similar to: Restarting of B-channel on span 1

Hello All, Hope you all are fine and good ... I m facing an odd problem. I am able to dial some local and Mobile Number and some are not working. I also try to remove and re create dial plans , but it's not working. I live in New Delhi and Area Code is {011} but i am bale to call the number without adding {0} by just {11}, Previously i was thinking it was PRI problem, so i called the

Hi Guys, I have been checking logs and noticed this over the last night. Should I be concerned? and where to look for further details? Sample: [2010-04-13 04:31:27] VERBOSE[3844] logger.c: -- B-channel 0/1 successfully restarted on span 1 [2010-04-13 04:31:27] VERBOSE[3844] logger.c: -- B-channel 0/2 successfully restarted on span 1 [2010-04-13 04:31:27] VERBOSE[3844] logger.c: --

Hey everybody, I've, within the last 3 weeks, moved over to a PRI from SBC/AT&T. I've received several complaints about dropped calls. Reviewing the archives on PRI and dropped calls shows that I should set the resetinterval=never in the zapata.conf and restart. This hasn't helped. The dropped calls have to date only been on outbound calls. Usually within 2 to 3 minutes

Hi I''m trying to setup my app to use Feedtools amazing RSS powers, but it fails miserably on the following line: logger.debug("Description:#{FeedTools::Feed.open("http://rss.slashdot.org/Slashdot/slashdot").items.first.description}") The stack trace says: wrong number of arguments (0 for 1)

We are using AAH with Asterisk 1.2.7.1 with a TE405P as listed below. We are getting frequent restarts on the spans which lead to dropped calls. I have pasted some hopefully pertinent information below -- anyone have any clues that might help? Thanks Next line is repeated throughout messages, going through every channel in every connected span. asterisk/full.1:Apr 24 01:15:25 VERBOSE[4196]

<noobalert> Uh, yeah. So my noob self is showing through again. I''m trying to write a simple hello-world type web service with the ActionWebService thingy. I didn''t get very far. I''ve only done web services stuff in .NET, so I''m not exactly used to this sort of thing. Anyhow, I have a file, person_api.rb in my app/apis folder. I want to take a single

Good day everyone, I have Asterisk and Mitel 3300 ICP communicating via SIP. Calls to and from extensions on both sides are completing successfully (details on config coming soon). Problem is that calls from Asterisk to PSTN via E1 on Mitel 3300 ICP are rejected. What do I need to do to get Asterisk to route PSTN calls through it successfully? Here is an extract of the log on Asterisk whenever I

I upgraded from 12.2 to 12.4 and found that my happy little eventmachine processes die with: terminate called after throwing an instance of ''std::runtime_error'' what(): call SetTlsParms before calling StartTls Here''s what I believe is the relevant snippet of my code, which calls start_tls() in the connection_completed() callback: connection =

Hey, I'm up and running fine with 30 Polycom 500s connected to Asterisk 1.2Beta on Cent OS 4.1 with a Digium TE110 connected to a PRI line. Nearly every hour, almost on the hour I get this: Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/1 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/2 successfully restarted on span 1 Sep 29

Has anyone else had this symptom? It seems like my PRI ?restarts? each hour. I?ve got a PRI from Qwest in central California. What I don?t know is if this is on the * side or on the telco side. I don?t use my system enough to know if this is a big problem, a minor glitch or just an ?undocumented feature? - ? Any suggestions on how to track the source of these ?restarts?? Anybody else seen

Hey all, I am getting this error: NoMethodError (undefined method `gsub!'' for 2012-01-22 17:00:00 -0500..2012-01-23 00:00:00 -0500:Chronic::Span): in this code: date_range = Chronic.parse(the_date, :guess => false) reports.sum_distance_by_date(date_range).each do |d| u[:m] << d end def

Hi, Does anyone have any good empirical data suggesting what the maximum number of PRI calls (incoming and outgoing) without hardware echo cancellation can be handled on a single box is? I have a TE410P T1 (1st gen) card and I'm seeing interesting errors of D-Channels going down and then coming back up (See below). I've looked at the number of simultaneous calls at each of these points,

Hi all, I have a connect between a siemens hipath & Asterisk system over PRI The connection works perfectly I can call from the Hipath to an Asterisk Extension. I want to allow the hipath extensions to dial out over a SIP trunk on asterisk but I keep getting "The number you have dialed is not in service" In this CLI I'm dialing from hipath extension 9 (prefix for sip trunk)

I have a PRI E1 link between Asterisk 1.4.24 and Alcatel-Lucent OmniPCX Enterprise R9.0. As EuroISDN it works fine. However, I need to move to QSIG because of a firmware upgrade on the Alcatel PBX which doesn't support EuroISDN (please don't ask why). Besides, I've read somewhere that 2 B Channel Transfers "should" work with * 1.4, the latest 1.4 libpri and QSIG. So this

Hi, I set up a ring group. I would like for people who select a certain voice menu option to ring a list of extensions (I have just one extension in there at the moment) and if it doesn't answer to go to an extension's voice mail. I am using a version of asterisk from CVS, last updated a couple of weeks ago. This line in extensions_addtional.conf sends the call to ringgroup 3 if

Did you compile and install libpri *before* Asterisk? I had same problem (among others) b/c I didn't install in the correct order. Try the awesome asterisk_update.sh shell script. Are you trying to emulate CPE or NET? Try signalling=pri_cpe Check for whitespace behind the statement, zapata.conf seems bitchy about whitespace. hth -----Original Message----- From: Steve Totaro

Any ideas? 51] logger.c: [chan_zap.so] => (Zapata Telephony) Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata.conf': Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata.conf': Found Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata-auto.conf': Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing

hi i'm trying to configure a TE205P on asterisk@home i've edited /etc/sysconfig/zaptel adding this line: MODULES="$MODULES wct2xxp" now, when the system is loading, i can see that the wct2xxp module is loaded correctly but if i try the command: /usr/local/sbin/genzaptelconf i get: STOPPING ASTERISK STOPPING FOP SERVER Generating '/etc/zaptel.conf' Generating

Hi, I am using asterisk 1.4.28 with freepbx and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing "Unknown" when there is an incoming call. *My log file showing this while an incoming call on PSTN line:* tail -f /var/log/asterisk/full [Dec 30 06:36:16] DEBUG[2559] dsp.c: dsp busy pattern set to 0,0

Hi! This weekend we had a problem with our Asterisk Box which ran flawlessly for nearly 4 weeks. The Asterisk server sits between the PSTN and a Siemens PBX and bridges 2 BRI lines. No calls, not incoming, not outgoing. The admin rebooted the Dell Box and then everything worked fine again. Now, I'm analyzing log files to find the cause. During the Asterisk outage the logfiles only show