We are seeing some instances when we send a invite from the B-Leg back to FS and FS passes the invite to the A-Leg that the dialog on the A-Leg has not fully completed yet causing a return of a 491 from the A-Leg side causing the call to be disconnected. wanted to see if anyone else has seen something like this while running FS and if anyone had any suggestions on a fix? A-Leg Invite into Freeswitch 100 Trying back from FS to A-Leg 180 Ringing from FS to A-Leg 200 OK from FS to A-Leg at 15:08:14.638799 Invite from FS to A-Leg (From B-Leg below) at 15:08:14.749515 100 Giving a try form A-Leg to FS at 15:08:14.749757 491 From A-Leg to FS at 15:08:14.780968 ACK from FS to A-Leg at 15:08:14.781102 ACK from FS to A-Leg at 15:08:14.797143 BYE from A-LEG to FS B-Leg Invite from Freeswitch to B-Leg 100 Giving a try from B-Leg 180 Ringing from B-Leg 200 OK from B-Leg at 15:08:14.635670 ACK from FS to B-Leg at 15:08:14.637044 Invite from B-Leg to FS at 15:08:14.748623 100 Trying from FS to B-Leg at 15:08:14.748954 491 from FS to B-Leg at 15:08:14.782169 ACK from B-Leg to FS at 15:08:14.782372 BYE from FS to B-Leg
Hi, RE*-Invite from B-Leg to FS at 15:08:14.748623* ? Is you phone at B leg trying to do some kind of reinvite stuff..Disable this from FS like as in asterisk we had an option canreinvite=no likewise something in FS. I really hope that blocking RE-INVITES from FS will block new invites from the B leg and hence you'r call will stay established. BTW since both legs are in race condition,which do you think wins the race? j/k :P Thanks Sammy On Sat, Aug 25, 2012 at 12:53 AM, Mike Burlingame <mike.burlingame@me.com>wrote:
We are seeing some instances when we send a invite from the B-Leg back to FS and FS passes the invite to the A-Leg that the dialog on the A-Leg has not fully completed yet causing a return of a 491 from the A-Leg side causing the call to be disconnected. wanted to see if anyone else has seen something like this while running FS and if anyone had any suggestions on a fix?
A-Leg Invite into Freeswitch 100 Trying back from FS to A-Leg 180 Ringing from FS to A-Leg 200 OK from FS to A-Leg at 15:08:14.638799 Invite from FS to A-Leg (From B-Leg below) at 15:08:14.749515 100 Giving a try form A-Leg to FS at 15:08:14.749757 491 From A-Leg to FS at 15:08:14.780968 ACK from FS to A-Leg at 15:08:14.781102 ACK from FS to A-Leg at 15:08:14.797143 BYE from A-LEG to FS
B-Leg Invite from Freeswitch to B-Leg 100 Giving a try from B-Leg 180 Ringing from B-Leg 200 OK from B-Leg at 15:08:14.635670 ACK from FS to B-Leg at 15:08:14.637044 Invite from B-Leg to FS at 15:08:14.748623 100 Trying from FS to B-Leg at 15:08:14.748954 491 from FS to B-Leg at 15:08:14.782169 ACK from B-Leg to FS at 15:08:14.782372 BYE from FS to B-Leg
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Wait longer before you send it? Fs is not a proxy so you have to be more careful. On Aug 24, 2012 2:54 PM, "Mike Burlingame" <mike.burlingame@me.com> wrote:
We are seeing some instances when we send a invite from the B-Leg back to FS and FS passes the invite to the A-Leg that the dialog on the A-Leg has not fully completed yet causing a return of a 491 from the A-Leg side causing the call to be disconnected. wanted to see if anyone else has seen something like this while running FS and if anyone had any suggestions on a fix?
A-Leg Invite into Freeswitch 100 Trying back from FS to A-Leg 180 Ringing from FS to A-Leg 200 OK from FS to A-Leg at 15:08:14.638799 Invite from FS to A-Leg (From B-Leg below) at 15:08:14.749515 100 Giving a try form A-Leg to FS at 15:08:14.749757 491 From A-Leg to FS at 15:08:14.780968 ACK from FS to A-Leg at 15:08:14.781102 ACK from FS to A-Leg at 15:08:14.797143 BYE from A-LEG to FS
B-Leg Invite from Freeswitch to B-Leg 100 Giving a try from B-Leg 180 Ringing from B-Leg 200 OK from B-Leg at 15:08:14.635670 ACK from FS to B-Leg at 15:08:14.637044 Invite from B-Leg to FS at 15:08:14.748623 100 Trying from FS to B-Leg at 15:08:14.748954 491 from FS to B-Leg at 15:08:14.782169 ACK from B-Leg to FS at 15:08:14.782372 BYE from FS to B-Leg
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The M6 on the other side does not have any options to delay the invite also broadsoft support is saying due to the ACK on the B-Leg (Broadsoft side) that they see no reason to delay the invite. This only happens on 1 out of 30-45 calls and does not seem to be resource related I will try to work with Broadsoft to see if they can delay the invite Thanks for your feedback Sent from my iPad On Aug 25, 2012, at 11:00 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Wait longer before you send it?
Fs is not a proxy so you have to be more careful.
On Aug 24, 2012 2:54 PM, "Mike Burlingame" <mike.burlingame@me.com> wrote: We are seeing some instances when we send a invite from the B-Leg back to FS and FS passes the invite to the A-Leg that the dialog on the A-Leg has not fully completed yet causing a return of a 491 from the A-Leg side causing the call to be disconnected. wanted to see if anyone else has seen something like this while running FS and if anyone had any suggestions on a fix?
A-Leg Invite into Freeswitch 100 Trying back from FS to A-Leg 180 Ringing from FS to A-Leg 200 OK from FS to A-Leg at 15:08:14.638799 Invite from FS to A-Leg (From B-Leg below) at 15:08:14.749515 100 Giving a try form A-Leg to FS at 15:08:14.749757 491 From A-Leg to FS at 15:08:14.780968 ACK from FS to A-Leg at 15:08:14.781102 ACK from FS to A-Leg at 15:08:14.797143 BYE from A-LEG to FS
B-Leg Invite from Freeswitch to B-Leg 100 Giving a try from B-Leg 180 Ringing from B-Leg 200 OK from B-Leg at 15:08:14.635670 ACK from FS to B-Leg at 15:08:14.637044 Invite from B-Leg to FS at 15:08:14.748623 100 Trying from FS to B-Leg at 15:08:14.748954 491 from FS to B-Leg at 15:08:14.782169 ACK from B-Leg to FS at 15:08:14.782372 BYE from FS to B-Leg
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I would like to add an update to this and maybe someone has some other suggestions - It does look like that FS is disconnecting the call with cause code 31 [NORMAL_UNSPECIFIED] I did not notice the time stamp difference on the Bye when I first looked at the log's The basis of the issue still stands a 491 Results in the B-Leg being disconnected On Aug 24, 2012, at 12:53 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
We are seeing some instances when we send a invite from the B-Leg back to FS and FS passes the invite to the A-Leg that the dialog on the A-Leg has not fully completed yet causing a return of a 491 from the A-Leg side causing the call to be disconnected. wanted to see if anyone else has seen something like this while running FS and if anyone had any suggestions on a fix?
A-Leg Invite into Freeswitch 100 Trying back from FS to A-Leg 180 Ringing from FS to A-Leg 200 OK from FS to A-Leg at 15:08:14.638799 Invite from FS to A-Leg (From B-Leg below) at 15:08:14.749515 100 Giving a try form A-Leg to FS at 15:08:14.749757 491 From A-Leg to FS at 15:08:14.780968 ACK from FS to A-Leg at 15:08:14.781102 ACK from FS to A-Leg at 15:08:14.797143 BYE from A-LEG to FS 15:11:10.791963 481 Call Does Not Exist back to A-LEG
B-Leg Invite from Freeswitch to B-Leg 100 Giving a try from B-Leg 180 Ringing from B-Leg 200 OK from B-Leg at 15:08:14.635670 ACK from FS to B-Leg at 15:08:14.637044 Invite from B-Leg to FS at 15:08:14.748623 100 Trying from FS to B-Leg at 15:08:14.748954 491 from FS to B-Leg at 15:08:14.782169 ACK from B-Leg to FS at 15:08:14.782372 BYE from FS to B-Leg at 15:08:14.790714 with Cause Code 31
Hi, Can you please manage a proper sip trace for this ! I'd really love to troubleshoot and analyse the sip traces. What is the equipment on your A-leg ? BR Sammy On Fri, Aug 31, 2012 at 3:28 AM, Mike Burlingame <mike.burlingame@me.com>wrote:
I would like to add an update to this and maybe someone has some other suggestions - It does look like that FS is disconnecting the call with cause code 31 [NORMAL_UNSPECIFIED] I did not notice the time stamp difference on the Bye when I first looked at the log's
The basis of the issue still stands a 491 Results in the B-Leg being disconnected
On Aug 24, 2012, at 12:53 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
We are seeing some instances when we send a invite from the B-Leg back to FS and FS passes the invite to the A-Leg that the dialog on the A-Leg has not fully completed yet causing a return of a 491 from the A-Leg side causing the call to be disconnected. wanted to see if anyone else has seen something like this while running FS and if anyone had any suggestions on a fix?
A-Leg Invite into Freeswitch 100 Trying back from FS to A-Leg 180 Ringing from FS to A-Leg 200 OK from FS to A-Leg at 15:08:14.638799 Invite from FS to A-Leg (From B-Leg below) at 15:08:14.749515 100 Giving a try form A-Leg to FS at 15:08:14.749757 491 From A-Leg to FS at 15:08:14.780968 ACK from FS to A-Leg at 15:08:14.781102 ACK from FS to A-Leg at 15:08:14.797143 BYE from A-LEG to FS 15:11:10.791963
481 Call Does Not Exist back to A-LEG
B-Leg Invite from Freeswitch to B-Leg 100 Giving a try from B-Leg 180 Ringing from B-Leg 200 OK from B-Leg at 15:08:14.635670 ACK from FS to B-Leg at 15:08:14.637044 Invite from B-Leg to FS at 15:08:14.748623 100 Trying from FS to B-Leg at 15:08:14.748954 491 from FS to B-Leg at 15:08:14.782169 ACK from B-Leg to FS at 15:08:14.782372 BYE from FS to B-Leg at 15:08:14.790714 with Cause Code 31
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Sure I will put logs and sip dump up tomorrow A-Leg == Sonus B-Leg == Broadsoft M6 Sent from my iPhone 4S On Aug 30, 2012, at 10:24 PM, SamyGo <govoiper@gmail.com> wrote:
Hi, Can you please manage a proper sip trace for this ! I'd really love to troubleshoot and analyse the sip traces. What is the equipment on your A-leg ?
BR Sammy
On Fri, Aug 31, 2012 at 3:28 AM, Mike Burlingame <mike.burlingame@me.com> wrote:
I would like to add an update to this and maybe someone has some other suggestions - It does look like that FS is disconnecting the call with cause code 31 [NORMAL_UNSPECIFIED] I did not notice the time stamp difference on the Bye when I first looked at the log's
The basis of the issue still stands a 491 Results in the B-Leg being disconnected
On Aug 24, 2012, at 12:53 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
We are seeing some instances when we send a invite from the B-Leg back to FS and FS passes the invite to the A-Leg that the dialog on the A-Leg has not fully completed yet causing a return of a 491 from the A-Leg side causing the call to be disconnected. wanted to see if anyone else has seen something like this while running FS and if anyone had any suggestions on a fix?
A-Leg Invite into Freeswitch 100 Trying back from FS to A-Leg 180 Ringing from FS to A-Leg 200 OK from FS to A-Leg at 15:08:14.638799 Invite from FS to A-Leg (From B-Leg below) at 15:08:14.749515 100 Giving a try form A-Leg to FS at 15:08:14.749757 491 From A-Leg to FS at 15:08:14.780968 ACK from FS to A-Leg at 15:08:14.781102 ACK from FS to A-Leg at 15:08:14.797143 BYE from A-LEG to FS 15:11:10.791963 481 Call Does Not Exist back to A-LEG
B-Leg Invite from Freeswitch to B-Leg 100 Giving a try from B-Leg 180 Ringing from B-Leg 200 OK from B-Leg at 15:08:14.635670 ACK from FS to B-Leg at 15:08:14.637044 Invite from B-Leg to FS at 15:08:14.748623 100 Trying from FS to B-Leg at 15:08:14.748954 491 from FS to B-Leg at 15:08:14.782169 ACK from B-Leg to FS at 15:08:14.782372 BYE from FS to B-Leg at 15:08:14.790714 with Cause Code 31
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OK, and can you see whats happening inside the SONUS when it gets the second invite (debug logs etc)? Can you control the M6 to stop re-invites !? On Fri, Aug 31, 2012 at 10:29 AM, Mike Burlingame <mike.burlingame@me.com>wrote:
Sure I will put logs and sip dump up tomorrow A-Leg == Sonus B-Leg == Broadsoft M6
Sent from my iPhone 4S
On Aug 30, 2012, at 10:24 PM, SamyGo <govoiper@gmail.com> wrote:
Hi, Can you please manage a proper sip trace for this ! I'd really love to troubleshoot and analyse the sip traces. What is the equipment on your A-leg ?
BR Sammy
On Fri, Aug 31, 2012 at 3:28 AM, Mike Burlingame <mike.burlingame@me.com>wrote:
I would like to add an update to this and maybe someone has some other suggestions - It does look like that FS is disconnecting the call with cause code 31 [NORMAL_UNSPECIFIED] I did not notice the time stamp difference on the Bye when I first looked at the log's
The basis of the issue still stands a 491 Results in the B-Leg being disconnected
On Aug 24, 2012, at 12:53 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
We are seeing some instances when we send a invite from the B-Leg back to FS and FS passes the invite to the A-Leg that the dialog on the A-Leg has not fully completed yet causing a return of a 491 from the A-Leg side causing the call to be disconnected. wanted to see if anyone else has seen something like this while running FS and if anyone had any suggestions on a fix?
A-Leg Invite into Freeswitch 100 Trying back from FS to A-Leg 180 Ringing from FS to A-Leg 200 OK from FS to A-Leg at 15:08:14.638799 Invite from FS to A-Leg (From B-Leg below) at 15:08:14.749515 100 Giving a try form A-Leg to FS at 15:08:14.749757 491 From A-Leg to FS at 15:08:14.780968 ACK from FS to A-Leg at 15:08:14.781102 ACK from FS to A-Leg at 15:08:14.797143 BYE from A-LEG to FS 15:11:10.791963
481 Call Does Not Exist back to A-LEG
B-Leg Invite from Freeswitch to B-Leg 100 Giving a try from B-Leg 180 Ringing from B-Leg 200 OK from B-Leg at 15:08:14.635670 ACK from FS to B-Leg at 15:08:14.637044 Invite from B-Leg to FS at 15:08:14.748623 100 Trying from FS to B-Leg at 15:08:14.748954 491 from FS to B-Leg at 15:08:14.782169 ACK from B-Leg to FS at 15:08:14.782372 BYE from FS to B-Leg at 15:08:14.790714 with Cause Code 31
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The Sonus and M6 are out of my control they belong to my upstream and downstream so I have no access to them Broadsoft support will not work this issue because FS ACKs the 200 OK so dialog is completed at least on the B-Leg so that's when the M6 sends the 2nd invite however the dialog has not completed on the A-Leg but FS still passes the Invite along resulting in the 491 Sent from my iPhone 4S On Aug 30, 2012, at 11:42 PM, SamyGo <govoiper@gmail.com> wrote:
OK, and can you see whats happening inside the SONUS when it gets the second invite (debug logs etc)? Can you control the M6 to stop re-invites !?
On Fri, Aug 31, 2012 at 10:29 AM, Mike Burlingame <mike.burlingame@me.com> wrote:
Sure I will put logs and sip dump up tomorrow A-Leg == Sonus B-Leg == Broadsoft M6
Sent from my iPhone 4S
On Aug 30, 2012, at 10:24 PM, SamyGo <govoiper@gmail.com> wrote:
Hi, Can you please manage a proper sip trace for this ! I'd really love to troubleshoot and analyse the sip traces. What is the equipment on your A-leg ?
