[asterisk-users] call transfer to asterik.. asterisk as an end point
Hello All. I am having some trouble with call transfers when asterisk is the 2nd party called and I hope to benefit from your experience. I want to use asterisk for call park/pickup and have configured openser to relay calls made to ruri 700-720 to asterisk running on localhost:5069 Call flow: phone A calls phone B (both phones are polycom) Phone B answers then phone b user presses transfer and dials 700 asterisk plays back 701 as the parking lot location phone B user presses transfer again. at this time phone b is not disconnected from asterisk system phone A is also connected to asterisk and hears 702 as the parking lot location (as if asterisk places the user at priority 1 for that context) From phone C calling 702 will connect phone C to phone A. This was a specific example but this transfer problem is not limited to call park only. It happens any time asterisk is the second party called in call transfer. Thanks in advance for your help. -- Zahid On May 8, 2007, at 1:56 PM, Christian Schlatter wrote: I think I found out why this doesn't work as expected. After phone 1 receives REFER from phone 2, it sends a new INVITE to the asterisk server. This INVITE includes a Replaces: header that tells the receiver (asterisk) to replace an existing SIP dialog with the new one. RFC 3891 The SIP Replaces Header, Section 3 UAS Behavior, defines: the UA attempts to accept the new INVITE, reassign the user interface and other resources of the matched dialog to the new INVITE, and shut down the replaced dialog. But your SIP trace shows that asterisk doesn't shut down the replaced dialog (by sending a BYE), which is the reason why phone 2 does not get disconnected after hitting transfer the second time. Instead of creating a new call park slot (702) when phone 1 sends the Replaces: INVITE to asterisk, asterisk should be intelligent enough to figure out that this INVITE actually replaces the existing SIP dialog with phone 2. And asterisk should not create a new park slot 702 but directly put phone 1 on hold at park slot 701 and send a BYE to phone 2. Although asterisk supports the Replaces: header when used e.g. as a gateway, I have some doubts that the call park/pickup implementation does so too. Especially since it was designed to be used in PBX mode where asterisk acts as B2BUA for all involved call legs. Maybe this should be opened as a new feature/bug request on the asterisk bug tracker. Or maybe there is a asterisk setting that controls this behavior, I'm not really an asterisk expert myself ;-) -- The fact that an opinion has been widely held is no evidence that it is not utterly absurd; indeed, in view of the silliness of the majority of mankind, a widespread belief is more often likely to be foolish than sensible. -Bertrand Russell 8:00? 8:25? 8:40? Find a flick in no time with the Yahoo! Search movie showtime shortcut. http://tools.search.yahoo.com/shortcuts/#news___ --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] unable to register using SIP
Sorry for the duplicate post but I have hit a brick wall trying to get this to work. Is there anyone who can help me? I am having trouble trying to register with a voip provider using sip. I am able to connect using xlite softphone. in xlite i use domain/realm: providerdomain.com sip proxy: host.providerdomain.com:9000 this difference in domain and sip proxy host is whats causing problem for me. section from sip.conf [provider-out] type=peer secret=nn username=55439 fromuser=55439 fromdomain=providerdomain.com host=host.providerdomain.com port=9000 nat=No canreinvite=no when trying to make a call with xlite, i see that the to part in sip messages is using @xyz.provider.com where as in asterisk it uses host.xyz.provider.com (sip proxy host, NOT the domain/realm host). Another thing i notice is that if i use nat=yes then asterisk doesn't seem to be using the port=9000 and uses default 5060 for remote host. What am i doing wrong or missing? Can someone point me in the right direction? What will be the register = line for this? Also can someone provide info on [authentication] in sip.conf? any help will be greatly appreciated. thanks. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] sip registration question
I am a newbie and am having trouble trying to register with a voip provider using sip. I am able to connect using xlite softphone. in xlite i use domain/realm: providerdomain.com sip proxy: host.providerdomain.com:9000 this difference in domain and sip proxy host is whats causing problem for me. section from sip.conf [provider-out] type=peer secret=nn username=55439 fromuser=55439 fromdomain=providerdomain.com host=host.providerdomain.com port=9000 nat=No canreinvite=no when trying to make a call with xlite, i see that the to part in sip messages is using @xyz.provider.com where as in asterisk it uses host.xyz.provider.com (sip proxy host, NOT the domain/realm host). Another thing i notice is that if i use nat=yes then asterisk doesn't seem to be using the port=9000 and uses default 5060 for remote host. What am i doing wrong or missing? Can someone point me in the right direction? What will be the register = line for this? Also can someone provide info on [authentication] in sip.conf? any help will be greatly appreciated. thanks. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] sip registration question
I am a newbie and am having trouble trying to register with a voip provider using sip. I am able to connect using xlite softphone. in xlite i use domain/realm: providerdomain.com sip proxy: host.providerdomain.com:9000 this difference in domain and sip proxy host is whats causing problem for me. section from sip.conf [provider-out] type=peer secret=nn username=55439 fromuser=55439 fromdomain=providerdomain.com host=host.providerdomain.com port=9000 nat=No canreinvite=no when trying to make a call with xlite, i see that the to part in sip messages is using @xyz.provider.com where as in asterisk it uses host.xyz.provider.com (sip proxy host, NOT the domain/realm host). Another thing i notice is that if i use nat=yes then asterisk doesn't seem to be using the port=9000 and uses default 5060 for remote host. What am i doing wrong or missing? Can someone point me in the right direction? What will be the register = line for this? Also can someone provide info on [authentication] in sip.conf? any help will be greatly appreciated. thanks. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] need help configuring dlink dvg-1120M
Hi, I have a dlink dvg-1120M (mgcp) box that i will like to use with asterisk. Is it possible? has anyone done that? Here's a link to the product page at dlink. http://support.dlink.com/products/view.asp?productid=DVG%2D1120M Also, does anyone has or know where to get the firmware for Dlink DVG-1120S (sip model)? thanks. -- Zahid __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
