Re: [asterisk-users] Inquiry:Connect my Asterisk to external sip?
It looks to me like calls from your Dial will route back to the sip-outgoing context and Dial again... it's loop. You'd really need to provide more logging information to advise further. On Thu, Dec 24, 2009 at 5:16 AM, hadi motamedi wrote: > > > On Wed, Dec 23, 2009 at 1:55 PM, David Cunningham < > dcunning...@voisonics.com> wrote: > >> AsteriskWin32 does have SIP server functionality, same as the linux >> version. >> >> I can't think of any reason why having your CentOS Asterisk be both client >> and server and register with itself wouldn't work. >> Although I am wondering how much help all this will be in debugging a >> connection problem to another SIP provider... >> >> >> On Wed, Dec 23, 2009 at 12:55 PM, hadi motamedi wrote: >> >>> >>> >>> On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham < >>> dcunning...@voisonics.com> wrote: >>> Hadi, You could use Asterisk as a sip server, it's installable on Windows. Using "sip set debug on" might help you with the "Host '192.168.0.139' does not implement 'REGISTER'" problem. On Wed, Dec 23, 2009 at 9:12 AM, hadi motamedi wrote: > > > On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner wrote: > >> >> On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote: >> >> > >> > >> > >> > On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner >> wrote: >> > >> > On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote: >> > >> > > Dear All >> > > I have an application that calls for my Asterisk sip to be >> connected to an external sip server for voip routing . Please be informed >> that my Asterisk sip is at @192.168.0.2 and the external sip is at @ >> 192.168.0.139 . To this end , I modified my sip.conf & >> extensions.conf as the followings : >> > > Under sip.conf : >> > > - >> > > [general] >> > > register => toronto:welc...@192.168.0.139/osaka >> > > [osaka] >> > > type=friend >> > > secret=welcome >> > > context=osaka_incoming >> > > host=dynamic >> > > disallow=all >> > > allow=alaw >> > > [6672019] >> > > type=friend >> > > host=dynamic >> > > context=phones >> > > >> > >> > Try this: >> > >> > [general] >> > register => toronto:welc...@osaka >> > >> > [osaka] >> > type=friend >> > username=toronto >> > authname=toronto >> > secret=welcome >> > context=osaka_incoming >> > host=192.168.0.139 >> > disallow=all >> > allow=alaw >> > >> > Although your error shows the other server does not allow register. >> What is the other server? >> > >> > ---fred >> > http://qxork.com >> > >> > >> > Thank you for your reply . The other server is not an Asterisk sip >> server . It is a sip server inside a softswitch from a third party >> vendor . >> As the external sip server man is asking me to disable for the >> authentication at the first stage , can you please let me know how can I >> disable for the authentication at this stage (when the calls get through >> I >> will enable it again) ? >> > Thank you in advance >> > >> >> [general] >> ;register => toronto:welc...@osaka >> >> [osaka] >> type=friend >> ;username=toronto >> ;authname=toronto >> ;secret=welcome >> context=osaka_incoming >> host=192.168.0.139 >> disallow=all >> allow=alaw >> >> >> ---fred >> http://qxork.com >> >> >> >> >> >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > Thank you for your reply . Please be informed that I want to simulate > this case in the Laboratory , i.e. connecting my Asterisk sip to external > sip server with the guidelines you sent me . Can you please propose for an > Voip application sw that I can install on my MS Windows client and plays > the > external sip server side role ? It seems that Skype is not suitable for > this > case as it cannot be configured to play the role of external sip server . > Thank you in advance > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180
Re: [asterisk-users] Inquiry:Connect my Asterisk to external sip?
