Fail to compile DAHDI under Linux 3.0.0

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Just finished installing Kernel 3.0.0 and tried to compile DAHDI
Complete 2.4.1.2+2.4.1. Errors below: ________ /home/doug/Extracts/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi-base.c:95:2:
warning: #warning "No CONFIG_BKL is an experimental configuration."
CC [M]
/home/doug/Extracts/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/vpmadt032_loader/dahdi_vpmadt032_loader.o
SHIPPED
/home/doug/Extracts/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_32.o
LD [M]
/home/doug/Extracts/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi.o
CC [M]
/home/doug/Extracts/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi_dynamic.o
CC [M]
/home/doug/Extracts/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi_dynamic_loc.o
CC [M]
/home/doug/Extracts/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi_dynamic_eth.o
CC [M]
/home/doug/Extracts/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi_dynamic_ethmf.o
CC [M]
/home/doug/Extracts/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi_transcode.o
/home/doug/Extracts/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi_transcode.c:49:
error: ‘SPIN_LOCK_UNLOCKED’ undeclared here (not in a function)
make[3]: ***
[/home/doug/Extracts/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi_transcode.o]
Error 1
make[2]: ***
[_module_/home/doug/Extracts/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi]
Error 2
make[2]: Leaving directory `/usr/src/linux-3.0'
make[1]: ***…

Asterisk Users 4.2 years ago 1 Answer

dahdi channels busy/congested

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Dear all, i have a problem with a system running - Ubuntu 10.04 ( all updates done )
- ii asterisk 1:1.8.5.0-1digium1~lucid Open Source Private Branch Exchange (PBX)
- ii asterisk-dahdi 1:1.8.5.0-1digium1~lucid DAHDI devices support for the Asterisk PBX I also use freepbx 2.9 for the configuration. Hardware is a Dell R410 and a Digium Wildcard wcte12xp+ d161:8000 Wildcard TE121 The status is as follows,
- all drivers are loaded and the E1 card shows a GREEN LED and no alarms
- asterisk is up
- the provider is Bharti Airtel in India
-…

Asterisk Users 4.2 years ago 3 Answers

Error Code 101

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Hi, I have 8 port PRI Sangoma Card connected to the Server running
Asterisk 1.8.5 on CentOS 5.6. I see these errors in sangoma file under
/var/log/asterisk/
http://pastebin.ubuntu.com/651451/
Error code 101 is "Message not compatible with call state". The
explanation for this is "The remote equipment received an unexpected
message that does not correspond to the current state of the
connection. This is usually due to a D-channel error. Please suggest/guide and let me know if anyone needs any information
about configs. Regards, Kaushal

Asterisk Users 4.2 years ago 1 Answer

One way calling on asterisk to cisco

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I did duplicate cucm as cucm2. I was a bit confused as to what changed. However, it was the same results. I commented out the cucm1 instances so it was forced to use cucm2. however I still get the same results: == Using SIP RTP CoS mark 5

Asterisk Users 4.2 years ago 0 Answers

Security questions

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Hello everybody! I'd like to heard from those with more experience in Security if the following configuration is a good attempt to prevent hack: exten => CALLER,2,Set(header=${SIP_HEADER(User-Agent)}) exten => CALLER,3,NoOp(Cabecalho ${header}) exten => CALLER,4,GotoIf($["${header}"= "My User Agent"]?6:7) Considering I have only one type of IP phone in my scenario. I know, somebody with another IP phone will succeed in dial on my asterisk but I think it will limit at one only kind of IP phone. My question is , if there are some way to break it and use any kind of User Agent despite this configuratio.

Asterisk Users 4.2 years ago 2 Answers

Securing Asterisk

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I beg to differ. Digium is hiding from the real world and somebody is
going take the software and run with it. My customers lost in excess
of $50.000 and cut my pay in half, because of hackers. The hackers
figured out how to scan every asterisk for weak passwords or open
ports, and bang them real good. We need two things: a) disable in
sip.conf the reply for INVITES that have wrong user information, and
also, b) disable any response to any REGISTER packet altogether. Can
somebody please write patch? Or should…

Asterisk Users 4.2 years ago 3 Answers

Invalid argument

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Hi All; Asterisk version is 1.8.4
The call is going to the queue but it is not routing to the agent which is logged in. I am afraid that I am missing a parameter to be set or enable or disable in the queues.conf so it is causing not to route for the interface. I am getting the following error messages: [Jul 24 18:45:02] WARNING[13177]: pbx.c:4893 __ast_pbx_run: Don't know what to do with 'SIP/gwbilalhome-00000bfd' == Using SIP RTP CoS mark 5 [Jul 24 18:45:07] WARNING[13175]: acl.c:708 ast_ouraddrfor: Cannot connect
[Jul 24 18:45:07] WARNING[13175]: chan_sip.c:3280 __sip_xmit: sip_xmit of 0x7ff3…

Asterisk Users 4.2 years ago 1 Answer

DISA password

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Hello list, how can I give a simple password to the DISA-application ? The following : exten => 1000,n,DISA('123456',from-TEST) results in : [Jul 23 13:47:51] WARNING[2357]: app_disa.c:255 disa_exec: DISA password
file '123456' not found on chan SIP/test6-00000006

Asterisk Users 4.2 years ago 2 Answers

Securing Asterisk - How to avoid sending, "SIP/2.0 603 Declined"

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Hello, I am wondering if there is a way to drop SIP packets for generic
transactions? For example, only SIP PEERs are allowed to call in and receive
ACK or Declined rather that those inviting a call who are not PEERs at all. Currently my Asterisk setup sends, "*SIP/2.0 603 Declined" *to any stranger
invites because my dialplan includes Hangup(). Is there any way I can not
send a 603 declined so to mislead the probe runner? Thanks

Asterisk Users 4.2 years ago 13 Answers