[Asterisk-Users] Re: Asterisk using a h323 gateway
Shimul Kanti Barua
shimul at stitel.com
Wed Sep 17 03:58:28 MST 2003
----- Original Message -----
From: <asterisk-users-request at lists.digium.com>
To: <asterisk-users at lists.digium.com>
Sent: Saturday, September 13, 2003 7:55 PM
Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs
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> Today's Topics:
>
> 1. Re: Caller ID Problems (WipeOut .)
> 2. Re: IAX, IAX2 and authenticatyion (Dan)
> 3. RE: 7206 as SIP->PSTN Gateway? (Abdul Hakeem)
> 4. Re: IAX, IAX2 and authenticatyion (Brancaleoni Matteo)
> 5. Re: Dect Phone (Tjardick van der Kraan)
> 6. Monitoring an active channel (Timothy Soos)
> 7. Re: asterisk and defunct perl procs (Rich Adamson)
> 8. Re: Caller ID Problems (Rich Adamson)
> 9. UK Suppliers (Angel Gabriel)
> 10. RE: UK Suppliers (Lee Redmayne)
> 11. How to test * ? (Angel Gabriel)
> 12. Re: IAX, IAX2 and authenticatyion (dtoma at fx.ro)
> 13. Re: UK Suppliers (YO Internet Information)
> 14. Re: asterisk and defunct perl procs (Angel Gabriel)
> 15. Re: asterisk and defunct perl procs (Rich Adamson)
> 16. Re: Asterisk using a h323 gateway (Michael Manousos)
>
> --__--__--
>
> Message: 1
> From: "WipeOut ." <wipeout at linuxmail.org>
> To: asterisk-users at lists.digium.com
> Date: Sat, 13 Sep 2003 06:41:43 +0000
> Subject: Re: [Asterisk-Users] Caller ID Problems
> Reply-To: asterisk-users at lists.digium.com
>
> There are two things I can think of..
>
> 1. You are not paying for CallerID support from your telco on that line..
Its is not always a standard feature..
>
> 2. The CallerID that your telco provides is not compatible with the digium
card and Asterisk..
>
>
>
> > Dear Asterisk User,
> >
> > I am trying to use a Digium FXO Card to get the callerid but fail.
> >
> > Asterisk version: Asterisk CVS-09/03/03-11:15:03
> >
> > In my zapata.conf
> > usecallerid=yes
> > hidecallerid=no
> > callwaitingcallerid=yes
> > rxgain=3.0
> > txgain=3.0
> > ;callprogress=yes
> >
> > When I use my mobile (my mobile will show callerid) dial a call to the
system Zap/1-1 channel. Then I use "show channel zap/1-1" The callerid field
show "Caller ID: (N/A)"
> >
> > Please help ... Anywhere I can check and anywhere I done wrong?
> >
> > Thanks,
> Randal
> --
> ______________________________________________
> http://www.linuxmail.org/
> Now with e-mail forwarding for only US$5.95/yr
>
> Powered by Outblaze
>
> --__--__--
>
> Message: 2
> From: "Dan" <dtoma at fx.ro>
> To: <asterisk-users at lists.digium.com>
> Subject: Re: [Asterisk-Users] IAX, IAX2 and authenticatyion
> Date: Sat, 13 Sep 2003 09:49:13 +0300
> Organization: Personal Use
> Reply-To: asterisk-users at lists.digium.com
>
> Hi Martin,
>
> ----- Original Message -----
> From: "Martin Pycko" <martinp at digium.com>
> To: "Asterisk Users" <asterisk-users at lists.digium.com>
> Sent: Friday, September 12, 2003 11:11 PM
> Subject: Re: [Asterisk-Users] IAX, IAX2 and authenticatyion
>
>
> > IAX2 uses 4569 UDP port.
>
> How this port can be changed? There is no iax2.conf file...
>
> Dan
>
>
> --__--__--
>
> Message: 3
> From: "Abdul Hakeem" <alhakeem at blueyonder.co.uk>
> To: <asterisk-users at lists.digium.com>
> Subject: RE: [Asterisk-Users] 7206 as SIP->PSTN Gateway?
> Date: Sat, 13 Sep 2003 08:21:40 +0100
> Reply-To: asterisk-users at lists.digium.com
>
> Hi,
> You need the PA-VFC-2TE1+ cards. It supports 60 calls for codecs such as
> G723 and 120 calls for G729a and b(with the addition of a PA-MCX card).
>
> Cheers,
> Abdul
>
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Michael Kane
> Sent: 12 September 2003 18:30
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] 7206 as SIP->PSTN Gateway?
>
>
> Also, don't limit yourself to Cisco. There are many vendors out there
> that make SIP trunking gateways...
>
>
> ----- Original Message -----
> From: "David C. Troy" <dave at toad.net>
> To: <asterisk-users at lists.digium.com>
> Sent: Friday, September 12, 2003 1:24 PM
> Subject: [Asterisk-Users] 7206 as SIP->PSTN Gateway?
>
>
> >
> > All,
> >
> > I know you can use, say, a 2620 w/2 port FXO card as a SIP gateway.
> > Clearly you can use the 5300, 5800, and MGX8850 too. Does anyone know
>
> > which cards, if any, exist for a 7206VXR to act in a similar capacity,
>
> > either as a T1/PRI, DS3, or POTS FXO/FXS?
> >
> > What other Cisco routers can act as SIP gateways today?
> >
> > Thanks,
> > Dave
> >
> > =====================================================================
> > David C. Troy [dave at toad.net] 410-384-2500 Sales
> > ToadNet - Want to go fast? 410-544-1329 FAX
> > 570 Ritchie Highway, Severna Park, MD 21146-2925 www.toad.net
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --__--__--
>
> Message: 4
> Subject: Re: [Asterisk-Users] IAX, IAX2 and authenticatyion
> From: Brancaleoni Matteo <mbrancaleoni at espia.it>
> To: asterisk-users at lists.digium.com
> Organization: Espia - Emmegi Srl
> Date: Sat, 13 Sep 2003 09:52:34 +0200
> Reply-To: asterisk-users at lists.digium.com
>
> hi.
> actualy the iax2 conf file is the same of iax .
> iax2 port is hardcoded in channels/iax2.h, line 72 (more or less)
> You can change it & recompile.
>
> matteo.
>
> Il sab, 2003-09-13 alle 08:49, Dan ha scritto:
> > Hi Martin,
> >
> > ----- Original Message -----
> > From: "Martin Pycko" <martinp at digium.com>
> > To: "Asterisk Users" <asterisk-users at lists.digium.com>
> > Sent: Friday, September 12, 2003 11:11 PM
> > Subject: Re: [Asterisk-Users] IAX, IAX2 and authenticatyion
> >
> >
> > > IAX2 uses 4569 UDP port.
> >
> > How this port can be changed? There is no iax2.conf file...
> >
> > Dan
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> --
> Brancaleoni Matteo <mbrancaleoni at espia.it>
> Espia - Emmegi Srl
>
>
> --__--__--
>
> Message: 5
> From: "Tjardick van der Kraan" <tjardick at vanderkraan.net>
> To: <asterisk-users at lists.digium.com>
> Subject: Re: [Asterisk-Users] Dect Phone
> Date: Sat, 13 Sep 2003 10:47:29 +0200
> Reply-To: asterisk-users at lists.digium.com
>
>
> ----- Original Message -----
> From: "Robert Boardman" <robb at boardman.me.uk>
> To: <asterisk-users at lists.digium.com>
> Sent: Friday, September 12, 2003 10:26 PM
> Subject: [Asterisk-Users] Dect Phone
>
>
> > Hi
> >
> > I have a problem with a new DECT phone I have bought
> >
> > The key pad works like a Mobile phone where you dial first then pick up
> > the line, but it seems to dail too fast or spuriously, ie 012826736464
> > show on thew Asterisk console as 0012282677, could any one offer advice
> > how to fix?
>
> Have you tried hitting dial before typing the numbers ? My dect does give
me
> the dialtone then.
> (allthough i don't have the problem that digits go to quick, but maybe you
> can tweak that in an advanced menu setting on the phone).
