similar to: iaxpeers from Realtime

Hi all, Trying to do what should be a basic info retrieval from my asterisk network -- get the list of IAXPeers and break down by IP, name, and Status etc. Asterisk 1.0 Manager unfortunately. Using PHP. Easy to get the response -- but parsing it is impossible! That table format throws me for a loop ... Any idea of an easy way to seperate out the columns of that horrid 'table' reponse

Hi, I've just done up a quick proggy to show me the status of my IAX peers from my windows box. It plugs into the simple manager proxy. You can see more information (including a screenshot) at: http://www.sineapps.com/news.php?rssid=384 You can download it directly from: http://www.sineapps.com/down/IAXPeers.zip Could you please have a look and let me know your thoughts. -- Cheers,

Is there any reason why IAXPeers output is different from SIPPeers output? The response has no Eventlist: start Ej. Response: Success Eventlist: start Message: Peer status list will follow Event: PeerEntry Channeltype: SIP ObjectName: 1001 ChanObjectType: peer IPaddress: 192.168.175.1 IPport: 63772 Dynamic: yes Natsupport: no VideoSupport: no TextSupport: no ACL: no Status:

Asterisk 1.4.21.1 built by asterisk @ lnxca Fedora 7 Well, I did the upgrade from version 1.4.16.2 to 1.4.21.1, activated the qualify for iax users, and Asterisk is still loosing the iax user's registration. Once again this is really wired because no configurations changes have been done to the system, the only change was a reinstallation of asterisk with user "asterisk" instead of

All, I have spent the last couple of days looking through the mail archives and the documentation on the Wiki, but have not been able to find a solution to the problem. The version of code I am running is from CVS as of 6/30/04. What happens is that when I make an IAX call to another IAX client the caller receives a really bad echo. All of the documentation I found around using

Hi folks, I'm really confused, so please help me, or at least give me some pointers to clarify this issue. Can I mix "Static" and "Real" realtime? Is there a way to easily switch from one to another, say, for sip.conf? Which are the major benefits of "Real" realtime? Please help me! Thanks in advance, -- Andrea Spadaccini Multimedia Technologies Institute

Could someone provide some steps for troubleshooting Realtime? I can't see any signs that it's working. I followed and double-checked a few different guides around the net, but haven't been able to figure it out. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi! I am proving Asterisk 1.2.24 in realtime with MySQL 5.0.27 using Idefisk softphones. I followed the steps of "how to" of voip-org but always have this error: Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime:

Starting with Asterisk 1.2 I have always used realtime static to load my extensions.conf into Asterisk. It worked perfectly up to version 1.6.0.X but starting from 1.6.1.X and upwards it simply does nothing. I can see that the extensions.conf file is mapped to the database: == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': ==

Hello, With setvar statements in chan_dahdi.conf, we have a convenient way to store DAHDI channels specific values. Unfortunately, we don't have a function to access this data from the dialplan as easily as SIPPEER ou IAXPEER would for SIP or IAX trunks. Using AST_CONFIG, you can access DAHDI setvar value but: 1. only one setvar value (see bellow) 2. AST_CONFIG reads values from current

Hello, We use rsync to synchronize a large folder hierarchy between servers. Sometimes we notice that the permissions of some directories end up being wrong after rsync is run. The source directory has permissions 755 while target permissions sometimes end up being 700. We are using the --timeout option of rsync, and we have additional watchdog process that will terminate the process if a total

Hi! I am proving Asterisk 1.2.24 in realtime with MySQL 5.0.27 using Idefisk softphones. I followed the steps of "how to" of voip-org but always have this error: Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime:

After updating to the latest CVS Feb 17 15:20:03 WARNING[15317]: config.c:819 read_config_maps: The 'sipfriends' table is obsolete, update your config to use sipusers and sippeers, though they can point to the same table. == Binding sipusers to mysql/asterisk/sip == Binding sippeers to mysql/asterisk/sip Feb 17 15:20:03 WARNING[15317]: config.c:823 read_config_maps: The

Hi when i use MeetMe, i have this errors: app_meetme.c: Unable to open pseudo device Where is the problems ? i have too warning and error into my logs: [Nov 1 07:26:17] WARNING[18544] res_musiconhold.c: Unable to open pseudo channel for timing... Sound may be choppy. [Nov 1 07:26:17] WARNING[18544] config.c: Realtime mapping for 'iaxpeers' found to engine 'mysql', but

Is it just my box, or is there something flaky in the implementation of "show manager commands"? Note: I'm using putty. About half way through this, I toggled my KVM over to the desktop and logged in to try and recreate it. The output was the same as the last two entries in this dump. bebop*CLI> show manager commands bebop*CLPing Ping bebop*CLLogoff Logoff Manager

Hi all, A week or so down the list, i read that not many people were using realtime on an Asterisk18, so i had this afternoon a go at it... [sorry for the inconveneant line-wraps] First i did: mysql> create database asterisk; mysql> grant all on asterisk.* to 'voipadmin'@'localhost' identified by next i used the info from the wiki: CREATE TABLE `sip_devices` ( `id`

Hi, in the menu of voicemailmain, appear a lot of options, there is a way to leave only some of them? Also I want to know if there is a option that erase all message in a user box. Best REgards Ever Zalazar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060512/98a6f962/attachment.htm

I signed up for an IAXTEL account and have been trying, unsuccessfully, to get it working. In IAX.CONF I have: [iaxtel_out] type=peer host=iaxtel.com username=USERNAME secret=SECRET auth=rsa inkeys=iaxtel [iaxtel] type=friend context=incoming host=iaxtel.com auth=rsa inkeys=iaxtel However, when I start Asterisk, I get the following warning: [chan_iax2.so] => (Inter Asterisk eXchange

flight 5665 xen-unstable real [real] http://www.chiark.greenend.org.uk/~xensrcts/logs/5665/ Regressions :-( Tests which did not succeed and are blocking: build-amd64-oldkern 4 xen-build fail REGR. vs. 5640 build-amd64 4 xen-build fail REGR. vs. 5640 build-i386-oldkern 4 xen-build fail REGR. vs. 5640

Hi, I am setting up a small call center using *. I have ZAP setup for incoming calls and IAX setup for agents. Agents login using AgentCallbackLogin. When customers call, it's getting picked up and when queue is trying to call back the agents, I am getting error. I am using CVS HEAD, and updated just now. The error is: -- Executing Answer("Zap/1-1", "") in new