Re: [Freeswitch-users] dialpaln

Hi Brian, I tried as you suggested, but still my outbound is not yet thru, my log as follows, *default.xml* *Freeswitch Log* 2008-09-18 13:24:40 [INFO] mod_dialplan_xml.c:228 dialplan_hunt() Processing FreeSwitch->9894929942 in context default 2008-09

[Freeswitch-users] Can't find user (Anonymous calls)

When I make a call (from SJphone) to sip:[EMAIL PROTECTED] than I can see in Freeswitch: 2008-09-18 10:16:19 [WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [EMAIL PROTECTED] 2008-09-18 10:16:19 [DEBUG] sofia_reg.c:553 sofia_reg_handle_register() send challange for [EMAIL PROTEC

Re: [Freeswitch-users] How to get the content of SIP headers and sdp in dialplan?

Thank you for your replies. In checking the SIP messaging sequence, I saw that you are right about the source of the tag to the To-header: it is the phone, which as a response to the INVITE, adds the tag (and not, as I wrote, the FreeSWITCH). Still, we need a way to read it, and not the least to se

[Freeswitch-users] Turn on CN packet generation

Hello, Is there a way to let FS send RTP CN packets by itself? What I mean is for example when I turn on VAD for an outbound call and allow CN (suppress_cng=false) than lets FS sends out CN packets even when party A is not sending them. http://lists.freeswitch.org/mailman/listinfo/fre

[Freeswitch-users] billing platform

has any1 been able to use ASTPP billing with freeswitch and if so then my next question, is it possible to use it on a win2k3 platform and not linux as i have freeswitch running on a win2k3 server since a few days and works fine, i just need a prepaid type of script that basically stops a call onc

Re: [Freeswitch-users] billing platform

On Thu, Sep 18, 2008 at 4:24 PM, xbipin <[EMAIL PROTECTED]> wrote: > > has any1 been able to use ASTPP billing with freeswitch and if so then my yes, darren and a few others have, i have it under testing but not production > next question, is it possible to use it on a win2k3 platform and not

[Freeswitch-users] transcoding / g729 / g723.1

How hard would it be to add support for g729/g723.1 transcoding using the TC400B card (digium)? It's all part of the zaptel driver. Cristian ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/lis

Re: [Freeswitch-users] Test Proxy Media

check http://wiki.freeswitch.org/wiki/Bypass_Media -> "How to verify that it is working" opposite to the example you actually want to see media passing through. cristian msp wrote: > Hi all, > > I have enabled proxy media from dialplan. > After that, I can make calls same as it done before with

Re: [Freeswitch-users] transcoding / g729 / g723.1

it would be possible with proper funding. On Thu, Sep 18, 2008 at 6:30 AM, Cristian Talle <[EMAIL PROTECTED]> wrote: > How hard would it be to add support for g729/g723.1 transcoding using > the TC400B card (digium)? It's all part of the zaptel driver. > > Cristian > > __

Re: [Freeswitch-users] Can't find user (Anonymous calls)

Turn off the auth-calls = true in your profile if you do not require authentication for the inbound calls. or add accounts for your phone so it can make authenticated calls. btw, you appear to be using an older version of the code. You may want to update to help us beta test the 1.0.2 release.

[Freeswitch-users] mod_openzap & PRI

We have a lot of questions about mod_openzap & PRI. 1. If to see directory libs/openzap/src/isdn it is possible to see, there are defined such dialects : Q931, national, EvroISDN, DMS, 5ESS. $ ls *.c 5ESSmes.c DMSStateNT.c mfifo.c Q921.c Q931mes.c 5ESSState

[Freeswitch-users] mod_openzap

In a directory libs/openzap/src/isdn [EMAIL PROTECTED] isdn]$ ls *.c 5ESSmes.c DMSStateNT.c mfifo.c Q921.c Q931mes.c 5ESSStateNT.c DMSStateTE.c nationalmes.cQ931api.c Q931StateNT.c 5ESSStateTE.c EuroISDNStateNT.c nationalStateNT.c Q931

