Hi all! We set up a sipserver using asterisk X ix66 and need some test calls from around world to verify if it is working ok. If you can :-) please call us: sip:ipfone@sipserver.com.br > direct to snom200 or sip:asterisk@sipserver.com.br > to asterisk >> snom200 Thank?s for all Miklos iPFONE Telefonia IP Rua Caio Graco 735 S?o Paulo SP iPBX +55 11 3801-3702 FWD 64662 ICH 31451543 www.ipfone.com.br info@ipfone.com.br -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031124/9c604491/attachment.htm
On Mon, Nov 24, 2003 at 02:10:48PM -0200, listas iPfone wrote:> Hi all! > > We set up a sipserver using asterisk X ix66 and need some test calls from around world to verify if it is working ok. > > If you can :-) please call us: > > sip:ipfone@sipserver.com.br > direct to snom200 > > or > > sip:asterisk@sipserver.com.br > to asterisk >> snom200 > > Thank?s for all > > MiklosMiklos, OK, I just dialed, looks like you answered. However my * attempts a native bridge between my grandstream phone and your sipserver. Do you have a suggestions on how I can set up a stanza in sip.conf so I can call you and keep * from trying a native bridge? -->console log: -- Executing Dial("SIP/2400-3989", "sip/ipfone@sipserver.com.br|60") in new stack -- Called ipfone@sipserver.com.br -- SIP/sipserver.com.br-c906 is ringing -- SIP/sipserver.com.br-c906 is ringing -- SIP/sipserver.com.br-c906 is ringing -- SIP/sipserver.com.br-c906 is ringing -- SIP/sipserver.com.br-c906 answered SIP/2400-3989 -- Attempting native bridge of SIP/2400-3989 and SIP/sipserver.com.br-c906 -->extensions.conf: exten => 900000,1,Dial(sip/ipfone@sipserver.com.br|60) exten => 900000,2,Hangup Thanks, Walker -- ******** DataCrest, Inc. -- Technically Superior ****************** Walker Haddock http://www.datacrest.com DataCrest, Inc. e-mail: wh@datacrest.com 1634A Montgomery Hwy. phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 ***********************************************************************
Walker Haddock wrote:>On Mon, Nov 24, 2003 at 02:10:48PM -0200, listas iPfone wrote: > > >>Hi all! >> >>We set up a sipserver using asterisk X ix66 and need some test calls from around world to verify if it is working ok. >> >>If you can :-) please call us: >> >>sip:ipfone@sipserver.com.br > direct to snom200 >> >>or >> >>sip:asterisk@sipserver.com.br > to asterisk >> snom200 >> >>Thank?s for all >> >>Miklos >> >> > >Miklos, > >OK, I just dialed, looks like you answered. However my * attempts a native bridge between my grandstream phone and your sipserver. Do you have a suggestions on how I can set up a stanza in sip.conf so I can call you and keep * from trying a native bridge? > >-->console log: > > -- Executing Dial("SIP/2400-3989", "sip/ipfone@sipserver.com.br|60") in new stack > -- Called ipfone@sipserver.com.br > -- SIP/sipserver.com.br-c906 is ringing > -- SIP/sipserver.com.br-c906 is ringing > -- SIP/sipserver.com.br-c906 is ringing > -- SIP/sipserver.com.br-c906 is ringing > -- SIP/sipserver.com.br-c906 answered SIP/2400-3989 > -- Attempting native bridge of SIP/2400-3989 and SIP/sipserver.com.br-c906 > >-->extensions.conf: > >exten => 900000,1,Dial(sip/ipfone@sipserver.com.br|60) >exten => 900000,2,Hangup > > >Thanks, Walker > >Adding "canreinvite=no" to your sip.conf for that phone should do it.. Later..
Hi Miklos, I have the same as Walker. Dave -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Walker Haddock Sent: 24 November 2003 18:02 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] test call request On Mon, Nov 24, 2003 at 02:10:48PM -0200, listas iPfone wrote:> Hi all! > > We set up a sipserver using asterisk X ix66 and need some test calls fromaround world to verify if it is working ok.> > If you can :-) please call us: > > sip:ipfone@sipserver.com.br > direct to snom200 > > or > > sip:asterisk@sipserver.com.br > to asterisk >> snom200 > > Thank?s for all > > MiklosMiklos, OK, I just dialed, looks like you answered. However my * attempts a native bridge between my grandstream phone and your sipserver. Do you have a suggestions on how I can set up a stanza in sip.conf so I can call you and keep * from trying a native bridge? -->console log: -- Executing Dial("SIP/2400-3989", "sip/ipfone@sipserver.com.br|60") in new stack -- Called ipfone@sipserver.com.br -- SIP/sipserver.com.br-c906 is ringing -- SIP/sipserver.com.br-c906 is ringing -- SIP/sipserver.com.br-c906 is ringing -- SIP/sipserver.com.br-c906 is ringing -- SIP/sipserver.com.br-c906 answered SIP/2400-3989 -- Attempting native bridge of SIP/2400-3989 and SIP/sipserver.com.br-c906 -->extensions.conf: exten => 900000,1,Dial(sip/ipfone@sipserver.com.br|60) exten => 900000,2,Hangup Thanks, Walker -- ******** DataCrest, Inc. -- Technically Superior ****************** Walker Haddock http://www.datacrest.com DataCrest, Inc. e-mail: wh@datacrest.com 1634A Montgomery Hwy. phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *********************************************************************** _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
Hi ! Thank you for the call I think that you have to Put reinvite=no in your sip.conf for the given friend/user/peer to keep * from trying a native bridge. I tryed to call you ( sip:2400@home.datacrest.com and sip:2400@192.168.254.179) but the call timeout Thank you again Miklos ----- Original Message ----- From: "Walker Haddock" <whaddock@datacrest.com> To: <asterisk-users@lists.digium.com> Sent: Monday, November 24, 2003 4:02 PM Subject: Re: [Asterisk-Users] test call request> On Mon, Nov 24, 2003 at 02:10:48PM -0200, listas iPfone wrote: > > Hi all! > > > > We set up a sipserver using asterisk X ix66 and need some test callsfrom around world to verify if it is working ok.> > > > If you can :-) please call us: > > > > sip:ipfone@sipserver.com.br > direct to snom200 > > > > or > > > > sip:asterisk@sipserver.com.br > to asterisk >> snom200 > > > > Thank?s for all > > > > Miklos > > Miklos, > > OK, I just dialed, looks like you answered. However my * attempts anative bridge between my grandstream phone and your sipserver. Do you have a suggestions on how I can set up a stanza in sip.conf so I can call you and keep * from trying a native bridge?> > -->console log: > > -- Executing Dial("SIP/2400-3989", "sip/ipfone@sipserver.com.br|60")in new stack> -- Called ipfone@sipserver.com.br > -- SIP/sipserver.com.br-c906 is ringing > -- SIP/sipserver.com.br-c906 is ringing > -- SIP/sipserver.com.br-c906 is ringing > -- SIP/sipserver.com.br-c906 is ringing > -- SIP/sipserver.com.br-c906 answered SIP/2400-3989 > -- Attempting native bridge of SIP/2400-3989 andSIP/sipserver.com.br-c906> > -->extensions.conf: > > exten => 900000,1,Dial(sip/ipfone@sipserver.com.br|60) > exten => 900000,2,Hangup > > > Thanks, Walker > -- > ******** DataCrest, Inc. -- Technically Superior ****************** > Walker Haddock http://www.datacrest.com > DataCrest, Inc. e-mail: wh@datacrest.com > 1634A Montgomery Hwy. phone: 1-888-941-3282, 1-205-335-8589 > Birmingham, AL 35216 fax: 1-205-823-7838 > *********************************************************************** > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
Hi dave I think that is a problem with nat, calls direct to the snom phone trough ix66 works well but from asterisk don?t. Thanks for the call Miklos ----- Original Message ----- From: "David J Carter" <david.carter@codepipe.com> To: <asterisk-users@lists.digium.com> Sent: Monday, November 24, 2003 5:04 PM Subject: RE: [Asterisk-Users] test call request> Hi Miklos, > > I have the same as Walker. > > Dave > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Walker Haddock > Sent: 24 November 2003 18:02 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] test call request > > On Mon, Nov 24, 2003 at 02:10:48PM -0200, listas iPfone wrote: > > Hi all! > > > > We set up a sipserver using asterisk X ix66 and need some test callsfrom> around world to verify if it is working ok. > > > > If you can :-) please call us: > > > > sip:ipfone@sipserver.com.br > direct to snom200 > > > > or > > > > sip:asterisk@sipserver.com.br > to asterisk >> snom200 > > > > Thank?s for all > > > > Miklos > > Miklos, > > OK, I just dialed, looks like you answered. However my * attempts anative> bridge between my grandstream phone and your sipserver. Do you have a > suggestions on how I can set up a stanza in sip.conf so I can call you and > keep * from trying a native bridge? > > -->console log: > > -- Executing Dial("SIP/2400-3989", "sip/ipfone@sipserver.com.br|60")in> new stack > -- Called ipfone@sipserver.com.br > -- SIP/sipserver.com.br-c906 is ringing > -- SIP/sipserver.com.br-c906 is ringing > -- SIP/sipserver.com.br-c906 is ringing > -- SIP/sipserver.com.br-c906 is ringing > -- SIP/sipserver.com.br-c906 answered SIP/2400-3989 > -- Attempting native bridge of SIP/2400-3989 and > SIP/sipserver.com.br-c906 > > -->extensions.conf: > > exten => 900000,1,Dial(sip/ipfone@sipserver.com.br|60) > exten => 900000,2,Hangup > > > Thanks, Walker > -- > ******** DataCrest, Inc. -- Technically Superior ****************** > Walker Haddock http://www.datacrest.com > DataCrest, Inc. e-mail: wh@datacrest.com > 1634A Montgomery Hwy. phone: 1-888-941-3282, 1-205-335-8589 > Birmingham, AL 35216 fax: 1-205-823-7838 > *********************************************************************** > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >