- FreeSWITCH-dev - FreeSWITCH Mailing Lists

INFO RE: MODENDP-81
by Michael Collins 02 Feb '08

02 Feb '08

Mike et al.: I just wanted you to know that this issue has to do with the PRI dialing failures we've been experiencing. (The mystery octets, as it were.) I'm pretty sure that I've located the source of the extra bytes by parsing the Q931 SETUP message, specifically the called party number IE: the span, channel, and slashes from the bride app data seem to be making it into the called party number field. The details are in jira. I figured that although this was probably trivial for you guys to fix it would be good to have the issue on record. -MC P.S. - Mike, thanks for all of your time on this one. I hope this research saves you some energy.

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MWI support for SIP gateway
by sanjeev mudholkar 01 Feb '08

01 Feb '08

Hi, I have configured Freeswitch to register to a sipX server. Did a quick modification in sofia_reg.c to send a SUBSCRIBE with event:message-summary as soon as the registration is successful. With this change I see a SUBSCRIBE sent out, then receives a NOTIFY which is responded with '200 OK'(automatically) and then a '202 Accepted' is received for the SUBSCRIBE. But the issue is that the incoming NOTIFY is notified as 'nua_i_error' Here is the exact capture from console on receiving a NOTIFY "2008-02-01 17:44:41 [DEBUG] sofia.c:183 sofia_event_callback() event [nua_i_error] status [500][Responding to a Non-Existing Request] session: n/a" The event expected is 'nua_i_notify', so as to interpret the MWI Can someone give me some pointers to this issue. I am currently using svn Revision: 7401 Appreciate your help Thanks in advance -Sanjeev

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Interop list for crypto-enabled devices
by Alois Komenda 01 Feb '08

01 Feb '08

Hello, is there a document listing all devices (softphones and hardphones) which work with FreeSWITCH in encrypted (TLS + SRTP) mode? If not, could you please tell me with which devices you tested encrypted calls and what the results of these test are? Who of you got eyeBeam working with FS when encrypting signalling or/and media? Thanks a lot -- Alois Komenda Fraunhofer-Einrichtung für Systeme der Kommunikationstechnik ESK

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Re: [Freeswitch-dev] SIP Over TCP/TLS
by Imthiyaz Ahmed 31 Jan '08

31 Jan '08

Thanks Brian I am going to use sangoma modules for PSTN connectivity. I think I don't need asterisk in this scenario. Microsoft Mediation server <---------->FreeSwitch with Sangoma Modules installed. SIP over TCP/TLS <--------------->TDM That works right ? Thanks Imthiyaz

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Ultrawideband
by Brian West 31 Jan '08

31 Jan '08

Anyone know of any devices and what codecs currently support ultra- wideband? Btw portaudio@32k via FS@32k to FS@32k works. /b

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LPC10 Codec
by Brian West 30 Jan '08

30 Jan '08

Does anyone really find any use for this codec? It's currently broken in tree and wondered if it was worth the time fixing? Chime in. /b

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SIP Over TCP/TLS
by Imthiyaz Ahmed 28 Jan '08

28 Jan '08

---------- Forwarded message ---------- From: Imthiyaz Ahmed <imthiyazg(a)gmail.com> Date: Jan 28, 2008 7:07 PM Subject: SIP Over TCP/TLS To: freeswitch-dev(a)lists.freeswitch.org. Hello Does Free switch supports SIP Over TCP/TLS ? I am thinking of building a Media Gateway for OCS 2007 . Thanks Imthiyaz

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Re: [Freeswitch-dev] Call Hold ans swapping on FXS
by Anthony Minessale 28 Jan '08

28 Jan '08

I changed it to be options you have to set.    in <settings> for global or in <span> for local *note* you cannot have both of these analog-options at once because they compete. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: sanjeev mudholkar <sanjeev.mudholkar(a)gmail.com> To: freeswitch-dev(a)lists.freeswitch.org Sent: Friday, January 25, 2008 1:18:29 AM Subject: [Freeswitch-dev] Call Hold ans swapping on FXS Hi, I'am trying to use Freeswitch(with FXS) as a UA. Everything is fine for a outgoing and incoming call. 1)On doing a hold (flash) on the FXS the call is put on hold but there no SIP signaling (i.e re-invite is not sent to the peer). How can I force freeswitch to send a re-invite? 2) I'am not able to establish a consultation call as it is conferenced with the first one as soon as the second call is connected. This also prevents from swaping between calls. The call swapping feature seems to have been removed by introducing the transfer/conference feature. Is there away to achieve call swapping ? Appreciate your help. Thanks in advance -Sanjeev ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

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Re: [Freeswitch-dev] Call Hold ans swapping on FXS
by Anthony Minessale 25 Jan '08

25 Jan '08

We do not send a reinvite when you put a sip call on hold. if you have the hold-music param set in the settings section or in each span you can deliver moh to the on-hold party. The call swapping concept and the 3-way feature are competing features so we would need to design config options to expose them. Judging from your numerous feature requests on TDM you should consider sponsoring the OpenZAP project because our resources are limited and TDM users such as yourself should appreciate the goals of a BSD-licensed unified TDM library. Everything you see so far in OpenZAP has been done out of faith that sponsors will step up and try to steer the project towards it's eventual goals of bigger and better ISDN support/ss7 etc. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: sanjeev mudholkar <sanjeev.mudholkar(a)gmail.com> To: freeswitch-dev(a)lists.freeswitch.org Sent: Friday, January 25, 2008 1:18:29 AM Subject: [Freeswitch-dev] Call Hold ans swapping on FXS Hi, I'am trying to use Freeswitch(with FXS) as a UA. Everything is fine for a outgoing and incoming call. 1)On doing a hold (flash) on the FXS the call is put on hold but there no SIP signaling (i.e re-invite is not sent to the peer). How can I force freeswitch to send a re-invite? 2) I'am not able to establish a consultation call as it is conferenced with the first one as soon as the second call is connected. This also prevents from swaping between calls. The call swapping feature seems to have been removed by introducing the transfer/conference feature. Is there away to achieve call swapping ? Appreciate your help. Thanks in advance -Sanjeev ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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Call Waiting not working!!!!!
by Irshad Ahamed 25 Jan '08

25 Jan '08

Hi, I am working on Freeswitch@ rev 7309. I remember with earlier revision call waiting was working fine. But now on getting the second incoming call FXS is returning 502 (BAD gateway error). Was there any change on FXS side pertaining to number of calls allowed or something??? Freeswitch Log: 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:144 switch_core_standard_on_execute() sofia/default/6502@192.168.2.51:5060 Execute set(dialed_ext=6601) 2008-01-25 15:48:17 [DEBUG] mod_dptools.c:442 set_function() SET [dialed_ext]=[6601] 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:144 switch_core_standard_on_execute() sofia/default/6502@192.168.2.51:5060 Execute set(hangup_after_bridge=true) 2008-01-25 15:48:17 [DEBUG] mod_dptools.c:442 set_function() SET [hangup_after_bridge]=[true] 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:144 switch_core_standard_on_execute() sofia/default/6502@192.168.2.51:5060 Execute set(bypass_media=false) 2008-01-25 15:48:17 [DEBUG] mod_dptools.c:442 set_function() SET [bypass_media]=[false] 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:144 switch_core_standard_on_execute() sofia/default/6502@192.168.2.51:5060 Execute bridge(openzap/1/1) 2008-01-25 15:48:17 [ERR] mod_openzap.c:833 channel_outgoing_channel() No channels available 2008-01-25 15:48:17 [ERR] switch_ivr_originate.c:827 switch_ivr_originate() Cannot Create Outgoing Channel! cause: DESTINATION_OUT_OF_ORDER 2008-01-25 15:48:17 [DEBUG] switch_ivr_originate.c:1226 switch_ivr_originate() Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] 2008-01-25 15:48:17 [INFO] mod_dptools.c:1325 audio_bridge_function() Originate Failed. Cause: DESTINATION_OUT_OF_ORDER 2008-01-25 15:48:17 [NOTICE] mod_dptools.c:1352 audio_bridge_function() Hangup sofia/default/6502@192.168.2.51:5060 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2008-01-25 15:48:17 [DEBUG] switch_channel.c:1317 switch_channel_perform_hangup() Kill sofia/default/6502@192.168.2.51:5060 [KILL] 2008-01-25 15:48:17 [DEBUG] switch_core_session.c:643 switch_core_session_signal_state_change() Kill sofia/default/6502@192.168.2.51:5060 [BREAK] 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:413 switch_core_session_run() sofia/default/6502@192.168.2.51:5060 Running State Change CS_HANGUP 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:439 switch_core_session_run() (sofia/default/6502@192.168.2.51:5060) State HANGUP 2008-01-25 15:48:17 [DEBUG] mod_sofia.c:299 sofia_on_hangup() Channel sofia/default/6502@192.168.2.51:5060 hanging up, cause: DESTINATION_OUT_OF_ORDER 2008-01-25 15:48:17 [DEBUG] mod_sofia.c:332 sofia_on_hangup() Responding to INVITE with: 502 2008-01-25 15:48:17 [DEBUG] sofia.c:183 sofia_event_callback() event [nua_i_state] status [502][Call state] session: sofia/default/6502@192.168.2.51:5060 2008-01-25 15:48:17 [DEBUG] sofia.c:1430 sofia_handle_sip_i_state() Channel sofia/default/6502@192.168.2.51:5060 entering state [terminated] 2008-01-25 15:48:17 [DEBUG] switch_core_state_machine.c:45 switch_core_standard_on_hangup() Standard HANGUP sofia/default/6502@192.168.2.51:5060, cause: DESTINATION_OUT_OF_ORDER 2008-01-25 15:48:17 [DEBUG] switch_core_session.c:701 switch_core_session_thread() Session 3 (sofia/default/6502@192.168.2.51:5060) Locked, Waiting on external entities 2008-01-25 15:48:17 [NOTICE] switch_core_session.c:707 switch_core_session_thread() Session 3 (sofia/default/6502@192.168.2.51:5060) Ended 2008-01-25 15:48:17 [NOTICE] switch_core_session.c:709 switch_core_session_thread() Close Channel sofia/default/6502@192.168.2.51:5060 [CS_HANGUP] 2008-01-25 15:48:36 [DEBUG] zap_analog.c:665 process_event() EVENT [ONHOOK][1:1] STATE [UP] 2008-01-25 15:48:36 [DEBUG] zap_analog.c:691 process_event() Changing state on 1:1 from UP to DOWN 2008-01-25 15:48:36 [DEBUG] zap_analog.c:351 zap_analog_channel_run() Executing state handler on 1:1 for DOWN 2008-01-25 15:48:36 [DEBUG] mod_openzap.c:1037 on_fxs_signal() got FXS sig [STOP] 2008-01-25 15:48:36 [NOTICE] mod_openzap.c:1115 on_fxs_signal() Hangup OpenZAP/777 [CS_EXECUTE] [NORMAL_CLEARING] 2008-01-25 15:48:36 [DEBUG] switch_channel.c:1317 switch_channel_perform_hangup() Kill OpenZAP/777 [KILL] 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:643 switch_core_session_signal_state_change() Kill OpenZAP/777 [BREAK] 2008-01-25 15:48:36 [DEBUG] zap_analog.c:649 zap_analog_channel_run() ANALOG CHANNEL thread ended. 