similar to: Open files / socket leak

This file is 13 years old and NV01/03 isn't even supported by Nouveau. This patch remove README.NV1. --- a/README.NV1 2009-08-02 18:19:25.000000000 +0200 +++ b/README.NV1 1970-01-01 01:00:00.000000000 +0100 @@ -1,42 +0,0 @@ - Information for NVidia NV1 / SGS-Thomson STG2000 Users - - David McKay - - 20th March 1997 - -1.

I've a problem. I've some spa3000 and spa2100. Asterisk 1.2.4. Prefered codec g711u in both. Calleng from a fxs of spa2100 to the fxo of spa3000, all works ok. Then I call from a sip phone configured for using g729, to the fxo of spa3000, it also works ok. The problem is that after this, when, making again a new call from spa2100 to spa3000, spa2100 receives only white noise. I suspect a

Dear all I am intrested in making iterative estimation (thro' loop statements) of, say, linear regression model. For this purpose, I have written the following programme and that I have made use of a sample data (viz., exp.txt): ? Programme: ? # Linear regression modelling with sample data (try5.txt) # Repeated estimation through loop statement

Ok, this is highly confusing. hestia*CLI> sip show users Username Secret Accountcode Def.Context ACL NAT 2944030 2944030 oneeighty_start No RFC3581 2944035 2944035 oneeighty_start No RFC3581 sip users (type=friend) are in sip.conf. I have nat=no

I have multiple sipura 2100 boxes connected to my * box and for some reason that i cannot figure out when making a call to one and answering it and then hanging up results in the line be permanently busy (the phone called is permanently busy until * is rebooted). Any idea where to start with this one? It seems to me that either the SPA2100 is not registering the end of the call or * isn't.

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"

Hi all, I have around 100 SPA2100 registered in my provider openSER. I'd like to add an Asterisk registered into openSER, to the network, to deploy voicemail service for those SPAs. Due to administration access levels, I have no access to SER box, so I'm wondering if that possible: - Some foreign user (say A) calls one of my SPA (say B). - B don't answer. So.. B SPA is setted up to

do not know why. Anybody has a clue what could be wrong ? Is this a bug ? [I rebooted asterisk, and now it works.] Regards Axelle. Logs of failed registration: > sip show users Username Secret Accountcode Def.Context ACL NAT IMSI208011234567890 sip-local No RFC3581 IMSI208302141472352 sip-external No

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"

I tried GET /channels/{channelid}/variable?variable=CHANNEL(pjsip,call-id) But it responds with "message": "Channel not in Stasis application" Since I want to get the call-id for a channel not in stasis I guess that won’t work. Similarly, I can’t force the channel through my own code in the dialplan, so the PJSIP_HEADER function won’t work. So it looks like I’ll

On Sat, Jun 17, 2023 at 2:55 PM TTT <lists at telium.io> wrote: > Based on postings it should be possible to get the SIP Call-ID header > value from the ARI. At what point is this value available ? As well, how > do I retrieve that value – something like > > > > GET /channels/{channelId}/pjsip_header?key=Call-Id > > > > But that doesn’t work. >

Hi all, I hava SER with many clients (sipura SPA2100). One of these is an Asterisk which have others clients (sipuraSPA2100). I also have a Cisco GW which give me access to the PSTN. I make calls to all IP phones in my network, but I can't call PSTN numbers. After I dial, I hear 2 ringbacks but at the same time Asterisk says: Called pstn_number@SER_ip_address SIP/SER_ip_address-ec75 is

I have been running Asterisk for years on a machine with a public IP. Most recently, I have been running 1.2.17, from the day it came out, with no (noticeable) problems. Yesterday, I switched over to a new server that is on the same public subnet, one higher than the original server. I built 1.2.17 from source on that machine (as I did on the old server). My firewall on the new machine is

Hi List!! Thanks for the colaboration, especially to Richard Cavanna who gave me the necessary support. I followed your indications and the comunication was better for the test users. The warning indication is no jumping anymore and the voice is not delayed. This is my sip.conf: [general] context=default ;allowguest=no ;realm=mydomain.tld bindport=5060 bindaddr=0.0.0.0 srvlookup=yes

Hello, did you got your issue solved? I am suffering of the same issue.... On 4/28/07, Hadar Pedhazur <hadar@unorthodox.com> wrote: > > I snipped all of the previous data, as I'm trying to "boil down" > this problem to its essence... > > I turned off the firewall for a few seconds, and still got no > audio. For those that will be suspicious, the commands

Hello, I have a Nvidia Riva 128 ZX graphic card. But renouveau need the nvidia proprietary driver to work and there is no nvidia proprietary driver for Riva 128 or Riva 128 ZX. How can I send you informations that can help you to write a free and open source driver for this nvidia graphic chip. At http://users.tkk.fi/~jpakkane/ren/, I found nothing for NV3. But there is some information for NV1

A patch is attached to fix this problem. It removes the deprecated reference to README.NV1 and properly adds src/nv_rop.h Thanks, Rico Tzschichholz -------------- next part -------------- A non-text attachment was scrubbed... Name: 0001-Fix-make-dist.patch Type: text/x-patch Size: 976 bytes Desc: not available URL:

I have two asterisk machines. One of them has a Digium board (server A) and the other is simply using ztdummy (server B). Server A is running on Debian and Server B is running Gentoo. Server A is running Asterisk CVS-Nv1-0-7-06/01/05-01:27:25 and Server B is running Asterisk 1.0.7. The problem I have is that when I try to transfer a call into a meetme room in server B, it simply hangs

Hi all, i have observed, that sometimes all BChannels on my Zaptel Pri device (E400P) will be restarted. The E400P is connected to another pri switch. In the traces from the other side (pri switch) i can see that libpri request for the channelid is 255. Is this a bug or a feature ...? Or, can it be a bug on the other side (terminator switch) ? Have anyone an idea ? Thanks, Thomas.

Based on postings it should be possible to get the SIP Call-ID header value from the ARI. At what point is this value available ? As well, how do I retrieve that value - something like GET /channels/{channelId}/pjsip_header?key=Call-Id But that doesn't work. -------------- next part -------------- An HTML attachment was scrubbed... URL: