similar to: Sip to ATA?

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

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

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

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

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

Hi Does anyone know the clear advantages over using asterisk rather than asterisk@home. Is the home version limited in anyway etc? Many thanks in Advance Scott

What are people's thoughts on asterisk 1.2.26? Any show stopping bugs? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080117/57d1002d/attachment.htm

Asterisk Users, I am thinking about selecting CALLWITHUS as my sip provider. Has anybody ever used them? How was the call quality? DTMF Tones issues? Thanks in advance. -John _________________________________________________________________ Gear up for Halo? 3 with free downloads and an exclusive offer. http://gethalo3gear.com?ocid=SeptemberWLHalo3_MSNHMTxt_1

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

Please chime in if you've ever wished for digium to make a 4-port daughter board with a combination of 2FXO AND 2FXS ports on the same card. When using the 800 series cards, one must either choose 4-port permutations of FXS/FXO, OR one must give up 2 valuable ports. In other words, when you add ONE 100-series daughter board, you give up TWO of your physical ports. Is there a technical

I'm running RedHat 9 with a TDM400 (2FXO, 2FXS). Before I implemented Asterisk, some users were using Bell services to set-up 5 way calling: The user would set up a three way call on one line, switch to the second line, set up another 3 way call and then link the two lines together with the Flash key, thus establishing a 5 way call (the user, 2 others on line 1, another 2 on line 2). How

Hello again Asterisk people. I am running Asterisk 1.42 on an old PowerPC ibook. I have had this deployed for several years now, with pretty good results. Recently I added a callerid service to my landline (qwest). I am using the audiocodes MP114 2fxo/2fxs gateway, which is an outstanding piece of hardware once it's configured (lol). Anyhow, I can see that the gateway is passing

Hello, I connected a Micronet SP5014 2FXS + 2FXO gateway to the asterisk, the problem is i can make call but can't receive calls. If i make a "sip show peers" it shows the micronet is not connected to the asterisk. Does anybody knows how to configure the micronet and asterisk to solve this problem ? Thank you

We are experiencing problems with FXO modules on TDM400P. From time to time they stop responding to incoming rings although they work fine if we use them to dial out. It's been verified at least in two different installations (using different mainboards) in two different locations. The only solution to the problem is to stop asterisk, then unload and reload kernel drivers. The problem appears

I have set up asterisk@home <mailto:asterisk@home> with Digium TDM400P 2FXO/2FXS. I am unable to seize my trunks from either soft or analog phones. Inbound calls result in answer/disconnection. I see the following error code on my asterisk server INIT: Id "s0" respawning too fast: disable for 5 minutes Does anyone have any suggestions for me? I?d really appreciate some help on

I have a TDM-400 from digium with 2FXO+2FXS ports. Any idea on how much power will this drain from the 12 and 5 V connector when all ports are in use? -- Gustavo Felisberto (HumpBack) Web: http://dev.gentoo.org/~humpback Blog: http://blog.felisberto.net/ ------------ It's most certainly GNU/Linux, not Linux. Read more at http://www.gnu.org/gnu/why-gnu-linux.html . -------------

I'm running AsteriskNow 1.7.1 with a OpenVox 2FXO/2FXS card, SPA942 SIP phone and outgoing SIP and IAX routes. When I dial local PSTN numbers from the SPA942 using the FXO channels I observe a 5-7 second delay between when the PSTN number answers the call and when Asterisk connects the call at my end. There's enough delay time that I hear an additional ring after the PSTN number has