >>>> 10.10.10.138 = FS >>> >>> Those are private addresses; there's a NAT device involved somewhere, and >>> that's likely to be where your problems are. >>> >>> >>> _______________________________________________ >>> 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] rtp timeout yet
by Madovsky 23 Mar '10

23 Mar '10

Holly sh....! , I found the problem ! I didn't know that the fact I don't plug a mic on the legA and talk in it can causes a rtp timeout !!!! so I have a question : what's happened if in a conversation the legB talks talks and talks but legA listens and don't say anything longer as the rtp timeout ? :D I tried it, and indeed, there's an rtp timeout everytime I stop to talk. ----- Original Message ----- From: "Madovsky" <infos(a)madovsky.org> To: <freeswitch-users(a)lists.freeswitch.org> Sent: Tuesday, March 23, 2010 12:26 PM Subject: Re: [Freeswitch-users] rtp timeout yet > thanks for your answer Jason. > yes the problem is yet there. before > to submit a bug I want to be sure it's a mistake I did. > so when I set inbound-bypass-media to true, the problem disappear. > it's for sure a media problem. > A phone with a public IP ( I tested on 6 PCs) registers well, but make a > rtp timeout > A phone behind a firewall (tested on 6 PCs too) registers well and doesn't > make rtp timeout. > I stopped all IP rules like iptables etc... without success. > I'd like to understand > > Thanks > > > > ----- Original Message ----- > From: "Jason White" <jason(a)jasonjgw.net> > To: <freeswitch-users(a)lists.freeswitch.org> > Sent: Tuesday, March 23, 2010 4:32 AM > Subject: Re: [Freeswitch-users] rtp timeout yet > > >> Madovsky <infos(a)madovsky.org> wrote: >>> to be sure I'm silly I downgraded from trunk 17001 >>> and rtp timeout disappeared. >> >> Try the latest trunk. If the problem still occurs, capture SIP traces and >> submit a bug report, making sure the configurations are identical. >> >> >> _______________________________________________ >> 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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Paging - AASTRA phones
by Aloysius Lloyd 23 Mar '10

23 Mar '10

Hi All, How to build a dial plan for paging for groups of AASTRA phones. Here is my environment. FreeSWITCH on a Public IP address. All the AASTRA phones behind the Linksys WRT 54GL(Tomato firmware) . I see the following sample dial plan come with FreeSWITCH. <extension name="rtp_multicast_page"> <condition field="destination_number" expression="^*pagegroup* $|^7243$"> <action application="answer"/> <action application="esf_page_group"/> </condition> </extension> What is the *pagegroup *variable referring to? Thanks Lloyd

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Re: [Freeswitch-users] rtp timeout yet
by Madovsky 23 Mar '10

23 Mar '10

And to explain why it was working behind a firewall, I was using another softphone (x-lite) that maybe sends fake packets to keep the line on... ----- Original Message ----- From: "Madovsky" <infos(a)madovsky.org> To: <freeswitch-users(a)lists.freeswitch.org> Sent: Tuesday, March 23, 2010 12:58 PM Subject: Re: [Freeswitch-users] rtp timeout yet > Holly sh....! , I found the problem ! > I didn't know that the fact I don't plug a mic on the legA and talk in it > can causes a rtp timeout !!!! > so I have a question : what's happened if in a conversation the legB > talks talks and talks > but legA listens and don't say anything longer as the rtp timeout ? :D > I tried it, and indeed, there's an rtp timeout everytime I stop to talk. > > > > > ----- Original Message ----- > From: "Madovsky" <infos(a)madovsky.org> > To: <freeswitch-users(a)lists.freeswitch.org> > Sent: Tuesday, March 23, 2010 12:26 PM > Subject: Re: [Freeswitch-users] rtp timeout yet > > >> thanks for your answer Jason. >> yes the problem is yet there. before >> to submit a bug I want to be sure it's a mistake I did. >> so when I set inbound-bypass-media to true, the problem disappear. >> it's for sure a media problem. >> A phone with a public IP ( I tested on 6 PCs) registers well, but make a >> rtp timeout >> A phone behind a firewall (tested on 6 PCs too) registers well and >> doesn't make rtp timeout. >> I stopped all IP rules like iptables etc... without success. >> I'd like to understand >> >> Thanks >> >> >> >> ----- Original Message ----- >> From: "Jason White" <jason(a)jasonjgw.net> >> To: <freeswitch-users(a)lists.freeswitch.org> >> Sent: Tuesday, March 23, 2010 4:32 AM >> Subject: Re: [Freeswitch-users] rtp timeout yet >> >> >>> Madovsky <infos(a)madovsky.org> wrote: >>>> to be sure I'm silly I downgraded from trunk 17001 >>>> and rtp timeout disappeared. >>> >>> Try the latest trunk. If the problem still occurs, capture SIP traces >>> and >>> submit a bug report, making sure the configurations are identical. >>> >>> >>> _______________________________________________ >>> 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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FreeSwitch & Vegastream TLS interoperability problems?
by Yehavi Bourvine 23 Mar '10

