Thanks Luciano.
But there is no active ALG on the modem.
Attached the call flow, including the ACK.
Em 22/09/2020 14:41, Luciano Moreira escreveu:
Roberto
Check your router if ALG or similar feature is enabled. Disable and test.
Also, on SNGREP check if both parties are getting ACK correctly
The problem has been detected.
FXS equipment is causing the fall. Most likely from some bad contact.
Thank you all for your help.
Roberto.
Em 22/09/2020 14:41, Luciano Moreira escreveu:
Roberto
Check your router if ALG or similar feature is enabled. Disable and test.
Also, on SNGREP check if
20 17:12, Dovid Bender escreveu:
Is there anything in the Asterisk logs? Which side sends the BYE? Were
you able to capture the traffic with sngrep/wireshark to see if any
side stopped sending/getting RTP? What did the other side see?
On Mon, Sep 21, 2020 at 3:22 PM Roberto
<
localnet = 191.0.0.0 / 24
localnet = 201.0.0.0 / 24
localnet = 177.0.0.0 / 24
localnet = 179.0.0.0 / 24
Thanks
Roberto.
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Check out the new
[SOLVED]!!!
My function that changed the callerid was returning an invalid number.
Although the asterisk sends the call, the SIP header was wrong and the
extension did not ring
Thanks.
Em 18/08/2020 09:07, Joshua C. Colp escreveu:
On Tue, Aug 18, 2020 at 9:00 AM Roberto
oad"
Make same calls, and opening the file only the following appears:
[2020-08-18 08:46:47.778] Asterisk 17.6.0 built by root @
asterisk-homolog on a x86_64 running Linux on 2020-08-13 22:40:11 UTC\
Em 17/08/2020 18:57, Joshua C. Colp escreveu:
On Mon, Aug 17, 2020 at 6:16 PM Roberto
&
Hello.
I am having a lot of problems with SIP through NAT. So, I decided to
adopt PJSIP. However, I am not able to make the extensions ring when
receiving a call from the queue. I'm using telnet to include the
extension and on the asterisk console, it even shows Called PJSIP/6001,
but the ex
On 11/05/2015 08:56 PM, Russ Meyerriecks wrote:
> On Thu, Nov 5, 2015 at 11:36 AM, Roberto Fichera wrote:
>> The question is, can I call the dahdi_transmit() from the TX DMA callback
>> and the dahdi_receive() from
>> the RX DMA callback or should use a particular order for
se a particular order for them?
Looking at the DAHDI drivers it seems that the sequence is always:
dahdi_ec_chunk() -> dahdi_receive()
and then dahdi_transmit().
Thanks in advance,
Roberto Fichera.
--
_
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has been
configured in PtP mode.
Cheers,
Roberto Fichera.
> On 4 Aug 2014 14:36, "Roberto Fichera" <mailto:ker...@tekno-soft.it>> wrote:
>
> On 08/04/2014 01:21 PM, David Duffett wrote:
>>
>> If the OpenVox card can supply the voltage, then it will a con
On 08/04/2014 05:02 PM, Roberto Fichera wrote:
> On 08/04/2014 04:29 PM, Mc GRATH Ricardo wrote:
>> Roberto
>>
>> Could you provide more details about Panasonic PBX test? model unit and
>> configuration details?
> It's a KX-TD1232 model. The conf is nothing spe
On 08/04/2014 04:29 PM, Mc GRATH Ricardo wrote:
> Roberto
>
> Could you provide more details about Panasonic PBX test? model unit and
> configuration details?
It's a KX-TD1232 model. The conf is nothing special! It's exactly the same used
to connect the
BT Versatility.
On 08/04/2014 03:03 PM, David Duffett wrote:
>
> Please come back to let us know if this actually does fix the issue.
>
Yep! Sure!
> On 4 Aug 2014 14:36, "Roberto Fichera" <mailto:ker...@tekno-soft.it>> wrote:
>
> On 08/04/2014 01:21 PM, David Duffett
rt the external voltage supply device.
> I was going to point you to the Xorcom Astribank, which I know supplies the
> required voltage.
