similar to: Random Dropped Called

I just had the same exact problem this morning. The only thing I've done in the last couple of days is update update zaptel. I rolled back my zaptel to 2/11/04 from 3/8/04. And kept my libpri from 3/8/04. I never had this error before updated. I had other issues, but not this one. -sb -----Original Message----- From: asterisk-users-admin@lists.digium.com

Configuration: Linux wpbx 2.4.9-13 #1 Tue Oct 30 20:11:04 EST 2001 i686 unknown P4 2.5 GHz, 1 GB RAM T400P with 3 T1s plugged in. A flow of 46 calls (spread out over 3 T1s). Each call gets transferred (Dial) to the SIP platform and stays for 5 min. Case 1. Asterisk built out of CVS Mar. 19. Test was running for 3 days. Segmentation fault. Case 2. Asterisk built out of CVS Apr. 1. Test was running

I've this error How i can find the problem? Apr 23 12:24:43 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:24:47 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:24:48 WARNING[131081]: PRI: !! Got a UA, but i'm in state 1 Apr 23 12:24:53 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:25:02 WARNING[131081]: PRI:

Hi! I have the chance to play with a couple of E400P cards, each installed in a IBM e330 XSeries servers (2 x 1GHz P-III CPU 2 Gb RAM, 36Gb SCSI HDD with RH8.0 2.4.18-smp kernel), and I'm trying to test/benchmark this e330/E400P combo generating calls thru /var/spool/asterisk/outgoing One e400P if doing the carrier work making calls and the other just receives the calls: Server#1

Hi all, Sorry to bother you with the potentially not too bright question, but since I cannot get it to work, I'll have to ask someone.. I've got this vector: > token [1] sal skal skal bak b_r d_ gal kal l_r n_r pak p_r [13] sal sl_r sn_r spak sp_r st_r s_ bak d_ gal l_r pak [25] p_r sak sl_r sp_r t_ kal n_r s_ st_r sak kal

Hi, Urgently response would be wonderful, system is a Fedora Core 2. I have a Ericsson BP250 connected to 1 port on the TE405P and another connected to a local telco ISDN30. I have been running CVS-HEAD from about a 2 months ago and upgraded it again just in cause it was a version issue (didn't fix it) but this is what I am getting. When a person calls out from an extension on the BP250 to

Dear List i have upgrade my * box with the latest CVS version of Asterisk Stable 1.0 and zaptel/libpri my system is MDK9.2 with 1 TE410P and seems work well for now but i have a little amount of traffic (25 IN/OUT calls) i only notice this Warning.. What kind of error is? ------------------------------- Apr 20 21:28:49 WARNING[147466]: chan_zap.c:5979 zt_pri_error: PRI: !! Got reject for

Hi all- HELP! This is actually a revisit of a problem that I had earlier with E400P's at a customer site. Customer still gets locked up channel problem, but has learned to live with it (channels clear themselves after several minutes). The symptoms, which I believe are directly related: I'm having problems with tons of framing and "read" errors on my E1 connections (and

I just tried to go from CAS to PRI on my T1 (Sangoma), and failed pretty badly. Seemingly everything worked -- Asterisk would see the incoming call (including CID and DID info), try to route it, and fail -- giving me a telco (not Asterisk) call failure message. My zapata.conf and zaptel.conf files are at http://pastebin.com/558349 Below's the log dump. Note that, because I was simply going

Hi all, I apologies in advance if I am missing something very simple here, but since I failed at resolving this myself, I'm sending this question to the list. I would appreciate any help in understanding how the rpart function is (exactly) computing the "improve" (which is given in fit$split), and how it differs when using the split='information' vs split='gini'

I'm in the process of turning up a PRI in one of my markets and have run into a problem I have never seen before. I am unable to place a local outgoing call. Long Distance over the same PRI works fine. When I attempt to place a local call using the PRI I see Asterisk attempt to dial, and am greeted with a busy signal. This signal appears to originate on the telco's switch. I have had

Hello all, I got a problem at an asterisk server, with dropping calls, losing all channels and reaktivating all channels and beeing back up. This problem seems to occure randomly over the whole day, when it gots traffic on the card. After looking @ google I found several hints but none did work fine. To avoid problems with the phone line (german E1) I called the provider, he did a 45 min. route

Hi, We have a Ericsson BP250 Phone system setup witht he following configuration Telco <-> Asterisk E405P <-> BP250 The system seem to work perfectly on 1.0.9 for a very long time but there is some functionality we wanted to take advantage of in the 1.2 version branch so we upgraded. Currently running Asterisk 1.2.4 Zaptel 1.2.3 (noticed 1.2.4 is out will upgrade next

I've got a client who would love to be able to leave an asterisk server running sompelace, and achieve telephone connectivity using an IP phone from within his Motorhome in his words "I want to be able to work from a mountaintop in Glacier National Park" I've done some initial testing, and a SNOM200 SIP phone comes close enough to working that I have not ruled out the idea as

I've noticed that the max.col function with the default "random" option often gives unexpected results. For instance, in this test, it seems clear what the answer should be: > # second col should always be max > x1 = cbind(1:10, 2:11, -Inf) > > # this works fine > max.col(x1, "first") [1] 2 2 2 2 2 2 2 2 2 2 > > # this gives random answers >

Anyone know what would cause this error? !! Got reject for frame 39, retransmitting frame 39 now, updating n_r! !! Got reject for frame 39, retransmitting frame 40 now, updating n_r! I assume this would cause audio issues as well. Thanks, Steve > Message type: CALL PROCEEDING (2) > [18 03 a9 83 89] > Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0

Dear list, While trying to identify the root of a problem I am having with garbage collected variables, I have come across the following oddity: depending on whether --verbose is set or not, I obtain different results. I have made a small standalone example to demonstrate it. The example is very artificial, but I had a hard time reproducing reliably the problem. So when I do: (the content of

Hi- I've posted this on the bugs list, but I'd also like to see if others have had similar problems when connecting via E1 trunks (E400P). I'm getting numerous errors like the following during inbound calls to my E1 channels. These occur when the system is under medium load: WARNING[196621]: File chan_zap.c, Line 5404 (zt_pri_error): PRI: !! Got reject for frame 78, retransmitting

Hi, we have a E1 pri from Citylink, (they are using Ericsson Engine exchange), that are restarting after 5 - 15 minutes, before and after that we can make calls in and out w/o problems. The cards have been tested in two computers (Atholon XP 2200+ and Celeron 2.6Ghz), are on there own IRQ, not showing any IRQ misses in zttool. It's all correctly configuered - zaptel and zapata (attached). We

Hello everybody, Just was wondering if somebody can help for G711 fax passthrough w/ asterisk. The issue I have is regarding codec upspeed when the call is already connected using G729 for example. The setup is fax---ATA---asterisk---Cisco---fax When codec upspeed should happen, ATA or Cisco will send a G711 reINVITE causing the codec to be switched over, but asterisk does NOT