Not so fast. Can other internal users calls the user the FXS port is connected to? Does it actually say it is registered in sipx? Calling internally "to" other users might not be an issue even if not registered. Calling "to" this user still might be. DNS could still be an issue if the sipura is not using it's dial plan correctly. If it has a logging function, I would use it and see what it tells you. Can it be assumed the sipura and the sipx are on the same network and do not have any firewall in between? Enabling the dial plan is not the same as activating it. Did you activate it (it restarts services once this is invoked, and does make a difference)?
>>> From: "Alberto Furtado" <[EMAIL PROTECTED]> To:<[email protected]> Date: 11/3/2008 7:26 AM Subject: Re: [sipx-users] Unable to dial out, 404 Not Found Code Mike, 1. The User/FXS is getting registered and I can call other extensions. 2. I created the gateway putting the Name/IP/Port (Sipura/198.168.0.138/5060) I added the gateway to the emergency and local dial plans on the gateway and enabled the plans. 3. I don't think DNS has any affect on this since the gateway uses IP address directly. I did notice that the sipxcallresolver-agent is disabled but I don't know if this is related since I saw that in other posts that this was also listed as disabled. Initially I was thinking this was a 3102 problem since other posts indicate problems with FXS/FXO ATAs. But looking at the sipwiewer I noticed it doesn't contact the 3102 at all. The fact that the server does not try to contact the Sipura and gives a 404 error leads to me believe that it thinks there isn't a path defined for this call. Either I did something wrong with the dial plans or the gateway configuration. But I have no clues to what I did wrong. I even added the Emergency that doesn't need permissions and the dial plan is simpler!!! Greetings from Rio de Janeiro and Thanks very much for your input Mike, Alberto ----- Original Message ----- From: Picher, Michael ( mailto:[EMAIL PROTECTED] ) To: Alberto Furtado ( mailto:[EMAIL PROTECTED] ) ; [EMAIL PROTECTED] ( mailto:[email protected] ) Sent: Monday, November 03, 2008 7:29 AM Subject: RE: [sipx-users] Unable to dial out, 404 Not Found Code Alberto, The Sipuras are definitely frustrating unitsÂ… First, did you verify that the FXS is actually getting registered to the PBX? Second, did you activate your dial plan after you added the gateway to the dial plan entries? Third most common culprit is DNS configuration. Mike From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Furtado Sent: Sunday, November 02, 2008 1:04 PM To: [email protected] Subject: [sipx-users] Unable to dial out, 404 Not Found Code Hi All, I am testing a basic system for a future implementation. I have tested several configurations with 2 Sipura 3102(1FXS/1FXO) and a SipXecs 3.10.2 CentOS5 Server. But I am unable to dial out using the PSTN gateway. This is my setup: 1. Only one User configured. 2. One Sipura configured to register to this User. 3. One Unmanaged Gateway configured to the address of second Sipura PSTN port. 4. Second Sipura configured not to register. 5. Added the Gateway to Emergency and Local Dial Plan. The User side seems OK, I can dial 101 (or other extensions, when configured), but when I try to dial anything else I get a busy signal. Using sipviewer I can see that the sip.myserver.com-SipRegistrar returns a 404 Not Found Code, and that the server never tries to contact the Second Sipura. I have read the manuals, the wiki and the posts and found nothing similar to my problem. I am probably missing out on something very simple in the configuration of the server. I would REALLY appreciate any suggestions!!! Thanks in advance for the patience with a newcomer, Alberto PS: Its been as exhausting week after being unable to install on Fedora, unable to use the auto-installation that only boots in machines that have Via chipsets and crash. But I was able to get very interesting configurations to work, like configuring 2 Users in each Sipura 3102, one on the Voip1(FXS) an another on the Voip2(FXO) using the VoIP->PSTN gateway. When I dial the user in Voip2 port I get the line out dial tone and can call out.And using the PSTN->VoIP when I call in to the PSTN port I get the tone signal of the SipXecs. Installation problems are complicated but the SipXecs seems to be very powerful and rich in features. No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.175 / Virus Database: 270.8.5/1762 - Release Date: 2/11/2008 09:51
_______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