>
Ah! Ok! I was thinking you was giving me something like a temporary solution
for supply the voltage to the PBX.
Cheers,
Roberto Fichera.
> All the b
ity takes or not the voltage from the NT-1 and
it does! So the problem looks really this!
Can you point me to how supply voltage to the PBX while I'm waiting the given
OpenVox adapter being delivered?
Cheers,
Roberto Fichera.
>
> All the best,
>
>
>
> On Sunday, August
ply it, but I can point you to a
> solution if required.
The OpenVOX B400M can supply external voltage via dedicated connector.
Cheers,
Roberto Fichera
>
> All the best,
>
>
>
> On Sunday, August 3, 2014, Mc GRATH Ricardo <mailto:mcgra...@mail2web.com>> wrote:
>
&
e OpenVOX card I can
see lots of errors in the log.
Regarding the LED things I'll check tomorrow morning.
Cheers,
Roberto Fichera
> Good luck
>
> Mc GRATH Ricardo
> E-Mail mcgra...@mail2web.com
--
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-- Band
Il 01/08/2014 21.13, Richard Mudgett ha scritto:
>
>
>
> On Fri, Aug 1, 2014 at 1:26 PM, <mailto:ker...@tekno-soft.it>> wrote:
>
> On Fri, 1 Aug 2014 12:39:18 -0500, Richard Mudgett wrote:
>
> On Fri, Aug 1, 2014 at 12:03 PM, Roberto Fichera wrote:
ved
switchtype = euroisdn
signalling = bri_cpe
channel => 10-11
context=from-NT1
the /etc/asterisk/chan_dahdi.conf
has the default settings and include the dahdi-channels.conf at the end.
Does anyone know how can I solve this problem?
Thanks in advance.
Roberto Fichera.
--
__
cript to take the standard output?
> When I take the standard output, I'll do the grep to see if there is a code
> 450.
> Right?
>
> Il 22/08/12 11:56, Roberto Piola ha scritto:
>
>> no. when you issue
gument :p
>
> Il 22/08/12 07:02, Roberto Piola ha scritto:
>
>>
>> I would simply send the message with sendmail -v and then grep the output
>> for the error message
>>
>
> --
> _
> --
I would simply send the message with sendmail -v and then grep the output
for the error message
Il giorno 22/ago/2012 04:19, "Raj Mathur (राज माथुर)"
ha scritto:
> On Tuesday 21 Aug 2012, Ruben Rögels wrote:
> > Hello,
> >
> > no problem at all, I think this is the tricky part.
> >
> > A smtp dia
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> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update op
http://www.api-digital.com --
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> http://www.asterisk.org/hello
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>
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>http://lists.digium.c
In Italy, you must enable overlapdial=yes
On Thu, Feb 3, 2011 at 7:45 PM, Cassius Smith wrote:
> Hello,
> I have an installation in Austria; ISDN service provided by Austria Telekom.
> The main number of the service is 6 digits. Incoming calls may contain 2
> additional digits, which I then use t
stion about this
issue, I'm sorry but want to ask if anyone can indicate the correct
place .
Thanks
--
_______
Roberto Linck
robertoli...@gmail.com
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>
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> Hard drive speed may differ between 5400rpm and 7200rpm.
in high performance server environment, you can get 1 and 15000rpm
drives as well... the fastest, the better (and the more expensive).