BR Sammy
On Fri, Aug 31, 2012 at 3:28 AM, Mike Burlingame <mike.burlingame@me.com> wrote:
I would like to add an update to this and maybe someone has some other suggestions - It does look like that FS is disconnecting the call with cause code 31 [NORMAL_UNSPECIFIED] I did not notice the time stamp difference on the Bye when I first looked at the log's
The basis of the issue still stands a 491 Results in the B-Leg being disconnected
On Aug 24, 2012, at 12:53 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
We are seeing some instances when we send a invite from the B-Leg back to FS and FS passes the invite to the A-Leg that the dialog on the A-Leg has not fully completed yet causing a return of a 491 from the A-Leg side causing the call to be disconnected. wanted to see if anyone else has seen something like this while running FS and if anyone had any suggestions on a fix?
A-Leg Invite into Freeswitch 100 Trying back from FS to A-Leg 180 Ringing from FS to A-Leg 200 OK from FS to A-Leg at 15:08:14.638799 Invite from FS to A-Leg (From B-Leg below) at 15:08:14.749515 100 Giving a try form A-Leg to FS at 15:08:14.749757 491 From A-Leg to FS at 15:08:14.780968 ACK from FS to A-Leg at 15:08:14.781102 ACK from FS to A-Leg at 15:08:14.797143 BYE from A-LEG to FS 15:11:10.791963 481 Call Does Not Exist back to A-LEG
B-Leg Invite from Freeswitch to B-Leg 100 Giving a try from B-Leg 180 Ringing from B-Leg 200 OK from B-Leg at 15:08:14.635670 ACK from FS to B-Leg at 15:08:14.637044 Invite from B-Leg to FS at 15:08:14.748623 100 Trying from FS to B-Leg at 15:08:14.748954 491 from FS to B-Leg at 15:08:14.782169 ACK from B-Leg to FS at 15:08:14.782372 BYE from FS to B-Leg at 15:08:14.790714 with Cause Code 31
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Well that narrow downs the possible solutions to doing some stuff on FS only ! - I really think that Stopping or applying a work-around for the re-invite from Broadsoft will get this fixed. I did some googling on the SIP 491 code and re-invites in FS and mostly they point to session timers or some media_bridging option in dialplan. On Fri, Aug 31, 2012 at 12:03 PM, Mike Burlingame <mike.burlingame@me.com>wrote:
The Sonus and M6 are out of my control they belong to my upstream and downstream so I have no access to them
Broadsoft support will not work this issue because FS ACKs the 200 OK so dialog is completed at least on the B-Leg so that's when the M6 sends the 2nd invite however the dialog has not completed on the A-Leg but FS still passes the Invite along resulting in the 491
Sent from my iPhone 4S
On Aug 30, 2012, at 11:42 PM, SamyGo <govoiper@gmail.com> wrote:
OK, and can you see whats happening inside the SONUS when it gets the second invite (debug logs etc)? Can you control the M6 to stop re-invites !?
On Fri, Aug 31, 2012 at 10:29 AM, Mike Burlingame <mike.burlingame@me.com>wrote:
Sure I will put logs and sip dump up tomorrow A-Leg == Sonus B-Leg == Broadsoft M6
Sent from my iPhone 4S
On Aug 30, 2012, at 10:24 PM, SamyGo <govoiper@gmail.com> wrote:
Hi, Can you please manage a proper sip trace for this ! I'd really love to troubleshoot and analyse the sip traces. What is the equipment on your A-leg ?
BR Sammy
On Fri, Aug 31, 2012 at 3:28 AM, Mike Burlingame <mike.burlingame@me.com>wrote:
I would like to add an update to this and maybe someone has some other suggestions - It does look like that FS is disconnecting the call with cause code 31 [NORMAL_UNSPECIFIED] I did not notice the time stamp difference on the Bye when I first looked at the log's
The basis of the issue still stands a 491 Results in the B-Leg being disconnected
On Aug 24, 2012, at 12:53 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
We are seeing some instances when we send a invite from the B-Leg back to FS and FS passes the invite to the A-Leg that the dialog on the A-Leg has not fully completed yet causing a return of a 491 from the A-Leg side causing the call to be disconnected. wanted to see if anyone else has seen something like this while running FS and if anyone had any suggestions on a fix?
A-Leg Invite into Freeswitch 100 Trying back from FS to A-Leg 180 Ringing from FS to A-Leg 200 OK from FS to A-Leg at 15:08:14.638799 Invite from FS to A-Leg (From B-Leg below) at 15:08:14.749515 100 Giving a try form A-Leg to FS at 15:08:14.749757 491 From A-Leg to FS at 15:08:14.780968 ACK from FS to A-Leg at 15:08:14.781102 ACK from FS to A-Leg at 15:08:14.797143 BYE from A-LEG to FS 15:11:10.791963
481 Call Does Not Exist back to A-LEG
B-Leg Invite from Freeswitch to B-Leg 100 Giving a try from B-Leg 180 Ringing from B-Leg 200 OK from B-Leg at 15:08:14.635670 ACK from FS to B-Leg at 15:08:14.637044 Invite from B-Leg to FS at 15:08:14.748623 100 Trying from FS to B-Leg at 15:08:14.748954 491 from FS to B-Leg at 15:08:14.782169 ACK from B-Leg to FS at 15:08:14.782372 BYE from FS to B-Leg at 15:08:14.790714 with Cause Code 31
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I do have an OpenSIPs proxy in front of and behind the A and B Leg's - but the issue still seems to be in FS Sonus_UAC --> OpenSIPs --> Freeswitch --> OpenSIPs --> Broadsoft_M6 Log and SIP trace from FS Log file posted to PasteBin: http://pastebin.freeswitch.org/19799 On Aug 31, 2012, at 12:14 AM, SamyGo <govoiper@gmail.com> wrote:
Well that narrow downs the possible solutions to doing some stuff on FS only ! - I really think that Stopping or applying a work-around for the re-invite from Broadsoft will get this fixed. I did some googling on the SIP 491 code and re-invites in FS and mostly they point to session timers or some media_bridging option in dialplan.
On Fri, Aug 31, 2012 at 12:03 PM, Mike Burlingame <mike.burlingame@me.com> wrote: The Sonus and M6 are out of my control they belong to my upstream and downstream so I have no access to them
Broadsoft support will not work this issue because FS ACKs the 200 OK so dialog is completed at least on the B-Leg so that's when the M6 sends the 2nd invite however the dialog has not completed on the A-Leg but FS still passes the Invite along resulting in the 491
Sent from my iPhone 4S
On Aug 30, 2012, at 11:42 PM, SamyGo <govoiper@gmail.com> wrote:
OK, and can you see whats happening inside the SONUS when it gets the second invite (debug logs etc)? Can you control the M6 to stop re-invites !?
On Fri, Aug 31, 2012 at 10:29 AM, Mike Burlingame <mike.burlingame@me.com> wrote: Sure I will put logs and sip dump up tomorrow A-Leg == Sonus B-Leg == Broadsoft M6
Sent from my iPhone 4S
On Aug 30, 2012, at 10:24 PM, SamyGo <govoiper@gmail.com> wrote:
Hi, Can you please manage a proper sip trace for this ! I'd really love to troubleshoot and analyse the sip traces. What is the equipment on your A-leg ?
BR Sammy
On Fri, Aug 31, 2012 at 3:28 AM, Mike Burlingame <mike.burlingame@me.com> wrote: I would like to add an update to this and maybe someone has some other suggestions - It does look like that FS is disconnecting the call with cause code 31 [NORMAL_UNSPECIFIED] I did not notice the time stamp difference on the Bye when I first looked at the log's
The basis of the issue still stands a 491 Results in the B-Leg being disconnected
On Aug 24, 2012, at 12:53 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
We are seeing some instances when we send a invite from the B-Leg back to FS and FS passes the invite to the A-Leg that the dialog on the A-Leg has not fully completed yet causing a return of a 491 from the A-Leg side causing the call to be disconnected. wanted to see if anyone else has seen something like this while running FS and if anyone had any suggestions on a fix?
A-Leg Invite into Freeswitch 100 Trying back from FS to A-Leg 180 Ringing from FS to A-Leg 200 OK from FS to A-Leg at 15:08:14.638799 Invite from FS to A-Leg (From B-Leg below) at 15:08:14.749515 100 Giving a try form A-Leg to FS at 15:08:14.749757 491 From A-Leg to FS at 15:08:14.780968 ACK from FS to A-Leg at 15:08:14.781102 ACK from FS to A-Leg at 15:08:14.797143 BYE from A-LEG to FS 15:11:10.791963 481 Call Does Not Exist back to A-LEG
B-Leg Invite from Freeswitch to B-Leg 100 Giving a try from B-Leg 180 Ringing from B-Leg 200 OK from B-Leg at 15:08:14.635670 ACK from FS to B-Leg at 15:08:14.637044 Invite from B-Leg to FS at 15:08:14.748623 100 Trying from FS to B-Leg at 15:08:14.748954 491 from FS to B-Leg at 15:08:14.782169 ACK from B-Leg to FS at 15:08:14.782372 BYE from FS to B-Leg at 15:08:14.790714 with Cause Code 31
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An idea I had you can test if you update to latest set the variable sip_wait_for_aleg_ack on the b leg in the {} when you make the outbound call <action application="bridge" data="{sip_wait_for_aleg_ack=true}sofia/internal/foo@bar.com"/> In theory this should make the B leg delay sending the ACK until it sees that the A leg has recv'd an ack On Fri, Aug 31, 2012 at 2:33 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
I do have an OpenSIPs proxy in front of and behind the A and B Leg's - but the issue still seems to be in FS
Sonus_UAC --> OpenSIPs --> Freeswitch --> OpenSIPs --> Broadsoft_M6
Log and SIP trace from FS Log file posted to PasteBin: http://pastebin.freeswitch.org/19799
On Aug 31, 2012, at 12:14 AM, SamyGo <govoiper@gmail.com> wrote:
Well that narrow downs the possible solutions to doing some stuff on FS only ! - I really think that Stopping or applying a work-around for the re-invite from Broadsoft will get this fixed. I did some googling on the SIP 491 code and re-invites in FS and mostly they point to session timers or some media_bridging option in dialplan.
On Fri, Aug 31, 2012 at 12:03 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
The Sonus and M6 are out of my control they belong to my upstream and downstream so I have no access to them
Broadsoft support will not work this issue because FS ACKs the 200 OK so dialog is completed at least on the B-Leg so that's when the M6 sends the 2nd invite however the dialog has not completed on the A-Leg but FS still passes the Invite along resulting in the 491
Sent from my iPhone 4S
On Aug 30, 2012, at 11:42 PM, SamyGo <govoiper@gmail.com> wrote:
OK, and can you see whats happening inside the SONUS when it gets the second invite (debug logs etc)? Can you control the M6 to stop re-invites !?
On Fri, Aug 31, 2012 at 10:29 AM, Mike Burlingame <mike.burlingame@me.com> wrote:
Sure I will put logs and sip dump up tomorrow A-Leg == Sonus B-Leg == Broadsoft M6
Sent from my iPhone 4S
On Aug 30, 2012, at 10:24 PM, SamyGo <govoiper@gmail.com> wrote:
Hi, Can you please manage a proper sip trace for this ! I'd really love to troubleshoot and analyse the sip traces. What is the equipment on your A-leg ?
BR Sammy
On Fri, Aug 31, 2012 at 3:28 AM, Mike Burlingame <mike.burlingame@me.com> wrote:
I would like to add an update to this and maybe someone has some other suggestions - It does look like that FS is disconnecting the call with cause code 31 [NORMAL_UNSPECIFIED] I did not notice the time stamp difference on the Bye when I first looked at the log's
The basis of the issue still stands a 491 Results in the B-Leg being disconnected
On Aug 24, 2012, at 12:53 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
We are seeing some instances when we send a invite from the B-Leg back to FS and FS passes the invite to the A-Leg that the dialog on the A-Leg has not fully completed yet causing a return of a 491 from the A-Leg side causing the call to be disconnected. wanted to see if anyone else has seen something like this while running FS and if anyone had any suggestions on a fix?
A-Leg Invite into Freeswitch 100 Trying back from FS to A-Leg 180 Ringing from FS to A-Leg 200 OK from FS to A-Leg at 15:08:14.638799 Invite from FS to A-Leg (From B-Leg below) at 15:08:14.749515 100 Giving a try form A-Leg to FS at 15:08:14.749757 491 From A-Leg to FS at 15:08:14.780968 ACK from FS to A-Leg at 15:08:14.781102 ACK from FS to A-Leg at 15:08:14.797143 BYE from A-LEG to FS 15:11:10.791963
481 Call Does Not Exist back to A-LEG
B-Leg Invite from Freeswitch to B-Leg 100 Giving a try from B-Leg 180 Ringing from B-Leg 200 OK from B-Leg at 15:08:14.635670 ACK from FS to B-Leg at 15:08:14.637044 Invite from B-Leg to FS at 15:08:14.748623 100 Trying from FS to B-Leg at 15:08:14.748954 491 from FS to B-Leg at 15:08:14.782169 ACK from B-Leg to FS at 15:08:14.782372 BYE from FS to B-Leg at 15:08:14.790714 with Cause Code 31
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Cool I will nail that up on my test box and see if that works Thanks for the suggestion Sent from my iPhone 4S On Aug 31, 2012, at 12:50 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
An idea I had you can test if you update to latest
set the variable sip_wait_for_aleg_ack on the b leg in the {} when you make the outbound call
<action application="bridge" data="{sip_wait_for_aleg_ack=true}sofia/internal/foo@bar.com"/>
In theory this should make the B leg delay sending the ACK until it sees that the A leg has recv'd an ack
On Fri, Aug 31, 2012 at 2:33 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
I do have an OpenSIPs proxy in front of and behind the A and B Leg's - but the issue still seems to be in FS
Sonus_UAC --> OpenSIPs --> Freeswitch --> OpenSIPs --> Broadsoft_M6
Log and SIP trace from FS Log file posted to PasteBin: http://pastebin.freeswitch.org/19799
On Aug 31, 2012, at 12:14 AM, SamyGo <govoiper@gmail.com> wrote:
Well that narrow downs the possible solutions to doing some stuff on FS only ! - I really think that Stopping or applying a work-around for the re-invite from Broadsoft will get this fixed. I did some googling on the SIP 491 code and re-invites in FS and mostly they point to session timers or some media_bridging option in dialplan.
On Fri, Aug 31, 2012 at 12:03 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
The Sonus and M6 are out of my control they belong to my upstream and downstream so I have no access to them
Broadsoft support will not work this issue because FS ACKs the 200 OK so dialog is completed at least on the B-Leg so that's when the M6 sends the 2nd invite however the dialog has not completed on the A-Leg but FS still passes the Invite along resulting in the 491
Sent from my iPhone 4S
On Aug 30, 2012, at 11:42 PM, SamyGo <govoiper@gmail.com> wrote:
OK, and can you see whats happening inside the SONUS when it gets the second invite (debug logs etc)? Can you control the M6 to stop re-invites !?
On Fri, Aug 31, 2012 at 10:29 AM, Mike Burlingame <mike.burlingame@me.com> wrote:
Sure I will put logs and sip dump up tomorrow A-Leg == Sonus B-Leg == Broadsoft M6
Sent from my iPhone 4S
On Aug 30, 2012, at 10:24 PM, SamyGo <govoiper@gmail.com> wrote:
Hi, Can you please manage a proper sip trace for this ! I'd really love to troubleshoot and analyse the sip traces. What is the equipment on your A-leg ?
BR Sammy
On Fri, Aug 31, 2012 at 3:28 AM, Mike Burlingame <mike.burlingame@me.com> wrote:
I would like to add an update to this and maybe someone has some other suggestions - It does look like that FS is disconnecting the call with cause code 31 [NORMAL_UNSPECIFIED] I did not notice the time stamp difference on the Bye when I first looked at the log's
The basis of the issue still stands a 491 Results in the B-Leg being disconnected
On Aug 24, 2012, at 12:53 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
We are seeing some instances when we send a invite from the B-Leg back to FS and FS passes the invite to the A-Leg that the dialog on the A-Leg has not fully completed yet causing a return of a 491 from the A-Leg side causing the call to be disconnected. wanted to see if anyone else has seen something like this while running FS and if anyone had any suggestions on a fix?
A-Leg Invite into Freeswitch 100 Trying back from FS to A-Leg 180 Ringing from FS to A-Leg 200 OK from FS to A-Leg at 15:08:14.638799 Invite from FS to A-Leg (From B-Leg below) at 15:08:14.749515 100 Giving a try form A-Leg to FS at 15:08:14.749757 491 From A-Leg to FS at 15:08:14.780968 ACK from FS to A-Leg at 15:08:14.781102 ACK from FS to A-Leg at 15:08:14.797143 BYE from A-LEG to FS 15:11:10.791963
481 Call Does Not Exist back to A-LEG
B-Leg Invite from Freeswitch to B-Leg 100 Giving a try from B-Leg 180 Ringing from B-Leg 200 OK from B-Leg at 15:08:14.635670 ACK from FS to B-Leg at 15:08:14.637044 Invite from B-Leg to FS at 15:08:14.748623 100 Trying from FS to B-Leg at 15:08:14.748954 491 from FS to B-Leg at 15:08:14.782169 ACK from B-Leg to FS at 15:08:14.782372 BYE from FS to B-Leg at 15:08:14.790714 with Cause Code 31
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
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On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame <mike.burlingame@me.com>wrote:
Cool I will nail that up on my test box and see if that works
Please report back on whether it works or not and then be prepared to pay the wiki tax. :) I'll be glad to assist with getting this documented although I think you're in the best position to give that documentation some real-world context.
-MC
No worries I will be out this weekend for the long weekend I will work on getting the test box upgraded and a test case setup on Tuesday I will report back the results mid to late next week and provided everything works as I hope it will I will be happy to pay the Wiki tax :) On Aug 31, 2012, at 1:42 PM, Michael Collins <msc@freeswitch.org> wrote:
On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame <mike.burlingame@me.com> wrote: Cool I will nail that up on my test box and see if that works
Please report back on whether it works or not and then be prepared to pay the wiki tax. :) I'll be glad to assist with getting this documented although I think you're in the best position to give that documentation some real-world context.
-MC _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
as promised here is the update testing and enabling {sip_wait_for_aleg_ack=true} causes a log delay before the audio starts to flow to the B-Leg endpoint with multi 200 OK's being sent to FS from the B-Leg as well as an abnormally long delay in getting an ACK from the A-Leg I can open a JIRA case on this and provide the console log file / PCAP's ect if that would help Call Flow with out {sip_wait_for_aleg_ack=true} 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 0.000652 FreeSwitch -> A-LEG SIP Status: 100 Trying 0.042837 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.043059 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.290947 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.292890 FreeSwitch -> A-LEG SIP Status: 180 Ringing 0.490220 B-Leg -> FreeSwitch SIP Status: 180 Ringing 4.481038 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 4.482310 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 4.483474 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session description 4.533691 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@FreeSwitch:5070;transport=udp 19.953061 A-LEG -> FreeSwitch SIP Request: BYE sip:+13605551212@FreeSwitch:5070;transport=udp 19.954592 FreeSwitch -> A-LEG SIP Status: 200 OK 19.955454 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 19.956014 B-Leg -> FreeSwitch SIP Status: 200 OK Call Flow with {sip_wait_for_aleg_ack=true} enabled 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 0.000651 FreeSwitch -> A-LEG SIP Status: 100 Trying 0.039455 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.039709 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.244269 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.245607 FreeSwitch -> A-LEG SIP Status: 180 Ringing 0.363325 B-Leg -> FreeSwitch SIP Status: 180 Ringing 4.718173 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 4.918915 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.117917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.518902 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 6.318053 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 7.917921 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 11.118917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 15.102917 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 15.104488 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session description 15.173265 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@FreeSwitch:5070;transport=udp 29.566504 A-LEG -> FreeSwitch SIP Request: BYE sip:+13605551212@FreeSwitch:5070;transport=udp 29.568902 FreeSwitch -> A-LEG SIP Status: 200 OK 29.570603 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 29.571240 B-Leg -> FreeSwitch SIP Status: 200 OK On Aug 31, 2012, at 1:52 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
No worries I will be out this weekend for the long weekend I will work on getting the test box upgraded and a test case setup on Tuesday I will report back the results mid to late next week and provided everything works as I hope it will I will be happy to pay the Wiki tax :)
On Aug 31, 2012, at 1:42 PM, Michael Collins <msc@freeswitch.org> wrote:
On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame <mike.burlingame@me.com> wrote: Cool I will nail that up on my test box and see if that works
Please report back on whether it works or not and then be prepared to pay the wiki tax. :) I'll be glad to assist with getting this documented although I think you're in the best position to give that documentation some real-world context.
-MC _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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update and try again, if it still doesn't work open a jira On Wed, Sep 5, 2012 at 12:18 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
as promised here is the update testing and enabling {sip_wait_for_aleg_ack=true} causes a log delay before the audio starts to flow to the B-Leg endpoint with multi 200 OK's being sent to FS from the B-Leg as well as an abnormally long delay in getting an ACK from the A-Leg
I can open a JIRA case on this and provide the console log file / PCAP's ect if that would help
Call Flow with out {sip_wait_for_aleg_ack=true} 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 0.000652 FreeSwitch -> A-LEG SIP Status: 100 Trying 0.042837 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.043059 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.290947 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.292890 FreeSwitch -> A-LEG SIP Status: 180 Ringing 0.490220 B-Leg -> FreeSwitch SIP Status: 180 Ringing 4.481038 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 4.482310 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 4.483474 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session description 4.533691 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@FreeSwitch:5070;transport=udp 19.953061 A-LEG -> FreeSwitch SIP Request: BYE sip:+13605551212@FreeSwitch:5070;transport=udp 19.954592 FreeSwitch -> A-LEG SIP Status: 200 OK 19.955454 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 19.956014 B-Leg -> FreeSwitch SIP Status: 200 OK
Call Flow with {sip_wait_for_aleg_ack=true} enabled 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 0.000651 FreeSwitch -> A-LEG SIP Status: 100 Trying 0.039455 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.039709 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.244269 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.245607 FreeSwitch -> A-LEG SIP Status: 180 Ringing 0.363325 B-Leg -> FreeSwitch SIP Status: 180 Ringing 4.718173 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 4.918915 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.117917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.518902 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 6.318053 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 7.917921 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 11.118917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 15.102917 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 15.104488 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session description 15.173265 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@FreeSwitch:5070;transport=udp 29.566504 A-LEG -> FreeSwitch SIP Request: BYE sip:+13605551212@FreeSwitch:5070;transport=udp 29.568902 FreeSwitch -> A-LEG SIP Status: 200 OK 29.570603 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 29.571240 B-Leg -> FreeSwitch SIP Status: 200 OK
On Aug 31, 2012, at 1:52 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
No worries I will be out this weekend for the long weekend I will work on getting the test box upgraded and a test case setup on Tuesday I will report back the results mid to late next week and provided everything works as I hope it will I will be happy to pay the Wiki tax :)
On Aug 31, 2012, at 1:42 PM, Michael Collins <msc@freeswitch.org> wrote:
On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
Cool I will nail that up on my test box and see if that works
Please report back on whether it works or not and then be prepared to pay the wiki tax. :) I'll be glad to assist with getting this documented although I think you're in the best position to give that documentation some real-world context.
-MC _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
The change seems to have broken the ability for the call to connect - would you like me to open a jira up with the current log files or before the change was made today? commit d45db898339e1b2212f5befff1af714abcec034f Author: Anthony Minessale <anthm@freeswitch.org> Date: Wed Sep 5 13:11:32 2012 -0500 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 0.002715 FreeSwitch -> A-LEG SIP Status: 100 Trying 0.062694 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.062976 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.238255 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.239830 FreeSwitch -> A-LEG SIP Status: 180 Ringing 0.456828 B-Leg -> FreeSwitch SIP Status: 180 Ringing 4.642993 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.637738 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.638091 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 6.037804 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 6.839818 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 8.438750 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 11.638797 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 24.925970 FreeSwitch -> A-LEG SIP Status: 480 Temporarily Unavailable 24.926257 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@A-LEG:5060 24.926321 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 24.926580 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 24.927029 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist 27.078016 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 27.078467 FreeSwitch -> A-LEG SIP Status: 100 Trying 27.123233 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 27.123445 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 27.315802 B-Leg -> FreeSwitch SIP Status: 180 Ringing 27.317391 FreeSwitch -> A-LEG SIP Status: 180 Ringing 27.529142 B-Leg -> FreeSwitch SIP Status: 180 Ringing 31.520118 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 31.769831 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 31.920832 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 32.319816 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 33.120808 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 34.720813 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 37.920852 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 49.362647 A-LEG -> FreeSwitch SIP Request: CANCEL sip:+13605551212@A-LEG:5060 49.362952 FreeSwitch -> A-LEG SIP Status: 200 OK 49.363196 FreeSwitch -> A-LEG SIP Status: 487 Request Terminated 49.363307 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@A-LEG:5060 49.365975 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 49.366171 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 49.366522 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist On Sep 5, 2012, at 11:10 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
update and try again, if it still doesn't work open a jira
On Wed, Sep 5, 2012 at 12:18 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
as promised here is the update testing and enabling {sip_wait_for_aleg_ack=true} causes a log delay before the audio starts to flow to the B-Leg endpoint with multi 200 OK's being sent to FS from the B-Leg as well as an abnormally long delay in getting an ACK from the A-Leg
I can open a JIRA case on this and provide the console log file / PCAP's ect if that would help
Call Flow with out {sip_wait_for_aleg_ack=true} 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 0.000652 FreeSwitch -> A-LEG SIP Status: 100 Trying 0.042837 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.043059 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.290947 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.292890 FreeSwitch -> A-LEG SIP Status: 180 Ringing 0.490220 B-Leg -> FreeSwitch SIP Status: 180 Ringing 4.481038 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 4.482310 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 4.483474 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session description 4.533691 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@FreeSwitch:5070;transport=udp 19.953061 A-LEG -> FreeSwitch SIP Request: BYE sip:+13605551212@FreeSwitch:5070;transport=udp 19.954592 FreeSwitch -> A-LEG SIP Status: 200 OK 19.955454 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 19.956014 B-Leg -> FreeSwitch SIP Status: 200 OK
Call Flow with {sip_wait_for_aleg_ack=true} enabled 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 0.000651 FreeSwitch -> A-LEG SIP Status: 100 Trying 0.039455 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.039709 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.244269 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.245607 FreeSwitch -> A-LEG SIP Status: 180 Ringing 0.363325 B-Leg -> FreeSwitch SIP Status: 180 Ringing 4.718173 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 4.918915 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.117917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.518902 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 6.318053 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 7.917921 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 11.118917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 15.102917 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 15.104488 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session description 15.173265 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@FreeSwitch:5070;transport=udp 29.566504 A-LEG -> FreeSwitch SIP Request: BYE sip:+13605551212@FreeSwitch:5070;transport=udp 29.568902 FreeSwitch -> A-LEG SIP Status: 200 OK 29.570603 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 29.571240 B-Leg -> FreeSwitch SIP Status: 200 OK
On Aug 31, 2012, at 1:52 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
No worries I will be out this weekend for the long weekend I will work on getting the test box upgraded and a test case setup on Tuesday I will report back the results mid to late next week and provided everything works as I hope it will I will be happy to pay the Wiki tax :)
On Aug 31, 2012, at 1:42 PM, Michael Collins <msc@freeswitch.org> wrote:
On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
Cool I will nail that up on my test box and see if that works
Please report back on whether it works or not and then be prepared to pay the wiki tax. :) I'll be glad to assist with getting this documented although I think you're in the best position to give that documentation some real-world context.
-MC _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
ok, update one more time, if it still does not work just go right to jira with the latest (not before today's changes) On Wed, Sep 5, 2012 at 1:37 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
The change seems to have broken the ability for the call to connect - would you like me to open a jira up with the current log files or before the change was made today?
commit d45db898339e1b2212f5befff1af714abcec034f Author: Anthony Minessale <anthm@freeswitch.org> Date: Wed Sep 5 13:11:32 2012 -0500
0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 0.002715 FreeSwitch -> A-LEG SIP Status: 100 Trying 0.062694 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.062976 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.238255 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.239830 FreeSwitch -> A-LEG SIP Status: 180 Ringing 0.456828 B-Leg -> FreeSwitch SIP Status: 180 Ringing 4.642993 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.637738 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.638091 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 6.037804 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 6.839818 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 8.438750 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 11.638797 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 24.925970 FreeSwitch -> A-LEG SIP Status: 480 Temporarily Unavailable 24.926257 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@A-LEG:5060 24.926321 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 24.926580 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 24.927029 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist 27.078016 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 27.078467 FreeSwitch -> A-LEG SIP Status: 100 Trying 27.123233 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 27.123445 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 27.315802 B-Leg -> FreeSwitch SIP Status: 180 Ringing 27.317391 FreeSwitch -> A-LEG SIP Status: 180 Ringing 27.529142 B-Leg -> FreeSwitch SIP Status: 180 Ringing 31.520118 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 31.769831 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 31.920832 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 32.319816 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 33.120808 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 34.720813 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 37.920852 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 49.362647 A-LEG -> FreeSwitch SIP Request: CANCEL sip:+13605551212@A-LEG:5060 49.362952 FreeSwitch -> A-LEG SIP Status: 200 OK 49.363196 FreeSwitch -> A-LEG SIP Status: 487 Request Terminated 49.363307 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@A-LEG:5060 49.365975 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 49.366171 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 49.366522 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist
On Sep 5, 2012, at 11:10 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
update and try again, if it still doesn't work open a jira
On Wed, Sep 5, 2012 at 12:18 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
as promised here is the update testing and enabling {sip_wait_for_aleg_ack=true} causes a log delay before the audio starts to flow to the B-Leg endpoint with multi 200 OK's being sent to FS from the B-Leg as well as an abnormally long delay in getting an ACK from the A-Leg
I can open a JIRA case on this and provide the console log file / PCAP's ect if that would help
Call Flow with out {sip_wait_for_aleg_ack=true} 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 0.000652 FreeSwitch -> A-LEG SIP Status: 100 Trying 0.042837 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.043059 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.290947 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.292890 FreeSwitch -> A-LEG SIP Status: 180 Ringing 0.490220 B-Leg -> FreeSwitch SIP Status: 180 Ringing 4.481038 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 4.482310 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 4.483474 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session description 4.533691 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@FreeSwitch:5070;transport=udp 19.953061 A-LEG -> FreeSwitch SIP Request: BYE sip:+13605551212@FreeSwitch:5070;transport=udp 19.954592 FreeSwitch -> A-LEG SIP Status: 200 OK 19.955454 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 19.956014 B-Leg -> FreeSwitch SIP Status: 200 OK
Call Flow with {sip_wait_for_aleg_ack=true} enabled 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 0.000651 FreeSwitch -> A-LEG SIP Status: 100 Trying 0.039455 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.039709 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.244269 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.245607 FreeSwitch -> A-LEG SIP Status: 180 Ringing 0.363325 B-Leg -> FreeSwitch SIP Status: 180 Ringing 4.718173 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 4.918915 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.117917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.518902 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 6.318053 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 7.917921 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 11.118917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 15.102917 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 15.104488 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session description 15.173265 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@FreeSwitch:5070;transport=udp 29.566504 A-LEG -> FreeSwitch SIP Request: BYE sip:+13605551212@FreeSwitch:5070;transport=udp 29.568902 FreeSwitch -> A-LEG SIP Status: 200 OK 29.570603 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 29.571240 B-Leg -> FreeSwitch SIP Status: 200 OK
On Aug 31, 2012, at 1:52 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
No worries I will be out this weekend for the long weekend I will work on getting the test box upgraded and a test case setup on Tuesday I will report back the results mid to late next week and provided everything works as I hope it will I will be happy to pay the Wiki tax :)
On Aug 31, 2012, at 1:42 PM, Michael Collins <msc@freeswitch.org> wrote:
On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
Cool I will nail that up on my test box and see if that works
Please report back on whether it works or not and then be prepared to pay the wiki tax. :) I'll be glad to assist with getting this documented although I think you're in the best position to give that documentation some real-world context.
-MC _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Looks much much better Thank you -- Now to conduct more testing 0.000000 A-Leg -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-Leg:5060, with session description 0.000639 FreeSwitch -> A-Leg SIP Status: 100 Trying 0.051096 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.051351 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.286387 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.288216 FreeSwitch -> A-Leg SIP Status: 180 Ringing 0.475452 B-Leg -> FreeSwitch SIP Status: 180 Ringing 5.311144 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.313775 FreeSwitch -> A-Leg SIP/SDP Status: 200 OK, with session description 5.385087 A-Leg -> FreeSwitch SIP Request: ACK sip:+13605551212@FreeSwitch:5070;transport=udp 5.385796 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 12.027026 A-Leg -> FreeSwitch SIP Request: BYE sip:+13605551212@FreeSwitch:5070;transport=udp 12.029232 FreeSwitch -> A-Leg SIP Status: 200 OK 12.030707 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 12.033888 B-Leg -> FreeSwitch SIP Status: 200 OK On Sep 5, 2012, at 12:01 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
ok,
update one more time, if it still does not work just go right to jira with the latest (not before today's changes)
On Wed, Sep 5, 2012 at 1:37 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
The change seems to have broken the ability for the call to connect - would you like me to open a jira up with the current log files or before the change was made today?
commit d45db898339e1b2212f5befff1af714abcec034f Author: Anthony Minessale <anthm@freeswitch.org> Date: Wed Sep 5 13:11:32 2012 -0500
0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 0.002715 FreeSwitch -> A-LEG SIP Status: 100 Trying 0.062694 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.062976 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.238255 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.239830 FreeSwitch -> A-LEG SIP Status: 180 Ringing 0.456828 B-Leg -> FreeSwitch SIP Status: 180 Ringing 4.642993 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.637738 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.638091 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 6.037804 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 6.839818 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 8.438750 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 11.638797 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 24.925970 FreeSwitch -> A-LEG SIP Status: 480 Temporarily Unavailable 24.926257 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@A-LEG:5060 24.926321 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 24.926580 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 24.927029 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist 27.078016 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 27.078467 FreeSwitch -> A-LEG SIP Status: 100 Trying 27.123233 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 27.123445 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 27.315802 B-Leg -> FreeSwitch SIP Status: 180 Ringing 27.317391 FreeSwitch -> A-LEG SIP Status: 180 Ringing 27.529142 B-Leg -> FreeSwitch SIP Status: 180 Ringing 31.520118 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 31.769831 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 31.920832 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 32.319816 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 33.120808 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 34.720813 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 37.920852 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 49.362647 A-LEG -> FreeSwitch SIP Request: CANCEL sip:+13605551212@A-LEG:5060 49.362952 FreeSwitch -> A-LEG SIP Status: 200 OK 49.363196 FreeSwitch -> A-LEG SIP Status: 487 Request Terminated 49.363307 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@A-LEG:5060 49.365975 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 49.366171 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 49.366522 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist
On Sep 5, 2012, at 11:10 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
update and try again, if it still doesn't work open a jira
On Wed, Sep 5, 2012 at 12:18 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
as promised here is the update testing and enabling {sip_wait_for_aleg_ack=true} causes a log delay before the audio starts to flow to the B-Leg endpoint with multi 200 OK's being sent to FS from the B-Leg as well as an abnormally long delay in getting an ACK from the A-Leg
I can open a JIRA case on this and provide the console log file / PCAP's ect if that would help
Call Flow with out {sip_wait_for_aleg_ack=true} 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 0.000652 FreeSwitch -> A-LEG SIP Status: 100 Trying 0.042837 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.043059 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.290947 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.292890 FreeSwitch -> A-LEG SIP Status: 180 Ringing 0.490220 B-Leg -> FreeSwitch SIP Status: 180 Ringing 4.481038 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 4.482310 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 4.483474 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session description 4.533691 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@FreeSwitch:5070;transport=udp 19.953061 A-LEG -> FreeSwitch SIP Request: BYE sip:+13605551212@FreeSwitch:5070;transport=udp 19.954592 FreeSwitch -> A-LEG SIP Status: 200 OK 19.955454 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 19.956014 B-Leg -> FreeSwitch SIP Status: 200 OK
Call Flow with {sip_wait_for_aleg_ack=true} enabled 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 0.000651 FreeSwitch -> A-LEG SIP Status: 100 Trying 0.039455 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.039709 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.244269 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.245607 FreeSwitch -> A-LEG SIP Status: 180 Ringing 0.363325 B-Leg -> FreeSwitch SIP Status: 180 Ringing 4.718173 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 4.918915 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.117917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.518902 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 6.318053 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 7.917921 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 11.118917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 15.102917 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 15.104488 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session description 15.173265 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@FreeSwitch:5070;transport=udp 29.566504 A-LEG -> FreeSwitch SIP Request: BYE sip:+13605551212@FreeSwitch:5070;transport=udp 29.568902 FreeSwitch -> A-LEG SIP Status: 200 OK 29.570603 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 29.571240 B-Leg -> FreeSwitch SIP Status: 200 OK
On Aug 31, 2012, at 1:52 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
No worries I will be out this weekend for the long weekend I will work on getting the test box upgraded and a test case setup on Tuesday I will report back the results mid to late next week and provided everything works as I hope it will I will be happy to pay the Wiki tax :)
On Aug 31, 2012, at 1:42 PM, Michael Collins <msc@freeswitch.org> wrote:
On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
Cool I will nail that up on my test box and see if that works
Please report back on whether it works or not and then be prepared to pay the wiki tax. :) I'll be glad to assist with getting this documented although I think you're in the best position to give that documentation some real-world context.
-MC _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Anthony, shouldn't it be the default behavior? I see no reason to send ACK to B leg before we get ACK from A-leg. On Wednesday 05 September 2012 12:28:42 Mike Burlingame wrote:
Looks much much better Thank you -- Now to conduct more testing
0.000000 A-Leg -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-Leg:5060, with session description 0.000639 FreeSwitch -> A-Leg SIP Status: 100 Trying 0.051096 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.051351 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.286387 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.288216 FreeSwitch -> A-Leg SIP Status: 180 Ringing 0.475452 B-Leg -> FreeSwitch SIP Status: 180 Ringing 5.311144 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.313775 FreeSwitch -> A-Leg SIP/SDP Status: 200 OK, with session description 5.385087 A-Leg -> FreeSwitch SIP Request: ACK sip:+13605551212@FreeSwitch:5070;transport=udp 5.385796 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 12.027026 A-Leg -> FreeSwitch SIP Request: BYE sip:+13605551212@FreeSwitch:5070;transport=udp 12.029232 FreeSwitch -> A-Leg SIP Status: 200 OK 12.030707 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 12.033888 B-Leg -> FreeSwitch SIP Status: 200 OK
On Sep 5, 2012, at 12:01 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
ok,
update one more time, if it still does not work just go right to jira with the latest (not before today's changes)
On Wed, Sep 5, 2012 at 1:37 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
The change seems to have broken the ability for the call to connect - would you like me to open a jira up with the current log files or before the change was made today?
commit d45db898339e1b2212f5befff1af714abcec034f Author: Anthony Minessale <anthm@freeswitch.org> Date: Wed Sep 5 13:11:32 2012 -0500
0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 0.002715 FreeSwitch -> A-LEG SIP Status: 100 Trying 0.062694 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.062976 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.238255 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.239830 FreeSwitch -> A-LEG SIP Status: 180 Ringing 0.456828 B-Leg -> FreeSwitch SIP Status: 180 Ringing 4.642993 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.637738 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.638091 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 6.037804 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 6.839818 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 8.438750 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description>> 11.638797 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 24.925970 FreeSwitch -> A-LEG SIP Status: 480 Temporarily Unavailable 24.926257 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@A-LEG:5060 24.926321 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 24.926580 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 24.927029 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist 27.078016 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 27.078467 FreeSwitch -> A-LEG SIP Status: 100 Trying 27.123233 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 27.123445 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 27.315802 B-Leg -> FreeSwitch SIP Status: 180 Ringing 27.317391 FreeSwitch -> A-LEG SIP Status: 180 Ringing 27.529142 B-Leg -> FreeSwitch SIP Status: 180 Ringing 31.520118 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 31.769831 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 31.920832 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 32.319816 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 33.120808 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 34.720813 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 37.920852 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 49.362647 A-LEG -> FreeSwitch SIP Request: CANCEL sip:+13605551212@A-LEG:5060 49.362952 FreeSwitch -> A-LEG SIP Status: 200 OK 49.363196 FreeSwitch -> A-LEG SIP Status: 487 Request Terminated 49.363307 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@A-LEG:5060 49.365975 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 49.366171 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 49.366522 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist>> On Sep 5, 2012, at 11:10 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
update and try again, if it still doesn't work open a jira
On Wed, Sep 5, 2012 at 12:18 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
as promised here is the update testing and enabling {sip_wait_for_aleg_ack=true} causes a log delay before the audio starts to flow to the B-Leg endpoint with multi 200 OK's being sent to FS from the B-Leg as well as an abnormally long delay in getting an ACK from the A-Leg
I can open a JIRA case on this and provide the console log file / PCAP's ect if that would help
Call Flow with out {sip_wait_for_aleg_ack=true} 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 0.000652 FreeSwitch -> A-LEG SIP Status: 100 Trying 0.042837 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.043059 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.290947 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.292890 FreeSwitch -> A-LEG SIP Status: 180 Ringing 0.490220 B-Leg -> FreeSwitch SIP Status: 180 Ringing 4.481038 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 4.482310 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 4.483474 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session description 4.533691 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@FreeSwitch:5070;transport=udp 19.953061 A-LEG -> FreeSwitch SIP Request: BYE sip:+13605551212@FreeSwitch:5070;transport=udp 19.954592 FreeSwitch -> A-LEG SIP Status: 200 OK 19.955454 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 19.956014 B-Leg -> FreeSwitch SIP Status: 200 OK
Call Flow with {sip_wait_for_aleg_ack=true} enabled 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 0.000651 FreeSwitch -> A-LEG SIP Status: 100 Trying 0.039455 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.039709 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.244269 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.245607 FreeSwitch -> A-LEG SIP Status: 180 Ringing 0.363325 B-Leg -> FreeSwitch SIP Status: 180 Ringing 4.718173 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 4.918915 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.117917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.518902 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 6.318053 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 7.917921 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 11.118917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 15.102917 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 15.104488 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session description 15.173265 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@FreeSwitch:5070;transport=udp 29.566504 A-LEG -> FreeSwitch SIP Request: BYE sip:+13605551212@FreeSwitch:5070;transport=udp 29.568902 FreeSwitch -> A-LEG SIP Status: 200 OK 29.570603 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 29.571240 B-Leg -> FreeSwitch SIP Status: 200 OK
On Aug 31, 2012, at 1:52 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
No worries I will be out this weekend for the long weekend I will work on getting the test box upgraded and a test case setup on Tuesday I will report back the results mid to late next week and provided everything works as I hope it will I will be happy to pay the Wiki tax :)
On Aug 31, 2012, at 1:42 PM, Michael Collins <msc@freeswitch.org> wrote:
On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame <mike.burlingame@me.com>
wrote:
Cool I will nail that up on my test box and see if that works
Please report back on whether it works or not and then be prepared to pay the wiki tax. :) I'll be glad to assist with getting this documented although I think you're in the best position to give that documentation some real-world context.
-MC _______________________________________________________________________ __ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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_______________________________________________________________________ __ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
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________________________________________________________________________ _ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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After about 20K test calls this seems to have addressed the issue - I will keep running my test's for today and put this box in a production environment tomorrow to validate it still holds up with load. I will report back after that is completed. Thanks On Sep 5, 2012, at 12:28 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
Looks much much better Thank you -- Now to conduct more testing
0.000000 A-Leg -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-Leg:5060, with session description 0.000639 FreeSwitch -> A-Leg SIP Status: 100 Trying 0.051096 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.051351 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.286387 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.288216 FreeSwitch -> A-Leg SIP Status: 180 Ringing 0.475452 B-Leg -> FreeSwitch SIP Status: 180 Ringing 5.311144 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.313775 FreeSwitch -> A-Leg SIP/SDP Status: 200 OK, with session description 5.385087 A-Leg -> FreeSwitch SIP Request: ACK sip:+13605551212@FreeSwitch:5070;transport=udp 5.385796 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 12.027026 A-Leg -> FreeSwitch SIP Request: BYE sip:+13605551212@FreeSwitch:5070;transport=udp 12.029232 FreeSwitch -> A-Leg SIP Status: 200 OK 12.030707 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 12.033888 B-Leg -> FreeSwitch SIP Status: 200 OK
On Sep 5, 2012, at 12:01 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
ok,
update one more time, if it still does not work just go right to jira with the latest (not before today's changes)
On Wed, Sep 5, 2012 at 1:37 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
The change seems to have broken the ability for the call to connect - would you like me to open a jira up with the current log files or before the change was made today?
commit d45db898339e1b2212f5befff1af714abcec034f Author: Anthony Minessale <anthm@freeswitch.org> Date: Wed Sep 5 13:11:32 2012 -0500
0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 0.002715 FreeSwitch -> A-LEG SIP Status: 100 Trying 0.062694 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.062976 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.238255 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.239830 FreeSwitch -> A-LEG SIP Status: 180 Ringing 0.456828 B-Leg -> FreeSwitch SIP Status: 180 Ringing 4.642993 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.637738 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.638091 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 6.037804 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 6.839818 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 8.438750 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 11.638797 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 24.925970 FreeSwitch -> A-LEG SIP Status: 480 Temporarily Unavailable 24.926257 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@A-LEG:5060 24.926321 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 24.926580 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 24.927029 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist 27.078016 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 27.078467 FreeSwitch -> A-LEG SIP Status: 100 Trying 27.123233 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 27.123445 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 27.315802 B-Leg -> FreeSwitch SIP Status: 180 Ringing 27.317391 FreeSwitch -> A-LEG SIP Status: 180 Ringing 27.529142 B-Leg -> FreeSwitch SIP Status: 180 Ringing 31.520118 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 31.769831 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 31.920832 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 32.319816 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 33.120808 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 34.720813 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 37.920852 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 49.362647 A-LEG -> FreeSwitch SIP Request: CANCEL sip:+13605551212@A-LEG:5060 49.362952 FreeSwitch -> A-LEG SIP Status: 200 OK 49.363196 FreeSwitch -> A-LEG SIP Status: 487 Request Terminated 49.363307 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@A-LEG:5060 49.365975 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 49.366171 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 49.366522 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist
On Sep 5, 2012, at 11:10 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
update and try again, if it still doesn't work open a jira
On Wed, Sep 5, 2012 at 12:18 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
as promised here is the update testing and enabling {sip_wait_for_aleg_ack=true} causes a log delay before the audio starts to flow to the B-Leg endpoint with multi 200 OK's being sent to FS from the B-Leg as well as an abnormally long delay in getting an ACK from the A-Leg
I can open a JIRA case on this and provide the console log file / PCAP's ect if that would help
Call Flow with out {sip_wait_for_aleg_ack=true} 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 0.000652 FreeSwitch -> A-LEG SIP Status: 100 Trying 0.042837 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.043059 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.290947 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.292890 FreeSwitch -> A-LEG SIP Status: 180 Ringing 0.490220 B-Leg -> FreeSwitch SIP Status: 180 Ringing 4.481038 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 4.482310 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 4.483474 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session description 4.533691 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@FreeSwitch:5070;transport=udp 19.953061 A-LEG -> FreeSwitch SIP Request: BYE sip:+13605551212@FreeSwitch:5070;transport=udp 19.954592 FreeSwitch -> A-LEG SIP Status: 200 OK 19.955454 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 19.956014 B-Leg -> FreeSwitch SIP Status: 200 OK
Call Flow with {sip_wait_for_aleg_ack=true} enabled 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 0.000651 FreeSwitch -> A-LEG SIP Status: 100 Trying 0.039455 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.039709 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.244269 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.245607 FreeSwitch -> A-LEG SIP Status: 180 Ringing 0.363325 B-Leg -> FreeSwitch SIP Status: 180 Ringing 4.718173 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 4.918915 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.117917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.518902 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 6.318053 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 7.917921 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 11.118917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 15.102917 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 15.104488 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session description 15.173265 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@FreeSwitch:5070;transport=udp 29.566504 A-LEG -> FreeSwitch SIP Request: BYE sip:+13605551212@FreeSwitch:5070;transport=udp 29.568902 FreeSwitch -> A-LEG SIP Status: 200 OK 29.570603 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 29.571240 B-Leg -> FreeSwitch SIP Status: 200 OK
On Aug 31, 2012, at 1:52 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
No worries I will be out this weekend for the long weekend I will work on getting the test box upgraded and a test case setup on Tuesday I will report back the results mid to late next week and provided everything works as I hope it will I will be happy to pay the Wiki tax :)
On Aug 31, 2012, at 1:42 PM, Michael Collins <msc@freeswitch.org> wrote:
On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
Cool I will nail that up on my test box and see if that works
Please report back on whether it works or not and then be prepared to pay the wiki tax. :) I'll be glad to assist with getting this documented although I think you're in the best position to give that documentation some real-world context.
