similar to: NAT/SIP solution?

Hi all. I have come to the conclusion that there just isn't anything out there for allowing SIP and NAT to work together nicely. This is rather amazing considering that as far back as March 2000 there are documents describing how to do it. So I've started a really simple SIP and RTP proxy project, SaRP, on sourceforge.net. Yesterday we uploaded 0.2 of the perl based release. This is the

I'm having trouble configuring these services the way I want. Basically I prefer using iLBC before GSM, however Iaxtel only want to talk GSM. It _seems_ that Voicepulse prefers GSM also, because even if I put ILBC before GSM in the "allow=" part of iax.conf, if GSM is mentioned, Voicepulse will use it. If I don't allow GSM Voicepulse will use ILBC. Does anyone know how to

For some reason I now seem unable to control which codec is chosen. The problem happens with outgoing calls to Stanaphone. Even if I chose disallow=all and allow=ulaw as the only codecs it connects with GSM. Has anyone else got problems with these settings? Any suggestions? As I recalled it, such a setup would not establish a call if the ulaw-codec was not offered by the provider. Stanaphone has

Hi guies, I know I know this subject have been The most written subject about VoIP Right... but I just want to make clear, just one time ! If Asterisk is on a Public IP Address and a softphone behind the nat, sip.conf must contains for this phone: nat=yes .... Now if I want to configure my sipphone (X-Lite) placing behing the NAT, it must have in "Domain/Realm" the external IP

Hello, We have 2 grandstream GX 2000 phones behind NAT and Asterisk outside this natted network. We have the issue with calls to these SIP phones - no audio. It is probably the problem with port forwarding on router - but I am not sure how can I forward same sip ports (5004 to 5100) to two phones (nat addresses?)? Any help appreciated! Z. Zivanovic -------------- next part

Normally icy-metaint is a line in the http header, but some stream only sends it in the response data is self. Where can I find such a stream? Is that compliant to the standard? Implementation that also parses the response data http://stackoverflow.com/questions/8621353/textview-not-updating-shoutcast-stream-metadata-in-android-2-2 Thank Stig -- Best Regards / Mvh Stig Christensen stig

I have recently set to * to play with and I'm getting the hang of it.. I've seen various posts about problems where the SIP clients are behind NAT but I anted to ask a slightly different question. At the current CVS can the * SIP client register with a SIP provider (such as ICH) when the * box itself is behind a NAT. I have an ICH account which I can configure fine on an ATA 186 using

To make a long story short, after a Solaris reinstall I needed to access a disk from the previous install. I realize now I should have done a zpool import, but instead I recreated the pool, thinking this would bring my file system back. I destroyed the new pool, but other than that I have done nothing to overwrite data. Is it possible to recover the data from the old pool? --Stig

Check your FW-1 tracker and see if any sip packets are dropped during call initiation. I had this problem and it went away when I upgraded the BT's firmware to the latest (16). Beware, though, that people on the list claim that this firmware breaks functionality of the message button and autoanswer. I haven't checked this yet, cause I can't afford to go back a version. I prefer a

Hi, VoiceGlo is comercial version of Asterisk? :))) loooooooooollllllllllllllllllll Take a loock on http://www.voiceglo.com/ The softphone is IAX :) Best regards, Chris HARIGA Techselesta Inc. http://www.techselesta.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031201/307c10e9/attachment.htm

I know this issue has been covered with at least 2 different patches, and probably a dozen different discussions, however I'm a bit unclear as to what my options are. I have a DSL line coming in with 8 IP addresses going to an OpenBSD firewall doing 1:1 NAT for machines behind the firewall. My asterisk box is one of these machines, and I'd like to allow foreign SIP clients

Hello all, I''ve been running Xen 2.0x for some few months and have written a simple top(1)-like tool - xenConsole - which continously displays all running domains, along with their CPU-consumption, load averages, percentual CPU-/memory-usage, domain uptimes and so forth. A screenshot probably explains it better than me: http://www.gothix.dk/xenConsole/xcon-0.10b1-ok.PNG It requires

> -----Original Message----- > From: Jason Ricles > Sent: Thursday, December 04, 2014 11:23 > To: CentOS mailing list > Subject: Re: [CentOS] Firefox fails to authenticate .mil > sites with New DoDCAC > > I thought DoD used RHEL and not Centos, or did Centos did approved > DADEMS recently? DADMS is a Navy system, but yes Centos is approved for use by DISA. You would

On Linux (and others that define DONT_TRY_OTHER_AF) x11_create_display_inet() will only use the first entry returned by getaddrinfo(). When binding sockets to "ANY" this is fine on Linux since a PF_INET6 socket bound to ANY will also include IPv4. However when x11_use_localhost (X11UseLocalhost) is set, this is a problem. getaddrinfo() will then return an AF_INET6 entry with IPv6 address

Ravi Varadhan and I have been looking at UCMINF to try to identify why it gives occasional (but not reproducible) errors, seemingly on Windows only. There is some suspicion that its use of DBLEPR for finessing the Fortran WRITE() statements may be to blame. While I can find DBLEPR in Venables and Ripley, it doesn't get much mention after about 2000 in the archives, though it is in the R FAQ

Ravi Varadhan and I have been looking at UCMINF to try to identify why it gives occasional (but not reproducible) errors, seemingly on Windows only. There is some suspicion that its use of DBLEPR for finessing the Fortran WRITE() statements may be to blame. While I can find DBLEPR in Venables and Ripley, it doesn't get much mention after about 2000 in the archives, though it is in the R FAQ

Hi, I upgraded my two asterisk boxes today to the latest cvs (up from 5/3/04). These two boxes talk to eachother via sip, not iax. Since the upgrade, I get the error "Failed to authenticate on INVITE" trying to make calls to/from either box. Removing the secret from each box's sip config seems to work but is utterly braindead. Has anyone seen this? - Eric

This is a follow-up question for PR#15052 <http://bugs.r-project.org/bugzilla3/show_bug.cgi?id=15052> There is another thing I would like to discuss wrt how nlminb() should proceed with NAs. The question is: What would be a successful way to deal with an evaluation point of the objective function where the gradient and the hessian are not well defined? If the gradient and the hessian both

> -----Original Message----- > From: Jason Ricles > Sent: Thursday, December 04, 2014 11:42 > > Gotcha, I also work with DoD for Navy systems and was surprised by > that. So you mean if we don't want to pay RHEL licensing fees, we can > use Centos? Since we are paying about $100 per RHEL license. But you will still need a (self?) support plan to be STIG compliant. >

Dear All, About a week ago; I posted a proposal over on the centos-devel mailing list, the proposal is for a SIG 'CentOS hardening', there were a few of the members of the community who are also interested in this. Therefore, I am extending that email to this community; where there is a larger community. Some things that we will like to achieve are as follows: SSH: disable root