similar to: Teliax Quality of Service

Has anybody ever integrated Skype with Asterisk? If you have, which software would you recommend to accomplish such a task? ChanSkype? And how reliable are the calls? Did the DTMF tones work? Thanks in advance. _________________________________________________________________ Discover sweet stuff waiting for you at the Messenger Cafe.? Claim your treat today!

Asterisk Users, I am thinking about selecting CALLWITHUS as my sip provider. Has anybody ever used them? How was the call quality? DTMF Tones issues? Thanks in advance. -John _________________________________________________________________ Gear up for Halo? 3 with free downloads and an exclusive offer. http://gethalo3gear.com?ocid=SeptemberWLHalo3_MSNHMTxt_1

Asterisk Users, Has anybody use Voicepulse Connect for Asterisk? I am trying to cover all my bases because in the past, I got burned with poor quality of service, along with failed DTMF tones with 3 different SIP Providers (Vitelity, Broadvoice, and Teliax). I am running Asterisk 1.2.13 on the Debian Etch system, using the SIP protocol. Any insights would be great. Thanks. -John

I am having a problem with DTMF when sending calls through Teliax (SIP). In the peer for teliax I defined dtmfmode=rfc2833 and for the most part it is working. The problem always happens when a user is trying to call a conference system. They simply cannot get into the conference because DTMF is not understood. If I dial from a land line I can get in with no problems. Any tweaks recommended

Not to point the finger at Teliax, but I'm having some unique problems with their service that are as yet unexplained. Incoming calls are fine. Outgoing calls don't work, though they did at one time. As of today, I'm running the latest code from CVS. -- Called teliax/13143212222 -- Call accepted by 208.139.204.245 <http://208.139.204.245> (format ulaw) -- Format for call is

I'm struggling to get my VOIP installation to be acceptable. I'm looking for advice on what else I can look for. My system: o Teliax VOIP service, voip-ny1 proxy o RCN Cable Internet Service (3Mbps download, 500kbps upload, 6ms average jitter) o 3.2 GHZ P4 Server (runs asterisk, firewall, other stuff) o server lightly loaded o Linux kernel 2.6.19.2 o Shorewall Firewall software with

Hi I am looking at purchasing some DID lines from Teliax to install it on my asterisk. i would like to know some feed back on "Teliax" before i purchase. suggest me if there are better sevice providers. thanks Giridhar Bandi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Everyone, I'm planning my first asterisk box, and I'd like to know what SIP providers everyone likes. Voipjet? Gizmo? Somebody else? Thanks, Alex

Good evening all, hope everyone in the US had a nice Thanksgiving! On one of our internal servers, I decided to make the leap from 1.4.2x to 1.6.2.0-rc6 so I could start learning about the changes and new features that have been implemented. I upgraded all the configs, removed all the deprecated stuff, etc -- well went well. However, I noticed after the upgrade, when dialing into an

One might also conclude that during the outage the support people were focusing on getting the system back up and were not near phones. At least that is what I would bet on. Just a thought considering how most of the smaller ITSPs seem to work. Cheers, Wiley -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of

We are unable to call certain 800 numbers through Teliax but I thought I would post this here and see if anyone else had the same problem with either Teliax or other carriers. The 800 numbers causing problems pick-up the call right away and are in the US - American Airlines (8004337300) and Staples (800-378-2753) - we can call many other 800 numbers just fine. Our asterisk setup has a 4-port

We are setting up asterisk with Teliax and having trouble getting the incoming call to work all the time, the outgoing does not seem to have a problem. I have worked with their support but since they say that we are getting the initial call to our server they want to charge to take a look. They did a tcpdump and we are seeing an attempt but no CLI most of the time. Some times we see this but it

We are having issues setting up our Asterisks server with Teliax service. We are able to place calls, but cannot seem to get our Asterisk box to answer from Teliax service. We are using Asterisk with the latest AMP interface. Teliax's examples are for single SIP phone, not the voice response systems, nor do they provide any support of Asterisk other than basic sample scripts...

We are looking for SIP trunks for our * pbx for our business. Being able to port our numbers is an absolute requirement. teliax can do it, but I am unsure of the others. Anyone have experiences (good, bad) with the above mentioned providers to share? Eg reliability, quality, etc. -Dan

I am using Teliax to terminate my calls, and I have 3 licenses' for g729 from Digium. "show translations" verifies that the registration took place. When I place a call, having "allow=g729" as the only allow option in iax.conf, I get the following error: WARNING[361]: chan_iax2.c:6017 socket_read: Call rejected by 208.139.204.228: Unable to negotiate codec If I place a

I really hate to have to make a post like this, but I feel I have little choice but to relay to the group my experience with Teliax, and explain why I recommend against using them as a primary Voip-> PSTN provider. I hope that a letter like this will inspire companies like Teliax to work harder at customer service, as well as circuit stability. We need more companies that offer the types of

We have a Teliax IAX trunk that we use as an overflow for our four regular business lines into our local Asterisk PBX (Trixbox). We have our Teliax account set up so that it goes to a Teliax voicemail box if it cannot reach our Asterisk server, and we have the channel set up for 5 simultaneous connections. Occasionally, calls are sent to the Teliax voicemail box for no apparent reason. In

Has anyone had problems getting their preffered codecs on the Teliax web interface taking effect? I have two accounts, two separate yet similarly configured * servers. On one account the settings took right away - on another server I am getting no result. In fact, no matter what I change the settings to, only the old codecs are usable (otherwise * says it can't negotiate a codec). Teliax

I'm getting really wierd errors from them, like bad packet checksums: Jun 27 18:22:56 NOTICE[29051]: rtp.c:435 ast_rtp_read: RTP: Received packet with bad UDP checksum - a -- "I didn't see it then, but it turned out that getting fired was the best thing that could have ever happened to me. The heaviness of being successful was replaced by the lightness of being a beginner

I'm trying to setup asterisk to accept call from Teliax, request the 10 digit number from user, then dial it thru the VoIPJet. If I'm not wrong I will be charged by both providers because both connection is active during conversation. So my question is can I set the things so that I pay only to VoIPJet? Specific configuration snippets will be greatly appeciated. Thank you.