Hi Giovanni,
I put everything you aked for in archive and attached it to the bug report
at http://jira.freeswitch.org/browse/MODSKYPIAX-28
Hope it'll help to resolve this issue.
Best Regards, Dmitry
Giovanni Maruzzelli-3 wrote:
>
> Ciao Dmitry,
>
> The warnings are unharmful, I've just fixe
Thank you Dmitry,
I'll have a look into it this evening (6 hours from now :-) )
Sincerely,
Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039
On Mon, Mar
Use SVN, or wait for the next release, fs_cli+siptrace rocks :)
On Sun, Mar 8, 2009 at 12:03 PM, Nik Middleton
wrote:
> That's exactly what I was looking for, many thanks
>
> Regards,
>
> -Original Message-
> From: freeswitch-users-boun...@lists.freeswitch.org
> [mailto:freeswitch-users-b
The IP550, 650 and 670 DO NOT support any G722.1 codecs at this
point... expect support for those later in the year... right now they
only support G722.
/b
On Mar 9, 2009, at 1:44 AM, zhaoxxqq wrote:
Hello,
I'm a newbe of Freeswitch. I have tried to config Polycom's
soundpoint IP550 to u
that means you should report it to jira not the mailing list.
On Sun, Mar 8, 2009 at 1:28 AM, Diego Viola wrote:
> Oh, I noticed the billing actually works, it discounts from my credit
> but I still get that message, even if the update works.
>
> "2009-03-08 00:37:02 [CRIT] mod_nibblebill.c:286
Hi,
This issue is that our mod_iax is using the only freely available iax2
stack. A client library that was only designed for small softphones.
There is a patch in jira to add registration support but it was not done
correctly and we have not had much time to work on it.
We've already had to add s
Hello,
following scenario:
-Phone A is redirected unconditionally to phone C
-Phone B calls A
-Phone A replys with 302 and Dieversion header
-FS detects the 302 and sends out a new INVITE to C
I found that FS doesnt' include the received diversion sip header into
the new INVITE sent to phone C.
Thank you,
I appriciate that you get some benifits from our efforts.
We only recommend irc because it's an easy-to-access multi user chat where
we can put all of the people who need help in the same room in real time so
they can help each other and we can help them.
I tried not to get annoyed ab
Hi everybody,
I'm trying to use Sangoma A500 BRI card with OpenZap, but it does not work.
Can somebody help to to configure it?
Problem log (Incoming call):
2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event()
Unhandled event 2
2009-03-06 14:58:14 [WARNING] ozmod_
> [devices]
> wanpipe1 = WAN_AFT_TE1, Comment
>
> [interfaces]
> wp1aft1 = wanpipe1, auto, API, Comment
> wp1aft2 = wanpipe1, auto, API, Comment
>
> [wanpipe1]
> CARD_TYPE = AFT
> S514CPU = A
> CommPort = PRI
> AUTO_PCISLOT = NO
> PCISLOT = 4
> PCIBUS
it's not released yet,
please wait for the announcement that it has been completed sometime in the
next week or 2.
On Mon, Mar 9, 2009 at 1:41 PM, Sergey Kirillov
wrote:
> Hi everybody,
>
> I'm trying to use Sangoma A500 BRI card with OpenZap, but it does not work.
>
> Can somebody help to to c
Hi Giovanni,
Finally I was able to manage the problem. It was my fault. I didn't
realized, that the value of the "name" parameter in this line: should strictly correspond to skype name you
regester in startskype.sh script. It can not be arbitrary chosen, as I
thought before. After I fixed that,
Ok, done.
Thanks.
On Mon, Mar 9, 2009 at 11:18 AM, Anthony Minessale
wrote:
> that means you should report it to jira not the mailing list.
>
>
> On Sun, Mar 8, 2009 at 1:28 AM, Diego Viola wrote:
>>
>> Oh, I noticed the billing actually works, it discounts from my credit
>> but I still get tha
Hi all,
I'm looking to implement an admin panel much like the one used at
http://conference.freeswitch.org. Now I obviously cannot login and see the
"admin" part of the panel but I'm pretty sure whats in there.
I have xml_rpc running and can connect via http and issue commands. I've
searched the
Hi,
I have the FS worked perfectly under NAT. And when I moved it to a server
with public IP, things getting wrong.
This is the error message that I got:
2009-03-09 21:31:23 [ERR] sofia_glue.c:559 sofia_glue_ext_address_lookup() STUN
Failed! stun.freeswitch.org:3478 [Remote Address Error!]
--
Sounds like DNS failure maybe... might wanna remove the ext-sip-ip and
ext-rtp-ip setting out of external.xml to take care of that.
west philadelfia born and raised?
/b
On Mar 9, 2009, at 8:36 PM, Will Smith wrote:
> Hi,
> I have the FS worked perfectly under NAT. And when I moved it to a
Will Smith wrote:
> I tried whatever I can think of like;
> set the
> or
> but still got the error.
> Could you please give me some guide how to fix this.
Change external_sip_ip and external_rtp_ip settings in vars.xml or in your
external SIP profile. By default these are configured to use
stu
Thank you Brian, it works like a champ.
Yes, west philadelfia born and raised?
--- On Mon, 3/9/09, Brian West wrote:
From: Brian West
Subject: Re: [Freeswitch-users] STUN error
To: freeswitch-users@lists.freeswitch.org
Date: Monday, March 9, 2009, 6:52 PM
Sounds like DNS failure maybe... mig
Small joke :P Do you get that a lot?
/b
On Mar 9, 2009, at 10:09 PM, Will Smith wrote:
Thank you Brian, it works like a champ.
Yes, west philadelfia born and raised?
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://li
Hi all,
I wrote a ruby script. it works for me. The script is in /scripts/
contrib/seven/sip/.
All ideas and suggestions are welcome!
Comment in script:
Now and then we need to look at sip traces to see want happened on a
failed call. There are lots of ways
to monitor sip messages. However,
Hi,
I wanted to use A2Billing on FS, but I noticed it uses some AGI stuff
for dialling and to check how much credit the user has, etc. I heard
you could use A2B by just importing the FS CDR data into it, but that
wont work, so I come to the conclusion that I have no way of doing
billing on FS yet.
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