similar to: SIP INFO vs RFC2833?

worked came to one ring only now. Thank you very much. If I use TE410 or TE405 instead of X100P. do it make that first ring disappear? Shakil -----Original Message----- From: tony@softins.clara.co.uk [mailto:tony@softins.clara.co.uk] Sent: Tuesday, April 20, 2004 12:27 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Auto Answering PSTN --> Asterisk using X100PCard In

Dears; I am having same problem, that when I place a call from the H323 end point (even if it is not added in the ooh323.conf), then asterisk handle the call and play the wave file in the default context. Also I added endpoint to the ooh323.conf and same thing, it keep goes for default context whatever the context placed. My Asterisk vesion is 1.4.25 My Asterisk add-on version is: 1.4.8 What I

I'm not sure if this is a -users or a -dev question, since the answer probably comes down to something in the code. I'm running the latest CVS-STABLE, and am subscribed to PSTN service using IAX2 via Voiptalk in the UK. I've just been alerted by a customer that the sending of DTMF from my asterisk box to a remote PSTN user doesn't work, although it used to. To test it, I have

I've just noticed a strange behaviour with a MeetMe conference. I have a pair of phones (GS BT102) on my desk, and dialled both of them into a conference on speakerphone. If I spoke or made a sound, I heard it replayed from both speakers together a split second later, as expected. I went away for about 15 minutes, leaving the conference running. When I came back any sound I made came back

Has anyone else experienced a problem with the LED for span 1 on a TE410P or TE405P? I had a TE410P on which the span 1 LED would not light red, but once the span was connected, it did correctly light green. I RMAed the board to our UK distrbutor and received a replacement. However, the replacement board displayed the same problem! Wondering if it was related to the computer I was putting it

Does anyone know how a program, script or shell user can best determine whether the machine is running on bare metal or is a VMware guest? Cheers Tony -- Tony Mountifield Work: tony at softins.co.uk - http://www.softins.co.uk Play: tony at mountifield.org - http://tony.mountifield.org

Ronald, Grandstream products have a one year warrantee. If you don't have any luck with Pulver, contact us and we can probably get your phones exchanged. Please don't assume that your experience with Grandstream is typical. We sell a lot of these phones and the overwhelming majority of the purchasers are very happy with their units. The quality has improved tremendously over the last

I want to use fail2ban on CentOS 6 to monitor Apache with the standard default logfile format ("combined"). Has anyone here succeeded in doing so? The format has the IP at the start of the line, followed by two dashes (if no authentication) and THEN the timestamp. What I've read on the fail2ban wiki seems to say that the timestamp must ALWAYS be at the start of the line, followed by

Has anyone here done any work on clustering Meetme conferences over multiple Asterisk boxes? The scenario I am thinking of is where there are two or more boxes connected to a set of PRIs that all answer to the same PSTN number, and where it's not possible to know in advance on which box a call would arrive. So it would be possible to have some calls on one box and some on another, that should

[apologies if this comes through twice - the original doesn't seem to have shown up even after 16 hours] In the handling of agents, when using AgentCallbackLogin, a call placed to an agent needs to be accepted by the agent pressing the '#' key. I'm trying to replicate that kind of operation in a non-agent scenario: I want to call Dial() from my dialplan, play an announcement to

I have an Asterisk system running on T1 PRI trunks using a TE405P. It seems to be running ok, but one thing puzzles me. Every so often I get a raft of messages like this: -- B-channel 0/1 successfully restarted on span 1 -- B-channel 0/2 successfully restarted on span 1 ....... -- B-channel 0/22 successfully restarted on span 1 -- B-channel 0/23 successfully restarted on span 1 I could

I am seeing some curious behaviour with a 1.2.32 system, which I do not understand and so can't work out how to fix it. I have a PRI routed to context default. Here is the complete default context: [default] exten => _9X.,1,Dial(IAX2/m1peer/${EXTEN:1}) exten => _20XX,1,Dial(IAX2/sipeer/${EXTEN}) exten => _X.,1,Dial(IAX2/m1peer/${EXTEN}) exten =>

I'm just setting up a SuperMicro system which has twin SATA disks on an Intel IHC7 RAID-capable controller. The system came with Fedora 5 pre-installed, which I will be removing and replacing with CentOS 4.5. But before doing so, I've been having a look at how the original vendor configured it. When I've built systems previously, I've disabled any RAID controller and used kernel

Hi, I have a project coming up which will need to interface Asterisk to E1 trunks in the UK. I have a couple of questions which I hope someone can answer, or give me some pointers: 1. If I want two E1 trunks, is there anything to choose, performance-wise, between using two ports on a single TE405P, and using two E100P cards? 2. How can I test the E1 operation in the lab, which doesn't

I have been writing a lot of AGI programs in C with good success. I would like somehow to have an AGI program continue in the background while the pbx execution returns to the dialplan and continues. Is this possible? I was thinking that perhaps I could fork or create another thread within the AGI prog. The reason I want to do so is in order to monitor external information (e.g. credit limit and

Roberto estoy en uruguay en estoos momentos. Recien lllego el miercoles El mar 31, 2013 1:59 p.m., <asterisk-users-request at lists.digium.com> escribi?: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users

Does anyone here know anything about GEN-GEN analogue circuits, also known as Manual Ring-Down (MRD)? Apparently they are widely used in Hoot'n'Holler systems for financial dealer-boards. I have been asked to try and interface to such circuits, and have been having great difficulty locating any specifications for the interface. Apparently, they are always-on 2-wire analogue circuits with

I have used Grandstream phones for years, and have just started testing a Cisco 7940 (with SIP firmware 7.4). I have found something puzzling and don't know whether it's just a limitation or something I haven't done correctly. The Asterisk server is directly on the Internet with a public IP. The phones are on a private LAN with a NAT router to the Internet. The sip.conf entries for

I have today reported a bug with the latest channel.c (1.134) that affects music-on-hold for the first user in a MeetMe room when calling from a Grandstream BT102. The music is broken up about 5-10 times a second. It doesn't happen when calling from Firefly. It is also fine on both clients with 1.133 of channel.c. I am using the ALAW codec. Mark at Digium can't reproduce the problem,

I have a small number of boxes in different locations, and currently have a fairly crude cron job running on each, which does a ping of one or more of the other boxes, and if the ping fails, it emails me to say the other box might be down. It then emails me again the next time the other box appears to be up. Of course, this can't distinguish between the remote box really being down and there