Need Assistance With Asterisk/ Grandstream ATA Registration

I need some help. I have asterisk 18.11.3, and I have two call centric accounts that both registered to it just fine. I have a grandstream HT802 that with one callcentric account registered to it in port 1, but the other callcentric account that is registered to asterisk will not register in port 2. All credentials are identical on the ATA side. What could be going on?

what do you mean with

what error do you get when it try to registre

I don’t understand how an ITSP registering to an ATA has any relevance to Asterisk.

However the likely cause is that the peer is being identified by IP address, once registered.

When I did a trace log on the ata the error message was saying a number. 401 or 404. I am wondering if I had something thing configured wrong on extensions.conf. I can attach the trace log and the conf files if it helps

David551 if that was to be the case how would I correct it?

Here is what extensions.conf looks like, I am wondering if this might be the cause[from-callcentric]
exten => 17778379007,1,Progress()
same => n,Dial(PJSIP/DeskPhone1)
; same => n,Dial(PJSIP/DeskPhone1,m(ringback))
same => n,Goto(${DIALSTATUS},1)
exten => BUSY,1,PlayTones(busy)
same => n,Wait(60)
same => n,Hangup()
exten => CONGESTION,1,PlayTones(congestion)
same => n,Wait(60)
same => n,Hangup()
exten => CHANUNAVAIL,1,PlayTones(congestion)
same => n,Wait(60)
same => n,Hangup()
exten => _[A-Z].,1,PlayTones(congestion)
same => n,Wait(60)
same => n,Hangup()

exten => 17778756010,1,Progress()
same => n,Dial(PJSIP/DeskPhone2)
; same => n,Dial(PJSIP/DeskPhone2,m(ringback))
same => n,Goto(${DIALSTATUS},1)
exten => BUSY,1,PlayTones(busy)
same => n,Wait(60)
same => n,Hangup()
exten => CONGESTION,1,PlayTones(congestion)
same => n,Wait(60)
same => n,Hangup()
exten => CHANUNAVAIL,1,PlayTones(congestion)
same => n,Wait(60)
same => n,Hangup()
exten => _[A-Z].,1,PlayTones(congestion)
same => n,Wait(60)
same => n,Hangup()
exten => 01,1,Answer()
same => n,Background(custom/signal/dialtone)
exten => _9XX,1,Dial(PJSIP/${EXTEN}@callcentric)
exten => _9XX,1,Dial(PJSIP/${EXTEN}@callcentric2)
same => n,Hangup()
exten => _NXXXXXX,1,Goto(1252${EXTEN},1)
exten => _1NXXNXXXXXX,1,Dial(PJSIP/${EXTEN}@callcentric)
exten => _1NXXNXXXXXX,1,Dial(PJSIP/${EXTEN}@callcentric2)
same => n,Hangup()

You need to understand the issue more precisely. In the extreme case, you may need to use more than one ITSP.

As I said, I do not understand how Asterisk fits into this; with Asterisk and ATAs only Asterisk talks to the ATA. The ITSP talks to Asterisk and is completely unaware of the ATA.

I’m beginning to conclude that “callcentric” has nothing to do with callcentric. If it was really supposed to be grandstream, the relevant part of the Asterisk configuration is in pjsip.conf, not extensions.conf, with the reservation that ATAs cannot forward dialled digits, which suggests that, if these are Grandstream devices, they are not ATAs, or you are trying to use them as FXO gateways.

I don’t know how Grandstream gateways distinguish the analogue line to use when the caller, for both, has the same address. It could be on the details of the dialled number. It could be based on a port number on the Grandstream. It could be on a port number (requires multiple transports) on the caller. You therefore need a clear understanding of how the Grandstream handles this case.

I should have also added that I have the same set up with one other callcentric account on asterisk and was using the same grandstream ht802 which both registered. I will show you the pjsip.conf file as well

The HT802 is a pure ATA, so there is no way that it is going to be able to do anything useful with a PSTN number sent from the VoIP side.

401 is a request for password, and is normal.

404 means that the dialled number is not a valid number.

Looking at it probably uses the ATA’s SIP port number to distinguish between traffic for the two lines (Local SIP Port).

Even if i am already using the same ata in port 1 with another callcentric account and everything registered exactly the same?

Your configuration doesn’t make sense to me. If you want to connect Asterisk to callcentric, you do not use an analogue line in any form.

Please provide a detailed diagram of your telephony network.

We come back to the position that I see no role for Asterisk when using callcentric and an ATA.

Here is my current pjsip.conf file which is active

; -----------------------------------
; -----------------------------------






type = endpoint
disallow = all
allow = ulaw
rtp_symmetric = yes
force_rport = yes
rewrite_contact = yes
direct_media = no
inband_progress = yes
tos_audio = ef
device_state_busy_at = 1
trust_id_outbound = no
trust_id_inbound = no
notify_early_inuse_ringing = yes
context = from-internal ; context in the dialplan in which this user originates a call = yes
subscribe_context = phreaknet-hints

type = aor
max_contacts = 1
qualify_frequency = 30

type = auth

; every user needs to have an AOR (address of record) section, auth section, and endpoint section. To minimize clutter, we use templates for each.

username = DeskPhone1
password =

context = from-internal-pstn
callerid = “Gregg Carberry” <2526920234> ; change the CNAM and caller ID of your line here
media_encryption = sdes ; for SRTP (voice path), if your endpoint supports it. Make sure to connect to the TLS port so the signaling is encrypted, to>
auth = DeskPhone1
aors = DeskPhone1
;mailboxes = 2368@vmcontext ; for voicemail MWI

There is nothing in there that references the Grandstream. Diagram, please.

Asterisk is registering to callcentric, as one might expect, not callcentric to Asterisk.

You don’t have a callcentric2 endpoint, so the dialplan references to it will fail.

There are some strange things in the configuration, which may or may not be the result of your not using pre-formatted text mark up for the forum.

I would like to start another asterisk with a call centric account. I am using a nortel 616 ksu so at some point I will need ti use an ATA can someone tell me how to get this done?

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