similar to: SIP Re-Invite behind a NAT

Hi, I still have the same problem trying to configure ITSP failover in extensions.conf for a connected PRI. Any comments thoughts or direction would be greatly appreciated. I sympathize with wanting inbound DID failover. If we have a client with multiple DIDs we will spread them across two or three ITSPs so that all inbound connectivity will not be lost if one of them has an issue. I

SetUp: - Asterisk behind a NAT, - Red Hat 9.0 - Asterisk 1.2.14 My Asterisk box is behind a NAT and I have a DiD from an ITSP. I have my dial plan set up so that when outside callers dial the DiD, the call is answered by my auto-attendant. The caller can then select who they'd like to speak to and the call is transferred to the external line associated with that person (usually a mobile

http://bugzilla.mindrot.org/show_bug.cgi?id=982 Summary: scp doesn't work with password authentication when copying from remote to remote Product: Portable OpenSSH Version: 3.9p1 Platform: All OS/Version: Linux Status: NEW Severity: normal Priority: P2 Component: scp

Depending on an R computation I would like to include an Sweave documents in the main Sweave document. How can I do it? So I was thinking .... to use Latex features : \newif\ifpaper \ifpaper \SweaveInput{"my1.Rnw"} \else \SweaveInput{"my2.Rnw"} \fi But how do I set paper to true or false given an \Sexpr ?? \papertrue % or \paperfalse Any ideas? cheers -- Witold

Hello, I problem is in the format of the date, my time series is like this: 2006070100 1244 61 62 2006070101 1221 60 60 2006070102 1214 60 60 2006070103 1194 59 59 2006070104 1182 58 58 2006070105 1178 58 58 2006070106 1176 58 58 2006070107 1173 58 58

Hello, I problem is in the format of the date, my time series is like this: 2006070100 1244 61 62 2006070101 1221 60 60 2006070102 1214 60 60 2006070103 1194 59 59 2006070104 1182 58 58 2006070105 1178 58 58 2006070106 1176 58 58 2006070107 1173 58 58

Hi List I have UAs registered with Asterisk and make outbound calls via ITSP1, everything is fine without re-INVITE. When people call 178, the actual number 112233445566 at ITSP1 network will be called. When UA or called telephone (112233445566) hang up, the call and associated channels are cleared. Sip.conf [general] canreinvite=no nat=no [ITSP1] type=peer host=A.B.C.D

I'm using Asterisk (1.2.14, RedHat 9) but I've been having trouble with SIP re-invites. I have a DiD from an ITSP and when someone calls in, Asterisk plays a menu recording and transfers the call to the external line the caller selects. Since both sides of the call are external, I want to use re-invite to avoid the rtp packets from going through my server after the call is bridged. I

This has two purposes : - cleaner code - reduce the diff between nv30 and nv40 exa for a possible nvfx_exa merge ? The main differences seem to be that nv30 uses rect texture format (and does not support repeat on that). Then there are some minor changes in TX_FORMAT RT_FORMAT and TEX_FILTER usage. And NVAccelInitNVx0TCL look complete different. Tested with: ./rendercheck -t

I have an ITSP we are trying to work with that has an "Unusual" way of working, but that said my understanding of their behaviour is that it is fully RFC compliant. Can someone suggest how I might be able to interoperate under these circumstances: We register fine with them, and send the default asterisk Contact: header of: Contact: <sip:s at x.x.x.x> This then causes ALL

Asterisk 1.2.7 RedHat 9.0 I frequently have the need to redirect calls that come in on a DiD provisioned by my ITSP, back to the ITSP so that they can terminate the call on the PSTN. For example when an external call comes in, I often have to send it to a cell phone. I believe that this is referred to as "hairpinning" the call. I do this by answering the incoming call and then I use

Asterisk 1.2.7 RedHat 9.0 Hi, I've run into some voice degradation problems with IAX2: I frequently have calls come in on a DiD provided by an ITSP. I often have to redirect these calls back out to the PSTN (i.e. to a cell phone). When this happens, I don't want my server in the media path, I want to hand it off to my ITSP instead and let them handle both ends of the call. I've

We are using using what Cisco's Port Address Translation, so that all SIP traffic is done through %EXTERNIP%. ?To any outside box, it should look like the asterisk server is actually on %EXTERNIP%. My SIP packet gets sent to the ITSP with a Call-ID: 2fd557964ca936b66661d72f1328c918@%EXTERNIP% , but the SIP 200 OK reply from ITSP has Call-ID: 2fd557964ca936b66661d72f1328c918@%INTERNIP%. I can

* Kill the A8+A8 hack. Recent enough X servers (>=1.7) fall back to ARGB glyphs for drivers not supporting A8 render targets. * Kill all the global state. It doesn't matter a lot yet but it might if we get multicard working at some point. * Other random clean-ups with no functional changes. Some numbers from x11perf -aa10text -aa24text -comppixwin10 -comppixwin500: * Before, with A

There are a number of them, try Comwave, Voxip or Wiztel. Depends on what you need we may also provide it... email me privately if you're interested. Some provide IAX, some only SIP, H323, & MGCP... -----Original Message----- From: hugolivude [mailto:hugolivude@gmail.com] Sent: Thursday, February 02, 2006 7:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:

https://issues.asterisk.org/jira/browse/ASTERISK-20841 The patch was already posted by someone but then was deleted because of guide lines. Is it really that hard to fix? Since 1.8 there is this problem but nobody seems to care about. Asterisk isnt only used with itsp who dont care about fromdomain. Or are the developers saying, we dont care about people who are using Asterisk in smaller

If a call comes into my Asterisk server on a DiD provided by an ITSP and the dialplan sends that call to another external number throught the same ITSP's network, I don't want the RTP packets to pass through my server once the call is bridged. I have had great success getting this to work using IAX, but I have not been able to get this to work with SIP. The call is bridged OK (media at

I have an Asterisk 11.5 system, using SIP Realtime and operating as a ITSP. One of my customer's endpoints is a NetVanta 7100 PBX system that has a SIP trunk connection to my Asterisk box. The NV 7100 has a public IP on it that doesn't have any NAT between it and my Asterisk system. When the customer transfers a call from one handset to a voicemail box, the NV 7100 sends a RE-INVITE to

Hi, [I sent this to the plyr mailing list (late) last night, but it seems to be lost in the moderation queue, so here's a shot to the broadeR community] Apologies in advance for being more verbose than necessary, but I'm not even sure how to ask this question in the context of plyr, so ... here goes. As meaningless as this might be to do with the `iris` data, the spirit of it is what

-----Mensaje original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]En nombre de asterisk-users-request@lists.digium.com Enviado el: jueves, 02 de febrero de 2006 10:15 Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users Digest, Vol 19, Issue 15 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To