March 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

To Proxy Media or Not
by Max Clark 09 Mar '11

09 Mar '11

Hello, I have a couple of FreeSWITCH boxes functioning as SBCs on our network. We need them to: - Route Calls - Function as B2BUAs for call logging - Hide topology from the endpoints They do not need to: - Intercept/play audio - Manipulate the SDP - Transcode - Process anything other than SIP So these machines have been running with Proxy Media enabled in the dialplan. Is the the proper/recommended way of configuring for our environment/needs? Is the minimal gain of resources by not processing the RTP worth it? Thanks, Max

5 10

Duplicate UUIDs in voicemail_msgs
by Troy Anderson 09 Mar '11

09 Mar '11

Hi All, I am regularly getting multiple copies of voicemails when checking voicemail on one of the extensions of a system, and, trying to diagnose the situation, seeing duplicate UUIDs in voicemail_msgs database. I suspect that's why I get multiple "copies" of the message. I'm trying to figure out what would cause multiple entries in the database for the same message. I checked the version of mod_voicemail.c I'm using on this machine (from Nov 18, 2010) against the latest version and none of the few changes seem remotely related. I would love to update to the latest and see if the issue is resolved, but this is on a client's production system, so that's not possible ATM. Any ideas on what may be causing this, or how to diagnose further? Thanks! Troy

1 0

sofia.c:5172 Codec Error! v=0
by Max Clark 09 Mar '11

09 Mar '11

How do I identify the source of this error and fix it? 2011-03-09 16:02:42.845215 [ERR] sofia.c:5172 Codec Error! v=0 o=MxSIP 0 28211 IN IP4 x.x.x.x s=SIP Call c=IN IP4 x.x.x.x t=0 0 m=audio 5238 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 I'm actually surprised that I'm even seeing this with proxy_media=true enabled. The codec list is correct and what are supported by both endpoints. <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA"/> Thanks, Max

2 2

Pre-Answer / Sending early media Command not found
by Max Clark 09 Mar '11

09 Mar '11

I upgraded to current last night, since then I've noticed these errors on the console: -ERR 2011-03-09 15:49:15.629671 [NOTICE] sofia.c:4794 Pre-Answer sofia/external/xxxx@y.y.y.y:5060! Command not found! freeswitch@internal> 2011-03-09 15:49:15.630671 [INFO] switch_ivr_originate.c:3365 Sending early media -ERR 2011-03-09 15:49:15.630671 [INFO] switch_ivr_originate.c:3365 Sending early media Command not found! Do I need to add/adjust configuration on my side? Thanks, Max

2 3

FreeSWITCH Conference Call Today: Special Presentation
by Michael Collins 09 Mar '11

09 Mar '11

Hello all! Today we are having a special presentation by Jim Gettys and Dave Taht of the Bufferbloat project. Please join us and learn more about this subject - you'll be glad you did. The official agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_03_09 Talk to you soon! -Michael

1 0

Re: [Freeswitch-users] tls/srtp goes straight to voicemail
by Mitch Johnson 09 Mar '11

09 Mar '11

It turns out that the Eyebeam app was the cause of the issue. Using the Acrobits client showed no issues whatsoever. This was discovered with a lot of help from another Mitch. > From: Michael Collins <msc(a)freeswitch.org> > Date: March 8, 2011 4:49:26 PM EST > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Subject: Re: [Freeswitch-users] tls/srtp goes straight to voicemail > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > Are you sure that the Eyebeam clients support SRTP? It looks like you're getting a 406: > > 2011-03-05 23:21:08.650486 [NOTICE] sofia.c:5323 Hangup sofia/internal/sip:1005@192.168.60.135:53664 [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED] > > You probably need to look directly at the SIP traces and see if there is any more information as to what's going on. > > -MC > > > >

1 0

send and receive call for non registered endpoint.
by Gary Chen 09 Mar '11

09 Mar '11

I have a IAD with PRI interface and multiple SIP users. I do not want to register each user on Freeswitch. I do still want each user to be able to send call and receive call. For sending call from IAD to FS, since all the sip calls have to go through SIP-router (Kamailio) and then routed to FS, I am thinking using X-AUTH-IP in ACL so no auth required. For receiveing call from FS, I am thinking either to setup each user on FS and manually create contact information and make them never expire or add cidr to each user and modify dial-string to use cidr instead of sip contact. Are these approaches workable? What is the best way to do this? Gary

1 0

incoming call and callee busy
by Madovsky 09 Mar '11

09 Mar '11

What's the A leg SIP return status of the callee if the calle is already in early media status with another call ? Thanks

2 1

FreeTDM and Libpri with a Digium TE122 ... some calls not being answered...
by Matt Paine 09 Mar '11

09 Mar '11

Hi Guys. Current setup is a FS box with a Digium TE122 installed, with DAHDI kernel drivers and using FreeTDM with a libpri span... == autoload_configs/freetdm.conf.xml == <configuration name="freetdm.conf" description="FreeTDM Configuration"> <settings> <param name="debug" value="0"/> <param name="sip_headers" value="true"/> </settings> <libpri_spans> <span name="ISDN"> <param name="mode" value="user"/> <param name="dialect" value="q931"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> </span> </libpri_spans> </configuration> == freetdm.conf == [general] cpu_monitor => yes cpu_monitoring_interval => 1000 cpu_set_alarm_threshold => 80 cpu_reset_alarm_threshold => 70 cpu_alarm_action => warn [span zt ISDN] trunk_type => E1 b-channel => 1-15 d-channel => 16 b-channel => 17-21 == zt.conf == [defaults] codec_ms => 20 wink_ms => 150 flash_ms => 750 echo_cancel_level => 128 rxgain => 0.0 txgain => 0.0 == /etc/dahdi/system.conf == span=1,1,0,ccs,hdb3,crc4 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 # Global data loadzone = au defaultzone = au ========= And finally some logs from two calls, a successful one, and a failed one... http://pastebin.freeswitch.org/15533 If anyone could offer any suggestions as to what I can do to answer every incoming call into the box that would very appreciated. Of course if I need to provide more information I am willing to, and any suggestions for settings that need changing to test out will also be great. Thank you in advance Matt

2 2

native default sounds
by Madovsky 08 Mar '11

08 Mar '11

are default us callie sounds in speex already exist somewhere ? or should I need to convert them on my own ? Thanks Franck

2 1