similar to: Not Registering Port with VSP.

Hello All, For some reason my asterisk server is not registering a port number with my VSPs. This is causing problems where people are not able to dial in from any of my SIP or IAX VSPs. I do have one VSP that has hard coded my IP and port so I can get incoming calls but this still leaves a problem with my other VSPs. Hose can I get asterisk to register my IP and port? I have been

Hi All, I think I have missed something as I am resisted with 4 VSPs and I can not dial in using any one of them using the corresponding VoIP numbers assigned with the VSP. I can make outbound calls to another VoIP number to the same provider. The weird thing is that I have a DID with a VSP and I have that working fine but try using the associate VoIP number and nothing happens. When

Hi, I have Grandstream GXP-2000 connected to Asterisk, and Asterisk has trunk to VSP for PSTN calls. When ever I place local PSTN call, the landline doesn't hang up right away (40 sec), when I hang up the GXP-2000. The GXP-2000 seems to have problems making international calls as well. Where it hangs up soon as the other party picks up. I have used different IP phones, VSP's and etc.

I'm sure this might not be the correct place to ask and I have done a Google but I can't seem to find anything that says there is a VSP that will work with * in the Ukraine. I have a friend that lives in Kiev and basically want a phone number there to be able to talk to him and have him call me. If anyone has any information on it and they are willing to share please advise.

All, I'm hoping someone can direct me as to why when someone calls my DID Asterisk tries to authenticate the incoming call on my outbound context. If I remove the GoTalk context I can receive incoming calls. Outbound calls work fine while I have the GoTalk context in place. The error I am getting when someone calls the DID is WARNING[16072]: chan_sip.c:8272 check_auth: username mismatch,

Has anyone tried using BroadVoice for VSP? I have Asterisk configured for a home office & I've been trying to decide which VoIP provider to go with for a little while now. I had heard you could get sub $.01 calls but I have not found that to be true yet (not saying it's not possible, I just haven't found it!). Also I'm not sure if BV will support multiple lines. Any

I have a list of VSPs that I use. Some are not able to terminate to different locations. It appears they are returning this error message: Sep 13 00:01:43 WARNING[22093]: chan_iax2.c:6835 socket_read: Call rejected by x.x.x.x: No such context/extension I would like to find out what the dialstatus is on this so I can try a different VSP that is able to terminate the call. Right now I have this

Bear with me this is a bit long winded. I am having some issues making automated outbound calls over Broadvoice from my Asterisk 1.4.2 server. For reference, none of the below issues happen when I make the calls to VoIP phones attached to the Asterisk server. What I am trying to do is call, using a .call file, out via the SIP trunk we have setup, and when the party picks up use AMD to

Help! I have 10 nodes: Ocfs2 1.2.4 RHEL4 U4 (AMD64) Kernel 2.6.9-42.ELsmp While running fsck.ocfs2 I am getting a lot of these. o2fsck_mark_cluster_allocated: Internal logic faliure !! duplicate cluster <bunch of numbers> Then I get a deluge of "Bad magic number in inode while trying to verify the type of directory entry <filename>" Is fsck failing to

On Thu, Feb 13, 2014 at 5:52 PM, Renato Golin <renato.golin at linaro.org> wrote: > On 13 February 2014 13:47, Evgeniy Stepanov <eugenis at google.com> wrote: >> Hm, I see that -funwind-tables on arm-linux-androideabi target >> replaces this "cantunwind" with a proper unwind table. >> Hence http://llvm-reviews.chandlerc.com/D2762. > > If Android is

Due to an issue I'm having with 7.x, and trying to track it down, I spent tonight getting my server setup to allow my to break into the debugger when it hangs, and hopefully dump core ... But, although I *think* I've got it all, I'm obviously missing something, as it isn't breaking ... First ... I'm running a proliant server, and when I connect via SSH to ILO on that

Greg, Chris, and Jay, Thanks! You've given me plenty of info to digest. I really appreciate the responses. Apologies if my list manners aren't up to snuff! Thanks again, Jason

Hi. We have an odd problem with incoming SIP calls. I have attached a SIP debug log, with some asterisk verbosity as well, demonstrating the problem, below. Is this a known bug? Vital stats: - Asterisk 1.2.3 - Sipura SPA-841, SPA-941 phones - Fedora core 3 The problem manifests itself with these symptoms: - an internal SIP extension receives a call from our PRI - the SIP phone answers the

Hi everyone Firstly, apologies for the cross post, I am not sure which list is most appropriate for this question. I should also point out, that I did not install OCFS2 and I am not the person that normally looks after these kind of things, so please can you bear that in mind when you make any suggestions (I will need a lot of detail!) The problem: accessing certain directories within the

Hi everyone Firstly, apologies for the cross post, I am not sure which list is most appropriate for this question. I should also point out, that I did not install OCFS2 and I am not the person that normally looks after these kind of things, so please can you bear that in mind when you make any suggestions (I will need a lot of detail!) The problem: accessing certain directories within the

Hi, I'm experiencing the same problem. We have 2 office locations and the Asterisk server is at one of them. At the other location, all SPA941 access the Asterisk server over an Internet link. All phones are set to "nat=yes" at the remote location. So my problem is that the MWI doesn't work at the remote location. The Sipsak messages are sent properly, but it's sent to the

I am trying to build cross compiler for custom processor (say XYZ) but on compilation it is giving error llvm-build: error: invalid native target: XYZ (not in project) I have tried configuring like these 1. ./configure --target=XYZ 2. ./configure --target=XYZ --enable-targets=XYZ 3. ./configure --enable-targets=XYZ But every time it is not recognising the XYZ processor. What could be the

Hi all, I've just been told from an ex workmate that my VSP (who I used to work for) has put an anti flooding limit of 80 SIP messages per IP per minute in place. I run the phone system for a facility that has a lot of extensions, but would rarely have more than 4 or 5 simultaneous external calls. Am I in danger of tripping over this limit? It sounds dangerously low to me.

Dear All, I like to simulate a physiologically based pharmacokinetics model using R but am having a problem with the daspk routine. The same problem has been implemented in Berkeley madonna and Winbugs so that I know that it is working. However, with daspk it is not, and the numbers are everywhere! Please see the following and let me know if I am missing something... Thanks a lot in advance,

Hi, I am trying to do my first puppet configuration, already installed the puppetserver and client, in this link show my configuration and my puppet structure: http://paste.pocoo.org/show/212227/ But when i run the client side daemon i get this message: info: /Class[main]/Node[basenode]/Class[inittab]/File[inittab]/source: No specified sources exist err: