[Asterisk-Users] DISA & SPA3000 issues
Hi, These days I run into something quite odd. I have an [EMAIL PROTECTED] that was modified to meet our requirements. We have a completely funtional DISA which we use pretty much all the time. I works flawlessly with incomming SIP calls from several providers, IAX calls from FWD and with ZAP. Recently we came out with a situation where it doesn't work... with a SPA3000 PSTN Line. You can call, navigate de IVR, log in into our app, and then when you go to DISA, and DISA plays the dialtone... whatever you dial is not recognized... This was REALLY odd... so I made a network capture with Ethereal, and... the SPA actually STOPS sending the RTP Events after the second dialtone... To verify this, I created an IVR which played the dialtone, and verified that it was true no RTP DTMF events (RFC2833) are sent after the SPA listens the second dialtone. I just reviewed the 87 pages PDF of the SPA3000... and didn't find anything about such "feature". Now I am going to try to figure out if it has something to do with the tones recognition of the SPA. I the meanwhile I had to write a little DISA-like app, based on something I found on this forum, without the dialtone. Did anyone find out anything about this issue before? REGARDS!!! Alchaemist ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SIP behind NAT to pub Asterisk, best solution?
Hi, The VERY BEST solution, is AIX... SIP + NAT tends to be lousy... For softphone you can try the simplest one, DIAX, it is extremely basic, but does a good job. Now... If SIP is a must, then it really depends a lot on the NAT you have there.. the best way to do it is using qualify=yes and nat=yes in the config for your extensions, and even better if you can forward a fixed SIP and RTP port to the inner IP of the phone. You have all the info in www.voip-info.org http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions Cheers! Alchaemist "Blake Krone" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]... What is the best solution? I dont want to have modify firewall's at all or do port fowarding. Ideally I would like a solution that with either a softphone or wireless hardphone one could connect via friends, family, or hotspots without reconfiguring their devices. What are people using? STUN? SER? Thanks in advance! -blake ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: VOIP in Japan using Freebit
Have you tried: [EMAIL PROTECTED]:[EMAIL PROTECTED] [EMAIL PROTECTED]:mypass:[EMAIL PROTECTED] [EMAIL PROTECTED]:mypass:[EMAIL PROTECTED]/05075034132 ? Sometimes SIP providers require the realm in the username, so the first part should have the @blah Then, the third part, is the callerid so it shouldn;t be required, and the last part, is the extension notification or something like that, I never use it. Always include the pass. Regards! Alchaemist "Pikoro" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] > Has anyone had any experience using a VOIP provider in Japan? > > No matter what I try, my REGISTER string kicks back one of 2 errors: > Got SIP response 481 "Call/Transaction Does Not Exist" back from x.x.x.x > or > Got SIP response 400 "Bad Request" back from x.x.x.x > > My register string is as follows: > [EMAIL PROTECTED] > > I have tried the following also: > 05075034132:[EMAIL PROTECTED] > [EMAIL PROTECTED]/05075034132 > 05075034132:[EMAIL PROTECTED]/05075034132 > myuserid:[EMAIL PROTECTED] > > and variations of the above. > > Is there any other information I could provide in order to get some help? > > I guess another thing I am looking for is a list of possible registration > strings.. I'll try them all :D > > Cheers > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: Asterisk and a SPA3000behindNATpeerregistration
Hi,
Lets see... dynamic IP, means mainly two options:
1- PPPoE in the same machine as asterisk
In that case, you can get the IP locallyin the shell
2- Whatever protocol, in a router
In that case you must rely in querying your router or an external system
(like www.myipaddress.com)
Now, in the confs, you need in sip.conf a line like "#include
sip_nat.conf"
Then, in your script you would need something like this php with
wrappers enabled
And finally you need to setup the right cron with the script.
CASE 2 - External IP source
Put the right file path in $ConfFile, and make this script run in your
cron, often... the interval you setup will be the maximum "SIP trouble time"
you will have.
Just in case don't put it with an interval too low, because it might be
something nasty to do to a website like this.
Also, if you want to change it for your router webpage, then all you have
to do, is to update the URL and the regex. If it has password, all you have
to do is to use an URL like http://user:[EMAIL PROTECTED]/path/to/page
#-
#!/usr/bin/php -q
http://www.myipaddress.com');
$Matches = array();
preg_match('/.*IP address is\:\*<\/a>\s*(\d+\.\d+\.\d+\.\d+)<\/b>/i',$Webpage,$Matches);
$IP = @ $Matches[1];
if ($IP){
echo "Your IP is: $IP\n";
if ($FH = fopen($ConfFile,'w')){
fputs($FH,"externip=$IP\n");
fclose($FH);
$Reload = `asterisk -rx \"sip reload\"`;
echo "Reload Status: $Reload\n";
}
else{
echo "Unable to save file.\nSomething odd is going on, check permissions
and paths.\n";
}
}
else{
echo "Unable to detect IP.\nPerhaps you are not connected, the website is
down, or it changed its HTML.\n";
}
#-
CASE 1 - Similar to case 2, except that you get the IP from a local command,
I've seen an SH script that does this work, in this very list, somewhere.
Another difference is that instead of letting the cron regularly run it, you
can/must run it every time the os detects an IP change/interface reload,
init.d something, I cannot recall it now. Anyway case 2 will work as well.
Cheers!
Alchaemist
- Original Message -
From: "razza" <[EMAIL PROTECTED]>
Newsgroups: gmane.comp.telephony.pbx.asterisk.user
Sent: Thursday, September 22, 2005 5:24 AM
Subject: RE: Re: Asterisk and a SPA3000behindNATpeerregistration
>
> Alchaemist Wrote:
>>Now... if you have dynamic IP in the asterisk... things change
> because
>>Asterisk must know in sip.conf the external IP.
>>I think I read in this list, that the best (only?) way to get arround,
> is to
>>place a script that detects the external IP when it changes,updates
> sip.conf
>>(preferably another conf like sip_nat.conf that is included into
> sip.conf),
>>and then uses the CLI to reload the sip configuration, you have a
> command I
>>cannot recall right now to only reload that part instead of the whole
> bunch
>>of confs.
>
> The problem is no one has come back with a script which does the job :o(
> The command to reload the sip.conf is -
> asterisk -rx "sip reload"
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: POP3 and TTS (Festival?)
Hi, I plan, if I have time, to work on it this weekend, so in that case I will post the code, no problem at all. I want to add reply with record functionality first. Regards! Alchaemist "Michiel van Baak" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] > On 17:34, Wed 21 Sep 05, Alchaemist wrote: >> Hi, >> >> Has anybody seen a non commercial, or freeware, or GPL, or >> even >> CHEAP... POP/IMAP to Text-to-speech? >> >> I have a working version for POP3 using festival. It DOES >> work... it even cleans the email contents to get the actual content. It >> works great with Outlook emails and similar, and skips non >> multipart/alternative (that would be mainly SPAM, where the email is just >> html or multipart without the text/plain alternative). >> >> BUT, festival does really sound BAD when reading... >> >> AND... IMAP would be preferable, because you can mark emails >> as >> read, instead I have to go over all the emails I have. So if I can get >> past >> this without further coding... would be great. >> >> I tested with Cepstral.. it seems quite good... perhaps >> somebody >> has another option, preferable as cheap as cepstral... >> >> In, short... any experiences with this? > > Hi, > > Care to share the code you already have for POP3 using > festival ? If so I can help to port it to IMAP (I'm using > IMAP here for mail). > > Greetz > > -- > Michiel van Baak > http://michiel.vanbaak.info > [EMAIL PROTECTED] > GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D > > "Why is it drug addicts and computer afficionados are both called users?" > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Get SIP to work over very limited network access
Hi, Some way of VPN seems to be the only solution. But, you should try something really silly first. Try to setup your asterisk to listen in one of the open ports (ie 21, 22) with SIP you will require two connections, thus two open ports, instead with IAX2, one port will do it, (You can try DIAX, its nice, and really simple). There is a chance, (there always is), that your friend's ISP is allowing all traffic over port 21 and 22, TCP AND UDP see? Perhaps even before setting asterisk, you should open an UDP listener, and try to send traffic to see if it passes through his ISP. Otherwise all you have is VPN or SSH port forwarding. Regards! Alchaemist PS: Asterisk over TCP? no way...not supported yet. "Benjamin Lawetz" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] > I've got a friend who's spending 6 months on the other side of the world. > So > before he