moreover, if you have 6 disks in raid 1+0, you have better write
performance than 3 disks in raid
t; ___
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>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
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I do not have examples, but if you are using the 1700 series router in order
to originate the ipsec vpn, you may use command qos pre-classify (please
search for it on cco.cisco.com)
On Thu, May 7, 2009 at 9:54 PM, Brent Davidson
wrote:
> I've got multiple satellite office all linked back to the m
Tim Panton ha scritto:
> [ ... snip .. ]
I'm interested to use it as IAX2 API within my UI, so something
like:
- open IAX2 channel
- call 123456
- answer a call
- close IAX2 channel
>>> It is definitely capable of that with an added class or
Tim Panton ha scritto:
> On 14 Jan 2009, at 18:02, Roberto Fichera wrote:
>
>
>> Tim Panton ha scritto:
>>
>>> On 14 Jan 2009, at 17:07, Roberto Fichera wrote:
>>>
>>>
>>>
>>>> Tim Panton ha scritto:
>>&
Tim Panton ha scritto:
> On 14 Jan 2009, at 17:07, Roberto Fichera wrote:
>
>
>> Tim Panton ha scritto:
>>
>>> It isn't really in a state for novices at the present
>>> you'd need:
>>> 1) a java compiler
>>> 2
Tim Panton ha scritto:
> It isn't really in a state for novices at the present
> you'd need:
> 1) a java compiler
> 2) a java code signing certificate (java applets can't read from the
> mic
> without being signed)
> 3) appropriate javascript and DHTML to implement
ursday 14 August 2008 03:09:42 pm roberto wrote:
>> I'm looking for some "free" LATA X Area Code database.
>>
>> Anyone have any idea where can i found?
>
> this site ha
Hi All,
I'm looking for some "free" LATA X Area Code database.
Anyone have any idea where can i found?
Thanks,
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Register Now: h
> PSTN,3,Background(enter-ext-of-person) ; input an extension
exten => PSTN,n,WaitExten(20) ; Adjust wait, default 5 sec
exten => PSTN,n,Goto(internal,${EXTEN},1) ; Goto the correct extension
exten => PSTN,n,Hangup() ; End the call
where PSTN is your sipura SIP name (1002 i think)
Cia
uot;SIP/1003-b5f0b840", "SIP/
1009/1442302") in new stack
second
is the line cord plugged in and working?
Because
-- Got SIP response 503 "Service Unavailable" back from
192.168.0.111
is what you get when there is no line attached/working.
Ciao
Roberto
___
by the 9 (that is the :1
after EXTEN)
Now 9 is standard in USA for outside line, in some other countries is
0, you choose
Ciao
Roberto
On May 21, 2008, at 7:52 AM, RoLaNd RoLaNd wrote:
Hello Roberto,
first of all, id like to thank you for your help with this..
secondly, i tried the conf
search. I don’t know what it does, but stuff seems to work. Help?
FXO Timer Values (sec):
..PSTN Answer Delay: 5 < Delay so that you can get the CID data.
NghtShd at http://forum.voxilla.com/linksys-sipura-voip-support-forum/starter-spa3102-asterisk-setup-18612.html
claims that 5 seconds i
Problem solved!
msmtp from the command line was using the config file ~/.msmtprc (the
one I configured) when called from asterisk msmtp uses /opt/local/etc/
msmtprc
so I copied the config in there and voila the emails worked as a champ.
Ciao
Roberto
On May 14, 2008, at 9:14 PM, Roberto
did that and still no email
is asterisk logging something somewhere about errors in saving files
or so?
nothing shows up in the /tmp directory anyway
I have verbose and debug set to 100 in the CLI but I see no error
messages
HELP!
Roberto
On May 15, 2008, at 1:29 AM, Tzafrir Cohen wrote
nt default : aspinet
Adn in voicemail.conf
add
mailcmd=/usr/bin/msmtp -t
Also you can try to configure sendmail for smtp relay with your ISP
This doc was very useful when I try it.
http://cri.ch/linux/docs/sk0009.html
Regards
2008/5/14 Tilghman Lesher <[EMAIL PROTECTED]>:
On Wednesday 14 M
does the /tmp directory need to have some specific kind of mode/
ownership?
mine is linked to /private/tmp and is lrwxr-xr-x root admin
Ciao
Roberto
On May 14, 2008, at 8:34 PM, Roberto Milani wrote:
> That's what I thought,
> and my voicemail.conf is:
>
> [general]
>
&
;max and min length of a message
maxmessage = 180
maxlogins = 3
[default]
100 => 4711,Front Desk,[EMAIL PROTECTED],,attach=yes
the voicemail works, I get also the MWI working perfectly
but no email
Roberto
On May 14, 2008, at 6:37 PM, Tilghman Lesher wrote:
> On Wednesday 14 May 20
Good hint but I tested that too
I sent the command line to the link called sendmail and I got my mail
just right
is there any other configuration in asterisk that might prevent it to
send mails?
