hadi motamedi
2009-Dec-19 11:51 UTC
[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:welcome at 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 => _XXXXXXX,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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091219/d1292120/attachment-0001.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: log-sip Type: application/octet-stream Size: 17815 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20091219/d1292120/attachment-0001.obj
Fred Posner
2009-Dec-19 12:07 UTC
[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:welcome at 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:welcome at 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
hadi motamedi
2009-Dec-19 12:57 UTC
[asterisk-users] Inquiry:Connect my Asterisk to external sip?
On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner <fred at teamforrest.com> 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:welcome at 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:welcome at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091219/2e95af0a/attachment.htm
Fred Posner
2009-Dec-19 13:33 UTC
[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 <fred at teamforrest.com> 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:welcome at 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:welcome at 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:welcome at 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
hadi motamedi
2009-Dec-23 09:12 UTC
[asterisk-users] Inquiry:Connect my Asterisk to external sip?
On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner <fred at teamforrest.com> wrote:> > On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote: > > > > > > > > > On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner <fred at teamforrest.com> > 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:welcome at 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:welcome at 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:welcome at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091223/de416063/attachment.htm