February 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Greetings and a couple of questions
by Doc 18 Feb '10

18 Feb '10

First, to all, greetings. I am just beginning a quest in trying to setup a simple FS box to route my incoming skype account to a SIP ATA (SPA-1001). I have this installed on Ubuntu 8.04 (since I read that it came with a stock tickless kernel with 100HZ tick). I have (as far as I can tell) compiled the alsa 1.0.20 drivers and included the mod_skypiax dummy file. I followed install instructions in the FS wiki for Hardy & FS & skypiax. 1) I am able to see a call come in, and it gets routed to the sample IVR to start. The first thing off is that when I dial from PSTN -> Skype-In, it does not let me push any buttons. If I launch a second skype client, and dial the skype user on FS directly, it works fine. Any ideas? 2) No matter how Skype works, when I hang up, it throws 3 or 4 errors (2010-02-16 20:41:45.384633 [ERR] mod_skypiax.c:826 rev 16619M[(nil)|37 ][ERRORA 826 ][interface1][-1, 5,21] EXIT? sent=-1), and when I try subsequent dials, I get pages and pages of this error, and no audio. Eventually it crashes FS. Any ideas? 3) When dialing extension to extension, or even testing out the IVR, it all works fine - no errors. The only thing it does make me do, and I haven't tracked it down yet is it waits like 30 seconds before responding (unless I press the # after I type the extension number). Any ideas? 4) When I run startskype.sh, I see these errors (and are they worth concerning?) : expected keysym, got XF86KbdLightOnOff: line 70 of pc expected keysym, got XF86KbdBrightnessDown: line 71 of pc expected keysym, got XF86KbdBrightnessUp: line 72 of pc Could not init font path element /usr/share/fonts/X11/cyrillic, removing from list! Could not init font path element /var/lib/defoma/x-ttcidfont-conf.d/dirs/TrueType, removing from list! ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM null ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM null Thanks for the patience in letting me email to the group. I used Asterisk in the past, and even though I am finding a bit of a learning curve using FS, I am enjoying it. Hopefully someone will have some insight at least into where or how I can settle the above items down, and I can start building a proper dialplan. Thanks, Doc

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Re: [Freeswitch-users] One more thing..
by Anthony Minessale 18 Feb '10

18 Feb '10

Are you doing this with the latest revision? You would have to supply more info like a console trace on debug level with siptrace enabled. On Feb 17, 2010 1:51 AM, "Doc" <ledoktre(a)meanie.us> wrote: Greetings one more time, I just remembered. I also ran into one other issue in my testing. When I route the incoming skype call to an extension (Sipura SPA 1001), it plays hold music on the callers side (good....), and when I pick up the extension, calling party drops, internal phone is left with a fast busy signal : 2010-02-16 21:09:31.687204 [NOTICE] sofia.c:4690 Channel [sofia/internal/sip:1001@10.24.72.12:5060] has been answered 2010-02-16 21:09:31.987714 [WARNING] mod_sofia.c:918 We were told to use ptime 3 but what they meant to say was 20 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.. 2010-02-16 21:09:32.027243 [ERR] switch_core_io.c:121 sofia/internal/sip:1001@10.24.72.12:5060 has no read codec. First thing I notice is this unusual error about the ptime? and the one that hangs me up I think is the "sofia... has no read codec". I have been poking around my configuration - any suggestions? Thanks, Doc _______________________________________________ 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] One more thing..
by Anthony Minessale 18 Feb '10

