similar to: breaking into asterisk channel

Hi, we're running Asterisk 1.2.10 and have been randomly being left blank voicemails with long messages that we can't hear. I've searched and searched but cannot find a solution. This is what happens: Internal Server runs Asterisk 1.2.10 where our mailboxes are Incoming Server (behind a firewall) runs Asterisk 1.2.13 and calls are bridged between this server and our internal server.

Yesterday we moved one of our servers to a new IP. We updated DNS and various adapters configured to register to that server registered to the new IP correctly. All seemed to be well. This evening I discovered that with one exception, all of the adapters are getting a SIP/2.0 401 Unauthorized message back from asterisk. The exception is an Innomedia adapter -- Linksys PAP2's and (I

Hi! I am currently working on setting up an Asterisk system, and I was wondering if anyone has worked on an alternate mp3 player to mpg123. We have a library of MP3 files that we would like people to be able to select and play over the phone -- and this will require pause & resume, as well as fast forward / reverse (jump forward / jump back). It doesn't seem like mpg123 can do this. Is

Hi, I'm running asterisk 1.4.22 on a debian server. I have php5 installed and it works correctly command line. When trying to run a php script via AGI, I get messages such as: GI Tx >> I> AGI Rx << #!/usr/bin/php5 -q AGI Tx >> 510 Invalid or unknown command The scripts are completely executable and owned by asterisk -rwxr-xr-x 1 asterisk asterisk Googling is not helping

We've been running an Asterisk-based phone system here in our office for a year and a half, and it's pretty much been running smoothly. One employee who works out of the office has a problem that she can't make outgoing calls on a temporary basis every so often (a few times a day). No one else has this problem, her settings are fine, and she regains the capability spontaneously with no

I've just moved into 3.3v PCI servers and found that my clone X100P cards were lying about the 3.3v supported notch. Can I use a Wildcard TDM400P without any modules as a timer for MeetMe in a 64 bit 3.3v server? Will I still need to plug the hard disk power cable into it? Is there a better cheaper 3.3v MeetMe timer? (Boss doesn't trust the kernel timer.) -HJC

One of my user is praising Skype!!! I cannot figure out anymore what I can improve! This users sip show peers is jumping from 65 msec to 1800 all the time. Of course his voice quality is like a morse code with dashes or dots of connection time. The next minute he calls me via Skype and it works fine !!!! What indicates that there is no fault on his Internet connection!!! He is using his

Hi Chaps, Is there a way to detect/highlight poor quality voice calls going through an asterisk server? Was thinking of picking up a cdr record or some other variable showing poor quality on calls when the internet is having issues. Is there any qos or poor audio quality variables available? Cheers, Taff. ___________________________________________________________ Yahoo! Answers - Got

Hi, Pretty sure I'm missing something simple, but I've seen references to this feature but not found documentation for it: I have a queue set up so that many people are contacted (ringall) when a call comes in. I would like the answering party to confirm that he is a human being rather than cellphone voicemaill by pressing a digit. This is somewhat similar to the 2nd macro example

We need to conference together a call center agent, a customer and a third party IVR and send DTMF tones from the agent to the IVR. MeetMe has been eating our DTMFs so we'd like to try app_conference. Has anybody setup such a configuration in app_conference and how did you configure it? -HJC

The options are not seperated by commas. exten => s,1,Dial(SIP/50,23,r,d) should be exten => s,1,Dial(SIP/50,23,rd) -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of John Klimek Sent: Thursday, June 22, 2006 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]

cpp[BytePerPlane] can't describe the 10bit data format correctly, So we use bpp[BitPerPlane] to instead cpp. Signed-off-by: Sandy Huang <hjc at rock-chips.com> --- drivers/gpu/drm/nouveau/dispnv04/crtc.c | 7 ++++--- drivers/gpu/drm/nouveau/dispnv50/base507c.c | 4 ++-- drivers/gpu/drm/nouveau/dispnv50/ovly507e.c | 2 +- 3 files changed, 7 insertions(+), 6 deletions(-) diff --git

Has anybody created a crontab script to check the health of an Asterisk server? The part I'm struggling with is some sort of IAX "ping" to test the connection to each provider without making a call. -HJC

Would anybody mind if the the following command options where added to MeetMeAdmin? 0 - 9, * and # I'm considering hacking the code to add these commands to play the DTMFs to the specified user as tones and hope that the SIP or IAX channels then work with these correctly. -HJC

[Jul 2 09:31:16] VERBOSE[2682] logger.c: == Refreshing DNS lookups. [Jul 2 09:31:16] NOTICE[2682] dnsmgr.c: host 'outbound1.vitelity.net' changed from 64.2.142.17 to 64.2.142.29 [Jul 2 09:31:23] DEBUG[2711] jitterbuf.c: Attempting to exceed Jitterbuf max 600 timeslots And the calls are dropped. I fixed this by turning off enable in dnsmgr.conf My question is: Do you attempt to

Hi, I am using Polycom 501 and I came across a problem. As soon as I have incominglimit=1 in sip.conf, which is necessary for buddy watching, I cannot transfer calls. On the console it tells me: Call from user '3052' rejected due to usage limit of 1. Can someone please tell me how to get around this problem? (I don't know if this is relevant, but in the phone.cfg file, I have

I have a very, very simple Asterisk setup in my house. I have a Sipura 3000 with a PSTN line connected and one analog phone connected. The [incoming] context looks like this: exten => s,1,Dial(SIP/50,23,r) exten => s,2,VoiceMail(u50@default) exten => s,3,Playback(vm-goodbye) exten => s,4,Hangup As you can see, when somebody calls in if I don't answer in 23 seconds then they are

On 21.11.2018 19:19, Laurent Pinchart wrote: > Hi Ville, > > Thank you for the patch. > > On Tuesday, 20 November 2018 18:13:42 EET Ville Syrjala wrote: >> From: Ville Syrjälä <ville.syrjala at linux.intel.com> >> >> Make life easier for drivers by simply passing the connector >> to drm_hdmi_avi_infoframe_from_display_mode() and >>

Hi all, I'm planning to buy a X100P clone and would like some feedback about this card. Does anyone already used this card? Does anyone recommend it ? or not? I'd appreciate any comments. Thanks. Ronaldo.

Hi, Let's say you have an Asterisk server running. Which parameters would you check to improve service continuity ? I was thinking of : - telco lines status (make sure every is up) - registered hardphones - config files backup (compare live and saved configuration files, if files differ, notifies the administration team) - systems variables (disk and CPU) - log files (trigger an alarm for