Hi Im trying to make SIP to H.323 calls, redirecting from SER to Asterisk and then to GNUGK I tried hard to do it but the documentation is not very explicit.
As SIP Im using a ZyXel Prestige 2000P and X lite, and as H.323 im using an Aethra.
What do I need in the SER configuration (ser.cfg) to redirect the H.323 calls to asterisk?
I tried the rewritehostport:
rewritehostport("Asterisk_ip : Asterisk_port"); t_relay();
.... but Asterisk didnt received the calls.
Can anyone tell me how to redirect H.323 calls to Asterisk?
Thanks in advance Joao Pereira
Joao,
--- Joao Pereira joao.pereira@fccn.pt wrote:
Im trying to make SIP to H.323 calls, redirecting from SER to Asterisk and then to GNUGK I tried hard to do it but the documentation is not very explicit.
<snip>
rewritehostport("Asterisk_ip : Asterisk_port");
<snip>
Can anyone tell me how to redirect H.323 calls to Asterisk?
The rewritehostport should work for you. Have you installed the h323 channel driver for Asterisk? If yes, which one? Experience from people say that Asterisk's native h323 channel driver has issues bridging calls between SIP and H323. You need to use the channel driver from inaccessnetworks (chan_oh323).
Here's a caveat if you are installing oh323 driver: Read the documentation carefully before installing pwlib and openh323 libraries for the driver. Unless you choose the right versions and follow the readme, you'll never be able to install the driver. Just search through the * archives to see how many people have faced problems installing the driver.
Best,
===== Girish Gopinath gr_sh2003@yahoo.com
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Hi I am using this condition that works very well for me:
if(uri=~"^sip:6607.@(my.domain.com.pe|a.b.c.d)"){ rewritehostport("w.x.y.z:5060"); ## --->asterisk IP:port log(1,"n digit expression match - SIP_to_H323 "); route(2); break; };
route[2]{ log(1,"route[2]:SIP-to-* call routed"); if(!t_relay()){ sl_reply_error(); };
in sip.conf:
[general] context=default autocreatepeer=yes canreinvite=no
[mic-inout] type=friend secret=****** username=asterisk ; Authentication user for outbound proxies fromuser=asterisk ; Many SIP providers require this! host=my.domain.com.pe dtmfmode=rfc2833 insecure=very ; needed if we want to allow incoming SER calls to bypass authentication
...you have to register Asterisk as a gateway in GNUGK, see oh323.conf...
in extensions use exten => _00NXX.,1,Dial,OH323/${EXTEN}
and if you are using oh323 from inaccessnetworks in asterisk try *CLI> oh323 debug toggle ...maybe provide you some useful logs to solve your problem...
rafael
PS: I recommend this versions: - PWLIB : pwlib_1.5.2.tar.gz - OpenH323 : openh323_1.12.2.tar.gz - Inaccessnetworks-asterisk-oh323 : asterisk-oh323-0.7.0.tar.gz (see readme file, there is a note about a patch) - Asterisk 1.0+ - GNUGK (from www.gnugk.org) versions 2.2+ or 2.0.9, note: radius billing for sip-to-h323 calls does not work properly with older versions...
On Sat, 11 Dec 2004 03:38:21 -0800 (PST), Girish gr_sh2003@yahoo.com wrote:
Joao,
--- Joao Pereira joao.pereira@fccn.pt wrote:
Im trying to make SIP to H.323 calls, redirecting from SER to Asterisk and then to GNUGK I tried hard to do it but the documentation is not very explicit.
<snip>
rewritehostport("Asterisk_ip : Asterisk_port");
<snip>
Can anyone tell me how to redirect H.323 calls to Asterisk?
The rewritehostport should work for you. Have you installed the h323 channel driver for Asterisk? If yes, which one? Experience from people say that Asterisk's native h323 channel driver has issues bridging calls between SIP and H323. You need to use the channel driver from inaccessnetworks (chan_oh323).
Here's a caveat if you are installing oh323 driver: Read the documentation carefully before installing pwlib and openh323 libraries for the driver. Unless you choose the right versions and follow the readme, you'll never be able to install the driver. Just search through the * archives to see how many people have faced problems installing the driver.
Best,
===== Girish Gopinath gr_sh2003@yahoo.com
Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Hi thanks for the answer, I tried your condition in ser.cfg but it had errors and I could not fix it.
I also think that the rewritehostport("Asterisk_ip : Asterisk_port") should work, maybe the problem is in the Asterisk configuration...
In the extensions.conf, I must put the extension ( exten => _00NXX.,1,Dial,OH323/${EXTEN} ) in one context, In witch context should I put it?
