similar to: Trouble getting sound from a call

Displaying 20 results from an estimated 100 matches similar to: "Trouble getting sound from a call"

2007 Oct 11
0
Alert_INFO x2 => 400 Bad Request
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good evening, I have something strange, when I add an ALERT_INFO variable to a ring group, the invite generated contains 2 lines with Alert-Info and my phones return a 400 Bad Request... I've checked in my config files, there is only one line with Set(__ALERT_INFO..... Any idea?? PS: I'm using Asterisk Asterisk 1.4.13-BRIstuffed-0.4.0-test4
2011 Mar 28
1
DTMF input while waiting in queue...
Hey all! I'm trying to figure out how to have a queue accept an inbound caller's key press to action on. At first I'm just trying to implement a "Press 1 to leave a voice mail" announced and at any time in the queue, the user can press 1 and go to the queue's voicemail. Later I'd like to have it accept "Press 1 if this is an x issue, press 2 if this a y
2002 Jun 28
0
troubles with PDC
Hello everybody, I hope somebody can tell me what is going wrong and how it can be fixed. To be honest: I have walked through the test document, and smbd and nmbd seem to work quite properly. However this is the case: Recently one of our smb-servers was suffering from a very bad HD. So I took another and installed linux on it (Slackware 8.1) and took most of the config files (among which
1998 Nov 23
0
Freebsd + NT/Nt-Client am Linux-Server
Pam_SMB allows Linux clients to validate their passwords against an NT PDC, so the only thing you have to do is set up the accounts on the Linux side with an '*' in the /etc/passwd entry. This can be done using a list of users: #!/bin/bash for i in `cat myuserlist`; do /usr/sbin/adduser -p '*' $i with various other command line options, such as "-s
2005 Feb 03
1
Multiple mailbox on the same SIP extension
I'm wondering if there's a way it will show on the phone when there's a new message. Here's what I'm trying to do : in my extensions.conf when someone call from a PSTN line on my TDM04B card they have a choice. When someone press 1 for sales then I have 3 phones ringing at the same time. Each phone as already there own mailbox because if someone know there extension
2009 Feb 23
1
Inbound call to IVR drops after 21 seconds?
Does anyone know why? ThePBX*CLI> -- Executing [310-456-7890 at from-trunk:1] Set("SIP/202.101.202.101-b763ce60", "__FROM_DID=310-456-7890") in new stack -- Executing [310-456-7890 at from-trunk:2] ExecIf("SIP/202.101.202.101-b763ce60", "1 |Set|CALLERID(name)=310-456-0987") in new stack -- Executing [310-456-7890 at from-trunk:3]
2008 Dec 26
2
[Bug 19299] New: urxvt does not render fonts with background
http://bugs.freedesktop.org/show_bug.cgi?id=19299 Summary: urxvt does not render fonts with background Product: xorg Version: git Platform: x86-64 (AMD64) OS/Version: Linux (All) Status: NEW Severity: normal Priority: medium Component: Driver/nouveau AssignedTo: nouveau at lists.freedesktop.org
2002 Apr 27
1
rsync md4sum code.
