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Connect external phone
by Sean Holt 02 May '10

02 May '10

Hello list, I keep getting this authentication error when I attempt to connect a device from the outside. I followed the wiki that suggest creating a separate profile to handle external device connections. Here¹s my external xml file <profile name="gexternal5090">  <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <aliases> <alias name="outbound"/> </aliases> <domains> <domain name="$${domain}" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5090"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${moh_uri}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="71.133.39.219"/> <param name="ext-sip-ip" value="71.133.39.219"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile> I did a siptrace on the incoming connection attempt ------------------------------------------------------------------------ REGISTER sip:71.133.39.219:5090 SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5090;branch=z9hG4bKvvaun9r6.1u8ej Max-Forwards: 70 From: <sip:1001@71.133.39.219>;tag=i9w3pt4bm4tpsl To: <sip:1001@71.133.39.219> Call-ID: fo33ia7sk6qy9vnk6h CSeq: 75203 REGISTER Contact: <sip:1001@x.x.x.x:5090;line=20880> Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK Expires: 600 User-Agent: snom-m3-SIP/02.02 (MAC=0004132AD0BE; HW=255) Content-Length: 0 ------------------------------------------------------------------------ send 591 bytes to udp/[x.x.x.x]:5090 at 01:26:47.109165: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.10.231:5090;branch=z9hG4bKvvaun9r6.1u8ej;received=207.47.31.108 From: <sip:1001@71.133.39.219>;tag=i9w3pt4bm4tpsl To: <sip:1001@71.133.39.219>;tag=Zc7mBaBN723Bp Call-ID: fo33ia7sk6qy9vnk6h CSeq: 75203 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="71.133.39.219", nonce="bd1ecf94-5589-11df-b3f4-c1a375adf51a", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ Thoughts? Thanks Sean

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Choppy voive in conference from fritzbox to freeswitch (ptime:30 msecs)
by Nicola von Thadden 01 May '10

01 May '10

Hi, is there anything new since august? I have the same problem with latest FritzBox firmenware and latest FreeSwitch installed (via Quick & Dirty install). FritzBox always sends 30ms in one paket, FS starts with 20 and then switches to 30 (maybe thats the problem?) Calling the Demo IVR at 5000 is (with default options and latest FS) no problem, both uses 30ms ptime, have no idea why. Calling the tetris-demo at 9998 is chopped up, FS using 20ms ptime, FBox 30. Conferences are also choppy. Adding <param name="rtp-autofix-timing" value="false" /> to 1000.xml (account used at the FBox) does not help. Anyone got an idea? Nico

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get a special var from accepted call
by Madovsky 01 May '10

01 May '10

Hi, I'd like to set a var (trunk ID) in the bridge that I can get from my SIP phone when the call is accepted. channel variables in set / export ? Thanks Franck

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fs_cli
by Madovsky 01 May '10

01 May '10

Hi all, when I run from bash as "root" I get this error: /usr/local/freeswitch/bin/fs_cli -x "any_command" [ERROR] libs/esl/fs_cli.c:1181 main() Error Connecting [Socket Connection Error] Freeswitch is running without problems. I think I forgot to something.... svn17134M Thanks Franck

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How to get GMT date/time in channel variable "start_stamp", instead of local
by Vitalii Colosov 01 May '10

01 May '10

Hi, Right now channel variable "start_stamp" represents *local *date/time (to remind, it contains value in following format: 2010-05-01 16:31:58) Is there a similar variable (like start_stamp_gmt?) that will provide *GMT*formatted date/time? Or maybe there are some settings that will make FreeSWITCH provide all date / time formatted variables in GMT instead of local. Thank you, Vitalie

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event after hangup (Javascript)
by Alberto Escudero 01 May '10

01 May '10

Hi, Is there a way to trigger a customized event after the user hangup the call? Although i define the function on_hangup(hup_session, how) it seems i can not trigger and event inside of it although i can run logging. Hints are welcome. -aep -- Stopping junk mailers is good for the environment

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 46, Issue 347
by guru singh 01 May '10

01 May '10

> > From: "Dave Stevenson" <stevendt(a)primrosebank.net> > To: <freeswitch-users(a)lists.freeswitch.org> > Date: Fri, 30 Apr 2010 15:53:26 +0100 > Subject: Re: [Freeswitch-users] Picking up voicemail > Hi Brian, > > thanks for the pointer. > > The SPA-3012 is pretty strong on configuration options, but the documentation is very light on what they actually mean. Just so that I know, what is AVT actually doing ? > > regards > Dave DTMF Tx Method Select the method to transmit DTMF signals to the far end: InBand, AVT, INFO, Auto, InBand+INFO, or AVT+INFO. InBand sends DTMF using the audio path. AVT sends DTMF as eypents. INFO uses the SIP INFO method. Auto uses InBand or AVT based on the outcome of codec negotiation. The default is Auto. Shamelessly pasted from http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/ata/administration/gu…. I was reading this manual when I saw your post. You should check it out if you haven't already. gs

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Error with media ptime
by leonardo alves 01 May '10

01 May '10

Hello, I am new to freeswitch and I have just installed the last version of freeswitch. I am doing some tests dialing with sip and when the call is answer I play an file. If I do 10 calls, in 5 or 6 of them I get this error in the console and the audio gets cut like with it was a bandwithd problem. 2010-04-27 22:20:56.236782 [WARNING] mod_sofia.c:999 We were told to use ptime 20 but what they meant to say was 30 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen. Does anyone knows if there is a way to fix this issue ? Thanks Leonardo

2 3

libpri not working
by Tom 01 May '10

01 May '10

Voip gods, having problem with libpri on incoming call it gives me this [WARNING] ozmod_libpri.c:760 --Duplicate Ring on channel 1:16 (ignored) then it wait a few seconds then i get a fast busy. If i turn off libpri i get this [CRIT] ozmod_isdn.c:710 Received Release Complete with no matching channel 0 Iam running FreeSWITCH Version 1.0.head (svn-17188M) wanpipe-3.5.10.14 dahdi-2.3.0 Sangoma B601de card Tom Baldwin

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Aastra and SCA
by Mark Maly 01 May '10

01 May '10

Hi, I've tried to patiently figure this out by reading the wiki and this list. Unfortunately, I've been unable to get it right. I have 2 Aastra 6731is and a 51i and trying to get SCA working. I'm experiencing problems similar to the Cisco thread from last month - outgoing calls implement SCA well. Incoming calls ring all lines and appearances work, but when one phone is answered, the line appearances are removed from the remaining phones. I am not attempting to use any DNS. My configuration has all three phones plus FS on a local LAN. Nothing too fancy. Each line is configured for Broadsoft SCA and SCA bridging is enabled globally for the phones. Trying to update/replace an old phone at my church. Any help would be greatly appreciated. Thanks, Mark Mark.maly(a)molcs.org

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