similar to: ${HASH(SIP_CAUSE, ...)} and peer name

Greetings, Recently a performance regression in chan_sip was discovered in Asterisk 1.8. The regression is caused by chan_sip setting MASTER_CHANNEL(HASH(SIP_CAUSE,<chan name>)) after each response received on a channel. That feature has been made optional in the latest 1.8 SVN code, but is currently still enabled by default. After some internal discussion, we decided to consider disabling

Hello, I read on the wiki : Asterisk 1.8 will allow to read SIP response codes in the dialplan via *${HASH(SIP_CAUSE,<channel-name>)}*. Additionally make sure you're using the destination channel, not the source channel. But when I use this in my dialplan, this 'variable' is empty. Dialplan : exten => h,n,NoOp(sip cause = ${HASH(SIP_CAUSE,${CHANNEL})}) exten =>

Hi, I am experimenting with getting hold of the sip cause and sip response from outgoing call. How could i make a userevent printing the sip cause and/or sip response. I have tried using hangupcause, sip_cause and such , but i am not getting any data. I would at least like to use the q.850 reason codes in the dialplan which i now am unable to do. Any help appreciated. [Beskrivning: Fogwise -

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 =>

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

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.

I was running benchmark on IO performance using iozone3. In my build, the dom0 resides on a small usb stick and all the storage comes from a NFS mount. I test NFS performance on both dom0 && domU, mounting from the same server. The dom0 test works just well, but the domU run suffers from unstable NFS mount. Since this is a NFS root, the domU just appear to be freezed. The log from both

I was running benchmark on IO performance using iozone3. In my build, the dom0 resides on a small usb stick and all the storage comes from a NFS mount. I test NFS performance on both dom0 && domU, mounting from the same server. The dom0 test works just well, but the domU run suffers from unstable NFS mount. Since this is a NFS root, the domU just appear to be freezed. The log from both

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

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

Hi I have some question about mysql realtime. 1) Anyone know exactly if there is a specific order to declare sip table column for realtime ? In which file can I find that order ? 2) In my extconfig.conf, [settings] are : sipusers => mysql,general,siptable sippeers => mysql,general,siptable so means that I use realtime dynamic exactly ? Is it normal if some parameters from sip.conf still

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

Hello Everyone. In my outbound contexts, I'm using "${DIALSTATUS}" to fail over to other routes if the chosen route rejects the call. Now, My current scenario is if I get "BUSY" back from the first provider, I send a busy back to my customer. If I get something like CHANUNAVAIL (Like a SIP 503) I advance to the next carrier and attempt the call. This works

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

Hello, How can I get peerip, recvip, from, uri, useragent, peername, t38passthrough variables in (within) my custom Asterisk application? I can't use chan_sip.c internal structures (such as sip_pvt) in my custom application, because there's no chan_sip.h and I can't include it into my application (maybe there's other way?). I can do like this: exten =>

try, database get SIP/Registry/<peername> it gives you a string which contains the info, then pass it to CUT to extract ip-adr and port Freddi > To do that you need to get the remote ip address and port of the sip peer! > > I found the function: > > ${SIPPEER(exten:ip) > > But how can I get the port??? > >

I went from asterisk 1.4.22 (which was working) to SVN and I am getting the message rejected because extension not found... How can I modify the print statement in chan_sip.c line 18388 to include not just the extension but the context its trying to find my extension in??? ast_log(LOG_NOTICE, "Call from '%s' to extension"

Hello, I upgraded from asterisk 1.6.2.7 to asterisk 1.6.2.17. I have context=defcontext set in sip.conf. For each peer I have context=outcontext in the peer definition since I want outgoing calls from registered SIP peers to go through context 'outcontext'. This used to work in the older version (1.6.2.7), but after upgrading this has stopped working. Now outgoing calls are going to

I need to access number received after a I dial a SIP or H323 call? suppose I get one of these: *404 Not found **486 Busy here **408 Request Timeout **480 Temporarily unavailable **480 Temporarily unavailable **403 Forbidden (+) ** 410 Gone **301 Moved Permanently **410 Gone ** 404 Not Found (=) **502 Bad Gateway **484 Address incomplete* How do I get the 404, 486, etc. F.A. -------------- next

The Asterisk Development Team announces the release of Asterisk 1.8.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.7.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! Please note that a significant numbers of changes and fixes have