similar to: How to set busy

Anyone have any info on the date (or bug tracker number) of the change from incominglimit to call-limit, and is it call_limit or call-limit? Does it work with SIP friends? Running CVS head 8/24 (right before 1.2 release). -------------- next part -------------- An HTML attachment was scrubbed... URL:

Just starting to enjoy the full features of asterisk, I do have a couple questions though, that I can't seem to find answers for in the wiki, just wondering if someone could light my way. after a caller has made their choice of options in the dial plan, I would like them to be placed on "hold" (music, not ringing) while the system processes through the rest of the dial plan

There is plenty of information on the wiki for setting asterisk up for transferring calls both from the Dail() command, and features.conf. What really seems to be missing, is simply how do you actually perform the transfer? Blind transfers are pretty simple as you only have two obvious steps. How though do you do attended transfers? 1.) You have a call 2.) You dial *2 or whatever you have

<font size="2">I''m working on a large Rails site and we want to move from RSpec 0.8.2 to the latest and greatest.&nbsp; So we ran the translator and yet we''re having a lot of trouble translating should_render.<br><br>I found this on the web:<br><br>We will NOT be supporting the following in the new syntax:<br>&nbsp;

Does anyone know anything about a 4GB size limit on Samba 3.0.4 running on AIX 5.2 with a 32-bit kernel? We currently have files being transferred from a Windows 2000 server to an AIX machine, and if the files are larger than 4GB, they are getting mangled. Running samba at a high debug level shows the file pointer rewinding or becoming negative once it reaches 4GB and md5sum indicates that the

Hey, I would like in the course of dial plan logic, to trigger a separate outbound call. If that outbound call is answered, and if that certain key response is detected then it will bridge the incoming call to the newly dialed outbound call. What I want to accomplish is that when a caller dials in, they can enter enter an extension that will call out to a callee's cell phone. When the

I'm having a real problem with one of my linksys pap2. On outgoing calls the callee will ring, but caller (pap2) will not here it ring When the callee answers, no audio is transmitted either way. Asterisk reports the call connected and bridged correctly. Now the kicker is that sometimes it works and other times it doesn't. I have had the most luck calling land lines, but sometime

I'm really confused on how to use these two options together: A while back: JustRumours edited this page: http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe and added a little section about dynamic conferences. the 'e' option is repeated all over the page as the savior of dynamic conferences, maybe I'm just dumb, but can someone tell me if a conference is created with the e

Hi all Apparently there is a patch for those 1.2.4/5 MeetMe Freezes: http://bugs.digium.com/view.php?id=5884 Haven't tried it out yet. Benoit Panizzon -- I m p r o W a r e A G - System Services ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz

I have a few weird problems with profiles on my samba PDC. Right now I'm just testing with two XP pro clients. Samba is Samba version 3.0.2a-Debian The problems that I'm having and I believe are related are: 1.) Profiles are saved to the server, but don't migrate to different clients. This is very odd, I can make all sorts of changes to the profile and I can see those changes

Hi Gang To increase security against phished passwords and similar attacks, we consider offering customers to define IP ranges (or GeoIP locations) from which their dynamic registrations are being accepted. I can already look at the source IP in the dial plan, so no issue with validate an INVITE against a source IP. But I would also like to prevent registrations from outside of this

Hi I would like to send a text to the called person when he picks up the phone before the call gets connected through. Is there a way to do this? Example: I'm registered to multiple SIP providers. They come in to a context each and then get through to my phone. Now I would like to send myself an announcement about from which SIP provider this call came from. -- Beno?t Panizzon,

Hi Joshua Thank you for your reply. Indeed, Ubuntu only ships with this old version. Upgraded to 16.2. via PPA. Problem persisted. Well, I already mentioned that this is a machine with two physical interfaces with different routes which on the 'external' side handles SIP customer registrations and has an 'internal' IC Trunk to a commercial Voice Switch via private IP Range. I

Hi all I noticed that most caller are quite confused by the standard voicemail announcement text. Especialy as the number read is the 'internal' number. Callers often hang up because they think having called the wrong number when they hear the announcement. Is there a way (like in many other PBXes) that the VoiceMail user could record his own announcement? (like, hello, this is the

Dear fellow list readers This is the situation: ISDN Devices => Patton ISDN to SIP GW => Asterisk PJSIP The Patton GW resides on a dynamic IP address, so I cannot really use match=ip in the identify section. The Patton does not send a line parameter. The ISDN Devices behind the patton have different MSN and should be able to send them in the From: Header, so the default endpoint

Hi Joshua thank you for the quick reply > Have you checked the Asterisk console when PJSIP is loaded to see if > the endpoint did not load for some reason? Does it show up in "pjsip > show endpoints"? Yes, the endpoint shows up. Endpoint: 11/(scrubbed from mail) Not in use 0 of inf InAuth: 11/11 Aor: 11

Hi Gang Server, two interfaces, routing to two different networks. Two transports defined, each bound to the corresponding ip assigned to the interface. But still, especially when an 183 message is sent, the Contact header does contain the wrong IP Address. Is this a known issue 13.18.3? Or is there a way to make absolutely sure the IP addresses within the Contact header is corresponding to

Hello We had this situation: Some bot-net did try to guess SIP logins and finally succeeded. The Asterisk Server was abused to call a large number of expensive destinations. It is clear that the sip logins have been passed to various persons (probably posted on a forum somewhere inviting to do 'free calls'). Right after the affected password was changed, the message log shows which

Hi, We've been using asterisk as our main telephone-communications platform for years now, and we wrote several extra scripts and features for it. Now we 're looking for a solution to limit the number of channels going to an external SIP provider. We recently upgraded our system from asterisk 1.0 to 1.2(.9 now) to be able to use such features, but nothing helped... When we configure a new

Hi Gang I have stumbled of this problem. I need the P-Asserted-Identity header in an AGI scrip. In the Dial-Plan I do: same => n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity)}) In the AGI I do: my $pai = $AGI->get_variable(PAI); This works fine, unless the PAI contains quotes: P-Asserted-Identity: <sip:1000 at 1.2.3.4:5060;user=phone> I get "<sip:1000 at