February 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Long call duration - Hangup issue
by Balamurugan Mahendran 24 Feb '12

24 Feb '12

HI all, While dialing-out (outbound call) if the call is terminated by the calling party(other side), then the call will never hang-up that reflects long call duration in CDR. Please let me know if Iam missing anything. Thanks, Bala

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can i get gtalk account online status on mod_dingaling?
by 熊子奇 24 Feb '12

24 Feb '12

HI All, im using mod_dingaling to connect to google talk, and the call from freeswitch to remote gtalk account has been done successfully. now here raise a question, can i get the remote account online/offline status before i call it?thanks for your attention. if any reply, im really appreciate . regards.--xiong

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Re: [Freeswitch-users] Overhead paging
by George Sapak 23 Feb '12

23 Feb '12

I have taken a look at that, the client is a non profit, we would like to do it on the cheap, we do have some ht286, just though that could be hacked instead of spending almost $400 CAN Thank You, George. ----- Original Message ----- From: "Tim Nelson" <tnelson(a)rockbochs.com> To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Sent: Thursday, February 23, 2012 9:44:55 AM Subject: Re: [Freeswitch-users] Overhead paging Yes, the product referenced would allow your VoIP implementation (Freeswitch) to work with your existing paging system. Did you read the info on the link I sent? --Tim ----- Original Message ----- > there is already a paging system in place. > > Thanks, George. > > > ----- Original Message ----- > From: "Tim Nelson" <tnelson(a)rockbochs.com> > To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> > Sent: Thursday, February 23, 2012 8:39:14 AM > Subject: Re: [Freeswitch-users] Overhead paging > > ----- Original Message ----- > > what is the easiest way to set this up, just need a way to pickup a > > set and have an amp output going from the switch going to the paging > > system, I have read somewhere that you could use HT503 for this, > > sounds confusing. > > > > You could use a VoIP to analog paging gateway, such as those made by > Cyberdata. See here: > > http://www.cyberdata.net/products/voip/legacyanalog/index.html > > I've used their products quite a bit recently, and have had fantastic > results. In the rare event I required support, the experience was top > notch.*** > > --Tim > > ***Please keep in mind I'm not affiliated with them in any way, but I > find the need to 'toot my horn' about a product/company that exceeds > my expectations as it is becoming a rarer thing these days. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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need more variables which xml_curl module sends
by Yuriy Nasida 23 Feb '12

23 Feb '12

Hello FS users, I search some way for recieving more variables which my php script recieves in array - $_REQUEST from xml_curl module (it is intralanman's scripts) in case of recieving sip message - REGISTER from some sip device. I see list of variables:http://wiki.freeswitch.org/wiki/Mod_xml_curl#Authorization But, I need more. For example I need to know source sip port of endpoint. Also I have to know full SIP contact header from REGISTER request. Please advise.Thanks.

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How to control custom header
by 오부경 23 Feb '12

23 Feb '12

Hi, users. Sorry about my poor english. I am a beginner of FS. I have a problem with IP-PBX. My customer wants to call mobile phone using FS connected IP-PBX. IP-PBX must set 100-rel mode. IP-PBX send to FS with custom SIP header(P-Charging-Vector), and I want to add this header to response SIP message. I read FS wiki and try many cases. but, i can't solve my problem. Follow is my call diagram: ======================================================================================================= SIP CLIENT : FreeSwitch : IP-PBX : Telecom(?) ======================================================================================================= INVITE --> <-- 100 TRYING INVITE --> <-- 100 TRYING <-- *183 Session Progress (with P-Charging-Vector Header)* *PRACK --> (without P-Charging-Vector Header) * <-- 183 (with P-Charging-Vector Header) *<-- 481 Call Leg/Transaction Does Not Exist * CANCEL --> <-- 200 OK <-- 487 Request Terminated ACK --> <-- 480 Temporarily Unavailable ACK --> ======================================================================================================= SIP Client is sofia based windows client. Please help me. Thank you. Regards, -- BooKyung Oh

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FreeSWITCH version: 1.0.head, sofia.c:6033 Reinvite Codec Error!
by Eric Masson 23 Feb '12

