- FreeSWITCH-users - FreeSWITCH Mailing Lists

DCS in IP telephony communication standards
by efim_serg 11 Jan '11

11 Jan '11

Could somebody tell me what does mean 'DCS' in IP telephony communication standards? I have looked at the internet and found that DCS is a distributed control system, but I think it's not relevant information when we talk about IP telephony communication standards. I also need find interrelation between DCS and FreeSwitch. Thanks. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/DCS-in-IP-telephony-communica… Sent from the freeswitch-users mailing list archive at Nabble.com.

1 0

Any one use FreeCanadianCalls with FreeSwitch?
by Alex Wang 11 Jan '11

11 Jan '11

Hi Any one is using FreeCanadianCalls with FreeSwitch? Would you mind to share the xml setting? Thanks Alex

2 2

Re: [Freeswitch-users] IVR menu not observing timeout value
by Kim Culhan 11 Jan '11

11 Jan '11

On Tue, January 11, 2011 1:20 am, Michael Collins wrote: > I was not able to reproduce this using the demo IVR. Can you try it with the > demo IVR and let us know what you get? Thanks for checking this, describing the problem on irc anthm asked for debugging output and made a code change a short time later. After the change the dtmf decode appears to be totally reliable. -kim

1 0

Re: [Freeswitch-users] IVR menu not observing timeout value
by Kim Culhan 11 Jan '11

11 Jan '11

On Mon, January 10, 2011 9:35 am, Kim Culhan wrote: > FreeSWITCH Version 1.0.head (git-4e95227 2010-12-26 09-09-14 -0600) Correction on the FS version, it should read: FreeSWITCH Version 1.0.head (git-3003489 2011-01-09 14-42-42 -0500) thanks -kim

2 1

Link2voip
by Darren Wiebe 11 Jan '11

11 Jan '11

Good Afternoon, I'm trying to get my freeswitch box talking to Link2voip. Does anybody have sample XML files for them? -- Darren Wiebe Aleph Communications -------------------- Phone: 1-877-702-2900 Fax: 1-866-274-4506 Email: darren(a)aleph-com.net

4 8

gateway context
by deniro 11 Jan '11

11 Jan '11

Hi, This is in regards to how to tell my new sip profile or gateway definition to use different context within new gateway definition. In conf/sip_profile/external.xml I see it tells to use public context <settings> <param name="context" value="public"/> I define a new sip_profile or gateway and within that I try to tell to use a different context like conf/sip_profile/external/incoming.xml <gateway> ........... <param name="context" value="incoming"/> <gateway/> (i put that "incoming" context under dialplan/public/xxx.xml file) The problem, it still it reads public context from external.xml (checked log files) How do I make my gateway/sip profile use a different context than the public context in external.xml file. thx

2 1

auto-nat is not working
by Alex Wang 10 Jan '11

10 Jan '11

Hi I am very new about the freeswitch and just finished setup a new freeswitch and config a test sip account from my voip provider. in the fs_cli, originate sofia/gateway/gw1/xxxxxx(my other number) &echo() I got the phone ring but no echo. originate sofia/user/1000 &echo() works fine. nat_map status shows; freeswitch@internal> nat_map status Nat Type: UNKNOWN, ExtIP: 0 total. freeswitch@internal> sofia status profile internal ================================================================================================= Name internal Domain Name N/A Auto-NAT false DBName sofia_reg_internal Just wonder how to fix the NAT issue on my freeswitch. The router is WRT54GL with Tomato 1.28 on it, which seems support upnp... Thanks a lot Alex

5 15

Problems with recorded messages cutting off at 60 seconds involving remote SDP
by Andy Ayers 10 Jan '11

10 Jan '11

Hi, I have a weird problem where on certain calls when someone tries to leave a recorded message the call hangs up exactly 60 seconds after the start of the recording. The recordFile is sending the stream to icecast using a shout: address in MP3 format. I've done a line by line compare of a successful call and one where the recording fails and the issue seems to be with the remote SDP. RECORDING SUCCEEDS WHEN SDP IS.. v=0 o=root 1060 1060 IN IP4 77.240.60.33 s=session c=IN IP4 77.240.60.33 t=0 0 m=audio 13850 RTP/AVP 8 0 3 97 7 110 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:7 LPC/8000 a=rtpmap:110 speex/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 RECORDING CUTS OFF AFTER 60 SECS WHEN SDP IS.. v=0 o=voip 302847118 302847118 IN IP4 77.240.60.4 s=voip c=IN IP4 77.240.60.4 t=0 0 m=audio 13622 RTP/AVP 8 0 3 97 7 110 5 10 18 112 111 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:7 LPC/8000 a=rtpmap:110 speex/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 Can anyone tell me the significance of the o=voip, s=voip as opposed to o=root,s=session? Is this likely to be the cause of the problem and any idea how I fix it? Many thanks as always Andy Ayers

1 0

Getting Error ORA-923 in oracle query execution
by Rajesh Bansal 10 Jan '11

10 Jan '11

Hi , I am getting error ORA -923 (from missing from statement) when i am tring to execute a sql query from Javascript file. Here i am successfully able to connect & execute queries with MYSQL. But in oracle connection i am getting this error even i can successfully make a connection with oracle. with isql & a program written in C i can connect and execute queries ok. I am using FreeSwitch 1.0.6 unixOdbc 2.3.0 Please tell me where is problem. Best Regards, Rajesh Bansal

3 6

Unable to get Freeswitch to register with Grandstream HT503
by Charles Bujold 10 Jan '11

10 Jan '11

Unable to get the HT-503 to register with Freeswitch. Does anybody know what configuration setting in the HT503 that are needed. Thanks cjb

3 4