- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] email core dump
by Mark Campbell-Smith 24 Jun '09

24 Jun '09

Thanks Brian, but still no luck with the email.. I have configured exim4 so that I can send messages from the command line using 'mail' command and these are sent successfully. I still get a core dump in the log when freeswitch is trying to send the mail: /bin/cat: write error: Broken pipe sh: line 1: 4492 Done(1) /bin/cat /tmp/mail.1245811149abdc 4493 Segmentation fault (core dumped) | /usr/local/bin/eximcompat.sh -t xxx(a)xx.com 2009-06-24 12:39:09.285351 [DEBUG] switch_utils.c:554 Emailed file [/tmp/mail.1245811149abdc] to [xxx(a)xx.com] 2009-06-24 12:39:09.285351 [DEBUG] mod_voicemail.c:2491 Sending message to xxx(a)xx.com eximcompat.sh is as described on the wiki: freeswitch:/# cat /usr/local/bin/eximcompat.sh #!/bin/bash exec exim4 -t Any other thoughts? From: Brian West <brian(a)freeswitch.org> > Subject: Re: [Freeswitch-users] email core dump > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: <7C7A8ED9-ECED-4100-87F6-0875C054EC64(a)freeswitch.org> > Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes > > http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings > > /b > > On Jun 21, 2009, at 6:27 AM, Mark Campbell-Smith wrote: > > > Hi! > > > > I am trying to email from > > 2009-06-21 20:43:24.273895 [DEBUG] switch_core_codec.c:122 Restore > > original codec. > > 2009-06-21 20:43:24.273895 [DEBUG] mod_voicemail.c:2321 Deliver VM to > > 1000(a)192.168.0.20 > > /bin/cat: write error: Broken pipe > > sh: line 1: 11975 Done(1) /bin/cat /tmp/mail. > > 124558382500b1 > > 11976 Segmentation fault (core dumped) | exim4 -t > myemail(a)xx.com > > 2009-06-21 20:43:24.670899 [DEBUG] switch_utils.c:554 Emailed file > > [/tmp/mail.12455810042c7f] to [myemail(a)xx.com] > > > > I can manually send an email to myself with exim4, but freeswitch > > fails. > > > > Any ideas what I have configured incorrectly? > > > > Thanks > > >

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Re: [Freeswitch-users] Transmit fax locally for test
by Tim B 23 Jun '09

23 Jun '09

Did anyone have any suggestions on this? Just to reiterate... - 8000 is a local extension defined in the default dialplan... see http://pastebin.freeswitch.org/9450 for definition - didn't work: originate sofia/default/8000(a)192.168.10.35 &txfax(storage/fax/test.tif) ... see http://pastebin.freeswitch.org/9440 for log - had to add the FS ip (192.168.10.35) to the domains acl... now it to works <list name="domains" default="deny"> <node type="allow" cidr="192.168.10.35/32"/> <node type="allow" domain="$${domain}"/> </list> Is this the proper way to configure? Tim From: timb0311(a)hotmail.com To: freeswitch-users(a)lists.freeswitch.org Subject: RE: Transmit fax locally for test Date: Mon, 22 Jun 2009 18:37:47 -0400 8000 is a local extension defined in the default dialplan. Tim > ------------------------------ > > Message: 2 > Date: Mon, 22 Jun 2009 15:05:20 -0400 > From: Brian West <brian(a)freeswitch.org> > Subject: Re: [Freeswitch-users] Transmit fax locally for test > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: <8618988E-BB27-4400-BDDF-99C87A26FE67(a)freeswitch.org> > Content-Type: text/plain; charset="us-ascii" > > what is 8000? is it local or is it a remote endpoint? > > /b > > On Jun 22, 2009, at 3:01 PM, Tim B wrote: > > > > > originate sofia/default/8000(a)192.168.10.35 &txfax(storage/fax/ > > test.tif) > > Brian West > brian(a)freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > Insert movie times and more without leaving Hotmail®. See how. _________________________________________________________________ Microsoft brings you a new way to search the web. Try Bing™ now http://www.bing.com?form=MFEHPG&publ=WLHMTAG&crea=TEXT_MFEHPG_Core_tagline_…