-MC _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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after putting the server in production and loading it up it just randomly crashed I am updating to the latest git will try to capture a core dump if one is created on exit nothing in the logs point to anything it was handling calls and just exited. However on a positive note the switch sip_wait_for_aleg_ack=true seems to do as expected with all my tests. On Sep 6, 2012, at 12:50 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
After about 20K test calls this seems to have addressed the issue - I will keep running my test's for today and put this box in a production environment tomorrow to validate it still holds up with load. I will report back after that is completed.
Thanks
On Sep 5, 2012, at 12:28 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
Looks much much better Thank you -- Now to conduct more testing
0.000000 A-Leg -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-Leg:5060, with session description 0.000639 FreeSwitch -> A-Leg SIP Status: 100 Trying 0.051096 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.051351 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.286387 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.288216 FreeSwitch -> A-Leg SIP Status: 180 Ringing 0.475452 B-Leg -> FreeSwitch SIP Status: 180 Ringing 5.311144 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.313775 FreeSwitch -> A-Leg SIP/SDP Status: 200 OK, with session description 5.385087 A-Leg -> FreeSwitch SIP Request: ACK sip:+13605551212@FreeSwitch:5070;transport=udp 5.385796 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 12.027026 A-Leg -> FreeSwitch SIP Request: BYE sip:+13605551212@FreeSwitch:5070;transport=udp 12.029232 FreeSwitch -> A-Leg SIP Status: 200 OK 12.030707 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 12.033888 B-Leg -> FreeSwitch SIP Status: 200 OK
On Sep 5, 2012, at 12:01 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
ok,
update one more time, if it still does not work just go right to jira with the latest (not before today's changes)
On Wed, Sep 5, 2012 at 1:37 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
The change seems to have broken the ability for the call to connect - would you like me to open a jira up with the current log files or before the change was made today?
commit d45db898339e1b2212f5befff1af714abcec034f Author: Anthony Minessale <anthm@freeswitch.org> Date: Wed Sep 5 13:11:32 2012 -0500
0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 0.002715 FreeSwitch -> A-LEG SIP Status: 100 Trying 0.062694 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.062976 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.238255 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.239830 FreeSwitch -> A-LEG SIP Status: 180 Ringing 0.456828 B-Leg -> FreeSwitch SIP Status: 180 Ringing 4.642993 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.637738 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.638091 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 6.037804 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 6.839818 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 8.438750 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 11.638797 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 24.925970 FreeSwitch -> A-LEG SIP Status: 480 Temporarily Unavailable 24.926257 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@A-LEG:5060 24.926321 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 24.926580 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 24.927029 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist 27.078016 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 27.078467 FreeSwitch -> A-LEG SIP Status: 100 Trying 27.123233 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 27.123445 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 27.315802 B-Leg -> FreeSwitch SIP Status: 180 Ringing 27.317391 FreeSwitch -> A-LEG SIP Status: 180 Ringing 27.529142 B-Leg -> FreeSwitch SIP Status: 180 Ringing 31.520118 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 31.769831 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 31.920832 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 32.319816 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 33.120808 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 34.720813 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 37.920852 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 49.362647 A-LEG -> FreeSwitch SIP Request: CANCEL sip:+13605551212@A-LEG:5060 49.362952 FreeSwitch -> A-LEG SIP Status: 200 OK 49.363196 FreeSwitch -> A-LEG SIP Status: 487 Request Terminated 49.363307 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@A-LEG:5060 49.365975 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 49.366171 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 49.366522 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist
On Sep 5, 2012, at 11:10 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
update and try again, if it still doesn't work open a jira
On Wed, Sep 5, 2012 at 12:18 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
as promised here is the update testing and enabling {sip_wait_for_aleg_ack=true} causes a log delay before the audio starts to flow to the B-Leg endpoint with multi 200 OK's being sent to FS from the B-Leg as well as an abnormally long delay in getting an ACK from the A-Leg
I can open a JIRA case on this and provide the console log file / PCAP's ect if that would help
Call Flow with out {sip_wait_for_aleg_ack=true} 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 0.000652 FreeSwitch -> A-LEG SIP Status: 100 Trying 0.042837 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.043059 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.290947 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.292890 FreeSwitch -> A-LEG SIP Status: 180 Ringing 0.490220 B-Leg -> FreeSwitch SIP Status: 180 Ringing 4.481038 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 4.482310 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 4.483474 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session description 4.533691 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@FreeSwitch:5070;transport=udp 19.953061 A-LEG -> FreeSwitch SIP Request: BYE sip:+13605551212@FreeSwitch:5070;transport=udp 19.954592 FreeSwitch -> A-LEG SIP Status: 200 OK 19.955454 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 19.956014 B-Leg -> FreeSwitch SIP Status: 200 OK
Call Flow with {sip_wait_for_aleg_ack=true} enabled 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 0.000651 FreeSwitch -> A-LEG SIP Status: 100 Trying 0.039455 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.039709 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.244269 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.245607 FreeSwitch -> A-LEG SIP Status: 180 Ringing 0.363325 B-Leg -> FreeSwitch SIP Status: 180 Ringing 4.718173 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 4.918915 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.117917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.518902 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 6.318053 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 7.917921 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 11.118917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 15.102917 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 15.104488 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session description 15.173265 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@FreeSwitch:5070;transport=udp 29.566504 A-LEG -> FreeSwitch SIP Request: BYE sip:+13605551212@FreeSwitch:5070;transport=udp 29.568902 FreeSwitch -> A-LEG SIP Status: 200 OK 29.570603 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 29.571240 B-Leg -> FreeSwitch SIP Status: 200 OK
On Aug 31, 2012, at 1:52 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
No worries I will be out this weekend for the long weekend I will work on getting the test box upgraded and a test case setup on Tuesday I will report back the results mid to late next week and provided everything works as I hope it will I will be happy to pay the Wiki tax :)
On Aug 31, 2012, at 1:42 PM, Michael Collins <msc@freeswitch.org> wrote:
On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame <mike.burlingame@me.com> wrote: > > Cool I will nail that up on my test box and see if that works > Please report back on whether it works or not and then be prepared to pay the wiki tax. :) I'll be glad to assist with getting this documented although I think you're in the best position to give that documentation some real-world context.
-MC _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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FYI: You need to get the backtrace before you update or the core file becomes useless. It dumps core by default so you should have one but if you updated, its tool late unless you can checkout the exact previous rev. On Fri, Sep 7, 2012 at 12:00 PM, Mike Burlingame <mike.burlingame@me.com>wrote:
after putting the server in production and loading it up it just randomly crashed I am updating to the latest git will try to capture a core dump if one is created on exit nothing in the logs point to anything it was handling calls and just exited.
However on a positive note the switch sip_wait_for_aleg_ack=true seems to do as expected with all my tests.
On Sep 6, 2012, at 12:50 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
After about 20K test calls this seems to have addressed the issue - I will keep running my test's for today and put this box in a production environment tomorrow to validate it still holds up with load. I will report back after that is completed.
Thanks
On Sep 5, 2012, at 12:28 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
Looks much much better Thank you -- Now to conduct more testing
0.000000 A-Leg -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-Leg:5060, with session description 0.000639 FreeSwitch -> A-Leg SIP Status: 100 Trying 0.051096 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.051351 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.286387 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.288216 FreeSwitch -> A-Leg SIP Status: 180 Ringing 0.475452 B-Leg -> FreeSwitch SIP Status: 180 Ringing 5.311144 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.313775 FreeSwitch -> A-Leg SIP/SDP Status: 200 OK, with session description 5.385087 A-Leg -> FreeSwitch SIP Request: ACK sip:+13605551212@FreeSwitch:5070;transport=udp 5.385796 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 12.027026 A-Leg -> FreeSwitch SIP Request: BYE sip:+13605551212 @FreeSwitch:5070;transport=udp 12.029232 FreeSwitch -> A-Leg SIP Status: 200 OK 12.030707 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 12.033888 B-Leg -> FreeSwitch SIP Status: 200 OK
On Sep 5, 2012, at 12:01 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
ok,
update one more time, if it still does not work just go right to jira with the latest (not before today's changes)
On Wed, Sep 5, 2012 at 1:37 PM, Mike Burlingame < mike.burlingame@me.com> wrote:
The change seems to have broken the ability for the call to connect - would you like me to open a jira up with the current log files or before the change was made today?
commit d45db898339e1b2212f5befff1af714abcec034f Author: Anthony Minessale <anthm@freeswitch.org> Date: Wed Sep 5 13:11:32 2012 -0500
0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 0.002715 FreeSwitch -> A-LEG SIP Status: 100 Trying 0.062694 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.062976 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.238255 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.239830 FreeSwitch -> A-LEG SIP Status: 180 Ringing 0.456828 B-Leg -> FreeSwitch SIP Status: 180 Ringing 4.642993 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.637738 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.638091 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 6.037804 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 6.839818 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 8.438750 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 11.638797 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 24.925970 FreeSwitch -> A-LEG SIP Status: 480 Temporarily Unavailable 24.926257 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@A-LEG :5060 24.926321 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 24.926580 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 24.927029 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist 27.078016 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip: +13605551212@A-LEG:5060, with session description 27.078467 FreeSwitch -> A-LEG SIP Status: 100 Trying 27.123233 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 27.123445 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 27.315802 B-Leg -> FreeSwitch SIP Status: 180 Ringing 27.317391 FreeSwitch -> A-LEG SIP Status: 180 Ringing 27.529142 B-Leg -> FreeSwitch SIP Status: 180 Ringing 31.520118 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 31.769831 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 31.920832 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 32.319816 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 33.120808 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 34.720813 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 37.920852 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 49.362647 A-LEG -> FreeSwitch SIP Request: CANCEL sip:+13605551212@A-LEG:5060 49.362952 FreeSwitch -> A-LEG SIP Status: 200 OK 49.363196 FreeSwitch -> A-LEG SIP Status: 487 Request Terminated 49.363307 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212 @A-LEG:5060 49.365975 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 49.366171 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 49.366522 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist
On Sep 5, 2012, at 11:10 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
update and try again, if it still doesn't work open a jira
On Wed, Sep 5, 2012 at 12:18 PM, Mike Burlingame < mike.burlingame@me.com> wrote: > as promised here is the update testing and enabling > {sip_wait_for_aleg_ack=true} causes a log delay before the audio starts to > flow to the B-Leg endpoint with multi 200 OK's being sent to FS from the > B-Leg as well as an abnormally long delay in getting an ACK from the A-Leg > > I can open a JIRA case on this and provide the console log file / PCAP's ect > if that would help > > > Call Flow with out {sip_wait_for_aleg_ack=true} > 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE > sip:+13605551212@A-LEG:5060, with session description > 0.000652 FreeSwitch -> A-LEG SIP Status: 100 Trying > 0.042837 FreeSwitch -> B-Leg SIP/SDP Request: INVITE > sip:13605551212@B-Leg, with session description > 0.043059 B-Leg -> FreeSwitch SIP Status: 100 Giving a try > 0.290947 B-Leg -> FreeSwitch SIP Status: 180 Ringing > 0.292890 FreeSwitch -> A-LEG SIP Status: 180 Ringing > 0.490220 B-Leg -> FreeSwitch SIP Status: 180 Ringing > 4.481038 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session > description > 4.482310 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg > 4.483474 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session > description > 4.533691 A-LEG -> FreeSwitch SIP Request: ACK > sip:+13605551212@FreeSwitch:5070;transport=udp > 19.953061 A-LEG -> FreeSwitch SIP Request: BYE > sip:+13605551212@FreeSwitch:5070;transport=udp > 19.954592 FreeSwitch -> A-LEG SIP Status: 200 OK > 19.955454 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg > 19.956014 B-Leg -> FreeSwitch SIP Status: 200 OK > > Call Flow with {sip_wait_for_aleg_ack=true} enabled > 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE > sip:+13605551212@A-LEG:5060, with session description > 0.000651 FreeSwitch -> A-LEG SIP Status: 100 Trying > 0.039455 FreeSwitch -> B-Leg SIP/SDP Request: INVITE > sip:13605551212@B-Leg, with session description > 0.039709 B-Leg -> FreeSwitch SIP Status: 100 Giving a try > 0.244269 B-Leg -> FreeSwitch SIP Status: 180 Ringing > 0.245607 FreeSwitch -> A-LEG SIP Status: 180 Ringing > 0.363325 B-Leg -> FreeSwitch SIP Status: 180 Ringing > 4.718173 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session > description > 4.918915 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session > description > 5.117917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session > description > 5.518902 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session > description > 6.318053 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session > description > 7.917921 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session > description > 11.118917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session > description > 15.102917 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg > 15.104488 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session > description > 15.173265 A-LEG -> FreeSwitch SIP Request: ACK > sip:+13605551212@FreeSwitch:5070;transport=udp > 29.566504 A-LEG -> FreeSwitch SIP Request: BYE > sip:+13605551212@FreeSwitch:5070;transport=udp > 29.568902 FreeSwitch -> A-LEG SIP Status: 200 OK > 29.570603 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg > 29.571240 B-Leg -> FreeSwitch SIP Status: 200 OK > > On Aug 31, 2012, at 1:52 PM, Mike Burlingame < mike.burlingame@me.com> wrote: > > No worries I will be out this weekend for the long weekend I will work on > getting the test box upgraded and a test case setup on Tuesday I will report > back the results mid to late next week and provided everything works as I > hope it will I will be happy to pay the Wiki tax :) > > > On Aug 31, 2012, at 1:42 PM, Michael Collins <msc@freeswitch.org> wrote: > > > > On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame < mike.burlingame@me.com> > wrote: >> >> Cool I will nail that up on my test box and see if that works >> > Please report back on whether it works or not and then be prepared to pay > the wiki tax. :) I'll be glad to assist with getting this documented > although I think you're in the best position to give that documentation some > real-world context. > > -MC >
> Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > >
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> Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >
-- Anthony Minessale II
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I figured I would update to the latest git due to in the past I have seen "make sure your running the latest git" the odd thing is no core dump was created on the last crash. I will post the backtrace if a core dump is generated after the next crash On Sep 7, 2012, at 10:16 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
FYI: You need to get the backtrace before you update or the core file becomes useless. It dumps core by default so you should have one but if you updated, its tool late unless you can checkout the exact previous rev.
