If your firewall has a packet capture facility, you can do a pcap on the WAN interface and see what they are sending.
I would suspect if anyone has a working teliax config they will share it. On Tue, Sep 7, 2010 at 6:15 PM, Tony Graziano <tgrazi...@myitdepartment.net>wrote: > I think unless you are wed to them, it would be easier to switch to a > "normal" provider. Supported providers in the templates usually take 5 > minutes to setup. I HOPE your firewall is doing manual versus automatic NAT. > > I looked at Teliax and they seem "residentially" focused, and really > expensive for business plans. > > > On Tue, Sep 7, 2010 at 6:12 PM, Stiles Watson <wat...@datatek-net.com>wrote: > >> Unfortunately, there is no way in the Teliax portal to even see if you >> are registered, much less what port. >> >> The reason I had 5060 forwarded to sipx was this was how I had Trixbox CE >> setup and working. There is nothing in my Teliax setup which I changed to >> force 5060. >> >> Thanks for the pdf. With the exception of the SIP port, I think I have >> everything setup correctly. I changed my NAT rules to forward 5080 instead >> of 5060 and the call acted exactly the same. >> >> I've also asked Teliax if they have config info for sipX and they said no, >> but many are using the two together successfully. Here is their exact >> response: >> >> "We do not have a have a configuration for them. However, I know that many >> customers have used SIPXECS without a problem. The main information you need >> is the username, secret, and host that you are registering to." >> >> I've asked them what port they are sending the INVITE on and am waiting on >> a response. >> >> Any other suggestions/thoughts? >> >> Stiles >> >> Tony Graziano wrote: >> >> It means they are not acking the call. I suspect this is because >> sipxbridge may not be involved in the call, and only sipxproxy is, which >> would be problematic for a lot of call scenarios (like transfers). >> >> I'm confused though, because it seems you are breaking "rule #1" when >> using sipxbridge... you are having the calls sent to port 5060 instead of >> 5080. >> >> When you register with teliax, can you see on their portal what port you >> are registering on? Can you confirm they are sending to you on a specific >> port? If so, what port? >> >> You should peek at this: >> >> >> http://blog.myitdepartment.net/wp-content/uploads/2009/11/Call-Setup-Example-sipXecs-through-ITSP1.pdf >> >> Somehow I don't believe you are doing it quite like that. >> >> >> >> On Tue, Sep 7, 2010 at 5:18 PM, Stiles Watson <wat...@datatek-net.com>wrote: >> >>> Running >>> >>> - sipXecs v 4.2.1 >>> - ITSP is Teliax >>> - SIP ports 5060 & 5061 are routed to sipX server >>> - RTP ports 30000-31000 are routed to sipX server >>> - Polycom IP 335 hardphone >>> >>> I'm able to place incoming and outgoing calls through Teliax, but calls >>> consistently drop after 1 min. 29 sec. >>> >>> Teliax device config change attempts: >>> >>> - Enable DNIS (teliax sends number in sip invite instead of user) >>> - result: calls still drop after 1 min. 29 sec., but made call >>> routing easier via a custom DID! >>> - Entered public IP under "Your IP" >>> - This is optional and resulted in not being able to make inbound >>> calls (I read in the archives that this is recommended with Teliax - >>> is >>> there a sipX config change needed to make this work?) >>> >>> sipX config for teliax SIP trunk Gateway: >>> >>> - Configuration >>> - Enabled: yes >>> - Name: teliax >>> - SBC Route: sipXbridge-1 >>> - Address: den.teliax.net (this has to match with the proxy >>> setting in your teliax account) >>> - Port: 0 >>> - Transport protocol: Auto >>> - Location: all >>> - Shared: yes >>> - Caller ID >>> - Default Caller ID: set this to the number from Teliax >>> - use default for all other settings >>> - Dial Plan >>> - Enabled and added both Local & Long Distance dial plans to this >>> gateway >>> - ITSP Account >>> - Username: use teliax username >>> - Authentication Username: same as Username >>> - Password: use teliax device password >>> - Register on init: yes >>> - ITSP server address: same as Config-->Address above >>> - Use public address for call setup: yes (I tried both yes and no, >>> calls completed either way and did not effect disconnect problem) >>> - Strip private headers: default >>> - Use default asserted identity: default >>> - Asserted identity: default >>> - Use default preferred identity: default >>> - Preferred identity: default >>> - User part of INVITE SIP URI is a phone number: NO >>> - ITSP Registrar Address: default >>> - ITSP Registrar Port: default >>> - Registration interval: default >>> - Session Timer Interval: default >>> - Method to use for SIP keepalive: Empty SIP message (also tried >>> None) >>> - Method to use for RTP keepalive: Replay last sent packet (also >>> tried None) >>> - Route by To Header: default >>> >>> Any thoughts as to why the calls would drop after 1 min. 29 sec.? >>> >>> Stiles >>> >>> _______________________________________________ >>> sipx-users mailing list >>> sipx-users@list.sipfoundry.org >>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>> >> >> >> >> -- >> ====================== >> Tony Graziano, Manager >> Telephone: 434.984.8430 >> sip: tgrazi...@voice.myitdepartment.net >> Fax: 434.984.8431 >> >> Email: tgrazi...@myitdepartment.net >> >> LAN/Telephony/Security and Control Systems Helpdesk: >> Telephone: 434.984.8426 >> sip: helpd...@voice.myitdepartment.net >> Fax: 434.984.8427 >> >> Helpdesk Contract Customers: >> http://www.myitdepartment.net/gethelp/ >> >> Why do mathematicians always confuse Halloween and Christmas? >> Because 31 Oct = 25 Dec. >> >> ------------------------------ >> >> _______________________________________________ >> sipx-users mailing listsipx-us...@list.sipfoundry.org >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> >> >> _______________________________________________ >> sipx-users mailing list >> sipx-users@list.sipfoundry.org >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> > > > > -- > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: tgrazi...@voice.myitdepartment.net > Fax: 434.984.8431 > > Email: tgrazi...@myitdepartment.net > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: helpd...@voice.myitdepartment.net > Fax: 434.984.8427 > > Helpdesk Contract Customers: > http://www.myitdepartment.net/gethelp/ > > Why do mathematicians always confuse Halloween and Christmas? > Because 31 Oct = 25 Dec. > > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
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