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Problem with handling unanswered calls for a managed redirect
by Richard Lamkin 23 Jun '09

23 Jun '09

Can anyone suggest a good way to do the following; 1 - I want to be alerted [via the event API] to a new incoming call. 2 - I do not want that incoming call to be answered but just stay ringing. 3 - Then via the API I want to send a redirect command to push the call off to a new destination of my choice, I do not want to use the answer/deflect sequence. So far I've managed 1 - I see on the incoming call on the event API 2 - I used sleep 180000 (3 mins) see rule below. 3 - failed - because the rule is executing a sleep command and I cannot break in with my redirect. <extension name="Trunk_Line1"> <condition field="destination_number" expression="^012345$"> <action application="set" data="domain_name=x.x.x.x"/> <action application="ring_ready" /> <action application="sleep" data="180000"/> </condition> </extension> ============ I have tested the following works as single DP rule. Using the fixed dial plan rule below I do get the SIP signalling I want but of course it's a redirect immediately and to a fixed destination. The redirect causes FS to send a "302 moved temporarily", and the move works. <extension name="Trunk_Line1"> <condition field="destination_number" expression="^012345$"> <action application="set" data="domain_name=x.x.x.x"/> <action application="ring_ready" /> <action application="set" data="effective_caller_id_number= 00123456789"/> <action application="set" data="effective_caller_id_name= fred"/> <action application="redirect" data="sip:012345678901@${domain_name}"/> </condition> </extension> ============= Any suggestions would be gratefully received Richard Lamkin richard.lamkin(a)mettoni.com ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN *************************************************************************

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Polycom configuration problems?
by Lars Zeb 23 Jun '09

23 Jun '09

I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines on the phone. The first two are registered with a SwitchVox, the last with Freeswitch. When I select the 3rd line and begin to press numbers, pressing the 3rd digit automatically causes the phone to begin to dial. It does not matter which three numbers I press, the 3rd one is magic. However, if I do not select a line before dialing and key a 10-digit number into the phone, then select the 3rd line, it dials out fine. You can see from the debug console output that Processing begins before it hits any dialplan, so that cannot be the problem. I must have the line defined incorrectly for Freeswitch. Thanks for any suggestions, Lars. PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth. 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl "domains". Falling back to Digest auth. 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001(a)192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001(a)192.168.10.29) Running State Change CS_NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/1001(a)192.168.10.29 entering state [received][100] 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=- 1245682011 1245682011 IN IP4 192.168.10.101 s=Polycom IP Phone c=IN IP4 192.168.10.101 t=0 0 m=audio 2254 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/1001(a)192.168.10.29) State NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:115:32000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:107:16000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec sofia/internal/1001(a)192.168.10.29 PCMU/8000 20 ms 160 samples 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload to 101 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/1001(a)192.168.10.29) State Change CS_NEW -> CS_INIT 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001(a)192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001(a)192.168.10.29) Running State Change CS_INIT 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001(a)192.168.10.29) State INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/1001(a)192.168.10.29 SOFIA INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/1001(a)192.168.10.29) State Change CS_INIT -> CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1001(a)192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1001(a)192.168.10.29) State INIT going to sleep 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1001(a)192.168.10.29) Running State Change CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/1001(a)192.168.10.29) State ROUTING 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/1001(a)192.168.10.29 SOFIA ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 sofia/internal/1001(a)192.168.10.29 Standard ROUTING 2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing 1001->323 in context default Dialplan: sofia/internal/1001(a)192.168.10.29 parsing [default->unloop] continue=false Dialplan: sofia/internal/1001(a)192.168.10.29 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1001(a)192.168.10.29 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false

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voicemail problem
by Mark Campbell-Smith 23 Jun '09

23 Jun '09

Hi! I have a problem with voicemail in that freeswitch fails to let users leave their message. Something wrong in the config I guess. I see this in the logs: 2009-06-21 11:28:32.202912 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-06-21 11:28:32.238744 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-record_message] (en:en) 2009-06-21 11:28:32.290756 [ERR] mod_native_file.c:68 Error opening /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-record_message.PCMU 2009-06-21 11:28:32.402826 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-06-21 11:28:32.434733 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-goodbye.wav] (en:en) I assume the vm-record_message.PCMU is the file that will be created to record the voicemail. Is that correct and how can I fix this? Thanks!