G'day, I've been working on a Python interface to librsync and have noticed that it uses md4sum code borrowed from Andrew Tridgell and Martin Pool that comes via rsync and was originally written for samba. Is there anything special about this code compared to the RSA md4sum code that can be found with libmd <"http://www.penguin.cz/~mhi/libmd/">? Python uses the RSA
2010 Aug 26
1
Timecondition fallthrough on 2nd GSM Modem, First modem and ZAP's are all fine
Hello, we have an asterisk (1.4.21.2) with ZAP and mISDN channels, the mISDN are 2 incoming GSM Modems, each with 2 simcards. No, the mISDN line one and two are fine, but when I get a call on line 3 something with the time is wrong. Timeconditions fall through to off-hours even if the time of the call is clearly inside business hours, here a log excerpt: [Aug 26 11:04:36] VERBOSE[3112]
2008 Jul 08
0
Trouble with faxing using iaxmodem / hylafax
Hi all, I have just setup a trixbox system and I am implementing hylafax/iaxmodem solution for the faxing. When i send a fax to it by phoning in listening to the IVR and manually pressing start to initate the fax, the call gets picked up correctly as a fax and everything works well. When I try sending a fax by entering the phone number and pressing start to initiate dialing it sounds like
1998 Oct 26
0
SAMBA digest 1853
samba@samba.anu.edu.au wrote: > > SAMBA Digest 1853 > > For information on unsubscribing see http://samba.anu.edu.au/listproc > Topics covered in this issue include: > > 1) Re: long winded printing LARGE files soloution > by Heiko Nardmann <h.nardmann@secunet.de> > 2) Samba replacing NFS > by Jonathan Peterson
2009 Jul 03
2
normalised curve fitting with error bars
Dear List, My data consist of nine columns and about 50,000 rows. It looks like this. -9.0225 3.46464 2.80926 -0.3847 3.73735 1.1058 -2.98936 1.38901 -8.1846 -2.4315 -5.1189 1.8225 3.3798 1.7874 4.693 -3.9286 1.4266 5.7849 -3.4894 -4.0305 3.7879 3.5195 2.9186 2.8685 -6.126 4.978 4.9381 4.5282 3.62558 -3.0455 4.6518 1.39746 0.68652 3.5708 -3.6404 -4.2963 -1.3183 0.6752 -4.0382 -2.5386
2009 Sep 18
0
Blind Transfer Won't Hangup
I'm using FreePBX 2.5.2.2 with Asterisk 1.6.1.4. If I make a call and then decide to blind transfer them using ## my side of the call is not hung up. Instead it sends me to voicemail. If somebody calls me and then I blind transfer them with ## I am hung up on as expected. I called from 8678 to 28688. I then transferred the call to 8532. Asterisk acts like it wants to hang up, but then
1998 Nov 23
1
SAMBA digest 1883
> I am an analyst for a large corporation and am interested in using Samba > in > a project for mine. I need one piece of the puzzle answered for me > though. > Can anyone tell me what is the maximum size of a file system can Samba > see. > I'm talking in Terabytes. If anyone could answer this for me, I'd truly > appreciate it. Could you send any responses to
2007 May 23
0
IVR Loop on invalid input
We are running 1.2.14 with an IVR in the dialplan. If I connect to the IVR with a SIP phone (Polycom or Xlite) and press a couple of digits very rapidly (I found this with 33 on a sticky keypad) which are an invalid response, Allison will go into a loop saying 'I'm sorry, that is an invalid response, please try again.' over and over. This does not happen with a commercial
2006 Nov 08
1
Delay between DTMF Down & Detected Digit
Good Morning, I've recently gotten Asterisk installed and configured our IVR using FreePBX. Things seem to be going well except a few of our inbound callers are ending up in the wrong place when trying to connect to a specific extension. The example I had this morning was someone trying to call extension 212 and getting connected to the Sales queue which is option 2 on the IVR. I looked in
2006 Nov 05
1
asterisk DTMF detection
Hi, Hi All, I've just delved into the world of asterisk and I'm having a few dtmf issues. Internally, amongst sip phones, dtmf is fine. Externally, if you ring from a GSM mobile, DTMF is fine, however if you ring from a standard phone, DTMF fails to register. I am attempting to use a quad port HFC-4S Beronet Card. I've been searching the web most of the last week and
2003 May 08
5
MD4 bug-fix for protocol version 27
Hi, while implementing the rsync protocol in one of our projects I found that the current CVS version still has a MD4 bug. I'm using the FreeBSD libmd implementation and I still had checksum mismatches with protocol version 27 for files whose size was a multiple of 64 - 4 ( - 4 due to checksum_seed). A patch for todays CVS version is attached. Someone should also review the clean_fname()
1998 Mar 05
14
Browsing
I am configuring Linux (2.0.33) as a local router. I am wanting to use Samba (1.9.18p3) to "manage" the browse lists on my network so that computers on each subnet can browse resources on the other subnets. My local network consists of Win95 and NT40 workstations with NT40 servers. My test configuration consists one token ring segment on one side of the Samba box (network .192) and
2007 Dec 22
0
Dead Incoming call - Sangoma A200
Hello List, I am having a strange issue with a trixbox system we installed for a client, and I would appreciate any help on this one. The issue is that occasionally when they go to answer an inbound call from the Sangoma A200 - there is no one there, and they are presented with dial tone. The calling party is hung up. A bit of background: The client actually has two systems install (one at