similar to: Create a group of SIP acoount for outgoing calls ?

I use a E100P in France with a french operator E1. I can receive calls via the E1 and tranfer them to a VoIP phone, play IVR etc .... But outgoing calls doesn't work at all. I receive a RELEASE COMPLETE just after the SETUP. There is no pb with the operator (the E1 work well with an other Pbx). Here a call trace. Anyone have an idea ? (g1 is my group name for the 30 channels) -- Accepting

When I start my script: * - Creating classes on br1 for upload control ... * - tc class add dev br1 parent 2:0 classid 2:46 hfsc ls m1 576.0Kbit d 2000ms m2 192.0Kbit ul m2 384Kbit ... [ ok ] * - tc class add dev br1 parent 2:46 classid 2:47 hfsc sc umax 1500b dmax 30ms rate 80Kbit . [ ok ] * - tc class add dev br1 parent 2:46 classid 2:48 hfsc ls m2 152.0Kbit ul m2 152.0Kbit

hi i've configured a TE205P on asterisk at home this is my zaptel.conf span=1,0,0,ccs,hdb3,crc4,yellow span=2,0,0,ccs,hdb3,crc4,yellow bchan = 1-15, 17-31 dchan = 16 bchan = 32-46,48-62 dchan = 47 loadzone = it defaultzone = it and my zapata.conf signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master switchtype=national usecallerid=yes hidecallerid=no callwaiting=yes

Hi All. I connect asterisk and definity by manual at www.voip-info.org/tiki-index.php?page=Asterisk%20Avaya. (I just only have E1, not T1 card). I see, that card work (in definity trunk status, and at asterisk == D-Channel on span 1 up -- B-channel 1 successfully restarted on span 1 -- B-channel 2 successfully restarted on span 1 -- B-channel 3 successfully restarted on span 1

Hello...i want to find the empirical rate for type 1 error using fixed trimmed mean. To make it easy, i'm referring to journal given by this website http://www.academicjournals.org/ajmcsr/PDF/pdf2011/Yusof%20et%20al.pdf. I already run the programme and there is no error in it but i got zero for the empirical rate of type 1 error. The empirical rate for the type 1 error given in the journal

Hi all, I'm using asterisk CVS-08/14/03-22 on a box with a digium T1 connected to a channel bank and a digium E1 connected to the PSTN. I get occasional warnings from asterisk: WARNING[37909]: File chan_zap.c, Line 3197 (zt_read): zt_rec: Unknown error 500 This happens mosttimes in a loop like this: [netland_helpdesk] exten =>

Hello, I haven't found errors in your code. I implemented the test in the paper (the first, fixed symetric mean) and it also gives me zero Type I errors, when alpha = 0.05. Try to see the value of min(pv) or to plot the histogram of 'pv', hist(pv) and you'll see that there are no significant p-values, at that level. Anyway I'll continue to look at it, but my first

hi! I have a question of ISDN PRI "calling number" setting for outgoing call. I need the receiver to see the CLI that I have set. Eventough I've set the CLI as below, the receiver keep on getting a fixed number( used with the PRI to receive incoming calls). My ISDN PRI E1 provider says he has not done any restricions on the custom CLI side.

Hi @ all, i set a server to a costumer of mine with a TE207P for use with 2 E1 Lines. I set them together into one group in zaptel/zapata.conf The point is now, the customer has a free-volumina of 60k minutes per month, per line. How can i make a kind of load balancing, that both lines will be trafficed the same way ? I read something about DIAL(Zap/r1/.) for using round robin, and

Yes, I can make a call on that extension from other definity phone, if you mean it. -----Original Message----- From: Ken Godee [mailto:ken@perfect-image.com] Sent: 30 ÉÀÌÑ 2004 Ç. 19:14 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Connecting Asterisk and Avaya Definity By E1. Incoming work, but not outgoing Roman Bessyadovskii wrote: > Hi All. > > I connect

Hi, I've just upgraded by asterisk box from 1.0.7 to 1.0.8 / 1.0.9. I'm running Gentoo, and in the UK, on a BT PSTN line. The box has been running more or less fine for several months. Since upgrading asterisk has been failing to hangup inbound / outbound calls. I've kept my original config files. The sequence of events is roughly: - Place call from a cisco 7940 through the

Hi i test a new equipment on my backbone: a Cisco AS5300 with voice dsp ressource connected at a E1 Voice Link. I want that all call incoming on the cisco 5300 are sent to Asterisk and all Asterisk outgoing call are sent to Cisco AS5300. Actually, i configure the AS5300: isdn switch-type primary-net5 ! voice service voip sip ! voice class codec 400 codec preference 1 g711alaw codec

Hi List, I am from Peru, I have installed an asterisk server in my company with digium card E1 TE120P, I am having issues when i make calls, here the error from my server [Oct 18 09:13:50] WARNING[2377]: channel.c:3232 ast_request: No channel type registered for 'Zap' [Oct 18 09:13:50] WARNING[2377]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Zap' (cause

Nik, I'm not sure that "NOP" is correct, but I'm in the states so I'll to defer to someone who knows E1/PRI. When I run zttool I have "OK" under the alarms. Is there a way you can call the telco and confirm the settings? Make sure that framing, coding and D channels are set up on their end the same way you're set up. As for asterisk, here's what I get

Dear list, I have been studying R and I would like the aid of more experienced to solve the problems of the analysis below: r = gl(3, 8, label = c('r1', 'r2', 'r3')) e = rep(gl(2, 4, label = c('e1', 'e2')), 3) y = c(26.2, 26.0, 25.0, 25.4, 24.8, 24.6, 26.7, 25.2, 25.7, 26.3, 25.1, 26.4, 19.6, 21.1, 19.0, 18.6, 22.8, 19.4, 18.8, 19.2, 19.8, 21.4,

Hi it's possible to use a Digium TE110P Single T1 / E1 PCI Interface for supply a E1 link to a old PABX ? Thanks

Hi, I need little help with setting up new server i recently bought TE110P/E1. I have 1 PRI and I will be using Soft phones on the agents site. This is 1st time for me to work with Zaptel card and i am having some problem with that. if any one call help me with this or care to share extensions.conf with zap>sip sip>zap with me will be great. I want to be able to make call through SIP

The subject says it all. Is it possible, code wise, configuration wise, at all - to reverse the order in which the available zap channels are used for *outgoing* calls? Code wise, I looked at the channel structure and it appears as though there is only a next pointer, not a previous pointer, so to 'easily' to this in the code would require a change to the code that reads in zapata.conf?

Dear all, I connected a normal phone line to the FXO port but the call is not being processed. The following is the output to asterisk console when I dial 9150 "9 is the prefix I configured and 150 is a local service in to know the current time" *CLI> -- Executing Dial("SIP/4444-d365", "Zap/1/150") in new stack -- Called 1/150 -- Zap/1-1 answered

Dears members of R list, It would like that a more experienced statician in R helped me to complete the analysis to follow: r = gl(3, 8, label = c('r1', 'r2', 'r3')) e = rep(gl(2, 4, label = c('e1', 'e2')), 3) y = c(26.2, 26.0, 25.0, 25.4, 24.8, 24.6, 26.7, 25.2, 25.7, 26.3, 25.1, 26.4, 19.6, 21.1, 19.0, 18.6, 22.8, 19.4, 18.8, 19.2, 19.8, 21.4,