On Wed, Dec 23, 2009 at 1:55 PM, David Cunningham wrote: > AsteriskWin32 does have SIP server functionality, same as the linux > version. > > I can't think of any reason why having your CentOS Asterisk be both client > and server and register with itself wouldn't work. > Although I am wondering how much help all this will be in debugging a > connection problem to another SIP provider... > > > On Wed, Dec 23, 2009 at 12:55 PM, hadi motamedi wrote: > >> >> >> On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham < >> dcunning...@voisonics.com> wrote: >> >>> Hadi, >>> >>> You could use Asterisk as a sip server, it's installable on Windows. >>> >>> Using "sip set debug on" might help you with the "Host '192.168.0.139' >>> does not implement 'REGISTER'" problem. >>> >>> >>> On Wed, Dec 23, 2009 at 9:12 AM, hadi motamedi wrote: >>> On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner wrote: > > On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote: > > > > > > > > > On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner > wrote: > > > > On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote: > > > > > Dear All > > > I have an application that calls for my Asterisk sip to be > connected to an external sip server for voip routing . Please be informed > that my Asterisk sip is at @192.168.0.2 and the external sip is at @ > 192.168.0.139 . To this end , I modified my sip.conf & extensions.conf > as the followings : > > > Under sip.conf : > > > - > > > [general] > > > register => toronto:welc...@192.168.0.139/osaka > > > [osaka] > > > type=friend > > > secret=welcome > > > context=osaka_incoming > > > host=dynamic > > > disallow=all > > > allow=alaw > > > [6672019] > > > type=friend > > > host=dynamic > > > context=phones > > > > > > > Try this: > > > > [general] > > register => toronto:welc...@osaka > > > > [osaka] > > type=friend > > username=toronto > > authname=toronto > > secret=welcome > > context=osaka_incoming > > host=192.168.0.139 > > disallow=all > > allow=alaw > > > > Although your error shows the other server does not allow register. > What is the other server? > > > > ---fred > > http://qxork.com > > > > > > Thank you for your reply . The other server is not an Asterisk sip > server . It is a sip server inside a softswitch from a third party vendor > . > As the external sip server man is asking me to disable for the > authentication at the first stage , can you please let me know how can I > disable for the authentication at this stage (when the calls get through I > will enable it again) ? > > Thank you in advance > > > > [general] > ;register => toronto:welc...@osaka > > [osaka] > type=friend > ;username=toronto > ;authname=toronto > ;secret=welcome > context=osaka_incoming > host=192.168.0.139 > disallow=all > allow=alaw > > > ---fred > http://qxork.com > > > > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > Thank you for your reply . Please be informed that I want to simulate this case in the Laboratory , i.e. connecting my Asterisk sip to external sip server with the guidelines you sent me . Can you please propose for an Voip application sw that I can install on my MS Windows client and plays the external sip server side role ? It seems that Skype is not suitable for this case as it cannot be configured to play the role of external sip server . Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >>> >>> -- >>> David Cunningham >>> Voisonics >>> IVR development, VOIP consultancy >>> http://voisonics.com/ >>> US toll-free: +1 888 842 2720 >>> UK: +44 (0) 20 3411 5024 >>> Australia: +61 (0) 2 9037 2180 >>> >>> >>> ___ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> >> >> I downloaded & installed the AsteriskWin32 PBX but it doesn't have sip >> server functionality . Can you please propose for an alternative to be used >> on the MS Windows client as external sip ser
Re: [asterisk-users] Inquiry:Connect my Asterisk to external sip?
AsteriskWin32 does have SIP server functionality, same as the linux version. I can't think of any reason why having your CentOS Asterisk be both client and server and register with itself wouldn't work. Although I am wondering how much help all this will be in debugging a connection problem to another SIP provider... On Wed, Dec 23, 2009 at 12:55 PM, hadi motamedi wrote: > > > On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham < > dcunning...@voisonics.com> wrote: > >> Hadi, >> >> You could use Asterisk as a sip server, it's installable on Windows. >> >> Using "sip set debug on" might help you with the "Host '192.168.0.139' >> does not implement 'REGISTER'" problem. >> >> >> On Wed, Dec 23, 2009 at 9:12 AM, hadi motamedi wrote: >> >>> >>> >>> On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner wrote: >>> On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote: > > > > On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner wrote: > > On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote: > > > Dear All > > I have an application that calls for my Asterisk sip to be connected to an external sip server for voip routing . Please be informed that my Asterisk sip is at @192.168.0.2 and the external sip is at @ 192.168.0.139 . To this end , I modified my sip.conf & extensions.conf as the followings : > > Under sip.conf : > > - > > [general] > > register => toronto:welc...@192.168.0.139/osaka > > [osaka] > > type=friend > > secret=welcome > > context=osaka_incoming > > host=dynamic > > disallow=all > > allow=alaw > > [6672019] > > type=friend > > host=dynamic > > context=phones > > > > Try this: > > [general] > register => toronto:welc...