similar to: IAX2->SIP dropped calls

Hi, I'm having a bit of a problem with storing voicemail messages in an odbc database. I *think* I've got everything configured correctly but messages are stored on the asterisk server instread of in the database. System info 64 bit redhat RHEL 5.1 Asterisk 1.4.26 unixODBC installed used makemenuselect to instal res_odbc and cdr_odbc Back end database DB2 Database name voiceml

hi, everyone, i have been sufferred for the asterisk hang on problem for so long and i just reinstalled the whole thing yesterday, but again this morning the server hangs on again, you could not call in through PSTN line and the ppl also could not call out throught the server, there is simply engaged dial tone when you try to do so. and the only thing i can do is to restart asterisk server after

Hi all, I have recently installed Asterisk@home and outbound calling is working great. However I am strugglings with inbound calls. I have setup a trunk for my provider, voipfone and in the inbound area on AMP I have the following :- user context name = 3011XXXX context=from-pstn dtmfmode=rfc2283 fromdomain=voipfone.co.uk host=voipfone.co.uk insecure=very secret=XXXXXX type=user user=3011XXXX

Hi, I'm currently running CVS-HEAD 2005-09-03 I do plan to upgrade to the newest version, but need to do some testing with it first. In the mean time... does anyone know what these messages below are about? I've never seen it before, but when it happened it locked Asterisk up pretty good. Feb 10 10:16:51 DEBUG[14917] chan_zap.c: Echo cancellation already on Feb 10 10:16:57

I have a situation when I dial out my Zaptel I am getting a recording that I need to add a 1 or a 0 and the area code with this number. I have tried appending this and the number going out the zap is 1NXXNXXXXXX so it is going out with 1 and the area code. Someone has suggested that maybe the zaptel is dialing too fast. My question is how can I add a pause before dialing to test this out. I am

I know in advance maybe I'm overlooking something stupid, however I'm really lost and cannot find the solution... Situation: - asterisk-1.2.13 on a linux box with no iptables active. - one iax2 peer defined - one wildiax phone running on my laptop the soft phone is configured to connect & register on asterisk, however, I cannot get it working. What am I missing? Please help!!

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 am using Realtime on Asterisk 11.5 with a SQLite3 backend. While everything seems to be working fine I keep getting this error on my log files: [2014-02-17 19:55:18] WARNING[20569] res_config_sqlite3.c: Could not execute 'UPDATE "sip_buddies" SET "ipaddr" = '192.168.2.23', "port" = '5060', "regseconds" = '1392692118',

Hi guys, I'm using asterisk 1.2.1 since a week ago or so. today I found it crashed when making a call through teliax. This is how it looks: -- Called xxxxxxxxx@teliax/17075471770 -- Call accepted by 208.139.204.245 (format ulaw) -- Format for call is ulaw Jan 16 17:53:56 WARNING[5901]: chan_iax2.c:7535 socket_read: Received mini frame before first full voice frame Jan 16

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

Hello everyone. I can place outgoing calls no problem with my IP500 (using teliax as our provider). Thing is, when a call comes in, 90% of the time when I pick up the handset it drops the call immediately. I turned on SIP debug, and have listed my extension config from sip.conf. Any help is greatly appreciated.... sooo close.... TIA! -Ron [3004] type=friend username=3004 password=XXX

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 have several Asterisk servers (1.6.2.15) all configured for Realtime, all backed by the same database. The Asterisk servers are all listed under DNS SRV records, and SIP ATAs find us this way. Normally, no matter which Asterisk server an ATA connects to, we get our database fields filled out correctly, such as "regseconds", "lastms", "ipadr", etc. However, with

Hi, I am having a difficulty with getting two realtime user?s to bridge on answer. I have managed successfully to bridge the same two users/channels via the Bridge Manager api command and confirm that the two communicate directly bypassing the asterisk server (I confirmed this with Wireshark). Does anyone have some ideas? I have put some log entries below. I haven?t attached my

I am having some problems getting TelIax service to work with *. Outbound calls work just fine. When I try an inbound call the phone rings and there is no audio. Upon further investigation "iax2 show channels" indicates that the codec is "unknown" The provider confirmed that they are set for ulaw and so am I. Does anyone have an idea what could be causing the codecs to

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 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 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 am trying to use ChanIsAvail to detect the best route for a call. I am testing by dialing an extension that is then forwarded to the DID. Normally it will be an incoming PSTN call that is forwarded. When I try it, I get put on hold for a few seconds and miss the beginning of the recorded message. Any ideas what is going on? -- Executing ChanIsAvail("SIP/501-304d",

This is probably a good time to point out that there is a good litmus test for all Voip providers. PRIOR to purchasing anything, send them an email and request the sales information. Ask about their servers or their policies or anything you can think of. How they respond will tell you a lot. If it takes forever, you can tell that they are either really busy, really indifferent, or something in