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@192.1.1.20) 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
exten => _9X.,1,Dial(H323/${EXTEN}@192.168.1.20)
If it's not working it's worth looking at the reson:
h.323 debug
h.323 trace 3
regards
Martin
On Fri, 12 Sep 2003, 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@192.1.1.20)
>
> 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
>
>
>
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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@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.
Shimul Kanti Barua
2003-Sep-17 03:58 UTC
[Asterisk-Users] Re: Asterisk using a h323 gateway
----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, September 13, 2003 7:55 PM Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs> Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > > You can reach the person managing the list at > asterisk-users-admin@lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Asterisk-Users digest..." > > > 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@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@linuxmail.org> > To: asterisk-users@lists.digium.com > Date: Sat, 13 Sep 2003 06:41:43 +0000 > Subject: Re: [Asterisk-Users] Caller ID Problems > Reply-To: asterisk-users@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 digiumcard 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 thesystem 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@fx.ro> > To: <asterisk-users@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@lists.digium.com > > Hi Martin, > > ----- Original Message ----- > From: "Martin Pycko" <martinp@digium.com> > To: "Asterisk Users" <asterisk-users@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@blueyonder.co.uk> > To: <asterisk-users@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@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@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Michael Kane > Sent: 12 September 2003 18:30 > To: asterisk-users@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@toad.net> > To: <asterisk-users@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@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@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@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@espia.it> > To: asterisk-users@lists.digium.com > Organization: Espia - Emmegi Srl > Date: Sat, 13 Sep 2003 09:52:34 +0200 > Reply-To: asterisk-users@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@digium.com> > > To: "Asterisk Users" <asterisk-users@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@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > Brancaleoni Matteo <mbrancaleoni@espia.it> > Espia - Emmegi Srl > > > --__--__-- > > Message: 5 > From: "Tjardick van der Kraan" <tjardick@vanderkraan.net> > To: <asterisk-users@lists.digium.com> > Subject: Re: [Asterisk-Users] Dect Phone > Date: Sat, 13 Sep 2003 10:47:29 +0200 > Reply-To: asterisk-users@lists.digium.com > > > ----- Original Message ----- > From: "Robert Boardman" <robb@boardman.me.uk> > To: <asterisk-users@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 giveme> 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@americanisp.net> > Organization: XQL, LLC > To: asterisk-users@lists.digium.com > Date: Sat, 13 Sep 2003 05:13:54 -0600 > Subject: [Asterisk-Users] Monitoring an active channel > Reply-To: asterisk-users@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@CustomerSide > and answer the phone when it rings. > 2. Next, I dial from the console to activate monitoring: > *CLI> dial 3@CustomerSide > Unfortunately, the monitoring does not start, and I hear Asterisk sending3> 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@routers.com> > Subject: Re: [Asterisk-Users] asterisk and defunct perl procs > To: asterisk-users@lists.digium.com > Reply-To: asterisk-users@lists.digium.com > > FWIW, I just immplemented * on a RH9 box using the CVS without anyproblems> whatsoever. The RH9 box was built from CD's as a workstation (witheverything> installed), up2date ran to bring it reasonably current, etc. I hadinstalled> "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 systemother> then to ensure the running kernel and its "matching" source code wasinstalled.> 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 startingasterisk:> > > > > > export LD_ASSUME_KERNEL=2.4.1 > > > > > > Hope this works! > > > > > > -Scott > > > > > > Scott M. Stingel > > <snip> > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@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@routers.com> > Subject: Re: [Asterisk-Users] Caller ID Problems > To: asterisk-users@lists.digium.com > Reply-To: asterisk-users@lists.digium.com > > I'm having some of the same issues and it seems to be related totransmission> levels. CallerID worked fine prior to me messing with rxgain/txgain, butI'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 thatline.. Its is not always a> standard feature.. > > > > 2. The CallerID that your telco provides is not compatible with thedigium 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 thesystem 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@lycos.co.uk> > To: * Users <asterisk-users@lists.digium.com> > Date: 13 Sep 2003 13:01:32 +0100 > Subject: [Asterisk-Users] UK Suppliers > Reply-To: asterisk-users@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@nwva.org> > To: <asterisk-users@lists.digium.com> > Subject: RE: [Asterisk-Users] UK Suppliers > Date: Sat, 13 Sep 2003 13:11:52 +0100 > Reply-To: asterisk-users@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@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@lycos.co.uk> > To: * Users <asterisk-users@lists.digium.com> > Date: 13 Sep 2003 13:22:11 +0100 > Subject: [Asterisk-Users] How to test * ? > Reply-To: asterisk-users@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@lists.digium.com > Subject: Re: [Asterisk-Users] IAX, IAX2 and authenticatyion > From: dtoma@fx.ro > Reply-To: asterisk-users@lists.digium.com > > At Sat, 13 Sep 2003 09:52:34 +0200 , asterisk-users@lists.digium.comwrote:> > >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@yointernet.com> > To: <asterisk-users@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@lists.digium.com > > http://www.telappliant.co.uk > > > > > ----- Original Message ----- > From: "Lee Redmayne" <lee.redmayne@nwva.org> > To: <asterisk-users@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@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@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@lycos.co.uk> > To: * Users <asterisk-users@lists.digium.com> > Date: 13 Sep 2003 13:24:52 +0100 > Reply-To: asterisk-users@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 anyproblems> > whatsoever. The RH9 box was built from CD's as a workstation (witheverything> > installed), up2date ran to bring it reasonably current, etc. I hadinstalled> > "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@routers.com> > Subject: Re: [Asterisk-Users] asterisk and defunct perl procs > To: asterisk-users@lists.digium.com > Reply-To: asterisk-users@lists.digium.com > > > > > FWIW, I just immplemented * on a RH9 box using the CVS without anyproblems> > > whatsoever. The RH9 box was built from CD's as a workstation (witheverything> > > installed), up2date ran to bring it reasonably current, etc. I hadinstalled> > > "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 opensource.> > > > > --__--__-- > > Message: 16 > Date: Sat, 13 Sep 2003 16:32:32 +0300 > From: Michael Manousos <manousos@inaccessnetworks.com> > Organization: inAccess Networks > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Asterisk using a h323 gateway > Reply-To: asterisk-users@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@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 callsare> > routed to the PSTN perfectly. > > > > What is the correct way to route some calls from Asterisk to anotherh323> > 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@xxx.xxx.xxx.xxx) ; Crilium --------- exten => _9XXXXXXXXXX,1,Dial(OH323/BYEXTENSION@xxx.xxx.xxx.xxx) Shimul> > > > --__--__-- > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > > > End of Asterisk-Users Digest >
Why do I need to read all the other stuff just to get to a 3 liner ? On Wednesday 17 September 2003 12:58 pm, Shimul Kanti Barua wrote:> ----- Original Message ----- > From: <asterisk-users-request@lists.digium.com> > To: <asterisk-users@lists.digium.com> > Sent: Saturday, September 13, 2003 7:55 PM > Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs >...
Hi all, Thank you for your help, finally we have found that it was a codec problem, now both systems are forced to use g711 ulaw and outbound calls are working fine. Best regards, Mark. ---------- Original Message ----------- From: "Cerrajetto" <cerrajetto@pyme.net> To: asterisk-users@lists.digium.com Sent: Fri, 12 Sep 2003 18:30:54 +0200 Subject: [Asterisk-Users] Asterisk using a h323 gateway> 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@192.1.1.20) > > 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 > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users------- End of Original Message -------