Re: [asterisk-users] Voicemail help when listening to messages
Hi, I have replicated this on a fresh 15.6.0 install, chan_sip, voicemail plain, vanilla config (whatever ships with the tucny.com RPMs) other than creating a SIP extension and voicemail user. Should I file a bug? > On 4/09/2018, at 3:46 PM, Nathan Ward wrote: > > Hi, > > I have a weird issue, unsure if it’s due to a bug, or configuration on my > end. We’re on 14.7.7. I’ve looked at the app_voicemail.c code, and see no > changes in this area of the code until the current version so don’t think age > of the code is an issue here (but happy to be proven wrong!). > > When hitting * for help when listening to messages (dial in, press 1, then > *), we get the following options: > > press 1 for new messages > press 2 to change folders > press 3 for advanced options > press 0 for mailbox options > press 1 to listen to new messages * > press 2 to access messages saved in other folders * > press 3 to record a message for another mailbox * > press 0 for greetings and password management * > press 5 to repeat the current message > press 6 to play the next message > press 7 to delete this message > press 8 to forward the message to another user > press 9 to save this message > press * for help or # to exit > > I’ve marked 4 items with *s, these are items which seem out of place, as the > other options largely work. > > I’ve been poking around, and it seems to be that this is happening because > `skipadvanced` is set to 1, though, it this is set to 0 it would (I think) be: > press 1 for new messages > press 2 to change folders > press 3 for advanced options > press 0 for mailbox options > press 3 for advanced options * > press 5 to repeat the current message > press 6 to play the next message > press 7 to delete this message > press 8 to forward the message to another user > press 9 to save this message > press * for help or # to exit > > Note the item with a * where advanced options is offered again (once from > vm-opts then later from vm-advopts). > > I am not clear on how this is supposed to work, as both with and without > skipadvanced set seems weird. Can anyone help? > > -- > Nathan Ward > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is it possible to retrieve header fields from a SIP UPDATE packet?
Working with a NEC switch. They send the initial INVITE to asterisk with a phantom NEC number in the From. NEC has told us they send the original caller id information as the From in an UPDATE packet. >From what I have read, only headers in the INVITE are accessible. Is it >correct that we cannot retrieve SIP headers from this UPDATE packet (occurs >after the INVITE)? This system has to be chan_sip because blind transfers outside of asterisk work using chan_sip. When we looked at PJSIP blind transfers to switches a couple years ago, we found problems with the Asterisk PJSIP support. Have a great day! Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup the _called_ channel ?
On 9/12/18 1:32 PM, Joshua Colp wrote:
On Wed, Sep 12, 2018, at 2:25 PM, sean darcy wrote:
On 9/12/18 1:22 PM, Joshua Colp wrote:
On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote:
I understand that HangUp() hangs up the calling channel. I want to
hangup the called channel.
SIP/mycall-x calls and bridges with DAHDI/1-1.
I send SIP/ to listen to a long, very long, file.
Define "send". How are you doing it?
GoSub(play-long-file,s,1)
You can't have a channel both in dialplan directly and also bridged to another
channel at the same time. There's not enough context or information to really
be able to answer without understanding fully.
Maybe this will help explain it. Here's the cli:
Executing [s@incoming:7] Dial("SIP/incall-0001",
"DAHDI/g0,55,tTD(:1)") in new stack
-- Called DAHDI/g0
-- DAHDI/1-1 answered SIP/incall-0001
-- Channel DAHDI/1-1 joined 'simple_bridge' basic-bridge
<5312c0a8-7697-4a97-b3ff-ff0484fbaf3d>
-- Channel SIP/incall-0001 joined 'simple_bridge' basic-bridge
<5312c0a8-7697-4a97-b3ff-ff0484fbaf3d>
-- SIP/incall-0001 Internal Gosub(long-file,s,1) start
-- Executing [s@long-file:1] Playback("SIP/incall-0001",
"long-file") in new stack
-- Playing 'long-file.slin' (language
'en')
-- Executing [s@long-file:2] Verbose("SIP/incall-0001",
"bridgepeer is DAHDI/1-1") in new stack
Executing [s@long-file:3] Hangup("SIP/incall-0001", "") in new stack
== Spawn extension (long-file, s, 3) exited non-zero on
'SIP/incall-0001'
[Sep 12 13:06:06] NOTICE[2217][C-0001]: app_stack.c:1082 gosub_run:
SIP/callcentric20-0001 Abnormal 'Gosub(long-file,s,1)' exit.
