similar to: iax2_read: I should never be called

Hi All, I'm having a problem after reinstalling the operating system. Festival works fine for SIP, but when IAX users are calling the same extension they don't hear the festival and I see the next message on console: NOTICE[3996]: chan_iax2.c:2995 iax2_read: I should never be called! I googled and couldn't find a solution, if somebody can help.... neobase*CLI>

Hello ! I 'm having a machine running asterisk 1.6.0.10 with IAX and dahdi. The calls are going in and out from IAX2 to dahdi (chan_dahdi + libpri) and vice versa. After a period of time, I got the following scenario: NOTICE[860] chan_iax2.c: I should never be called! WARNING[752] channel.c: Exception flag set on 'IAX2/iax-peer-13262', but no exception handler WARNING[752] channel.c:

We upgraded our asterisk server to 1.2.18 last night to pick up the security update. Since then, any calls coming in on IAX2 links get dropped if they try to enter a MeetMe conference room. The log shows this: Apr 26 08:33:16 NOTICE[27362]: chan_iax2.c:3167 iax2_read: I should never be called! Hanging up. I've temporarily worked around it by switching our inbound provider to use SIP

I attempted to install an X100P card but it was not correctly recognized by my Redhat 9 install. I had a test install running without any cards which was working great minus the outward dialing since no cards existed. Now that I have a card, I want to add it to the system. Do I have to scratch the whole current install in order to get the X100P running on my system or is there a way to get it

Is any one else having problems with Voicepulse today? Suddenly, I can't register and calls to my Voicepulse numbers get a fast busy. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815

I downloaded the astcc calling card program. Thanks, it is very easy to setup and works Excellent. Anyway, it says to use DeadAGI to run it rather than AGI. I don't know what I am doing wrong. I just updated my asterisk from cvs and rebuilt and reinstalled. I do not have an application called DeadAGI. I have searched the source, google, etc. but have not been able to find anything.

I am recieving inbound calls to my asterisk box over IAX. And getting "spawn extension....exited non-zero" errors. Im not entirely sure what this means, and would appreciate any help in fixing my problem. FYI: ********** is the inbound phone number x.x.x.x is a remote asterisk box calling my own asterisk box. When I choose it to dial an internal extension using this dialplan: exten

Does asterisk support using an H.323 provider for outgoing calls? From everything I have found, it looks like it does. However, I have had no success in getting it to work. I would really appreciate if somebody could give me a hand. I am using the channel that comes with asterisk. I have also tried using the channel from inaccessnetoworks but have not had any more success. My provider

Good day to all-- First off let me say that I have been very pleased with the mailing list. I have learned a ton of stuff just reading other peoples questions and comments. I really enjoyed the VOIP Conference call on Friday morning. Still working on figuring out the best approach to custom voicemail emails (the reason I joined this group); however, we have more pressing issues. I

I have been picking at Asterisk for about a week, and I think I'm close. I was hoping for a little guidance to bring this on home. I want to be able to make outgoing calls from my SJPhone clients using my VoicePulse Connect account. I have the two requisite items from Voice Pulse, but I've had no luck successfully integrating the VoicePulse settings into iax.conf. My current config:

I've got a number (10) Cisco 7960's connected to my network. All the phones work perfectly except one. The asterisk console keeps throwing up: Jun 22 15:39:15 NOTICE[-1147470928]: chan_sip.c:5887 sip_poke_noanswer: Peer '4001' is now UNREACHABLE! Jun 22 15:39:27 NOTICE[-1147470928]: chan_sip.c:4925 handle_response: Peer '4001' is now REACHABLE! Jun 22 15:42:08

how many calls can be handled with Asterisk Calling Card app? somebody knows that? --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.744 / Virus Database: 496 - Release Date: 2004-08-24 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041004/498fd5f8/attachment.htm

hi folks. we recently deployed 10 Cisco 7960G w/ SIP firmware 7.3 on our network and we start having problems of dropping calls (actually the calls wasn't dropped it just the sound was muted for about 5-10 seconds, but most users will think the call dropped and hangup/redial). i've check the console output. there was a lot of messages like the following: Sep 28 15:00:49 NOTICE[8182]:

Hello, I am logging directly into file and also to syslog. Here is snippet from my /etc/asterisk/logger.conf: messages => notice,warning,error,verbose syslog.local0 => notice,warning,error,verbose But the logs look different: VERBOSE[7609][C-00000013] pbx.c: NOTICE[3042] chan_sip.c: Peer '1111' is now UNREACHABLE! vs. VERBOSE[7609][C-00000013]: pbx.c:2925 in

In response to the postings by Andrew Kohlsmith and Ernest W. Lessenger: Andrew, I modified the exten line in extensions.conf as you suggested. Unfortunately, It still does not work... Ernest, I spent approx. 4 hours reading list archives (and anything else Google served up) on how to configure iax.conf and extensions.conf to work with Voicepulse. Then, I sent an email to voicepulse

Maybe someone could point in the right direction. I have a small facility that's running around 40 Polycom 301/501 phones, Asterisk 1.4.18 running under Mandriva 2007.1. The phones were assigned a DHCP address in the 10.10.10.x range. Today, the DHCP server failed and to get them back online, I loaded the dhcp-server onto another system (Also running Mandriva) and copied the dhcpd.conf

Hello, Currently we are using a new network with an old samba server. Previously we were using a standard hub based network. Now we have Cisco 3516 switches and a new configuration of V-lan masking. In our old non v-lan segmented network, I had no trouble setting up a samba server. Currently I can browse the samba server, but I can not attach to any shares. I have updated our smb.conf file

Hello; I've been playing with a nifty Open Source java based report writer called Datavision (datavision.sourceforge.net) and I've managed to write enough logic to calculate phone bills at different rates from the MySQL cdr's. (cdr_addon_mysql) Eventually I want to have sets of rate structures for each user of the system - so I can bill client A at 3 cents a minute and client B at 2

I'm having trouble setting up tls/srtp secure communications on my Asterisk server- I'm still rather new at working with Asterisk. I have enabled tls and encryption and I have csipsimple with tls build on the phone. I'm currently only testing one phone with this capability so far, and the rest still work in the current state. My logging looks like this with verbose turned up:

Hi list! I have a problem and I hope someone can help me... I configured an Asterisk on a VM to serve more accounts and act as a proxy to other SIP-providers. The first account running on my phone works without any problem. A second account, running on the phone of my wife, is always UNREACHABLE. I can just see in the log: [May 28 21:48:46] NOTICE[3646]: chan_sip.c:22933 sip_poke_noanswer: Peer