RE: [Asterisk-Users] Direct SIP connection to Vonage service
On Friday, October 22, 2004 2:40 PM Stewart Nelson wrote: I presently have a small VoIP network using H.323 and gnugk, and would like to upgrade it to an Asterisk-based system, primarily to take advantage of low cost unlimited calling plans offered by SIP providers such as Vonage. FYI these so called unlimited monthly plans are RARELY, if _EVER_ truly unlimited. They CAN (read the TOS), and WILL terminate you if you use too many minutes more then whatever average they calculated for when pricing the plan. I personally know several people who were using the Vonage unlimited calling plan and were terminated for _EXCESSIVE USAGE_ However, the carriers with good reputations for reliability and quality seem to require that you connect via a locked ATA device. SNIP As some other people have suggested, your best bet is to just use a VoIP provider who natively supports the InterAsteriskExchange protocol. Two that I know of are NuFone.net and tollfreeexpress.com . Try Google and the voip-info.org wiki for others. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Direct SIP connection to Vonage service
I presently have a small VoIP network using H.323 and gnugk, and would like to upgrade it to an Asterisk-based system, primarily to take advantage of low cost unlimited calling plans offered by SIP providers such as Vonage. However, the carriers with good reputations for reliability and quality seem to require that you connect via a locked ATA device. It appears that many other users of these services desire a direct SIP connection. Perhaps we can work together to achieve this. Some users have said that you can connect the ATA to an FXO port on your system; the extra conversions to analog and back to digital only slightly impair quality. IMHO, that is not true. Here are my Top Ten reasons why one shouldn't connect Asterisk to a VoIP service through the provider's ATA: 10. Increased noise caused by extra D/A and A/D conversion. 9. Unnecessary extra delay of at least 150 ms in speech path, causing problems when both parties try to speak at the same time, and making echo suppression difficult. 8. Extra echo source caused by needlessly converting from 4-wire to 2-wire and back again. 7. At least two seconds additional call setup time, because destination number must be sent via DTMF on outgoing, and you must wait for CLID data on incoming. 6. If ever need to use compression codecs, one would have two such codecs cascaded, resulting in abysmal quality. 5. Destroys answer supervision and other call progress info. 4. No ability to REINVITE, resulting in needless delay and load on server when external user connects VoIP. 3. Makes debugging more difficult (need managed switch or extra hub to capture SIP traffic from and to ATA). 2. ATA might stop responding; additional hardware is needed to reset it if user is not physically present. 1. It's not the Right Thing! Reminds me of the days when an anti-interface was needed to connect a modem to a DAA. So, how can one avoid these problems, without violating the Terms of Service or the law? It appears that Vonage does not explicitly prohibit use of other devices, although they reserve the right to do so. A sentence taken from http://www.vonage.com/features_terms_service.php says: If you decide to use the Service through an interface device not provided by Vonage, which Vonage reserves the right to prohibit in particular cases or generally, you warrant and represent that you possess all required rights, including software and/or firmware licenses, to use that interface device with the Service and you will indemnify and hold harmless Vonage against any and all liability arising out of your use of such interface device with the Service. However, it appears that Vonage will not supply the needed SIP credentials, and Google searches did not yield a way to get them. Here are some ideas on the subject; I'm a newbie in this area, so please excuse me if they are naive or simplistic. Technically, it should not be hard to obtain the SIP password. For example, an ATA-186 stores its firmware and data in a standard 28F040A flash memory. It is a 32-pin DIP, conveniently socketed. Just unplug it and dump its contents with a standard EPROM programmer. (Search a similar dump of an unlocked unit with a known password to determine where the needed data is stored.) The password in flash can't be encrypted, but it might be obfuscated and/or compressed. In that case, change the password on the unlocked box and see how the dump changes. Of course, this method would not be very useful if the Vonage password is changed regularly. If the carrier will provision an empty (unlocked, reset) ATA, you could just capture the TFTP traffic to get the configuration key, and subsequent encrypted configuration data. Or, get the key from the flash as above. However, I'm worried that the above methods may run afoul of the TOS, or perhaps even the DMCA. Or, it may simply take too much effort to get them to work. So I have been thinking about ways to get good performance without knowing the SIP secret. How about a software proxy that talks to Asterisk, the ATA, and the outside world? I believe that incoming calls could bypass the ATA completely, because no authentication is needed. Is that correct? For outgoing, the proxy would place the call through the FXO device and the ATA. But once connected, it would feed the audio packets from Vonage directly to Asterisk (and also to the ATA to keep it happy). Likewise, audio from Asterisk would be sent directly to Vonage (and also to the FXO device). A further refinement would eliminate the need for an FXO interface, and instead would use the ATA as just an authentication client. For example, you would connect the ATA to that old 14.4k modem. When making a call, you would take the ATA off hook and send it the desired number. You'd feed it the 407 challenge from Vonage, send back the resulting response, and hang up the ATA. The remainder of the process would be completed by Asterisk. Does anyone know if the Vonage SIP
Re: [Asterisk-Users] Direct SIP connection to Vonage service
On Fri, 22 Oct 2004 23:39:56 +0200, Stewart Nelson [EMAIL PROTECTED] wrote: I would appreciate your opinions on the feasibility of these techniques, and also about any other methods that have been tried to achieve direct SIP connectivity. If you are that desperate to use Vonage, then why don't you sign up for the secondary soft-client option which is $15 or so IIRC?! That will allow you to connect Asterisk directly to Vonage, although you pay extra for the privilege. I personally wouldn't bother and I wouldn't want to take my money to a company that uses a business model that I despise. So, vote with your wallet. Don't use Vonage. Use a true VoIP service. And while we are at it, support IAX: Use a provider that offers IAX. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Direct SIP connection to Vonage service
Do you have a list of those providers that use IAX? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin on Asterisk Mailing Lists Sent: Friday, October 22, 2004 4:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Direct SIP connection to Vonage service On Fri, 22 Oct 2004 23:39:56 +0200, Stewart Nelson [EMAIL PROTECTED] wrote: I would appreciate your opinions on the feasibility of these techniques, and also about any other methods that have been tried to achieve direct SIP connectivity. If you are that desperate to use Vonage, then why don't you sign up for the secondary soft-client option which is $15 or so IIRC?! That will allow you to connect Asterisk directly to Vonage, although you pay extra for the privilege. I personally wouldn't bother and I wouldn't want to take my money to a company that uses a business model that I despise. So, vote with your wallet. Don't use Vonage. Use a true VoIP service. And while we are at it, support IAX: Use a provider that offers IAX. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Direct SIP connection to Vonage service
[EMAIL PROTECTED] wrote: Do you have a list of those providers that use IAX? Check the: Asterisk to/from PSTN services section on the wiki page: http://www.voip-info.org/wiki-VOIP+Service+Providers -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin on Asterisk Mailing Lists Sent: Friday, October 22, 2004 4:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Direct SIP connection to Vonage service On Fri, 22 Oct 2004 23:39:56 +0200, Stewart Nelson [EMAIL PROTECTED] wrote: I would appreciate your opinions on the feasibility of these techniques, and also about any other methods that have been tried to achieve direct SIP connectivity. If you are that desperate to use Vonage, then why don't you sign up for the secondary soft-client option which is $15 or so IIRC?! That will allow you to connect Asterisk directly to Vonage, although you pay extra for the privilege. I personally wouldn't bother and I wouldn't want to take my money to a company that uses a business model that I despise. So, vote with your wallet. Don't use Vonage. Use a true VoIP service. And while we are at it, support IAX: Use a provider that offers IAX. rgds benjk Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Direct SIP connection to Vonage service
[EMAIL PROTECTED] wrote: Do you have a list of those providers that use IAX? http://www.voip-info.org/tiki-index.php?page=VOIP+Service+Providers is a good starting point.. Try a search on google, you would be surprised on how many of these will pop... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin on Asterisk Mailing Lists Sent: Friday, October 22, 2004 4:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Direct SIP connection to Vonage service On Fri, 22 Oct 2004 23:39:56 +0200, Stewart Nelson [EMAIL PROTECTED] wrote: I would appreciate your opinions on the feasibility of these techniques, and also about any other methods that have been tried to achieve direct SIP connectivity. If you are that desperate to use Vonage, then why don't you sign up for the secondary soft-client option which is $15 or so IIRC?! That will allow you to connect Asterisk directly to Vonage, although you pay extra for the privilege. I personally wouldn't bother and I wouldn't want to take my money to a company that uses a business model that I despise. So, vote with your wallet. Don't use Vonage. Use a true VoIP service. And while we are at it, support IAX: Use a provider that offers IAX. rgds benjk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Direct SIP connection to Vonage service
On Fri, 22 Oct 2004 16:07:08 -0600, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Do you have a list of those providers that use IAX? check the Wiki ... http://www.voip-info.org/tiki-index.php?page=VOIP%20Service%20Providers in addition to those entries which mention IAX there are also some entries which I know do IAX but it's not mentioned. So you may want to check out the links anf do your own research to get a more complete picture. -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