23 Mar '10

Hello, We have a Vegastream ATA adapter working fine with FreeSwitch as long as we do not enable TLS. Once TLS is enabled, I see the following behavior: - The adapter creates a TLS tunnel with Freeswitch and registers itself. However, I see one difference in the registration between this adapter and other adapters/phones: - The VegaStream adapter is registered as: Contact: "user" <sips:80632@10.64.1.6:5061> Status: Registered(TLS)(unknown) EXP(2010-03-23 08:50:26) - Other TLS phones register as: Contact: "user" <sips:80631@10.64.0.8:5061;transport=tls> Status: Registered(TLS)(unknown) EXP(2010-03-23 08:37:35) - What I see is the difference in the "contact" field in which the transport=tls clause is missing. - When a user on the ATA dials out, the request is sent via TLS and all is fine. - When someone calls an ATA extension, Freeswitch sends the request to the ATA via SIP/UDP and not via SIP/TLS I've contacted the manufacturer who claims that according to RFC5630 it should send the request via TLS as it is registered as TLS. I have a feeling that the erronous behaviour is because they do not append "transport=tls" to the contact field, although they claim it is not required. Who is right and who is wrong? Thanks! __Yehavi:

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Question about TOS/DSCP
by Helmut Kuper 23 Mar '10

23 Mar '10

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, is there a way in FS to set the TOS/DSCP values directly in FS instead of using iptables as described in FS wiki? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFLqH3C4tZeNddg3dwRArTKAKCMkiMNgL2MWcbnikdkRlHlD3kJzgCfZvEK Cv2gtKHQPYyeelW4mxwFpmg= =AJAv -----END PGP SIGNATURE-----

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recoding idividual contributions to a conference call
by Andy 23 Mar '10

23 Mar '10

Hi, Sorry I couldn't find this in the doucmentation. If I have 3 people in a conference call, can I record each of their contriubutions into 3 seperate files? Many thanks Andy

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Call gets hanged up with ORIGINATOR_CANCEL cause
by Maciej Aniserowicz 23 Mar '10

23 Mar '10

Hello, I have the following problem: I register with x-lite to my freeswitch and then call it by sending command bgapi originate {originate_timeout=15}[24cc_session_id=76dfb907-a0c3-4357-b1f9-ede89caf8a7e,ignore_early_media=true,max_forwards=999999,origination_uuid=d4f4e460-43c7-4e53-af18-46fa5226227d]sofia/gateway/agent_gateway/2500 &park() Then i dial through a gateway which points to another freeswitch instance: bgapi originate {originate_timeout=20}[24cc_session_id=3134a5ae-ec00-4164-94b1-a7fb5e0368fa,ignore_early_media=true,max_forwards=999999,origination_uuid=a85fd47e-051f-42ea-987e-f4e090c7ae78]sofia/gateway/prospect_gateway/1300671139878747511456 &park() The dialplan causes answering the phone, playing a file and hanging up. The two calls are connected with this command: bgapi uuid_transfer f0c0db16-f00f-4b0b-8a00-d3b108d121b2 intercept:a85fd47e-051f-42ea-987e-f4e090c7ae78,park inline Unfortunately after the second call hangs up, the first call gets hanged up too. I recently upgraded to FS trunk. Logs are in pastebin: first freeswitch: http://pastebin.freeswitch.org/12432 second freeswitch: http://pastebin.freeswitch.org/12433 Please take a look and help me figure this out.. -- View this message in context: http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp473… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Compiling freeswitch on Ubuntu 10.04
by Kees Varekamp 23 Mar '10

