Does Asterisk provide any SPC(Signalling Point Code)? Can anybody provide me tutorial on how to configure E1 card in Asterisk?
Hi
For which country ? and signalling ?
Ian
Dear ian,
my zaptel.conf is like this:
span=1,1,3,cas,hdb3
bchan=1-15,17-31
dchan=16
getting following error
31 channels to configure.
ZT_SPANCONFIG failed on span 1: Invalid argument (22)
Please help me
Hi
What does your zapata.conf look like and wha does dmesg show ?
Ian
[quote=“scream”]
my zaptel.conf is like this:
span=1,1,3,cas,hdb3
bchan=1-15,17-31
dchan=16
getting following error
31 channels to configure.
ZT_SPANCONFIG failed on span 1: Invalid argument (22)
Please help me[/quote]
Your zaptel.conf is set for channel associated signalling (cas). You will need to set it as SS7, if zaptel supports it, and you will probably need some additional SS7 software plus other configurations.
Why do you need an SPC?
Dallas
Hi Have I missed something here
when did the OP mention they wanted ss7 ?
thank for your prompt reply. i am using libss7 and asterisk-trunk. my card is digium T100P single span card. is it support E1? if i configure for T1 it works. but if i configure for E1 with signalling cas or ccs it gives that error. i mentioned signalling type ss7 in zapata.conf.
Can you give me any manual or step by step conf or sample zapata and zaptel conf file?
ian,
The OP didn’t mention SS7 specifically, but SPC is the signalling infrastructure use in SS7… prior telephony knowledge 
scream,
Sorry but I haven’t used the T100P card. It may be T1 only but don’t quote me.
Dallas
Thanks Dallas
Ok
This is the confusion, the OP never mentioned SS7 but did ask for SPC, and yes I did know this may have pointed to using SS7, having working on telephone networking for over a quarter of a century, But as it wasn’t mentioned and the zapata.conf was never posted so who could tell.
The T100P is a T1 card thats what the T stood for, there was an E100P but both of these were replaced with the TE110 and now the TE120
Also the zaptel.conf is wrong any was as it set to CAS channel associated signalling and not CCS common channel signalling.
If you want to use e1 then you will need a new card.
Ian
thanks ian
I’d say that sorts it all out for scream.
I knew that presentation on SS7 back in the mid 1970s would come in handy one day. I spent 2 days in a lecture theatre listening to all this confusing information about ‘digital telephony’ - meanwhile I was a long time FDM man.
How times have changed.
Cheers
Dallas
Dear Ian and Dallas,
I need to know one thing. do i need libss7 & asterisk-trunk to use ss7 or libpri and asterisk 1.4 do the job?
scream,
Let me preface my reply with “I don’t know”.
Having said that, if libss7 & libpri will co-exist I’d try that first. My reason for thinking this is that you probably still need a primary rate driver for the card as well as an ss7 driver for the signalling. (I may be completely wrong.)
I only used asterisk in a pure IP environment (SIP & IAX) as I pass off the PSTN gateway function to a traditional PBX/ACD via SIP trunks.
Ian may be more help.
Dallas
Hi Sceam.
If you want to setup your card using SS7 link, you have to install dahdi-trunk,dahdi-tools, libss7-trunk, asterisk-1.6.0.1.
You can download them on the digium web page.
I allready setup it successfully.
Please let me know if any more question.
Hi vu quoc viet,
i have configured Dahdi-linux, dahdi-tools, libss7, asterisk 1.6.x. i am using digium TE122 E1 card. after running command dahdi_scan i got
active=yes
alarms=OK
description=Wildcard TE122 Card 0
name=WCT1/0
manufacturer=Digium
devicetype=Wildcard TE122
location=PCI Bus 01 Slot 04
basechan=1
totchans=31
irq=233
type=digital-E1
syncsrc=0
lbo=0 db (CSU)/0-133 feet (DSX-1)
coding_opts=HDB3
framing_opts=CCS,CRC4
coding=HDB3
framing=CCS
and after running dahdi_cfg vvv i got
Channel 01: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 01)
Channel 02: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 02)
Channel 03: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 03)
Channel 04: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 04)
Channel 05: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 05)
Channel 06: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 06)
Channel 07: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 07)
Channel 08: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 08)
Channel 09: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 09)
Channel 10: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 10)
Channel 11: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 11)
Channel 12: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 12)
Channel 13: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 13)
Channel 14: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 14)
Channel 15: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 15)
Channel 16: D-channel (Default) (Echo Canceler: none) (Slaves: 16)
Channel 17: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 17)
Channel 18: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 18)
Channel 19: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 19)
Channel 20: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 20)
Channel 21: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 21)
Channel 22: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 22)
Channel 23: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 23)
Channel 24: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 24)
Channel 25: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 25)
Channel 26: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 26)
Channel 27: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 27)
Channel 28: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 28)
Channel 29: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 29)
Channel 30: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 30)
Channel 31: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 31)
when i connect with media gateway to transfer tdm calls, physical links shows up in bedia gateway but signalling link showing down. can you suggest anything?
when testing ss7 link i am getting “Segmentation Fault”. can anyone tell me how to debug
See http://www.asterisk.org/developers/bug-guidelines and also the file doc/backtrace.txt in the source tree.
First make sure that you don’t have mixed versions of your modules.
I’m assuming that it is asterisk that is suffering the fault.