June 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Nibblebill heartbeat on B-leg
by MaartenDM 22 Jun '09

22 Jun '09

Hello, I am using Nibblebill to bill bridged calls initiated via API. The problem is that the billing on the B-leg is only done when the call is terminated and not at heartbeat. For the a-leg it is working. I added the global_heartbeat variable at the b-leg but without success. I use now: originate {ignore_early_media=true,nibble_account=100,nibble_rate=0.02}sofia/external/12345@serverip &bridge({global_heartbeat=50,nibble_account=100,nibble_rate=0.02}sofia/external/45678@serverip) thx, MdM -- View this message in context: http://www.nabble.com/Nibblebill-heartbeat-on-B-leg-tp24144481p24144481.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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event_add_head warning message on cosole
by Lars Zeb 22 Jun '09

22 Jun '09

I just upgraded to version 13886. On the console the following messages appear every few minutes. I've looked at the code but it's way over my head. Why is it displaying? How can I turn it off? 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header -> 'toll_allow' = 'domestic,international,local' 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header -> 'accountcode' = '1000' 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header -> 'user_context' = 'default' 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header -> 'effective_caller_id_name' = 'Extension 1000' 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header -> 'effective_caller_id_number' = '1000' 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header -> 'outbound_caller_id_name' = 'FreeSWITCH' 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header -> 'outbound_caller_id_number' = '3235551212' 2009-06-21 17:39:26.881193 [WARNING] sofia_reg.c:1988 event_add_header -> 'callgroup' = 'techsupport' 2009-06-21 17:39:26.882195 [WARNING] sofia_reg.c:1988 event_add_header -> 'record_stereo' = 'true' 2009-06-21 17:39:26.882195 [WARNING] sofia_reg.c:1988 event_add_header -> 'default_gateway' = 'example.com' 2009-06-21 17:39:26.882195 [WARNING] sofia_reg.c:1988 event_add_header -> 'default_areacode' = '323' 2009-06-21 17:39:26.882195 [WARNING] sofia_reg.c:1988 event_add_header -> 'transfer_fallback_extension' = 'operator' Thanks, Lars

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svn update error
by Lars Zeb 21 Jun '09

21 Jun '09

I am currently running 13723 and want to get current. When I issue the "svn update" command, the following error appears: A src/mod/asr_tts/mod_unimrcp A src/mod/asr_tts/mod_unimrcp/mod_unimrcp.2008.vcproj A src/mod/asr_tts/mod_unimrcp/unimrcp.vsprops A src/mod/asr_tts/mod_unimrcp/Makefile.am A src/mod/asr_tts/mod_unimrcp/mod_unimrcp.c U src/mod/asr_tts/mod_pocketsphinx/mod_pocketsphinx.c U src/mod/event_handlers/mod_event_multicast/mod_event_multicast.c svn: Failed to add file 'src/mod/event_handlers/mod_event_multicast/Makefile': an unversioned file of the same name already exists What do I need to do? Should I delete the Makefile file? Thanks, Lars

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SIP gateway behind NAT
by Jan Kubr 21 Jun '09

21 Jun '09

We have a SIP gateway behind NAT which I haven't been able to set up to work with Freeswitch. The configuration I thought would work is: <include> <gateway name="nat"> <param name="username" value="user"/> <param name="password" value="pass"/> <param name="realm" value="11.12.13.10"/> <param name="proxy" value="78.24.13.197"/> </gateway> </include> This, however, sends REGISTER sip:78.24.13.197;transport=udp SIP/2.0 whereas I need REGISTER sip:11.12.13.10;transport=udp SIP/2.0 When I set "register-proxy" to the public address (78.24.13.197) and comment the "proxy" param out, it successfully registers. Though any other SIP messages are sent to 11.12.13.10 which obviously fails. Basically what I need Freeswitch to do is to send everything to the public address, but put the private one everywhere in the SIP messages. How do I do this? Thanks, Jan

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email core dump
by Mark Campbell-Smith 21 Jun '09

21 Jun '09

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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sofia external profile: external IP problem
by Nandy Dagondon 21 Jun '09

21 Jun '09

hi, i tested the latest SVN build (13884) using the sample configuration files ... no modifications whatsoever. but in sofia external profile, the IP address is my internal address instead of my external IP address. did i miss something here? tks. -nandy

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Open source Java based inbound event socket library available
by paul.degt＠gmail.com 21 Jun '09

21 Jun '09

http://versafon.com/versafonweb/Software.jsp Essentially it's a wrapper around inbound socket interface, not all events supported yet, and not all event parameters/variables. It's multi threaded and scaled well in testing. We offer commercial support and development for FreeSwitch as well.

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FS as a Class 5 switch
by Nandy Dagondon 21 Jun '09

21 Jun '09

hi everybody, i'm interested to know if anyone employed FS as a local exchange switch. i'm confident FS can handle several calls using RTP by-pass mode. however, i'm more concerned on handling the large dialplan with hundreds (or even a few thousand) exchange prefixes nationwide during call setup. i'd be glad to hear experiences and suggestions esp on the hardware dimensioning. we're talking a small exchange up to about 1,100 lines only, mostly linked to the main exchange via MFC-R2. tks, nandy

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Can it do it?
by JuanMa 21 Jun '09

21 Jun '09

Hi, I need to have the hability to negotiate the codec in a session (using proxy media or bypass media), unfortunally I've been unable to achive this due the documentation that I've found about it's vague. I've already tried using "absolute_codec_string" and everything that says in http://wiki.freeswitch.org/wiki/Codec_negotiation but it seams to ignore it when the media is in "bypass media" or "proxy media". I need to configure the FS as a SBC or as a pseudo proxy (I already know that FS is not intend to do it, but in the documentation says that it can). I've also tried to manually modify the SDP using: <param name="inbound-late-negotiation" value="true"/> And updating variables "switch_r_sdp" and "switch_l_dsp" but it also seams to ignore it. Here is the config: Endpoint1-->FS-->SWITCH-->FS-->Endpoint2 What I need, is to offer to the SWITCH only the codecs defined for Enpoint1 (ie. G729) and when the SWITCH send the INVITE to FS again, only offer the codecs available for Endpoint2. Eventually the SWITCH will do the transcoding. So, here is my question, is there any way to achive this? (Handle the invite codecs in bypass or proxy media), if so, is there any example to follow? o can you give a tip? Thanks in advance, Regards

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stucked with mod_fsv and mod_h26x
by Ing. Edwin Villarreal 20 Jun '09

20 Jun '09

Hello friends. i'm kind of stucked with mod_fsv and mod_h26x... there is no info on wiki yet. have u used them? Can u send examples of how-to? I need to join 2 endpoints with full HD video conference using PZT HD VideoCams to PC and SIP-UA. Thank you! EDWIN ChiPoLy

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