Re: [asterisk-users] Inbound calls from Asterisk to Asterisk with SIP "Forbidden" from '"asterisk"

[Rafael Canchola]

You can use next parameter:
Fromuser = VoipDirect777821



At 04:23 p.m. 01/12/2008, Shaun Wingrin wrote:

Please help.

Asterisk 1: Sip.conf
[VoipDirect777821]
type=friend
host=dfvvd.dyndns.org
username=VoipDirect777821
secret=
accountcode=5260477782
amaflags=billing
context=Incoming
disallow=all
allow=g729
;allow=alaw
;allow=ulaw
trunk=no
qualify=yes
qualifysmoothing=yes
nat=no
canreinvite=yes
dtmfmode=rfc2833
;directrtpsetup=no
t38pt_udptl = yes

Asterisk 2 sip.conf
  GNU nano 1.3.12  File: sip_custom.conf

[VoipDirect777821]
type=friend
host=141.122.139
username=VoipDirect777821
secret=wsPiOov8830
accountcode=5260477782
amaflags=billing
context=Incomming
disallow=all
allow=g729
;allow=alaw
;allow=ulaw
trunk=no
qualify=yes
qualifysmoothing=yes
nat=no
canreinvite=yes
dtmfmode=rfc2833
;directrtpsetup=no
t38pt_udptl = yes

sip show peers shows both as registered.

this is the error when try and place a call from Asterisk 1 to Asterisk 2:

- Executing [EMAIL PROTECTED]:1] Dial("Console/dsp", 
"SIP/VoipDirect777821|60|") in new stack

-- Called VoipDirect777821
[Dec  1 23:20:21] WARNING[25399]: chan_sip.c:12334 
handle_response_invite: Received response: "Forbidden" from 
'"asterisk" ;tag=as070b02e2'

-- SIP/VoipDirect777821-0876c360 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
-- Executing [EMAIL PROTECTED]:2] Hangup("Console/dsp", "") in new stack
  == Spawn extension (a1, 582, 2) exited non-zero on 'Console/dsp'
 << Hangup on console >>

I get the same error even if I include this on Asterisk 1:
register => VoipDirect777821:[EMAIL PROTECTED]

Please help

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--
RafaelCanchola
Product Development Engineer,
FonetGlobal Inc.
[EMAIL PROTECTED]
http://www.fonetglobal.com
Ph. + 52 800 022 10 21 ext. 214
  + 52 442 167 08 14
VoIP 523663801
d00d! cyberalph
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Re: [asterisk-users] VOIP Provider

[Rafael Canchola]

Hi.

We recommend Fonet Global, they work with Asterisk many years ago and 
provide sip termination, DIDs, etc.




At 03:39 p.m. 02/10/2008, Steve Totaro wrote:



2008/10/2 Gregory Malsack <[EMAIL PROTECTED]>

Hi All,



Can anyone recommend a good VOIP provider in the Milwaukee/Chicago 
area? We need flat rate billing per line/trunk, trunking, did's, and 
iax or G.729 compatibility.




Thanks,

Greg

No virus found in this outgoing message.
Checked by AVG.
Version: 7.5.524 / Virus Database: 270.7.5/1703 - Release Date: 
10/2/2008 7:46 AM



Bandwidth.com is good, has flat rate, trunk, not sure about their 
stock of DIDs.


--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] Terrible Experience Net2phone A-Z termination

[Rafael Canchola]

Hi.

You can test Fonet Global Inc., its a good company and provide you 
world termination, aditional services, good rates, etc. and they work 
with Asterisk many years ago.
www.fonetglobal.com


At 06:32 a.m. 25/09/2008, Igor Hernandez wrote:
>Bruno Castelo Branco wrote:
> > you can try inphonex.com
> >
> > Steve Totaro wrote:
> >> Try Bandwidth.com or Junction Networks.  You get what you pay for.
> >>
> >> If you want a lower end provider, go with Vitelity, Gafachi, or even
> >> VoicePulse.  I am not saying they are lower end on service
> >> necessarily, but on reputation and corporate image.  Vitelity tested
> >> very well in a very limited time frame.  VoicePulse was great too but
> >> they kept making changes that resulted in outages, if engineered
> >> properly, there should be no outage short of an act of God.
> >>
> >> Thanks,
> >> Steve Totaro
> >>
> >> On Thu, Sep 25, 2008 at 3:31 AM, broadband Voice
> >> <[EMAIL PROTECTED] > wrote:
> >>
> >> I'm using Net2phone termination and the experience has been
> >> horrible for the past 2 weeks, I have put in several tickets and
> >> nothing has been done. I get a lot of congestion, channel
> >> unavailable and calls not going through. Does anyone use them? I
> >> have been using SIP debug to try to resolve it but to no avail.
> >> Are there any tier A-Z termination partners out there,
> >>
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> >>
> >>
> >> 
> >>
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> > 
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>
>Funny Net2Phone comes up. We talked to them when we were starting out
>and they wanted to charge $500 "setup" fee because we had no volume. The
>guy said "We have to charge this because we had many people coming to us
>without volume, so we charge this setup fee in order to allow us to
>still provide them service." Like that makes any sense to anyone. Either
>way, they had the worst rates in the market and claimed extremely high
>quality. I'm glad we didn't go with them.
>
>Regards,
>
>--
>Igor Hernandez
>Escape Communications
>http://www.escapetel.com
>
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Re: [asterisk-users] Problem receiving audio with Asterisk 1.4.5 with SIP trunk

[Rafael Canchola]

Hi

You can tray to making a tcpdump for detect where stay the audio 
packets (RTP) and/or stopping the iptables.
Also you may check the out route (route -n) and get the default GW, 
it should be the Public GW.





At 09:17 a.m. 26/11/2007, Zaheer K. Master wrote:

Hi All,

I'm running asterisk 1.4.5 (on AsteriskNOW beta 6 appliance) and using Snom
360/370 phones with direct SIP trunking from bandwidth.com. I can make
outgoing calls, and the person on the receiving end can hear my voice, but I
cannot hear them. I also cannot receive incoming calls to my DID number.

Here is my current setup:
Asterisk is running on a dell poweredge server with an ip of 192.168.1.55
I have setup a 1:1 NAT for asterisk with my public IP of 72.127.218.XXX
The phones have IPs of 192.168.1.150-160
I can register my phones and they work correctly for intercom, voicemail,
etc.

I think the problem is that after the SIP session has initiated, the phones
are giving an IP of 192.168.1.151 for the return audio, and those packets
are getting dropped. I'm not sure where to go from here to get the incoming
calls/audio working. Do I have to give the Asterisk box a public IP? I tried
this, and when I did I was unable to get the phones to register - probably
since they had private IPs.

Any help or suggestions would be greatly appreciated. Thanks in advance!

Regards,
Zaheer


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--
RafaelCanchola
Product Development Engineer,
FonetGlobal Inc.
[EMAIL PROTECTED]
http://www.fonetglobal.com
Ph. + 52 800 022 10 21 ext. 214
  + 52 442 167 08 00
VoIP 523663899
d00d! cyberalph
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[asterisk-users] AsteriskNOW and TDM800P

[Rafael Canchola]

Hi all

I sold new TDM800P card with 8 FXO ports, someone know if can be use 
this card on AsteriskNOW or trixbox?

What can i do for use this card?

Thanks.



--
RafaelCanchola
Product Development Engineer,
FonetGlobal Inc.
[EMAIL PROTECTED]
http://www.fonetglobal.com
Ph. + 52 800 022 10 21 ext. 214
  + 52 442 167 08 00
VoIP 523663899
d00d! cyberalph
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Re: [asterisk-users] Grandstream GXV-3000

[Rafael Canchola]

Hi.

Check the codec allowing, disallow=all and allow=ulaw etc.


At 02:25 p.m. 25/10/2007, hin lee wrote:
>I am trying to set up a Grandstream GXV-3000 Video
>phone to Asterisk ver 1.2.21.1.  The problem I'm
>having is that it can call other SIP phones, but not
>vice versa.  Can someone tell me where is the problem?
>TIA!
>
>Here's part of my configurations:
>
>--
>sip.conf
>--
>; 113 is the Grandstream phone
>[113]
>type=friend
>username=113
>secret=secret
>context=default
>dtmfmode = rfc2833
>host = dynamic
>qualify = yes
>allow = h263
>video=yes
>videosupport=yes
>
>; 112 is the X-Lite phone
>[112]
>type=friend
>host=dynamic
>user=112
>username=112
>secret=secret
>allow=all
>nat=no
>
>-
>extensions.conf
>-
>exten => 112,1,Dial(SIP/112)
>exten => 112,2,Playback(vm-nobodyavail)
>exten => 112,102,Playback(tt-allbusy)
>exten => 112,103,Voicemail([EMAIL PROTECTED])
>
>exten => 113,1,Dail(SIP/113)
>exten => 113,2,Playback(vm-nobodyavail)
>exten => 113,102,Playback(tt-allbusy)
>exten => 113,103,Voicemail([EMAIL PROTECTED])
>
>__
>Do You Yahoo!?
>Tired of spam?  Yahoo! Mail has the best spam protection around
>http://mail.yahoo.com
>
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Re: [asterisk-users] G729a codecs + Asterisk 1.4.11

[Rafael Canchola]

Hi:

You can check the next command: show g729
and you should see some like this "0/0 encoders/decoders of 2 
licensed channels are currently in use"

or
the command show translation
or check the asterisk log may be the module is not for you processor version.

Best Regards


At 09:06 a.m. 10/10/2007, you wrote:

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Good Morning,
Any help would be grateful to help me understanding what's wrong...

I have bought 2 g729a licenses to digium and I would like to have 
them works...
My processor is an Intel(R) Xeon(R) CPU   E5310  @ 1.60GHz 
(4 processors)
so I have downloaded the 
http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.4/x86-64/codec_g729a_v32_nocona.tar.gz 
codec
I have registered my license, copied the codec_g729a.so into the 
/usr/lib/asterisk/modules folder and restarted my asterisk


But on the CLI when I type
asterisk*CLI> show modules like 72
Module Description 
   Use Count

codec_g726.so  ITU G.726-32kbps G726 Transcoder 0
format_g729.so Raw G729 data0
format_g726.so Raw G.726 (16/24/32/40kbps) data 0
format_g723.so G.723.1 Simple Timestamp File Format 0

The codec_g729a.so doesn't appear..


Any idea how to solve the problem.

Thanks

Best Regards,

Marc LEURENT
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (GNU/Linux)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFHDNxdqjpLE0HiOBYRAug5AJ4qjE57UcgHEsmAVQFwPSyMn/dyogCeP3qG
UKXWhR9ebm2iw2Ao8VLuSEk=
=7O/k
-END PGP SIGNATURE-

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--
RafaelCanchola
Product Development Engineer,
FonetGlobal Inc.
[EMAIL PROTECTED]
http://www.fonetglobal.com
Ph. + 52 800 022 10 21 ext. 214
  + 52 442 167 08 00
VoIP 523663899
d00d! cyberalph
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Re: [asterisk-users] inbound call voip providers

[Rafael Canchola]

Yes, you can receive call from a 800s number or local DID, I am a DID
Provider please check next webpage

www.fonetglobal.com


At 03:40 p.m. 08/10/2007, srgqwerty wrote:
Hello:
I want to have a local telephone number that, when the people calls this

number (via mobile or normal PSTN), the voip provider stablishes a SIP

session to my asterisk box.
It is possible?
If yes...
What providers have this service in Europe?
It is difficult to configure and get things working ok?
Will my asterisk box see the mobile or normal PSTN phone# that is calling
the 
number  (callerID)?
Who many SIMULTANEOUS sessions (calls) can be active in this
scenario?
It is possible to have multiple asterisk boxes and tell it to the voip

provider in order to "balancing" the asterisks?
Can my asterisk box have IVR support under this scenario?
Thanks a lot.
Regards
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RafaelCanchola
Product Development Engineer,

FonetGlobal Inc.
[EMAIL PROTECTED]  

http://www.fonetglobal.com
Ph.
+ 52 800 022 10 21 ext. 214
  + 52 442 167 08 00
VoIP
523663899
d00d!
cyberalph




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Re: [asterisk-users] how to route outgoing calls on IP-level

[Rafael Canchola]

Check the "route" command on your Linux system. The gateway
route should be the ethX and network whatever you want.

At 01:41 p.m. 14/09/2007, Drew Gibson wrote:
Kate Kretz wrote: 
Dear Sirs,
out asterisk server has multiple network cards.
I want some outgoing calls (from several extensions) to use one IP
address, and others to go through
another address.
is there a way to achive that using asterisk ? 
Cheers,
Kate
This is the job of your network, not Asterisk. Policy-based
routing is not much fun (unless you think the Cisco CLI is "really
cool") but it can be done.
regards,
Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com

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RafaelCanchola
Product Development Engineer,

FonetGlobal Inc.
[EMAIL PROTECTED]  

http://www.fonetglobal.com
Ph.
+ 52 800 022 10 21 ext. 214
  + 52 442 167 08 00
VoIP
523663899
d00d!
cyberalph




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Re: [asterisk-users] International Carriers

[Rafael Canchola]

Hi:
I am working in a VoIP Carrier Company, I could provider you the service
for your internationals calls. 
Please visit

www.fonetglobal.com and call me, my phone number is +52 442 167 08 00
x214 Rafael Canchola.
Thanks.
At 09:54 a.m. 26/01/2007, Facundo Ameal wrote:
Hello everyone!
I 've looking for carriers which can terminate my international
calls.
They must accept payments from Argentina and give me interconection
to
my Asterisk. I'd appreciate your help or recomendations.

Regards.
-- 
Facundo Ameal.
famealgmailcom
Linux User #395088
Share your knowledge, use free software.
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RafaelCanchola
Product Development Engineer,

FonetGlobal Inc.
[EMAIL PROTECTED]  

http://www.fonetglobal.com
Ph.
+ 52 800 022 10 21 ext. 214
  + 52 442 167 08 00
VoIP
523663899
d00d!
cyberalph



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[Asterisk-Users] i need limit my outgoing context

[Rafael Canchola]

Hi.
I have a problem or require in my Asterisk, I need limit the out calls
from my outgoing context. I have configure a outgoing peer in
sip.conf  [outgoing-xxx], but I need that in this peer out calls 4
only.
I configure my outgoing peer with registry parameters for out calls with
my SIP provider and I prove changing the configuration next:
outgoinglimit=4 and incominglimit=4, but this not found. This parameters
limit the extensions only, but not all SIP calls that out for my SIP
provider context.
Please help me with this issue.
Thanks.


Rafael Canchola
Medina
Soporte Tecnico FonetGlobal Inc.
[EMAIL PROTECTED]
01 800 022 10 21 ext. 120 



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