@osaka > > [osaka] > type=friend > username=toronto > authname=toronto > secret=welcome > context=osaka_incoming > host=192.168.0.139 > disallow=all > allow=alaw > > Although your error shows the other server does not allow register. What is the other server? > > ---fred > http://qxork.com > > > Thank you for your reply . The other server is not an Asterisk sip server . It is a sip server inside a softswitch from a third party vendor . As the external sip server man is asking me to disable for the authentication at the first stage , can you please let me know how can I disable for the authentication at this stage (when the calls get through I will enable it again) ? > Thank you in advance > [general] ;register => toronto:welc...@osaka [osaka] type=friend ;username=toronto ;authname=toronto ;secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw ---fred http://qxork.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >>> >>> Thank you for your reply . Please be informed that I want to simulate >>> this case in the Laboratory , i.e. connecting my Asterisk sip to external >>> sip server with the guidelines you sent me . Can you please propose for an >>> Voip application sw that I can install on my MS Windows client and plays the >>> external sip server side role ? It seems that Skype is not suitable for this >>> case as it cannot be configured to play the role of external sip server . >>> Thank you in advance >>> >>> >>> ___ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> >> >> -- >> David Cunningham >> Voisonics >> IVR development, VOIP consultancy >> http://voisonics.com/ >> US toll-free: +1 888 842 2720 >> UK: +44 (0) 20 3411 5024 >> Australia: +61 (0) 2 9037 2180 >> >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > I downloaded & installed the AsteriskWin32 PBX but it doesn't have sip > server functionality . Can you please propose for an alternative to be used > on the MS Windows client as external sip server for my Asterisk on CentOS ? > Thank you > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://l
Re: [asterisk-users] Inquiry:Connect my Asterisk to external sip?
On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham < dcunning...@voisonics.com> wrote: > Hadi, > > You could use Asterisk as a sip server, it's installable on Windows. > > Using "sip set debug on" might help you with the "Host '192.168.0.139' does > not implement 'REGISTER'" problem. > > > On Wed, Dec 23, 2009 at 9:12 AM, hadi motamedi wrote: > >> >> >> On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner wrote: >> >>> >>> On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote: >>> >>> > >>> > >>> > >>> > On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner >>> wrote: >>> > >>> > On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote: >>> > >>> > > Dear All >>> > > I have an application that calls for my Asterisk sip to be connected >>> to an external sip server for voip routing . Please be informed that my >>> Asterisk sip is at @192.168.0.2 and the external sip is at @ >>> 192.168.0.139 . To this end , I modified my sip.conf & extensions.conf >>> as the followings : >>> > > Under sip.conf : >>> > > - >>> > > [general] >>> > > register => toronto:welc...@192.168.0.139/osaka >>> > > [osaka] >>> > > type=friend >>> > > secret=welcome >>> > > context=osaka_incoming >>> > > host=dynamic >>> > > disallow=all >>> > > allow=alaw >>> > > [6672019] >>> > > type=friend >>> > > host=dynamic >>> > > context=phones >>> > > >>> > >>> > Try this: >>> > >>> > [general] >>> > register => toronto:welc...@osaka >>> > >>> > [osaka] >>> > type=friend >>> > username=toronto >>> > authname=toronto >>> > secret=welcome >>> > context=osaka_incoming >>> > host=192.168.0.139 >>> > disallow=all >>> > allow=alaw >>> > >>> > Although your error shows the other server does not allow register. >>> What is the other server? >>> > >>> > ---fred >>> > http://qxork.com >>> > >>> > >>> > Thank you for your reply . The other server is not an Asterisk sip >>> server . It is a sip server inside a softswitch from a third party vendor . >>> As the external sip server man is asking me to disable for the >>> authentication at the first stage , can you please let me know how can I >>> disable for the authentication at this stage (when the calls get through I >>> will enable it again) ? >>> > Thank you in advance >>> > >>> >>> [general] >>> ;register => toronto:welc...@osaka >>> >>> [osaka] >>> type=friend >>> ;username=toronto >>> ;authname=toronto >>> ;secret=welcome >>> context=osaka_incoming >>> host=192.168.0.139 >>> disallow=all >>> allow=alaw >>> >>> >>> ---fred >>> http://qxork.com >>> >>> >>> >>> >>> >>> >>> ___ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> >> >> Thank you for your reply . Please be informed that I want to simulate this >> case in the Laboratory , i.e. connecting my Asterisk sip to external sip >> server with the guidelines you sent me . Can you please propose for an Voip >> application sw that I can install on my MS Windows client and plays the >> external sip server side role ? It seems that Skype is not suitable for this >> case as it cannot be configured to play the role of external sip server . >> Thank you in advance >> >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > -- > David Cunningham > Voisonics > IVR development, VOIP consultancy > http://voisonics.com/ > US toll-free: +1 888 842 2720 > UK: +44 (0) 20 3411 5024 > Australia: +61 (0) 2 9037 2180 > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > I downloaded & installed the AsteriskWin32 PBX but it doesn't have sip server functionality . Can you please propose for an alternative to be used on the MS Windows client as external sip server for my Asterisk on CentOS ? Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Connect my Asterisk to external sip?
