Unfortunately, I don't think I will be able to provide a solution or workaround on this case. Our Engineering team has confirmed that Asterisk can't be configured to operate either way particularly; it reacts to what the other side is doing, whether it's strict or loose.
I suggest to seek assistance with A2Billing, perhaps they have seen this issue before and could be able to provide a solution.
Could you provide a detailed explanation about the difficulty that you are experiencing with Asterisk? I'm not familiarized with these terms "loose routing or strict routing" therefore I not sure how to properly answer your question.
Hello, this the difference between loose routing and strict routing: Loose and strict are different methods of routing SIP messages. Loose routing is new in SIP version 2. When you use loose routing, the R-URI is never changed and backwards compatibility is maintained with the older method (strict routing RFC2543).
The problem with strict routing is in the process of specifying the entire proxy set in the initial request before starting the SIP dialog. The processing throws away the information contained in the received R-URI. The behavior of UAs with outbound-proxy was problematic. The whole system would fail if there was a failure in one of the elements.
My problem is :
Iam using Asterisk 1.8 with A2Billing 2.0, i have connected a client to my switch with giving him a sip trunk, my customr uses opensips, the inerconnection between my switch asterisk and his switch opensips is ok, the problem is that my customer is an opertor and he has a lot of customers, when one of his customers dial a number to call it and after making the call, if this customer ( calling party) hangup first the switch of my customer ( the operator) opensips billed correctly the call and my customer got the traces of tha call but when the called party hangup first the call my customer can't get the traces of the call and he can't bill correctlyb the call because our switch asterisk didn't send in this case the necessary information in the header of the sip message ( called number, duration ,source ip) this is why i would like to know how to activate loose routing in order to force asterisk to send these informations like freeswitch and opensips. I hope i have explained sitaution to you. I hope you could help me.
Unfortunately, I don't think I will be able to provide a solution or workaround on this case. Our Engineering team has confirmed that Asterisk can't be configured to operate either way particularly; it reacts to what the other side is doing, whether it's strict or loose.
I suggest to seek assistance with A2Billing, perhaps they have seen this issue before and could be able to provide a solution.
All
Unfortunately, I don't think I will be able to provide a solution or workaround on this case. Our Engineering team has confirmed that Asterisk can't be configured to operate either way particularly; it reacts to what the other side is doing, whether it's strict or loose.
I suggest to seek assistance with A2Billing, perhaps they have seen this issue before and could be able to provide a solution.
All Answers
Could you provide a detailed explanation about the difficulty that you are experiencing with Asterisk? I'm not familiarized with these terms "loose routing or strict routing" therefore I not sure how to properly answer your question.
this the difference between loose routing and strict routing:
Loose and strict are different methods of routing SIP messages. Loose routing is new in SIP version 2. When you use loose routing, the R-URI is never changed and backwards compatibility is maintained with the older method (strict routing RFC2543).
The problem with strict routing is in the process of specifying the entire proxy set in the initial request before starting the SIP dialog. The processing throws away the information contained in the received R-URI. The behavior of UAs with outbound-proxy was problematic. The whole system would fail if there was a failure in one of the elements.
My problem is :
Iam using Asterisk 1.8 with A2Billing 2.0, i have connected a client to my switch with giving him a sip trunk, my customr uses opensips, the inerconnection between my switch asterisk and his switch opensips is ok, the problem is that my customer is an opertor and he has a lot of customers, when one of his customers dial a number to call it and after making the call, if this customer ( calling party) hangup first the switch of my customer ( the operator) opensips billed correctly the call and my customer got the traces of tha call but when the called party hangup first the call my customer can't get the traces of the call and he can't bill correctlyb the call because our switch asterisk didn't send in this case the necessary information in the header of the sip message ( called number, duration ,source ip) this is why i would like to know how to activate loose routing in order to force asterisk to send these informations like freeswitch and opensips. I hope i have explained sitaution to you. I hope you could help me.
Best Regards
Mettichi Bassem
Unfortunately, I don't think I will be able to provide a solution or workaround on this case. Our Engineering team has confirmed that Asterisk can't be configured to operate either way particularly; it reacts to what the other side is doing, whether it's strict or loose.
I suggest to seek assistance with A2Billing, perhaps they have seen this issue before and could be able to provide a solution.