Re: [Freeswitch-users] xml_curl and gateways

Peter P GMX wrote: > According to http://wiki.freeswitch.org/wiki/Mod_xml_curl I'v tried to > set up gateways dynamically, but I do net get it to work: > I always get : "Invalid profile" > > My assumptions for a right xml answer back to FS are as follows, but I > think at least one of it is false

Re: [Freeswitch-users] mod_openzap

Evgeniy, I've only got a few minutes so I have to be quick. Right now we only support the "user" or TE (terminal) side of PRI. Still working on the network side. If you need to do a back-to-back setup for testing then your options are a bit limited, but I do know that YATE and Asterisk both have

Re: [Freeswitch-users] mod_openzap & PRI

Evgeniy, I'm not sure why EuroISDN isn't listed in the enum or string below because I know that Stefan (stkn on IRC) has been working on it for some time. I'll follow up with him and Mike Jerris to see what's going on there. As for seeing the dialect at run-time, right now you can't, although I d

Re: [Freeswitch-users] mod_openzap & PRI

the dialects are not all done. we so far can do DMS100 and euro please join #openzap on irc.freenode.net and join the effort. On Thu, Sep 18, 2008 at 8:48 AM, Evgeniy Zolotov <[EMAIL PROTECTED]> wrote: > We have a lot of questions about mod_openzap & PRI. > > 1. If to see directory libs/openz

Re: [Freeswitch-users] xml_curl and gateways

Nobody has an idea anbout this? Peter P GMX schrieb: > According to http://wiki.freeswitch.org/wiki/Mod_xml_curl I'v tried to > set up gateways dynamically, but I do net get it to work: > I always get : "Invalid profile" > > My assumptions for a right xml answer back to FS are as follows, but I

Re: [Freeswitch-users] Can't find user (Anonymous calls)

Anthony Minessale-2 wrote: > > Turn off the auth-calls = true in your profile if you do not require > authentication for the inbound calls. or add accounts > for your phone so it can make authenticated calls. > In which file do I need to do that ? (Have gone through most of them, but can't fin

Re: [Freeswitch-users] Can't find user (Anonymous calls)

in the sofia profile auth-calls is set to true and context is set to public change auth-calls to false and change context to default On Thu, Sep 18, 2008 at 9:54 AM, henkoegema <[EMAIL PROTECTED]>wrote: > > > Anthony Minessale-2 wrote: > > > > Turn off the auth-calls = true in your profile if

Re: [Freeswitch-users] Test Proxy Media

Bypass isn't the same as proxy media. Bypass the media will never touch FreeSWITCH and flows p2p. Proxy the media will proxy thru FreeSWITCH without looking at the packets. Press F8 and look at the debug logs or type "console loglevel 8" You should see something about proxy codec. /b On

Re: [Freeswitch-users] Turn on CN packet generation

Hello, to answer for my own question: Tamas írta: > Hello, > > Is there a way to let FS send RTP CN packets by itself? > No, FS just forwards the CN packets (or absorbs). > What I mean is for example when I turn on VAD for an outbound call and > allow CN (suppress_cng=false) than lets FS send

Re: [Freeswitch-users] billing platform

I've been trying to install. The problem is the lack of documentation on how to integrate with FS. Jair Santos > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On > Behalf Of xbipin > Sent: Thursday, September 18, 2008 3:25 AM > To: freeswitch-users@li

[Freeswitch-users] Accessing the variable shown by the application "info"?

