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bridging 2 SIP destinations from PBX behind NAT
by Michal Zubáč 05 Jan '12

05 Jan '12

Hi. I encountered problem with bridging two SIP calls. My FreeSwitch is behind NAT, I'm not sure if endpoint switches are, but I think it doesn't matter now. Situation looks like this: SIP1 <--> NAT <--> PBX <--> NAT <--> SIP2 SIP1 is caller switch, SIP2 is destination switch so call is initiated like this: SIP1->PBX->SIP2 I tested both endpoints separately using ISDN endpoint and they worked. Like this ISDN1->PBX->SIP2 & SIP1->PBX->ISDN2 But when I connected SIP1 & SIP2 together, call is estabilished, but no audio was going through our firewall. Both audio streams from SIP1 and SIP2 were filtered out by our firewall, because Freeswitch (PBX) wasn't sending any initial packets to SIP endpoints, so no NAT holes were created. It look strange to me. I expected FreeSwitch to at least send some "empty" RTP packets to SIP endpoints as soon as call estabilishment is confirmed on SIP channel. But Freeswitch doesn't do it and only sends ringing indication. I had problems in scenario, where SIP2 endpoint doesn't ring and immediately answers the call. Call was estabilished, but audio was stuck in our firewall, because Freeswitch haven't initiated any RTP communication yet and was probably waiting for something. I managed to solve this problem by adding <action application="set" data="instant_ringback=true"/> between my <action application="ring_ready" /> and <action application="bridge" /> commands in dialplan. Now it works, but I'm feeling kind of uncertain with this solution, because I don't really understand why. What was Freeswitch waiting for, before I added instant_ringback? Why can't FreeSwitch prepare NAT holes (as described before) as soon as SDP are interchanged? Thanks for clarification. Regards -- Michal Zubac ComGate Interactive s.r.o. Prague Marina Office Center Jankovcova 1596/14a 17000 Praha 7, Czech Republic

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help with wait_for_silence
by Hector Geraldino 05 Jan '12

05 Jan '12

Hi, I'm trying to use the wait_for_silence application to stop recording audio from a channel, and I need to use it not from a FS dialplan but from an ESL application. So far I can execute the wait_for_silence application using 'execute wait_for_silence arguments' with no issues. I can even see the result of the execution of this command on the fs_cli console: 2012-01-05 19:04:40.266001 [DEBUG] switch_ivr.c:576 sofia/internal/5512(a)192.168.8.11 Command Execute wait_for_silence(200 15 15 5000) EXECUTE sofia/internal/5512(a)192.168.8.11 wait_for_silence(200 15 15 5000) The problem is that I don't know if this is working or not, as it is not raising any event that can be received by the application. I don't see any event related to this application (or to detect silence using any other mechanism) on the wiki Events List page [http://wiki.freeswitch.org/wiki/Event_list] so I'm not sure it is usable in an socket outbound mode. Resuming: is it possible to detect the silence on a channel and then perform an action in an ESL application, using wait_for_silence app or any other mechanism? Thanks, Hector

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SIP attacks from 188.161.101.73
by Karol 05 Jan '12

05 Jan '12

Anyone else see this?? http://pastebin.freeswitch.com/18078

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How to troubleshoot "503 Maximum Calls In Progress"?
by Chris Fowler 05 Jan '12

05 Jan '12

Hi, On our Production FreeSWITCH box: FreeSWITCH Version 1.0.head (git-4c435ec 2011-03-14 11-54-08 -0500) UP 0 years, 14 days, 6 hours, 28 minutes, 44 seconds, 474 milliseconds, 751 microseconds 12519 session(s) since startup 2 session(s) 0/30 1000 session(s) max min idle cpu 0.00/100.00 At 14:30:37 a reloadxml command was issued. At 14:30:43 the box started rejecting all calls with 503 error. At 14:38 it started working again. Looking at the code I see three conditions can trigger this behavior: if (sess_count >= sess_max || !sofia_test_pflag(profile, PFLAG_RUNNING) || !switch_core_ready()) { nua_respond(nh, 503, "Maximum Calls In Progress", SIPTAG_RETRY_AFTER_STR("300"), TAG_END()); switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "No more sessions allowed at this time.\n"); goto done; } Logs show it wasn't lack of available sessions. How can sofia_test_pflag(profile, PFLAG_RUNNING) or switch_core_ready() fail and busy the system? Thoughts? Thx, Chris.

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Sangoma A200 FXO Outgoing Problem
by Ryan V 05 Jan '12

05 Jan '12

Hi, I have configured A200 and able to receive incoming calls. I am unable to dial out. Here are the logs. 2011-12-08 16:39:15.990089 [DEBUG] ftmod_analog.c:62 [s3c1][2:1] Changed state from DOWN to DIALING 2011-12-08 16:39:15.990089 [DEBUG] ftmod_analog.c:439 [s3c1][2:1] ANALOG CHANNEL thread starting. 2011-12-08 16:39:15.990089 [DEBUG] ftmod_analog.c:459 [s3c1][2:1] Initialized DTMF detection 2011-12-08 16:39:15.990089 [DEBUG] ftmod_analog.c:640 [s3c1][2:1] Completed state change from DOWN to DIALING in 0ms 2011-12-08 16:39:15.990089 [DEBUG] ftmod_analog.c:646 [s3c1][2:1] Executing state handler on 3:1 for DIALING 2011-12-08 16:39:21.009503 [DEBUG] ftmod_analog.c:505 [s3c1][2:1] Changed state from DIALING to BUSY 2011-12-08 16:39:21.030044 [DEBUG] ftmod_analog.c:640 [s3c1][2:1] Completed state change from DIALING to BUSY in 20ms 2011-12-08 16:39:21.030044 [DEBUG] ftmod_analog.c:646 [s3c1][2:1] Executing state handler on 3:1 for BUSY 2011-12-08 16:39:21.030044 [DEBUG] ftmod_analog.c:802 [s3c1][2:1] Changed state from BUSY to DOWN 2011-12-08 16:39:21.049928 [DEBUG] ftmod_analog.c:640 [s3c1][2:1] Completed state change from BUSY to DOWN in 20ms 2011-12-08 16:39:21.049928 [DEBUG] ftmod_analog.c:646 [s3c1][2:1] Executing state handler on 3:1 for DOWN 2011-12-08 16:39:21.049928 [DEBUG] ftmod_analog.c:944 [s3c1][2:1] Going onhook2011-12-08 16:39:21.049928 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/3:1/9449905000) Running State Change CS_HANGUP Here is our dial plan <extension name="outgoing_bsnl"> <condition field="destination_number" expression="^9(\d{10})$"> <action application="bridge" data="freetdm/3/a/$1"/> </condition> </extension> I suspect FS is not detecting the dial tone on analogue channel. BTW, outgoing on our digital span works fine. Any suggestions? Thanks, Ryan

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Flite speed and pitch increase after update.
by Brandon McGinty 05 Jan '12

05 Jan '12

Hi everyone. We upgraded Freeswitch on the 30th of December, using the git branch revision from the 28th of December. However, while the conference features improved, the flite package has had its pitch and rate increased for all tts speech. We are using the mod_tts module in our conferences, and it now sounds extrordinarily sped up and pitched up, similar to what one would consider a chipmunk sound. I've gone through the mod_tts sources, the flite sources, the modules directory, the autoload_modules directory, the configs for the conference and flite, and I can't find anywhere how to change the pitch and rate of the synthesizer. I've tried tts-rate and tts-pitch, etc, to no affect. I've also tried google, and noone else seems to be dealing with this strangeness. We are all using soft-phones or pots to sip devices, so the speech output is quite important. Any help you all can provide would be truly be appreciated. Brandon McGinty-Carroll

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Torrents for the weekly conference
by Karol 04 Jan '12

04 Jan '12

Has everyone given up on this method? I have 3 torrents frozen, in their tracks. - Karol

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 67, Issue 7
by Sharad Garg 04 Jan '12

04 Jan '12

All these are basically IP based PBX systems having PBX features. Regards ----- Original Message ----- From: <freeswitch-users-request(a)lists.freeswitch.org> To: <freeswitch-users(a)lists.freeswitch.org> Sent: Tuesday, January 03, 2012 7:47 PM Subject: FreeSWITCH-users Digest, Vol 67, Issue 7 > 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. Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, > CallWeaver, and YATE. (Kaushal Shriyan) > 2. Re: mod_sofia not binding to port 5060 (Brandon McGinty) > 3. Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, > CallWeaver, and YATE. (Zohair Raza) > 4. Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, > CallWeaver, and YATE. (Giovanni Maruzzelli) > 5. Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, > CallWeaver, and YATE. (Kaushal Shriyan) > 6. Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, > CallWeaver, and YATE. (Zohair Raza) > 7. Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, > CallWeaver, and YATE. (Vitaly Nikolaev) > -------------------------------------------------------------------------------- > _______________________________________________ > 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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Raspberry Pi & ISDN
by lists.freeswitch.org＠sjau.ch 04 Jan '12

04 Jan '12

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi there Soon the Raspberry Pis will be available and I'm planning to buy a bunch of them. I think with a 4GB SD Card, ethernet port and 2x USB port it should be fine. However in one office that I plan on setting up FS is still ISDN being used. There are no plans on changing to VoIP however having an internal FS system would be nice. My question is, how can I hookup the Raspberry Pi to the ISDN lines. I guess I'd need some USB<->ISDN Adapter. The only thing I could find this far were USB ISDN modems. But modems are for data transmission and not voice - at least that's what I think. So, any of you have suggestions how I can hook up the Raspberry Pi to ISDN? Thx Stephan -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.11 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAk8EYAEACgkQ9KcSxJrwrxgIjgCfaMa4sYLN1TD17+YMXd7WY0YW /CUAnAgPR6v0Dr2RhM7D9aWhO5770wST =m666 -----END PGP SIGNATURE-----

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Debugging intermitent BLF problems
by Yehavi Bourvine 04 Jan '12

04 Jan '12

Hello, I am "stuck" with FreeSWITCH version from 15-Oct since all newer vesions that I tried have intermitent BLF problems. The BLF gets stuck once in a while (stuck in Ringing, busy or free) until the next SUBSCRIBE refresh (once every hour). It is not determenistic, and it happens on a profile on which I have over 180 extensions (it does not happen on a test profile I built and put one phone on it). Before I open a JIRA: 1. Has anyone noticed this? 2. What debugs shall I enable so there will be a reasonable amount of data? BTW, most phones are Polycoms (and one SNOM...). Thanks, __Yehavi:

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