similar to: IAX user register problem

hi, i want to call PC2PC between to IAX client without authentication i want to allow every user to use PC2PC no any password required. Please let me know what i have need to do in IAX.conf or any other file to allow any user to call Pc2Pc. My IAX.conf [guest] type=user context=default callerid="Guest IAX User" My extensions.conf [default]

Dear Sanjay, Sorry sanjay i miss to explain completely. My PC2PC mean is Dialer2Dialer i want to allow call between Dialer with out any registry and authentication through IAX. so i need to setup Asterisk accept calls from any user and users can call to each other without any password and registration. please help how can i configure Asterisk using IAX in this regards. thanks, Asif Message: 9

First I like to apologize if this is common knowledge, but I'm unable to get MSN messenger 4.6 to register with asterisk. I configured MSN messenger to use UDP and the IP of my asterisk server I edited the registry entry - for pC2PC calls under Windows98. What I'm I missing ? Asterisk version information Asterisk CVS-04/25/03-05:37:19 sip.conf [pingtel] type=friend

I have some stuck channels that I think I'm going to have to bounce Asterisk to get rid of, but am curious to know what they are and how they've managed to accumulate. The show up with a channel identifier of '(None)' as in the output below, and do not show up in the soft hangup list, and so can't be cleared by that method. Here is the output from iax2 show channels:

The situation is one of my asterisk servers is behind a NAT firewall and one is not. Both servers have multiple IAX peers. The NAT firewall has port 4569 mapped through to the asterisk server behind. But, the natted server is almost permanently unreachable from this non-natted server, even though, the non-natted server is almost permanently _reachable_ from the natted server. Details are below

To answer my own message, I figured out a solution (untested) about 10 minutes after posting and leaving the office. Doh! Anyway, the solution (now tested) was to make the Asterisk server behind the NAT register with its peers. Despite reserving port 4569 in the firewall, that was not enough in this particular NAT firewall - it was only being reserved for one connection. Kind regards, Sebastian

Hi all, Some time ago I posted an issue regarding the hangup of active calls from the CLI and someone told me that "soft hangup" should work. Well, in fact it does work, but only if the channel is known, i.e. it doesn't work for zombie channels. For example, I have this scenario (CLI output of command "iax2 show channels") IP-AM-PBX*CLI> iax2 show channels Channel

I'm using the the Microsoft Real-Time Communications Client API SDK using Visual Studio 6 and . NET 2003 SE to make SIP calls. Using the examples provided I can make unregistered SIP calls fine, however I am having trouble registering with Asterisk. I have to produce an XML Profile to use when registering with a registrar. The one I use is... <provision

http://bugs.freedesktop.org/show_bug.cgi?id=15233 Summary: geforce 7800gs and AGP 3.0 DBI function Product: xorg Version: unspecified Platform: Other OS/Version: All Status: NEW Severity: normal Priority: medium Component: Driver/nouveau AssignedTo: nouveau at lists.freedesktop.org

I seem to be having a problem that I have narrowed down to a disagreement on codec negotiation or codec setup of some kind in an IAX peering arrangement. Here's a non-ASCII art version of the setup: DID origination provider via SIP/gsm to Call routing asterisk server via IAX/gsm to Client asterisk server via SIP/ulaw to Polycom 501 UA The problem that occurs

Hi, I've tagged the 8.0.1-rc4 release, please begin testing. This will (hopefully) be the last release candidate. If all goes well, I will tag the final release next Wednesday. -Tom

I am having a problem when a SIP registration fails. I get the following messages in the log: Sep 25 18:51:02 NOTICE[1125329600]: File chan_sip.c, Line 2874 (sip_reg_timeout): Registration for '<user>@fwd.pulver.com@65.39.205.114' timed out, trying again Sep 25 18:51:02 NOTICE[1125329600]: File chan_sip.c, Line 5119 (handle_request): Registration from

Hi, The 8.0.1 final release has been tagged. Testers please upload the final binaries. -Tom

Hi! I'm trying an asterisk-FWD connection. I'm using X-Lite OR SIPPS as the IP phone. I configured the X-Lite and SIPPS to use GSM codec. Whe I call FWD, I get this info on the channels when the call has not been stablished yet: sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 192.246.69.223 613 1770bf3430d 00102/00000

I know I'm getting tired of looking at code, but why is the function register_verify defined in 2 different files? chan_iax2.c line 3860 static int register_verify(int callno, struct sockaddr_in *sin, struct iax_ies *ies) chan_sip.c line 4869 /*--- register_verify: Verify registration of user */ static int register_verify(struct sip_pvt *p, struct sockaddr_in *sin, struct sip_request *req,

Hi list, on my previous (lenny or squeeze) machine, I had it up and running, but that machine and configs are gone - and on my new jessie machine (see device and config details below) I cannot start the driver: # /lib/nut/bestups -DDDDD -a Patriot_Pro_II_750 0.000000 debug level is '5' 0.000806 send_to_all: SETINFO device.type "ups" 0.000842 send_to_all:

Hi all, Below is what I did to run Asterisk in pass-thru mode: sip.conf: [general] disallow=all allow=ulaw canreinvite=yes For each channel, canreinvite=yes is enabled. No dial command has 't' option. However, it seems that Asterisk still stay in the media path and bridge the 2 end points. Am I missing something??? sip*CLI> show channels Channel (Context Extension

Hi, I'm slowly getting to grips with *. My next quest is to get IAX2/FWD calls working. I've setup a fwd account and added IAX capability to it via the website. I got the email saying it had been done with some welcome text and sample phone numbers to try, such as 10001 for the answer phone. I followed the setup example on the fwd site for configuring * to work with fwd's IAX.

Has anyone seen this before? This channel is consistently present on both of my asterisk servers. Sometimes they disappear for a few seconds and then come back. It always has the same Call ID. voip1*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 192.168.0.102 (None) df92fb1b-8a 00101/03059 00000ms 0000ms UNKN

I'm experiencing one-way voice paths, followed by a hangup on one softphoine and not the other. Also, caller ID has odd outputs -- and I wonder if the problems are related. My configuration has Asterisk and a Linphone softphone running on the same box. I have a PC, and on that PC I use X-Lite or SJPhone to connect to the Linphone instance. When I call from the PC to Linphone: * I call