similar to: Urgent help with Sip <------> H323 on FREEBSD

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

I have build asterisk from latest CVS HEAD-05/09/05 with H323 support as described in README file. Open H.323 version v1.17.1 and PWLib v1.9.0 on Mandrake Linux 10.2 kernel-2.6.11 I tested it with following phones: -- XLite (SIP softphone) -- QMix SIP IP phone (PA168F) -- SJPhone (H323 softphone) -- QMix H323 IP phone (PA168F) -- FireFly (IAX2 softphone) Everything works fine except a problem

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.

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

I just upgrade quintum A800 with new SIP firmware ---------- Product Name: Tenor Analog A800 Multipath Switch - 8 ports (Rev. B) Gatekeeper Status: Mini GK Calls Allowed: 8 Feature Bit Status: -PS/+RB/-ER Languages allowed: 1 Serial Number: A002-00308F Ethernet Address: 00-30-E1-00-30-8F IP Address: 10.101.0.10 Subnet Mask: 255.255.255.0 Default Gateway: 10.101.0.1 System Software Version:

Hi, I am using asterisk CVS 2004-06-16 with oh323-0.6.3a I have a strange problem if I start asterisk with oh323 loaded /usr/sbin/asterisk -vvvvvc once I am in the console and issue "restart now" or "reload" asterisk hangs and it not stoping or restarting at all, below is the console logging when it happens, as you can see it stucks on "Destroying any remaining

Hi, Could someone help me on configuring a H.323 trunk. I am trying to set up the following scenario: [SIPphone(2004)]--[asterisk/oh323/asterisk-oh323]--H323Trunk--[PBX]--[H323phone/(8004)] I am using the following versions: Linux CentOS 3.3/2.4.21-.EL.co asterisk 1.0.1 pwlib_1.5.2 openh323_1.12.2 asterisk-oh323-0.6.3b Calling from Asterisk (2004) to the

After much reading and fiddling - I have the gnugk GateKeeper running and can make calls from the H323 phone to the sip phone. Voice works bi-directionally.. Calling from SIP to H323 gives me a problem... Both gnuGK and Asterisk are on the same box. Someone said this was OK. Others said No. I added a second IP (eth0:1) and told gnuGK that was HOME. How do I lock asterisk to the other (eth0) IP -

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, I have a weird problem. I have asterisk setup using the G729 Codec to receive Incoming calls both from a SIP Gateway (SER and Quintum) and via ISDN using i4l and have rules setup in extensions.conf for sending calls out either back via the SIP Gateway or ISDN. What I want to do is have PSTN calls come in via the SIP Gateway, be answered by the auto-attendant and then sent back out to

Hi i'm trying to connect to the PSTN in the following way sip ATA -> * -> gnugk -> Cisco AS5300 -> PSTN I'm using asterisk CVS-HEAD-06/01/05-14:33:15 running over RH EL3 Asterisk-Oh323 0.7.2 pre1 Open H323 v1.13.5 pwlib v1.6.6 and I'm having a lot of trouble, gnugk and * both have public ips and are not behind any type of firewall, the sip ATA is behind a firewall and

Hello, I just tried to compile OH323 0.6.3b on a RH9 machine with Asterisk 1.0 RC2 installed but failed. I applied the patch to the required OpenH323 library according to the instructions, and set the proper directories in the Makefile. Here is what I receive after I issue make: ******************************* g++ -DP_USE_PRAGMA -fno-rtti -ffunction-sections -fdata-sections -D_REENTRANT -

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 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