left I configured him a softphone on his laptop to connect to my > asterisk so he can call home free of charge. > > Unfortunately, he just found out he has horrible internet connection. > Bandwith and latency is ok, the problem is the stop almost all > connections. > He has to connect to a proxy server for his web access. The only thing > that > seems to go through straight away is ssh or ftp. Thought about doing a ssh > tunnel, but that doesn't work over udp. He has linux and windows on his > machine, and is full configurable. The server (and my net connection) is > under my control and fully configurable. > > So does anyone have any idea how to get him to connect to my asterisk > server > through a TCP connection ? > > thanks > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk and a SPA3000 behind NATpeerregistration
Hi Lan, SPA 3000 <> NAT <-> Internet < NAT> Asterisk That is two NATs... so, as it is, it will NEVER work, so you have only one way to go. This is the recipe: 1- Asterisk side, MUST have SIP/RTP ports forwarded in your router 2- RTP ports must be a fixed range in rtp.conf [general] ; RTP start and RTP end configure start and end addresses rtpstart=5061 (whatever range you want to add here...) rtpend=5099 3- Asterisk must be told that is using NAT in sip.conf nat=yes externip=123.123.123.123 (your external IP address) localnet=192.168.0.1/255.255.255.0 (your local IP range) 4- The extensions must be configured for NAT qualify=yes nat=yes 5- Configure a fixed port range for the RTP in the Sipura 6- Forward the SIP ports (Line 1 & PSTN Line) AND the RTP portson the router That way should work flawlessly. If it is too much bother to forward ports or ranges of ports, and you don't care for security, particullarly in the asterisk, then configure the DMZ of both routers to their respective device IP (Asterisk and SPA). Without Forwarding or DMZ you will never be able to do it... Now... if you have dynamic IP in the asterisk... things change because Asterisk must know in sip.conf the external IP. I think I read in this list, that the best (only?) way to get arround, is to place a script that detects the external IP when it changes,updates sip.conf (preferably another conf like sip_nat.conf that is included into sip.conf), and then uses the CLI to reload the sip configuration, you have a command I cannot recall right now to only reload that part instead of the whole bunch of confs. Cheers! Alchaemist "Maps" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] > Dear Richard and supporters! > I see that you guys could be able to setup the SPA 3000 to connect to the > asterisk thru the NAT. I don't know how would to do this. As my > understand > is that the SPA 3000 is just able to configure with the SIP that not NAT > aware in Asterisk. > I am trying to configure the SPA 3000 <> NAT <-> Internet < NAT> > Asterisk, but I am unsuccess to do that. > Also I have the SPA2100 and try to do the same thing. > > If you guys could be able to do so. Would you PLEASE tell me how could I > do > that? or please direct me to the information where I can find out the way > to setup that up! I am very appreciated > > > Thanks in advance! > > Lan > > >> >> > Hi, >> > >> > I have a little situation here :( Perhaps somebody can give me >> > a >> > hand with it. >> > >> > I have an Asterisk working, and in another office, a Sipura >> > SPA-3000. >> > I configured the SPA and I have the extension working, the > incomming >> > trunk working, but the outgoing trunk (peer) does not work. >> > >> > The issue is that I have a dynamic IP where the SPA is, and > neither >> > the SPA nor my router have DynamicDNS. >> > So, if I manually change the host for the peer for the SPA PSTN >> > Line, then everything works fine, and I can make local calls through >> > it. >> > BUT when my router's IP changes... I am lost. (EVEN while the >> > extension SPAN Line 1, or the incomming trunk are registered and DO > work.) >> > >> > According to what I read, if I setup the outgoing trunk to > type=peer >> > with host=dynamic, then I can make the peer to register itself into the >> > Asterisk, so the asterisk will know where to contact the peer. >> > But I cannot figure out how to make the SPA register as the >> > peer? I can make it register as a type=friend for the incomming >> > traffic, > but >> > not as a peer.. >> >> Don't bother with the peer. >> >> Define both the Line1 and PSTN ports on the SPA to register with asterisk >> using different UserID/Passwords for each, use port 5060 for one and 5061 >> for the other, and in asterisk's sip.conf file define them as >> type=friend. >> >> It does work just fine that way. >> >> >> ___ >> --Bandwidth and Colocation sponsored by Easynews.com -- >> >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > A
[Asterisk-Users] Re: How can i call to a cellphone here in Mexico?
Hi again...
How are you
dialling this?
90446612345678 ? or 0446612345678 ?
Also
another possibility is that the card is sending the DTMF when it haven't yet get
the tone from your PSTN? just thinking... in that case you can use the 'w' in
the dialstring to get a wait delay of 0.5 secs. (I don't think it might be this
one because your problem is speciffic to cells, but just thinking
aloud)
Good
luck!
Alchaemist
"Claudio Canseco" <[EMAIL PROTECTED]>
wrote in message news:[EMAIL PROTECTED]...
Hi, thanks for your replay Alex:
Right now a have an Asterisk server on a Dell Optiplex GX110 (PIII
666MHz, 320 RAM) with no soundcard.
With an X100P clone card (an ambient modem).
Everything looks good, I've been able to make local calls trough PSTN,
IAX, SIP.
I only have 1 POTS line, and 4 SIP softphones (X-lite) running all
right.
The only problem so far I have noticed (or realized of :P), it is
that i can make calls
to cellularphone numbers, * tries to connect but i get redirected to the
emergency service number 066.
I don't think it is because of my dialplan, eventhough I tried several
configurations. Anyways here is part of the dialplan
where my softphones make calls:
;;# Llamadas
salientes [outgoing]
#;
[outgoing]include => toPSTNinclude => iaxtelinclude
=> fwd-iax
; -> PSTN
[toPSTN]
; Permite hacer llamadas locales (7-digitos sin contar 9)ignorepat =>
9
exten => _92XX,1,NoOp("Call for "${EXTEN:1})exten =>
_92XX,2,Dial(Zap/1/${EXTEN:1})
exten => _904466,1,NoOp("Call for "(${EXTEN:1})
;Llamadas a Celularexten =>
_904466,2,Dial(Zap/1/ww${EXTEN:1})
; -> IAXTEL
[iaxtel]exten => _1700XXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED] )exten =>
_1888NXX,1,Dial( IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED] )exten
=> _1877NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED] )exten =>
_1866NXX,1,Dial( IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED] )exten =>
_1800NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED] )
; -> FWD
[fwd-iax]exten => _3.,1,SetCallerId,${FWDCIDNAME}exten =>
_3.,2,Dial(
IAX2/${FWDNUMBER}:[EMAIL PROTECTED]/${EXTEN:1},60,tr)
exten => _3.,3,Congestion
;#;# Softphone
x-lite
#;#
[x-lite] ; Note: SIP extensions are defined here as "66" followed
by any two digitsinclude => defaultinclude =>
serviciosinclude => outgoing
exten => 6600,1,NoOp(Llamada saliente maneja IAX2)exten =>
6600,2,Macro(dial,kano00,IAX2/kano00,20,tr)
exten => _X,1,NoOp(Llamada saliente maneja SIP)exten =>
_X,2,Macro(dial,667${EXTEN},SIP/667${EXTEN},20,tr)
All softphones working are SIP, and are directed to the [x-lite]
context.
This is my zapata.conf:
[channels]language=escontext=incomingsignalling=fxs_ks.usecallerid=yeshidecallerid=nocallwaiting=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescancallforward=yescallreturn=yes
echocancel=yesechocancelwhenbridged=yesechotraining=800rxgain=0.0txgain=25.0
group=1pickupgroup=1immediate=yesmusiconhold=defaultrelaxdtmf=yes
; Relajar el DTMF, poner si asterisk salta o duplica algún DTMF,
; dando lugar a un número incorrecto.channel =>
1
And my simple configuration file,
zaptel.conf:
loadzone=mxdefaultzone=mxfxsks=1
As you can see this aren't complicated configurations because i only have 1
X100P card, and I am currently using little extensions.Also, I am not
using AMP but I'm thinking to installing it over my current
installation. I installed asterisk and zaptel from instructions i
got from several documentations sites (voip-info wiki, digium, etc).
Well, I hope this info can help to look down the problem. Thanks
again,
Regards,
Claudio
___--Bandwidth and
Colocation sponsored by Easynews.com --Asterisk-Users mailing
listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo
UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: How can i call to a cellphone here in Mexico?
Hi Claudio (Hola)
The reason is
surely that you have a conflict with the prefix commonly used in mexico for cell
phones (044)
You will
have to review all your extensions.conf and related files, to make sure the
calls are routed correctly.
Regards!
Alchaemist
"Claudio Canseco" <[EMAIL PROTECTED]>
wrote in message news:[EMAIL PROTECTED]...
Hi, thanks for your replay Alex:
Right now a have an Asterisk server on a Dell Optiplex GX110 (PIII
666MHz, 320 RAM) with no soundcard.
With an X100P clone card (an ambient modem).
Everything looks good, I've been able to make local calls trough PSTN,
IAX, SIP.
I only have 1 POTS line, and 4 SIP softphones (X-lite) running all
right.
The only problem so far I have noticed (or realized of :P), it is
that i can make calls
to cellularphone numbers, * tries to connect but i get redirected to the
emergency service number 066.
I don't think it is because of my dialplan, eventhough I tried several
configurations. Anyways here is part of the dialplan
where my softphones make calls:
;;# Llamadas
salientes [outgoing]
#;
[outgoing]include => toPSTNinclude => iaxtelinclude
=> fwd-iax
; -> PSTN
[toPSTN]
; Permite hacer llamadas locales (7-digitos sin contar 9)ignorepat =>
9
exten => _92XX,1,NoOp("Call for "${EXTEN:1})exten =>
_92XX,2,Dial(Zap/1/${EXTEN:1})
exten => _904466,1,NoOp("Call for "(${EXTEN:1})
;Llamadas a Celularexten =>
_904466,2,Dial(Zap/1/ww${EXTEN:1})
; -> IAXTEL
[iaxtel]exten => _1700XXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED] )exten =>
_1888NXX,1,Dial( IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED] )exten
=> _1877NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED] )exten =>
_1866NXX,1,Dial( IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED] )exten =>
_1800NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED] )
; -> FWD
[fwd-iax]exten => _3.,1,SetCallerId,${FWDCIDNAME}exten =>
_3.,2,Dial(
IAX2/${FWDNUMBER}:[EMAIL PROTECTED]/${EXTEN:1},60,tr)
exten => _3.,3,Congestion
;#;# Softphone
x-lite
#;#
[x-lite] ; Note: SIP extensions are defined here as "66" followed
by any two digitsinclude => defaultinclude =>
serviciosinclude => outgoing
exten => 6600,1,NoOp(Llamada saliente maneja IAX2)exten =>
6600,2,Macro(dial,kano00,IAX2/kano00,20,tr)
exten => _X,1,NoOp(Llamada saliente maneja SIP)exten =>
_X,2,Macro(dial,667${EXTEN},SIP/667${EXTEN},20,tr)
All softphones working are SIP, and are directed to the [x-lite]
context.
