similar to: TLS, was [IANA #1182277] AutoReply: Port Number (3493) Modification

I would like to suggest that if we modify the IANA registration of nut/3493, we do it formally using the official IANA form https://www.iana.org/form/ports-services , and that we agree on the form contents before submitting it. For example, who is the new assignee and who is the new contact? The last thing the IANA services manager needs is to talk to an unstructured crowd of us. Roger On

IANA ups/401 was registered by Charles Bennett for "Uninterruptibe Power Supply" but I have been unable to find any details of any protocol that might have been developed. The IANA ups/401 contact address corresponds to Charles Bennett's employment by Ohio University as an IT technologist for the Chemistry and Biochemistry Departments. I believe Charles Bennet died in March

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote: :On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: :> ;; dialtone in the background isn't there any more :> ;; dialed '305' :> ;; everything from here is exactly as expected. : :OK, I missed this in the first email you sent... Asterisk is playing :dialtone *on top* of the background message the first

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote: :On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: :> ;; dialtone in the background isn't there any more :> ;; dialed '305' :> ;; everything from here is exactly as expected. : :OK, I missed this in the first email you sent... Asterisk is playing :dialtone *on top* of the background message the first

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote: :On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: :> ;; dialtone in the background isn't there any more :> ;; dialed '305' :> ;; everything from here is exactly as expected. : :OK, I missed this in the first email you sent... Asterisk is playing :dialtone *on top* of the background message the first

Hi guys, I don't know if this is the question ... But I have my softphones set up to use ilbc (because I found that it is better for me) and I'm trying to connect them to my provider that provides me termination through g.729. I don't have any T1 or whatever card in my server. But everytime I try to connect them they use gsm ... I don't know why ... So, that's the reason of my

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote: :On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: :> ;; dialtone in the background isn't there any more :> ;; dialed '305' :> ;; everything from here is exactly as expected. : :OK, I missed this in the first email you sent... Asterisk is playing :dialtone *on top* of the background message the first

Hi guys, I don't know if this is the question ... But I have my softphones set up to use ilbc (because I found that it is better for me) and I'm trying to connect them to my provider that provides me termination through g.729. I don't have any T1 or whatever card in my server. But everytime I try to connect them they use gsm ... I don't know why ... So, that's the reason of my

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote: :On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: :> ;; dialtone in the background isn't there any more :> ;; dialed '305' :> ;; everything from here is exactly as expected. : :OK, I missed this in the first email you sent... Asterisk is playing :dialtone *on top* of the background message the first

Hi guys, I don't know if this is the question ... But I have my softphones set up to use ilbc (because I found that it is better for me) and I'm trying to connect them to my provider that provides me termination through g.729. I don't have any T1 or whatever card in my server. But everytime I try to connect them they use gsm ... I don't know why ... So, that's the reason of my

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote: :On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: :> ;; dialtone in the background isn't there any more :> ;; dialed '305' :> ;; everything from here is exactly as expected. : :OK, I missed this in the first email you sent... Asterisk is playing :dialtone *on top* of the background message the first

Hi! Thank you all for the info! But I think I haven't explained my scenario well enough. I am not relaying the calls to SIP. What happens is the following (the scenario is: a call started from an ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the remote party): 1. A CONNECT comes from the PSTN containing a Display IE (which has info sent by the telco that is used for

Andy Wiggin has contribued a Python program that reads http://www.iana.org/assignments/ipv4-address-space and creates a list of reserved subnets suitable for inclusion in /etc/shorewall/rfc1918. The list produced by Andy''s program will be included in the rfc1918 file included in version 1.3.2 (it''s available now from CVS). Thanks Andy! -Tom -- Tom Eastep \ Shorewall -

Hello, I am in the process of enhancing the work of Luiz Angelo Daros de Luca (https://github.com/luizluca/nut-snmpagent). For that matter, nut needs an official IANA enterprise-ID. Currently Luiz is using the enterprise-ID from his employer, which is obviously not a good idea. Does anybody mind if I apply for a ent.-ID for myself and assign nut a sub-ID from my ID given by IANA? Any better

Hi! Thank you all for the info! But I think I haven't explained my scenario well enough. I am not relaying the calls to SIP. What happens is the following (the scenario is: a call started from an ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the remote party): 1. A CONNECT comes from the PSTN containing a Display IE (which has info sent by the telco that is used for

On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote: :On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote: :> ;; dialtone in the background isn't there any more :> ;; dialed '305' :> ;; everything from here is exactly as expected. : :OK, I missed this in the first email you sent... Asterisk is playing :dialtone *on top* of the background message the first

Hi guys, I don't know if this is the question ... But I have my softphones set up to use ilbc (because I found that it is better for me) and I'm trying to connect them to my provider that provides me termination through g.729. I don't have any T1 or whatever card in my server. But everytime I try to connect them they use gsm ... I don't know why ... So, that's the reason of my

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix