similar to: Default caller ID

The asterisk wiki states that it needs SIP, IAX2, IAX and RTP open to the world to work. Is this necessarily true, or does it only need some of these outgoing? I'm concerned as anyone that could guess an extension number&password could use my server to make outgoing calls. It would help if the extensions had a netmask/allowable IP setting like the iax.conf file uses, but there

Hi All, Having searched the archives, I can see there has been much discussion at various points regarding capture of caller id information from good old BT. If I understand correctly, it seems that not only do the drivers not currently support it, but my X101P possibly/probably can't do it anyway due to hardware? So, that leaves me with the modem route, which seems more and more unlikely,

So I just saw this VoIP-centric article at slashdot (http://slashdot.org/article.pl?sid=04/05/22/1840220) which mentions e164.org. It's a "non-profit public DNS root designed to map phone numbers to Internet protocols." Is anyone on this list actually using this? They have asterisk config instructions: http://www.e164.org/config.php I wonder if someone can help me understand

Hello *, is the rtpip patch still valid for asterisk 1.6 (with some code changes, obviously)? https://issues.asterisk.org/view.php?id=8161 Or, in asterisk 1.6 there is an alternative to using it? This is the difffile I produced for chan_sip.c in asterisk 1.6.2.11 --- chan_sip.c 2010-10-12 13:47:49.000000000 +0200 +++ chan_sip.c.orig 2010-10-12 13:47:27.000000000 +0200 @@ -987,9 +987,6 @@

Folks, I've been trying to handle the problem where blocked callerids appear as coming from asterisk <asterisk> on the email notification, and the message envelope simply doesn't say anything (does not actually play the vm-unknown message). So, following the tip provided by several previous posters, I tried putting this in my extensions.conf (the xx's are my DID,

ztdummy successfully compiles under kernel 2.6, but when I load it I get ztdummy: Unknown symbol fill_td ztdummy: Unknown symbol insert_td_horizontal ztdummy: Unknown symbol uhci_devices ztdummy: Unknown symbol uhci_interrupt ztdummy: Unknown symbol alloc_td ztdummy: Unknown symbol unlink_td ztdummy: Unknown symbol delete_desc I had a quick look at the source, and it looks like these function

Hi, I have access to two providers. On one of them the authuser is the same as the username, so outgoing works. On the other one I can only get incoming - what ever combination I try for outgoing I get an error. The register command has the ability to specify both usernames (which is why incoming works) but outgoing doesn't seem to, and without that I'm stuck. They are defined as:

why can't we compile the asterisk coading in windows, it's done in c++ so it should work in windows as well

Hello, I am relatively new to Asterisk and I need to compile the G.723.1 codec for Asterisk. I downloaded the ITU source code, placed it in the codecs directory, but apparently Asterisk needs a rather different library than the one provided from ITU. As I've seen in the mailing list archives, there are quite a few users who were able to compile G.723.1 in *, so, could someone kindly share it

I have been using Sixtel from the beginning of the year and service was getting worse and worse. Yesterday I tried to access the website to get the CDR and I got an error saying that the domain no longer exists. I checked the whois and it says that the domain is on hold. Have they finally folded? -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel:

Hi all, I've updated Asterisk trunk LDAP schema [0] [1] to include queues and other attributes needed for a working LDAP backend (I'll open a bug to include these changes on svn). SIP users and dialplan are perfectly working, but when I call a queue the whole Asterisk (1.6.2.0) crashes: on extconfig: [settings] sipusers => ldap,"dc=nodomain",sip sippeers =>

I'd also like to register my disappointment that the slides and videos aren't available. On Friday, October 16, 2009 4:46 PM, David Greene wrote: > When I agreed to be a speaker, I signed off on having my > talk made publicly available. There does seem to be a > double-standard here and that's concerning. There are few things about this whole situation that aren't

Hi list, My need is to append a site specific parameter to the Contact: header on all INVITEs exiting * via a SIP trunk. I'd like it to look something like this: Contact: <bob:3125551212 at 10.10.10.10;SITE-ID=us.here> where SITE-ID=us.here is set in a config file that * parses on startup. Or in a Dial() command option? Or I don't care exactly how. :-) It is possible to

We are a telco so when we receive calls via ISDN and the number is witheld we see the correct presentation value but also still see the actual callers number in the callerid(num) variable. I am trying to forward some of these calls over to one of our other boxes via SIP but I have found that if the number is withend then the sip packet contains :- From: "Anonymous" <sip:Anonymous

Hi all as per the new release notice for 13.33.0 received today - can anyone advise me the max limit of the string to the Dial Command - see * [ASTERISK-27946 <BLOCKED::https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't I have been fighting with this issue for months trying to find a solution I

Hi. Since a customer requested us that feature, I wrote this little patch for app_dial to allow to play an announcement to the called party, as soon he answers. you can define the file to play in the dial() option, using A(filename). for example: exten => blah,1,Dial(Zap/blah,30,rA(/my/own/announce)Tt) that doesn't break anything ... feel free to blame me for anything bad this patch

I'm getting this message when I start Asterisk chan_capi.c:2205 capi_handle_msg: Command.Subcommand = 0x5.0x81 but when I try and recompile I get this chan_capi.c:60: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) any help would be greatly appreciated. -- Dave Cotton <dcotton@linuxautrement.com>

I notice in the wiki you can have password sceme overides (such as {PLAIN}password). Would this work with the password fields in SQL databases (as it isn't clear in the wiki or conf file)? Regards Andrew -- Andrew Hutchings Linux Guru Netserve Consultants Ltd. http://www.domaincity.co.uk/

On Wed, 2019-02-06 at 11:56 -0800, H. Peter Anvin wrote: > CAUTION: This email originated from outside of the organization. Do not click links or open attachments unless you recognize the sender and know the content is safe. > > > On 2/6/19 11:44 AM, Joakim Tjernlund wrote: > > On Wed, 2019-02-06 at 11:34 -0800, H. Peter Anvin wrote: > > > > Great, that tree now

hey guys I thought I was making progress on my dialplan when I realized that the class features that are available for zap channels aren't available for SIP channels. I see references in the archives to adding pattern matches in the dialplan for CLASS features which has raised a couple questions. 1. Is implementing CLASS like features via the dialplan the currently recommended way to do