Re: [Freeswitch-users] FS installed, but no external calls

You'll need to set from-user on freeswitch also. /b On Jun 4, 2008, at 7:26 PM, Brian B wrote: Here are the Elastix/Asterisk settings that were working. Brian West remarked that I have to "add it to the dialplan in addition to registration". Can anyone give me a pointer on that? Inboun

Re: [Freeswitch-users] FS installed, but no external calls

Here are the Elastix/Asterisk settings that were working. Brian West remarked that I have to "add it to the dialplan in addition to registration". Can anyone give me a pointer on that? *Inbound* registration string: 16643:[EMAIL PROTECTED]/16643 User context: from-trunk User Details: canreinvit

Re: [Freeswitch-users] LDAP Internal & FreeSwitch

Nicola wrote: > thanks for your answer the problem is that FreeSwitch NOT MADE no > query the LDAP server. Consequently, I can not know whether the query is > wrong or not. I tried to follow the instructions you have given me kindly > ... but freeswitch not going to make any queries on LDAP: I

Re: [Freeswitch-users] FS installed, but no external calls

thanks for the pointer ... so the only difference between the below and Brian B's is the register param? On Wed, Jun 4, 2008 at 7:20 PM, Brian West <[EMAIL PROTECTED]> wrote: > I just want to point out one thing.. if you called the gateway > did.voip.les.net then the ream and proxy values can be

Re: [Freeswitch-users] FS installed, but no external calls

I just want to point out one thing.. if you called the gateway did.voip.les.net then the ream and proxy values can be left out. We will just use that. /b On Jun 4, 2008, at 6:14 PM, Nick Temple wrote: ___ Frees

Re: [Freeswitch-users] FS installed, but no external calls

Thanks Brian, Klaus, and Peder for your help! Here's the les2.xml file which is now showing REGED. (Thanks Peder! For any potential Les.net users: your username is the longer peer name, not your shorter les username). I tried to figure out dialplan and what to do or even show there, but can't

Re: [Freeswitch-users] FS installed, but no external calls

Here's my setup: conf/sip_profiles/outbound/lesnet.xml Then: sessionx.originate(session, "sofia/gateway/lesnet/1502XXX", 30); HTH, Nick On Wed, Jun 4, 2008 at 6:30 PM, Klaus Teller <[EMAIL PROTECTED]> wrote: > Hi Brian, > > I created a LES profile loca

Re: [Freeswitch-users] FS installed, but no external calls

how about you reply with the settings you currently use for Asterisk? /b On Jun 4, 2008, at 5:30 PM, Klaus Teller wrote: > Hi Brian, > > I created a LES profile located at sip_profiles/external/les.xml > with following content: > > > > > > > > > > > My dialing is actually done in a

Re: [Freeswitch-users] inconsistent Event XML (Event PLAIN workingfine)

do you mean noevents before or after you hangup? cos noevents first would obviously kill the events before you ever saw them. I ask because I ran script in 2 shells telneted to 8021 on a box one "events plain" and the other "events xml" Then I called 3011 and hungup again and all the events were

Re: [Freeswitch-users] FS installed, but no external calls

Hi Brian, I created a LES profile located at sip_profiles/external/les.xml with following content: My dialing is actually done in a Javascript file in the following way: session.execute("bridge","sofia/gateway/did.voip.les.net/14156113200"); Since you already have LES workin

Re: [Freeswitch-users] FS installed, but no external calls

Brian, Just registering doesn't input the routing info into the dialplan. You'll need to add it to the dialplan in addition to registration. Can you clarify how you did your registration? Maybe post your XML without user/pass in them so we can point out why it failed? /b On Jun 4, 200

[Freeswitch-users] Help Wikify Default Config

Community, We need your help. We are in the middle of documenting the default installation of FreeSWITCH. The wiki page that we're working on now is located at http://wiki.freeswitch.org/wiki/Default_config ... We need help with documenting ALL of the modules, especially the ones that are

Re: [Freeswitch-users] FS installed, but no external calls

Can you send your dialplan and directory files. It sounds like you are calling into a context with no phones in it. Brian B wrote: > FS is definitely up and running on my Lylix VPS, which is exciting. > X-Lite is dialed in and can talk to the IVR etc. But no inbound or > outbound calls. >

Re: [Freeswitch-users] inconsistent Event XML (Event PLAIN workingfine)

These steps instruct how to manually reproduce these symptoms using a VoIP endpoint (IPPhone/SoftPhone) and a terminal (telnet/putty). In step 7 you physically put the phone down (on-hook). I recommended entering in the terminal "NOEVENTS" to stop receiving events so you can scroll up and compare

[Freeswitch-users] FS installed, but no external calls

FS is definitely up and running on my Lylix VPS, which is exciting. X-Lite is dialed in and can talk to the IVR etc. But no inbound or outbound calls. I tried configuring les.net (which I got working in * on the same box) but couldn't get that working, and sofia status is NOREG. I got gizmo to

Re: [Freeswitch-users] inconsistent Event XML (Event PLAIN working fine)

I dont get step 7? hangup the call and send noevents ? On Wed, Jun 4, 2008 at 12:09 AM, UV <[EMAIL PROTECTED]> wrote: > We've been having various problems intercepting events using the > Event_Socket mod where we found different events to be missing from time to > time. This included events con

[Freeswitch-users] weekly conference call reminder

Everyone remember the call that's coming up in about two hours. The agenda was sent out yesterday and is attached below for your reference. We look forward to hearing from everyone then. Thanks, -Ray * 1 we still need to finish adopting the orphaned pages

Re: [Freeswitch-users] Asterisk extensions

Rizwan Hisham wrote: Hi all, Can i use asterisk's extensions.conf with freeswitch? yes If yes, How? http://wiki.freeswitch.org/wiki/Mod_dialplan_asterisk -- Best Regards Rizwan Hisham _

Re: [Freeswitch-users] VoiceXML?

Here are the new standards.. heavily backed by IBM and Nuance VXML2 and SSML/SCXML, Xhtml + Voice Profile, (also known as Xhtml+Voice1.) or X+V And yes, you can do it all in JS, if you preferred -E ___ Freeswitch-users mailing list Freeswitch-users@li

[Freeswitch-users] Asterisk extensions

Hi all, Can i use asterisk's extensions.conf with freeswitch? If yes, How? -- Best Regards Rizwan Hisham ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:ht

Re: [Freeswitch-users] LDAP Internal & FreeSwitch

thanks for your answer the problem is that FreeSwitch NOT MADE no query the LDAP server. Consequently, I can not know whether the query is wrong or not. I tried to follow the instructions you have given me kindly ... but freeswitch not going to make any queries on LDAP: In fact, when I try to

Re: [Freeswitch-users] LDAP Internal & FreeSwitch

Nicola wrote: > Hello, I am writing from italy ... > > I'm trying to use FreeSwitch with ldap ... > So far I managed, after many attempts to create LDAP server on the schema. > That schema was meant as an example only. I think the only attrs the module uses are: param paramvalue variable varia

[Freeswitch-users] g729

Hi, I'd like to know what's needed to add support for G729, I know there's a bounty, but I couldn't make sense out of what's posted on the wiki. I'm really interested in this, as one of my current VoIP providers restricts me to using only this codec, which limits me to using Asterisk, hence I can't

[Freeswitch-users] Bug in mod_wanpipe.c ?

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, when I try to compile mod_wanpipe I get this error: Compiling mod_wanpipe.c... mod_wanpipe.c: In function ‘wanpipe_send_dtmf’: mod_wanpipe.c:912: error: request for member ‘duration’ in something not a structure or union make[1]: *** [mod_wanp

[Freeswitch-users] VoIP Engineer in Beijing, China

Since I know there are a number of people in the community in China or with connections in 中国, I post this here. Apologizes for the spam. We are looking for a lead engineer for our VoIP Systems. Its FreeSWITCH based but with many other components that integrate with our Web based server