- FreeSWITCH-users - FreeSWITCH Mailing Lists

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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What am I missing for IVR Display Name?
by mehmet ozisik 18 Feb '14

18 Feb '14

Hello, For days I am trying to change IVR's display name when user called IVR on freeswitch on 200 OK response. I am always seeing "Outbound Call" because it is setted as default. I have tried many combiantions , searched the wiki , but nothing was effective. I would like to try to write down all combinations which I have tried, could you please look at and tell me what I am missing? PS : I have activated ignore_display_update variable not to get sip UPDATE. I have edited dialpan for IVR <extension name="IVR"> <condition field="destination_number" expression="^IVR$"> <action application="socket" data="192.168.170.102:8084 async full"/> <action application="export" data="effective_caller_id_name=Mehmet1"/> <action application="export" data="origination_callee_id_name=Mehmet2"/> <action application="export" data="origination_callee_id_number=Mehmet3"/> <action application="export" data="caller_id_name=Mehmet4"/> <action application="export" data="caller_id_number=Mehmet5"/> <action application="export" data="effective_caller_id_number=Mehmet6"/> <action application="export" data="effective_callee_id_name=Mehmet7"/> </condition> </extension> <extension name="IVR"> <condition field="destination_number" expression="^IVR$"> <action application="socket" data="192.168.170.102:8084 async full"/> <action application="bridge" data="effective_caller_id_name=Mehmet1"/> <action application="bridge" data="origination_callee_id_name=Mehmet2"/> <action application="bridge" data="origination_callee_id_number=Mehmet3"/> <action application="bridge" data="caller_id_name=Mehmet4"/> <action application="bridge" data="caller_id_number=Mehmet5"/> <action application="bridge" data="effective_caller_id_number=Mehmet6"/> <action application="bridge" data="effective_callee_id_name=Mehmet7"/> </condition> </extension> <extension name="IVR"> <condition field="destination_number" expression="^IVR$"> <action application="socket" data="192.168.170.102:8084 async full"/> <action application="set" data="effective_caller_id_name=Mehmet1"/> <action application="set" data="origination_callee_id_name=Mehmet2"/> <action application="set" data="origination_callee_id_number=Mehmet3"/> <action application="set" data="caller_id_name=Mehmet4"/> <action application="set" data="caller_id_number=Mehmet5"/> <action application="set" data="effective_caller_id_number=Mehmet6"/> <action application="set" data="effective_callee_id_name=Mehmet7"/> </condition> </extension> <extension name="IVR"> <condition field="destination_number" expression="^IVR$"> <action application="socket" data="192.168.170.102:8084 async full"/> <action application="bridge" data="{effective_caller_id_name='Mehmet1'}sofia/<mydomain>/ sip:IVR@192.168.170.102:15060"/> <action application="bridge" data="{origination_callee_id_name='Mehmet2'}sofia/<mydomain>/ sip:IVR@192.168.170.102:15060"/> <action application="bridge" data="{origination_callee_id_number='Mehmet3'}sofia/<mydomain>/ sip:IVR@192.168.170.102:15060"/> <action application="bridge" data="{caller_id_name='Mehmet4'}sofia/<mydomain>/ sip:IVR@192.168.170.102:15060"/> <action application="bridge" data="{caller_id_number='Mehmet5}sofia/<mydomain>/ sip:IVR@192.168.170.102:15060"/> <action application="bridge" data="{effective_caller_id_number='Mehmet6'}sofia/<mydomain>/ sip:IVR@192.168.170.102:15060"/> <action application="bridge" data="{effective_callee_id_name='Mehmet7'}sofia/<mydomain>/ sip:IVR@192.168.170.102:15060"/> </condition> </extension> <extension name="IVR"> <condition field="destination_number" expression="^IVR$"> <action application="socket" data="192.168.170.102:8084 async full"/> <action application="export" data="{effective_caller_id_name='Mehmet1'}sofia/<mydomain>/ sip:IVR@192.168.170.102:15060"/> <action application="export" data="{origination_callee_id_name='Mehmet2'}sofia/<mydomain>/ sip:IVR@192.168.170.102:15060"/> <action application="export" data="{origination_callee_id_number='Mehmet3'}sofia/<mydomain>/ sip:IVR@192.168.170.102:15060"/> <action application="export" data="{caller_id_name='Mehmet4'}sofia/<mydomain>/ sip:IVR@192.168.170.102:15060"/> <action application="export" data="{caller_id_number='Mehmet5}sofia/<mydomain>/ sip:IVR@192.168.170.102:15060"/> <action application="export" data="{effective_caller_id_number='Mehmet6'}sofia/<mydomain>/ sip:IVR@192.168.170.102:15060"/> <action application="export" data="{effective_callee_id_name='Mehmet7'}sofia/<mydomain>/ sip:IVR@192.168.170.102:15060"/> </condition> </extension> <extension name="IVR"> <condition field="destination_number" expression="^IVR$"> <action application="socket" data="192.168.170.102:8084 async full"/> <action application="set" data="{effective_caller_id_name='Mehmet1'}sofia/<mydomain>/ sip:IVR@192.168.170.102:15060"/> <action application="set" data="{origination_callee_id_name='Mehmet2'}sofia/<mydomain>/ sip:IVR@192.168.170.102:15060"/> <action application="set" data="{origination_callee_id_number='Mehmet3'}sofia/<mydomain>/ sip:IVR@192.168.170.102:15060"/> <action application="set" data="{caller_id_name='Mehmet4'}sofia/<mydomain>/ sip:IVR@192.168.170.102:15060"/> <action application="set" data="{caller_id_number='Mehmet5}sofia/<mydomain>/ sip:IVR@192.168.170.102:15060"/> <action application="set" data="{effective_caller_id_number='Mehmet6'}sofia/<mydomain>/ sip:IVR@192.168.170.102:15060"/> <action application="set" data="{effective_callee_id_name='Mehmet7'}sofia/<mydomain>/ sip:IVR@192.168.170.102:15060"/> </condition> </extension> regards Mehmet

