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 <[EMAIL PROTECTED]> 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." <[EMAIL PROTECTED]> 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/