18 Feb '10

One hint is to add keywords to the subject like "problems with skypeiax" so the guy will see your issue he doesn't read every mail looking for skype issues. Even better, open a new issue on jira.freeswitch.org under skypeiax related category...... On Feb 17, 2010 6:23 PM, "Doc" <ledoktre(a)meanie.us> wrote: I have updated to "FreeSWITCH Version 1.0.trunk (16679)" which seems to have cleared up part # 2 of my question (issues with the errors being thrown on the screen during a call, and then subsequent calls failing). About the DTMF issues, for fun, I got onto Skype chat. I was told by the technician there that Skype does not support DTMF on SkypeIn -OR- Skype for SIP. He said sorry, but you won't get anywhere. I was also told they are not doing any blocking, that the reason its not working is probably due to all the transcoding that is taking place has degraded the signal too much. I guess I might try and see if there is a module available for a voice driven IVR? (any suggestions?) And on the rest of my first posting, any thoughts on why I must push a # sign when dialing an extension, etc, in FS? If I don't, I have to wait like 30 seconds. Also, the errors when I run startskype,sh (I am thinking must be no big deal) listed, are they any problem : expected keysym, got XF86KbdLightOnOff: line 70 of pc expected keysym, got XF86KbdBrightnessDown: line 71 of pc expected keysym, got XF86KbdBrightnessUp: line 72 of pc Could not init font path element /usr/share/fonts/X11/cyrillic, removing from list! Could not init font path element /var/lib/defoma/x-ttcidfont-conf.d/dirs/TrueType, removing from list! ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM null ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM null Thanks again you guys for all your help! Doc Anthony Minessale wrote: > keep updating, the maintainer of mod_skypeiax is adding new patches > e... > You can also join irc on irc.freenode.net <http://irc.freenode.net> > #freeswitch and #freeswitch-dev to interact with him live. > > > One hint around here when we ask ... _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists...

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Calls being parked on DTMF (2)
by Eelco Brölman 17 Feb '10

17 Feb '10

Hi all, I have the same problems as described in the thread "Calls being parked on DTMF". We have a lab-test setup, which consist of the following: 1) a box with 4 E1 trunks with a call generator application (let's call it CG) 2) connected with a FreeSwitch box (Sangoma A104D card) (let's call that FS) The CG dials a number on the FS box (number 921000), which is bridged out back to the CG server (number 1000). The initiating application sends some DTMF, which is replied (echo-ed) by the receiving application (both on the CG server). In roughly about 80% of the calls, FS just disconnects the call: 2010-02-17 14:30:14.677125 [DEBUG] switch_ivr_originate.c:3105 Originate Resulted in Success: [OpenZAP/1:1/1000] 2010-02-17 14:30:14.677125 [DEBUG] switch_core_session.c:638 Send signal OpenZAP/1:1/1000 [BREAK] 2010-02-17 14:30:14.677125 [DEBUG] switch_core_session.c:638 Send signal OpenZAP/1:60/921000 [BREAK] 2010-02-17 14:30:14.677125 [DEBUG] switch_ivr_bridge.c:1175 (OpenZAP/1:1/1000) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2010-02-17 14:30:14.677125 [DEBUG] switch_core_session.c:1018 Send signal OpenZAP/1:1/1000 [BREAK] 2010-02-17 14:30:14.677125 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/1000) Running State Change CS_EXCHANGE_MEDIA 2010-02-17 14:30:14.677125 [DEBUG] switch_core_state_machine.c:351 (OpenZAP/1:1/1000) State EXCHANGE_MEDIA 2010-02-17 14:30:14.677125 [DEBUG] mod_openzap.c:558 CHANNEL EXCHANGE_MEDIA 2010-02-17 14:30:14.687135 [DEBUG] switch_core_session.c:699 Send signal OpenZAP/1:1/1000 [BREAK] 2010-02-17 14:30:14.687135 [DEBUG] switch_core_session.c:699 Send signal OpenZAP/1:60/921000 [BREAK] 2010-02-17 14:30:15.986882 [DEBUG] mod_openzap.c:684 queue DTMF [7] 2010-02-17 14:30:16.026733 [DEBUG] zap_io.c:1870 1:1 GENERATE DTMF [7] 2010-02-17 14:30:16.106418 [DEBUG] switch_ivr_bridge.c:469 OpenZAP/1:1/1000 ending bridge by request from write function 2010-02-17 14:30:16.106418 [DEBUG] switch_core_session.c:638 Send signal OpenZAP/1:60/921000 [BREAK] 2010-02-17 14:30:16.126335 [DEBUG] switch_ivr_bridge.c:475 OpenZAP/1:1/1000 ending bridge by request from read function 2010-02-17 14:30:16.126335 [DEBUG] switch_core_session.c:638 Send signal OpenZAP/1:1/1000 [BREAK] 2010-02-17 14:30:16.126335 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD DONE [OpenZAP/1:60/921000] 2010-02-17 14:30:16.126335 [DEBUG] switch_ivr_bridge.c:582 Send signal OpenZAP/1:1/1000 [BREAK] 2010-02-17 14:30:16.126335 [DEBUG] switch_ivr_bridge.c:562 BRIDGE THREAD DONE [OpenZAP/1:1/1000] 2010-02-17 14:30:16.126335 [DEBUG] switch_ivr_bridge.c:582 Send signal OpenZAP/1:60/921000 [BREAK] 2010-02-17 14:30:16.126335 [NOTICE] switch_ivr_bridge.c:634 Hangup OpenZAP/1:1/1000 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2010-02-17 14:30:16.126335 [DEBUG] switch_channel.c:2063 Send signal OpenZAP/1:1/1000 [KILL] Where the line "ending bridge by request from write function" is curious of course. I'm running: - Freeswitch trunk (r16674) (updated, make clean, reconfigured and make install this morning) - Wanpipe 3.5.10.smg-2 (see http://wiki.sangoma.com/wanpipe-SmgPriInstallation) - Using the ozmod_sangoma_boost SCTP If anyone needs more information, tests or config files, I'll be happy to provide them. Kind regards, Eelco Brölman

