similar to: CallerID issue

Scratching my head and trying this. Asterisk Version: Asterisk 1.4.21.2 Tried: exten => 4771,1,ExtenSpy(4724 at mbb) exten => 4771,2,Hangup Also tried: exten => 4771,1,Answer exten => 4771,2,ExtenSpy(4724 at mbb) exten => 4771,3,Hangup Also tried many variations including option ,b I think most calls I make are 'bridged' extensions 4771 and 4724 are both in mbb context.

I'm trying to move the asterisk server to an Amazon Web instance. We have teliax for our sip provider. I'd like for our DID lines to be connected to a users cell phone. Seems simple enough, but I'm getting the dreaded one-way audio, even with nat=yes everyplace I can think of. The dialplan is real easy: [from-teliax-sip] exten => _j.,1,NoOp("From teliax sip with exten

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

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:

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

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

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

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

Greetings! to all Now, with some hard time and help from many genurous people's in the list, I have come to this point with my TDM20B card & my teliax's IAX2 account. I hope someone may help me with this issue mentioned below. I have already selected my codec as gms in my iax.conf as well as in teliax's "my account page" but still i have the same error when I attempt

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

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

I have had good success with my efforts to send faxes over voip using ulaw, surprisingly, and I want to move it from testing to reality. I have an account with Teliax, who have been very good. For voice I use g729 and ulaw, but for faxing I can only allow ulaw. However, Teliax only sets the codec preferences by account. I have another account, but I can't see a way to register two accounts

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...

Asterisk Users, I recently ran into some problems with the quality of service with Teliax. This occurred on August 1, 2007 with a dropped outbound call, audio quality isse on the callee side- not hearing me well on callee side, and sending DTMF tones (configured for RFC2833). Am I the only Teliax customer having this problem? It seems like when I am ready to go live with my Asterisk

Hi, I can't receive/send calls with Asterisk. Could someone please give me a few pointers on my configuration? Regards, Norman Zhang ; sip.conf [general] disallow=all allow=ulaw port=5060 bindaddr=0.0.0.0 externip=x.x.x.x localnet=192.168.22.0 mask=255.255.255.0 context=inbound-sip maxexpirey=180 defaultexpirey=160 tos=reliability srvlookup=yes register =>

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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

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

I use the Teliax service with the IAX2 protocol. I noticed 2 days ago that I was not registered with the Teliax server. I used the "iax2 show registry" command and found I was not registered with Teliax. I issued a reload command in asterisk in order to connect again. I went to the Teliax website and noticed a message which stated clients may need to reboot due to changes made with

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