similar to: Problem With zaphfc

Hi, I've been learning asterisk for a couple of weeks now - and it worked for me as faar as standard configurations where concerned (sip/iax outbound/isdn4linux & capi with AVM Fritz!, Digium X100P FXO). Now I recently I'evaluating to use asterisk as a replacemnt for our companies (15 employees) legacy pbx system and I'm experiencing multiple problems with the hfc isdn cards:

Hello all, if I try to call from one phone on the internal S0 to another on the same S0 using zaphfc, the bus is hung up. The called phone is ringing, but I can't talk from one phone to the other. The error I get is: -- Executing Dial("Zap/2-1", "ZAP/1/55|15|tr") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called 1/55 -- Channel 0/1,

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

I've got a 4 port T1 card in my Asterisk box with a PRI from Qwest as my PSTN interface. I'm experiencing random dropped calls on the various SIP devices I have tested. Network connectivity to the SIP devices looks ok, and I have tried a variety of the devices including all of the following. Grandstream 286 Grandstresm 486 Sipura SPA 1000 Mediatrix 2102 Some example lines from my logs

I managed to get Asterisk working with my PBX using T1, now I am moving on to trying to make PRI work. I have my zaptel.conf and zapata.conf configured as follows: Zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone=us defaultzone=us Zapata.conf: [channels] transfer=yes immediate=yes callprogress=yes language=en context=default switchtype=national signalling=pri_net group=1

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

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

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

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

I've having problems with a BT 2 span ISDN-30/Digium TE205P asterisk setup with outgoing calls not completing and requiring an Asterisk reset to 'unstick' span 1. Sorry this is a bit long but I'm completely out of my depth :-( This system has been in use for some while and I recently upgraded it to asterisk 1.4.24, zaptel 1.4.11 and libpri 1.4.9. I didn't change

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

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

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

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

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

Hi I have a small ISDN PABX , (BRI) at home - Siemens Gigaset 4175 which has cordless DECT extensions. I have set up asterisk on FC3 with two HFC cards and I am using the latest bristuff. I am trying to use * between my ISDN line and my PABX. One card is in TE mode and can receive and make calls OK. I can make and receive external calls to a SIP phone. I cannot get the other card to talk to my

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 all, I really hope that you guys can help, because I've been tearing my hair out for the past 5 hours on this one. I have a Zaphfc (BRI) card in TE mode connected to the S-Bus of a Nortel Meridian phone system. Phone calls from the Nortel to say MSN 510 are correctly being sent to the right SIP phone. When asterisk dials say Zap/g2/224 (a Nortel internal extension) the call goes

Hi All, The meetme.conf have created as below: [rooms] conf => 101 conf => 102 and extensions.conf as below: exten => _1XX,1,MeetMe,${EXTEN} why the warning printed when I called 101. WARNING[27660]: File pbx.c, Line 1051 (pbx_extension_helper): No application 'MeetMe' for extension (ipcentrex, 101, 1) And I found asterisk have not load the meetme.conf when it starts up.

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