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Help with Socket event again
by Max Bridgewater 23 Jun '09

23 Jun '09

Any help our there? I'm still trying to get this piece working. Essentially what i wan to do is, when a call comes in (from registered devices as well as unregistered devices), notify the my server socket. Somehow it's not working. The change i made compared to the standard Freeswitch settings are the following: 1) Added following extension that in /usr/local/freeswitch/conf/dialplan/public/mysocket.xml: <include> <extension name="mysocket"> <condition field="destination_number" expression="^242.*$" > <action application="socket" data="192.168.50.67:10000 full" /> </condition> </extension> </include> 2) Changed file: /usr/local/freeswitch/conf/autoload_configs/event_socket.conf to: <configuration name="event_socket.conf" description="Socket Client"> <settings> <param name="nat-map" value="false"/> <param name="listen-ip" value="0.0.0.0"/> <param name="listen-port" value="8021"/> <param name="password" value="1234"/>  </settings> </configuration> I noticed that with this extension, all calls received from external providers (e.g. Les.net, Gafachi, etc.) are indeed forwarded to my socket. But calls from registered devices and initiated using the socket interface are not forwarded. Is there something that need to be changed in the profiles? or is something wrong with my dial string? {origination_caller_id_number=12000}sofia/internal/242424(a)192.168.1.62. In the logs, i cannot see that that my extension is being matched. Any idea, Max.

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ClueCon 2009 - Important Information
by Michael Collins 23 Jun '09

23 Jun '09

I know you are all eagerly anticipating the arrival of the coolest conference around! We want to make sure that everyone is aware of the following information: * The last day to get the early-bird registration is Wednesday, July 1. Early birds get into the conference for only $499. After July 1 the price is $699 per person. Please call 877.742.CLUE and get registered today! * The last day to book a hotel room at the Wyndham is Tuesday, July 21. Be sure to use expedia.com to get the best deal available. The ClueCon team is working hard to make this a very special event and we hope to have more announcements soon. You don't want to miss ClueCon 2009 - it will be the best conference you attend this year, bar none! -The ClueCon Team http://www.cluecon.com 877.742.CLUE

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Sound file or lua script not played under load
by Nik Middleton 23 Jun '09

23 Jun '09

Hi Guys, Scratching my head on this one, under load FS is not playing an audio file, OR and lua script is not getting executed. Not all the time, just some. I've changed ulimit -n to 99999 but no diff, and ideas where else I might look? Regards,

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channel variable sip_to_tag
by Christian Löschenkohl 23 Jun '09

23 Jun '09

hello do someone know how to get the sip_to_tag from an active call? the sip_from_tag is available as a channel variable but sip_to_tag isn't. i don't know if it is available at call setup, the fist time i see the tag=... in the sip header is the challenge response answer from fs i need this to get my aoc (advice-of-charge) implementation running, this one is based on sip info messages and has to contain the same tag's as the active call. br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

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-ERR SERVICE_NOT_IMPLEMENTED
by Edmar Cruz 23 Jun '09

23 Jun '09

Hi, API CALL [originate sofia/external/1001(a)116.50.456.212] -ERR SERVICE_NOT_IMPLEMENTED I receiving this error i dont know y? Can u help mo on this? I dialing a mobile number on this sometimes it works... Sometimes it destroys the call [CALL_DESTROY] Thanks -- View this message in context: http://www.nabble.com/-ERR-SERVICE_NOT_IMPLEMENTED-tp24143545p24143545.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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