similar to: Calls not hanging up

My last problem was nicely solved through this mailing list so hopefully this new problem will have the same happy outcome. My situation is that I have many extensions. Here is a sample: [client-phone](!) type=friend host=dynamic secret=XXXXXXXXXX dtmfmode=auto disallow=all allow=ulaw allow=gsm allow=g723 allow=ilbc subscribemwi=no [4165555555](client-phone) secret=xxxxxxxxxxxxxxxxxxxxxxxxxx

On 2017-04-18 02:42 AM, Pete Mundy wrote: > Try this: > > asterisk -r > core set verbose 10 > [get user to trigger fault] > [examine console output, and post to list if still unclear] > > If you don't solve it yourself, then we'll be able to help further once > we've seen the output. I can't see much more than at my previous debug level but here it is

On 2017-04-18 08:17 PM, Pete Mundy wrote: >> On 19/04/2017, at 7:58 am, D'Arcy Cain <darcy at VybeNetworks.com >> <mailto:darcy at VybeNetworks.com>> wrote: >> >> <snip> >> Everything looks the same as another one that works except for two >> things. The one that works doesn't have the "Probation passed" lines. >> I am

We have a template for extensions and voicmail. They look like this: exten => %ACCOUNT%,1,Verbose(0,Entering extension %ACCOUNT%) same => n(DialDesk),Verbose(0,${CALLERID(all)} Calling ${EXTEN}) same => n,Dial(SIP/%ACCOUNT%,30) same => n(VoiceMail),Set(CDR(userfield)=VoiceMail) same => n,Verbose(0,${CALLERID(all)} going into voice mail for %ACCOUNT%)

I have this in my sip.conf: exten => *98,1,Verbose(0,CALLERID number is "${CALLERID(num)}") same => n,VoicemailMain(${CALLERID(num)}@LocalSets,s) same => n,Hangup However, my extensions are set up so that they always show the external number, not the extension: [foobar2](client-phone) secret=xxxxxxxxxxxxxxxxxxxxxxxxxxxxx callerid=Candace <5555551212>

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.

On 12/3/19 3:04 PM, Joshua C. Colp wrote: > >     The AMI command, after the login, looks like this: > > > >     Action: Originate > >     Channel: SIP/outgoing/%%(destination)s > >     Context: LocalSets > >     CallerID: Vybe Consulting Inc Fax Service <5555551212> > >     Exten: sendfax > >   

On 12/2/19 11:52 AM, Joshua C. Colp wrote: > So I know that AMI is listening and I can talk to it.  Here is the > main log" > > [Nov 27 06:16:00] VERBOSE[101155] asterisk.c: Remote UNIX connection > [Nov 27 06:16:00] VERBOSE[101245] asterisk.c: Remote UNIX connection > disconnected > [Nov 27 06:16:01] VERBOSE[101244] manager.c: Manager

On 2017-04-19 11:57 AM, J Montoya or A J Stiles wrote: > I fished this out of an old extensions.conf from a defunct project. It might > be relevant to your use case: > > exten => 1571,1,NoOp(Call to 1571: voicemail retrieval) > exten => 1571,n,AGI(lookup_caller_id.agi,${CALLERID(num)}) > exten => 1571,n,NoOp(CLID is ${clid}) > exten =>

I recently upgraded from Asterisk 13.19 to 16.6.1. Everything is working fine with a few minor tweaks except outgoinf fax. Incoming works fine. I do outgoing faxing through an AMI call. Here is the output from the security log: [Nov 27 06:16:05] SECURITY[101222] res_security_log.c:

On 2017-04-20 05:14 AM, J Montoya or A J Stiles wrote: > This is just screaming "configuration mismatch" -- or, possibly, "latent bug > whereby things parsed in separate places should be treated the same, but are > actually getting treated differently". I really don't want to be the "my system isn't working so there must be a bug in Asterisk" guy

I am having troubles with sending faxes. I hope someone can help me work out a better method. Basically we have a special address that our users can send to. It winds up on our Asterisk server which runs a Python script that parses the message for attachments and the phone number from the recipient address. The attachments are converted to TIFF and stored in a folder with various information

Asterisk 11.1.0 One queue with strategy=leastrecent. (Full queues.conf below.) Occasionally (several times today), a caller will get "stuck" in the queue - there are operators available to take the call, but the caller stays in the queue for a long time. Any idea what might cause this, or where I can start looking to debug it? I'm going to start digging through the queue log

On Tue, 30 Aug 2016 10:39:14 -0400 Eric Wieling <ewieling at nyigc.com> wrote: > The dialplan below cannot go to voicemail, either something else is Of course not. It's the individual extensions that have voice mail. I have a similar problem when one of those destinations is a cell phone but I know that there is no Asterisk solution for that problem. If the cell phone answers and

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 If you are using PJSIP it's more complex same => n,Set(EndPoint=${CHANNEL(endpoint)}) ; Get the peer same =>

Hello, I have Asterisk version: Asterisk SVN-branch-11-r420435 I have the following code: exten => 303,1,NoOp(Dialing ${EXTEN}) ? ? ? ? same => n,NoOp(DBKey = ${DBKey}) ? ? ? ? same => n,DB_DELETE(office/${DBKey}) ? ? ? ? ? same => n,Playback(auth-thankyou) ? ? ? ? same => n,Hangup() And I get the following error: [2014-08-09 18:00:30] WARNING[4338][C-00000067]: pbx.c:4869

I have an extension that looks like this: exten => 5555551111,1,Verbose(Door buzzer calling) same => n,Dial(SIP/user1&SIP/user2&SIP/user3) The idea is that any of the three users can answer the phone to let someone in. The problem is that if, say, user2 unplugs his phone then the call immediately goes to his voice mail and the other two do not have the ability to open the door.

Running 11.10.2 on NetBSD 6.1.4 but I observed this on 11.11.0 as well. I have a directory which, through a combination of NULL mount and UNION mount contains everything in the installed config directory /usr/pkg/etc/asterisk except for my modified versions of those files. Basically I mount_null /usr/pkg/etc/asterisk on /usr/local/etc/asterisk and then mount_union my SVN directory with my

Hi, If agents is already logged in via AgentLogin() and users dialled extension 300 which will be placed in Queue(support-queue). How to find out which agent is available I can put their Agent id in AgentRequest() ? If this is not a good approach then how it should be done? Agent should automatically get next call when he/she is available. extensions.conf [LocalSets] exten =>

I am seeing this in my log file :[Aug 7 21:35:24] ERROR[19582] acl.c: Cannot create socket [Aug 7 21:35:24] WARNING[19582][C-00000283] res_rtp_asterisk.c: Unable to allocate RTP socket: Too many open files [Aug 7 21:35:24] NOTICE[19582][C-00000283] chan_sip.c: Failed to authenticate device "677"<sip:677 at IP>;tag=3637370132313231383238343335 [Aug 7 21:35:24] WARNING[19734]