Hi all, I tried with defaultzone and loadzone = uk also...but it also dosn't seem to be working..... Right now i m using busydetect which seems to be working fine....But its not the proper solution....If u are using IVR ...then busydetect also even not work. The main problem is with zaptel hardware is that its not able to recognize the Hangup signal. I tried to contact with the main devloper of zaptel also. He replied:
Mr. Jim(Devloper of Zaptel) mailed me:- The biggest thing that is needed for your application is for you to get them(etisalat) to turn on the CPC pulse at the end of the call (thats the zero-battery pulse). LOL!!! Thanks & Regards, Parag Srivastava --- In [email protected], "Michael S." <[EMAIL PROTECTED]> wrote: > > Hi jama, > > No, iax protocol is working fine. I stepped away from SIP because of > too many troubles routing the RTP protocol. I did get it working once > by setting nat=yes in the sip.conf channel descriptions. But it didn't > seem to work anymore afterwards. > > Then I found that IAX softphone support is coming up rapidly so I > switched to IAX protocol completely. > > Anyway that's not the problem. IAX to IAX is working fine, I can leave > messages on voicemail and all that. > > The problem lies here: I make a call to a landline, asterisk makes the > call over the TDM card. But it does NOT detect that the person on the > other line picks up! Therefore does not connect the IAX and ZAP > channel together and the person doesn't hear me talking. I do here the > other person talking because one way the connection is already open > because I need to hear the ringing tone. > > So I need to fine tune the zapata.conf to properly detect when a phone > on the other end of the telephone line is picked up. > > I'll keep you posted! > > Michael > > > --- In [email protected], Jama I Barreh <jbarreh@> wrote: > > > > Hello Michael, > > > > One way voice could be due to blocked ports either on your router or > Etisalat (most probably > > your router if your calling device is behind it). > > Depending on whether you are using SIP or IAX, open the appropriate > port on your > > router. Note: for SIP you have to open range of ports between 10,000 > to 20,000 depending > > on how your asterisk is set, in addition to port 5060 (default). > Port 4569 is the default > > for IAX. > > I do have extra FXO's for Digium TDM4XX card but too bad I am not in > UAE. I can ship > > one to you if you pay the shipment cost (DHL) on delivery. > > Please DO let us know the outcome of your trial. > > > > Jama > > > > > > --- "Michael S." <chas3r@> wrote: > > > > > Hi Jama, > > > > > > It seems to be working better now. Don't really know what I changed. > > > PCI interrupts should be ok, I checked and the card is using resources > > > that are free. It's the only pci card in the box so that should be > fine. > > > > > > Incoming calls on the digium card are working now. But I have a > > > problem making outgoing calls. The Digium card doesn't seem to notice > > > that the other end picked up! So I hear the person on the other end of > > > the landline picking up, talking, but he doesn't hear me! Moreover, > > > there is still some kind of ringing tone coming over the channel. > > > > > > I guess it has something to do with the way Etisalat signals a line > > > pickup. Some tweaking here and there is definitely needed, but overall > > > the thing is starting to work. > > > > > > Is anyone going to order something from Digium soon? Because we need > > > another FXO module for the TDM400P. But it's a bit silly to pay $60 > > > shipment costs for just one $85 module. > > > Also if anyone has an unused FXO module for the TDM400P and wants to > > > trade it for an FXS module (which we have spare) let me know! > > > > > > Michael > > > > > > --- In [email protected], Jama I Barreh <jbarreh@> wrote: > > > > > > > > Hello, > > > > > > > > Is it only IAX that drops the calls ? > > > > Did you try SIP at all ? > > > > Random loss of connection can some times be due to interrupt > conflict > > > > on the PCI bus. > > > > > > > > Jama > > > > > > > > --- "Michael S." <chas3r@> wrote: > > > > > > > > > Hi John, > > > > > > > > > > I think it's important to know some facts about the Etisalat > telephone > > > > > network. Because a lot of how the cards work depend on which > network > > > > > type is being used. This goes from voltage ranges to signal > timings > > > > > and all that. Also the way a call is "sensed". Either there is > a brief > > > > > polarity switch on the incoming phone wires or there is a ring > signal. > > > > > > > > > > I noticed that in order to get things working, we need to get > as much > > > > > technical information as possible from the Etisalat telephone > network. > > > > > Anyone that can help us with that? > > > > > > > > > > I asked the guy who came to rewire our incoming lines a > question as > > > > > well. In the /etc/zaptel.conf you have to specify which > country you > > > > > are using. This determines dial timings (signal timings) and ring > > > > > types etc. It turns out that the Etisalat network is a lot > like the UK > > > > > telephone network. Since AE is not a supported zone in Zaptel > yes, set > > > > > loadzone=uk and defaultzone=uk. > > > > > > > > > > That seems to prevent calls from being dropped more. > > > > > > > > > > I do have another problem and that's when I am calling someone IAX > > > > > protocol -> landline, after about 10 minutes or so the other > person > > > > > doesn't hear me talking anymore. That problem could be related > to many > > > > > things though. > > > > > > > > > > Michael > > > > > > > > > > > > > > > --- In [email protected], John Joseph <jjk_saji@> wrote: > > > > > > > > > > > > > > > > > > --- "Michael S." <chas3r@> wrote: > > > > > > > By the way, my Asterisk Digium card arrived today. > > > > > > > I'm installing it now. > > > > > > > > > > > > > > > > > > > I had faced problem on connecting to the PSTN line , > > > > > > my incoming calls get droped off > > > > > > I plan to give a try again on Sunday > > > > > > are u using AMP > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > ___________________________________________________________ > > > > > > Win a BlackBerry device from O2 with Yahoo!. Enter now. > > > > > http://www.yahoo.co.uk/blackberry > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Yahoo! Groups Links > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Time-table for Garadag (0224000 only - no country code no 011) > > > > 8:30AM ilaa 12:00 Noon > > > > 6:00PM ilaa 12:00 Midnight > > > > > > > > > > > > > > > > > > > > > > > > > > > > Yahoo! Groups Links > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Time-table for Garadag (0224000 only - no country code no 011) > > 8:30AM ilaa 12:00 Noon > > 6:00PM ilaa 12:00 Midnight > > > Yahoo! Groups Links <*> To visit your group on the web, go to: http://groups.yahoo.com/group/dubailug/ <*> To unsubscribe from this group, send an email to: [EMAIL PROTECTED] <*> Your use of Yahoo! Groups is subject to: http://docs.yahoo.com/info/terms/