This is my zapata.conf:
[channels]language=escontext=incomingsignalling=fxs_ks.usecallerid=yeshidecallerid=nocallwaiting=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescancallforward=yescallreturn=yes
echocancel=yesechocancelwhenbridged=yesechotraining=800rxgain=0.0txgain=25.0
group=1pickupgroup=1immediate=yesmusiconhold=defaultrelaxdtmf=yes
; Relajar el DTMF, poner si asterisk salta o duplica algún DTMF,
; dando lugar a un número incorrecto.channel =>
1
And my simple configuration file,
zaptel.conf:
loadzone=mxdefaultzone=mxfxsks=1
As you can see this aren't complicated configurations because i only have 1
X100P card, and I am currently using little extensions.Also, I am not
using AMP but I'm thinking to installing it over my current
installation. I installed asterisk and zaptel from instructions i
got from several documentations sites (voip-info wiki, digium, etc).
Well, I hope this info can help to look down the problem. Thanks
again,
Regards,
Claudio
___--Bandwidth and
Colocation sponsored by Easynews.com --Asterisk-Users mailing
listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo
UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] POP3 and TTS (Festival?)
Hi, Has anybody seen a non commercial, or freeware, or GPL, or even CHEAP... POP/IMAP to Text-to-speech? I have a working version for POP3 using festival. It DOES work... it even cleans the email contents to get the actual content. It works great with Outlook emails and similar, and skips non multipart/alternative (that would be mainly SPAM, where the email is just html or multipart without the text/plain alternative). BUT, festival does really sound BAD when reading... AND... IMAP would be preferable, because you can mark emails as read, instead I have to go over all the emails I have. So if I can get past this without further coding... would be great. I tested with Cepstral.. it seems quite good... perhaps somebody has another option, preferable as cheap as cepstral... In, short... any experiences with this? Cheers! Alchaemist ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and a SPA3000 behind NAT peer registration
Hi,
I have a little situation here :( Perhaps somebody can give me a
hand with it.
I have an Asterisk working, and in another office, a Sipura
SPA-3000.
I configured the SPA and I have the extension working, the incomming
trunk working, but the outgoing trunk (peer) does not work.
The issue is that I have a dynamic IP where the SPA is, and neither
the SPA nor my router have DynamicDNS.
So, if I manually change the host for the peer for the SPA PSTN
Line, then everything works fine, and I can make local calls through it.
BUT when my router's IP changes... I am lost. (EVEN while the
extension SPAN Line 1, or the incomming trunk are registered and DO work.)
According to what I read, if I setup the outgoing trunk to type=peer
with host=dynamic, then I can make the peer to register itself into the
Asterisk, so the asterisk will know where to contact the peer.
But I cannot figure out how to make the SPA register as the
peer? I can make it register as a type=friend for the incomming traffic, but
not as a peer..
The Line 1 part of the SPA is registering and working neatly.
BTW I have [EMAIL PROTECTED]
Any idea, suggestion, or whatever you can think about it?
Thanks!!!