Asterisk is already as a gateway in GNUGK as shown in the gnugk monitorization: RCF|Asterisk_ip:Asterisk_port|asterisk:h323_ID=ASTERISK:h323_ID=664:dialedDi gits|gateway|8478_endp
I installed oh323 for Asterisk. The versions I have are the following: pwlib-1.6.6-0_11.rh9 openh323-1.13.5-0_13.rh9 gnugk-2.0.8-linux-x86
Joao Pereira
----- Original Message ----- From: "Rafael J. Risco G.V." rafael.risco@gmail.com To: "Joao Pereira" joao.pereira@fccn.pt; serusers@lists.iptel.org Sent: Sunday, December 12, 2004 6:33 AM Subject: Re: [Serusers] SER forwarding to Asterisk
Hi I am using this condition that works very well for me:
if(uri=~"^sip:6607.@(my\.domain\.com\.pe|a\.b\.c\.d)"){ rewritehostport("w.x.y.z:5060"); ## --->asterisk
IP:port
log(1,"n digit expression match - SIP_to_H323 "); route(2); break; };
route[2]{ log(1,"route[2]:SIP-to-* call routed"); if(!t_relay()){ sl_reply_error(); };
in sip.conf:
[general] context=default autocreatepeer=yes canreinvite=no
[mic-inout] type=friend secret=****** username=asterisk ; Authentication user for outbound
proxies
fromuser=asterisk ; Many SIP providers require this! host=my.domain.com.pe dtmfmode=rfc2833 insecure=very ; needed if we want to allow incoming SER calls to bypass authentication
...you have to register Asterisk as a gateway in GNUGK, see oh323.conf...
in extensions use exten => _00NXX.,1,Dial,OH323/${EXTEN}
and if you are using oh323 from inaccessnetworks in asterisk try *CLI> oh323 debug toggle ...maybe provide you some useful logs to solve your problem...
rafael
PS: I recommend this versions:
- PWLIB : pwlib_1.5.2.tar.gz
- OpenH323 : openh323_1.12.2.tar.gz
- Inaccessnetworks-asterisk-oh323 : asterisk-oh323-0.7.0.tar.gz
(see readme file, there is a note about a patch)
- Asterisk 1.0+
- GNUGK (from www.gnugk.org) versions 2.2+ or 2.0.9, note: radius
billing for sip-to-h323 calls does not work properly with older versions...
On Sat, 11 Dec 2004 03:38:21 -0800 (PST), Girish gr_sh2003@yahoo.com
wrote:
Joao,
--- Joao Pereira joao.pereira@fccn.pt wrote:
Im trying to make SIP to H.323 calls, redirecting from SER to Asterisk
and then to GNUGK
I tried hard to do it but the documentation is not very explicit.
<snip>
rewritehostport("Asterisk_ip : Asterisk_port");
<snip>
Can anyone tell me how to redirect H.323 calls to Asterisk?
The rewritehostport should work for you. Have you installed the h323
channel driver for Asterisk?
If yes, which one? Experience from people say that Asterisk's native
h323 channel driver has
issues bridging calls between SIP and H323. You need to use the channel
driver from
inaccessnetworks (chan_oh323).
Here's a caveat if you are installing oh323 driver: Read the
documentation carefully before
installing pwlib and openh323 libraries for the driver. Unless you
choose the right versions and
follow the readme, you'll never be able to install the driver. Just
search through the * archives
to see how many people have faced problems installing the driver.
Best,
===== Girish Gopinath gr_sh2003@yahoo.com
Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
--
rrgv
Hi,
--- Joao Pereira joao.pereira@fccn.pt wrote:
I also think that the rewritehostport("Asterisk_ip : Asterisk_port") should work, maybe the problem is in the Asterisk configuration...
In the extensions.conf, I must put the extension ( exten => _00NXX.,1,Dial,OH323/${EXTEN} ) in one context, In witch context should I put it?
I dont think that the problem is with SER. I never faced any problems in using rewritehostprot for call re-direction. Most probably your problem could be in the Asterisk dialplan.
exten => _.,1,Dial,OH323/${EXTEN}
Try adding that in the default context. Enable sip debug on the Asterisk console and check if the packets from SER are reaching Asterisk. If they reach, see why they are dropped. My guess is that the packets reach an incorrect context in Asterisk dialplan and hence dropped. Also, it could be a codec issue.
Remove the above dial statement once you figure out the problem. Adding such a line in the default context is not suggested.
Good Luck!
===== Girish Gopinath gr_sh2003@yahoo.com
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