Hi! When I try to call from a SIP phone to a PBX phone I get this error: chan_oh323.c [1004] Couldn`t call 483377839 and if I get the messages from SIP debug, I have a 403 message. The configuration of my system is: SIP Phone ---- ASterisk ---- Gatekeeper ----- Gateway ----- PBX ----- Phone Have someone any idea of what is going on?. It will be very nice if someone helps... it`s been more than a week that I can`t solve this problem. Best Regards, Mireia
Hi Mieria, Mireia Munoz de jesus wrote:>Hi! > >When I try to call from a SIP phone to a PBX phone I get this error: > >chan_oh323.c [1004] Couldn`t call 483377839 > >and if I get the messages from SIP debug, I have a 403 message. The >configuration of my system is: > >SIP Phone ---- ASterisk ---- Gatekeeper ----- Gateway ----- PBX ----- Phone > >Have someone any idea of what is going on?. It will be very nice if someone >helps... it`s been more than a week that I can`t solve this problem. > >Best Regards, > >Mireia >Could it be that you are using a *SIP* phone? Although you can add H.323 to Asteriskm, SIP and H.323 are different protocols... HTH, Martin
I believe your gatekeeper or your gateway is refusing the call. This can be a authorization problem in the gatekeeper or codec problem in the gateway. You need to see where your call is failing. Try to do the following: 1 - Turn on the oh323 trace in the oh323.conf file adding these lines to your configuration: wrapLibTraceLevel=3 libTraceLevel=3 libTraceFile=/var/log/asterisk/oh323.log 2 - Make a call from your SIP Phone to your PBX 3 - Look into the /var/log/asterisk/oh323.log and verify if the call is failing in the Admission Request or in the Setup message. 4 - If it fails in the Admission Request (you will see a Admission Reject into the log) the problem is in the configuration of your gatekeeper. 5 - If it fails in the Setup message (you will see a Release Complete into the log) the problem is in the configuration of your gateway Other thing you can see is if your asterisk box is registered with your gatekeeper. With the information you supplied this is what I remember you can check to see what is wrong. Regards, Vinicius -----Mensagem original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]Em nome de Mireia Munoz de jesus Enviada em: quarta-feira, 10 de mar?o de 2004 16:46 Para: asterisk-users@lists.digium.com; Martin Mielke Cc: asterisk-users@lists.digium.com Assunto: Re: [Asterisk-Users] 403 Forbidden Hi, Thanks for your answer, but my asterisk is working as a H.323 - SIP gateway and calls between SIP clients (phone and soft clients) are working all right. The only problem I have, is like I have said in my mail is between sip phones and PBX. Best Regards, Mireia PS: Someone have other ideas? Quoting Martin Mielke <martin.mielke@thales-is.com>:> Hi Mieria, > > Mireia Munoz de jesus wrote: > > >Hi! > > > >When I try to call from a SIP phone to a PBX phone I get this error: > > > >chan_oh323.c [1004] Couldn`t call 483377839 > > > >and if I get the messages from SIP debug, I have a 403 message. The > >configuration of my system is: > > > >SIP Phone ---- ASterisk ---- Gatekeeper ----- Gateway ----- PBX -----Phone> > > >Have someone any idea of what is going on?. It will be very nice ifsomeone> >helps... it`s been more than a week that I can`t solve this problem. > > > >Best Regards, > > > >Mireia > > > > Could it be that you are using a *SIP* phone? Although you can add > H.323 to Asteriskm, SIP and H.323 are different protocols... > > > HTH, > > Martin > > > _______________________________________________ > 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 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 --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.602 / Virus Database: 383 - Release Date: 01/03/2004 --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.602 / Virus Database: 383 - Release Date: 01/03/2004
