Re: [asterisk-users] voicemail message not deleted
> Am 26.07.2021 um 07:28 schrieb Fourhundred Thecat <400the...@gmx.ch>: > > Hello, > > I have this in my voicemail.conf: > > attach=yes > > delete=yes > > I do get an email when new voicemail is received, and I do get the > voicemail message as attachment. > > However, the original message is not deleted from the sevber. > > How do I delete the message, after it has been sent per email as > attachment? I don't want to store messages on the server indefinitely. > > thanks, > > -- I think you need to set "delete=yes" as option per mailbox account. 100 => 1234,Test,,,delete=yes The global setting is only an example. Michael http://www.mksolutions.info -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] problems with natted phones
I currently disabled also RTSP ALG and rebooted the router. Fixed for now. I do not know for how long. Marek 2021-07-26 8:54 GMT+02:00, Marek Greško : > Hmm, back to original problem. My happines was premature. Today one of > the phones have no audio again. I see packets from lan segment again. > > I double checked the router configuration. SIP ALG is disabled. There > are also another ALGs present: > > NAT ALG > RTSP ALG > PPTP ALG > IPSEC ALG > > Which of them are neede to be disabled? > > As of my observations these problems are triggered by reboots on > asterisk side. How could this be related? (I may be wrong.) > > Thanks > > Marek > > > > 2021-07-23 14:54 GMT+02:00, Marek Greško : >> I achieved a partial success adding --use-compact-form option. >> >> Marek >> >> >> 2021-07-23 13:47 GMT+02:00, Marek Greško : >>> Hello, >>> >>> your suggestion to turn off SIP ALG on provider's router was probably >>> correct. no problem until now. Thank you very much. >>> >>> I just found out another issue. I had a pjsue client in that network >>> which called specific number when turned on. It was working perfectly >>> with the old provider with working SIP ALG. But now with this provider >>> and SIP ALG disabled I am not able to make the call using pjsua >>> client. >>> >>> My pjsua config looks like this: >>> --id sip:ext@asterisk.domain >>> --registrar sip:asterisk.domain >>> --proxy sip:asterisk.domain >>> --outbound sip:asterisk.domain >>> --realm * >>> --username username >>> --password password >>> --null-audio >>> --no-tcp >>> --max-calls=1 >>> --no-vad >>> >>> The pjsua client successfully registers but is unable to call. >>> >>> I see the following: >>> IP address change detected for account 1 >>> (localip:5060-->nattedip:newport). Updating registration (using method >>> 4) >>> Temporary failure in sending Request msg INVITE/cseq=, will try >>> next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT) >>> >>> What could be the problem? How can I convince pjsue to work correctly >>> behind nat? >>> >>> Thanks >>> >>> Marek >>> >>> >>> >>> >>> >>> 2021-07-10 11:08 GMT+02:00, Marek Greško : Hello, I just disabled. Currently it is working. I shloud give it some time to confirm the problem has gone. Maybe one month would be enough to confirm. Thanks Marek 2021-07-09 20:11 GMT+02:00, Abdenasser Ghomri : > Yes just disable the SIP ALG and see if it helps, Thanks. > > Best Regards, > > On Fri, Jul 9, 2021, 09:10 Antony Stone < > antony.st...@asterisk.open.source.it> wrote: > >> On Friday 09 July 2021 at 08:47:46, Marek Greško wrote: >> >> > Hello, >> > >> > yes SIP ALG are anbled on the router. Should I disable? >> >> In my opinion, always. >> >> Antony. >> >> -- >> I don't know, maybe if we all waited then cosmic rays would write all >> our >> software for us. Of course it might take a while. >> >> - Ron Minnich, Los Alamos National Laboratory >> >>Please reply to >> the >> list; >> please >> *don't* >> CC >> me. >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> 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 > >>> >> > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] problems with natted phones
Hmm, back to original problem. My happines was premature. Today one of the phones have no audio again. I see packets from lan segment again. I double checked the router configuration. SIP ALG is disabled. There are also another ALGs present: NAT ALG RTSP ALG PPTP ALG IPSEC ALG Which of them are neede to be disabled? As of my observations these problems are triggered by reboots on asterisk side. How could this be related? (I may be wrong.) Thanks Marek 2021-07-23 14:54 GMT+02:00, Marek Greško : > I achieved a partial success adding --use-compact-form option. > > Marek > > > 2021-07-23 13:47 GMT+02:00, Marek Greško : >> Hello, >> >> your suggestion to turn off SIP ALG on provider's router was probably >> correct. no problem until now. Thank you very much. >> >> I just found out another issue. I had a pjsue client in that network >> which called specific number when turned on. It was working perfectly >> with the old provider with working SIP ALG. But now with this provider >> and SIP ALG disabled I am not able to make the call using pjsua >> client. >> >> My pjsua config looks like this: >> --id sip:ext@asterisk.domain >> --registrar sip:asterisk.domain >> --proxy sip:asterisk.domain >> --outbound sip:asterisk.domain >> --realm * >> --username username >> --password password >> --null-audio >> --no-tcp >> --max-calls=1 >> --no-vad >> >> The pjsua client successfully registers but is unable to call. >> >> I see the following: >> IP address change detected for account 1 >> (localip:5060-->nattedip:newport). Updating registration (using method >> 4) >> Temporary failure in sending Request msg INVITE/cseq=, will try >> next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT) >> >> What could be the problem? How can I convince pjsue to work correctly >> behind nat? >> >> Thanks >> >> Marek >> >> >> >> >> >> 2021-07-10 11:08 GMT+02:00, Marek Greško : >>> Hello, >>> >>> I just disabled. Currently it is working. I shloud give it some time >>> to confirm the problem has gone. Maybe one month would be enough to >>> confirm. >>> >>> Thanks >>> >>> Marek >>> >>> >>> 2021-07-09 20:11 GMT+02:00, Abdenasser Ghomri : Yes just disable the SIP ALG and see if it helps, Thanks. Best Regards, On Fri, Jul 9, 2021, 09:10 Antony Stone < antony.st...@asterisk.open.source.it> wrote: > On Friday 09 July 2021 at 08:47:46, Marek Greško wrote: > > > Hello, > > > > yes SIP ALG are anbled on the router. Should I disable? > > In my opinion, always. > > Antony. > > -- > I don't know, maybe if we all waited then cosmic rays would write all > our > software for us. Of course it might take a while. > > - Ron Minnich, Los Alamos National Laboratory > >Please reply to the > list; > please > *don't* > CC > me. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > 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 >>> >> > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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
