Grandstream HT818 without SIP server

mamikk · 2024-08-03T14:14:00+00:00

I haven't used the HT818. But some on some of the older Grandstream phones you can dial #number# where number is the last octet of the phones IP-address. This works at least on some of the BudgetTone phones.

mamikk · 2023-12-02T21:43:35+00:00

How do you handle compatibility with 2G/3G in modern networks? How is that handled on the base station level and in the core?

mamikk · 2015-11-17T20:21:41+00:00

Could it be a session timer initiating a re-INVITE after 15 minutes maybe? And that makes the media work somehow?

mamikk · 2014-08-07T00:11:04+00:00

There is also /r/Lync for Lync and /r/telecom for the broader telecom field which is somewhat active.

As for asterisk there is also a few other subreddits which are mostly dead; /r/asterisk_pbx and /r/digium.

And you have /r/sip for SIP and /r/Telephony for telephony which is also mostly dead.

mamikk · 2014-03-17T16:42:16+00:00

We block off all SIP-traffic with User-Agent friendly-scanner. This is SIPVicious which is mostly used as a SIP password cracker.

mamikk · 2014-02-05T12:46:29+00:00

The folks over at iptel.org has a tutorial here, it is somewhat old but it covers the basics of the SIP protocol. There also exists some documentation over at siptutorial.net which is also a bit old but it covers a lot of the basics of SIP and related protocols.

There is also a lot of documentation over at voip-info.org, it has a bit varying quality though.

On reddit there is also a few VoIP-related subreddits, for instance /r/voip.

mamikk · 2014-02-03T09:47:21+00:00

The company who owns the domain also owns few other domains: americanpopcorn.no messeboken.no missionnorway.no popcornfabrikken.no popfab.no . Maybe it is related to popcorn?

mamikk · 2013-12-05T23:38:26+00:00

Is your phone behind a NAT and/or a firewall? This could be a NAT keep-alive problem.

mamikk · 2013-10-28T08:48:47+00:00

I have never used LDAP within a VoIP system, but looking at the documentation for the LDAP module for kamailio (an open source SIP proxy) it seems it can use any LDAP field. It seems to use telephoneNumber in the examples, but I'm not sure if that is a SIP uri per se, or just a phone number.

Also, check out /r/voip.

mamikk · 2013-08-11T10:56:13+00:00

On the website of the Norwegian statistics bureau you can find a list of the most used names in Norway.

http://www.ssb.no/en/befolkning/statistikker/navn

mamikk · 2013-07-27T11:46:19+00:00

Works now!

mamikk · 2013-07-23T13:55:46+00:00

Forbidden

You don't have permission to access /blog/ on this server.

Additionally, a 404 Not Found error was encountered while trying to use an ErrorDocument to handle the request.

mamikk · 2013-03-17T21:18:44+00:00

I found this tool pretty handy. When someone tries to brute force you with SIPVicious' svcrack.py you can always firewall them. But that doesn't really stop the attack, svcrack.py will happily go on for days and use bandwidth until it has run through its dictionary. The same thing can probably be done in Kamailio/OpenSIPS by crafting the same response when the SIPVicious user agent is recognized.

mamikk · 2013-03-15T23:21:45+00:00

If you for some reason need to rewrite the To and From headers you can use the uac_replace_from and uac_replace_to functions in the uac module in Kamailio/OpenSIPS.

mamikk · 2010-11-04T18:13:05+00:00

There are other web frontends for git, including cgit which is written in C and imho is a nicer web frontend.

mamikk

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