Re: [asterisk-users] Call Transfers in SER + Asterisk architecture
Hi Ricardo, Could you post a specific example where your problem happens. That way would be easier for me to try to help you on this. Does asterisk is registred into SER , or you have trust based relationship between them? On 11/23/06, Ricardo Carvalho [EMAIL PROTECTED] wrote: Hi, I'm deploying a SER + Asterisk architecture, where SER is used as SIP registrar, and Asterisk is used for voicemail and PSTN gateway. This system is already able to make Call Transfers (Blind and Attended) internally between phones registered in SER, although, I can't make Call Transfers in some scenarios involving PSTN numbers (which need to pass through Asterisk). The problem is that when the REFER message (that carries the Refer-To number to whom the call should be transferred) gets to Asterisk, it replies with a 404 Not Found message, and the Call Transfer isn't established! Any ideas on how can I solve this problem? Thanks in advance, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Transfers in SER + Asterisk architecture
Hi Marco, Ser has IP of Asterisk server in his trusted table, Asterisk isn't registered in Ser. When Ser needs to use Asterisk, it simply rewrites the IP destination with Asterisk's IP, and routes them to him. For example, here's one failed attempt in transferring a call PSTN - VoIP - VoIP: PSTN Asterisk Ser phone_A phone_B |INVITE| | | | | -- | | | | | 100 Trying | | | | | --- | | | | | | INVITE| | | | | -- |INVITE | | | | | --- | | | | |100 trying | | | 100 trying | --- | | | 100 trying | --- | 180 Ringing | | | -- | 180 Ringing | --- | | | 180 Ringing | -- | | | | -- | | | | | ACK | | | | | --- | ACK | | | | | --- | ACK | | | | | --- | | | | RTP | | | | == | | | | | | | | | | REFER | | | | REFER| --- | | | | -- | | | | | 404 Not Found | | | | | --- | 404 Not Found | | | | | -- | | | | | | | In this example, phone_A answers the PSTN originated call, and wants to transfer the call to phone_B. A REFER message is than routed backwards to Asterisk, and he replies with those 404 Not Found messages. Phone_B never gets called! Should Asterisk be registered in Ser so this can work properly? How can that be done? Thanks, Ricardo. Marco Mouta wrote: Hi Ricardo, Could you post a specific example where your problem happens. That way would be easier for me to try to help you on this. Does asterisk is registred into SER , or you have trust based relationship between them? On 11/23/06, *Ricardo Carvalho* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, I'm deploying a SER + Asterisk architecture, where SER is used as SIP registrar, and Asterisk is used for voicemail and PSTN gateway. This system is already able to make Call Transfers (Blind and Attended) internally between phones registered in SER, although, I can't make Call Transfers in some scenarios involving PSTN numbers (which need to pass through Asterisk). The problem is that when the REFER message (that carries the Refer-To number to whom the call should be transferred) gets to Asterisk, it replies with a 404 Not Found message, and the Call Transfer isn't established! Any ideas on how can I solve this problem? Thanks in advance, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options
Re: [asterisk-users] Call Transfers in SER + Asterisk architecture
do you have created Asterisk views to SER database? Are you using sip realtime on asterisk? please post your extensions.conf. By the way, I'm Portuguese:) Qualquer coisa manda mail pode ser q consiga ajudar. On 11/24/06, Ricardo Carvalho [EMAIL PROTECTED] wrote: Hi Marco, Ser has IP of Asterisk server in his trusted table, Asterisk isn't registered in Ser. When Ser needs to use Asterisk, it simply rewrites the IP destination with Asterisk's IP, and routes them to him. For example, here's one failed attempt in transferring a call PSTN - VoIP - VoIP: PSTN Asterisk Ser phone_A phone_B |INVITE| | | | | -- | | | | | 100 Trying | | | | | --- | | | | | | INVITE| | | | | -- |INVITE | | | | | --- | | | | |100 trying | | | 100 trying | --- | | | 100 trying | --- | 180 Ringing | | | -- | 180 Ringing | --- | | | 180 Ringing | -- | | | | -- | | | | | ACK | | | | | --- | ACK | | | | | --- | ACK | | | | | --- | | | | RTP | | | | == | | | | | | | | | | REFER | | | | REFER| --- | | | | -- | | | | | 404 Not Found | | | | | --- | 404 Not Found | | | | | -- | | | | | | | In this example, phone_A answers the PSTN originated call, and wants to transfer the call to phone_B. A REFER message is than routed backwards to Asterisk, and he replies with those 404 Not Found messages. Phone_B never gets called! Should Asterisk be registered in Ser so this can work properly? How can that be done? Thanks, Ricardo. Marco Mouta wrote: Hi Ricardo, Could you post a specific example where your problem happens. That way would be easier for me to try to help you on this. Does asterisk is registred into SER , or you have trust based relationship between them? On 11/23/06, *Ricardo Carvalho* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, I'm deploying a SER + Asterisk architecture, where SER is used as SIP registrar, and Asterisk is used for voicemail and PSTN gateway. This system is already able to make Call Transfers (Blind and Attended) internally between phones registered in SER, although, I can't make Call Transfers in some scenarios involving PSTN numbers (which need to pass through Asterisk). The problem is that when the REFER message (that carries the Refer-To number to whom the call should be transferred) gets to Asterisk, it replies with a 404 Not Found message, and the Call Transfer isn't established! Any ideas on how can I solve this problem? Thanks in advance, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Transfers in SER + Asterisk architecture
Hi, I'm deploying a SER + Asterisk architecture, where SER is used as SIP registrar, and Asterisk is used for voicemail and PSTN gateway. This system is already able to make Call Transfers (Blind and Attended) internally between phones registered in SER, although, I can't make Call Transfers in some scenarios involving PSTN numbers (which need to pass through Asterisk). The problem is that when the REFER message (that carries the Refer-To number to whom the call should be transferred) gets to Asterisk, it replies with a 404 Not Found message, and the Call Transfer isn't established! Any ideas on how can I solve this problem? Thanks in advance, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