Ciao
Roberto
On May 14, 2008, at 5:11 PM, Tilghman Lesher wrote:
> On Wednesday 14 May 2008
I do have a mail transport agent configured
It is msmtp and it is working just fine I tested it on the command
line and I receive the test email
I have a link from sendmail pointing to msmtp.
but it never get called.
Ciao
Roberto
On May 15, 2008, at 12:54 AM, gres wrote:
> i think
On May 14, 2008, at 1:18 PM, david wrote:
> david wrote:
>> Roberto Milani wrote:
>>
>>>>
>>>> Roberto - I noticed in your original email you had the lines
>>> something like
>>>>
>>>> mailcmd=/opt/local/bin/msmtp -
On May 14, 2008, at 1:00 PM, david wrote:
> Roberto Milani wrote:
>>>
>>> Roberto - I noticed in your original email you had the lines
>> something like
>>>
>>> mailcmd=/opt/local/bin/msmtp -t ; --from blah
>>> AND
>>> serverem
>
>Roberto - I noticed in your original email you had the lines
something like
>
>mailcmd=/opt/local/bin/msmtp -t ; --from blah
> AND
>serveremail=from=blah
>
>In mailcmd everything after the ; will be ignored as a comment
>In serveremail - well - it shou
ut attachment
It just does not get called from asterisk.
is there a way to debug it?
thanks
Roberto
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k don't send the emails.
any suggestions?
Thanks
Roberto
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Have anyone maided like 200 simultaneous connections to asterisk AMI
(manager). ??
How many connections can I made without problems ?
I’m using a Quad core DELL poweredge machine.
Roberto Fernandes Lopes
Diretor Presidente
Dialtech Telecom. e Sistemas Ltda.
(11) 6986-8886
No
At 19.56 28/05/2007, you wrote:
>On 5/28/07, Roberto Fichera <<mailto:[EMAIL PROTECTED]>[EMAIL PROTECTED]>
>wrote:
>At 19.19 28/05/2007, you wrote:
>
>
>>On 5/28/07, Roberto Fichera <<mailto:[EMAIL
>>PROTECTED]>[EMAIL PROTECTED]> wrote:
&g
At 19.19 28/05/2007, you wrote:
>On 5/28/07, Roberto Fichera <<mailto:[EMAIL PROTECTED]>[EMAIL PROTECTED]>
>wrote:
>At 17.09 28/05/2007, Richard Alam wrote:
>>Yes, we have some downloadable code. We are in the process of completing the
>>instructions (build/de
gt;
>
>Hoping for your feedbacks.
>
>
>
>Thanks.
>
>
>
>Blindside Project Team
>
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>asterisk-users mailing list
>To UNS
Hi all !!!
I would like to install asterisk in Xen domU using TDM400 hardware.
Somebody know a howto or tutorial about that ?
Thanks in advance
roberto
--
Ing. Roberto Pereyra
ContenidosOnline
http://www.contenidosonline.com.ar
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isk-users mailing list
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Other: sipX
http://www.sipfoundry.org/features.html
roberto
--
Ing. Roberto Pereyra
ContenidosOnline
http://www.contenidosonline.com.ar
_
sk for our call
centre solution but, since the bugtracker only grows and people still want
to stuck more and more features without solve CRITICAL and crash bugs.
Can someone answer my questions?
Roberto
-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de
[EMAIL
Hi
Which is most resistant to the loss of packages in a dirty link ? SIP or IAX ?
roberto
2006/10/27, Dave Cotton <[EMAIL PROTECTED]>:
On Thu, 2006-10-26 at 13:08 -0700, Martin Joseph wrote:
> On 2006-10-26 09:21:20 -0700, Dave Cotton <[EMAIL PROTECTED]> said:
&
dvance for the help,
Roberto.
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ard in the 2610?
Thanks in advance,
Roberto Fichera.
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2006/7/24, Steve Langstaff <[EMAIL PROTECTED]>:
I couldn't find an open source phone, but "NCH Express Talk" appears to be a
free download that supports G726-32.
Hi
I installed NCH Express but not support G726-32 like as says the
company's website.
Thanks
rober
nd all the extension
are in
the same subnet.
So, does anyone is having/had the same problem and/or could point me where
I could fix such problem?
Thanks in advance,
Roberto Fichera.
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aste
Anybody know about a softphone (open source) that support G726-32 codec ?
Thanks in advance
roberto
--
Ing. Roberto Pereyra
ContenidosOnline
Looking for Linux Virtual Private Servers ? Click here:
http://www.spry.com/hosting-affiliate/scripts/t.php?a_aid=426&a_bi
HiSomebody knows a hosted billing voip service ?roberto-- Ing. Roberto PereyraContenidosOnlineServidores BSD, Solaris y LinuxSoporte técnico ISPsJabber ID:
[EMAIL PROTECTED]For reliable and professional DNS, use DNS Made Easy!http://www.dnsmadeeasy.com/u/14989
Yes.
disallow=all
allow=g723
Allow only g723 codec.
roberto
2006/3/26, Mohammad Salaque <[EMAIL PROTECTED]>:
Hello list,Another newbie question,. if I put "disallow=all" and "allow=g723"my sip.cof does it mean that extension could only communicate usingg723 ?bel
Hi
Somebody knows a tutorial or help me for use a SPA3000 like fxo Asterisk interface ?
I would like to send and receive calls from/to my asterisk extensions from PSTN by spa3000 fxo.
Thanks in advance.
roberto-- Ing. Roberto PereyraContenidosOnlineServidores BSD, Solaris y LinuxSoporte
vekennedyuk / MSN [EMAIL PROTECTED]Euro Tech News Blog http://eurotechnews.blogspot.com___
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http://lists.digium.com/mailman/listinfo/asterisk-users-- In
Hi
I looking a good IAX service for a emerging
voip provider.
Better with a test account to try.
Thanks in advance.
roberto-- Ing. Roberto PereyraContenidosOnlineServidores BSD, Solaris y LinuxSoporte técnico ISPsJabber ID: [EMAIL PROTECTED]
For reliable and professional DNS
).
Thanks in advance.
roberto-- Ing. Roberto PereyraContenidosOnlineServidores BSD, Solaris y LinuxSoporte técnico ISPsJabber ID: [EMAIL PROTECTED]
For reliable and professional DNS, use DNS Made Easy!http://www.dnsmadeeasy.com/u/14989
Hi
I found this http://www.etntalk.com/callto/loginany/
Somebody has used it?
roberto2006/1/11, Derek Whitten <[EMAIL PROTECTED]>:
Miguel wrote:> Roberto Pereyra wrote:>>> Hi>>>> Someone knows a free web based SIP client for use with any provider ?>>>>
Hi
Someone knows a free web based SIP client for use with any provider ?
Thanks
roberto-- Ing. Roberto PereyraContenidosOnlineServidores BSD, Solaris y LinuxSoporte técnico ISPsJabber ID: [EMAIL PROTECTED]
For reliable and professional DNS, use DNS Made Easy!http
Hi,
I solved my last problem that was about receive calls.
Now I have another one, that's after a end a phone the zap extension
stay unavailable, until a restart on 1 minute.
Does anybody know what could be it?
Tkz,
Reiner
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channel Zap/31-1 to write format slin
Apr 19 20:08:21 DEBUG[4878]: Set channel SIP/6633-115a to read format slin
Apr 19 20:08:21 DEBUG[4878]: Ooh, format changed from unknown to gsm
On SJPhone shows a message with PCMA
Does anybody have more information?
Tkz,
Reiner
On 4/19/05, Roberto Reiner Uhry
Hi,
I have an Asterisk installed on a FC3 with a Digium e100p card and an
E1 (ISDN PRI).