I have tried to access the variables shown by the application info, and the variable name to be used is generally not the same as the name listed by info. I have listed nearly 100 variables below, where I guessed the "real" names, but a number are still missing, and I would like to know what the "r

[Freeswitch-users] Ringback disappeared

I have for more than 1 year now sent my inbound calls to an application running on another machine listening to FreeSWITCH's outbound socket. I have this in my dialplan: In my external application, I look at the called number, and redirects it back to FreeSWITCH with

Re: [Freeswitch-users] billing platform

are u interested in a paid implementation as im willing to post a bounty for it if there r others like us willing to use it with freeswitch. wikipbx is also good for a start but one thing i dont understand is if freeswitch is made for cross platform then y r other projects related to it limiting i

Re: [Freeswitch-users] Test Proxy Media

I know what the difference is but the example at hand was in the bypass section of wiki (of course for the opposite scenario). I'll keep in mind the proxy codec thing for the future though :) Brian West wrote: > Bypass isn't the same as proxy media. > > Bypass the media will never touch FreeSWITC

Re: [Freeswitch-users] Accessing the variable shown by the application "info"?

http://wiki.freeswitch.org/wiki/Channel_Variables Some of these aren't on that wiki page maybe you can help fill them in. I'll need to double check the direction and state ones but that should be it. > Unique-ID---Unique_ID---FALSE ${uuid} > Call-Direction---Call_Direction---FALSE ${direct

Re: [Freeswitch-users] transcoding / g729 / g723.1

Any idea what are we looking at? I already have a couple of those cards available. Cristian 416 548 7916 ext 120 Anthony Minessale wrote: > it would be possible with proper funding. > > > On Thu, Sep 18, 2008 at 6:30 AM, Cristian Talle <[EMAIL PROTECTED] > > wrote: >

Re: [Freeswitch-users] Test Proxy Media

Yes, I understand you might have known the difference but for the sake of the archive and everyone else that might come across this thread. I thought I would clarify since the link was about bypass media and not proxy media ;) /b On Sep 18, 2008, at 11:00 AM, Cristian Talle wrote: > I kno

Re: [Freeswitch-users] mod_openzap & PRI

Thanks for all. But I am in perplexity now - how we can khow, what any certain dialect is carried out? Look for this example: > If we'll specify for nonexistent dialects: > > > > > > > > > > > - then >load mod_openzap is carried out successfully and >oz dump

Re: [Freeswitch-users] dialpaln

Your dialplan is fixed now, this is an issue with openzap now, can you clarify your openzap configuration and what kind of line it is hooked too please? Mike On Sep 18, 2008, at 3:55 AM, Gopal krishnan wrote: Hi Brian, I tried as you suggested, but still my outbound is not yet thru,

Re: [Freeswitch-users] dialpaln

post openzap.conf.xml openzap.conf and outbout of ztcfg -vv note: openzap does not ask for opposite names like asterisk does in openzap.conf if it's fxo say fxo not fxs On Thu, Sep 18, 2008 at 11:18 AM, Michael Jerris <[EMAIL PROTECTED]> wrote: > Your dialplan is fixed now, this is an issue

Re: [Freeswitch-users] How to get the content of SIP headers and sdp in dialplan?

On Sep 18, 2008, at 4:53 AM, Jon Bruel wrote: Thank you for your replies. In checking the SIP messaging sequence, I saw that you are right about the source of the tag to the To- header: it is the phone, which as a response to the INVITE, adds the tag (and not, as I wrote, the FreeSWITCH). S

Re: [Freeswitch-users] billing platform

On Sep 18, 2008, at 12:00 PM, xbipin wrote: > > are u interested in a paid implementation as im willing to post a > bounty for > it if there r others like us willing to use it with freeswitch. > wikipbx is > also good for a start but one thing i dont understand is if > freeswitch is > made

Re: [Freeswitch-users] mod_openzap & PRI

euro should work fine with the q931 dialect. Miike On Sep 18, 2008, at 12:04 PM, Evgeniy Zolotov wrote: Thanks for all. But I am in perplexity now - how we can khow, what any certain dialect is carried out? Look for this example: > If we'll specify for nonexistent dialects: > > > >

Re: [Freeswitch-users] Ringback disappeared

On Sep 18, 2008, at 11:53 AM, Ivan C Myrvold wrote: > I have for more than 1 year now sent my inbound calls to an > application running on another machine listening to FreeSWITCH's > outbound socket. > I have this in my dialplan: > > > In my external application, I look at

Re: [Freeswitch-users] xml_curl and gateways

Hello Raymond, I had a look at your code. You are submitting a complete profile for sofia.conf, right? Best regards Peter Raymond Chandler schrieb: > Peter P GMX wrote: > >> According to http://wiki.freeswitch.org/wiki/Mod_xml_curl I'v tried to >> set up gateways dynamically, but I do net

Re: [Freeswitch-users] mod_openzap & PRI

> Is it possible to add our own dialect PRI (for another submission of > SUBBADDRESS, for example), if we'll add > new files *mess.c, *StateNT.c, *StateTE and definition * into the > Q931Dialect_t ? We have added new dialect ( li_ewsd, similar to Q.931, but different that bytes of data SUBADDR

Re: [Freeswitch-users] mod_openzap & PRI

Thanks, Mike. But what dialect works when we make or How we can define it (See example below) - Original Message - From: Michael Jerris To: freeswitch-users@lists.freeswitch.org Sent: Thursday, September 18, 2008 7:43 PM Subject: Re: [Free

Re: [Freeswitch-users] billing platform

Hello, We did actually start writing a billing platform for FreeSwitch back when it didnt even have the Socket API as a C module to FreeSwitch (compiled on linux platform as such), but the core developer moved away and the project is in deserted mode. So we do have a lead of it, if someone want

[Freeswitch-users] DID partially working

I have DID assigned to extension 1000 in public.xml Calling this DID from extension 1001 works but when calling from a regular phone it doesn't work. Does anybody have any idea about what I am missing ? thanks Jair Santos __

Re: [Freeswitch-users] xml_curl and gateways

Peter P GMX wrote: Hello Raymond, I had a look at your code. You are submitting a complete profile for sofia.conf, right? correct. not only a profile, but the entire sofia.conf with all of its profiles and all of their gateways, settings, aliases, etc -Ray Best regards Peter Raymond

Re: [Freeswitch-users] Can't find user (Anonymous calls)

Anthony Minessale-2 wrote: > > in the sofia profile > > auth-calls is set to true > and context is set to public > > change auth-calls to false > and change context to default > Changed the above settings in ./conf/sip-profiles/internal.xml and now ..:drunk: Henk -- View this me

[Freeswitch-users] Hooking into the Audio Stream

All, Due to the limited audio abilities implemented thus far in FreeSwitch I am needing to handle the audio streams outside of FreeSwitch. So can I use Lua to pass the audio back and forth to an outside library as streamRecord and FilePlay do not provide the functionality needed? And if so what F

Re: [Freeswitch-users] How to get the content of SIP headers

As I see it, the NOTIFY methods mentioned are used when there is a call, i.e. at the earliest when the phone has responded with a RINGING or a similar response to the INVITE. By the way, the response to the INVITE also includes a list of events (Header: Allow-Events) which the phone responds to. I

[Freeswitch-users] At the least

All, To overcome audio limitations at the least can one use a callback in Lua to pull and push raw audio? And if so could both the push and pull be done simultaneously per caller? Thanks, Bob ___ Freeswitch-users mailing list Freeswitch-users@lists.fre

Re: [Freeswitch-users] DID partially working

How did you try to troubleshoot, Here comes use of ngrep ngrep port 5060 Follow the output, This to check 1, If you are getting the call on your switch? 2. How is your switch responding to the call? 3. If everything is fine until here then look in your dialplan. Jai www.didforsale.com *Buy DIDs

[Freeswitch-users] Asynchronous communication with FreeSWITCH's mod_event_socket

I've been doing a lot of work recently with FreeSWITCH's mod_event_socket, and I wanted to comment a bit about the syntax used for commands through the socket while using asynchronous mode. I haven't tried the synchronous mode yet, as I always want to be free to be able to execute commands without

Re: [Freeswitch-users] mod_openzap & PRI

my guess would be that it just fails to load. what does the debug say? Mike On Sep 18, 2008, at 1:19 PM, Evgeniy Zolotov wrote: Thanks, Mike. But what dialect works when we make or How we can define it (See example below) _

