October 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Automate wancfg
by DN Singh 11 Oct '13

11 Oct '13

Hello list, Has anyone tried to automate the wancfg configuration with freeswitch? I have tried, but gets automatically configured with T1, even if I give it parameters for E1. I have done this by using wancfg_zaptel with some arguments. Please let me know if more details would be required. Cheers!! DN

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Integrate openfire to freeswitch
by Groupe SOGO 11 Oct '13

11 Oct '13

Dear Friends, I installed freeswitch latest version on CentOS 6.4 and everything is working well. The default sip internal accounts can make and receive calls. I also installed openfire 3.8.1 on the same machine and it is working well. I can create users with spark client 2.6.3 and they can chat and create conference. So everything works very well. Now I want to integrate openfire to freeswitch. Has anyone tried it? What did he do to achieve that? What can i do to achieve the integration? thanks in advance for your help. Labolinux K. Amouzou

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ANI missing in cli originate
by François 11 Oct '13

11 Oct '13

Hello, I've seen some FS users using ANI for billing purposes, and I'm not sure if it is normal that the ANI variable is missing on calls originated from CLI (or other originate with no session associated). Would a small patch copying the specified Caller ID Number to ANI in those cases be an acceptable solution? Thanks, François.

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originate not creating xml_cdr record
by Matt Broad 11 Oct '13

11 Oct '13

Hi, I have setup xml_cdr and it is working great, both aleg and bleg calls are logged. I am using the api originate method, but this does not create an xml_cdr record (though a record is logged in /log/cdr-csv/master/csv). Is this correct? thanks Matt

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mod_snmp and freeswitch MIB
by Vik Killa 11 Oct '13

11 Oct '13

FreeSWITCH MIB is now included in Solarwind's Orion MIB I only had to submit the request to mibs(a)solarwinds.com :-)

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Re: [Freeswitch-users] call park blf
by fs 11 Oct '13

11 Oct '13

sorry, I forgot to paste in the settings in the dial plan <extension name="valet_park"> <condition field="destination_number" expression="^(park\+\d)$"> <action application="answer"/> <action application="valet_park" data="my_lot(a)myhome.pbx2net.ca $1"/> </condition> </extension> I use park+{slot} to park calls using keys on snom and grandstream phones. The busy lamp field lights but goes out shortly after, about 5 seconds. However, the call is still on hold and can be picked up by pressing the illuminated light. When I do a sip trace, there is a notify message saying the park+{slot} has been terminated. Any ideals. Incidentally, all other BLF's work prefectly. Darcy Primrose

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FreeTDM - Using ftdm_channel_wait with FTDM_EVENTS
by Andrea Coslovich 11 Oct '13

11 Oct '13

Dear guys, I'm new to the list so first of all let me thank the FreeSWITCH developers for their great work. But now I must come to the point. I have an A104 Sangoma board and I'm developing an application based on FreeTDM library. I've started by using and modifying the examples "freeswitch/libs/freetdm/src/test*.c" and now I'm able to retreive and send the audio data from/to the board. I'm using the "ftdm_channel_wait" to know when I can read/write from the freetdm's queues and all seems to work well. But I have a big issue when I try to use "ftdm_channel_wait" with the flag "FTDM_EVENTS" (flags=(ftdm_wait_flags_t)FTDM_EVENTS). Basically, what happen is that the "ftdm_channel_wait(ftdmchan,&flags,-1)" never exits even if the channel receives DTMF or is been hung-up. I've also tried to use "ftdm_channel_command(ftdmchan,FTDM_COMMAND_ENABLE_DTMF_DETECT,NULL)" before the "ftdm_channel_wait" but nothing changed. Now, my questions are the following: 1. What kind of events can cause the exit of "ftdm_channel_wait" function? I've seen the "ftdm_event_type_t" that defines "FTDM_EVENT_DTMF" and "FTDM_EVENT_OOB". Is it the answer to my question? 2. Is there something that I need to do before call "ftdm_channel_wait" to enable the "events generation"? I hope to be in the right place for these questions. Otherwise, please let me know where I can find someone that can help me. I haven't put a lot in this mail because I wasn't really sure if this is the right mail-list but, of course, if you need more information and details to help me don't esitate to ask. Excuse me for my bad english. Have a nice day! Andrea

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Possible Polycom bridging issues in all recent FreeSWITCH versions
by Emrah 11 Oct '13

11 Oct '13

Hi, I have been experiencing the following issue since I upgraded my version of FS. In a scenario where a called is made from one Polycom VVX1500 to another, the video works both ways with only one way audio. A hold / unhold results in 2 way audio with 1 way video. I obviously suspected my configs, especially after subsequently trying the latest git, the 1.4 beta and the stable 1.2.10. So far, the only way I found to solve this issue is by using an old copy of FreeSWITCH with my current configs. FreeSWITCH Version 1.3.13b+git~20130119T155523Z~6cb3be7d8b (git 6cb3be7 2013-01-19 15:55:23Z) To clarify, the same config with any of the latest versions of FS does not work. Questions: How can I troubleshoot this further and help resolve this issue? Any one has an idea of what has happened? Thanks!

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Fax calls drop in FreeSWITCH 1.2.12
by John Platts 10 Oct '13

10 Oct '13

We have had fax calls drop in FreeSWITCH 1.2.12, but we have been able to get the problem solved by enabling proxy media for fax calls. The problem occurs when neither proxy media nor bypass media are enabled. We need to get fax to email and T.38 support working. We need to get these features working correctly. We also need to have faxes working consistently. We need a solution that will allow faxes to work consistently when neither proxy media nor bypass media are enabled.

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troubleshooting packet loss
by Peter Hartmann 10 Oct '13

10 Oct '13

Hi, I just want to make sure I'm understanding something about troubleshooting RTP packet loss. I've done a capture between a handset and the Freeswitch server on a local network and during the call there were obvious audio drop outs from the incoming side. The analysis in Wireshark showed there were zero packets lost in either direction. My question: is this what 'outside' problems look like? Thanks, Peter

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