similar to: 1.4.26.1, 1.6.0.13, 1.6.1.4

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, I've been fighting with this seems like forever now and can't make it work in 1.6.x. In 1.4.x, I could make sure a particular voip provider was always first in the list by making him as an #include and putting it last. Now in 1.6.x I can never get this to take. I really don't want to run TWO asterisk servers just for some IAX trunking. Real problem: my voip provider

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.

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 =>

The Asterisk Development Team is pleased to announce the releases of 1.2.34, 1.4.26.1, 1.6.0.12, and 1.6.1.4. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of 1.6.1.4 fixes a remote crash security vulnerability in the SIP stack. Although this crash was not demonstrated in any other version, the details of the

Hi all, I'm running 1.6.2.0-rc6, and I'm running into a problem: sometimes the audio vanishes in the middle of listening to an IVR background prompt. This happens with both analog (Digium card) and IAX2 incoming calls. The prompts are stored in ulaw format (and the IAX2 calls use ulaw). The asterisk console claims that the IVR prompts are proceeding in the expected fashion, but I

I am running asterisk 1.4.26.1 and using ALSA not oss dahdi 2.2.0 and libpri-1.4.10 I am calling into console/dsp I hear the audio just fine then after the hangup I hear ringing on the console/dsp. Why would that be? I found this bug for OSS https://issues.asterisk.org/view.php?id=13686 Does the same thing exist in ALSA??? some traces below Jerry == Parsing

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 anyone know where i can find all internatinal numbering plan in csv and for free or small price ? thanks Jpc

G'day all, I'm using Asterisk 1.4 and am trying to work out a way to bring people into a conference call. In the ideal scenario two people would be talking and one of them would push some keys, then a phone number and then the three of them would be in a conference. From there they should be able to bring in other people as well. This seems to be what the Asterisk n-way call HOWTO

Dear All Can you please do me favor and let me know how I can hide the subs number being displayed on his phone when he goes off hook ? I mean when the subs goes off hook he sees his assigned number on his phone and I need to disable this feature . I don't know from which configuration file this feature is coming so please let me know how can I disable it . Regards H.Motamedi --------------

Dear all, I'm trying to install Asterisk-stat (ASTERISK CDR ANALYSER) following: http://www.voip-info.org/wiki/index.php?page=Asterisk+CDR+Areski+GUI however it fails to run properly - lots of lines like: **Notice**: Undefined variable: s in */home/lukasz/DATA/www/asterisk-stat/cdr.php* on line *26* **Notice**: Undefined variable: t in */home/lukasz/DATA/www/asterisk-stat/cdr.php* on line

I have a trunk, and its host=dynamic dns. The problem is, when the IP changes the Sip show peers Still show the old IP of the DNS, I have to reload and save the configuration again so that asterisk recognize the new IP of the DNS. Any idea how to automate such a thing? Or how can I keep asterisk to deal with NAMES as NAMES, and IPs as IPs. Let me know. Thanks. -------------- next

Hi, I tried getting our server setup for 400-500 simultaneous calls, calls were going through properly but at around 200-250 calls, mysql (connect ...) statement was taking at least 5-10 sec to connect to the database. I optimized all possible parameters in my.cnf: max_connection=1000 wait_timeout=60 query_cache_type=1 query_cache_limit=4M query_cache_size=512M interactive_timeout=120

Is there any program Asterisk users use to calculate ASR and ACD ?? Thanks for any comments. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090910/af1f9656/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, I was looking round on the Internet and saw there was no definitive list of free applications available for use with Asterisk, so I thought I'd compile a list for you all. If there's anything that you know of that is actively maintained but not in the list below, let me know (bear in mind I'm not including distros or Asterisk packagings in this list). Hopefully there are a few

Hi! I want to use Asterisk as load generator to test quality degradation with increased load (e.g. testing other SIP equipment or IP-links). Is anybody aware of such a setup with Asterisk - is it possible to get RTP statistics out of Asterisk (e.g. jitter, packet loss, reordering ...)? Thanks Klaus