similar to: trunk=yes with recent CVS head problems

I've been having 'gappy' audio problems with nufone for about a week now but I think I've nailed it down. Setup: office* - iax2 - colo* - iax2 - nufone office* and colo* are identical physical hardware (Xeon 2.8, dual ethernet, solely used for Asterisk) -- they are joined together through their second ethernet ports over a dedicated 2meg SDSL link. One hop between office* and

I am trying to install the gls package (a wrapper for GNU scientific library special functions) package under Ubuntu 5.10. I have gls-bin (the debian GNU Scientific Library binary package). When I try to install the R package, I receive the following. > install.packages("gsl",dependencies=T) Warning in install.packages("gsl", dependencies = T) : argument

I have two questions for the group. One is very concrete, and is dangerously close to a "please do my homework" posting. The second follows from the first one but is more general. I would welcome the advice of experienced R users. As for the first one: I have a data frame with two variables X Y A, chris D, chris B, chris B, chris C, andrew E, andrew C, andrew B, beth

Martin and I will soon release a new version of the Matrix package with a modified API. This will affect the authors of any packages that use calls to the C function R_GetCCallable to directly access C functions in the DLL or shared object object in the libs directory of the Matrix package. (If you didn't understand that last sentence, relax - it means that you can ignore this message.) We

I just upgraded to RC1 from a two-three month old CVS , and noticed that during IAX2 calls to my service provider there are periods (usually less than 10 seconds long, minutes apart) during which the caller can not hear me, but I can hear the caller fine. Inter-office calls (SIP-to-SIP) does not appear to have this issue. Has any other users experienced this? Marcus Adolfsson TreoCentral

Hi, With reference to Andrew Kohlsmith's problem with calls over IAX2 going dead after 1minute 5 seconds: This is caused by a bug in the "optimized bridging" code in chan_iax2.c interacting badly with the IAX2 jitter buffer. This problem only affect calls where: 1) There are 2 or more servers doing optimised bridging between the end-points of the call. ie a call like A-end

I have a simple little asterisk setup: FXS is a PhoneJack PCI, FXO is an X100P. I have a regular old cordless phone plugged into the FXS port. I can dial anything and it's picked up properly by *. I can call FWD and it picks up the DTMF properly. I call out the FXO but nothing I call through there can hear the DTMF clearly. Bell Canada's Call Answer service, for instance, can't

Using FWD and accessing it via this extension: exten => _*8.,1,Dial(SIP/${EXTEN:2}@fwd.pulver.com) This works *perfectly* with SIP phones. However with a regular phone plugged into an FXS card (PhoneJack PCI in my case) the '.' traps the first number dialled after *8 and tries calling that. I've tried setting a digit timeout but it doesn't seem to help. Changing that to

The subject says it all... I have an X100P and I have (for now anyway) Bell Canada's Call Answer which will notify you through one of those nifty CLASS signals that you either do or do not have voicemail. This is not only a "stuttering dialtone" but some actual signal passed so CLASS-aware phones can detect it and flash their message waiting indicator. I'd like to detect

I'm trying to use Jeremy's suggestion of dialling using just the peer name instead of user:pass@peer but I'm running into some really funky issues. It does the same thing with both VoicePulse and another * server I have. [voicepulse] type=peer host=gw5.voicepulse.com trunk=yes user=USERNAME pass=PASSWORD and in my dialplan: Dial(IAX2/voicepulse/${EXTEN:2}@VPWS,90,r) The log shows

http://www.imagestream.com/PCI_720.html Regards, Andrew

I searched the 22490 messages I have in my own personal asterisk-users archive and have not found the answer, and it also does not appear on the wiki. I have a SIP phone and a regular phone on a TDM400P FXS interface. Extensions are 100 and 101, respectively. On the SIP phone I can hit #, get the "Transfer" prompt and enter an extension I want to transfer to. No problem. I can do

http://www.mixdown.ca/~asterisk/ The card seems to be working fine. Don't expect warranty on it though. :-) Why? Because I wanted a quad-span T1 card which worked in the high-end systems I was intending it for, but that I could also keep a few in stock and if the system mainboard failed, pull the card an put it in a standard PC system. I didn't want to have to stock 3.3V and 5V

T100P with an Adit600 channel bank. 16FXS, 8FXO. Been doing this with the past month or so's worth of CVS HEAD, probably longer. I have a weird problem when I bridge calls... not always but often enough to be nasty. Call comes in FXO, * calls my home via IAX2... times out so picks up another FXO port and dials my cell. 7 times out of 10 it bridges just fine, but about 30% of the time

I'm trying to set up asterisk so any phone connected to channel 1-16 of my Adit600 channel bank can pick up a call coming in on channel 24. I do not wish to ring any of the 16 channels on an incoming call -- this is strictly so I can pick up the line if I see it ringing and wish to answer at work. I have channel 24 in call group 3, and channels 1-16 in pickup groups 1 and 3. However

CVS HEAD from about 1 week ago. TDM30P and call through Nufone. I was talking and wanted to park the call and move to another phone to pick it up. I hit #701 instead of #700 though -- after a pause, I got a fast busy and the call was gone. When I called the person back, she said that Alison told HER that 701 was an invalid extension. I should have heard that though, not her. If I dial

The problem: T100P connected to an Adit600. Channel 1-16 are FXS, 17-24 FXO. I have Zap/24 in callgroup 3 and Zap/1-16 in pickupgroup 3. When a call comes in on Zap/24 I cannot pick it up with *8 from Zap/1-16. *CLI> show version Asterisk CVS-04/27/04-23:48:08 built by root@tuck on a i686 running Linux The zapata.conf and extensions.conf are located here:

I'm running asterisk CVS HEAD from 20040601 with spandsp 0.0.1k and libtiff 3.6.0 (no other copies are installed). I've put the audio files up at http://www.mixdown.ca/~andrew/dump/akohlsmith-faxsegfault.tgz -- the machine I am faxing from is a Canon IR3300 printer/copier/fax, but I get similar crashes from $29.95 fax machines too. :-) I'm trying to get a decent backtrace but

I understand from the archives that * does this occassionally, but I'm trying to figure out why. * didn't do this at all for two days, and then it's gone and done it 3 times in the past hour. It does not seem to be affecting calls, I'm just curious as to the reasoning behind the B channel resets and why they are so erratic. Regards, Andrew

I'm looking for a way to detect when a VOIP provider is unable to complete a call and thus try another VOIP provider (failover/backup type situation). using qualify is NOT sufficient, since the provider could very well be reachable but not be able to complete the call for other reasons. A perfect example: setting my caller ID number to my real number and calling a local number causes the