similar to: TIMEOUT() function missing

I have a new Sangoma A102 and I'm trying to get it running in asterisk. A look through the dmesg log shows the card is detected and the various channels created. However, when I start asterisk I get the error below. Any ideas? My zapata.conf is below. Thanks, MD == Registered custom function SIPCHANINFO == Registered custom function CHECKSIPDOMAIN == Manager registered action

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 1.6.2.11-rc2 I've noticed a bunch of DEBUG statements on startup, such as: == Registered custom function 'SIP_HEADER' [Aug 9 07:01:23] DEBUG[17330]: xmldoc.c:1727 xmldoc_build_field: Cannot find variable 'SIPPEER' in tree 'description' == Registered custom function 'SIPPEER' [Aug 9 07:01:23] DEBUG[17330]: xmldoc.c:1727 xmldoc_build_field: Cannot find

I have an Ubuntu system and went into Synaptic and checked asterisk for installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc and got the following output with several errors and notices. Do I need to do more or are these ok? I expected to have some conf files in /etc/asterisk but there is nothing there. Thanks! Created by Mark Spencer <markster@digium.com>

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 everyone, I'm Andrea, and I've started working with Asterisk a couple of weeks ago, so I'm still a newbie. :) I was reading about Asterisk Realtime, and I was wondering if I can mix Static realtime and "Real" realtime configuration. For instance: can I have a "Static Realtime" extensions.conf and use "Real Realtime" sippeers and sipusers? Moreover,

Hi guys, I just ran into a funny issue here. I'm trying to virtualize our asterisk pbx onto vmware esxi. Here's a quick glance of the system: * Ubuntu 9.10 i386 with linux-rt kernel (to get 1000Hz timer) everything up2date. * Asterisk 1.6.2.6 If I run asterisk using the debian init script in contrib/init.d/, top shows asterisk is using 99.x% CPU doing nothing. If I run asterisk with

Hi I'm struggling with setting up SIP domains. If I specify a domain and a context in [general], that context overrides any set in type=peer blocks elsewhere. This results in incoming calls from PSTN gateways I use arriving in the wrong context. If I don't specify a context (which the docs I've found suggest is valid), then I get: 2006-05-12 07:36:16 WARNING[95290]:

Hi folks, I used the ast2sql.pl script (found on www.voip-info.org) to put into the database a simple sip.conf. Among other entries, you could find: [general] context=sip-in ;incoming sip calls Well, the script put the comment into the database entry, and asterisk started complaining about a 'sip-in ;incoming sip calls' context not found in extensions.conf. IMHO the comments should be

Hello *, do queues allow me to set an announce like the A() option of the Dial() cmd? The announce that I've found is a message that is heard by the caller. I'd like to send a message to the member of the queue that picks up the call. Thanks in advance, -- Dott. Andrea Spadaccini Multimedia Technologies Institute - MTI S.r.l.

Hello everybody, I've got a 56k usb modem, lsusb says: Bus 002 Device 002: ID 0572:130 Conexant Systems (Rockwell), Inc. I'd like to let it work with Asterisk. I think that I should use chan_modem and/or chan_modem_bestdata, but I found little or no documentation. Can anybody please post some instructions? Thanks in advance, -- Dr. Andrea Spadaccini Multimedia Technologies Institute

Hello *, I have a strange problem with the MAPI proxy AstManProxy: sometimes it happens that I send a request and I receive a response to ANOTHER request that it got in the frame time between my request and my response. Did anyone else notice this behaviour? How can this be solved? I've been reading the source code, but I didn't find a solution. Thanks in advance, -- Dr. Andrea

So I saw mention of a way to allow dialing using SIP URI's on Dave McNett's site at http://slacker.com/~nugget/projects/asterisk/page7 Wow, awesome, I can call anywhere now. However, I think there is a more elegant way of figuring out whether or not the local * server should handle a given domain. Specifically, Dave compares a series of domains within extensions.conf to figure out how to

Hello everybody, I'd like to have more detailed records for calls related to queues. For instance, if A enters in queue X, waits for Y secs and then talks to peer Z for T seconds, I'd like to have two entries in my CDR: - src: A, dst: X, duration: Y, state: ANSWERED - src: A, dst: Z, duration: T, state: ANSWERED This independently from how many peers the Queue app calls without success

You can use the following: switch3*CLI> show function SIPCHANINFO switch3*CLI> -= Info about function 'SIPCHANINFO' =- [Syntax] SIPCHANINFO(item) [Synopsis] Gets the specified SIP parameter from the current channel [Description] Valid items are: - peerip The IP address of the peer. - recvip The source IP address of the peer. - from

2008/11/4 Igor Zamocky <asterix at ponozky.sk> > > Did You tried http://www.voip-info.org/wiki/view/Asterisk+sip+limitonpeers > ? > I didn't. Now I did and it's working the way I wanted. Meanwhile, I had found a (complex) workaround using GROUP, GROUP_COUNT and SIPPEER but limitonpeers is much more concise. Thanks a lot. > > > Hi, > > > In this

Hi, In this thread http://lists.digium.com/pipermail/asterisk-users/2008-October/219592.html , I wondered whether SIPPEER curcalls was working. I could test this anew today. Here are my findings : Alice, Bob and Carol ar all using SIP Phones. Whenever Alice is calling Bob, - if Carol is calling Alice, SIPPEER(Alice:curcalls) equals 0 - if Carol is calling Bob, SIPPEER(Bob:curcalls) equals 1

Hi, I'm using Asterisk (1.2.4) as a voicemail system for our Softswitch. When forwarding a call to Voicemail, here is somehow what the softswitch sends to Asterisk : In INVITE : Vm Phone Number ( to route the call ) In To : Person who has been called ! In From : Person who was calling ! Of course, I need to send the call into the "Called User" Mailbox (Thus To SIP header) ! So

Hello, I've enabled BT-200's SYSLOG logging, and I get some message whose meaning is obscure to me. In particular, in a day I got the "Deletion of invalid timer" message almost ten times from one phone, which has some call problems. Can someone point me to a resource on BT200 error codes? Thanks, -- Andrea Spadaccini Multimedia Technologies Institute s.r.l.

Hi folks, I'm really confused, so please help me, or at least give me some pointers to clarify this issue. Can I mix "Static" and "Real" realtime? Is there a way to easily switch from one to another, say, for sip.conf? Which are the major benefits of "Real" realtime? Please help me! Thanks in advance, -- Andrea Spadaccini Multimedia Technologies Institute