February 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

http://jira.freeswitch.org/browse/FS-6240
by Brian West 19 Feb '14

19 Feb '14

This patch needs testers, Please anyone that is interested test and feedback on the JIRA comments. -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

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ClueCon Weekly Today Adhearsion and mod_rayo update, and the FreeSWITCH Dev Team Discussing ICMP MTU and related networking Issues
by Ken Rice 19 Feb '14

19 Feb '14

Greetings FreeSWITCHers, Join us today at 1PM EST (10AM PST) for ClueCon Weekly! Today we¹ll have Ben and Ben from Adhearsion along with Chris Rienzo author of mod_rayo updating us on their latest developments. Following the Adearsion Crew, the FreeSWITCH Core Dev Team will take over to with a discussion of ICMP and MTU related networking issues. Join us today! Call sip:888@conference.freeswitch.org or hit http://fs0.us/call888 for PSTN access numbers! We¹ll be starting promptly so don¹t be late! K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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Auth by ACL/CIDR doesn't find the right user from directory
by Biswas, Dipak 19 Feb '14

19 Feb '14

Hi All, I guess this issue has been discussed before but I could not find a suitable solution. I've this one user with the CIDR declared in directory/default/1000.xml <include> <user id="1000" cidr="12.34.56.78/32,20.0.0.0/8">  <params> <param name="password" value="1234"/> <param name="vm-password" value="1000"/> </params> <variables> <variable name="accountcode" value="1000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1000"/> <variable name="effective_caller_id_number" value="1000"/> </variables> </user> Another user 2000 is defined but no CIDR is added there. <user id="2000" cidr="12.34.56.78/32,20.0.0.0/8"> <params> <param name="password" value="1234"/> <param name="vm-password" value="2000"/> </params> <variables> <variable name="accountcode" value="2000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 2000"/> <variable name="effective_caller_id_number" value="2000"/> </variables> </user> Now, when user 2000 dials out to another extension, ACL is still authenticating the user as 1000. How can I associate 2000 and it's associated params and variables to this call instead of 1000's? -- Thanks, Dipak

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Stream file to both legs
by Alex Lake 19 Feb '14

19 Feb '14

I'm sure I've done this before, but can't find it now - how can I play a file to more than one channel (eg. an a-leg and b-leg)

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inherit_codec fails in recent FS GIT head update
by Emrah 19 Feb '14

19 Feb '14

Hi all, It looks like my recent FS update to the GIT head broke late negotiation of codecs. With inherit_codec=true, my FS consistently transcodes instead of correlating codecs to find the best suitable one. Is anybody experiencing the same issue? This seems to occur only on a device to device call with late negotiation and inherit codec enabled. I can still manipulate my codecs with absolute_codec if I wan to enforce things. Thanks for your help, Best, Emrah

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JIRA Issues
by Brian West 19 Feb '14

19 Feb '14

FreeSWITCHers, Over the next few weeks I’m going thru the back log of JIRA issues filed. If you’ve filed something and its incomplete or needs additional information and hasn’t been responded to in the past 180 days it will be closed as such, feel free to re-open if you can provide more info or comment that the issue has been resolved. I’ve noticed a large number of issues being filed with incomplete data, terse descriptions. Do not reference the words ‘odd’, ‘weird’ or ‘strange’ in any bug report. If the issue wasn’t odd, strange or weird you probably wouldn’t be opening a ticket in the first place. Please keep descriptions clear and attach complete logs from start to finish of the issue happening along with relevant revision information, config settings such as dial plan or other changes to the config files that may contribute to your issue you’re filing. If the issue is something that you can’t quite replicate or aren’t sure don’t post to the list yet, file a JIRA with the relevant information and then reference that JIRA when you post to the list, You’ll find this will probably work out faster for you and makes sure we don’t lose the report in the sea of emails on the list. Do not attach core files, gzip’ed log files or anything that can’t be read properly from a mobile devices web browser. If we check JIRA on our phones (we do this a lot) when we are out and about and its frustrating when you try to read something and you can’t do so with ease. I was serious about dropping support for OpenBSD, NetBSD and FreeBSD (its on its last get almost), The response to maintainers has underwhelming and not many people are interested or state they do not have the skill set required to accomplish the task. If you need help I’m pretty sure we can get Richard Neese to assist you as he used to maintain those ports in the FreeBSD tree and they’ve expired and were removed because they would no longer build. Remember if you don’t even try you can’t fail… so what do you have to lose? ;) Remember we had to start somewhere to learn this auto goop you use for cross platform build systems. I would love to see complete support for OpenBSD/NetBSD and FreeBSD and anything else that you wanna toss in but we need your help to maintain, test and continue supporting those platforms. I’m going to be sorting thru these JIRA reports, data points and all facts required and design a tool set that we can deploy that will help you collect all the information required for a bug report so that we can work the issues faster, its going to take me a few months to wrap my head around an approach we can use in production environments that makes sense. The entire team will be on the Weekly call again tomorrow I would love to discuss this and the ICMP/MTU issue a little tomorrow too. As always feel free to respond and get involved. ATTENTION: If you have patches pending in JIRA please make sure they apply cleanly to MASTER, and update if needed so we can work thru testing them faster. Thanks, -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

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shared line
by Jason Holden 18 Feb '14

18 Feb '14

All, I am experience speratic problems with fs 1.5 and shared line on POlycom phones. The packet capture shows that the appearance information is being sent out by fs but phones are not properly lighting. The appearance index is empty but the status is changing between idle, active etc. Is their a bug in fs 1.5, a kerupt db due to incorrect shutdown? Any advise would be appreciated. Thanks.

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Week in Review(Master Branch) Feb 9-15
by Kathleen King 18 Feb '14

18 Feb '14

Hello, again. This week in the FreeSWITCH master branch we had 55 commits. Most of the commits this week were miscellaneous and several of those were related to promoting the speex codec to the core. We also got some new features, including additional options for voicemail storage organization. All in all, it was a good week and 18 Jira tickets were resolved. The following bugs were squashed: -- fixed bug to allow for comma separation of email addresses in mod_voicemail -- fixed bug with AWS S3 URL validation in mod_http_cache -- fixed bug by removing a hard coded timezone in mod_xml_radius -- work around the Zoiper bug were all subscribes have an expires of 0 New features that were added: -- added a parameter vm-domain-storage-dir to mod_voicemail -- added a parameter storage-dir-shared to mod_voicemail -- added SIP dialing to just a domain Improvements in cross platform build supports: -- fixed bug SRTP audio issue on Windows 7 -- disabled the SRTP warning for Windows -- build fix for hostname lookups in mod_sofia and mod_command's gethost -- updated to OpenSSL 1.0.1f in Windows build -- fixed compiler issue on Windows with mod_managed In terms of stability these were the use cases that were fixed: -- fixed segfault in mod_sofia related to SIP presence when the presence domain could not be found -- fixed segfault in sofia related to inbound SIP registrations Feedback welcome and the referenced commits are in the attached text file with corresponding Jira links. -- Kathleen King Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Cell: (703) 859-3757 kathleen.king(a)quentustech.com

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ICTFax Version 2.2.0 released
by Tahir Almas 18 Feb '14

18 Feb '14

ICTFax Version 2.2.0 released with improved features and bug fix , Please visit following link for more detail http://www.ictinnovations.com/content/new-release-ictfax *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT

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I can't read messages from voicemail
by FERNANDO VILLARROEL 18 Feb '14