23 Mar '10

Hi there, I am trying to compile Freeswitch on Ubuntu 10.04 but am getting stranded as follows: g++: /usr/lib/gcc/i486-linux-gnu/4.4.1/../../../../lib/crti.o: No such file or directory g++: /usr/lib/gcc/i486-linux-gnu/4.4.1/crtbeginS.o: No such file or directory g++: /usr/lib/gcc/i486-linux-gnu/4.4.1/crtendS.o: No such file or directory g++: /usr/lib/gcc/i486-linux-gnu/4.4.1/../../../../lib/crtn.o: No such file or directory make[2]: *** [libfreeswitch.la] Error 1 make[2]: Leaving directory `/usr/src/freeswitch.trunk' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/freeswitch.trunk' make: *** [freeswitch] Error 2 I think this is related to the gcc version; ubuntu 10.04 comes with version 4.4.3 rather than 4.4.1. Does anyone know how to fix this?

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Re: [Freeswitch-users] Monitoring Registrations/Users
by Dave Redmore 22 Mar '10

22 Mar '10

Nice - This is what I was looking for. So, Freeswitch sends an options Ping to the endpoints, using the nat-options-ping option and then will unregister if an endpoint doesn't respond. Then I can alert on an unregistered device. Sounds perfect. Thanks. Will experiment with that. Dave ----- Original Message ----- From: "Gabriel Kuri" <gkuri(a)ieee.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, March 22, 2010 6:16:12 PM GMT -06:00 US/Canada Central Subject: Re: [Freeswitch-users] Monitoring Registrations/Users Try and use "unregister-on-options-fail" and "nat-options-ping" ? Write a little app that connects to the Event Socket and wait for a device to be unregistered as a trigger for doing whatever you want (ie send an email, snmp trap, etc.). http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#unregister-on-opt… Cheers, Gabe On Mon, Mar 22, 2010 at 4:01 PM, Dave Redmore <dave.redmore(a)spigotsystems.com> wrote: > Thanks for the reply. I think that for me, the reason that the > registrations are getting dropped or calls are getting dropped is because we > are running across "best effort:" Internet connections - DSL and Cable - and > we are using off-the-shelf ATAs from Grandstream and Linksys. So, stuff > happens and we lose calls - but if I could Poll those far ends, I can be > more proactive when that happens. If I could send a SIP packet of some sort > every minute and make sure I get a response, then I know those devices are > ready to receive a call and if they aren't I can proactively find out why. > > Dave > > ----- Original Message ----- > From: "Michael Collins" <msc(a)freeswitch.org> > To: freeswitch-users(a)lists.freeswitch.org > Sent: Monday, March 22, 2010 5:03:04 PM GMT -06:00 US/Canada Central > Subject: Re: [Freeswitch-users] Monitoring Registrations/Users > > You can monitor in various ways, but a simple way to start out is by using > either a script with ESL or just doing: fs_cli -x "sofia status profile > internal" and parsing the results to look for whichever SIP registrations > are of interest. > > As far as finding this useful, I'm sure there are some who would find it > useful in at least a few cases. However, my guess is that they would be more > interested in know why the registrations are getting dropped in the first > place... > > -MC > > On Sun, Mar 21, 2010 at 6:45 PM, Dave Redmore > <dave.redmore(a)spigotsystems.com> wrote: >> >> Been working with Freeswitch to implement a simple T38 proxy to provide >> outbound faxing to some customers. Works very well, on initial testing, >> using the proxy-media option. Nice NAT detection/SBC functionality "out of >> the box". Generally very easy to get my head around and get it to do what I >> wanted, >> >> One bit of functionality I've been looking for in a softswitch for a >> while, and have never seen, is to somehow monitor the status of >> users/registrations and alert somehow if there is a problem. In my >> situation, we have connected some customers with VoIP phone service for >> their small offices. Typically, just a simple ATA that gives them 1-4 >> trunks and a couple of DIDs. The problem has always been that if their ATA >> drops it's registration or is not available, no one knows it until a >> customer tells them through some other means that their phones aren't >> working. This is a huge problem, obviously, and happens more than I would >> like. It seems like it wouldn't be too difficult to create some sort of >> monitoring function which can send an Invite packet every little while and >> make sure the UA responds with a 100/Trying and then disconnect. Basically >> polling any devices that I deem as needing to always be available and >> sending an email or something if one isn't. Does this seem useful to anyone >> else? Has anyone seen anything like this out there? >> >> Thanks, >> >> Dave Redmore >> Spigot Networks, Inc. >> >> _______________________________________________ >> 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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