>
> Tj
>
>
> --__--__--
>
> Message: 6
> From: Timothy Soos <XQL at americanisp.net>
> Organization: XQL, LLC
> To: asterisk-users at lists.digium.com
> Date: Sat, 13 Sep 2003 05:13:54 -0600
> Subject: [Asterisk-Users] Monitoring an active channel
> Reply-To: asterisk-users at lists.digium.com
>
> Hello All,
>
> I am still having some difficulty working to monitor an already active
> channel. I did some experimenting with the Monitor application without
> achieving my desired results.
>
> Here are the relevant parts of my extensions.conf file:
> [CustomerSide]
> exten => 2,1,StopMonitor
> exten => 3,1,Monitor(wav,Test_Recording_1)
>
> This is what happens:
> 1. From the console, I dial to the phone connected to the TDM400P card:
> *CLI> dial 1234 at CustomerSide
> and answer the phone when it rings.
> 2. Next, I dial from the console to activate monitoring:
> *CLI> dial 3 at CustomerSide
> Unfortunately, the monitoring does not start, and I hear Asterisk sending
3
> DTMF tones to the phone.
>
> What am I doing wrong that prevents the monitoring from starting?
>
> Is it required to start the monitoring from another phone (hard or soft
> phone)?
> --
> Thanks,
> Tim
>
> --__--__--
>
> Message: 7
> Date: Sat, 13 Sep 2003 06:42:03 -0600
> From: Rich Adamson <radamson at routers.com>
> Subject: Re: [Asterisk-Users] asterisk and defunct perl procs
> To: asterisk-users at lists.digium.com
> Reply-To: asterisk-users at lists.digium.com
>
> FWIW, I just immplemented * on a RH9 box using the CVS without any
problems
> whatsoever. The RH9 box was built from CD's as a workstation (with
everything
> installed), up2date ran to bring it reasonably current, etc. I had
installed
> "ser" a few weeks ago and it worked properly as well. Ser was shutdown
(still
> remains installed) and * is running now.
>
> I did not have to export anything or do anything special with the system
other
> then to ensure the running kernel and its "matching" source code was
installed.
> That was required due to the Digium X100P card installation needs,
otherwise
> * installed and ran correctly the first time.
>
> ------------------------
> > Yes, this is RH9. Thank you for the info.
> >
> > On Fri, Sep 12, 2003 at 02:59:46PM -0700, Scott Stingel wrote:
> > > If you're running RedHat 9, there is a known problem.
> > >
> > > Try executing the following line in the shell before starting
asterisk:
> > >
> > > export LD_ASSUME_KERNEL=2.4.1
> > >
> > > Hope this works!
> > >
> > > -Scott
> > >
> > > Scott M. Stingel
> > <snip>
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
> ---------------End of Original Message-----------------
>
>
>
> --__--__--
>
> Message: 8
> Date: Sat, 13 Sep 2003 06:56:51 -0600
> From: Rich Adamson <radamson at routers.com>
> Subject: Re: [Asterisk-Users] Caller ID Problems
> To: asterisk-users at lists.digium.com
> Reply-To: asterisk-users at lists.digium.com
>
> I'm having some of the same issues and it seems to be related to
transmission
> levels. CallerID worked fine prior to me messing with rxgain/txgain, but
I've
> not gone back to verify what I did to muck it up as yet.
>
> ------------------------
> > There are two things I can think of..
> >
> > 1. You are not paying for CallerID support from your telco on that
line.. Its is not always a
> standard feature..
> >
> > 2. The CallerID that your telco provides is not compatible with the
digium card and Asterisk..
> >
> >
> >
> > > Dear Asterisk User,
> > >
> > > I am trying to use a Digium FXO Card to get the callerid but fail.
> > >
> > > Asterisk version: Asterisk CVS-09/03/03-11:15:03
> > >
> > > In my zapata.conf
> > > usecallerid=yes
> > > hidecallerid=no
> > > callwaitingcallerid=yes
> > > rxgain=3.0
> > > txgain=3.0
> > > ;callprogress=yes
> > >
> > > When I use my mobile (my mobile will show callerid) dial a call to the
system Zap/1-1 channel.
> Then I use "show channel zap/1-1" The callerid field show "Caller ID:
(N/A)"
> > >
> > > Please help ... Anywhere I can check and anywhere I done wrong?
