similar to: Problems with * and IAXTel/FWD

I've been looking at how our budgetone's have been failing and have found the following: A quick layout -- Latest CVS as of tonight. Sip phone behind NAT. * server with public IP address. -------from sip.conf for my phone: [1747xxxxxxx] username=xxxxx secret=xxxxx host=dynamic type=friend nat=yes ------- -------from the * log messages Nov 6 01:50:07 DEBUG[4101]: File chan_sip.c,

[ Sorry, I incorrectly copied some Reference headers into this post and tacked it onto the wrong thread. -Steve ] So far, I have been able to receive incoming iaxtel calls via my assigned 1-700-xxx-xxxx number, but only when using md5 authentication in iax.conf: [iaxtel] type=user ; Incoming calls only context=incoming auth=md5 secret=<mysecret> ; Required for

Ok folks I have another question. So far I have gotten my IAXTEL number and I have been able to make calls from my asterisk system to any IAXTEL number and even to FWD numbers. I also got FWD to work and I now can get calls to my main system. It's great when these things work. But when I call my own IAXTEL number 17005441100 all I get is a message saying the user is un registered or un

On Sun, Aug 17, 2003 at 03:44:21AM -0500, Gnophone Support wrote: > Hello, and thank you for registering at gnophone.com. Your login > information is listed below: > > Username: miernik > Password: ******* > IAX Phone Number: 17002916107 > > Please login as soon as possible to > http://x.linux-support.net/directory/ to complete the

Hello I am still having problems with IAXTELL and FWD configuration. I get the following when I dial 17009965342 which is set as an example to dial to FWD people. 1+700+99+ 5 digit number. I have placed XXXXX where my passwords are. CLI> Executing Dial("Zap/14-1", "IAX/abatista:xxxxxx@iaxtel.com/917009965342@iaxtel") in new stack -- Calling using options

(I have some problems with my mailing-list alias, I hope this doesn't get sent twice) On Sat, Nov 15, 2003 at 04:35:20PM +0100, Philipp von Klitzing wrote: Thank you for your comments Philipp: > > - with a SIP phone configured as 192.168.1.190, and with its SIP > > server being 192.168.1.190 > > That doesn't look right. Do you have another "SIP

Hi, my setup is quite simple: an asterix CVS of 2003-11-15 on a 2.4.21-debian-5 GNU/Linux box in an internal network (192.168.1.0/24, asterisk is 192.168.1.10). - with a SIP phone configured as 192.168.1.190, and with its SIP server being 192.168.1.190 - with an ISDN AVM c4 i4l card on an ISDN connection with 2 channels. I try to: - dial-in from ISDN, then transfer to the SIP

I'll admit it. I'm a asterisk newbie (but no stranger to telephony). The setup is simple: two Grandstream BudgeTel 100 phones (SIPPhone specials) on a private segment calling to a Linux box acting as the segment's firewall with a leg on our public network. The phones are setup as SIP/phone1 (x1000) and SIP/phone2 (x1001), respectively (thanks to the Asterisk HOWTO). Getting IAX

Has anybody had a problem registering their IAXtel account? I just signed up for an account and followed the documentation on iaxtel.org and my registration is always rejected. When I type "iax show registry", I get the following output: Host Username Perceived Refresh State 12.37.165.130:5036 xxxxxxxx <Unregistered> 60 Rejected

Hi, Some calls I make trough my PSTN asterisk gateway just hangup after some minutes. Even if I'm using sip or iax. I have callprogress=no busydetect=no in my zapata.conf. Anyone help? Or tell me what to look at /var/log/asterisk/debug. I didn't find anything wrong. [endpoint]---iax or sip----[asterisk]----E&M----PSTN. As endpoint I had tested another asterisk box (with a FXS),

Hi, Every time I dial into my asterisk box i hear nothing but asterisk jittering. The following is an example of what I get on the asterisk CLI Thanks *CLI> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user

I'm new to the Asterisk software but have successfully set it up to make and receive calls using FXO cards, voicemail transfer etc. I can successfully call the Digium test IAX using the examples provided. I have signed up for an IAX tel account and got a number. The extensions have been set up as per the examples from IAX tel. However when I try to place a call this is what I get: --

Hi all, I've been running Asterisk with a TDM400P for about 6months, no problems. 2 in/outgoing analog lines, one analog phone. Recently I was messing with the XTEN client, got to finagling with things, and not knowing what was wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was testing various things, and found everything worked except outgoing calls. So I checked

Is anyone other than me having trouble dialing out via IAXTEL? I havn't changed my config files in weeks but seems that IAXTel calls (800 and FWD) stopped working in the past week sometime. Robert

Hello, I have Asterisk running on a RH9 box; Everything seems to be working as it should, except for Festival. Every time that Festival is called from Asterisk, Asterisk silently shuts down. Festival doesn't give any error indication and Asterisk just plain dies without a peep. Festival was installed per the Wiki, using source and patched with festival-1.4.3-diff; it works fine at the

<BLOCKQUOTE style="PADDING-LEFT: 8px; MARGIN-LEFT: 8px; BORDER-LEFT: blue 2px solid"> <DIV>Hello,</DIV> <DIV>&nbsp;can someone tell me what's wrong with this? I can't make toll free calls via iaxtel. Here's the definition in my extensions.conf</DIV> <DIV>&nbsp;</DIV><FONT size=2> <DIV>[iaxtel-trunks]</DIV>

Last night Mark and I made some changes to the IAXTEL tollfree outbound, and inbound access. The inbound access number has changed to: 248-724-0700. (This number is in Pontiac, MI Ratecenter, and is supplied by Telesthetic LLC, a next gen phone compnay) This number will say "Please dial your number now" at that point you can dial your 1-700-XXX-XXXX IAXTEL number assigned. In the

I have tried for over a month off and on to get iaxtel for inbound to work... and tonight after alot of troubleshooting we noticed this: iaxtel inbound will use the last entry in your iax.conf to auth against. So if [iaxtel] is at the top and say [voicepulse] at the bottom. An inbound call will try to auth against that [voicepulse] entry even with the [iaxtel] entry at the top of the file. Has

When I try to call iaxtel it goes to codec ADPCM even though I have define in iax.conf gsm Call accepted by 69.73.19.178 (format ADPCM) -- Format for call is ADPCM My settings: [general] port=4569 register => xxxx:xxxx@iaxtel.com bandwidth=high jitterbuffer=no tos=lowdelay [voipjet] type=peer host= xxx.xxx.xxx.xx secret= xxx auth=md5 notransfer=yes context=incoming disallow=all ;

I signed up for an IAXTEL account and have been trying, unsuccessfully, to get it working. In IAX.CONF I have: [iaxtel_out] type=peer host=iaxtel.com username=USERNAME secret=SECRET auth=rsa inkeys=iaxtel [iaxtel] type=friend context=incoming host=iaxtel.com auth=rsa inkeys=iaxtel However, when I start Asterisk, I get the following warning: [chan_iax2.so] => (Inter Asterisk eXchange