similar to: Parallel dialoog with different Alert-Info headers

On 7/22/2023 4:51 PM, Dirk-Willem van Gulik wrote: > We have a couple of parallel ring settings (and this has worked well for eons). > > Either in the form of > > same => n,Dial(SIP/1001 & SIP/1002 & SIP/1003 …..) > > Or via a subroutine (below) that has a bit of extra logic: > > FOO = 1010 & 1019 & 1017 & 1033 > ... > same =>

> On 22 Jul 2023, at 23:40, asterisk at phreaknet.org wrote: > > On 7/22/2023 4:51 PM, Dirk-Willem van Gulik wrote: >> We have a couple of parallel ring settings (and this has worked well for eons). >> >> Either in the form of >> >> same => n,Dial(SIP/1001 & SIP/1002 & SIP/1003 …..) >> >> Or via a subroutine (below) that has a bit

On 7/23/2023 12:32 PM, Dirk-Willem van Gulik wrote: >> On 22 Jul 2023, at 23:40, asterisk at phreaknet.org wrote: >> >> I'm assuming you mean at the device level, and that you want to send >> only the relevant header to each device? >> Use pre-dial handlers; a unique handler runs on each destination >> channel. With PJSIP, you're forced to do this

Hello there, I appreciate any help about this problem that I can't figure out... I need to record all my calls: this is pretty easy using Monitor() before the Dial(). eg: exten => 425,n,Monitor(wav49,/var/spool/asterisk/monitor/425/${EPOCH}_${CALLERID(num)}_in,mb) exten => 425,n,Dial(${PHONE1},10) Now, I want to create a call group: I mean, I want a number (eg 800) that makes

According to the instructions given at https://www.asterisksounds.org/de I converted and installed German prompts successfully and for numbers, I can successfully listen to a German female voice counting or telling the date/time. But unlikily, somehow the voicemail prompt is still English, although my general language settings are "de". I use pjsip.conf, not sip.conf. In

I have this in my config: exten => _800XXXXXXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN}) same => n,Dial(SIP/tollfree/1${EXTEN}) exten => _1800XXXXXXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN}) same => n,Dial(SIP/tollfree/${EXTEN}) exten => _NXXNXXXXXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN}) same => n,Dial(SIP/trunk/1${EXTEN}) exten =>

Hello I change this from chan_sip.conf (see ipsvr): static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, const char *username, const char *fullcontact, int expirey) { char port[10]; char ipaddr[20]; char regseconds[20]; time_t nowtime; -> char ipsvr[20]; time(&nowtime); nowtime += expirey;

You can use a custom device state to do it. [dnd] ;DND Toggle exten => *363,1,Answer() same => n,Set(CURRENT_PRESENCE=${DEVICE_STATE(Custom:DND${CHANNEL(peername)})}) same => n,GotoIf($[${CURRENT_PRESENCE}=NOT_INUSE]?*78,1:*79,1) ;DND On exten => *78,1,NoOP(Turning DND On) same => n,Set(DEVICE_STATE(Custom:DND${CHANNEL(peername)})=BUSY) same =>

So the call used Alaw as Codec. > Am 13.06.2020 um 17:23 schrieb Luca Bertoncello <lucabert at lucabert.de>: > > Am 13.06.2020 um 13:47 schrieb Michael Keuter: > > Hi > >> Try "sip show peer <peername>" for a phone. > > So: > > mobile phone: > bpi*CLI> sip show peer 0049177xxxxxxx > > > > > * Name :

Hello, I'm trying to figure out what was the return code of SIP for a call. The problem is that HASH(SIP_CAUSE) require a peer name, but when I try to retrieve the peer name using ${CHANNEL(peername)}, I have an error message that CHANNEL does not have peername or it is not available to be used. I tried to print it with NOOP on a live channel, and also after hangup, both with the same error

On Fri, 12 Jun 2015 11:49:05 -0700 John Kiniston <johnkiniston at gmail.com> wrote: > Try this for CHAN_SIP: > > same => n,Set(Peer=${SIPCHANINFO(peername)}) ; Get the peer > same => n,Set(MailBox=${SIPPEER(${Peer},mailbox)}); Get the > mailbox same => n,VoicemailMain(${MailBox}@LocalSets,s) ; If we > have a mailbox defined log into it Perfect.

If I remove the password, how can anyone access the mailbox if the 'mailbox' prompt is not played? Nabeel On 30 Jul 2016 3:19 p.m., "D'Arcy J.M. Cain" <darcy at vex.net> wrote: > On Sat, 30 Jul 2016 06:43:47 +0100 > Nabeel <nabeelshikder at gmail.com> wrote: > > I am using Asterisk voicemail on a CentOS 7 server. I would like to > > be able to

Hi! I've added member to a queue like this, from queues.conf: member => SIP/1070@peername It works OK. But, after restaring I see in show queue that Members: SIP/1070@peername (Invalid) ... What does it mean? Why is it Invalid? BTW, reload command fixes it, so the member receives queue calls. Thanks! PS. 1.2.9.1 -- DSS5-RIPE DSS-RIPN 2:550/5068@fidonet 2:550/5069@fidonet

I tried your extension definition as suggested: exten => *98,1,Verbose(0,${CHANNEL(peername)} calling voicemail) same => n,VoicemailMain(${CHANNEL(peername)}@VoiceMail,s) same => n,Hangup But there was no change in the prompts asked, ie. the voice first asked for 'mailbox', and then 'password' as before. The prompts are not removed. Please clarify what you mean by the

Am 13.06.2020 um 13:47 schrieb Michael Keuter: Hi > Try "sip show peer <peername>" for a phone. So: mobile phone: bpi*CLI> sip show peer 0049177xxxxxxx * Name : 0049177xxxxxxx Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : default Record On feature : automon

Is there a recommended way to find out the cause of DIALSTATUS = CONGESTION for PRI/BRI channels? Currently I am evaluating the DIALSTATUS variable and I also count the active ISDN channels for the ISDN trunk in question. Counting the active ISDN channels seems somewhat clumsy as the mapping to a specific trunk must be done by hand (or write even more code). I have a setup where outgoing calls

https://bugzilla.mindrot.org/show_bug.cgi?id=2240 Bug ID: 2240 Summary: Secure PIN entry for smartcards through the keypad on the reader (patch) Product: Portable OpenSSH Version: -current Hardware: All OS: All Status: NEW Severity: enhancement Priority: P5

I hate to be a pest, but are there any revelations on file descriptor "overusage" by dovecot-auth? ----- Forwarded message from Steven F Siirila <sfs at tc.umn.edu> ----- Date: Sat, 20 Jan 2007 12:42:50 -0600 From: Steven F Siirila <sfs at tc.umn.edu> To: Dovecot Mailing List <dovecot at dovecot.org> User-Agent: Mutt/1.4.2.1i Subject: Re: [Dovecot] dovecot-auth file

The Asterisk Development Team has announced the release of Asterisk 1.6.2.17. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.17 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 1.6.2.17. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.17 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this