similar to: On kernel 3.16.2 : dahdi_rec: Invalid argument

Displaying 20 results from an estimated 4000 matches similar to: "On kernel 3.16.2 : dahdi_rec: Invalid argument"

2009 Jan 16
1
pstn hangs up: MWI no message waiting ??
pstn incoming on a TDM400P, sometimes i* won't answer, going into a loop like this: -- Starting simple switch on 'DAHDI/4-1' [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 18 (Ring Begin)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2 (Ring/Answered)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: Channel 4
2008 Oct 12
3
setup for fax machine
Becasue of all the issues with fax over voip, we want to use pstn for our fax machine, but not dedicate a line just to fax. I'm thinking of having asterisk answer the pstn line, check for fax tones, and route appropriately. In zapata ( chan_dahdi ) set faxdetect=incoming then the dial plan would have [incoming-pstn] exten => fax,1,Dial(DAHDI/1) ; the fax machine exten =>
2014 Sep 13
2
VGA resume & thaw (wake up from S3 & S4) broken - kernel(nouveau) exclusively
On 13.09.2014 07:02, poma wrote: > On 13.09.2014 06:57, poma wrote: >> >> Actually I have nothing to show cause logs are all OK. >> Haha, it seems to me that the bugs become intelligent. >> >> 3.15.10-201.fc20.x86_64 >> 3.16.2-200.fc20.x86_64 >> 3.17.0-0.rc4.git3.2.fc22.1.x86_64 >> nouveau [ DRM] suspending display... >> nouveau [
2008 Sep 05
1
dahdi & tdm400p: no luck
As best i could figure it out, I've installed dahdi and rc4. My TDM400P doesn't answer fxo or fxs. /etc/dahdi/system.conf: loadzone = us defaultzone=us fxoks=1,2 fxsks=4 /etc/asterisk/chan_dahdi.conf: [house-phones] context=internal ; Uses the [internal] context in extensions.conf signalling=fxo_ks ; fxo_ks Use FXO signalling for an FXS chanel dahdichan => 1 ;
2011 Nov 11
2
10.0.0-rc1: dahdi doesn't see card
From asterisk -cvvvvv == Parsing '/etc/asterisk/chan_dahdi.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found -- Automatically generated pseudo channel [Nov 11 17:43:28] WARNING[5756]: chan_dahdi.c:18155 process_dahdi: Ignoring any changes to 'userbase' (on reload) at line 23. [Nov 11 17:43:28] WARNING[5756]: chan_dahdi.c:18155 process_dahdi:
2011 Apr 15
2
1.8.4-rc2: ReceiveFAX fails
On a test fax: -- Executing [s at incoming-fax:1] Set("DAHDI/4-1", "FAXFILE=/var/spool/asterisk/fax/20110415_1825") in new stack -- Executing [s at incoming-fax:2] Answer("DAHDI/4-1", "") in new stack -- Executing [s at incoming-fax:3] ReceiveFAX("DAHDI/4-1", "/var/spool/asterisk/fax/20110415_1825.tif") in new stack
2015 Nov 05
2
DAHDI driver question for custom card
Hi All, Not sure if this is the right mailing list since the dahdi-dev seems not really active, so I'll try here. I'm developing a new DAHDI driver for a custom board. In this card I've implemented the reading of the TDM slots by 2 DMA channels, TX and RX. Each DMA channel has its own callback that decompose the each slot into a per-channel linear buffer useful for calling the
2009 May 23
1
1.6.0.9: Unknown signalling method 'pri_cpe' ??
I'm trying to upgrade from 1.4.24.1 to 1.6.0.9 with a TE120P card. I can't make any connection over the T1. From CLI: ERROR[26017]: chan_dahdi.c:14300 process_dahdi: Unknown signalling method 'pri_cpe' at line 37. cat chan_dahdi.conf cat chan_dahdi.conf [trunkgroups] [channels] language=en ;internationalprefix = 00 ;nationalprefix = 0 context=from-pstn switchtype=national
2014 Sep 07
5
[Bug 83587] New: 3.14.18: FAN control: none / external -- 3.16.2: FAN control: PWM
https://bugs.freedesktop.org/show_bug.cgi?id=83587 Priority: medium Bug ID: 83587 Assignee: nouveau at lists.freedesktop.org Summary: 3.14.18: FAN control: none / external -- 3.16.2: FAN control: PWM QA Contact: xorg-team at lists.x.org Severity: normal Classification: Unclassified OS: Linux (All)
2010 Jun 18
6
Why asterisk down when inet server down?
We have a 10.10.0.0 internal network. The asterisk server - 10.10.10.180 - has a PRI connection to a T-1. Another server is the router to the internet. All phones in the office and the workstations are on the network. Most of the internal phones are aastra 9133i's. Here the network config from a phone: Network Settings Basic Network Settings DHCP [ ]
2014 Dec 21
2
11.5.0: blindxfer problems [Spam score:10%]
Have you enabled DTMF logging and seen the DTMF codes being recognised by Asterisk? I had a bunch of soft phones that I had to change to using ?sip info? for the DTMF signalling as the RFC signalling was not always being recognised. This would cause transfers to appear as if the user had not dialled any digits. On 20/12/2014 20:52, "sean darcy" <seandarcy2 at gmail.com> wrote:
2018 Aug 29
2
getting invites to rtp ports ??
On 08/29/2018 09:42 AM, Carlos Rojas wrote: > Hi > > Probably somebody is trying to hack your system, you should block that > ip on your firewall. > > Regards > > On Wed, Aug 29, 2018 at 9:34 AM, sean darcy <seandarcy2 at gmail.com > <mailto:seandarcy2 at gmail.com>> wrote: > > I'm getting invites to very high ports every 30 seconds from a
2018 Aug 29
3
getting invites to rtp ports ??
On 08/29/2018 11:59 AM, Telium Support Group wrote: > Block a single IP is the wrong approach (whack-a-mole). You should consider a more comprehensive approach to securing your VoIP environment. Have a look at this wiki: > > https://www.voip-info.org/asterisk-security/ > > > > -----Original Message----- > From: asterisk-users [mailto:asterisk-users-bounces at
2011 Dec 26
5
how to listen on different sip port for a device?
I'm trying to allow access to the office from home. But the ip provider (cablevision) blocks udp 5060. I can see the register packets leaving on wireshark, but nothing received by office. Changed to port to 6111 and now the packets show up. In the server I've set port=6111 in the device in sip.conf, but * is NOT listening for 6111: netstat -an | grep 5060 tcp 0 0
2014 Dec 02
2
On Fedora, kernel update resets /var/run/asterisk owner to root.root
On 12/02/2014 02:46 PM, Jeffrey Ollie wrote: > On Tue, Dec 2, 2014 at 1:22 PM, sean darcy <seandarcy2 at gmail.com> wrote: >> >> Or do I >> find a new place to put asterisk.pid? > > Also, if you use the native systemd unit file, you no longer need a > PID file, although you still need /run/asterisk to store the control > socket. > So systemd is taking
2012 Mar 09
2
dreaded one-way audio with nat=yes
I'm trying to move the asterisk server to an Amazon Web instance. We have teliax for our sip provider. I'd like for our DID lines to be connected to a users cell phone. Seems simple enough, but I'm getting the dreaded one-way audio, even with nat=yes everyplace I can think of. The dialplan is real easy: [from-teliax-sip] exten => _j.,1,NoOp("From teliax sip with exten
2014 Dec 02
3
On Fedora, kernel update resets /var/run/asterisk owner to root.root
On Fedora 20, every time the kernel updates, /var/run/asterisk owner is set to root.root. I'm running asterisk under user asterisk. Is there any way to keep /var/run/asterisk as asterisk.asterisk. Or do I find a new place to put asterisk.pid? sean
2014 Dec 20
2
11.5.0: blindxfer problems
On 12/19/2014 09:42 AM, Rusty Newton wrote: > On Wed, Dec 17, 2014 at 1:09 PM, sean darcy <seandarcy2 at gmail.com> wrote: >> I've got a confbridge set up which works if dialed locally: >> >> -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack >> -- Executing [266 at internal:2] SendDTMF("DAHDI/1-1",
2014 Sep 13
4
VGA resume & thaw (wake up from S3 & S4) broken - kernel(nouveau) exclusively
On 13.09.2014 22:58, Ilia Mirkin wrote: > On Sat, Sep 13, 2014 at 4:52 PM, poma <pomidorabelisima at gmail.com> wrote: >> On 13.09.2014 07:02, poma wrote: >>> On 13.09.2014 06:57, poma wrote: >>>> >>>> Actually I have nothing to show cause logs are all OK. >>>> Haha, it seems to me that the bugs become intelligent. >>>>
2013 Feb 11
1
how to join calls - not barge?
I'd like to have an extension "join" a call. That is, C can join A and B, just as if it were an analog extension phone. ChanSpy works, sort of. The problem is that once A or B hangs up, the channel is gone. With an analog extension, C would remain connected with B if A hung up. Can I throw A and B into a confbridge and then add C? Create a new channel that grabs the A