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cmrabet's Achievements


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  1. Yes you are right, I fixed the issue editing manually the config.xml file. I did this because support from Skype told me that Skype connect doesn't support TLS ports, and he told me to set it to 5060...wrong guy. Fixed. Thanks.
  2. Thanks for the document but we don't have those options. Our PBX version is (Linux), with a license for 10 extensions. I'm not sure if we can upgrade it to the same version as the one shown on your document. I fixed the issue anyway, I had to set the RTP port from 8000 to 8010, and then in the trunk, set the "Assume that call comes from user" to the Skype Connect user account number, since it seems that without this parameter, the user sent to Skype is the extension one, and therefore FORBIDDEN from Skype. In my case, placing calls from extension 101, was doing 101@skype which was rejected. Thanks.
  3. A long day with the SkyPe support department and there is no way to make this to work. Ports 5060, and 8000-8010 opened at the router side. At the PBX side I've changed the RTP ports to 8000 to 8010 as requested by them, and restarted the service. I created SIP register trunk with my user account and so, and it registers correctly. However when placing calls we just get a busy tone...looking at the log I see PBXnSIP is using SIP TLS 5061 port....Skype guys claim their service is not compatible with TLS... So I don't know what else to do..
  4. The only thing I did was setting the SIP TLS port to the same as the regular SIP one, 5060. I restarted and since then I'm having this problem. The daemon doesn't come up, so I don't have access to the Web interface anymore.
  5. root@ITE-Server:/usr/local/pbxnsip# ./pbxctrl --dir /usr/local/pbxnsip/ --config config.xml --log log.file --no-daemon Starting up [0] 20110712154847: UDP: bind() to port 5060 failed [0] 20110712154847: FATAL: Could not open UDP port 5060 for SIP But I restarted many times since the first time, and the daemon never came up since then, no process is running for PBXnSIP, and NETSAT doesn't show anything.... ????
  6. I can't understand what is going on. I changed the SIP ports, then went to the console of our Debian server where PBXnSIP is installed, and restarted the process as follows: root@ITE-Server:/etc/init.d# ./pbxnsip restart Shutting down pbxnsip daemon:/usr/local/pbxnsip/pbxctrl: no process killed Starting pbxnsip daemon The went to the Web browser and did as usually : That's the IP of the PBXnSIP server, and the port we configured for web access... AND NOTHING, the web is not UP. I went back to our Linux server, and it doesn't start any service, no error, no nothing. The PBXnSIP daemon doesn't restart!!! Any help please?
  7. Hi I have a new problem. I detected someone form outside our network registering in one of our extensions to place calls through one of our trunks (CallWithUS) running out our credit. The CallWithUS call records show calls made from this extension to a crazy list of different countries (that we do not call to). I changed the SIP password for the extensions but this happened again. So far 3 times. What is your recommendation to protect the PBXnSIP server against this? I had also set a Trusted IP Addresses to every extension but this did not make the trick either. Thanks.
  8. What ports are required to register PBXnSIP with third party services such CallWithUS when is not under DMZ? I solved my ASA issue here http://forum.pbxnsip.com/index.php?showtop...amp;#entry17147 but now I can not register. Thanks!
  9. I found the PBXnSIP configuration setup field "SIP IP Replacement List" set to (private static PBnSIP IP/our public static IP). I erased the content of this field, restarted the PBXnSIP server and voila!, sound during the calls and everything working now, except: - 408 Request Timeout (Registration failed, retry after 60 seconds), when trying to register to CallWithUS - 408 Request Timeout (Registration failed, retry after 60 seconds), when trying to rgister to TelSome The PBXnSIP had no issue before to register to these services when it was working under DMZ with the former router. Also the SIP port is opened 5060, so I don't understand what could be the reason for this. Any clue? Thanks.
  10. Yes I can ping the PBX server, actually I use the Web interface to control it. As a matter of fact all the SIP phones are correctly registered to it....so obvisouly it is available to at least the internal network. On the other hand, there is a question that nobody answered yet: Why this happens internally? I can understand if happened from an external caller (a SIP phone in internet trying to access to my LAN in order to register to the PBX server), but INTERNAL extensions? I mean, SIP phones that are located at or that are registered to a PBX server at, but no sound when calling.... ???
  11. UPDATE: The CISCO ASA reseted to factory defaults, working perfectly with a Web server (NAT for port 80), allowing ping between all the machines in the netwrok, everything brand new from scratch, no ACCESS RULES (so all the traffic is allowed): - PBXnSIP server can not register to CallWithUs or other services we have like VOIPRAIDER (Timeout) - No sound when placing calls There is no limitation in the traffic but the PBXnSIP server still can not work. DOES THIS THING MANDATORY NEED A DMZ? WHAT IS THE REASON THE CALLS CAN NOT TAKE PLACE INTERNALLY WHEN NO ACCESS RULES ? PLease, I will really appreciate your help, since this is really a desperate situation.
  12. This is driving me crazy. I have switched back to our former router and set up the PBXnSIP server the same way I did with the CISCO ASA box. It did not work! it behaved the same exact way. However I moved the PBXnSIP server under a DMZ configuration, and guess what. IT WORKED. I think this is what is happening in the CISCO firewall too. I need to set a DMZ for the PBXnSIP server. But the question is WHY? Is there any other port that we the user do not know ???? 5060, 5061, etc...???
  13. I still don't understand: 1) All the SIP phones (internal extensions) register correctly to the PBX server (a computer in the same LAN) 2) All SIP phones (internal extensions) can ring to others (you can hear the ringing sound) 3) When an extension calls to another one and this one answers, you can see the time ticking (so the call is taking place), BUT NO SOUND (the media ports??) I can understand a FIREWALL problem if this was happening from an outsider extension trying to register to our internal PBXnSIP server, but the issue is internally!!!! Another thing that I don't understand is that when the PBXnSIP was under DMZ with the former router everything was working great. Now with the CISO ASA box, with all the ports required correctly open, no MEDIA is transfered between PBXnSIP server and rest of extensions. Two more details: 4) If you call from outside (using a regular phone line) to our number, our PBXnSIP auto attendant answers correctly and you can hear all the different options you can dial. However it doesn't matter if you click any number, nothing happens, the auto attendant keeps talking and talking. 5) The PBXnSIP can not register with CallWithUS and other providers we have. With the former router it could (but we have all the ports opened!) Any advice please? Don't you think there is something wrong at the PBXnSIP side? Thanks.
  14. But it was registering correctly just 5 minutes before switching from the regular Linksys WAG router. This started to happen when using the CISCO ASA 5500. Thanks.
  15. I noticed also that the PBX failed to register with external trunks we have such CallWithUS.... mmm Is this fact any help to locate the issue? Thanks.
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