On Fri, Sep 7, 2012 at 12:00 PM, Mike Burlingame <mike.burlingame@me.com> wrote: after putting the server in production and loading it up it just randomly crashed I am updating to the latest git will try to capture a core dump if one is created on exit nothing in the logs point to anything it was handling calls and just exited.
However on a positive note the switch sip_wait_for_aleg_ack=true seems to do as expected with all my tests.
On Sep 6, 2012, at 12:50 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
After about 20K test calls this seems to have addressed the issue - I will keep running my test's for today and put this box in a production environment tomorrow to validate it still holds up with load. I will report back after that is completed.
Thanks
On Sep 5, 2012, at 12:28 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
Looks much much better Thank you -- Now to conduct more testing
0.000000 A-Leg -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-Leg:5060, with session description 0.000639 FreeSwitch -> A-Leg SIP Status: 100 Trying 0.051096 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.051351 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.286387 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.288216 FreeSwitch -> A-Leg SIP Status: 180 Ringing 0.475452 B-Leg -> FreeSwitch SIP Status: 180 Ringing 5.311144 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.313775 FreeSwitch -> A-Leg SIP/SDP Status: 200 OK, with session description 5.385087 A-Leg -> FreeSwitch SIP Request: ACK sip:+13605551212@FreeSwitch:5070;transport=udp 5.385796 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 12.027026 A-Leg -> FreeSwitch SIP Request: BYE sip:+13605551212@FreeSwitch:5070;transport=udp 12.029232 FreeSwitch -> A-Leg SIP Status: 200 OK 12.030707 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 12.033888 B-Leg -> FreeSwitch SIP Status: 200 OK
On Sep 5, 2012, at 12:01 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
ok,
update one more time, if it still does not work just go right to jira with the latest (not before today's changes)
On Wed, Sep 5, 2012 at 1:37 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
The change seems to have broken the ability for the call to connect - would you like me to open a jira up with the current log files or before the change was made today?
commit d45db898339e1b2212f5befff1af714abcec034f Author: Anthony Minessale <anthm@freeswitch.org> Date: Wed Sep 5 13:11:32 2012 -0500
0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 0.002715 FreeSwitch -> A-LEG SIP Status: 100 Trying 0.062694 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.062976 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.238255 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.239830 FreeSwitch -> A-LEG SIP Status: 180 Ringing 0.456828 B-Leg -> FreeSwitch SIP Status: 180 Ringing 4.642993 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.637738 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.638091 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 6.037804 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 6.839818 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 8.438750 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 11.638797 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 24.925970 FreeSwitch -> A-LEG SIP Status: 480 Temporarily Unavailable 24.926257 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@A-LEG:5060 24.926321 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 24.926580 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 24.927029 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist 27.078016 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 27.078467 FreeSwitch -> A-LEG SIP Status: 100 Trying 27.123233 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 27.123445 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 27.315802 B-Leg -> FreeSwitch SIP Status: 180 Ringing 27.317391 FreeSwitch -> A-LEG SIP Status: 180 Ringing 27.529142 B-Leg -> FreeSwitch SIP Status: 180 Ringing 31.520118 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 31.769831 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 31.920832 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 32.319816 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 33.120808 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 34.720813 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 37.920852 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 49.362647 A-LEG -> FreeSwitch SIP Request: CANCEL sip:+13605551212@A-LEG:5060 49.362952 FreeSwitch -> A-LEG SIP Status: 200 OK 49.363196 FreeSwitch -> A-LEG SIP Status: 487 Request Terminated 49.363307 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@A-LEG:5060 49.365975 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 49.366171 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 49.366522 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist
On Sep 5, 2012, at 11:10 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
update and try again, if it still doesn't work open a jira
On Wed, Sep 5, 2012 at 12:18 PM, Mike Burlingame <mike.burlingame@me.com> wrote: > as promised here is the update testing and enabling > {sip_wait_for_aleg_ack=true} causes a log delay before the audio starts to > flow to the B-Leg endpoint with multi 200 OK's being sent to FS from the > B-Leg as well as an abnormally long delay in getting an ACK from the A-Leg > > I can open a JIRA case on this and provide the console log file / PCAP's ect > if that would help > > > Call Flow with out {sip_wait_for_aleg_ack=true} > 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE > sip:+13605551212@A-LEG:5060, with session description > 0.000652 FreeSwitch -> A-LEG SIP Status: 100 Trying > 0.042837 FreeSwitch -> B-Leg SIP/SDP Request: INVITE > sip:13605551212@B-Leg, with session description > 0.043059 B-Leg -> FreeSwitch SIP Status: 100 Giving a try > 0.290947 B-Leg -> FreeSwitch SIP Status: 180 Ringing > 0.292890 FreeSwitch -> A-LEG SIP Status: 180 Ringing > 0.490220 B-Leg -> FreeSwitch SIP Status: 180 Ringing > 4.481038 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session > description > 4.482310 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg > 4.483474 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session > description > 4.533691 A-LEG -> FreeSwitch SIP Request: ACK > sip:+13605551212@FreeSwitch:5070;transport=udp > 19.953061 A-LEG -> FreeSwitch SIP Request: BYE > sip:+13605551212@FreeSwitch:5070;transport=udp > 19.954592 FreeSwitch -> A-LEG SIP Status: 200 OK > 19.955454 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg > 19.956014 B-Leg -> FreeSwitch SIP Status: 200 OK > > Call Flow with {sip_wait_for_aleg_ack=true} enabled > 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE > sip:+13605551212@A-LEG:5060, with session description > 0.000651 FreeSwitch -> A-LEG SIP Status: 100 Trying > 0.039455 FreeSwitch -> B-Leg SIP/SDP Request: INVITE > sip:13605551212@B-Leg, with session description > 0.039709 B-Leg -> FreeSwitch SIP Status: 100 Giving a try > 0.244269 B-Leg -> FreeSwitch SIP Status: 180 Ringing > 0.245607 FreeSwitch -> A-LEG SIP Status: 180 Ringing > 0.363325 B-Leg -> FreeSwitch SIP Status: 180 Ringing > 4.718173 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session > description > 4.918915 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session > description > 5.117917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session > description > 5.518902 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session > description > 6.318053 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session > description > 7.917921 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session > description > 11.118917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session > description > 15.102917 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg > 15.104488 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session > description > 15.173265 A-LEG -> FreeSwitch SIP Request: ACK > sip:+13605551212@FreeSwitch:5070;transport=udp > 29.566504 A-LEG -> FreeSwitch SIP Request: BYE > sip:+13605551212@FreeSwitch:5070;transport=udp > 29.568902 FreeSwitch -> A-LEG SIP Status: 200 OK > 29.570603 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg > 29.571240 B-Leg -> FreeSwitch SIP Status: 200 OK > > On Aug 31, 2012, at 1:52 PM, Mike Burlingame <mike.burlingame@me.com> wrote: > > No worries I will be out this weekend for the long weekend I will work on > getting the test box upgraded and a test case setup on Tuesday I will report > back the results mid to late next week and provided everything works as I > hope it will I will be happy to pay the Wiki tax :) > > > On Aug 31, 2012, at 1:42 PM, Michael Collins <msc@freeswitch.org> wrote: > > > > On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame <mike.burlingame@me.com> > wrote: >> >> Cool I will nail that up on my test box and see if that works >> > Please report back on whether it works or not and then be prepared to pay > the wiki tax. :) I'll be glad to assist with getting this documented > although I think you're in the best position to give that documentation some > real-world context. > > -MC > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
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Ok got another crash oddly enough it's kinda the same as yesterday's Yesterday the last message I see in the log file is a 401 Unauthorized before the log file stopped recording data and fs_cli exited to prompt today same thing happened but this time it looks like it did a core dump. Opening JIRA case now On Sep 7, 2012, at 10:23 AM, Mike Burlingame <mike.burlingame@me.com> wrote:
I figured I would update to the latest git due to in the past I have seen "make sure your running the latest git" the odd thing is no core dump was created on the last crash.
I will post the backtrace if a core dump is generated after the next crash
On Sep 7, 2012, at 10:16 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
FYI: You need to get the backtrace before you update or the core file becomes useless. It dumps core by default so you should have one but if you updated, its tool late unless you can checkout the exact previous rev.
On Fri, Sep 7, 2012 at 12:00 PM, Mike Burlingame <mike.burlingame@me.com> wrote: after putting the server in production and loading it up it just randomly crashed I am updating to the latest git will try to capture a core dump if one is created on exit nothing in the logs point to anything it was handling calls and just exited.
However on a positive note the switch sip_wait_for_aleg_ack=true seems to do as expected with all my tests.
On Sep 6, 2012, at 12:50 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
After about 20K test calls this seems to have addressed the issue - I will keep running my test's for today and put this box in a production environment tomorrow to validate it still holds up with load. I will report back after that is completed.
Thanks
On Sep 5, 2012, at 12:28 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
Looks much much better Thank you -- Now to conduct more testing
0.000000 A-Leg -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-Leg:5060, with session description 0.000639 FreeSwitch -> A-Leg SIP Status: 100 Trying 0.051096 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.051351 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.286387 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.288216 FreeSwitch -> A-Leg SIP Status: 180 Ringing 0.475452 B-Leg -> FreeSwitch SIP Status: 180 Ringing 5.311144 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.313775 FreeSwitch -> A-Leg SIP/SDP Status: 200 OK, with session description 5.385087 A-Leg -> FreeSwitch SIP Request: ACK sip:+13605551212@FreeSwitch:5070;transport=udp 5.385796 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 12.027026 A-Leg -> FreeSwitch SIP Request: BYE sip:+13605551212@FreeSwitch:5070;transport=udp 12.029232 FreeSwitch -> A-Leg SIP Status: 200 OK 12.030707 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 12.033888 B-Leg -> FreeSwitch SIP Status: 200 OK
On Sep 5, 2012, at 12:01 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
ok,
update one more time, if it still does not work just go right to jira with the latest (not before today's changes)
On Wed, Sep 5, 2012 at 1:37 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
The change seems to have broken the ability for the call to connect - would you like me to open a jira up with the current log files or before the change was made today?
commit d45db898339e1b2212f5befff1af714abcec034f Author: Anthony Minessale <anthm@freeswitch.org> Date: Wed Sep 5 13:11:32 2012 -0500
0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 0.002715 FreeSwitch -> A-LEG SIP Status: 100 Trying 0.062694 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.062976 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.238255 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.239830 FreeSwitch -> A-LEG SIP Status: 180 Ringing 0.456828 B-Leg -> FreeSwitch SIP Status: 180 Ringing 4.642993 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.637738 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.638091 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 6.037804 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 6.839818 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 8.438750 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 11.638797 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 24.925970 FreeSwitch -> A-LEG SIP Status: 480 Temporarily Unavailable 24.926257 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@A-LEG:5060 24.926321 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 24.926580 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 24.927029 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist 27.078016 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description 27.078467 FreeSwitch -> A-LEG SIP Status: 100 Trying 27.123233 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 27.123445 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 27.315802 B-Leg -> FreeSwitch SIP Status: 180 Ringing 27.317391 FreeSwitch -> A-LEG SIP Status: 180 Ringing 27.529142 B-Leg -> FreeSwitch SIP Status: 180 Ringing 31.520118 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 31.769831 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 31.920832 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 32.319816 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 33.120808 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 34.720813 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 37.920852 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 49.362647 A-LEG -> FreeSwitch SIP Request: CANCEL sip:+13605551212@A-LEG:5060 49.362952 FreeSwitch -> A-LEG SIP Status: 200 OK 49.363196 FreeSwitch -> A-LEG SIP Status: 487 Request Terminated 49.363307 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@A-LEG:5060 49.365975 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 49.366171 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 49.366522 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist
On Sep 5, 2012, at 11:10 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
> update and try again, if it still doesn't work open a jira > > > On Wed, Sep 5, 2012 at 12:18 PM, Mike Burlingame <mike.burlingame@me.com> wrote: >> as promised here is the update testing and enabling >> {sip_wait_for_aleg_ack=true} causes a log delay before the audio starts to >> flow to the B-Leg endpoint with multi 200 OK's being sent to FS from the >> B-Leg as well as an abnormally long delay in getting an ACK from the A-Leg >> >> I can open a JIRA case on this and provide the console log file / PCAP's ect >> if that would help >> >> >> Call Flow with out {sip_wait_for_aleg_ack=true} >> 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE >> sip:+13605551212@A-LEG:5060, with session description >> 0.000652 FreeSwitch -> A-LEG SIP Status: 100 Trying >> 0.042837 FreeSwitch -> B-Leg SIP/SDP Request: INVITE >> sip:13605551212@B-Leg, with session description >> 0.043059 B-Leg -> FreeSwitch SIP Status: 100 Giving a try >> 0.290947 B-Leg -> FreeSwitch SIP Status: 180 Ringing >> 0.292890 FreeSwitch -> A-LEG SIP Status: 180 Ringing >> 0.490220 B-Leg -> FreeSwitch SIP Status: 180 Ringing >> 4.481038 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session >> description >> 4.482310 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg >> 4.483474 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session >> description >> 4.533691 A-LEG -> FreeSwitch SIP Request: ACK >> sip:+13605551212@FreeSwitch:5070;transport=udp >> 19.953061 A-LEG -> FreeSwitch SIP Request: BYE >> sip:+13605551212@FreeSwitch:5070;transport=udp >> 19.954592 FreeSwitch -> A-LEG SIP Status: 200 OK >> 19.955454 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg >> 19.956014 B-Leg -> FreeSwitch SIP Status: 200 OK >> >> Call Flow with {sip_wait_for_aleg_ack=true} enabled >> 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE >> sip:+13605551212@A-LEG:5060, with session description >> 0.000651 FreeSwitch -> A-LEG SIP Status: 100 Trying >> 0.039455 FreeSwitch -> B-Leg SIP/SDP Request: INVITE >> sip:13605551212@B-Leg, with session description >> 0.039709 B-Leg -> FreeSwitch SIP Status: 100 Giving a try >> 0.244269 B-Leg -> FreeSwitch SIP Status: 180 Ringing >> 0.245607 FreeSwitch -> A-LEG SIP Status: 180 Ringing >> 0.363325 B-Leg -> FreeSwitch SIP Status: 180 Ringing >> 4.718173 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session >> description >> 4.918915 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session >> description >> 5.117917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session >> description >> 5.518902 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session >> description >> 6.318053 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session >> description >> 7.917921 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session >> description >> 11.118917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session >> description >> 15.102917 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg >> 15.104488 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session >> description >> 15.173265 A-LEG -> FreeSwitch SIP Request: ACK >> sip:+13605551212@FreeSwitch:5070;transport=udp >> 29.566504 A-LEG -> FreeSwitch SIP Request: BYE >> sip:+13605551212@FreeSwitch:5070;transport=udp >> 29.568902 FreeSwitch -> A-LEG SIP Status: 200 OK >> 29.570603 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg >> 29.571240 B-Leg -> FreeSwitch SIP Status: 200 OK >> >> On Aug 31, 2012, at 1:52 PM, Mike Burlingame <mike.burlingame@me.com> wrote: >> >> No worries I will be out this weekend for the long weekend I will work on >> getting the test box upgraded and a test case setup on Tuesday I will report >> back the results mid to late next week and provided everything works as I >> hope it will I will be happy to pay the Wiki tax :) >> >> >> On Aug 31, 2012, at 1:42 PM, Michael Collins <msc@freeswitch.org> wrote: >> >> >> >> On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame <mike.burlingame@me.com> >> wrote: >>> >>> Cool I will nail that up on my test box and see if that works >>> >> Please report back on whether it works or not and then be prepared to pay >> the wiki tax. :) I'll be glad to assist with getting this documented >> although I think you're in the best position to give that documentation some >> real-world context. >> >> -MC >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > googletalk:conf+888@conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Crash was resolved this weekend by a code update via jira case - no more crashes after the update - some more dial plan edits and a little more testing I will be ready to head over to pay the Wiki tax Thanks guys for adding this feature it seems to be working as expected and my 491 issues on the B-Leg / A-Leg have pretty much disappeared On Sep 8, 2012, at 9:36 AM, Mike Burlingame <mike.burlingame@me.com> wrote:
Ok got another crash oddly enough it's kinda the same as yesterday's
Yesterday the last message I see in the log file is a 401 Unauthorized before the log file stopped recording data and fs_cli exited to prompt today same thing happened but this time it looks like it did a core dump.
Opening JIRA case now
On Sep 7, 2012, at 10:23 AM, Mike Burlingame <mike.burlingame@me.com> wrote:
I figured I would update to the latest git due to in the past I have seen "make sure your running the latest git" the odd thing is no core dump was created on the last crash.
I will post the backtrace if a core dump is generated after the next crash
On Sep 7, 2012, at 10:16 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
FYI: You need to get the backtrace before you update or the core file becomes useless. It dumps core by default so you should have one but if you updated, its tool late unless you can checkout the exact previous rev.
On Fri, Sep 7, 2012 at 12:00 PM, Mike Burlingame <mike.burlingame@me.com> wrote: after putting the server in production and loading it up it just randomly crashed I am updating to the latest git will try to capture a core dump if one is created on exit nothing in the logs point to anything it was handling calls and just exited.
However on a positive note the switch sip_wait_for_aleg_ack=true seems to do as expected with all my tests.
On Sep 6, 2012, at 12:50 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
After about 20K test calls this seems to have addressed the issue - I will keep running my test's for today and put this box in a production environment tomorrow to validate it still holds up with load. I will report back after that is completed.
Thanks
On Sep 5, 2012, at 12:28 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
Looks much much better Thank you -- Now to conduct more testing
0.000000 A-Leg -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-Leg:5060, with session description 0.000639 FreeSwitch -> A-Leg SIP Status: 100 Trying 0.051096 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description 0.051351 B-Leg -> FreeSwitch SIP Status: 100 Giving a try 0.286387 B-Leg -> FreeSwitch SIP Status: 180 Ringing 0.288216 FreeSwitch -> A-Leg SIP Status: 180 Ringing 0.475452 B-Leg -> FreeSwitch SIP Status: 180 Ringing 5.311144 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description 5.313775 FreeSwitch -> A-Leg SIP/SDP Status: 200 OK, with session description 5.385087 A-Leg -> FreeSwitch SIP Request: ACK sip:+13605551212@FreeSwitch:5070;transport=udp 5.385796 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg 12.027026 A-Leg -> FreeSwitch SIP Request: BYE sip:+13605551212@FreeSwitch:5070;transport=udp 12.029232 FreeSwitch -> A-Leg SIP Status: 200 OK 12.030707 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg 12.033888 B-Leg -> FreeSwitch SIP Status: 200 OK
On Sep 5, 2012, at 12:01 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
ok,
update one more time, if it still does not work just go right to jira with the latest (not before today's changes)
On Wed, Sep 5, 2012 at 1:37 PM, Mike Burlingame <mike.burlingame@me.com> wrote: > The change seems to have broken the ability for the call to connect - would you like me to open a jira up with the current log files or before the change was made today? > > commit d45db898339e1b2212f5befff1af714abcec034f > Author: Anthony Minessale <anthm@freeswitch.org> > Date: Wed Sep 5 13:11:32 2012 -0500 > > 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description > 0.002715 FreeSwitch -> A-LEG SIP Status: 100 Trying > 0.062694 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description > 0.062976 B-Leg -> FreeSwitch SIP Status: 100 Giving a try > 0.238255 B-Leg -> FreeSwitch SIP Status: 180 Ringing > 0.239830 FreeSwitch -> A-LEG SIP Status: 180 Ringing > 0.456828 B-Leg -> FreeSwitch SIP Status: 180 Ringing > 4.642993 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description > 5.637738 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description > 5.638091 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description > 6.037804 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description > 6.839818 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description > 8.438750 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description > 11.638797 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description > 24.925970 FreeSwitch -> A-LEG SIP Status: 480 Temporarily Unavailable > 24.926257 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@A-LEG:5060 > 24.926321 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg > 24.926580 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg > 24.927029 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist > 27.078016 A-LEG -> FreeSwitch SIP/SDP Request: INVITE sip:+13605551212@A-LEG:5060, with session description > 27.078467 FreeSwitch -> A-LEG SIP Status: 100 Trying > 27.123233 FreeSwitch -> B-Leg SIP/SDP Request: INVITE sip:13605551212@B-Leg, with session description > 27.123445 B-Leg -> FreeSwitch SIP Status: 100 Giving a try > 27.315802 B-Leg -> FreeSwitch SIP Status: 180 Ringing > 27.317391 FreeSwitch -> A-LEG SIP Status: 180 Ringing > 27.529142 B-Leg -> FreeSwitch SIP Status: 180 Ringing > 31.520118 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description > 31.769831 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description > 31.920832 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description > 32.319816 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description > 33.120808 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description > 34.720813 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description > 37.920852 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session description > 49.362647 A-LEG -> FreeSwitch SIP Request: CANCEL sip:+13605551212@A-LEG:5060 > 49.362952 FreeSwitch -> A-LEG SIP Status: 200 OK > 49.363196 FreeSwitch -> A-LEG SIP Status: 487 Request Terminated > 49.363307 A-LEG -> FreeSwitch SIP Request: ACK sip:+13605551212@A-LEG:5060 > 49.365975 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg > 49.366171 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg > 49.366522 B-Leg -> FreeSwitch SIP Status: 481 Call leg/transaction does not exist > > > On Sep 5, 2012, at 11:10 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote: > >> update and try again, if it still doesn't work open a jira >> >> >> On Wed, Sep 5, 2012 at 12:18 PM, Mike Burlingame <mike.burlingame@me.com> wrote: >>> as promised here is the update testing and enabling >>> {sip_wait_for_aleg_ack=true} causes a log delay before the audio starts to >>> flow to the B-Leg endpoint with multi 200 OK's being sent to FS from the >>> B-Leg as well as an abnormally long delay in getting an ACK from the A-Leg >>> >>> I can open a JIRA case on this and provide the console log file / PCAP's ect >>> if that would help >>> >>> >>> Call Flow with out {sip_wait_for_aleg_ack=true} >>> 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE >>> sip:+13605551212@A-LEG:5060, with session description >>> 0.000652 FreeSwitch -> A-LEG SIP Status: 100 Trying >>> 0.042837 FreeSwitch -> B-Leg SIP/SDP Request: INVITE >>> sip:13605551212@B-Leg, with session description >>> 0.043059 B-Leg -> FreeSwitch SIP Status: 100 Giving a try >>> 0.290947 B-Leg -> FreeSwitch SIP Status: 180 Ringing >>> 0.292890 FreeSwitch -> A-LEG SIP Status: 180 Ringing >>> 0.490220 B-Leg -> FreeSwitch SIP Status: 180 Ringing >>> 4.481038 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session >>> description >>> 4.482310 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg >>> 4.483474 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session >>> description >>> 4.533691 A-LEG -> FreeSwitch SIP Request: ACK >>> sip:+13605551212@FreeSwitch:5070;transport=udp >>> 19.953061 A-LEG -> FreeSwitch SIP Request: BYE >>> sip:+13605551212@FreeSwitch:5070;transport=udp >>> 19.954592 FreeSwitch -> A-LEG SIP Status: 200 OK >>> 19.955454 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg >>> 19.956014 B-Leg -> FreeSwitch SIP Status: 200 OK >>> >>> Call Flow with {sip_wait_for_aleg_ack=true} enabled >>> 0.000000 A-LEG -> FreeSwitch SIP/SDP Request: INVITE >>> sip:+13605551212@A-LEG:5060, with session description >>> 0.000651 FreeSwitch -> A-LEG SIP Status: 100 Trying >>> 0.039455 FreeSwitch -> B-Leg SIP/SDP Request: INVITE >>> sip:13605551212@B-Leg, with session description >>> 0.039709 B-Leg -> FreeSwitch SIP Status: 100 Giving a try >>> 0.244269 B-Leg -> FreeSwitch SIP Status: 180 Ringing >>> 0.245607 FreeSwitch -> A-LEG SIP Status: 180 Ringing >>> 0.363325 B-Leg -> FreeSwitch SIP Status: 180 Ringing >>> 4.718173 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session >>> description >>> 4.918915 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session >>> description >>> 5.117917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session >>> description >>> 5.518902 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session >>> description >>> 6.318053 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session >>> description >>> 7.917921 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session >>> description >>> 11.118917 B-Leg -> FreeSwitch SIP/SDP Status: 200 OK, with session >>> description >>> 15.102917 FreeSwitch -> B-Leg SIP Request: ACK sip:13605551212@B-Leg >>> 15.104488 FreeSwitch -> A-LEG SIP/SDP Status: 200 OK, with session >>> description >>> 15.173265 A-LEG -> FreeSwitch SIP Request: ACK >>> sip:+13605551212@FreeSwitch:5070;transport=udp >>> 29.566504 A-LEG -> FreeSwitch SIP Request: BYE >>> sip:+13605551212@FreeSwitch:5070;transport=udp >>> 29.568902 FreeSwitch -> A-LEG SIP Status: 200 OK >>> 29.570603 FreeSwitch -> B-Leg SIP Request: BYE sip:13605551212@B-Leg >>> 29.571240 B-Leg -> FreeSwitch SIP Status: 200 OK >>> >>> On Aug 31, 2012, at 1:52 PM, Mike Burlingame <mike.burlingame@me.com> wrote: >>> >>> No worries I will be out this weekend for the long weekend I will work on >>> getting the test box upgraded and a test case setup on Tuesday I will report >>> back the results mid to late next week and provided everything works as I >>> hope it will I will be happy to pay the Wiki tax :) >>> >>> >>> On Aug 31, 2012, at 1:42 PM, Michael Collins <msc@freeswitch.org> wrote: >>> >>> >>> >>> On Fri, Aug 31, 2012 at 12:59 PM, Mike Burlingame <mike.burlingame@me.com> >>> wrote: >>>> >>>> Cool I will nail that up on my test box and see if that works >>>> >>> Please report back on whether it works or not and then be prepared to pay >>> the wiki tax. :) I'll be glad to assist with getting this documented >>> although I think you're in the best position to give that documentation some >>> real-world context. >>> >>> -MC
On Tue, Sep 11, 2012 at 11:39 AM, Mike Burlingame <mike.burlingame@me.com>wrote:
Crash was resolved this weekend by a code update via jira case - no more crashes after the update - some more dial plan edits and a little more testing I will be ready to head over to pay the Wiki tax
Thanks guys for adding this feature it seems to be working as expected and my 491 issues on the B-Leg / A-Leg have pretty much disappeared
Thanks for checking in and letting us know where you stand. I really like it when these threads have some sort of resolution. If you have any issues with wiki editing please let me know or hop on IRC for an assist. -MC
So it seems all is not well - missing ACK and forward of 200OK when 2 FS servers are in the mix when sip_wait_for_aleg_ack=true is enabled basic flow is FS Server A FS Server B |-----------------DID Channel---------------------| |---------Termination Channel (RCF)----------| Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> Carrier Both FS running latest GIT as of this morning - Both FS servers using sip_wait_for_aleg_ack=true - Everything on FS B looks ok (besides not getting an ACK back from FS_A) FS Server A Logs http://pastebin.freeswitch.org/19897 FS Server B SIP Logs http://pastebin.freeswitch.org/19896 On Sep 11, 2012, at 4:56 PM, Michael Collins <msc@freeswitch.org> wrote:
On Tue, Sep 11, 2012 at 11:39 AM, Mike Burlingame <mike.burlingame@me.com> wrote: Crash was resolved this weekend by a code update via jira case - no more crashes after the update - some more dial plan edits and a little more testing I will be ready to head over to pay the Wiki tax
Thanks guys for adding this feature it seems to be working as expected and my 491 issues on the B-Leg / A-Leg have pretty much disappeared
Thanks for checking in and letting us know where you stand. I really like it when these threads have some sort of resolution. If you have any issues with wiki editing please let me know or hop on IRC for an assist.
-MC _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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sigh, you forgot "console loglevel debug" this is only info loglevel......... On Thu, Sep 13, 2012 at 5:49 PM, Mike Burlingame <mike.burlingame@me.com>wrote:
So it seems all is not well - missing ACK and forward of 200OK when 2 FS servers are in the mix when sip_wait_for_aleg_ack=true is enabled basic flow is
FS Server A FS Server B |-----------------DID Channel---------------------| |---------Termination Channel (RCF)----------| Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> Carrier
Both FS running latest GIT as of this morning - Both FS servers using sip_wait_for_aleg_ack=true - Everything on FS B looks ok (besides not getting an ACK back from FS_A)
FS Server A Logs http://pastebin.freeswitch.org/19897
FS Server B SIP Logs http://pastebin.freeswitch.org/19896
On Sep 11, 2012, at 4:56 PM, Michael Collins <msc@freeswitch.org> wrote:
On Tue, Sep 11, 2012 at 11:39 AM, Mike Burlingame <mike.burlingame@me.com>wrote:
Crash was resolved this weekend by a code update via jira case - no more crashes after the update - some more dial plan edits and a little more testing I will be ready to head over to pay the Wiki tax
Thanks guys for adding this feature it seems to be working as expected and my 491 issues on the B-Leg / A-Leg have pretty much disappeared
Thanks for checking in and letting us know where you stand. I really like it when these threads have some sort of resolution. If you have any issues with wiki editing please let me know or hop on IRC for an assist.
-MC _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
I currently have 2 servers out of production right now I can have Opensips redirect calls to the two servers for my ANI and the Dest number and re-run the output On Sep 13, 2012, at 3:58 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
sigh, you forgot "console loglevel debug" this is only info loglevel.........
On Thu, Sep 13, 2012 at 5:49 PM, Mike Burlingame <mike.burlingame@me.com> wrote: So it seems all is not well - missing ACK and forward of 200OK when 2 FS servers are in the mix when sip_wait_for_aleg_ack=true is enabled basic flow is
FS Server A FS Server B |-----------------DID Channel---------------------| |---------Termination Channel (RCF)----------| Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> Carrier
Both FS running latest GIT as of this morning - Both FS servers using sip_wait_for_aleg_ack=true - Everything on FS B looks ok (besides not getting an ACK back from FS_A)
FS Server A Logs http://pastebin.freeswitch.org/19897
FS Server B SIP Logs http://pastebin.freeswitch.org/19896
On Sep 11, 2012, at 4:56 PM, Michael Collins <msc@freeswitch.org> wrote:
On Tue, Sep 11, 2012 at 11:39 AM, Mike Burlingame <mike.burlingame@me.com> wrote: Crash was resolved this weekend by a code update via jira case - no more crashes after the update - some more dial plan edits and a little more testing I will be ready to head over to pay the Wiki tax
Thanks guys for adding this feature it seems to be working as expected and my 491 issues on the B-Leg / A-Leg have pretty much disappeared
Thanks for checking in and letting us know where you stand. I really like it when these threads have some sort of resolution. If you have any issues with wiki editing please let me know or hop on IRC for an assist.
-MC _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
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FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
So what I am seeing is if the call flow is like Carrier --> OpenSIPs --> FS --> OpenSIPs --> Off Network things seem fine but if the flow is Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> PSTN things seem broken now if I remove wait for ACK from Server A's B-Leg the call works as expected FS_SERVER_A http://pastebin.freeswitch.org/19898 FS_SERVER_B http://pastebin.freeswitch.org/19899 On Sep 13, 2012, at 3:58 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
sigh, you forgot "console loglevel debug" this is only info loglevel.........