23 Feb '12

Hello, On a fresh 1.0.head (FreeSWITCH version: 1.0.head (git-303de52 2012-02-14 18-59-37 -0600), outgoing calls never succeed and the following error is logged in loop : > 2012-02-15 11:34:52.894480 [DEBUG] sofia.c:5525 Channel sofia/external/0251685292 entering state [received][100] > 2012-02-15 11:34:52.894480 [DEBUG] sofia.c:5536 Remote SDP: > v=0 > o=root 14865 14867 IN IP4 192.168.22.13 > s=session > c=IN IP4 83.136.163.36 > t=0 0 > m=audio 59418 RTP/AVP 8 0 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=nortpproxy:yes > > 2012-02-15 11:34:52.894480 [DEBUG] sofia_glue.c:4801 Audio Codec Compare [PCMA:8:8000:20:64000]/[GSM:3:8000:20:13200] > 2012-02-15 11:34:52.894480 [DEBUG] sofia_glue.c:4801 Audio Codec Compare [PCMU:0:8000:20:64000]/[GSM:3:8000:20:13200] > 2012-02-15 11:34:52.894480 [DEBUG] sofia_glue.c:4801 Audio Codec Compare [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200] > 2012-02-15 11:34:52.894480 [DEBUG] sofia_glue.c:4915 Set 2833 dtmf send payload to 101 > 2012-02-15 11:34:52.894480 [ERR] sofia.c:6033 Reinvite Codec Error! > 2012-02-15 11:34:52.894480 [DEBUG] switch_core_session.c:875 Send signal sofia/external/0251685292 [BREAK] > 2012-02-15 11:34:52.894480 [DEBUG] switch_core_session.c:875 Send signal sofia/external/0251685292 [BREAK] > 2012-02-15 11:34:52.894480 [DEBUG] sofia.c:5525 Channel sofia/external/0251685292 entering state [ready][488] Codecs prefs are the following : <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM,H263"/> I don't really understand the codec negociation issue, any idea, anyone ? Kind Regards Éric Masson

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mod_xmlr_cdr, resend failed CDR's.
by Henrik Aagaard Sørensen 23 Feb '12

23 Feb '12

I'm using the mod_xml_cdr module which works like a charm. It send the data to a web server and if it fails to do so, it saves the data as a .xml-file. My question is know, is it somehow possible to have FS check for failed CDR's and resend them once every hour or so? Or is this a job for a homemade cronjob? Could be nice if FS automatically had this feature as it already is able to push data to a web server.

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How to disable two 'm=' lines
by Mihail 23 Feb '12

23 Feb '12

Hello. I try to connect Asterisk as a client to Freeswitch using TLS and SRTP. Outbound direction (from Asterisk) - TLS/SRTP works fine. There is an SRTP issue with inbound calls. FS in SDP offer sends two 'm=' lines: m=audio 23036 RTP/SAVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:bdDpQbqc5iaGvVhCilOd5nXOKHLfGGm8J3mjsLIa a=ptime:20 m=audio 23036 RTP/AVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Receiving that Asterisk reply: WARNING[9480]: chan_sip.c:8847 process_sdp: Multiple audio streams are not supported How to solve it from Freeswitch side? I need to offer only RTP/SAVP. Also, looking ahead, is it possible to add and modify a=crypto: line, I need to send SHA1_80 in offer or both? Asterisk 1.8.9.2 FreeSWITCH Version 1.0.head (git-e6bfa11 2012-02-09 16-47-32 -0600) SRTP enabled in dialplan/public.xml <action application="export" data="nolocal:sip_secure_media=true"/> Please, advise. Thanks!

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G722 16k or 8k?
by Stephen Dame 23 Feb '12

23 Feb '12

Question on G722, trying to see if the quality of basic g.722 is much better than the standard g729 or g711. Looking at wiki docs. "G722 - G722 16kHz using default 20ms ptime. (multiples of 10)" However, when I connect to freeswitch using blink client I see G722/8000? I thought this would be 16000. I know when I connect using speex wb, it shows 16000. 2012-02-23 10:20:07.116606 [DEBUG] mod_sofia.c:750 Local SDP sofia/internal/1004(a)107.XX.XX.XX: v=0 o=FreeSWITCH 1329993171 1329993172 IN IP4 107.XX.XXX.XXX s=FreeSWITCH c=IN IP4 107.XX.XXX.XXX t=0 0 m=audio 17236 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv Shouldn't this be 16000? Thanks Stephen

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Voicemail - User Recorded greetings Directory
by Joe Flemmings 23 Feb '12

23 Feb '12

By default, voicemail user recordings are stored in $${base_dir}/storage/voicemail/default/$${domain}/<user>/ How can u change this directory location?

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