I'm in Brazil and using Embratel as carrier.
After few troubles I get it working to make calls, from a SIP channel
to an Fone through the carrier. But when I receive a call, this one
is transfered to the SIP
I forgot!
/etc/asterisk/zapata.conf
[trunkgroups]
trunkgroup => 1,16
spanmap => 1,1,1
[channels]
switchtype=euroisdn
signalling=pri_cpe
language=us
defaultzone=us
group = 1
musiconhold = default
echocancel=yes
channel => 1-15,17-31
On 4/18/05, Roberto Reiner Uhry <[EMAIL PROTE
Hi,
I have an Asterisk installed on a FC3 with a Digium e100p card and an
E1 (ISDN PRI).
I'm in Brazil and using Embratel as carrier.
After few troubles I get it working to make calls, from a SIP channel
to an Fone through the carrier. But when I receive a call, this one
is transfered to the SI
Span: Implicit
Actual Confinfo: Num/0, Mode/0x
Actual Confmute: No
Actual Hookstate: Onhook
I don't know because Asterisk doesn't enable echo cancelation.
Roberto Vargas.
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I fear that list digest did not forward to me all the messages...
buying cell phone adapters is quite unfeasible at this point, since the
installation at hand uses 8 BRI for outgoing calls, and the customer
negotiated very special rates for handling all the traffic through his voice
carrier. Moreo
We have set up an HP DL380 with 3 4BRI cards, Fedora core 2 (kernel 2.6.10)
and asterisk (bristuff-0.2.0-RC7f with asterisk 1.0.5). 4 ports are
configured in TE mode and connected to the PSTN; the other 8 are in NT mode
and connected to isdn phones.
the other outbound calls to PSTN are fine, howev
IO-APIC-level megaraid
24: 100936929 0 0 0 IO-APIC-level t4xxp
Roberto Grasso
Roberto Grasso
Technical Account Manager
Puntocontatto s.r.l.
Via Alessandrini 9
20016 Pero
tel e fax +39 02 38101310
Cell. 333-5253086
__
A great universe to explore...
I tried some 6 months ago but there isn´t any great voice project in
portuguese (Brazilian)... and CMU release a test code to windows... sphinks
+ festival + portuguese
If you have any news about shpinx+asterisk please let us know...
Happy new year,
Mario
- Or
We got the same problems in connecting to uni.it .
the problems were solved (under indications of Mr. Cardone, from Unidata) by
using the ip address of the machine as the SIP account, and by using two
different IP addresses, on the asterisk server: one for outgoing calls and
one for incoming ones.
0880
Answering with preferred capability 0x4(ULAW)
Answering with non-codec capability 0x1(G723)
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 217.18.104.75:5060;branch=z9hG4bK2f983a20;rport
From: "Roberto Piola" ;tag=as2fb0ecc6
To:
Contact
We use asterisk coupled with swyxIt via a gnugk (version 2.0.8) and oh323
module: both swyx server and asterisk register on the gnugk. asterisk
receives sip calls from the exterior and routes them to the gk. I've set up
a prefix on swyx so that if I prepend +996 to my phone numerb, the call
gests r
you can login as root only on the console or on the lines listed in
/etc/securetty
if you want to log in remotely as root, you can either:
- log in as a regular user and then issue the "su -" command in order to
become root
- use a ssh client (secure shell) instead of telnet (well, you can disable
This evening I tried to install asterisk RC1 .rpm on a Fedora box... but how
can I re-add openh323 support? or does it contain an alternate h323 support?
thanks in advance
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also tried to enable the [Proxy] function on the gatekeeper, but the
result is the same
I tried to search the internet for the message, but I got no results
Roberto Piola, Ph.D.
Senior Network Engineer
Divisione VAIPS
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SOFTPEOPLE
We have set up an Asterisk PBX managing a EuroPRI in Italy.
We have conneccted to the asterisk PBX some Cisco IP Phones and a Panasonic
PBX with 10 analogic phones.
If we dial an unassigned telephone number we are not able to listen to
PSTN
Operator message telling that the subscriber does not ex
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