hi all, just had a terrible and sleepless weekend at the office trying to get asterisk going, its just tough love ;) have tried several asterisk versions but i currently have the following setup on debian lenny that kind of works. asterisk-1.6.2.0 dahdi-linux-complete-2.2.0.2+2.2.0 libpri-1.4.10.2 freepbx-2.6.0 setting up the sip devices is no problem at all, the difficulty i have is setting up 6xisdn2 lines with 2xb410p cards. besides the fact that i have no clue about what i'm doing i find the available documentation very very confusing, but i finally managed to make outgoing calls to my mobile this morning, sort off. when calling my mobile i hear a ringtone on my sip device and my mobile actually rings, YEAH!!! however, when i accept the call on my mobile my sip device keeps on ringing and my mobile gives no sound at all, when cancelling the call it simply cancels. except, and this i don't understand, i issue asterisk -rv (only with the v option), then i can connect and talk to myself, i often talk to myself when i spent a weekend at the office but this time its justifiable ;) anybody has a clue what could trigger this behavior???
On 02/01/2010 08:38 AM, randall wrote:> hi all, > > just had a terrible and sleepless weekend at the office trying to get > asterisk going, its just tough love ;) > > have tried several asterisk versions but i currently have the following > setup on debian lenny that kind of works. > asterisk-1.6.2.0 > dahdi-linux-complete-2.2.0.2+2.2.0 > libpri-1.4.10.2 > freepbx-2.6.0 > > setting up the sip devices is no problem at all, the difficulty i have > is setting up 6xisdn2 lines with 2xb410p cards. > > besides the fact that i have no clue about what i'm doing i find the > available documentation very very confusing, but i finally managed to > make outgoing calls to my mobile this morning, sort off. > > when calling my mobile i hear a ringtone on my sip device and my mobile > actually rings, YEAH!!! > however, when i accept the call on my mobile my sip device keeps on > ringing and my mobile gives no sound at all, when cancelling the call it > simply cancels. > > except, and this i don't understand, i issue asterisk -rv (only with the > v option), then i can connect and talk to myself, i often talk to myself > when i spent a weekend at the office but this time its justifiable ;) > > anybody has a clue what could trigger this behavior??? > , >update!!! apparently it sometimes does work, randomly, guess the -v was a very lucky shot, i repeated it 20 times. i do seem to get this message everytime a connection fails [Feb 1 09:25:14] ERROR[2867] chan_dahdi.c: XXX Message longer than it should be?? XXX after applying this patch below the problem seem to have dissapeared for now https://issues.asterisk.org/view.php?id=16048
On Mon, Feb 01, 2010 at 08:38:36AM +0100, randall wrote:> hi all, > > just had a terrible and sleepless weekend at the office trying to get > asterisk going, its just tough love ;) > > have tried several asterisk versions but i currently have the following > setup on debian lenny that kind of works. > asterisk-1.6.2.0 > dahdi-linux-complete-2.2.0.2+2.2.0 > libpri-1.4.10.2 > freepbx-2.6.0 > > setting up the sip devices is no problem at all, the difficulty i have > is setting up 6xisdn2 lines with 2xb410p cards. > > besides the fact that i have no clue about what i'm doing i find the > available documentation very very confusing, but i finally managed to > make outgoing calls to my mobile this morning, sort off. > > when calling my mobile i hear a ringtone on my sip device and my mobile > actually rings, YEAH!!! > however, when i accept the call on my mobile my sip device keeps on > ringing and my mobile gives no sound at all, when cancelling the call it > simply cancels.You try to connect two devices, ISDN and SIP. Both have their own complexities. I would sugest that you start by breaking this into two: first make sure an incoming ISDN call can make it into your PBX. e.g. into a simple IVR. Also make sure you can call your phone from Asterisk: In the Asterisk CLI: originate SIP/your-peer-name Application Playback demo-instruct Or: originate SIP/your-peer-name Application Echo -- Tzafrir Cohen icq#16849755 jabber:tzafrir.cohen at xorcom.com +972-50-7952406 mailto:tzafrir.cohen at xorcom.com http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir