similar to: Trouble with T38/Dialogic

Hi, I have got to my hands an excellent book by Michael J. Crawley ``Statistical Computing: An Introduction to Data Analysis using S-Plus'' (John Wiley & Sons, Ltd, ISBN 0-471-56040-5). Its beauty for me is in the fact, that it is more of ``An Introduction to Data Analysis'' than ``using S-Plus'', but I guess that it may be of interest for many others. Most of the

Hello, I am using fitdistr function for parameter estimation. When I use fd<-fitdistr(V2,"gamma") I get following error: Error in optim(x = c(0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, : initial value in 'vmmin' is not finite fd<-fitdistr(V2,"weibull") Error in optim(x = c(0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, :

Hello, I am trying to fit distribution for data consisting of 421 readings.It is basically no of requests arrived per minute.It contains many 0 entries as no of requests.When i use fd<-fitdistr(V2,"gamma") I get following error: Error in optim(x = c(0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, : initial value in 'vmmin' is not finite What should I do ? I need

Hello, I am using fitdistr function for parameter estimation. When I use fd<-fitdistr(V2,"gamma") I get following error: Error in optim(x = c(0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, : initial value in 'vmmin' is not finite fd<-fitdistr(V2,"weibull") Error in optim(x = c(0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, 0L, :

Hi List, I am having trouble connecting two * boxes together via SIP. It looks like * is authenticating the hostname, not the username. The Sip.conf looks fine on both sides, but I get: Jun 2 14:44:52 WARNING[2407]: chan_sip.c:6829 handle_response: Forbidden - wrong password on authentication for INVITE to '"+1XXXXXXXXXX" <sip:+1XXXXXXXXXX@206.80.70.56>;tag=as562b672b'

Hello everyone. I've seen postings for connecting asterisk to vonage but I'm still having trouble achieving that. I have a vonage softphone and I'm trying to register to vonage using asterisk. I have not had any luck. I am behind a firewall. I've successfully gotten xlite to connect and work from the same network. When I change the port setting in [general] to 5061, I am able to

Hi, I have trouble getting MOH to work after an upgrade from asterisk 1.4 to 1.6.1.4. The call goes on hold, MOH is started, and then stops right away. Here are the files both of type .raw: Tsunami*CLI> moh show files Class: default File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-2 File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-1 These files

Skipped content of type multipart/alternative-------------- next part -------------- Nov 21 14:17:47 VERBOSE[32580] logger.c: <-- SIP read from 172.25.103.222:5060: INVITE sip:1715;phone-context=exp_net.ascom@ascom.be:5060;maddr=172.25.96.48;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From:

hey * folk, am trying to configure a WellGate 3504A FXS SIP ATA (http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set up two SIP clients in sip.conf as follows: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = default ; Default context for incoming calls [1235] host = dynamic secret = somepass context = default type

I am getting the following information and asterisk 1.0.7 is not answering the call. Any ideas? jerry ------------------ Sip read: INVITE sip:2828;phone-context=cdp.udp@qg.com:5060;maddr=161.49.198.102;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From: <sip:3173241052;phone-context=+1@qg.com;user=phone>;tag=c22da8c0-13c4-429efa71-657b8da-2ed To:

All, I am connecting to a CS 1000 nortel PBX. I can call out, I have limited success with call in. I get debug traffic that a call is coming in but I get the message "Unable to create/find channel". I was expecting that incoming calls over the trunk would be handled from my sip definition and goto the nortel context. It is not. Below is the actual incoming call debug information. I am

Greetings to All! Help to solve a problem: There is an asterisk and two VoIP a sluice (NSGate 800 2FXS 2FXO, NSGate 800 2FXS). In sip.conf they are registered so: [3301] type=friend host=172.222.135.11 username=3301 secret=0000 defaultip=172.222.135.11 dtmfmode=rfc2833 context=it callerid="VoIPGate2Line1" <3301> allow=g723.1 [3302] type=friend host=172.222.135.11 username=3302

Hi eveybody again! I don't want to be annoying, but if nobody can help me with this, I'll have to desist of working with SIP.I have some questions about SIP, as I wrote in another mail. I have a SIP Gateway and I have two phones (an analog one and a DECT one) conected to it.Also, I have two Dlink dg102s with four phones conected to them. The main problems are two. Calls between the

I am attempting to get a planet VIP-150T to register with asterisk 1.2.4. After searching google I've found what appear to be instructions in German, Russian and Spanish. Has anyone perhaps seen this before? Asterisk is kicking back the following error: Feb 14 09:59:32 NOTICE[21765]: chan_sip.c:10851 handle_request_register: Registration from '<sip:101@192.168.100.240>'

I'm trying to debug a NAT issue and I can't make up my mind if the problem is with my Vigor 2800 or Asterisk 1.6.2. I know the Draytek is alleged to suffer from nat 'issues' but I did not have the issue with 1.6.1 - so I'm wondering if something has changed? The Draytek offers 'NAT & Routed' on a single device - so my Asterisk sits on a Public IP, and I have a

Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming calls just fine. However, using outgoing call files the CS1000 is hanging up after I answer the call. I dont know why? Thanks, for any assistance. Jerry my sip.conf entry is: [Nortel] type=friend dtmfmode=rfc2833 username=XXXXXXXXX disallow=all allow=ulaw allow=alaw

Good evening, I just signed up with Nufone and I am able to receive calls with no problem via my 800 number. Outgoing calls are not going through though. My extensions.conf is as follows: [nufone-out] exten => _91NXXNXXXXXX,1,SetCallerID(mynumber) exten => _91NXXNXXXXXX,2,Dial(IAX2/user:pass@switch-2.nufone.net/${EXTEN:1}) exten => _91NXXNXXXXXX,3,Congestion Whenever I try to

Greetings, I have a very basic equal-weight ring-all queue set up in queues.conf: [sales-queue] ;music = default strategy = ringall periodic-announce-frequency = 20 announce-holdtime = no timeout = 15 maxlen = 0 member => SIP/1xxxxxxxxxx at junction_networks,1 member => SIP/1xxxxxxxxxx at junction_networks,1 member => SIP/dude,1 member => SIP/homie,1 member => SIP/fellow,1 But

I need some extensions logic assistance, I'm trying to dial out one of multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only allow 1 call per trunk) and roll over to a second or third depending on that busy status Here's what I've got for a macro thusfar, but it's not working(fails if the 1st trunk is busy) extensions.conf: [globals] trunk_1 => SIP/trunk1

Hi everyone, I'm having a hard time figuring this issue out, we just switched from a T1 PRI to a SIP trunk provider and that's when the issue started. Now when someone forwards all calls on their phone to a cellphone, when a customer calls in, Asterisk correctly calls the cellphone and connects the call, but there is a long delay before the audio starts, basically for the first 6-10