Re: [Freeswitch-users] mod_openzap & PRI

On Sep 18, 2008, at 1:13 PM, Evgeniy Zolotov wrote: > Is it possible to add our own dialect PRI (for another submission of SUBBADDRESS, for example), if we'll add > new files *mess.c, *StateNT.c, *StateTE and definition * into the Q931Dialect_t ? We have added new dialect ( li_ewsd, simi

Re: [Freeswitch-users] Hooking into the Audio Stream

On Sep 18, 2008, at 3:08 PM, Robert Clayton wrote: > All, > > Due to the limited audio abilities implemented thus far in FreeSwitch > I am needing to handle the audio streams outside of FreeSwitch. > So can I use Lua to pass the audio back and forth to an outside > library as streamRecord and Fil

Re: [Freeswitch-users] DID partially working

in which case you'll want to use 5080 and not 5060. /b On Sep 18, 2008, at 1:17 PM, Jai Rangi wrote: How did you try to troubleshoot, Here comes use of ngrep ngrep port 5060 Follow the output, This to check 1, If you are getting the call on your switch? 2. How is your switch responding to th

Re: [Freeswitch-users] DID partially working

Thanks to Brian who said he made no changes and just reload FS it is now working. :-) I am still wondering how and why because calling the DID from an internal extension was working. The problem was when calling from PSTN. Jair Santos -Original Message- From: [EMAIL PROTECTED] [m

Re: [Freeswitch-users] Alternative to directory lookup using mod_xml_curl

I gave it a try though... and it works nicely (at least for directory entries for now). I've added 3 more commands to xml_curl: cache_on, cache_off and cache_delete_key. - cache_on - enables caching: it will store in a switch_hash_t structure all xml strings returned by curl, using @ as a key

Re: [Freeswitch-users] Alternative to directory lookup using mod_xml_curl

I gave it a try though... and it works nicely (at least for directory entries for now). I've added 3 more commands to xml_curl: cache_on, cache_off and cache_delete_key. - cache_on - enables caching: it will store in a switch_hash_t structure all xml strings returned by curl, using @ as a key

Re: [Freeswitch-users] Alternative to directory lookup using mod_xml_curl

sorry, I meant switch_xml_parse_str() not switch_parse_xml_from_str() Cristian Talle wrote: > I gave it a try though... and it works nicely (at least for directory > entries for now). I've added 3 more commands to xml_curl: cache_on, > cache_off and cache_delete_key. > > - cache_on - enables cac

Re: [Freeswitch-users] Alternative to directory lookup using mod_xml_curl

Do you have any performance tests to show the difference? Mike On Sep 18, 2008, at 6:50 PM, Cristian Talle wrote: I gave it a try though... and it works nicely (at least for directory entries for now). I've added 3 more commands to xml_curl: cache_on, cache_off and cache_delete_key. - ca

Re: [Freeswitch-users] Alternative to directory lookup using mod_xml_curl

Not yet, I'm still getting familiar with it - I haven't covered the dialplan yet but I really like the voicemail module. I will set up a real server some time this week-end and I'll try stressing it. cristian Michael Jerris wrote: > Do you have any performance tests to show the difference? > >

Re: [Freeswitch-users] Asynchronous communication with FreeSWITCH's mod_event_socket

I have not worked with FS events, but this seems like a sound premise. Anyone care to enlighten an alternative? -Original Message- From: Luke Graybill <[EMAIL PROTECTED]> Sent: September 18, 2008 2:49 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Asynchronous co

Re: [Freeswitch-users] dialpaln

Hi Anthony, My conf as follows, *openzap.conf* [span wanpipe] trunk_type => e1 b-channel => 1:1-15 d-channel=> 1:16 b-channel => 1:17-31 *openzap.conf.xml* *ztcfg -vv *Still I have not installed zaptel, I have went thru that for openzap with sangoma