similar to: Site to site VPN problems

Hi All. Does anybody know if Google/Android has an API I can sign up for that will allow us to query the caller ID and find out if it is spam or a robocaller? I ask because we've had increase in spam calls and I'd like to simply play dead air or something really annoying. Thanks all, Alex -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello Asterisk-users, [2013-08-18 05:56:29] NOTICE[17089][C-000000a8] chan_sip.c: Failed to authenticate device 390<sip:390 at xx.xx.xxx.xxx>;tag=2762c06e [2013-08-18 05:56:34] NOTICE[17089][C-000000a9] chan_sip.c: Failed to authenticate device 390<sip:390 at xx.xx.xxx.xxx>;tag=7b909220 I keep getting messages like this where the IP, xx.xx.xxx.xxx, is my own IP. How do I figure

Hello, 2 questions: 1. I see from the archives that Vista support requires downloading an updated TAP driver from OpenVPN.net. I have just downloaded tinc 1.0.13 and was not able to get the TAP driver to work on my Windows 7 computer. Should I assume that 1.0.13 doesn't have the most current TAP driver and I should use the one I downloaded from OpenVPN ? There seems to be

Hi All, I have polycom phones setup in an office connected to a cloud asterisk server. The polycom phones can call out just fine - audio just fine. However a call coming into the cloud asterisk answers fine - get the autoattendant, enter the extension and the polycom does not ring. The CLI shows that the correct SIP extension is being Dialed (SIP/524) Looks like I'm getting a 401 permission

I was quite surprised to find a message in my in box from SellVOIP a day or two ago. It indicated I was running out of credit which was a surprise as I thought they'd gone under a large number of months back. So I ran upstairs, added their entry back to sip.conf, uncommented a couple of lines in extensions.conf and I'm again using sellvoip to make outgoing calls. The reason I was

The Asterisk Development Team would like to announce the release of Asterisk 15.2.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.2.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

I get this when my brother in law tries to call in from his box to mine. WARNING[4855]: chan_sip.c:12675 check_auth: username mismatch, have <100>, digest has <s> or after changing the register line: WARNING[4855]: chan_sip.c:12675 check_auth: username mismatch, have <100>, digest has <199> I have done everything I can think of and still failure. Currently the

Yesterday I upgraded my Zaptel to 1.2.26 or I think that was it, the latest 1.2 version at downloads.digium.com. I have a Digium 4 card populated with 4 red FXO cards using channels 1,2 and 4. Channel 3 is not used. It's been working fine for a few years. After upgrading to 1.2.26 calls stopped coming in on channel 1, Channel 2 still worked fine and I could get dialtone and make calls

Hello Asterisk, Would someone be kind enough as to add the issue: grp_lock error when compiling against pjproject and solution: delete the rogue install in /usr/local/include To the WIKI page about installing pjsip. I tried to update the WIKI but don't seem to have a way to do it. I know it's not supposed to happen and I know what I did wrong, but it's hard to imagine

Hello Asterisk, I've been running a CENTOS 5 box with Asterisk 14 and am trying to move to Asterisk 17.2 on a new Fedora Server 31 box. I built Asterisk from Source as I've always done and copied all the configuration files and other stuff from the old box. Everything comes up as expected and it all seems to work except I have one way audio. I'm still using SIP, not pjsip. As soon as

On 12/14/19 11:29 AM, Greg Troxel wrote: > sean darcy <seandarcy2 at gmail.com> writes: > >>> There is also the ObiHai OBi202 with an OBiLine, which provides an FXO >>> port remoted over SIP. (I am not sure if this is discontinued.) >> >> "FXO port remoted over SIP"? >> >> I have an analog phone system. I can use the obi202 to

I show the file size as 19M on the FTP site. 95MB is the gunzipped size. I suspect something unzipped it as it downloaded, I've seen browsers do that. I also pulled the file personally, and found the sizes lined up: ira at ira-t430:~/Downloads [/dev/pts/1](64/0)$ ls -la samba-4.1.19.tar.gz -rw-rw-r--. 1 ira ira 19558250 Jun 23 20:16 samba-4.1.19.tar.gz ira at ira-t430:~/Downloads

On 12/14/2017 10:18 PM, Leslie Zhai wrote: > Hi GCC and LLVM developers, > > I am learning Register Allocation algorithms and I am clear that: > > * Unlimited VirtReg (pseudo) -> limited or fixed or alias[1] PhysReg > (hard) > > * Memory (20 - 100 cycles) is expensive than Register (1 cycle), but > it has to spill code when PhysReg is unavailable > It might be

Hello Asterisk, Back in 2009 I built a small Intel Atom based computer running Centos 5 for my asterisk system. 5 phones, 2 people 1 POTs line and six or so SIP numbers. So basically no load. I'm feeling like it's time to build another machine. It's probably silly, but it's been six years and I can't upgrade the OS which is falling behind. I'd likely just put

Thanks for the quick reply, Guus. Well the debug helped a little bit. So did re-reading the manual again. I think I got further ahead, but unfortunately I still cannot quite get to complete the connection between the two nodes. It looks like the two nodes are communicating - they seem to be able to exchange RSA keys & MetaKeys, but for some reason, they start trying to exchange data on

I'm filtering that is being deliverd to postfix mail server with amavisd-new . I want spam with spam f level 1 - 8 to ad a tag any everything above to be delete is this posebol? If yes how? Met vriendelijk groet, Bas van Dikkenberg GISkit bv BFVD1-RIPE Tel: +3130-6340430 Fax: +3130-6342433 Prive Tel: +3130-6372769 Mob:

[ add Ira ] On Wed, Jun 26, 2019 at 5:27 AM Christoph Hellwig <hch at lst.de> wrote: > > The code hasn't been used since it was added to the tree, and doesn't > appear to actually be usable. > > Signed-off-by: Christoph Hellwig <hch at lst.de> > Reviewed-by: Jason Gunthorpe <jgg at mellanox.com> > Acked-by: Michal Hocko <mhocko at suse.com>

The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1. This release marks the beginning of the testing process for the eventual release of Asterisk 1.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any