similar to: In/out calls from/to same sip provider

Hi. We are trying to set up asterisk to service a wireless community in our town. We have about 30/40 wireless working nodes each one with a 10.34.x.x/24 subnet for users. Each one of these addresses can potentially have a 192.168.x.x/x subnet. On top, the wireless nodes, themselves, are linked in 172.16.x.x/x subnets. On top of the top, there is internet and cool things for people, like

Hello friends: I am facing cutoffs randomly when negotiating calls. The PBX dials the destination, the provider (softswitch) receives the request *[1]* and sudenly the PBX hangs up the call* [2]* while the provider is still dialing it, as a consequence the remote peer receives a ghost call. Along the atempt I could see six times a messages regarding NAT isuues *[3]* I hope anyone can give me an

Anyone help me here? I am a newbie so be gentle ;-)...... It worked once and then I played with the configs. I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22 I have the 7690 with a SIP iamge (Whatever latest is ) I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server. Here

Hallo Group! My Name is Guenther Sohler and I registred to this group, because I think asterisk could be interesting for me. I have got a small server at home running linux. It does NAT and a Firewall. There is an intranet with my home PC and a hardware SIP phone. This SIP phone registers at mujtelefon.cz Now I got another account at sipgate.at My idea is following: I want to be reachable at

Sorry if this comes in twice. Wasn't subscribed first time :-( Anyone help me here...... It worked once :-( I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22 I have the 7690 with a SIP iamge (Whatever latest is ) I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server.

My phone still says Not-Registered. I have a Polycom SoundPoint 600 SIP phone. Here is my sip.conf file: ; ; SIP Configuration ; [general] context=default ; Default context for incoming calls port=5060 ;added bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ;

Does anyone kmow what these errors mean or how they can be fixed. I'm using asterisk on a Fedora Core 2 box with a TDM400P with 2 fxo and 2 fxs ports. Dec 29 17:17:52 WARNING[6019]: chan_zap.c:5469 ss_thread: Channel Zap/4-1 in prering state, but I have nothing to do. Terminating simple switch, should be restarted by the actual ring. -- Hungup 'Zap/4-1' == Starting post

I am setting up a box with a T100P. Everything is going well. The company I am working with has their one phone switch gear. They provisioned me a E&M Wink T1. Cannot do PRI unfortunately. We chose E&M so we could pass an unlimited number of DIDs to the trunk as apposed to FXS loopstart signaling. I can make outbound calls no problem, but I am having problems with the dial plan for inbound

I am using version: CVS-v1-0-12/13/04-18:46:23 with a TDM400p (2 fxo, 2 fxs ports) and I keep getting errors along with phantom calls: Dec 21 16:02:07 NOTICE[5872]: chan_zap.c:5363 ss_thread: Got event 17 (Polarity Reversal)... Dec 21 16:02:14 WARNING[5872]: chan_zap.c:5434 ss_thread: CallerID returned with error on channel 'Zap/4-1' my analog phone reads caller ID info fine when

I am trying to setup a simple pstn gateway using Asterisk and a X100p card. I have got everything installed using Redhat 9 and am able to load Asterisk. I also configured sip and I am able to connect to the asterisk gateway with Xlite on the windows side. I am able to dial 1000 and get the welcome message. What I am NOT able to do is dial a seven digit local or 10 digit long distance number and

Hi, I have three Xen hosts running Xen 4.0.2 (OpenSuSE 11.4 based). I also have one ''NFS'' server with an NFS export holding VM images and configuration files. Each host has a dedicated LAN link directly to the NFS server. I have another separated NFS export for VM locking (but I had the same issues before). When I attempt to live migrate a VM, it *looks''*like

Hi everyone, I'm trying to send a FAX with the following configuration: Analog FAX machine (OKI) <----->SPA21000<----->LAN<----->Asterisk<--------> PSTN I'm restricted to use passthru mode for faxing, instead of T.38 protocol, because the Asterisk box is running v1.2 and cannot be changed as it is in a heavy production environment. Anyway, it "should"

You are reading the instructions for the STABLE 1.0 version of asterisk and are using the CVS version. Goto the wiki and read the instructions for RealTime. -Matthew ----- Original Message ----- From: "Muhammad Rizwan Khan" <rizwan@advcomm.net> To: <Asterisk-Dev@lists.digium.com> Sent: Wednesday, January 05, 2005 12:42 PM Subject: [Asterisk-Dev] Asterisk with MySQL >

Hi all, I have the following 7 x 7 matrix. ?I am trying to figure out how to label the columns to something more descriptive other than [,1], [,2], etc. I have tried the c(x,y,z,) function, but I get a error returned stating that my vectors need to be the same length. Do I need to convert this to something else such as a list and then repack it? Thanks, Kindra ?? ? ? ? Volume Time ? ? ? ?[,1]

The following fit does not make sense to me, please, correct me if I have a logical error. > moddowsn Call: coxph(formula = Surv(start, stop, resp) ~ sn + matfac2, data = coxsn1, method = "efron") coef exp(coef) se(coef) z p sn2 0.0497 1.051 0.02030 2.450 1.4e-02 sn3 -0.0532 0.948 0.02038 -2.610 9.0e-03 sn4 -0.0410 0.960

Maurice, You say, "knocking my ssh session offline on all terminals and it blocks ssh from being able to connect again. Even restarting sshd doesn't help". Questions: * Is the network stack on the affected machine still active? (Can it reach other services or systems on the network?) * If the network is NOT reachable, does restarting the network stack make a difference? I ask

Anyone out there have a functional DUNDi configuration using SIP for the inter-Asterisk transport? I've gotten it to work with IAX2, but if I change it to SIP it does not pass the call over even though it knows where to send it. Thanks. The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally

Hi folks, man page says "By default, rsync will delete any partially transferred file if the transfer is interrupted" I have (reproducible) a partial file left, if I do CTRL-C source-dir: mounted LVM XFS dest-dir: see source-dir Ubuntu 12.04.1 (LTS) kernel 3.2.0-39-generic command: rsync -a

Hi, Yes InternalExtension is the context and 2093 the extension. Just to explain something odd that?s happening (and I?m very stumped with this) .I think my contexts are definately the reason that I can?t interrupt the menu for incoming pstn calls to choose a submenu: My users register with my sip proxy (SER). Therefore when I create an entry for them in sip.conf I set only one context. Also to

Hi, I''m having some trouble downloading a .csv file from a particular website. The file isn''t part of a url, you need to click on a link in order to get the file sent. I don''t know how to get mechanize to correctly identify that. Here is the link to the file I''m trying to retrieve: <td style="vertical-align: bottom; text-align: center;">