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Connect three parties together using bridge?
by Ali Pey 17 Feb '14

17 Feb '14

Hello, Is it possible to connect three parties together using bridge or uuid_bridge? Thanks, Ali Pey

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Integrating FreeSWITCH with Open IMS Core
by Jaytirth S Khairatkar 15 Feb '14

15 Feb '14

Hello, I wish to integrate Open IMS Core and FreeSWITCH such that an IMS user can call a FreeSWITCH user and vice versa. I have succeed in establishing a call from an IMS user to a FreeSWITCH user. IMS provides a feature of forwarding a SIP INVITE request based on a filter. But I am struggling to establish a call from a FreeSWITCH user towards IMS. Does FreeSWITCH have a feature where it can forward an INVITE request based on a filter? I will greatly appreciate any help that can lead me closer to the solution. thanks and regards Jaytirth *Disclaimer* ------------------------------------------------------------ Notice: This e-mail and / or attachments (communication) may contain information that is confidential and proprietary to The Tata Power Company Limited, Strategic Engineering Division (Tata Power SED). If you are not the intended recipient of this communication, please do not use or disseminate the information contained therein, notify the sender by e-mail or telephone and delete the communication (permanently) from your system. Any information in this communication that does not relate to the official business of Tata Power SED shall be understood as neither given nor endorsed by it. Please note that e-mails are susceptible to change and Tata Power SED shall not be liable for the improper or incomplete transmission of the information contained in this communication nor for any delay in its receipt. This communication is not guaranteed to be free from computer viruses and it is recommended that you check for all viruses before download. Print this communication only if absolutely essential. Thank you for your co-operation. ------------------------------------------------------------

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not taking var in dialplan
by Jason Holden 14 Feb '14

14 Feb '14

Hi, Anyone provide any advise. The following dial plan chunk is not taking the EXT number and placing it in my bridge statement. <extension name="Voicemail_Get_Admin"> <condition field="destination_number" expression="^\*86(7\d{3})$"/> <condition field="${phone_type}" expression="1"> <action application="set" data="dialed_number=$1"/> <action application="bridge" data="{origination_callee_id_name='Voicemail'}sofia/gateway/vm/vm86-${phone_ type}-$1-${site}-$${lang}"/> </condition> </extension>

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timezones.conf.xml
by Brian West 14 Feb '14

14 Feb '14

I’ve checked in timezone-gen.pl into scripts/perl (Thanks to Nathan Neulinger) Run it on your systems, spot check it, Please give us feedback, we’ll be adding this to a periodic run to update going forward in a few weeks if nobody reports bugs with it. -- 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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SIP ALG paramenter
by John Schaeffer 14 Feb '14

14 Feb '14

Why does FREESWITCH fail with our router setting of SIP-ALG turned on? This seems to be opposite of Asterisk. John Schaeffer Telecommunications Engineer 503-727-9903(w)971-212-9431(c) [EB logo_no tagline_med]

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