similar to: audio cuts out during IVR

I'm trying to get call parking working with the lighted buttons on the SNOM 200. I have set the 5 buttons to "Park Orbit", for extensions 700-704. Pressing the first button (x700) does park the call. However, the remaining buttons (x701-704) don't allow me to pick up parked calls, or show parking status via the LEDs. I can only pick up parked calling by manually dialing the

With Asterisk 1.4 I could use commands like: /usr/sbin/asterisk -rx "sip notify reboot-snom mjc_home" to reboot a snom phone. Now, with 1.6, when I try that, I get: Unable to find notify type 'reboot-snom' Command 'sip notify reboot-snom mjc_home' failed. Do I need to add some magic to sip_notify.conf? I haven't quite figured out how to make it work. - Mike

Hi, Is it possible to use the hint priority to allow call parking slots to be monitored on (for example) Snom indicator lamps? How do you refer to the slots (i.e., what is the "channel") in the hint? - Mike

Hi all, I currently have an Asterisk test server behind a TZ170 Sonicwall firewall / NAT box, with several DIDs. I've found that inbound IAX2 calls don't work reliably (i.e., I get a busy tone) unless I enable "Use Consistent NAT" in the Sonicwall. This feature is poorly documented by Sonicwall, so I thought I'd pass it along. Has anyone else run into this, or figured

Hi all, One of my users reports frequently hearing echo on her Snom 360 phone, even while talking to other Snom phones (via Asterisk) on the same LAN (i.e., all-digital low-latency connection). I can never reproduce it though, and swapping the phone didn't help. Has anyone else seen "mystery echo" on Snom phones? Any suggestions for debugging? On my own Snom 360, I sometimes

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:

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, 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

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 --------------

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

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

Hello Anyone who have already use/configure Asterisk with CDRTool ? Or maybe can suggest another CDR GUI ? regards. Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090812/e3e9e675/attachment.htm

Dear All Can you please do me favor and let me know what is my problem with my Asterisk VoiceMail configuration as it doesn't work correctly in my case ? Please find below that part of my extensions.conf that I intend to make use of voice mail for No Answer reply : " [line-incoming] exten => _XXXXXXX,1,macro(dialuser,SIP/${EXTEN},${EXTEN}) [macro-dialuser] exten =>

Hi all, I'm looking for a reporting solution for Asterisk CDRs. I have a small Asterisk server that will eventually have 4 - 6 trunks. the system is up and the CDRs are being written to a MySQL DB. I tried installing Areski, but had no success .. I assume it's no longer supported... the last update was in March 2005 according to this page..

Hi, My Dial() is set to the following, but always stops about 30 seconds into the call even when I set it to try for 60 seconds. exten => dialnumber,1,Dial(${DialInfo},60) I am running on 1.6.1-r199820. Is there some other setting that is overriding mine? Or an issue with this release? Thanks for the help. JR -------------- next part -------------- An HTML attachment was

Hi All, I've just upgraded our CRM and it has an Asterisk Integration Module that I would like to test out. The CRM is running on one of our hosted servers in the cloud. The Asterisk server is running in my office. I am running Asterisk 1.4.21.2~dfsg-1ubuntu3. Reading the page http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20manager.conf got me a little concerned