similar to: IAX2 order

Folks, I've scoured the website and googled, but can't find a definitive answer: What's the difference? OK, the site says 1.4.26.1 is latest stable. Site also says 1.6.2.0-beta4 is latest beta. So what are the others? TIA, David A. Bandel -- Focus on the dream, not the competition. - Nemesis Air Racing Team motto Visit my blog at:

Folks, I know this must be a configuration problem. Just changed servers last nite -- an interim server running 1.6.1.6. Copied all of /etc/asterisk to the new server and fired it up. Now I'm getting: [Oct 14 12:28:10] ERROR[8471]: cdr_pgsql.c:309 pgsql_log: Failed to insert call detail record into database! [Oct 14 12:28:10] ERROR[8471]: cdr_pgsql.c:310 pgsql_log: Reason: ERROR: syntax

Folks, Not sure what's going on, but suddenly Asterisk 1.6.1.6 is crashing, usually when I exit the console or use asterisk -rx. The sip peers entry always shows duplicate entries (once I had an extension over half a dozen times) just before it crashes. 3182/3182 172.17.0.126 D N 5060 OK (14 ms) 3183/3183 172.17.0.128 D N 5060

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Folks, I looked through the archives, but didn't see anything like this. I have a network with a mix of Win95, Win98, and WinME clients (and one Linux system w/ Samba server version 2.2.2). All systems belong to a workgroup called "Oteima". All Win95 and Win98 clients see all other systems (including the Linux box) in one list.

Mark and Ed received word today that the long-awaited RFC for IAX2 has been approved by the IETF, and is now published: http://www.rfc-editor.org/authors/rfc5456.txt Thanks to Ed Guy, Mark Spencer, Brian Capouch, Frank Miller, and Kenny Shumard! Lots of revisions and discussions have paid off. JT --- John Todd email:jtodd at digium.com Digium, Inc. | Asterisk Open

Folks, I'm trying to implement a simple loop in a dialplan. The object is to set a counter, run through some IVR options, increment the counter, return to the start, then finally fall through to an operator or voicemail. Am using 1.4.10 and have reviewed doc/ exten => s,1,Set(TIMEOUT(digit)=5) exten => s,n,Set(TIMEOUT(response)=20) exten => s,n,Set(loop = 0) exten =>

I am investigating an issue with IAX2 extensions in Asterisk 1.8.x. My application connects to Asterisk via AMI and attempts to run an Originate command between an extension (such as SIP/5555 or IAX2/8888) and an application (in my case it is AgentLogin). This works correctly for SIP extensions, in all Asterisk versions. With IAX2 extensions, this worked correctly in Asterisk 1.6.2.20, but

> Asterisk Project Security Advisory - ASA-2007-013 > > +----------------------------------------------------------------------------------+ > | Product | Asterisk | > |----------------------+-----------------------------------------------------------| > | Summary | IAX2

> Asterisk Project Security Advisory - ASA-2007-013 > > +----------------------------------------------------------------------------------+ > | Product | Asterisk | > |----------------------+-----------------------------------------------------------| > | Summary | IAX2

Full_Name: Oliver Bandel Version: R version 2.7.1 (2008-06-23) OS: Linux (Debian Lenny) Submission from: (NULL) (88.73.82.147) Hello, I have written a small R-script. When I inserted one line of code, the behaviour was completely different! Instead of just printing one line more to the output, the complete call of the function in which this line was added, is different, and instead of one such

Hello, I have strings with date-/time-data from a logfile. For a sort-by-time (and other time-related analysis) I need something to convert the times to the Unix-epoche-time (better also with parts of seconds). Is there something that does that job (or a similar conversion) in R? TIA, Oliver Bandel

Hi i have a small problems on two Asterisk Server 1.6.4 : The first sent the call to the second, and in the second, i have a error : [Nov 15 15:30:12] ERROR[5113]: chan_iax2.c:4529 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address IP_FIRST_ASTERISK in the calltokenignore list or setting user 04TELNUMBER requirecalltoken=no on the second

I have an asterisk box using a SIP provider and IAX2 softphones clients. Audio is low and I need to apply some gain on it. How can I configure such gains - in/out on sip and iax2 channels? Thanks, Valter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101106/85417a17/attachment.htm

Folks, I installed a te110p to connect to an E-1. Ensured the jumper is on for E-1, installed the card, and got the following from the phone company: hdb3 encoding (verbally confirmed ccs) euroisdn switchtype pri signalling So I set up zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15 unused=17-31 dchan=16 loadzone=us defaultzone=us Card tries to come up but goes into RED alarm. Occasionally cycles

Hi all, I'm looking at my options for getting access to ISDN ISUP fields from DDI numbers, when connecting to a 3rd party Asterisk server. This is for a custom voicemail solution, and at this stage I want to avoid renting a PRI. The information I need to capture is: - Calling Number - Called Number (e.g. the DDI handling the call) - Redirecting Number (e.g. the device diverting to the

Hello, This is my 2nd attempt at running Samba on my linux box at work, but i thought i would ask some basic questions: David Bandel suggested i upgrade to 2.0, so i did..:) Now, what i want to do is the following: Be able to access the linux box from a 95/98 or NT workstation, now i know this can be done since my box "odie" showed up in the network neighborhood list on a few

I have setup iax2 between two servers without success. when I launch asterisk with the asterisk -vvvvvgcd command I see serveral wanings listed below. Is this why I cannot make connections?? My question is, how do I setup zaptel timming without any cards if possible? Does anyone have the steps? Thanks for any information. James [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) Apr

Hi List, I have a 2Mbps SDSL link which gets saturated during peak time because about I have about 3 E1 worth of g729 traffic going thru. So I'm planning to use IAX2 trunking to reduce bandwith requirement and squeeze out each and every bit of this (expensive) bandwith. I've set up two boxes (debian etch), one in a remote data center (which has plenty of bandwith) and one behing

Hi, I have an audio quality problem regarding IAX2. I have 2 Asterisk servers interconnected via 2 LAN trunks at 1Gbps (no nat, no firewall). One trunk is SIP and the other IAX2. Normally, I use IAX2 but have noticed easily reproducible audio quality problems (voice in/out is OK but there's a "third" noise overlapping with a "scratchy sound" as if it were some kind of

I am trying to set up an IAX2 trunk between two servers. Server A has the following in iax.conf: [general] ... [ServerB] type=friend trunk=yes host=dynamic secret=myPassword context=myContext Server B has the following in extensions.conf: [outgoing] exten=>_40X,1,Dial,IAX2/ServerB:myPassword@x.x.x.x/${EXTEN} I am using bmtools to monitor the bandwidth usage, and I am not seeing a difference.