On Thu, Sep 13, 2012 at 5:49 PM, Mike Burlingame <mike.burlingame@me.com> wrote: So it seems all is not well - missing ACK and forward of 200OK when 2 FS servers are in the mix when sip_wait_for_aleg_ack=true is enabled basic flow is
FS Server A FS Server B |-----------------DID Channel---------------------| |---------Termination Channel (RCF)----------| Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> Carrier
Both FS running latest GIT as of this morning - Both FS servers using sip_wait_for_aleg_ack=true - Everything on FS B looks ok (besides not getting an ACK back from FS_A)
FS Server A Logs http://pastebin.freeswitch.org/19897
FS Server B SIP Logs http://pastebin.freeswitch.org/19896
On Sep 11, 2012, at 4:56 PM, Michael Collins <msc@freeswitch.org> wrote:
On Tue, Sep 11, 2012 at 11:39 AM, Mike Burlingame <mike.burlingame@me.com> wrote: Crash was resolved this weekend by a code update via jira case - no more crashes after the update - some more dial plan edits and a little more testing I will be ready to head over to pay the Wiki tax
Thanks guys for adding this feature it seems to be working as expected and my 491 issues on the B-Leg / A-Leg have pretty much disappeared
Thanks for checking in and letting us know where you stand. I really like it when these threads have some sort of resolution. If you have any issues with wiki editing please let me know or hop on IRC for an assist.
-MC _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
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FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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You did it again... I need the DEBUG level in the logs sofia global siptrace on console loglevel debug fsctl loglevel debug On Thu, Sep 13, 2012 at 6:49 PM, Mike Burlingame <mike.burlingame@me.com>wrote:
So what I am seeing is if the call flow is like
Carrier --> OpenSIPs --> FS --> OpenSIPs --> Off Network things seem fine
but if the flow is
Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> PSTN things seem broken now if I remove wait for ACK from Server A's B-Leg the call works as expected
FS_SERVER_A http://pastebin.freeswitch.org/19898
FS_SERVER_B http://pastebin.freeswitch.org/19899
On Sep 13, 2012, at 3:58 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
sigh, you forgot "console loglevel debug" this is only info loglevel.........
On Thu, Sep 13, 2012 at 5:49 PM, Mike Burlingame <mike.burlingame@me.com>wrote:
So it seems all is not well - missing ACK and forward of 200OK when 2 FS servers are in the mix when sip_wait_for_aleg_ack=true is enabled basic flow is
FS Server A FS Server B |-----------------DID Channel---------------------| |---------Termination Channel (RCF)----------| Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> Carrier
Both FS running latest GIT as of this morning - Both FS servers using sip_wait_for_aleg_ack=true - Everything on FS B looks ok (besides not getting an ACK back from FS_A)
FS Server A Logs http://pastebin.freeswitch.org/19897
FS Server B SIP Logs http://pastebin.freeswitch.org/19896
On Sep 11, 2012, at 4:56 PM, Michael Collins <msc@freeswitch.org> wrote:
On Tue, Sep 11, 2012 at 11:39 AM, Mike Burlingame <mike.burlingame@me.com
wrote:
Crash was resolved this weekend by a code update via jira case - no more crashes after the update - some more dial plan edits and a little more testing I will be ready to head over to pay the Wiki tax
Thanks guys for adding this feature it seems to be working as expected and my 491 issues on the B-Leg / A-Leg have pretty much disappeared
Thanks for checking in and letting us know where you stand. I really like it when these threads have some sort of resolution. If you have any issues with wiki editing please let me know or hop on IRC for an assist.
-MC _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Sorry about that yesterday was a long day On Sep 14, 2012, at 12:08 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
You did it again...
I need the DEBUG level in the logs
sofia global siptrace on console loglevel debug fsctl loglevel debug
On Thu, Sep 13, 2012 at 6:49 PM, Mike Burlingame <mike.burlingame@me.com> wrote: So what I am seeing is if the call flow is like
Carrier --> OpenSIPs --> FS --> OpenSIPs --> Off Network things seem fine
but if the flow is
Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> PSTN things seem broken now if I remove wait for ACK from Server A's B-Leg the call works as expected
FS_SERVER_A http://pastebin.freeswitch.org/19898
FS_SERVER_B http://pastebin.freeswitch.org/19899
On Sep 13, 2012, at 3:58 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
sigh, you forgot "console loglevel debug" this is only info loglevel.........
On Thu, Sep 13, 2012 at 5:49 PM, Mike Burlingame <mike.burlingame@me.com> wrote: So it seems all is not well - missing ACK and forward of 200OK when 2 FS servers are in the mix when sip_wait_for_aleg_ack=true is enabled basic flow is
FS Server A FS Server B |-----------------DID Channel---------------------| |---------Termination Channel (RCF)----------| Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> Carrier
Both FS running latest GIT as of this morning - Both FS servers using sip_wait_for_aleg_ack=true - Everything on FS B looks ok (besides not getting an ACK back from FS_A)
FS Server A Logs http://pastebin.freeswitch.org/19897
FS Server B SIP Logs http://pastebin.freeswitch.org/19896
On Sep 11, 2012, at 4:56 PM, Michael Collins <msc@freeswitch.org> wrote:
On Tue, Sep 11, 2012 at 11:39 AM, Mike Burlingame <mike.burlingame@me.com> wrote: Crash was resolved this weekend by a code update via jira case - no more crashes after the update - some more dial plan edits and a little more testing I will be ready to head over to pay the Wiki tax
Thanks guys for adding this feature it seems to be working as expected and my 491 issues on the B-Leg / A-Leg have pretty much disappeared
Thanks for checking in and letting us know where you stand. I really like it when these threads have some sort of resolution. If you have any issues with wiki editing please let me know or hop on IRC for an assist.
-MC _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
It was getting tripped up from the bypass media invite. Try latest. On Fri, Sep 14, 2012 at 2:42 PM, Mike Burlingame <mike.burlingame@me.com>wrote:
Sorry about that yesterday was a long day
On Sep 14, 2012, at 12:08 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
You did it again...
I need the DEBUG level in the logs
sofia global siptrace on console loglevel debug fsctl loglevel debug
On Thu, Sep 13, 2012 at 6:49 PM, Mike Burlingame <mike.burlingame@me.com>wrote:
So what I am seeing is if the call flow is like
Carrier --> OpenSIPs --> FS --> OpenSIPs --> Off Network things seem fine
but if the flow is
Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> PSTN things seem broken now if I remove wait for ACK from Server A's B-Leg the call works as expected
FS_SERVER_A http://pastebin.freeswitch.org/19898
FS_SERVER_B http://pastebin.freeswitch.org/19899
On Sep 13, 2012, at 3:58 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
sigh, you forgot "console loglevel debug" this is only info loglevel.........
On Thu, Sep 13, 2012 at 5:49 PM, Mike Burlingame <mike.burlingame@me.com>wrote:
So it seems all is not well - missing ACK and forward of 200OK when 2 FS servers are in the mix when sip_wait_for_aleg_ack=true is enabled basic flow is
FS Server A FS Server B |-----------------DID Channel---------------------| |---------Termination Channel (RCF)----------| Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> Carrier
Both FS running latest GIT as of this morning - Both FS servers using sip_wait_for_aleg_ack=true - Everything on FS B looks ok (besides not getting an ACK back from FS_A)
FS Server A Logs http://pastebin.freeswitch.org/19897
FS Server B SIP Logs http://pastebin.freeswitch.org/19896
On Sep 11, 2012, at 4:56 PM, Michael Collins <msc@freeswitch.org> wrote:
On Tue, Sep 11, 2012 at 11:39 AM, Mike Burlingame < mike.burlingame@me.com> wrote:
Crash was resolved this weekend by a code update via jira case - no more crashes after the update - some more dial plan edits and a little more testing I will be ready to head over to pay the Wiki tax
Thanks guys for adding this feature it seems to be working as expected and my 491 issues on the B-Leg / A-Leg have pretty much disappeared
Thanks for checking in and letting us know where you stand. I really like it when these threads have some sort of resolution. If you have any issues with wiki editing please let me know or hop on IRC for an assist.
-MC _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Thanks updating and will report back On Sep 14, 2012, at 2:29 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
It was getting tripped up from the bypass media invite. Try latest.
On Fri, Sep 14, 2012 at 2:42 PM, Mike Burlingame <mike.burlingame@me.com> wrote: Sorry about that yesterday was a long day
On Sep 14, 2012, at 12:08 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
You did it again...
I need the DEBUG level in the logs
sofia global siptrace on console loglevel debug fsctl loglevel debug
On Thu, Sep 13, 2012 at 6:49 PM, Mike Burlingame <mike.burlingame@me.com> wrote: So what I am seeing is if the call flow is like
Carrier --> OpenSIPs --> FS --> OpenSIPs --> Off Network things seem fine
but if the flow is
Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> PSTN things seem broken now if I remove wait for ACK from Server A's B-Leg the call works as expected
FS_SERVER_A http://pastebin.freeswitch.org/19898
FS_SERVER_B http://pastebin.freeswitch.org/19899
On Sep 13, 2012, at 3:58 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
sigh, you forgot "console loglevel debug" this is only info loglevel.........
On Thu, Sep 13, 2012 at 5:49 PM, Mike Burlingame <mike.burlingame@me.com> wrote: So it seems all is not well - missing ACK and forward of 200OK when 2 FS servers are in the mix when sip_wait_for_aleg_ack=true is enabled basic flow is
FS Server A FS Server B |-----------------DID Channel---------------------| |---------Termination Channel (RCF)----------| Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> Carrier
Both FS running latest GIT as of this morning - Both FS servers using sip_wait_for_aleg_ack=true - Everything on FS B looks ok (besides not getting an ACK back from FS_A)
FS Server A Logs http://pastebin.freeswitch.org/19897
FS Server B SIP Logs http://pastebin.freeswitch.org/19896
On Sep 11, 2012, at 4:56 PM, Michael Collins <msc@freeswitch.org> wrote:
On Tue, Sep 11, 2012 at 11:39 AM, Mike Burlingame <mike.burlingame@me.com> wrote: Crash was resolved this weekend by a code update via jira case - no more crashes after the update - some more dial plan edits and a little more testing I will be ready to head over to pay the Wiki tax
Thanks guys for adding this feature it seems to be working as expected and my 491 issues on the B-Leg / A-Leg have pretty much disappeared
Thanks for checking in and letting us know where you stand. I really like it when these threads have some sort of resolution. If you have any issues with wiki editing please let me know or hop on IRC for an assist.
-MC _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
That addressed that failure - I am testing a few other failures and will report back thanks for the quick updates On Sep 14, 2012, at 2:38 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
Thanks updating and will report back
On Sep 14, 2012, at 2:29 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
It was getting tripped up from the bypass media invite. Try latest.
On Fri, Sep 14, 2012 at 2:42 PM, Mike Burlingame <mike.burlingame@me.com> wrote: Sorry about that yesterday was a long day
On Sep 14, 2012, at 12:08 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
You did it again...
I need the DEBUG level in the logs
sofia global siptrace on console loglevel debug fsctl loglevel debug
On Thu, Sep 13, 2012 at 6:49 PM, Mike Burlingame <mike.burlingame@me.com> wrote: So what I am seeing is if the call flow is like
Carrier --> OpenSIPs --> FS --> OpenSIPs --> Off Network things seem fine
but if the flow is
Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> PSTN things seem broken now if I remove wait for ACK from Server A's B-Leg the call works as expected
FS_SERVER_A http://pastebin.freeswitch.org/19898
FS_SERVER_B http://pastebin.freeswitch.org/19899
On Sep 13, 2012, at 3:58 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
sigh, you forgot "console loglevel debug" this is only info loglevel.........
On Thu, Sep 13, 2012 at 5:49 PM, Mike Burlingame <mike.burlingame@me.com> wrote: So it seems all is not well - missing ACK and forward of 200OK when 2 FS servers are in the mix when sip_wait_for_aleg_ack=true is enabled basic flow is
FS Server A FS Server B |-----------------DID Channel---------------------| |---------Termination Channel (RCF)----------| Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> Carrier
Both FS running latest GIT as of this morning - Both FS servers using sip_wait_for_aleg_ack=true - Everything on FS B looks ok (besides not getting an ACK back from FS_A)
FS Server A Logs http://pastebin.freeswitch.org/19897
FS Server B SIP Logs http://pastebin.freeswitch.org/19896
On Sep 11, 2012, at 4:56 PM, Michael Collins <msc@freeswitch.org> wrote:
On Tue, Sep 11, 2012 at 11:39 AM, Mike Burlingame <mike.burlingame@me.com> wrote: Crash was resolved this weekend by a code update via jira case - no more crashes after the update - some more dial plan edits and a little more testing I will be ready to head over to pay the Wiki tax
Thanks guys for adding this feature it seems to be working as expected and my 491 issues on the B-Leg / A-Leg have pretty much disappeared
Thanks for checking in and letting us know where you stand. I really like it when these threads have some sort of resolution. If you have any issues with wiki editing please let me know or hop on IRC for an assist.
-MC _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Please report issues to jira.freeswitch.org and be aware we have commercial support as explained in most of the footers on this mL. On Fri, Sep 14, 2012 at 4:42 PM, Mike Burlingame <mike.burlingame@me.com>wrote:
That addressed that failure - I am testing a few other failures and will report back thanks for the quick updates
On Sep 14, 2012, at 2:38 PM, Mike Burlingame <mike.burlingame@me.com> wrote:
Thanks updating and will report back
On Sep 14, 2012, at 2:29 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
It was getting tripped up from the bypass media invite. Try latest.
On Fri, Sep 14, 2012 at 2:42 PM, Mike Burlingame <mike.burlingame@me.com>wrote:
Sorry about that yesterday was a long day
On Sep 14, 2012, at 12:08 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
You did it again...
I need the DEBUG level in the logs
sofia global siptrace on console loglevel debug fsctl loglevel debug
On Thu, Sep 13, 2012 at 6:49 PM, Mike Burlingame <mike.burlingame@me.com>wrote:
So what I am seeing is if the call flow is like
Carrier --> OpenSIPs --> FS --> OpenSIPs --> Off Network things seem fine
but if the flow is
Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> PSTN things seem broken now if I remove wait for ACK from Server A's B-Leg the call works as expected
FS_SERVER_A http://pastebin.freeswitch.org/19898
FS_SERVER_B http://pastebin.freeswitch.org/19899
On Sep 13, 2012, at 3:58 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
sigh, you forgot "console loglevel debug" this is only info loglevel.........
On Thu, Sep 13, 2012 at 5:49 PM, Mike Burlingame <mike.burlingame@me.com
wrote:
So it seems all is not well - missing ACK and forward of 200OK when 2 FS servers are in the mix when sip_wait_for_aleg_ack=true is enabled basic flow is
FS Server A FS Server B |-----------------DID Channel---------------------| |---------Termination Channel (RCF)----------| Carrier --> OpenSIPs --> FS --> OpenSIPs --> OpenSIPs --> FS --> OpenSIPs --> Carrier
Both FS running latest GIT as of this morning - Both FS servers using sip_wait_for_aleg_ack=true - Everything on FS B looks ok (besides not getting an ACK back from FS_A)
FS Server A Logs http://pastebin.freeswitch.org/19897
FS Server B SIP Logs http://pastebin.freeswitch.org/19896
On Sep 11, 2012, at 4:56 PM, Michael Collins <msc@freeswitch.org> wrote:
On Tue, Sep 11, 2012 at 11:39 AM, Mike Burlingame < mike.burlingame@me.com> wrote:
Crash was resolved this weekend by a code update via jira case - no more crashes after the update - some more dial plan edits and a little more testing I will be ready to head over to pay the Wiki tax
Thanks guys for adding this feature it seems to be working as expected and my 491 issues on the B-Leg / A-Leg have pretty much disappeared
Thanks for checking in and letting us know where you stand. I really like it when these threads have some sort of resolution. If you have any issues with wiki editing please let me know or hop on IRC for an assist.
-MC
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
participants (5)
-
Anthony Minessale
-
Michael Collins
-
Mike Burlingame
-
SamyGo
-
Sergey Okhapkin