Hi, Thanks for your answer, but my asterisk is working as a H.323 - SIP gateway and calls between SIP clients (phone and soft clients) are working all right. The only problem I have, is like I have said in my mail is between sip phones and PBX. Best Regards, Mireia PS: Someone have other ideas? Quoting Martin Mielke <martin.mielke@thales-is.com>:> Hi Mieria, > > Mireia Munoz de jesus wrote: > > >Hi! > > > >When I try to call from a SIP phone to a PBX phone I get this error: > > > >chan_oh323.c [1004] Couldn`t call 483377839 > > > >and if I get the messages from SIP debug, I have a 403 message. The > >configuration of my system is: > > > >SIP Phone ---- ASterisk ---- Gatekeeper ----- Gateway ----- PBX ----- Phone > > > >Have someone any idea of what is going on?. It will be very nice if someone > >helps... it`s been more than a week that I can`t solve this problem. > > > >Best Regards, > > > >Mireia > > > > Could it be that you are using a *SIP* phone? Although you can add > H.323 to Asteriskm, SIP and H.323 are different protocols... > > > HTH, > > Martin > > > _______________________________________________ > 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 >
Hi - I'm new to Asterisk and I've got asterisk@ home running with a TDM12B with Grandstream 2000 phones. I can login to the phones using the browser, login to the AMP, but I can't make any calls either internal or external, via softphone or Grandstream. I get an error 403 Forbidden in the soft phone. The Grandstream gives a busy signal and 503 error or 403 on line3. The calls are being logged in the AMP as NO ANSWER. I just sent this in to Digium, but I don't think it's a related issue. Thanks for any ideas - Tim eMail to Digium: I'm getting an error that indicates 1 of my 3 cards on the TDM12B is not working. I bought it from Digium Canada. -- DID YOU REMEMBER TO PLUG IN THE HD POWER CABLE TO THE TDM400P?? Unable to do INITIAL ProSLIC powerup on module 0 ProSLIC on module 0 failed to powerup within 510 ms (0 mV only) -- DID YOU REMEMBER TO PLUG IN THE HD POWER CABLE TO THE TDM400P?? Unable to do INITIAL ProSLIC powerup on module 0 Module 0: FAILED FXS (FCC) Module 1: Not installed Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV I (2 modules) Registered tone zone 0 (United States / North America) usb.c: registered new driver wcusb Wildcard USB FXS Interface driver registered Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) ip_tables: (C) 2000-2002 Netfilter core team sis900.c: v1.08.06 9/24/2002 divert: allocating divert_blk for eth0 eth0: Realtek RTL8201 PHY transceiver found at address 1. eth0: Using transceiver found at address 1 as default eth0: SiS 900 PCI Fast Ethernet at 0x8800, IRQ 9, 00:0c:6e:0d:1e:10. ip_tables: (C) 2000-2002 Netfilter core team eth0: Media Link On 100mbps full-duplex parport0: PC-style at 0x378 (0x778) [PCSPP,TRISTATE] parport0: irq 7 detected lp0: using parport0 (polling). lp0: console ready usb.c: registered new driver serial usbserial.c: USB Serial support registered for Generic usbserial.c: USB Serial Driver core v1.4 audit subsystem ver 0.1 initialized lspci 00:00.0 Host bridge: Silicon Integrated Systems [SiS] 651 Host (rev 02) 00:01.0 PCI bridge: Silicon Integrated Systems [SiS] Virtual PCI-to-PCI bridge (AGP) 00:02.0 ISA bridge: Silicon Integrated Systems [SiS] SiS962 [MuTIOL Media IO] (rev 25) 00:02.5 IDE interface: Silicon Integrated Systems [SiS] 5513 [IDE] 00:02.7 Multimedia audio controller: Silicon Integrated Systems [SiS] Sound Controller (rev a0) 00:03.0 USB Controller: Silicon Integrated Systems [SiS] USB 1.0 Controller (rev 0f) 00:03.1 USB Controller: Silicon Integrated Systems [SiS] USB 1.0 Controller (rev 0f) 00:03.3 USB Controller: Silicon Integrated Systems [SiS] USB 2.0 Controller 00:04.0 Ethernet controller: Silicon Integrated Systems [SiS] SiS900 PCI Fast Ethernet (rev 91) 00:0e.0 Network controller: Unknown device e159:0001 01:00.0 VGA compatible controller: ATI Technologies Inc Radeon RV250 If [Radeon 9000] (rev 01) 01:00.1 Display controller: ATI Technologies Inc Radeon RV250 [Radeon 9000] (Secondary) (rev 01) [