Alchaemist
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: how to distinguish the "ringing" and"connected"for zap channel
This might be of help as well? http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+Disconnect+Supervision&diff=3 "Liu Peter" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] 1) how to config callprogress=yes ? in extensions.conf? could you give me an example? 2) you means record the call (via zaptel) into a file and analyze it with audio tool? thanks.. 2005/9/20, Alchaemist <[EMAIL PROTECTED]>: > Hi there, > >Basically, youare supposed to play arround with indications.conf >To have the extensions configured with callprogress=yes but, be > carefull because it is quite experimental. >Also, what I did was to get an audio program (Cooledit, Adobe > audition, or other), and you should use the spectral view (FFT Fast > Fourier > Transform), there you will be ableto see which frequencies the tones have, > and their duration. > >Now, having said that I was half successfull in making it work, > and I still have some problems, so if anybody else has a clear idea of > what > can be done, please shout in here ! :) > > >Give a look to this: > http://www.voip-info.org/tiki-index.php?page=Asterisk+config+indications.conf >Particullaryl the section: Generating a Tone Set > >And this: > http://www.speech.kth.se/wavesurfer/ > >Cheers! > Alchaemist > > > "Liu Peter" <[EMAIL PROTECTED]> wrote in message > news:[EMAIL PROTECTED] > I have a TDM card in a asterisk machine. > I found that once I used it to call out, the call status changed to > "connected" even the callee is still ring. > How could asterisk distinguish the "ringing" and "connected" in zap > channel? > > thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: how to distinguish the "ringing" and"connected"for zap channel
Hi Peter 1) how to config callprogress=yes ? in extensions.conf? could you give me an example? Not in extensions.conf, but in zapata. http://www.voip-info.org/wiki-Asterisk+config+zapata.conf In my case, busydetect=yes and busycount=5 were the key to getting it right. 2) you means record the call (via zaptel) into a file and analyze it with audio tool? In my case it was simpler. I had trouble with detecting the busy tone. So when I called the voicemail, it didn't detect the hangup, thus it kept recording the line (with the tone) until the tone stopped or the VM got tired of waiting. Then I received the WAV in my email... so in short I did not need to record it. Now, if you are detecting the busy tone, right, then you can record either by zaptel (I really don't know how) or an easier way, through the sound card of the machine you are calling from with a iax/sip softphone. Usually it is as easy as selecting the sound card record mode, and selecting the mixer channel. Now.. indications.conf, seems not only to define how to generate the tones, but also how to interpret them. At least, mine only worked once I changed those values. REGARDS! Alchaemist "Liu Peter" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] 1) how to config callprogress=yes ? in extensions.conf? could you give me an example? 2) you means record the call (via zaptel) into a file and analyze it with audio tool? thanks.. 2005/9/20, Alchaemist <[EMAIL PROTECTED]>: > Hi there, > >Basically, youare supposed to play arround with indications.conf >To have the extensions configured with callprogress=yes but, be > carefull because it is quite experimental. >Also, what I did was to get an audio program (Cooledit, Adobe > audition, or other), and you should use the spectral view (FFT Fast > Fourier > Transform), there you will be ableto see which frequencies the tones have, > and their duration. > >Now, having said that I was half successfull in making it work, > and I still have some problems, so if anybody else has a clear idea of > what > can be done, please shout in here ! :) > > >Give a look to this: > http://www.voip-info.org/tiki-index.php?page=Asterisk+config+indications.conf >Particullaryl the section: Generating a Tone Set > >And this: > http://www.speech.kth.se/wavesurfer/ > >Cheers! > Alchaemist > > > "Liu Peter" <[EMAIL PROTECTED]> wrote in message > news:[EMAIL PROTECTED] > I have a TDM card in a asterisk machine. > I found that once I used it to call out, the call status changed to > "connected" even the callee is still ring. > How could asterisk distinguish the "ringing" and "connected" in zap > channel? > > thanks. > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: how to distinguish the "ringing" and "connected"for zap channel
Hi there, Basically, youare supposed to play arround with indications.conf To have the extensions configured with callprogress=yes but, be carefull because it is quite experimental. Also, what I did was to get an audio program (Cooledit, Adobe audition, or other), and you should use the spectral view (FFT Fast Fourier Transform), there you will be ableto see which frequencies the tones have, and their duration. Now, having said that I was half successfull in making it work, and I still have some problems, so if anybody else has a clear idea of what can be done, please shout in here ! :) Give a look to this: http://www.voip-info.org/tiki-index.php?page=Asterisk+config+indications.conf Particullaryl the section: Generating a Tone Set And this: http://www.speech.kth.se/wavesurfer/ Cheers! Alchaemist "Liu Peter" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] I have a TDM card in a asterisk machine. I found that once I used it to call out, the call status changed to "connected" even the callee is still ring. How could asterisk distinguish the "ringing" and "connected" in zap channel? thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