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:186 audio_bridge_thread() read: OpenZAP/777 Bad Frame.... Bubye! 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:358 switch_core_session_receive_message() Kill OpenZAP/777 [BREAK] 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:224 audio_bridge_thread() Kill sofia/default/777(a)sipx.oneconv.com [BREAK] 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:227 audio_bridge_thread() BRIDGE THREAD DONE [OpenZAP/777] 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:693 switch_ivr_multi_threaded_bridge() sofia/default/777(a)sipx.oneconv.com State Change CS_LOOPBACK -> CS_RESET 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:643 switch_core_session_signal_state_change() Kill sofia/default/777(a)sipx.oneconv.com [BREAK] 2008-01-25 15:48:36 [DEBUG] switch_core_state_machine.c:413 switch_core_session_run() OpenZAP/777 Running State Change CS_HANGUP 2008-01-25 15:48:36 [DEBUG] switch_core_state_machine.c:439 switch_core_session_run() (OpenZAP/777) State HANGUP 2008-01-25 15:48:36 [DEBUG] mod_openzap.c:418 channel_on_hangup() OpenZAP/777 CHANNEL HANGUP 2008-01-25 15:48:36 [DEBUG] switch_core_state_machine.c:45 switch_core_standard_on_hangup() Standard HANGUP OpenZAP/777, cause: NORMAL_CLEARING 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:701 switch_core_session_thread() Session 1 (OpenZAP/777) Locked, Waiting on external entities 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:180 audio_bridge_thread() write: OpenZAP/777 Bad Frame....[160] Bubye! 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:358 switch_core_session_receive_message() Kill sofia/default/777(a)sipx.oneconv.com [BREAK] 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:224 audio_bridge_thread() Kill OpenZAP/777 [BREAK] 2008-01-25 15:48:36 [DEBUG] switch_ivr_bridge.c:227 audio_bridge_thread() BRIDGE THREAD DONE [sofia/default/777(a)sipx.oneconv.com] 2008-01-25 15:48:36 [NOTICE] switch_ivr_bridge.c:257 audio_bridge_on_loopback() Hangup sofia/default/777(a)sipx.oneconv.com[CS_RESET] [NORMAL_CLEARING] 2008-01-25 15:48:36 [NOTICE] switch_core_session.c:707 switch_core_session_thread() Session 1 (OpenZAP/777) Ended 2008-01-25 15:48:36 [NOTICE] switch_core_session.c:709 switch_core_session_thread() Close Channel OpenZAP/777 [CS_HANGUP] 2008-01-25 15:48:36 [DEBUG] switch_channel.c:1317 switch_channel_perform_hangup() Kill sofia/default/777(a)sipx.oneconv.com[KILL] 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:643 switch_core_session_signal_state_change() Kill sofia/default/777(a)sipx.oneconv.com [BREAK] 2008-01-25 15:48:36 [DEBUG] switch_core_state_machine.c:413 switch_core_session_run() sofia/default/777(a)sipx.oneconv.com Running State Change CS_HANGUP 2008-01-25 15:48:36 [DEBUG] switch_core_state_machine.c:439 switch_core_session_run() (sofia/default/777(a)sipx.oneconv.com) State HANGUP 2008-01-25 15:48:36 [DEBUG] mod_sofia.c:299 sofia_on_hangup() Channel sofia/default/777(a)sipx.oneconv.com hanging up, cause: NORMAL_CLEARING 2008-01-25 15:48:36 [DEBUG] mod_sofia.c:325 sofia_on_hangup() Sending BYE to sofia/default/777(a)sipx.oneconv.com 2008-01-25 15:48:36 [DEBUG] switch_core_state_machine.c:45 switch_core_standard_on_hangup() Standard HANGUP sofia/default/777(a)sipx.oneconv.com, cause: NORMAL_CLEARING 2008-01-25 15:48:36 [DEBUG] switch_core_session.c:701 switch_core_session_thread() Session 2 (sofia/default/777(a)sipx.oneconv.com) Locked, Waiting on external entities 2008-01-25 15:48:36 [NOTICE] switch_core_session.c:707 switch_core_session_thread() Session 2 (sofia/default/777(a)sipx.oneconv.com) Ended 2008-01-25 15:48:36 [NOTICE] switch_core_session.c:709 switch_core_session_thread() Close Channel sofia/default/777(a)sipx.oneconv.com [CS_HANGUP] Thanks in advance... BR, Irshad.

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