similar to: Sip to Sip Calls via Asterisk

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, I'm having problems to register Quintum Tenor DX 3020 on a Asterisk box with SIP. Asterisk always returns "Username/Password mismatch". I've tried all configurations that was on the Quintum's manual, but no success. I've tested the same username and password with a Linksys (PAP2-NA) with the same asterisk box, and it worked fine. Where is the problem ?

Hello, I got Quintum A800 with the P5-2-1 firmware. I configure my asterisk trunk as followed: [SIP_BD1] type=peer qualify=yes host=192.168.0.254 disallow=all context=from-pstn allow=h723 and inside the quantum I change the config sip useragent to 5060. Up to this part if I run sip show peers, I got: asterisk1*CLI> sip show peers Name/username????????????? Host??????????? Dyn Nat ACL

Hi Guys, Hope fully somebody out there will have experienced this and can shed some light on how it was overcome. Current setup includes asterisk 1.6.2.11, GNU GK and a Quintum Tenor CMS on the same lan. Earlier I was unable to make a sip call from the CMS back to a sip client registered on my asterisk box. So I moved onto passing the call from the Quintum CMS to a Quintum Tenore DX which is also

Hi all, Can someone share with me his experience in making asterisk-oh323 talk to quintum gateway without gatekeeper. My set up is QUINTUM GATEWAY ------IP----M ASTERISK (OH323) Both are gateways.. but I don't know what authentication I will set up in oh323.conf and how to set it up I will be glad if anyone can help Goksie

Hello *'s, I have question regarding Quintum Tenor AXT800 VOIP gateway can anyone integrate it with asterisk if anyone what is the scenerio i have scenerio which is quite simple but i am confused about it whether it is possible or not : I integrate it with asterisk for interanet no PSTN at all just only IPphones and analog phones connected on FXS port.Is it's neccassary to cannect with

Hi, We are testing a SIP solution * + ser solution for a large implementation. All the clients are nated. When a client is dialing outside the domain (to a FWD sip account for example) all is perfect ! ;-) But ,when a call is done to a sip account, the client is ringing, then the caller can hear the nated client very well, but the nated client does'nt hear anything. RTP issue no ? I've

Hello, My colleague installed a Asterisk home as company's SIP server and I would like to integrate the Quintum gateway (SIP) but the calls don't get through. Bellow is are the configurations on each side: Quintum ******** Primary Registrar = 202.69.190.244:5060 Primary Registrar User Name= sipquintum Primary Registrar Pwd= sipquintum Primary Proxy =

Hello *'s, I have question regarding Quintum Tenor AXT800 VOIP gateway can anyone integrate it with asterisk if anyone what is the scenerio? i have a scenerio which is quite simple but i am confused about it whether it is possible or not : I integrate it with asterisk for intranet no PSTN at all just only IPphones connected through ehternet port and analog phones connected on FXS port.Is

Hi, I have an asterisk and a quintum AFT200 with two FXO ports, and want to use it as a gateway to handle outgoing and incoming calls. I have found this thread, http://lists.digium.com/pipermail/asterisk-users/2005-February/084015.html But I think I need a little more help, could anyone knows where I can find the basic configuration for this quintum to get it connected to Asterisk using SIP, no

Hello, Maybe some of you tried the SIP support recently introduced by Quintum in their AS devices. I have one Asterisk machine connected to PSTN via E1. It works properly. On the other side I got an ADSL line, with NAT and few devices behind it, like computer with X-Lite client installed or mentioned Quintum device. It works great - calls initiated from there are OK, as well as PSTN originated

Hello all, I post this issue thinking too that could help other people on an asterisk deployment over distributed offices considering both quality, prices, devices and so. Well, i am working on a deployment of a telephony system based in asterisk. My company have a central office with seven remote offices connected all through a VPN. To reduce and evaluate costs i consider solutions like:

Hi, It's kind of off-topic , but still within Asterisk. I developed an asterisk module that send an authentication to a radius server for call authorization and process its reply (limited to User-Name and Cisco or Quintum VSA h323 attribute). My question, is when it make sense to use or include the attribute Password/User-Password? Looking on PDF's of Quintum and Cisco none of it really

Hi All I'm trying with no luck to connected the Quintum D series Gateway with the new SIP release to asterisk. Have anyone done this? If yes then how should I configure the sip.conf to accept the registration? maybe a sample config? Thanks /Krystian

Hi All, Is it possible to use Quintum AFT800 on Asterisk 1.4.29 as Trunk for Analog (like Digium Analog Card) ? And if it's possible, could any one please give me the reference how to configure it on Asterisk 1.4.29. Thanks Regards, Zoel Hairi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Well the subject line probably says it all. I have a Quintum D3000 which I'm supposed to be getting connected up to our Asterisk system. No matter what I try, neither username or authuser config works. I've also tried md5auth and it still refuses to register. Any one have a config they could share with me? Any help would be much appreciated. Neil

Please has anyone on this list had experience with getting Quintum equipment to talk to Asterisk? Specifically a D3000 in my case. It is refusing to register and I'm out of ideas. Any help appreciated. Neil

Hi, just wondering if there is anyone that can help me configure my quintum box to operate with asterisk. i have tried and made numerous attemtps configuring the tenor to work with asterisk@home but have been unlucky. anyone out there has a cheat sheet to configure this device. thanks.. for some reason i cannot get it to work. your help is appreciated.

Hi. I'm trying to configurate Quintum AX to work with Asterisk SIP egister/Proxy server and my problem is that only the first user account get logged in and only that user is able to make call correctly. It seems to be a problem with authorization - I have noticed no "Proxy-Authorization" information in SIP INVITE, ACK, CANCEL messages. I have also noticed that when I remove

I need some help with getting the following to work SipPhone <------> Asterisk <------> H323 GK (quintum) And H323Phone <------> Asterisk <------> H323 GK (quintum) I have tried to run the Asterisk from the newest ports and could after some digging around in the configs register the SipPone to Asterisk and Asterisk to the H323 GK. But when I try to make a call from