similar to: Intermitant delays on call setup.

We are running Suse Linux 8.2 with Samba 3.0.0.1 and Cups as our print server on a network with windows 98 and 2000 clients and HP1300, 2100, 2200 and 4050 printers. We are experiencing an intermitant error whereby the printer on the client machine will set itself to work off line and sometimes freeze up the pc when an attempt is made to print from it. This error does not happen in any consistant

Hi all, I have a problem with using remote MySQL database server with two Asterisk (1.4.17) servers. PhoneA registers with Asterisk#1 using realtime into MySQL on remote server and everything is working fine and when I call Phone A from Phone B (also registered with Asterisk#1) call is established. Problem is when I call PhoneA (which is registered with Asterisk#1) from PhoneC (which is

Hi, Below is my case. phoneA (PSTN) phoneB (SIP) phoneC (PSTN) phoneA --> asterisk --> phoneB phoneA (music on hold), phoneB --attended call transfer--> phoneC phoneA --connect-- phoneC after phone B hangup phoneA type some keys in keypad but phoneC always has wrong dtmf detection. In my case, I would like to know any factor that will cause the wrong dtmf detection.

Just to bug you all (feel free to rant at me), a client wants to set his caller*ID number for outbound calls though us to PSTN. the client is using SIP to us, he can set the caller*ID name fine. if he sets his caller*ID number to anything other than his account number (8440101), the call is dropped into the default context (and then hung up by our dial plan). To get around this i

Hello, I understand that only the destination of a call should do jitter buffering. So, if IAX2/PhoneA calls IAX2/PhoneB through my server (no transfers), PhoneA and PhoneB need to perform their own jitter buffering, and Asterisk will just forward the frames, correct? What happens if the peer does not support jitter buffering, but is close by so there's no need for jitter buffering? My

Hi, I got a weird behaviour in asterisk (original found in 1.8 but it is still the same in 11.15.0). I have three phones communicating via OpenSIPs with asterisk. Phone A dials 100 and asterisk calls SIP/phone-b. Phone B accepts the call. The User on Phone B places the call on hold, dials 200 and, while hearing the dial tone of ringing Phone C, places the handset on hook. Phone B sends a REFER,

I am having a problem with Polycom auto-answer. I have the auto-answer working between PhoneA and PhoneB, but when I try to use the intercom between more then one phone I start having problems. PhoneA dials *3 which calls PhoneB, PhoneC, and PhoneD. All the phones ring, but only one will pick up, the rest will hang up and I get this error on Asterisk: Got SIP response 500 "Internal Server

Can someone help me, im using latest CVS, asterisk and cdr_mysql, when I make a bridge call (using .call files in outgoing/) I always get 'billsec=12' in the cdr, both mysql and Master file even if the call lasted longer, watching the Master file while making a call I see it updated at 12 seconds even while im still 'in' the DIAL app and the call continues on just fine. Iv looked

Hi, I'm looking for a way to reject a call remotely using the Asterisk dialplan. For example, phone A is ringing - I'm at the other end of the room next to phone B, and I want to reject the call to Phone A by dialing an extension. I'm basically trying to reproduce the Polycom "reject" action but through the Asterisk dialplan. Reasons: 1. It would allow me to

I have just started to play with callgroups and pickupgroups.. I updates my * from CVS this morning (about 15 mins ago).. I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations in sip.conf.. I place a call from phoneA to phoneB, then I go to phoneC and dial *8# , the call does not get picked up by phoneC and continues to ring on phoneB.. Have I not configured

Hello all, Not sure if this mail belongs to this users or dev list. Sorry about that. We have the following scenario: PhoneA OpenSER Asterisk PhoneB PhoneC | | | | | | | | | | | | | |

Hello, I have a very odd issue happening, that I hope someone else might be able to give me pointers. I have two different networks running in two different locations, connected by a network vpn. In each network I have a test smb virtual machine. --------- Smb Machine 1: (smbtest1) (remote network) # cat /etc/redhat-release CentOS release 6.4 (Final) #rpm -qa | grep samba

Hi I have looked around but I cant find an answer for this, I randomly get the error 'TDM PCI Master abort' and the system locks up. All I have found so far are a couple other posts on it but no solution. Running fedora core 3, asterisk stable, zaptel stable. Any help will be appreciated. Morgan Gilroy

Oh. So how can I do this? If I write something in PHP, how do I make it output to an Asterisk variabel? I need to set a variable in asterisk to TRUE or FALSE based on the result of the PHP-script. ________________________________ Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Morgan Gilroy Sendt: 8. februar 2006 15:28 Til:

I can't seem to retrieve a parked call. Here is what I do: Call from phoneA to phoneB Answer phoneB Press Flash/Callwait on phoneB Press 700 to park the call A voice says that the call is parked at 701 When I try to dial 701, I don't get connected to the parked call Below is the asterisk output when I tried to park

Frank, Thanks for the reply so what I did in the interim was unistalled Hmisc and chron and used version 3.0-2 of Hmisc which doesn't have the dependency on chron, and reinstalled its version 1.1.4 However I still heave the issue, when I try to run the quantile command on the given dataset. Thanks, ~Lloyd Gilroy, Lloyd (GTI) wrote: > I currently have an instance of R running on

I have asterisk 1.0.2 and a Sipura SPA-2000 (firmware 2.0.6(c) ). Today it started to log "registration failed" at intermitent periods. It registers fine, after a few minutes it can no longer register, then after a few minutes it registers fine again. I am wondering if there is a known issue with either asterisk or that sipura firmware. Victor Perez

Some of the computers in our network keep having trouble connecting to samba I was able to restore the connectivity by placing the computer and IP address in the host file. Immediately after that the client was able to connect. It is my understanding that when Samba is using WINS if the WINS server fails to respond within a certain length of time then these are the types of errors I will get. Is

Hello, I am building an asterisk PBX with some stuff to make a usable VOIP / PSTN Gateway. I use the following devices: SIP Phones from GrandStream for VOIP side OpenLine4 from voicetronix for PSTN Side I am building things step by step with some priorities. I have now a working system able to place and receive calls from/to pstn. Before attempting to bring other functions (like voice

Hi all and thanks for reading. I am experiencing a frustrating issue with asterisk where on some calls the volume suddenly drops to inaudible o completely fades away for a time. If you hold on long enough (20 to 30 seconds) the sound will come back. My asterisk server is on a public IP, and basically acts as a VoIP bridge receiving calls from my customers (all of whom use Grandstream GXW400X