Popping routine return locations.
-- Channel SIP/incall left 'simple_bridge' basic-bridge
<5312c0a8-7697-4a97-b3ff-ff0484fbaf3d>
-- Channel DAHDI/1-1 left 'simple_bridge' basic-bridge
<5312c0a8-7697-4a97-b3ff-ff0484fbaf3d>
-- Hanging up on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
As you can see DAHDI/1-1 is not hungup until after Playback. I want to
hangup DAHDI/1-1 before the Playback.
Thanks,
--
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Re: [asterisk-users] hangup the _called_ channel ?
On Wed, Sep 12, 2018, at 2:25 PM, sean darcy wrote: > On 9/12/18 1:22 PM, Joshua Colp wrote: > > On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote: > >> I understand that HangUp() hangs up the calling channel. I want to > >> hangup the called channel. > >> > >> SIP/mycall-x calls and bridges with DAHDI/1-1. > >> > >> I send SIP/ to listen to a long, very long, file. > > > > Define "send". How are you doing it? > > > GoSub(play-long-file,s,1) You can't have a channel both in dialplan directly and also bridged to another channel at the same time. There's not enough context or information to really be able to answer without understanding fully. -- Joshua Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup the _called_ channel ?
On 9/12/18 1:25 PM, sean darcy wrote: On 9/12/18 1:22 PM, Joshua Colp wrote: On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote: I understand that HangUp() hangs up the calling channel. I want to hangup the called channel. SIP/mycall-x calls and bridges with DAHDI/1-1. I send SIP/ to listen to a long, very long, file. Define "send". How are you doing it? GoSub(play-long-file,s,1) [play-long-file] exten=s,1, ;;; Here I want to hangup DAHDI/1-1, the called channel same=n,Playback(very-long-file) same=n,Hangup() Is there a better way ? And I'm using dynamic features, applicationmap. play-file=*8,peer,GoSub,"pay-long-file,s,1" -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup the _called_ channel ?
On 9/12/18 1:22 PM, Joshua Colp wrote: On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote: I understand that HangUp() hangs up the calling channel. I want to hangup the called channel. SIP/mycall-x calls and bridges with DAHDI/1-1. I send SIP/ to listen to a long, very long, file. Define "send". How are you doing it? GoSub(play-long-file,s,1) [play-long-file] exten=s,1, ;;; Here I want to hangup DAHDI/1-1, the called channel same=n,Playback(very-long-file) same=n,Hangup() Is there a better way ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup the _called_ channel ?
On 9/12/18 1:22 PM, Joshua Colp wrote: On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote: I understand that HangUp() hangs up the calling channel. I want to hangup the called channel. SIP/mycall-x calls and bridges with DAHDI/1-1. I send SIP/ to listen to a long, very long, file. Define "send". How are you doing it? GoSub(play-long-file,s,1) [play-long-file] exten=s,1, ;;; Here I want to hangup DAHDI/1-1, the called channel same=n,Playback(very-long-file) same=n,Hangup() Is there a better way ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup the _called_ channel ?
On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote: > I understand that HangUp() hangs up the calling channel. I want to > hangup the called channel. > > SIP/mycall-x calls and bridges with DAHDI/1-1. > > I send SIP/ to listen to a long, very long, file. Define "send". How are you doing it? -- Joshua Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hangup the _called_ channel ?
I understand that HangUp() hangs up the calling channel. I want to hangup the called channel. SIP/mycall-x calls and bridges with DAHDI/1-1. I send SIP/ to listen to a long, very long, file. GoSub(play-long-file,s,1) [play-long-file] exten=s,1, ;;; Here I want to hangup DAHDI/1-1, the called channel same=n,Playback(very-long-file) same=n,Hangup() How do I hangup the called channel, and leave the calling channel listening to the file ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
