similar to: Strange problem with MusicOnHold - works outgoing - works with extension - but not incoming!

I'm trying to get Asterisk to accept incoming calls from budgetphone.nl. When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy tone. I tried X-lite, which worked perfect, so my modem (with nat) probably is not the problem. I did a sip debug and got the following output. Because I'm new to Asterisk I can't get the error why this is not working. To me it all

(this time with subject....) Hello, I?m trying to get Asterisk to accept incoming calls from budgetphone.nl. When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy tone. I tried X-lite, which worked perfect, so my modem (with nat) probably is not the problem. I did a sip debug and got the following output. Because I?m new to Asterisk I can?t get the error why this is not

Hi all. Just as a quote note, can I thank everyone on this list. I find my self finding pretty much every answer I am looking for on here. And a big thanks to all thoughs helping us out. Mass Respect :) Ok, I'm using a SIP provider (SipGate UK) to do my international dialing etc, working great from extension 8 on phones. However some more friends/contacts have started using SipGate also, and

Hello all, I've been trying everything I could find, but I can't seem to get my * server connected to my SIP provider (budgetphone.nl). Here's my sip.conf: [budgetphone] port=5060 bindaddr=0.0.0.0 context=from-budgetphone register => 31307110000:secret@budgetphone.nl/500 type=friend host=budgetphone.nl fromuser=31307110000 secret=secret fromdomain=budgetphone.nl

Hi all, I registered a SIP account at budgetphone.nl/talkin2ya.nl Receiving calls works like a charm, I even redirected my normal PSTN number to the number I got from them so everything ends up in my * server. Before I ask them to take over my normal phone number I wanted to test all of it, so I ordered some calling minutes to test. Now I cannot get outbound calling to work with them. Anyone here

Hi, I am currently trying out the asterisk@home (version 1) release of Asterisk, and I want to configure it as follows: Calls from regular telephony network (PSTN) come in through my VoIP provider over SIP and outgoing calls to the PSTN should be routed through the ViOP provider onto the PSTN network. I thus have no direct PSTN connection, but only a SIP connection. Incomming calls

hi there, i run asterisk 1.4 on my debian machine, which is in my internal 10.x.x.x network, behind my main computer, i cam make call, receive calls, all works fine, with all providers except sipgate.de, there i can receive call and make them, i can hear the other end but they can not hear me, this is only the case with sipgate.de i don#t know how to configure it and thought maybe someone can help

Hi there, I'm having a hard time getting outbound calling to my SIP-->PSTN gateway. I continuasly get the following result in my log files: Jun 5 10:07:50 WARNING[1568]: No such host: t2y Jun 5 10:07:50 NOTICE[1568]: Unable to create channel of type 'SIP' Jun 5 10:07:50 VERBOSE[1568]: == Everyone is busy/congested at this time I make the following context in my

Update on this: I found that by changing insecure = very to insecure = invite, adding the second trunk no longer stopped calls working. I've read the documentation on this switch and still don't see how it applies/is meant to get used. Anyway, with this change in place, the following may help: asterisk*CLI> sip show registry Host Username

Hello, I've found GlusterFS to be an interesting project. Not so much experience of it (although from similar usecases with DRBD+NFS setups) so I setup some testcase to try out failover and recovery. For this I have a setup with two glusterfs servers (each is a VM) and one client (also a VM). I'm using GlusterFS 3.4 btw. The servers manages a gluster volume created as: gluster volume

Dear list, I have a data frame like: > log2.ratios[1:3,1:4] Clone a1 a2 a3 1 GS1-232B23 -0.0207500 0.17553833 0.21939333 2 RP11-82D16 -0.1896667 0.02645167 -0.03112333 3 RP11-62M23 -0.1761700 0.08214500 -0.04877000 how to make it to look like: > log2.ratios[1:3,1:4] a1 a2

> Dear list, > I have a data frame like: > > > log2.ratios[1:3,1:4] > ID a1 a2 a3 > 1 GS1-232B23 -0.0207500 0.17553833 0.21939333 > 2 RP11-82D16 -0.1896667 0.02645167 -0.03112333 > 3 RP11-62M23 -0.1761700 0.08214500 -0.04877000 > 4 RP11-62M23 0.2761700 -0.15214500 -0.05877000 > the 3rd and

My BT100 won't register with my Asterisk server, it always comes back with a 403. I've included my sip_additional (only one to to have the username 2201) and a portion of the sniffer trace (packets 27 & 28). This has me puzzled as I have my SPA-3K working (incoming and outgoing). On my BT100 I get no dial tone, I can't call it (asterisk says the extension is busy) but I can call

Hi I have defined my SIP phones like this ... [Sip1] username=gs1 callerid= "Full name" <1001> etc etc Now, when I do this in a given extension exten => nnnn,1,NoOp(${CALLERIDNUM}) then I get "<gs1>" as callerid and not "<1001>" as defined with callerid= Sure, I could set the usernames to their respective extensions, but I don't want

I'm using the TSA package (along with all prerequisites) to do some GARCH work and for some reason, something which used to work for me has decided to up and stop. The code is as follows, after loading the package: " gs <- garch.sim(alpha=c(1.9,0.1), beta=c(0.700001, -0.0800003, -0.016),rnd = rnorm, n = 400, ntrans=500) gs1 <- garch.sim(alpha=c(1.9,0.1), beta=c(0.7, -0.08,

Hi, I am attempting to setup an asterisk server, current specs: CentOS release 5 (Final) Asterisk 1.4.11 Asterisk-gui checked out from SVN last week I started with a fairly basic setup involving one VOIP provider who provided one dial in number, and a couple of handsets. Config files are below. It was pretty much totally built by Asterisk-gui, except for the fact I had to add

I am messing around with gluster management and I've added a couple bricks and did a rebalance, first fix-layout and then migrate data. When I do this I seem to get a lot of failures: gluster> volume rebalance MAIL status Node Rebalanced-files size scanned failures status --------- -----------

Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D = 31999) -- Remote UNIX connection Verbosity is at least 8 -- Executing [00425298582 at numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582")

Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D = 31999) -- Remote UNIX connection Verbosity is at least 8 -- Executing [00425298582 at numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582")

Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D = 31999) -- Remote UNIX connection Verbosity is at least 8 -- Executing [00425298582 at numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582")