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FreeSWITCH.Managed.dll deletes on make distclean
by Puskás Zsolt 17 Feb '10

17 Feb '10

I usually do svn-clean than svn up when i compile a new version of fs. I noticed that if i do make distclean the file @ src/mod/languages/mod_managed/managed/FreeSWITCH.Managed.dll got deleted. When i do svn up it gets 'restored': Restored 'src/mod/languages/mod_managed/managed/FreeSWITCH.Managed.dll' Is this normal ? I don't use mod_managed btw.

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Re: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA Support
by Anthony Minessale 17 Feb '10

17 Feb '10

If by properly you mean the previously supported way then yes. We added support for the broadsoft method now as the default because its supported on more phones. You may want to try setting your phones to that mode to compare but setting the profile param to sylantro should restore the previous default behaviour. On Feb 17, 2010 8:46 AM, "Chris Chen" <chris.chen2004(a)gmail.com> wrote: Hi Tony, do you mean in the internal.xml under /usr/local/freeswitch/local/sip_profiles we should set <param name="manage-shared- appearance" value="sylantro"/> to get the SCA mode working properly? Please confirm Thanks, Chris On Wed, Feb 17, 2010 at 8:57 AM, Anthony Minessale < anthony.minessale(a)gmail.com> wrote: > > You h... _______________________________________________ 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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Announcement: FreeSWITCH Add Broadsoft SCA Support
by Michael Collins 17 Feb '10

17 Feb '10

We want to let everyone know that FreeSWITCH now supports the Broadsoft SCA method of doing shared lines. The story is here: http://www.freeswitch.org/node/227 Tony and Brian spent many hours laboring over this, so please be sure to show your appreciation to them for this new feature and all of the great things they do for the FreeSWITCH community and VoIP in general! -Michael

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freeswitch as proxy
by Madovsky 17 Feb '10

17 Feb '10

Hi, First I'd like to felicitate the huge work of the freeswitch founders, their patience and humbleness (Anthony Minessale (is there 1 and 2 ?), Michael Jerris, Brian West and Others ) and people need to know and understand how hard is to maintain open source for years... well, maybe I didn't look for very well in archives and google so I hope my request won't be an old one.... I set freeswitch with proxy_media and blind reg and auth on true since I want everybody who uses my network use a softphone to register their own sip account (whatever domain outside my network). user@hisdomain -> myproxy IP -> hisdomain registrar -> confirm register and wait call. user@hisdomain calls other@otherdomain everywhere in the world. other@otherdomain can also call user@hisdomain is it possible ? if yes is there any link example ? I guess maybe it's a dialplan rule.... Sorry I'm novice yet Regards Franck

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One more thing..
by Doc 17 Feb '10

17 Feb '10

Greetings one more time, I just remembered. I also ran into one other issue in my testing. When I route the incoming skype call to an extension (Sipura SPA 1001), it plays hold music on the callers side (good....), and when I pick up the extension, calling party drops, internal phone is left with a fast busy signal : 2010-02-16 21:09:31.687204 [NOTICE] sofia.c:4690 Channel [sofia/internal/sip:1001@10.24.72.12:5060] has been answered 2010-02-16 21:09:31.987714 [WARNING] mod_sofia.c:918 We were told to use ptime 3 but what they meant to say was 20 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.. 2010-02-16 21:09:32.027243 [ERR] switch_core_io.c:121 sofia/internal/sip:1001@10.24.72.12:5060 has no read codec. First thing I notice is this unusual error about the ptime? and the one that hangs me up I think is the "sofia... has no read codec". I have been poking around my configuration - any suggestions? Thanks, Doc

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mod_fax receives fax to file but logs error msg
by Robert Hadley 17 Feb '10

17 Feb '10

I have been playing around with mod_fax and can successfully receive a fax to file. However, while doing so mod_fax is logging an error message. Does anybody know what this error means? Dialplan: OpenZAP/2:1/1011 parsing [default->REH_test_fax_receive] continue=false Dialplan: OpenZAP/2:1/1011 Regex (PASS) [REH_test_fax_receive] destination_number(1011) =~ /^1011$/ break=on-false Dialplan: OpenZAP/2:1/1011 Action disable_ec() Dialplan: OpenZAP/2:1/1011 Action answer() Dialplan: OpenZAP/2:1/1011 Action playback(silence_stream://2000) Dialplan: OpenZAP/2:1/1011 Action rxfax(/tmp/rxfax.tif) Dialplan: OpenZAP/2:1/1011 Action hangup() 2010-02-16 16:27:08.542746 [DEBUG] switch_core_state_machine.c:122 (OpenZAP/2:1/1011) State Change CS_ROUTING -> CS_EXECUTE 2010-02-16 16:27:08.542746 [DEBUG] switch_core_session.c:1019 Send signal OpenZAP/2:1/1011 [BREAK] 2010-02-16 16:27:08.542746 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/2:1/1011) State ROUTING going to sleep 2010-02-16 16:27:08.542746 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/2:1/1011) Running State Change CS_EXECUTE 2010-02-16 16:27:08.542746 [DEBUG] switch_core_state_machine.c:348 (OpenZAP/2:1/1011) State EXECUTE 2010-02-16 16:27:08.542746 [DEBUG] mod_openzap.c:434 OpenZAP/2:1/1011 CHANNEL EXECUTE 2010-02-16 16:27:08.542746 [DEBUG] switch_core_state_machine.c:159 OpenZAP/2:1/1011 Standard EXECUTE 2010-02-16 16:27:08.542746 [DEBUG] switch_core_session.c:1521 Application disable_ec Requires media! pre_answering channel OpenZAP/2:1/1011 2010-02-16 16:27:08.542746 [DEBUG] switch_core_session.c:1523 OpenZAP/2:1/1011 receive message [PROGRESS] 2010-02-16 16:27:08.542746 [DEBUG] mod_openzap.c:960 Changing state on 2:1 from IDLE to UP 2010-02-16 16:27:08.542746 [NOTICE] mod_openzap.c:961 Channel [OpenZAP/2:1/1011] has been answered 2010-02-16 16:27:08.542746 [DEBUG] switch_channel.c:182 OpenZAP/2:1/1011 receive message [AUDIO_SYNC] 2010-02-16 16:27:08.542746 [DEBUG] switch_core_session.c:634 Send signal OpenZAP/2:1/1011 [BREAK] EXECUTE OpenZAP/2:1/1011 disable_ec() 2010-02-16 16:27:08.542746 [INFO] mod_openzap.c:2951 Echo Canceller Disabled EXECUTE OpenZAP/2:1/1011 answer() EXECUTE OpenZAP/2:1/1011 playback(silence_stream://2000) 2010-02-16 16:27:08.542746 [DEBUG] switch_ivr_play_say.c:1162 Codec Activated L16@8000hz 1 channels 20ms 2010-02-16 16:27:08.542746 [DEBUG] switch_core_io.c:652 OpenZAP/2:1/1011 receive message [TRANSCODING_NECESSARY] 2010-02-16 16:27:08.562747 [DEBUG] ozmod_analog.c:450 Executing state handler on 2:1 for UP 2010-02-16 16:27:08.562747 [DEBUG] mod_openzap.c:1463 got FXS sig [UP] 2010-02-16 16:27:10.522277 [DEBUG] switch_ivr_play_say.c:1454 done playing file EXECUTE OpenZAP/2:1/1011 rxfax(/tmp/rxfax.tif) 2010-02-16 16:27:10.522277 [DEBUG] mod_fax.c:591 Raw read codec activation Success L16 20000 2010-02-16 16:27:10.522277 [DEBUG] switch_core_codec.c:112 OpenZAP/2:1/1011 Push codec L16:10 2010-02-16 16:27:10.522277 [DEBUG] mod_fax.c:607 Raw write codec activation Success L16 2010-02-16 16:27:10.522277 [DEBUG] switch_channel.c:182 OpenZAP/2:1/1011 receive message [AUDIO_SYNC] 2010-02-16 16:27:10.542275 [DEBUG] switch_core_io.c:234 OpenZAP/2:1/1011 receive message [TRANSCODING_NECESSARY] 2010-02-16 16:28:14.982013 [DEBUG] ozmod_analog.c:788 EVENT [ONHOOK][2:1] STATE [UP] 2010-02-16 16:28:14.982013 [DEBUG] ozmod_analog.c:824 Changing state on 2:1 from UP to DOWN 2010-02-16 16:28:14.992014 [DEBUG] ozmod_analog.c:450 Executing state handler on 2:1 for DOWN 2010-02-16 16:28:14.992014 [DEBUG] mod_openzap.c:1463 got FXS sig [STOP] 2010-02-16 16:28:14.992014 [NOTICE] mod_openzap.c:1554 Hangup OpenZAP/2:1/1011 [CS_EXECUTE] [NORMAL_CLEARING] 2010-02-16 16:28:14.992014 [DEBUG] switch_channel.c:1976 Send signal OpenZAP/2:1/1011 [KILL] 2010-02-16 16:28:15.002011 [ERR] mod_fax.c:666 Cannot write frame [datalen: 320, samples: 160] 2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:167 ============================================================================ == 2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:174 Fax successfully received. 2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:185 Remote station id: 206 742 3831 2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:186 Local station id: SpanDSP Fax Ident 2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:187 Pages transferred: 1 2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:189 Total fax pages: 1 2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:190 Image resolution: 8031x3850 2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:191 Transfer Rate: 9600 2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:193 ECM status off 2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:194 remote country: 2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:195 remote vendor: 2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:196 remote model: 2010-02-16 16:28:15.002011 [DEBUG] mod_fax.c:198 ============================================================================ == I traced the error message as coming from switch_core_session_write_frame in src/switch_core_io.c which returns SWITCH_STATUS_FALSE. Thanks for any information, Robert

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