So you are using QSIG and connecting your Asterisk box to a legacy PBX over PRI E1?
Did you try "unknown"? Do you need to use QSIG (over euroisdn for instance)? Thanks, Steve Totaro On Wed, Sep 25, 2013 at 10:33 AM, Endri Stefani <[email protected]>wrote: > Hi Steve,**** > > ** ** > > There are no errors I need to be able to change TON(below my PRI debug ) > in international or subscriber. The change in chan_dahdi.conf did not do it > **** > > ** ** > > > Calling Number (len= 8) [ Ext: 0 TON: National Number (2) NPI: > ISDN/Telephony Numbering Plan (E.164/E.163) (1)**** > > > Presentation: Presentation permitted, user > number not screened (0) '1000' ]**** > > > [70 0c a1 30 30 36 36 39 31 31 30 30 30 30]**** > > > Called Number (len=14) [ Ext: 1 TON: National Number (2) NPI: > ISDN/Telephony Numbering Plan (E.164/E.163) (1) 'xxxxxx' ]**** > > ** ** > > ** ** > > Br**** > > ** ** > > ** ** > > *From:* [email protected] [mailto: > [email protected]] *On Behalf Of *Steve Totaro > *Sent:* Wednesday, September 25, 2013 4:24 PM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] Asterisk TON number**** > > ** ** > > I asked you before. What exactly are you trying to do that you cannot? > It helps to be fairly detailed when asking a question to the list. > Include error messages if you have any.**** > > ** ** > > The dialplan and your ISDN configs are different things. It sounds like > maybe you are having issues with with your dailplan and pattern matching. > You can probably do all of that from the Trixbox GUI. If you like > Trixbox, check out FreePBX since Trixbox is done. **** > > ** ** > > Thanks,**** > > Steve Totaro**** > > ** ** > > On Wed, Sep 25, 2013 at 9:30 AM, Endri Stefani <[email protected]> > wrote:**** > > Hi guys**** > > **** > > Thanks a lot, I am just getting used to it, my telco managers J don’t > trust stability for open source solution for voice(you give a headache to > calling parties if lag is more than 250ms J ) and I want to prove them > wrong. I have successfully integrated * with our system via ISDN and first > calls look fast and clear but it is required to be flexible with TON number > in order to be used. I tried unloading and loading chan-dahdi.conf like > Hans suggested with no success TON was not changed.**** > > Here is my chan_dahdi.conf, is there anything else I should do in order > for new pridailplan came into action :**** > > **** > > ;**** > > ; DAHDI telephony**** > > ;**** > > ; Configuration file**** > > **** > > [trunkgroups]**** > > **** > > [channels]**** > > **** > > switchtype = qsig**** > > context = pri_incoming**** > > group = 0**** > > signalling = pri_cpe**** > > channel => 1-15,17-31**** > > **** > > **** > > **** > > ;language=en**** > > ;context=from-zaptel**** > > ;signalling=fxs_ks**** > > ;rxwink=300 ;**** > > ; Whether or not to do distinctive ring detection on FXO lines**** > > ;**** > > ;usedistinctiveringdetection=yes**** > > **** > > usecallerid=yes**** > > hidecallerid=no**** > > callwaiting=yes**** > > usecallingpres=yes**** > > callwaitingcallerid=yes**** > > threewaycalling=yes**** > > transfer=yes**** > > cancallforward=yes**** > > callreturn=yes**** > > echocancel=yes**** > > echocancelwhenbridged=no**** > > echotraining=400**** > > rxgain=0.0**** > > txgain=0.0**** > > group=0**** > > callgroup=1**** > > pickupgroup=1**** > > immediate=no**** > > **** > > **** > > ;faxdetect=both**** > > faxdetect=incoming**** > > ;faxdetect=outgoing**** > > ;faxdetect=no**** > > **** > > ;Include setup-pstn configs**** > > #include dahdi-channels.conf**** > > **** > > group=1**** > > **** > > ;Include PBXconfig configs**** > > #include chan_dahdi_additional.conf**** > > **** > > **** > > unknown: Unknown**** > > private: Private ISDN**** > > local: Local ISDN**** > > national: National ISDN**** > > international: International ISDN**** > > dynamic: Dynamically selects the appropriate dialplan**** > > redundant: Same as dynamic, except that the underlying number is not > **** > > changed (not common)**** > > **** > > pridialplan = international**** > > prilocaldialplan = international**** > > **** > > **** > > **** > > *From:* [email protected] [mailto: > [email protected]] *On Behalf Of *$$ dave cantera > (android asus) > *Sent:* Wednesday, September 25, 2013 2:46 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion**** > > > *Subject:* Re: [asterisk-users] Asterisk TON number**** > > **** > > if you are a serious teleco guy, which it seems you are. you might > consider dumping trixbox in the near future. while trixbox does provide a > good entry level into the * world, there are limitations that will > eventually hold you back from enjoying the full breadth of utility that * > offers.**** > > > food for thought, > > Dave Cantera > (856)813-7098 mobile/txt > [email protected] > > Sent from my ASUS Pad > > Steve Totaro <[email protected]> wrote:**** > > **** > > **** > > On Wed, Sep 25, 2013 at 3:22 AM, Endri Stefani <[email protected]> > wrote:**** > > Hi**** > > **** > > Greeting to all you out there.**** > > **** > > I am new at asterisk, I have been working with PLMN platforms > telecommunication for 5 years with NSN and Huawei.**** > > We have recently built an asterisk PBX with Trixbox and connected it to > our MSS using Digium E1 cards(ISDN). Everything went smoothly as there are > tons of information out there, except for the TON number.**** > > If you have worked in Telecommunication you will know the importance of > TON flexibility. **** > > All the posts online suggested to update under Chan_dahdi.conf:**** > > pridialplan = international**** > > prilocaldialplan = international**** > > or other TON value ,restart the platform and then trixbox1*CLI> dialplan > reload**** > > I have already done this with no success. Are there other changes I have > to make in order to change dialplan?**** > > **** > > **** > > Br**** > > **** > > **** > > So what are you trying to do specifically?**** > > **** > > Thanks,**** > > Steve Totaro **** > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users**** > > ** ** > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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