similar to: asterisk users

Hi, I configured asterisk on redhat linux 9 box. I installed two different ip softphones (SJPHONE and X-PRO) and got them registered with asterisk. The call from one phone to another does get routed via asterisk, but there is one problem coming up. As soon as call is accepted by the end user , it is automatically disconnected with the error "cannot align media streams". If I enable SIP

a strange problem of multiple sip registrations and peer selection in sip.conf is calling for your suggestions!! let's examine this scenario: some numbers and passwords hidden with HHHs to protect the guilty :) I have 3 distinct sip subscriptions with cordiaip.net provider in US. For each of these i insert in sip.conf (with peer name differences and relefant number/password differences,

Hi all I have an Opensuse 11.1 64 bit acting as a PDC with all this pachages: samba-3.4.5-3.1 bind-9.5.0P2-18.1 dhcp-server-3.1.1-6.3 openldap2-2.4.12-5.3 It was working great for the first 6 months as a logon and file server for around 40 windows computers. But lately from time to time the samba server loses to be the master browser, and the only way to solve this problem is rebooting de entire

Heh heh heh. Whoops. I've spent a few hours digging around in the source and nearly posted a kludge of a patch I'd written to give a kind of half-baked attempt at getting group-usable subscriptions files. Then I came across something in the source........ In the config file, do something like the following: namespace public { separator = / prefix = Public/ location =

Hi All, I know, IAX is NAT friendly, but... I have a problem running gnophone from a box behind NAT firewall. I can register gnophone with * through NAT, but when I try to make a call it instantly disconnects. CLI iax show peers command tells me that peer is unreachable. However this peer is registred. Gnophone also tells me that it is registred. It seems that registration handshake has

http://shorewall.net/pub/shorewall/2.2-Beta/shorewall-2.2.0-RC1 ftp://shorewall.net/pub/shorewall/2.2-Beta/shorewall-2.2.0-RC1 Problems Corrected: 1. The syntax of the add and delete command has been clarified in the help summary produced by /sbin/shorewall. New Features: 1. TCP OpenVPN tunnels are now supported using the ''openvpn'' tunnel type. OpenVPN

Hello, I try configuring Quintum DX gateway as link to PSTN for *. Now, I can dial number which is connect to Quintum, and call is diverted to *. I don't know what I should set, if I want call from SIP_phone registred in Asterisk to PSTN via Quitnum. I set in sip.conf account for Quintum [sip_proxy-out] type=peer outboundproxy=QUINTUM_IP , and changed extensions.conf. When

Hi, Maybe I ask again for something what is wasn''t able to find anything. Is there simple way how to process authorizations? My imagine is that some action for controllers are allowed only to few users. E. g. Everybody can look category list, but only registred user can edit or add. In best way, when user is signup, rails will remember his rights like: adding_category = false

Hi All, i want to make an extension from pbx1 able to tlak to another extension from pbx2 or use pbx2's trunk to dial outside calls. so i edited in both servers accordinally the iax.conf: register => pbx1:pass at 172.16.200.175 <pbx1%3Apass at 172.16.200.175> [pbx2] type=friend host=dynamic trunk=yes sercret=pass context=[default] ; i used the biggest context to avoid confusion as

Help, I'm stuck. Lost in the woods. I have one Asterisk running on FreeBSD outside on the Wild Internet. One on the safe inside, behind a NAT firewall. The inside server registers with IAX to the outer one and can place calls. The outside one can't register to the one on the inside, since it can't be reached on the private network. Now to my problem: * How do I dial from outside to

Hi everyone! I have very simple question, how to limit SIP phone user making calls to for example longdistant calls? Maybe: Put in his context in sip.conf context which don't provide possibility to make such calls? Is it correct? thanks for any help, regards, Corvin

Hi Folks, Commit ac1d23ed7de01fb3a18b340536842a419b504d86 introduces a change in the way CodeGen and MC CommandFlags are handled. It's a change that may impact some devs, so I'd better give a small notice here. Basically previous approach was to bundle all options in a .inc file that declares a bunch of llvm::cl options. This file was lying in include/llvm and was to be included in

Hello, I just registred my e-mail adress and created a topic yesterday. But I couldnt find the reply button. Am I doing something wrong ? Regards.

Hi all, Perhaps someone has dealt with it before. I want to activate a bunch of my own scripts after someone has registred om my asterisk, or when his cient has de-registerded. have been skimming through AGI and AMI, and seen a lot of nice features, but not the (de-)registering events. Kind regards, Hans

Hello, Currenly I've ASterik@Home 1.5 running on DMZ. I can register SJphone there, good audio on 8200 (webmeet me calls) and i also can dial Zapata extensions. When I dial sip phone extensions nothing happens if the client that i'm calling is registred, if the client has voicemail it goes to voicemail. IMPORTANT: I get this error message on my Check Point Firewall: "sip

Hi all, I've 2 * tryning to connect each other Server A is already registred on server B But server B never registers in server A I always get this: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGREJ Timestamp: 00018ms SCall: 00004 DCall: 00003 [XXX.XXX.XXX.XX:4569] CAUSE : Registration Refused CAUSE CODE : 29 Any tip? Best regards, Marco Mouta

Hello, I installed Asterisk in a linonde cloud debian 5, and i'm trying to create a first call but when i try to set up the call i see the following message: -- Called 100 at 100 -- Now forwarding SIP/105-00000008 to 'Local/100 at default' (thanks to SIP/100-00000009) -- Executing [100 at default:1] Dial("Local/100 at default-c2a9;2", "SIP/100 at 100")

Tenho o wine instalado no Fedora 6 e quero rodar o VFP, mas quando executo o VFP aparece "erro ao conectar com SQL. No nodo texto aparece "err:CoGetClass {854d7-bc3d-11d0-b421-00a0c90f9dc4} not registred. O que eu posso fazer? cid:image001.jpg at 01C8625E.FF0BF240 Reginaldo Mendes da Cruz Analista de Suporte e Sistemas Fone:(011) 5503-4206 <mailto:reginaldo at

On 08/08/2023 11:20, Hans Schulze via samba wrote: > Hello, > > i am facing the same problem right now. I start from scratch for an > Samba AD. One question about this: I have registered a domain e.g. > "bla.org" extra/unique for AD, to have an placeholder and there are no > other external Services resolved over this, can i have an fqdn like >

Hello, all How can I write sound file with external G723.1 codec ( actually I have CISCO that can make H323 call to Asterisk box with G723.1 or G729 codec ) I am trying to start Record application by specifying in extensions.conf [writesound] exten => s,1, Answer exten => s,2,Record(soundexample:g723sf) or ...... ( soundexample:g729) I'am using oh323 channel driver, in oh323.conf