similar to: VoIP QoS monitoring and failover re-routing

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 *'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, 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 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

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 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, 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.

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 all, Im trying to build an small home system. I have 2 pots lines, and i need to make 8 extensions and be able to use my old analog phones. What would you recommend to use as the 8 FXS switch? I saw some equipment from quintum, they have a Tenor AS that offer 4 FXS ports. But i don't know if it is the best solution. Does anyone have a better solution to build this system? If an analog

Hi all, I am planning to use VoIP gateways to connect remote offices to Asterisk. Not having much experience with these and Asterisk I would appreciate any info/experience that you might share with me as to their interoperability with Asterisk. I am interested with in rather bigger gateways (order of E1's) from: AudioCodes - Mediant Mediatrix - 1531 Quintum tenor Multupath D3000 Has anyone

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, maybe someone out there already has some experience and can help me. I have just ordered an E100P card from Digium, I already have a basic asterisk setup up & running. My application is the following : I want to accept incoming calls from the PSTN to Asterisk, and without asking anything of the client just pass them immediately to a call gateway in USA, actually we are planning to use

Hi, I don't seem to be able to get music on hold to play normally. The sound gets often interrupted with a few seconds of silence then starts playing again. I'm using mpg123-0.59r and tried mp3 files with different sample rates with no luck. If that matters, endpoints are SIP ata186, SIP Cisco 7960 and H.323 (over chan_h323) Quintum Tenor. Sometimes it may play fine for a few minutes

I'm wondering if anyone knows of ISPs with service that has QoS features that would be good to use with VoIP stuff. Granted, the QoS would only be supported as long as you stayed within their network, but it might be better than nothing. --Eric

Hi! I have some problem in my asterisk 1.4.2, I've installed it on centOS 5.2 and this didn't accept voip QoS and can't route the packets having voip QoS. So I should change voip packets to be routing with centOS. I want to use iproute2 but i don't what to do after installing iproute2. Anyone could help me please? -------------- next part -------------- An HTML attachment was

I have currently following the instructions at- http://www.voip-info.org/wiki/view/QoS+Cisco I am using a Cisco 877 router with an ADSL2+ connection. My upstream bandwidth is around 1Mbps, and my downstream around 4-5Mbps. I have two PSTN side interfaces - atm0 and dialer0 (dialer0 is where the ISP details are configured and the interface that the WAN side IP is applied on) Now which

Hi i want see if my QoS are good because i am not very sure ... the VoIP quality are not very good when i download. I have on my Linux routeur/Firewall Asterisk .. and i have into my config : ================================================ tcdevices: eth0 2000kbit 2000kbit tcclasses: eth0 1 100kbit 180kbit 1 tos=0x68/0xfc,tos=0xb8/0xfc eth0 2 full/4 full

Hi, I'm looking for opinions on the "best value" router to use for home offices. It should work for a scenario in which there are 3 computers and 2 SIP phones, handling QoS so that the phones always have higher priority traffic than the PCs. (and not rely on the phones to do the QoS because some PCs may not be connected to the phones). QoS could be based on destination and source

Hello, I'm searching for a router for our company. Does anybody has a suggestion for a router with a SIP Application Layer Gateway and good working QoS (Upstream AND Downstream). Regards Bastian

I thought this might interest a few asterisk users. I don't use them so I have no idea about Adtran's but I know a heap of people on this list swear by them. Cheers, Dean ________________________________ From: vclass.coordinator@adtran.com [mailto:vclass.coordinator_adtran.com@mail.vresp.com] Sent: Monday, 19 September 2005 9:59 AM Subject: ADTRAN Virtual Classes: Ensuring