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...eg of the call as losing large numbers of packets. BTW, IIRC reinvites happen when a codec changes or the channel switches to T.38. Also Adtran SIP gateways appear not to support OPTIONS packets when running in SIP proxy mode, which is very annoying. At some point I'll try and arrange a slugfest between Digium and Adtran and they can figure out why it doesn't work. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Todd R. Sent: Monday, January 19, 2015 1:45 PM To: Asterisk-Users List Subject: Re: [asterisk-users] sip show...

...numbers of packets. BTW, > IIRC reinvites happen when a codec changes or the channel switches to T.38. > > > > Also Adtran SIP gateways appear not to support OPTIONS packets when > running in SIP proxy mode, which is very annoying. At some point I?ll > try and arrange a slugfest between Digium and Adtran and they can figure > out why it doesn?t work. > > > > *From:* asterisk-users-bounces at lists.digium.com [mailto: > asterisk-users-bounces at lists.digium.com] *On Behalf Of *Todd R. > *Sent:* Monday, January 19, 2015 1:45 PM > *To:* Asterisk-Users...

...ite leg of the call as losing large numbers of packets. BTW, IIRC reinvites happen when a codec changes or the channel switches to T.38. Also Adtran SIP gateways appear not to support OPTIONS packets when running in SIP proxy mode, which is very annoying. At some point I?ll try and arrange a slugfest between Digium and Adtran and they can figure out why it doesn?t work. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Todd R. Sent: Monday, January 19, 2015 1:45 PM To: Asterisk-Users List Subject: Re: [asterisk-users] sip show chan...

...>> IIRC reinvites happen when a codec changes or the channel switches to T.38. >> >> >> >> Also Adtran SIP gateways appear not to support OPTIONS packets when >> running in SIP proxy mode, which is very annoying. At some point I?ll >> try and arrange a slugfest between Digium and Adtran and they can figure >> out why it doesn?t work. >> >> >> >> *From:* asterisk-users-bounces at lists.digium.com [mailto: >> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Todd R. >> *Sent:* Monday, January 19, 2015 1:45...

...call as losing large numbers of packets. BTW, IIRC reinvites happen when a codec changes or the channel switches to T.38. > > Also Adtran SIP gateways appear not to support OPTIONS packets when running in SIP proxy mode, which is very annoying. At some point I'll try and arrange a slugfest between Digium and Adtran and they can figure out why it doesn't work. > > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Todd R. > Sent: Monday, January 19, 2015 1:45 PM > To: Asterisk-Users List > Subject: Re:...

I am seeing lots of lost packets when running the command sip show channelstats at the CLI. There are issues across multiple Asterisk servers I am trying to diagnose but everything I read seems to point to this command being pretty unreliable. Can I trust the info this command shows? I am showing lots of lost packets in sip show channelstats but I can't see any packet loss when pinging the