similar to: iax2_read: I should never be called - issue 8286

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Hi All, I'm having a problem after reinstalling the operating system. Festival works fine for SIP, but when IAX users are calling the same extension they don't hear the festival and I see the next message on console: NOTICE[3996]: chan_iax2.c:2995 iax2_read: I should never be called! I googled and couldn't find a solution, if somebody can help.... neobase*CLI>

Hi, Recently, one of my Asterisk servers stopped connecting calls and required a reboot to "fix it" (did not try to restart or reload). The log showed loads of this message: NOTICE[302] chan_iax2.c: I should never be called! This highly repeated message seems to be preceded by something like: WARNING[10767] channel.c: Exceptionally long voice queue length queuing to

We fairly recently switched service providers for our 4 PRI circuits. Since that time, we started to notice some failed inbound calls. These calls terminate with an ISDN cause code 47 'resource unavailble'. Most of the time I see this error on the first or second channel on the second span in a trunk group (This is the providers trunk group for hunting, not an Asterisk trunk group). All

We upgraded our asterisk server to 1.2.18 last night to pick up the security update. Since then, any calls coming in on IAX2 links get dropped if they try to enter a MeetMe conference room. The log shows this: Apr 26 08:33:16 NOTICE[27362]: chan_iax2.c:3167 iax2_read: I should never be called! Hanging up. I've temporarily worked around it by switching our inbound provider to use SIP

I am recieving inbound calls to my asterisk box over IAX. And getting "spawn extension....exited non-zero" errors. Im not entirely sure what this means, and would appreciate any help in fixing my problem. FYI: ********** is the inbound phone number x.x.x.x is a remote asterisk box calling my own asterisk box. When I choose it to dial an internal extension using this dialplan: exten

I've * in a small office with 10 internal sip extensions on aastra's. Outgoing is pstn over dahdi, voip over teliax and iax to another office. This morning no calls could be made: iax to branch offfices, voip iax over teliax, pstn, or even internal extensions. The aastra's showed "Not in Service". A "core restart now" got everything working again. Before I

Hello Previous context :- After Looking up sip and IAX2 that require configuration at router level which may cause some problems like connection break etc... so i left them ......... and start wondering if there is some thing that dont require configuration at router layer. The task to accomplish to make and recieve calls from outside local network using any protocol whose soft phones are

The Asterisk Development Team would like to announce the release of asterisk-18.21.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.21.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-18.21.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.21.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-20.6.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.6.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-20.6.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.6.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-21.1.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/21.1.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-21.1.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/21.1.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-18.20.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.20.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-18.20.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.20.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-20.5.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.5.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-20.5.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.5.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

Hi all! I downloaded right mpg123, chabged path to mpg123 binary in app_mp3.c, rebuilt app_mp3.so, and got MusicOnHold to work. But MP3Player refuses to do properly: -- Accepting AUTHENTICATED call from x.x.x.x, requested format = 1024, actual format = 1024 -- Executing Answer("IAX2/maxhome@maxhome/3", "") in new stack -- Executing

I'm trying the latest bri 0.1.0 RC2 drivers. In announce I see implementation of so long waited Transfer feature. But I can't make it work. When the person who is making transfer after talking with second party press "R" second time to establish 3 way call the person to which call supposed to be transfered being disconnected. Any ideas whats wrong? Thanks, Dmitry