>
>
>
> --__--__--
>
> Message: 9
> From: Angel Gabriel <badmangabriel at lycos.co.uk>
> To: * Users <asterisk-users at lists.digium.com>
> Date: 13 Sep 2003 13:01:32 +0100
> Subject: [Asterisk-Users] UK Suppliers
> Reply-To: asterisk-users at lists.digium.com
>
> Can anyone please direct me to UK based suppliers of equipment. Website
> URL's would be appreciated. TIA
> --
> *****
> Not everyone is touched by an Angel....
> .... Those that are, never forget the experience
> *****
>
>
> --__--__--
>
> Message: 10
> From: "Lee Redmayne" <lee.redmayne at nwva.org>
> To: <asterisk-users at lists.digium.com>
> Subject: RE: [Asterisk-Users] UK Suppliers
> Date: Sat, 13 Sep 2003 13:11:52 +0100
> Reply-To: asterisk-users at lists.digium.com
>
> I bought some Snom phones which work nicely with Asterisk from:
>
> ProVu Communications Ltd
> Bank House
> Marsden
> Huddersfield
> HD7 6BR
>
> 01484-840048
> info at provu.co.uk
> www.provu.co.uk
>
> -----Original Message-----
> From: Angel Gabriel
> Sent: 13 September 2003 13:02
> To: * Users
> Subject: [Asterisk-Users] UK Suppliers
>
> Can anyone please direct me to UK based suppliers of equipment. Website
> URL's would be appreciated. TIA
> --
> *****
> Not everyone is touched by an Angel....
> .... Those that are, never forget the experience
> *****
>
>
> --__--__--
>
> Message: 11
> From: Angel Gabriel <badmangabriel at lycos.co.uk>
> To: * Users <asterisk-users at lists.digium.com>
> Date: 13 Sep 2003 13:22:11 +0100
> Subject: [Asterisk-Users] How to test * ?
> Reply-To: asterisk-users at lists.digium.com
>
> I was wondering, can I test * using just a modem card? I was want to
> check ome of the features, before I go and buy some cards. (Thanks for
> th elink to the reseller page, you know who you are!)
> --
> *****
> Not everyone is touched by an Angel....
> .... Those that are, never forget the experience
> *****
>
>
> --__--__--
>
> Message: 12
> Date: Sat, 13 Sep 2003 15:27:31 +0300
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] IAX, IAX2 and authenticatyion
> From: dtoma at fx.ro
> Reply-To: asterisk-users at lists.digium.com
>
> At Sat, 13 Sep 2003 09:52:34 +0200 , asterisk-users at lists.digium.com
wrote:
>
> >hi.
> >actualy the iax2 conf file is the same of iax .
> >iax2 port is hardcoded in channels/iax2.h, line 72 (more or less)
> >You can change it & recompile.
> >
> >matteo.
> >
>
> Thanks a lot.
> Dan
> ...
>
> --__--__--
>
> Message: 13
> From: "YO Internet Information" <tan at yointernet.com>
> To: <asterisk-users at lists.digium.com>
> Subject: Re: [Asterisk-Users] UK Suppliers
> Date: Sat, 13 Sep 2003 13:30:41 +0100
> Organization: YO Internet Services Ltd
> Reply-To: asterisk-users at lists.digium.com
>
> http://www.telappliant.co.uk
>
>
>
>
> ----- Original Message -----
> From: "Lee Redmayne" <lee.redmayne at nwva.org>
> To: <asterisk-users at lists.digium.com>
> Sent: Saturday, September 13, 2003 1:11 PM
> Subject: RE: [Asterisk-Users] UK Suppliers
>
>
> I bought some Snom phones which work nicely with Asterisk from:
>
> ProVu Communications Ltd
> Bank House
> Marsden
> Huddersfield
> HD7 6BR
>
> 01484-840048
> info at provu.co.uk
> www.provu.co.uk
>
> -----Original Message-----
> From: Angel Gabriel
> Sent: 13 September 2003 13:02
> To: * Users
> Subject: [Asterisk-Users] UK Suppliers
>
> Can anyone please direct me to UK based suppliers of equipment. Website
> URL's would be appreciated. TIA
> --
> *****
> Not everyone is touched by an Angel....