18 Feb '14

Dear All. I use a Yealink T20 and i see message in display, but i can't read the messages, always received a message to say "Your session is incorrect" My user config /usr/local/freeswicth/conf/directory/default/56448908901.xml <?xml version="1.0" ?> <include> <user id="56448908901"> <params> <param name="password" value="Fernando2013"/> <param name="vmpassword" value="123"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="calls_max_intl" value="1"/> <variable name="accountcode" value="1"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 56448908901"/> <variable name="effective_caller_id_number" value="56448908901"/> <variable name="outbound_caller_id_name" value="56448908901"/> <variable name="outbound_caller_id_number" value="56448908901"/> <variable name="tarifa" value="1"/> <variable name="callgroup" value="1"/> </variables> </user> </include> I try to read messages from xlite but the registration for voicemail failed too. My dialplan: <extension name="vmain"> <condition field="destination_number" expression="^vmain$|^4000$|^\*98$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default ${domain_name}"/> </condition> </extension> But when i try to use users like 1000.xml the mailbox works fine so i can read messages. Any idea what is my problem or how i can access the mailbox for user 56448908901 My log: 2014-02-13 02:41:00.335758 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [es] 2014-02-13 02:41:00.575759 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [es] 2014-02-13 02:41:00.595771 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-enter_id.wav] (es:es) 2014-02-13 02:41:00.595771 [DEBUG] switch_ivr_play_say.c:1319 Codec Activated L16@8000hz 1 channels 20ms 2014-02-13 02:41:01.735759 [DEBUG] switch_rtp.c:5269 RTP RECV DTMF 5:960 2014-02-13 02:41:01.735759 [DEBUG] switch_channel.c:489 RECV DTMF 5:960 2014-02-13 02:41:01.735759 [DEBUG] switch_ivr_play_say.c:1723 done playing file /usr/local/freeswitch/sounds/es/mx/maria/voicemail/vm-enter_id.wav 2014-02-13 02:41:02.315758 [DEBUG] switch_rtp.c:5269 RTP RECV DTMF 6:800 2014-02-13 02:41:02.315758 [DEBUG] switch_channel.c:489 RECV DTMF 6:800 2014-02-13 02:41:04.115756 [DEBUG] switch_rtp.c:5269 RTP RECV DTMF 4:480 2014-02-13 02:41:04.115756 [DEBUG] switch_channel.c:489 RECV DTMF 4:480 2014-02-13 02:41:04.435757 [DEBUG] switch_rtp.c:5269 RTP RECV DTMF 4:640 2014-02-13 02:41:04.435757 [DEBUG] switch_channel.c:489 RECV DTMF 4:640 2014-02-13 02:41:05.535759 [DEBUG] switch_rtp.c:5269 RTP RECV DTMF 8:480 2014-02-13 02:41:05.535759 [DEBUG] switch_channel.c:489 RECV DTMF 8:480 2014-02-13 02:41:05.915756 [DEBUG] switch_rtp.c:5269 RTP RECV DTMF 9:480 2014-02-13 02:41:05.915756 [DEBUG] switch_channel.c:489 RECV DTMF 9:480 2014-02-13 02:41:06.295757 [DEBUG] switch_rtp.c:5269 RTP RECV DTMF 0:480 2014-02-13 02:41:06.295757 [DEBUG] switch_channel.c:489 RECV DTMF 0:480 2014-02-13 02:41:06.755757 [DEBUG] switch_rtp.c:5269 RTP RECV DTMF 8:480 2014-02-13 02:41:06.755757 [DEBUG] switch_channel.c:489 RECV DTMF 8:480 2014-02-13 02:41:07.175760 [DEBUG] switch_rtp.c:5269 RTP RECV DTMF 9:480 2014-02-13 02:41:07.175760 [DEBUG] switch_channel.c:489 RECV DTMF 9:480 2014-02-13 02:41:07.555757 [DEBUG] switch_rtp.c:5269 RTP RECV DTMF 0:480 2014-02-13 02:41:07.555757 [DEBUG] switch_channel.c:489 RECV DTMF 0:480 2014-02-13 02:41:08.595759 [DEBUG] switch_rtp.c:5269 RTP RECV DTMF 1:640 2014-02-13 02:41:08.595759 [DEBUG] switch_channel.c:489 RECV DTMF 1:640 2014-02-13 02:41:18.615756 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [es] 2014-02-13 02:41:18.635764 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-enter_pass.wav] (es:es) 2014-02-13 02:41:18.635764 [DEBUG] switch_ivr_play_say.c:1319 Codec Activated L16@8000hz 1 channels 20ms 2014-02-13 02:41:22.115757 [DEBUG] switch_ivr_play_say.c:1723 done playing file /usr/local/freeswitch/sounds/es/mx/maria/voicemail/vm-enter_pass.wav 2014-02-13 02:41:22.235757 [DEBUG] switch_ivr_play_say.c:251 Handle say:[#] (es:es) 2014-02-13 02:41:22.235757 [DEBUG] switch_ivr_play_say.c:1319 Codec Activated L16@8000hz 1 channels 20ms 2014-02-13 02:41:23.655758 [DEBUG] switch_ivr_play_say.c:1723 done playing file /usr/local/freeswitch/sounds/es/mx/maria/ascii/35.wav 2014-02-13 02:41:24.195758 [DEBUG] switch_rtp.c:5269 RTP RECV DTMF 1:1120 2014-02-13 02:41:24.195758 [DEBUG] switch_channel.c:489 RECV DTMF 1:1120 2014-02-13 02:41:24.695757 [DEBUG] switch_rtp.c:5269 RTP RECV DTMF 2:640 2014-02-13 02:41:24.695757 [DEBUG] switch_channel.c:489 RECV DTMF 2:640 2014-02-13 02:41:25.215759 [DEBUG] switch_rtp.c:5269 RTP RECV DTMF 3:640 2014-02-13 02:41:25.215759 [DEBUG] switch_channel.c:489 RECV DTMF 3:640 2014-02-13 02:41:35.235758 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [es] 2014-02-13 02:41:35.255765 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-fail_auth.wav] (es:es) 2014-02-13 02:41:35.255765 [DEBUG] switch_ivr_play_say.c:1319 Codec Activated L16@8000hz 1 channels 20ms 2014-02-13 02:41:37.375756 [DEBUG] switch_ivr_play_say.c:1723 done playing file /usr/local/freeswitch/sounds/es/mx/maria/voicemail/vm-fail_auth.wav 2014-02-13 02:41:37.575758 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [es] 2014-02-13 02:41:37.595765 [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-enter_id.wav] (es:es) 2014-02-13 02:41:37.595765 [DEBUG] switch_ivr_play_say.c:1319 Codec Activated L16@8000hz 1 channels 20ms Regards

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