Ok. I give in! I was tempted by the wiki that mentions the (very undocumented) VXML_URL and suggests it might be able to control the display on a Cisco phone during an incoming call using a SIP image. I've mucked around with this for over two hours and after scouring source code, google, and the archives have found nothing. Does anyone have any how to use this feature? Does it even really exist? I can see the header being set and hitting the phone - but I can't find documentation anywhere suggesting what format you can send it. Sleepless in the server room, --- Paul Andersen E-Gate Communications Inc. "The only thing to do with good advice is to pass it on. It is never any use to oneself" - Oscar Wilde
Thy looking at the VTGO-PC softphone by IPBlue at http://www.ipblue.com/downloadSales.htm... It's an SCCP softphone that emulates the Cisco 7960... they have a 30 day evaluation version that includes sample VXML scripts. ----- Original Message ----- From: "Paul Andersen" <asterisk@paul.ca> To: <asterisk-users@lists.digium.com> Sent: Saturday, March 13, 2004 3:15 PM Subject: [Asterisk-Users] VXML_URL and Cisco 7960 Phones?> > Ok. I give in! > > I was tempted by the wiki that mentions the (very undocumented) VXML_URL > and suggests it might be able to control the display on a Cisco phone > during an incoming call using a SIP image. > > I've mucked around with this for over two hours and after scouring source > code, google, and the archives have found nothing. > > Does anyone have any how to use this feature? Does it even really exist? I > can see the header being set and hitting the phone - but I can't find > documentation anywhere suggesting what format you can send it. > > Sleepless in the server room, > > --- > Paul Andersen > E-Gate Communications Inc. > > "The only thing to do with good advice is to pass it on. > It is never any use to oneself" - Oscar Wilde > > > _______________________________________________ > 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
> I was tempted by the wiki that mentions the (very undocumented) VXML_URL > and suggests it might be able to control the display on a Cisco phone > during an incoming call using a SIP image. > > I've mucked around with this for over two hours and after scouring source > code, google, and the archives have found nothing. > > Does anyone have any how to use this feature? Does it even really exist? I > can see the header being set and hitting the phone - but I can't find > documentation anywhere suggesting what format you can send it.It's my understanding, although I've no direct experience, the function does not exist in the SIP images. The limitation is highly likely related to Cisco marketing plans and not to real design/programming capability, etc. (How else would one sell proprietary systems?) Anyone have a disassembler?
I tried to get that working as well and also found it was not available in the SIP image. You can't do pushes either to the phone like you can with SCCP. -----Original Message----- From: Rich Adamson [mailto:radamson@routers.com] Sent: 14 March 2004 13:27 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] VXML_URL and Cisco 7960 Phones?> I was tempted by the wiki that mentions the (very undocumented) VXML_URL > and suggests it might be able to control the display on a Cisco phone > during an incoming call using a SIP image. > > I've mucked around with this for over two hours and after scouring source > code, google, and the archives have found nothing. > > Does anyone have any how to use this feature? Does it even really exist? I > can see the header being set and hitting the phone - but I can't find > documentation anywhere suggesting what format you can send it.It's my understanding, although I've no direct experience, the function does not exist in the SIP images. The limitation is highly likely related to Cisco marketing plans and not to real design/programming capability, etc. (How else would one sell proprietary systems?) Anyone have a disassembler? _______________________________________________ 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 ********* DISCLAIMER ********* This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person
I am currently using Asterisk behind Belkin NAT router. With what ever NAT router I have used, I have had difficulties in registration and audio problems with my SIP provider (Iconnect and Nikotel) It was suggested that I try to connect the asterisk box directly to the internet avoiding the NAT transition. As I will still need internet connectivity, I am trying to make the asterisk box the NAT gateway. I have an additional NIC for my Asterisk box. As I am no Linux or Asterisk expert, can anyone make suggestions as to this approach and any recommended steps to accomplish this? Also, how would Asterisk know which interface to bind to? I know there is a bindaddress= parameter in the SIP config, but the address to the internet is dynamic via DHCP from my cable provider. Thanks, Kevin