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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

2 1

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.

2 3

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!

2 4

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

4 5

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?

3 3

mod_curl data answer
by freeswitchtest 23 Feb '12

23 Feb '12

Hi, I am new in freeswitch but have used asterisk for more than 4 years. So I have scripts in php which connect to our internal mysql database an for example show who is calling. I trie in freeswitch with curl: <action application="curl" data="http://www.mywebseite.de/freeswitch/call_in.php/?id=1234356788 json"/> If I make it at the CLI I can see the correcct answer - the number of our customer But how can I read the correct answer in the freeswitch dialplan. I thought of something like: <action application="set" data="hurra=${curl_response_data}"/> I tried in different ways but never got the number. Where I am wrong ? Thanks for your help

2 2

ODBC ussing Dbh suddenly gives Invalid cursor state error after executing querys
by Michael Lutz 23 Feb '12

23 Feb '12

Hi, I just compiled the latest version on x64 and running it. Now suddenly al my queries log an error, even though they are executed correctly. Has something changed? I use the Dbh to connect via my system dsn to MSSql server. The exect error is: [STATE: 24000 CODE 0 ERROR: [Microsoft][ODBC SQL Server Driver]Invalid cursor state ] The log doesn't show anything else. Thanks, Mike.

4 6

Overhead paging
by George Sapak 23 Feb '12

23 Feb '12

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. Thank You, George.

3 4