> .... Those that are, never forget the experience
> *****
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --__--__--
>
> Message: 14
> Subject: Re: [Asterisk-Users] asterisk and defunct perl procs
> From: Angel Gabriel <badmangabriel at lycos.co.uk>
> To: * Users <asterisk-users at lists.digium.com>
> Date: 13 Sep 2003 13:24:52 +0100
> Reply-To: asterisk-users at lists.digium.com
>
> On Sat, 2003-09-13 at 13:42, Rich Adamson wrote:
> > FWIW, I just immplemented * on a RH9 box using the CVS without any
problems
> > whatsoever. The RH9 box was built from CD's as a workstation (with
everything
> > installed), up2date ran to bring it reasonably current, etc. I had
installed
> > "ser" a few weeks ago and it worked properly as well. Ser was shutdown
(still
> > remains installed) and * is running now.
>
> At the risk of sounding dumb, what's ser ?
> --
> *****
> Not everyone is touched by an Angel....
> .... Those that are, never forget the experience
> *****
>
>
> --__--__--
>
> Message: 15
> Date: Sat, 13 Sep 2003 07:49:22 -0600
> From: Rich Adamson <radamson at routers.com>
> Subject: Re: [Asterisk-Users] asterisk and defunct perl procs
> To: asterisk-users at lists.digium.com
> Reply-To: asterisk-users at lists.digium.com
>
>
> > > FWIW, I just immplemented * on a RH9 box using the CVS without any
problems
> > > whatsoever. The RH9 box was built from CD's as a workstation (with
everything
> > > installed), up2date ran to bring it reasonably current, etc. I had
installed
> > > "ser" a few weeks ago and it worked properly as well. Ser was shutdown
(still
> > > remains installed) and * is running now.
> >
> > At the risk of sounding dumb, what's ser ?
>
> From the 20,000 foot level:
>
> Asterisk is a PBX with lots of local features
>
> Ser is the Central Office switch ( http://www.iptel.org/ser/ )
>
> If you had hundreds/thousands of users and/or pbx's, ser typically handles
> the call routing. FWD uses ser as an example. Both are mostly open
source.
>
>
>
>
> --__--__--
>
> Message: 16
> Date: Sat, 13 Sep 2003 16:32:32 +0300
> From: Michael Manousos <manousos at inaccessnetworks.com>
> Organization: inAccess Networks
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Asterisk using a h323 gateway
> Reply-To: asterisk-users at lists.digium.com
>
> Cerrajetto wrote:
> > Hello:
> >
> > I am testing Asterisk with oh323.
> >
> > My question is: can Asterisk route some calls thru a second h323 gateway
(a
> > h323 <-> PSTN gw)?
> >
> > - Asterisk ip: 192.168.1.10
> > - h323<->PSTN gw: 192.168.1.20
> >
> > I've tried:
> >
> > exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20)
> >
> > or
> >
> > exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION at 192.1.1.20)
>
> I guess that "192.1.1.20" is a typo, right?
> You will have to give more info in order to be able to
> find the problem.
> Try to set these params in oh323.conf file:
>
> wrapLibTraceLevel=3
> libTraceLevel=3
> libTraceFile=/tmp/trace.txt
>
> Rerun and send me the "/tmp/trace.txt" file, "oh323.conf"
> and the screen log (off-list).
>
> >
> > but it does not work at all.
> >
> > If my h323 client directly uses 192.168.1.20 as h323 gateway, the calls
are
> > routed to the PSTN perfectly.
> >
> > What is the correct way to route some calls from Asterisk to another
h323
> > gateway?
> >
> > Thank you,
> > Mark
> >
>
>
> Michael.
>
Hi Mark,
Yes, it is possible. I have test it with Asterisk and oh323. We have routed
some calls thru a second h323 gateway (like Vegastream and Cirilium).
Following is the configuration:
; Vegastream
------------
exten => _01XXXXXXXXXX,1,Dial(OH323/BYEXTENSION at xxx.xxx.xxx.xxx)
; Crilium
---------
exten => _9XXXXXXXXXX,1,Dial(OH323/BYEXTENSION at xxx.xxx.xxx.xxx)
Shimul
>
>
>
> --__--__--
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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