similar to: Asterisk <------- Quintum SIP Registration

Hi All. I'm using RedHat 9 I configured the chan_h323 and asterisk from CVS. This is the scenario SJ_lab_phone(sip) ---------------> Asterisk -------------> H323 GK --------------> PSTN I have tried all codec's and always the same result, the called phone will ring without dropping for how ever I allow it to but as soon as it is answered it immediately gets disconnected.

I have compiled chan_oh323 and when starting * I get the following. [chan_oh323.so]Aug 15 12:40:00 WARNING[1076245120]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_oh323.so: undefined symbol: __use_ast_pthread_create_instead__ Aug 15 12:40:00 WARNING[1076245120]: loader.c:423 load_modules: Loading module chan_oh323.so failed! Can anyone tell me how to fix this, or what

Colleagues- I am working on the design of a fairly large samba deployment, and I am looking for feedback on some of my design ideas. I have 10 buildings spread out in and around a city, all interconnected via 1.5Mb leased lines. There are samba servers in each building. I have some users that move from building to building. We are using primarily windows 98 desktops, with a few 2K and XPP

Hi all I have asked this question before but have not got any helping input. I'm really new to this and need some explanation about ASTCC. So here is the question again. In the ASTCC web admin there are Trunks, Routes, IAXFriends, SIPFriends, Brands, Cards. As I understand Brands is not used, Cards just makes the cards. Routed in the dialplan and pricelist, Trunks is for ASTCC to

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 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

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 *'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

My fellows, We have Asterisk@home installed and we want to interconnect it with our existing quintum gateways.. any idea how to config that? Your time is very much appreciated.. Cheers, Jessie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050220/2518797c/attachment.htm

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

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

Let's try this again. I have a Quintum AX Tenor gateway sending calls to Asterisk from BT analogue lines connected to FXO. The agents hear an echo on their side but incoming callers hear the conversation fine. I can't seem to find the problem. Anyone seen this issue before? <p style="margin: 0; padding: 0; border-collapse: collapse; font-family: Tahoma, Arial, Sans-Serif;

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

Hi Couple months ago I started to have a strange problem with HTB. My setup is Fedora Core 2 + Pentium 2 233 + 128 MB of ram and its serving as a router. For some time since going to kernel 2.6 my HTB QoS Stalls for couple seconds, every couple minutes. If the connection load is bigger the stalling is more frequent and takes longer. I isolated the problem to be with HTB (CBQ works fine). The

Hi Is there a program which allow me to see how "my" traffic goes through my iptables rules? Which accept it, which deny? Right now my router has a little bit of traffic and its hard to see only mine traffic. -- Miłego Dnia Krystian Antoni _______________________________________________ LARTC mailing list LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/cgi-bin/mailman/listinfo/lartc

https://bugzilla.samba.org/show_bug.cgi?id=6573 Summary: Documentation Error: Section with exact title doesn't exist Product: rsync Version: 3.0.6 Platform: Other URL: http://samba.anu.edu.au/ftp/rsync/rsyncd.conf.html OS/Version: All Status: NEW Severity: normal Priority: P3

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