I have connected my Asterisk to 2 analogue extensions on my Definity and I can make and receive calls via those extensions. Only trouble is, no callerid is displayed on the VOIP phones. Does the Definity use standard US or UK (I am based in the UK) CallerID and how can I get the Asterisk to pick it up and pass it to the phones ?
Thanks
It depends on the analog card and your version of Definity.
Older cards and software versions had no analog caller ID. Older cards in newer versions still won’t.
Definity analog cards that supply caller ID include:
TN793B
Definity analog cards that won’t have caller ID include:
TN2793
TN2793B
TN742
TN746
Depending on the platform (G3R, G3i, G3si, Prologix, etc) you may require system version 8.2 or higher. Even if you have a TN793B, it will work in a system 6.3 or higher, but will only supply dialtone, and calling abilities.
My Definity is running g3si ver 6
The analogue cards are TN468B and TN2183
I guess I’ll just have to put up caller unknown then.
Many thanks
Paulreg 
You might consider using an analog CO card for your connection. Look up the Multi Frequency signalling parameters of your Definity. You may be able to match the signalling method, and transmit (and receive) the ANI that way.
To address the extensions natively, just use AAR routing, and add the extensions resident on the Asterisk to a UDP table in the Definity.
The downside to this method is that you end up waiting for all the signalling to occur every time you setup a call. So call connects will appear slower on both sides. (Asterisk to Definity, and Definity to Asterisk) Right now, if you dial an Asterisk extension, ringing begins immediately. If you used trunk style signalling, you’d be waiting for the system to flash the trunk, send touch tone digits, etc… before you heard ring-back from the Asterisk platform.
Of course the easiest (and best) method would be to use an ISDN T1, or an E1 card (TN464). ANI passes easily over D-channels that way, and calls setup immediately, so the calls appear to be more like they’re happening “on the same system”. (But those cards may be beyond your budget.) If does, however, offer much more in the way of feature transparency if you use Q.931 signalling.
Hi All,
I have successfully connected my Asterisk Box to an Avaya Definity G3r release 11 using a Sangoma A102 T1/E1 card and its working fine. I can make calls from Asterisk to avaya and from Avaya to Asterisk.
The issue now is, I don’t get callerID from Avaya. I keep getting “unknow” as CallerID. What can be wrong?
I am using TN2646 DS1 card in Avaya. I hope this card is capable of sending CallerID from Avaya to Asterisk.
Cheers.
In an Avaya system you have to setup the trunk group to send caller ID.
On Page 2 of the trunk group translations, look for the parameter:
Send calling number: N
Change that to Y.
Then enter the command
change isdn public-unknown-numbering
On the form that comes up, put in the number prefixes for the extensions of your PBX.
Alternatively, you can change the DS1 parameters to correspond to a PBX implementation, but that might be harder.
thanks dufus, it worked great.
Cheers.
Hi All,
I have been battling with another Avaya Asterisk integration for over a month now. Its was the same G3r like I did before but with software release 8. I am using TN2464 on the Avaya side and a Sangoma A102D Card on the Asterisk Card. I have done everything on both Avaya and Asterisk to get it to work, but D-channel is not coming up.
Presently, the status of the sangoma card is “Connected” and the Avaya trunk is “Out-of-service-NE”. I ave changed from “line-side” to “pbx” to “network” on the Avaya side with no luck. What else do i do or what am I missing.
Very funny, because I have successfully done the same integration earlier with the same TN2464 and sangoma A102D.
What else do i need to do?
Cheers to All.
Hi All,
After 1 month of config manipulations and testings, I got “connected” on span 1 (or Port 1 on the card) when I do “wanrouter status” and have all alarms off with " wanpipemon -i w1g1 -c Ta" command except for RAI and open circuit. I have both on ON. How do I make both go OFF?
I always get the following error; (with whatever I do)
[Nov 29 09:22:48] DEBUG[6858] chan_zap.c: Failed to read gains: Invalid argument
[Nov 29 09:22:48] VERBOSE[6858] logger.c: — Registered channel 1, ISDN PRI signalling
[Nov 29 09:22:48] DEBUG[6858] chan_zap.c: Failed to read gains: Invalid argument
[Nov 29 09:22:48] VERBOSE[6858] logger.c: — Registered channel 2, ISDN PRI signalling
[Nov 29 09:22:48] DEBUG[6858] chan_zap.c: Failed to read gains: Invalid argument
[Nov 29 09:22:48] VERBOSE[6858] logger.c: — Registered channel 3, ISDN PRI signalling
e.t.c
Again, my D channel comes up and goes off with the following errors;
[Nov 29 09:28:48] VERBOSE[6872] logger.c: == Primary D-Channel on span 1 up
[Nov 29 09:28:48] WARNING[6872] chan_zap.c: PRI Error on span 0: We think we’re the CPE, but they think they’re the CPE too.
[Nov 29 09:28:49] WARNING[6872] chan_zap.c: PRI Error on span 0: We think we’re the CPE, but they think they’re the CPE too.
[Nov 29 09:28:50] WARNING[6872] chan_zap.c: PRI Error on span 0: We think we’re the CPE, but they think they’re the CPE too.
and again a D-Channel error as follows;
[Nov 29 09:28:51] VERBOSE[6872] logger.c: == Primary D-Channel on span 1 down
[Nov 29 09:28:51] WARNING[6872] chan_zap.c: No D-channels available! Using Primary channel 16 as D-channel anyway!
[Nov 29 09:28:51] VERBOSE[6872] logger.c: == Primary D-Channel on span 1 up
[Nov 29 09:28:51] WARNING[6872] chan_zap.c: PRI Error on span 0: We think we’re the CPE, but they think they’re the CPE too.
[Nov 29 09:28:52] VERBOSE[7460] logger.c: == Parsing ‘/etc/asterisk/manager.conf’: [Nov 29 09:28:52] VERBOSE[7460] logger.c: Found
[Nov 29 09:28:52] VERBOSE[7460] logger.c: == Parsing ‘/etc/asterisk/manager_additional.conf’: [Nov 29 09:28:52] VERBOSE[7460] logger.c: Found
[Nov 29 09:28:52] VERBOSE[7460] logger.c: == Parsing ‘/etc/asterisk/manager_custom.conf’: [Nov 29 09:28:52] VERBOSE[7460] logger.c: Found
[Nov 29 09:28:52] WARNING[7460] config.c: Unknown directive ‘permit=192.168.1.0/255.255.255.0’ at line 18 of /etc/asterisk/manager_custom.conf
[Nov 29 09:28:52] VERBOSE[7460] logger.c: == Manager ‘admin’ logged on from 127.0.0.1
[Nov 29 09:28:53] VERBOSE[7460] logger.c: == Manager ‘admin’ logged off from 127.0.0.1
[Nov 29 09:28:54] VERBOSE[6872] logger.c: == Primary D-Channel on span 1 down
[Nov 29 09:28:54] WARNING[6872] chan_zap.c: No D-channels available! Using Primary channel 16 as D-channel anyway!
so, I changed my Asterisk config to have zapata-auto.conf to read this;
;Sangoma A102 port 1 [slot:3 bus:15 span:1]
switchtype=euroisdn
context=from-zaptel
group=0
signalling=pri_net
channel =>1-15,17-31
;Sangoma A102 port 2 [slot:3 bus:15 span:2]
switchtype=euroisdn
context=from-zaptel
group=0
signalling=pri_net
channel =>32-46,48-62
and my /etc/zaptel.conf is as follows;
#Sangoma A102 port 1 [slot:3 bus:15 span:1]
span=1,0,0,ccs,hdb3
bchan=1-15,17-31
hardhdlc=16
#Sangoma A102 port 2 [slot:3 bus:15 span:2]
span=2,0,0,ccs,hdb3
bchan=32-46,48-62
hardhdlc=47
so, the CPE error stopped. But, I still get the “Failed to read gains” error on all the channels. and Avaya always give “Out-of-service-NE” with “display trunk” command. Avaya config is set to “Network”.
What do i do to resolve this errors?
Cheers to All.
Hi All,
I just got it resolved. Its turns out that the trasmit side from the Avaya amphenol cable is not properly terminated. I ran “test board” and found out that my port 138 failed. I re-terminates it and it passed.
I can now call from Asterisk to Avaya and Avaya to Asterisk seamlessly.
Cheers to all.