On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham < dcunning...@voisonics.com> wrote: > Hadi, > > You could use Asterisk as a sip server, it's installable on Windows. > > Using "sip set debug on" might help you with the "Host '192.168.0.139' does > not implement 'REGISTER'" problem. > > > On Wed, Dec 23, 2009 at 9:12 AM, hadi motamedi wrote: > >> >> >> On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner wrote: >> >>> >>> On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote: >>> >>> > >>> > >>> > >>> > On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner >>> wrote: >>> > >>> > On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote: >>> > >>> > > Dear All >>> > > I have an application that calls for my Asterisk sip to be connected >>> to an external sip server for voip routing . Please be informed that my >>> Asterisk sip is at @192.168.0.2 and the external sip is at @ >>> 192.168.0.139 . To this end , I modified my sip.conf & extensions.conf >>> as the followings : >>> > > Under sip.conf : >>> > > - >>> > > [general] >>> > > register => toronto:welc...@192.168.0.139/osaka >>> > > [osaka] >>> > > type=friend >>> > > secret=welcome >>> > > context=osaka_incoming >>> > > host=dynamic >>> > > disallow=all >>> > > allow=alaw >>> > > [6672019] >>> > > type=friend >>> > > host=dynamic >>> > > context=phones >>> > > >>> > >>> > Try this: >>> > >>> > [general] >>> > register => toronto:welc...@osaka >>> > >>> > [osaka] >>> > type=friend >>> > username=toronto >>> > authname=toronto >>> > secret=welcome >>> > context=osaka_incoming >>> > host=192.168.0.139 >>> > disallow=all >>> > allow=alaw >>> > >>> > Although your error shows the other server does not allow register. >>> What is the other server? >>> > >>> > ---fred >>> > http://qxork.com >>> > >>> > >>> > Thank you for your reply . The other server is not an Asterisk sip >>> server . It is a sip server inside a softswitch from a third party vendor . >>> As the external sip server man is asking me to disable for the >>> authentication at the first stage , can you please let me know how can I >>> disable for the authentication at this stage (when the calls get through I >>> will enable it again) ? >>> > Thank you in advance >>> > >>> >>> [general] >>> ;register => toronto:welc...@osaka >>> >>> [osaka] >>> type=friend >>> ;username=toronto >>> ;authname=toronto >>> ;secret=welcome >>> context=osaka_incoming >>> host=192.168.0.139 >>> disallow=all >>> allow=alaw >>> >>> >>> ---fred >>> http://qxork.com >>> >>> >>> >>> >>> >>> >>> ___ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> >> >> Thank you for your reply . Please be informed that I want to simulate this >> case in the Laboratory , i.e. connecting my Asterisk sip to external sip >> server with the guidelines you sent me . Can you please propose for an Voip >> application sw that I can install on my MS Windows client and plays the >> external sip server side role ? It seems that Skype is not suitable for this >> case as it cannot be configured to play the role of external sip server . >> Thank you in advance >> >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > -- > David Cunningham > Voisonics > IVR development, VOIP consultancy > http://voisonics.com/ > US toll-free: +1 888 842 2720 > UK: +44 (0) 20 3411 5024 > Australia: +61 (0) 2 9037 2180 > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > With many thanks , please let me to ask you if I rely upon on my Asterisk 1.4 installation on my CentOS 5.0 and want to have this "external sip client" on my CentOS server as well so what will be the solution ? The one you told me was for the Laboratory test when the Asterisk on CentOS calls sip client on MS Windows but what will be the solution if the Asterisk on CentOS calls sip client on the same CentOS ? Is there a Voip application on the CentOS that can resemble this "external sip client" ? Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Connect my Asterisk to external sip?
Hadi, You could use Asterisk as a sip server, it's installable on Windows. Using "sip set debug on" might help you with the "Host '192.168.0.139' does not implement 'REGISTER'" problem. On Wed, Dec 23, 2009 at 9:12 AM, hadi motamedi wrote: > > > On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner wrote: > >> >> On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote: >> >> > >> > >> > >> > On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner >> wrote: >> > >> > On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote: >> > >> > > Dear All >> > > I have an application that calls for my Asterisk sip to be connected >> to an external sip server for voip routing . Please be informed that my >> Asterisk sip is at @192.168.0.2 and the external sip is at @192.168.0.139. >> To this end , I modified my sip.conf & extensions.conf as the followings : >> > > Under sip.conf : >> > > - >> > > [general] >> > > register => toronto:welc...@192.168.0.139/osaka >> > > [osaka] >> > > type=friend >> > > secret=welcome >> > > context=osaka_incoming >> > > host=dynamic >> > > disallow=all >> > > allow=alaw >> > > [6672019] >> > > type=friend >> > > host=dynamic >> > > context=phones >> > > >> > >> > Try this: >> > >> > [general] >> > register => toronto:welc...@osaka >> > >> > [osaka] >> > type=friend >> > username=toronto >> > authname=toronto >> > secret=welcome >> > context=osaka_incoming >> > host=192.168.0.139 >> > disallow=all >> > allow=alaw >> > >> > Although your error shows the other server does not allow register. What >> is the other server? >> > >> > ---fred >> > http://qxork.com >> > >> > >> > Thank you for your reply . The other server is not an Asterisk sip >> server . It is a sip server inside a softswitch from a third party vendor . >> As the external sip server man is asking me to disable for the >> authentication at the first stage , can you please let me know how can I >> disable for the authentication at this stage (when the calls get through I >> will enable it again) ? >> > Thank you in advance >> > >> >> [general] >> ;register => toronto:welc...@osaka >> >> [osaka] >> type=friend >> ;username=toronto >> ;authname=toronto >> ;secret=welcome >> context=osaka_incoming >> host=192.168.0.139 >> disallow=all >> allow=alaw >> >> >> ---fred >> http://qxork.com >> >> >> >> >> >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > Thank you for your reply . Please be informed that I want to simulate this > case in the Laboratory , i.e. connecting my Asterisk sip to external sip > server with the guidelines you sent me . Can you please propose for an Voip > application sw that I can install on my MS Windows client and plays the > external sip server side role ? It seems that Skype is not suitable for this > case as it cannot be configured to play the role of external sip server . > Thank you in advance > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Connect my Asterisk to external sip?
On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner wrote: > > On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote: > > > > > > > > > On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner > wrote: > > > > On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote: > > > > > Dear All > > > I have an application that calls for my Asterisk sip to be connected to > an external sip server for voip routing . Please be informed that my > Asterisk sip is at @192.168.0.2 and the external sip is at @192.168.0.139. To > this end , I modified my sip.conf & extensions.conf as the followings : > > > Under sip.conf : > > > - > > > [general] > > > register => toronto:welc...@192.168.0.139/osaka > > > [osaka] > > > type=friend > > > secret=welcome > > > context=osaka_incoming > > > host=dynamic > > > disallow=all > > > allow=alaw > > > [6672019] > > > type=friend > > > host=dynamic > > > context=phones > > > > > > > Try this: > > > > [general] > > register => toronto:welc...@osaka > > > > [osaka] > > type=friend > > username=toronto > > authname=toronto > > secret=welcome > > context=osaka_incoming > > host=192.168.0.139 > > disallow=all > > allow=alaw > > > > Although your error shows the other server does not allow register. What > is the other server? > > > > ---fred > > http://qxork.com > > > > > > Thank you for your reply . The other server is not an Asterisk sip server > . It is a sip server inside a softswitch from a third party vendor . As the > external sip server man is asking me to disable for the authentication at > the first stage , can you please let me know how can I disable for the > authentication at this stage (when the calls get through I will enable it > again) ? > > Thank you in advance > > > > [general] > ;register => toronto:welc...@osaka > > [osaka] > type=friend > ;username=toronto > ;authname=toronto > ;secret=welcome > context=osaka_incoming > host=192.168.0.139 > disallow=all > allow=alaw > > > ---fred > http://qxork.com > > > > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > Thank you for your reply . Please be informed that I want to simulate this case in the Laboratory , i.e. connecting my Asterisk sip to external sip server with the guidelines you sent me . Can you please propose for an Voip application sw that I can install on my MS Windows client and plays the external sip server side role ? It seems that Skype is not suitable for this case as it cannot be configured to play the role of external sip server . Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Connect my Asterisk to external sip?
On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote: > > > > On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner wrote: > > On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote: > > > Dear All > > I have an application that calls for my Asterisk sip to be connected to an > > external sip server for voip routing . Please be informed that my Asterisk > > sip is at @192.168.0.2 and the external sip is at @192.168.0.139 . To this > > end , I modified my sip.conf & extensions.conf as the followings : > > Under sip.conf : > > - > > [general] > > register => toronto:welc...@192.168.0.139/osaka > > [osaka] > > type=friend > > secret=welcome > > context=osaka_incoming > > host=dynamic > > disallow=all > > allow=alaw > > [6672019] > > type=friend > > host=dynamic > > context=phones > > > > Try this: > > [general] > register => toronto:welc...@osaka > > [osaka] > type=friend > username=toronto > authname=toronto > secret=welcome > context=osaka_incoming > host=192.168.0.139 > disallow=all > allow=alaw > > Although your error shows the other server does not allow register. What is > the other server? > > ---fred > http://qxork.com > > > Thank you for your reply . The other server is not an Asterisk sip server . > It is a sip server inside a softswitch from a third party vendor . As the > external sip server man is asking me to disable for the authentication at the > first stage , can you please let me know how can I disable for the > authentication at this stage (when the calls get through I will enable it > again) ? > Thank you in advance > [general] ;register => toronto:welc...@osaka [osaka] type=friend ;username=toronto ;authname=toronto ;secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw ---fred http://qxork.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Connect my Asterisk to external sip?
On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner wrote: > > On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote: > > > Dear All > > I have an application that calls for my Asterisk sip to be connected to > an external sip server for voip routing . Please be informed that my > Asterisk sip is at @192.168.0.2 and the external sip is at @192.168.0.139. To > this end , I modified my sip.conf & extensions.conf as the followings : > > Under sip.conf : > > - > > [general] > > register => toronto:welc...@192.168.0.139/osaka > > [osaka] > > type=friend > > secret=welcome > > context=osaka_incoming > > host=dynamic > > disallow=all > > allow=alaw > > [6672019] > > type=friend > > host=dynamic > > context=phones > > > > Try this: > > [general] > register => toronto:welc...@osaka > > [osaka] > type=friend > username=toronto > authname=toronto > secret=welcome > context=osaka_incoming > host=192.168.0.139 > disallow=all > allow=alaw > > Although your error shows the other server does not allow register. What is > the other server? > > ---fred > http://qxork.com > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > Thank you for your reply . The other server is not an Asterisk sip server . It is a sip server inside a softswitch from a third party vendor . As the external sip server man is asking me to disable for the authentication at the first stage , can you please let me know how can I disable for the authentication at this stage (when the calls get through I will enable it again) ? Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Connect my Asterisk to external sip?
On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote: > Dear All > I have an application that calls for my Asterisk sip to be connected to an > external sip server for voip routing . Please be informed that my Asterisk > sip is at @192.168.0.2 and the external sip is at @192.168.0.139 . To this > end , I modified my sip.conf & extensions.conf as the followings : > Under sip.conf : > - > [general] > register => toronto:welc...@192.168.0.139/osaka > [osaka] > type=friend > secret=welcome > context=osaka_incoming > host=dynamic > disallow=all > allow=alaw > [6672019] > type=friend > host=dynamic > context=phones > Try this: [general] register => toronto:welc...@osaka [osaka] type=friend username=toronto authname=toronto secret=welcome context=osaka_incoming host=192.168.0.139 disallow=all allow=alaw Although your error shows the other server does not allow register. What is the other server? ---fred http://qxork.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:Connect my Asterisk to external sip?
Dear All
I have an application that calls for my Asterisk sip to be connected to an
external sip server for voip routing . Please be informed that my Asterisk
sip is at @192.168.0.2 and the external sip is at @192.168.0.139 . To this
end , I modified my sip.conf & extensions.conf as the followings :
Under sip.conf :
-
[general]
register => toronto:welc...@192.168.0.139/osaka
[osaka]
type=friend
secret=welcome
context=osaka_incoming
host=dynamic
disallow=all
allow=alaw
[6672019]
type=friend
host=dynamic
context=phones
Under extensions.conf :
-
[osaka_incoming]
include=local-lines
[local-lines]
exten => _XXX,n,Dial(SIP/osaka/${EXTEN})
Please find attached the log captured when making calls (the call cannot get
through) .Can you please do me favor and let me know what is wrong in my sip
configuration ?
Let me thank you in advance
log-sip
Description: Binary data
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