similar to: TDMoE Configuration problems

Anybody have any issues running tdmoe on kernel 2.6+? I've got Suse 9.1 + 9.2 running 2.6.5 and 2.6.8 respectively, and when I enable dynamic spans between them, both boxes dump something similar to: Badness in local_bh_enable at kernel/softirq.c:141 [<c0120768>] local_bh_enable+0x48/0x60 [<c02952b0>] dev_queue_xmit+0x230/0x240 [<c02a0980>] eth_header+0x0/0x140

Hi, Is there any mailing list specifically for zaptel-users ? If yes, then, sorry to bother you people, i could not locate it. Can anybody post a link to it ? If not, then i hope, this is the right place to ask zaptel related questions. I'm using vanilla zaptel 1.4.3 on custom Linux 2.6.15.1. I've have a Redfone FoneBridge2 (TDMoE) gateway which i want to use with the zap drivers. It

Guys, I'm trying to configure a TE406P with MFC/R2. here goes my zaptel.conf: span=1,0,0,ccs,hdb3,crc4 cas=1-15:1101 dchan=16 cas=17-31:1101 span=2,0,0,ccs,hdb3,crc4 cas=32-46:1101 dchan=47 cas=48-62:1101 The first strange behavior is that the: zap show status shows this: Description Alarms IRQ bpviol CRC4 T4XXP (PCI) Card 0 Span 1

I was trying to use TDMoE and I lasted with two problems. First of all I can't configure the dynamic span to use CAS signalling but documentation (by Mark) says that you can use any type of signalling (and this includes CAS I guess). My second problem is related that my Linux system crashes frequently due to ztdynamic and friends. I'm using a 2.4.26 version kernel and zaptel

I'm just experimenting with dynamic spans using TDMoE, which looks ideal for clustering Asterisk boxes with low latency (i.e. 1ms for TDMoE compared with 20ms for IAX). When using normal E1s and T1s, I've always used ISDN signalling: either EuroISDN or NI2. So naturally I tried that over the TDMoE. I've found that I get a lot of HDLC errors and the D-channel is up and down like a

Hi all, We have our system hosted publicly and 4 phones are connected remotely at employee's home, and when they try to do a assisted transfer to one of the employee at the main office, the call is lost. For ex: person A calls person B, person B calls person C for assisted transfer, and as soon as person B hits transfer button again to transfer person A to C, the call is lost. But in the

Hi anyone have a idea for debug my digium TE405P card ? My zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone = fr defaultzone = fr My Zapata.conf: [channels] language=fr context=from-E1 switchtype = euroisdn pridialplan = unknown signalling = pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes

I have a box with 12 T1s (4 Te410P cards). The PSTN provider is reporting slips and ask me to update the clock source. I have my system.conf set as the following but when I run dahdi_scan only the ports on Card 1 are showing up with syncsrc=1 system.conf : span=1,1,0,esf,b8zs bchan=2-24 mtp2=1 span=2,2,0,esf,b8zs bchan=26-48 mtp2=25 span=3,3,0,esf,b8zs bchan=49-72

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]

Hi All, I am facing problem with E1 line. I have installed Asterisk (1.4.20.1) on a system with Digium TE420 card (Zaptel- 1.4.10) But every now and then I face problem of down E1's. The log show lot of entries like "pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2" This happens on a regular basis and the E1 becomes up after some time. My

Hi, Running 1.4.26.1 here. I have installed TE420B card in my server, and followed the appropriate steps (as far as I know to configure it). This TE420B is connected to a CLEC (T1s), so I am using pri_cpe as singalling type. When I dial out, I get this message: Dec 4 11:37:31] WARNING[27983]: app_dial.c:1275 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0

Hi, I have a digium card (igium, Inc. Wildcard TE405P quad-span T1/E1/J1 card 5.0V (rev 02)) 4 ports I want to make a loop test between digium card E1 to test the configuration of dahdi What I want to do scenario is I connect port 1 and port4 in the digium card with E1 cable SIPcall-->E1 Digium port 1--->(Loop)E1 port 2---->sip extension local. kindly can any can help me to

hello list, i have installed 2 diguim cards in my server using asterisk 1.4 (i use the old version with zapata.conf and zaptel.conf) i want to use the span 1 for group 1 and span 2-span 6 for the group 2 (i want to active the round-robin for span 2 and 6) in order to activate the WIMAX and FH please see the configuration below and tell me if there is anything wrong question 2: what is

Anybody sucessfully got stable 1000Hz clock without Digium harware and kernel 2.6? We need to consult some peoples how to clock asterisk stable with exactly 1000 Hz without much kernel/drives patching/tweaking. Some test results we made so far: 2.6 with digium card - stable 1000 Hz. 2.6 with ztdummy - uses RTC and the clock is 1024, not 1000. 2.6 with some Realtime kernel patch - provides stable

Hello- After configuring DAHDI and starting asterisk, I get the following message continuously on the Asterisk console: WARNING[2057]: chan_dahdi.c:4158 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! My card is a D410P configured for E1, only the first span is configured, and configuration snippets are as follows: From /etc/dahdi/system.conf:

I keep getting a red alarm when trying to setup asterisk to use my TE420B EC. I only have a blank context setup in my extensions.conf as I haven't started to config that until I can clear this red alarm. I don't have physical access to the server, so I can't go reseat the modules/card/ethernet cable, though I have hands on location that have done this a couple times

I have a 'CONGESTION' Status with R2 protocol. While testing this scenario sip GW--?Asterisk ?Digium E1 R2 Protocol?Cisco E1 R2 protocol?sip Gw Find below my error and configuration ,where are the errors in my configuration ? ========================================================================= Connected to Asterisk SVN-branch-1.6.2-r235775 currently running on

I'm scoping out HA for a relatively simple Office/Call Center PBX. Current setup uses a TE412P with 4 PRI our telco with SIP hard/soft phones for users. Some outbound also goes to a SIP provider. Active/Active looks to be too much hassle for an installation this size, so we're looking at adding an extra * in an active/passive configuration with Linux-HA in between them. Does anyone

I have a pair of sites tied together with a T1 line running zaptel PPP on either end. The system works, but I keep getting these messages in the kernel logs, and users are reporting connection problems (TCP timeouts, and hangs) especially under high usage. --snip-- HDLC Receiver overrun on channel WCT1/0/1 (master=WCT1/0/1) HDLC Receiver overrun on channel WCT1/0/1 (master=WCT1/0/1) HDLC

This afternoon I finally figured out more with regarding to a strange clock-slip problem we have on our asterisk box. We have two TE410s, in E1 mode: TE410P version c01a009b They have their own interrupts: 66: 781648298 783747388 IO-APIC-level t4xxp 233: 253890977 1311504670 IO-APIC-level t4xxp They have their full 31 channels: span=3,0,0,ccs,hdb3,crc4 bchan=63-77,79-93 dchan=78