Greetings. We are running * on RH9 using a Digium TDM400P four-port FXO card. We use only two ports on the card (ports 0 and 2 in this case). A consultant set this up for us, and it mostly works OK. However, outbound calls use our secondary number rather than our primary number first. This undesirable because of Caller ID; we'd like the primary number to appear instead. Yes, I can correct this by swapping the PSTN lines. But which config file assigns the physical FXO ports to PSTN lines? In other words, what file would I alter to assign different lines to each port? thanks dn
Try looking in your extensions.conf file. If you are using ports 0 and 2 then you should see somewhere in there something like zap/1 and zap/4 and those should tied to the dial commands. Hopefully these have been configure in the globals section of the extensions.conf. I am kind of a newb at this so if this is not clear, feel free to fire away with questions. Robert -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of David Newman Sent: Monday, January 31, 2005 2:09 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] line assignment on TDM400P Greetings. We are running * on RH9 using a Digium TDM400P four-port FXO card. We use only two ports on the card (ports 0 and 2 in this case). A consultant set this up for us, and it mostly works OK. However, outbound calls use our secondary number rather than our primary number first. This undesirable because of Caller ID; we'd like the primary number to appear instead. Yes, I can correct this by swapping the PSTN lines. But which config file assigns the physical FXO ports to PSTN lines? In other words, what file would I alter to assign different lines to each port? thanks dn _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
On Mon, 31 Jan 2005, Robert Webb wrote:> Try looking in your extensions.conf file. If you are using ports 0 and 2 > then you should see somewhere in there something like zap/1 and zap/4 > and those should tied to the dial commands. > > Hopefully these have been configure in the globals section of the > extensions.conf. > > I am kind of a newb at this so if this is not clear, feel free to fire > away with questions.Thanks for getting back to me. Unfortunately, there is no instance of the string "zap" in extensions.conf. And here is the entire globals section: [globals] USER1_EXTENS=SIP/user1&IAX2/user1&SIP/shadow&SIP/shadow2&SIP/shadow1&SIP/shadow3 USER1_BUSY=0 USER2_EXTENS=SIP/user2 [other sections...] Anywhere else I might look? thanks dn
Did you do a case sensitive search? There is a difference in the search criteria between zap and Zap. It should be as Zap and it can be in any form. I would have setup a trunk group and in the dial command it would be Zap/G1 or Zap/g1. Find that and just reverse the case of the letter g.(that reveres the order that uses idle channels.) If you need more help just holler and post the portion of your extensions.conf that has the Dial command for outgoing calls. Lyle ----- Original Message ----- From: "David Newman" <dnewman@networktest.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Monday, January 31, 2005 3:15 PM Subject: RE: [Asterisk-Users] line assignment on TDM400P> On Mon, 31 Jan 2005, Robert Webb wrote: > > > Try looking in your extensions.conf file. If you are using ports 0 and 2 > > then you should see somewhere in there something like zap/1 and zap/4 > > and those should tied to the dial commands. > > > > Hopefully these have been configure in the globals section of the > > extensions.conf. > > > > I am kind of a newb at this so if this is not clear, feel free to fire > > away with questions. > > Thanks for getting back to me. Unfortunately, there is no instance of the > string "zap" in extensions.conf. And here is the entire globals section: > > [globals] > > >USER1_EXTENS=SIP/user1&IAX2/user1&SIP/shadow&SIP/shadow2&SIP/shadow1&SIP/sha dow3> USER1_BUSY=0 > USER2_EXTENS=SIP/user2 > > [other sections...] > > Anywhere else I might look? > > thanks > > dn > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
> However, outbound calls use our secondary number rather than our primary > number first. This undesirable because of Caller ID; we'd like the primary > number to appear instead. > Yes, I can correct this by swapping the PSTN lines. But which config file > assigns the physical FXO ports to PSTN lines? In other words, what file > would I alter to assign different lines to each port?Since the ports will be assigned channels in order, swapping the cables may be the best option if you are using dialgroups. If you are not using dialgroups, you can find the dial command and change the channel number. Often these are handled like this: SECONDARY=ZAP/1 PRIMARY=ZAP/4 exten => _NXXNXXXXXX,1,Dial( ${PRIMARY}/${EXTEN} ) ; force use of line 1