March 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Time limit
by Juan Carlos Gonzalez Cardona 23 Mar '10

23 Mar '10

Hi everyone, is posible use call limite for calls, to some extensiones? In Asterisk is possible creating custom context with set time limit to specific trunk, but i see that Freeswitch is more stable code, i need to know if is possible create groups for limit calls and assign extensiones to that group, if is possible, how i do that, thank´s. -- Juan Carlos Gonzalez C.

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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 setup. FreeSWITCH on a Public IP address. All the AASTRA phones behind the Linksys WRT 54GL(Tomato firmware) . I see the following sample dialplan come with FreeSWITCH default.xml <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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Problem with bridge to multiple targets and loose race.
by Jonas Gauffin 23 Mar '10

23 Mar '10

Hello I got the following dial plan: http://pastebin.freeswitch.org/12527 Why do the registered phone loose the race after a second? The other number hasn't even started to ring yet. 336a7d1b-3815-4a10-8503-506765035dfe 2010-03-23 20:06:14.560000 [NOTICE] switch_channel.c:642 New Channel sofia/internal/u1000028 [336a7d1b-3815-4a10-8503-506765035dfe] 336a7d1b-3815-4a10-8503-506765035dfe 2010-03-23 20:06:15.300000 [NOTICE] switch_ivr_originate.c:3060 Hangup sofia/internal/u1000028 [CS_CONSUME_MEDIA] [LOSE_RACE] Log: http://pastebin.freeswitch.org/12528 //Jonas

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Simulate dtmf on a channel
by Fraser Redmond 23 Mar '10

23 Mar '10

>From the CLI/event-socket I want to simulate a dtmf press on an existing channel to make it perform an action as if the caller had pressed the key themselves (specifically they're in a javascript dialplan and listening to a session.streamFile() which is waiting for a dtmf input.) I've had a play with uuid_send_dtmf but all that seems to do is play the tone on the channel, even with it's undocumented third parameter for from|to|both. The tone is heard, but the action that would have been performed if the caller had pressed doesn't happen. Is there any way to do this? I'd really like to keep the call in the javascript dialplan, as breaking them out could get messy, with the other things that are going on. Cheers, Fraser

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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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