- This topic has 17 replies, 1 voice, and was last updated 13 years, 10 months ago by MikeM to Hasan Mahmud Riyad.
24th March 2006 at 10:36 #30817Bert-Guest
I need some help with my quintum
I’m unable to detect dialtone (test o 1 v) RJ11 is connected on pstn port 1.
i’m in France, so must i set a specific dialtone frequency ? If yes, How ?
When i try to make a call from a SIP phone, i can see in logs :
CH : 433608:[2: 1] sent message to cas: Setup
CAS : 433609:Received Message
CAS : 433609:mtype = 128
CAS : 433609:CAS: ISDN Message
CAS : 433609:[2,1] Process ISDN Message: imsg=0.
CAS : 433609:[2,1] Set Disc States
CAS : 433609:LocalDiscState = 0
CAS : 433609:RemoteDiscState = 0
CAS : 433609:In decode: message code = 0
CAS : 433609:[2,1] Decode SETUP
CAS : 433609:[2,1] casProc: idle-SETUP
CAS : 433609:[2,1] Send Seizure to Line
AB : 433609:[2,01]: t 1111
CAS : 433609:[2,1] Send ABCD = 15
CAS : 433609:[2,1] Start Timeout, ID = 1
CAS : 433609:[2,1] New State = 12
CAS : 434109:Timeout Event = 1
CAS : 434109:Stopped Timeout T = 1
CAS : 434109:[2,1] casProc: o1-T1
CAS : 434109:[2,1] No Responce to Seizure
CAS : 434109:RemoteDiscState = 0
CAS : 434109:[2,1] casProc: Disc Resp Proc
CAS : 434109:[2,1] Send Disc-Idle to Line
AB : 434109:[2,01]: t 0000
CAS : 434109:[2,1] Send ABCD = 0
CAS : 434109:[2,1] Start Delay, ID = 9
CAS : 434109:[2,1] New State = 20
CAS : 434189:Delay Event = 9
CAS : 434189:Stopped Delay D = 9
CAS : 434189:[2,1] casProc: x5x-D9
CAS : 434189:Sent Msg to Usid = 44
CAS : 434189:[2,1] Sent RELCOMP to CH
CAS : 434189:[2,1] New State = 0
CH : 434189:[2: 1] received message from cas: RelComp
CH : 434189:OBCSM: Release from peer=0xd82c04 cause=0x29 redir=.
CH : 434189:TBCSM: Release complete from peer=0xd8a80c.
CH : 434189:OBCSM: Trying another route.
CH : 434189:udp disconnect: 8 11
thank you for any help 🙂24th March 2006 at 15:53 #30818MikeM to BertGuest
Not sure if there is a big difference in the ring tone in France than any other place. One thing you can try is to connect the pstn port to the pbx port with a straight cable and run the test. This will at least say that the port is good or not.
Another thing to make sure is that there is no voice mail on the analog line that gives a stutter dial tone.
You did not mention the model type, but I am guessing from the log output and the command structure you gave for the test that it is a gen 1 unit (A400/A800) and not a Gen 2 product (AS/AX). There is no settings for different countries in the gen 1 product for the dial tone or type of analog line. There is a feature in gen 2 for line template and that may help.
Finally, you mentioned SIP. I am guessing that you downloaded the software from Quintum to support SIP on the gen 1 unit. I have not heard too much on this software yet, but it is beta and there may be some issues with the SIP side.24th March 2006 at 16:42 #30819Bert-Guest
Sorry for mistakes
Yes it is a gen 1 Analog Quintum A800
I call with a SIP phone, throught a nextone sswitch witch “translate” SIP to H.323.
I tried to link pstn and pbx : test is OK
i’m not sure if quintum call the good number (I suppose in dn:068196xxxx, dn means Dialed Number)
Another thing : i set ‘waitdialtone’ to 0 (no), so when i try to call with SIP phone, the nextone route well to my quintum, i can see the green pstn port led flashing , but the called phone doesn’t ring …
does A800 check for something before really call the “end phone” ?
Thank you for any help 🙂
Bert-.24th March 2006 at 17:00 #30820MikeM to BertGuest
If you have dialtonedetect set to Yes, then dial delay will do nothing. When the tenor receives a call from IP, it will route it to the first available pstn line (based on your hunt setting). It will go off-hook on the line and try to detect a standard dialtone. If it cannot detect this, then the call will be disconnected. You can try to disable dialtonedetect and set dialdelay for 500 (500ms) and see if that helps.29th March 2006 at 14:35 #30821Bert-Guest
dialtonedetect is set to 0 (no)
the issue is :
– if i run a test ( test o v), the dialtone is not detected.
– if i launch a voip call from my SIP phone, the quintum receive the request, with a valid num to dial. I see the little green light of the pstn port. i can wait 5 minuts, the phone called will never ring (and i hear nothing in the SIP phone).
The proper line is connected to the first pstn port, with a standard phone wire (when i plug a ‘real’ phone on this wire, i can hear dialtone and i can call who i want).
So i really don’t understand this issue, since apparently there is nothing else to configure.
But still no dialtone detected 🙁
i know i’m not a ‘Quintum Master’, but sincerely I really don’t know what i’m missing … 🙁
Thank anyone for any help 😉29th March 2006 at 15:44 #30822MikeM to BertGuest
It does not sound like you are missing anything here. For some reason, the tenor is just not detecting your dialtone. You said that you have dialtone detect set to 0, make sure you have dial delay set to 500 or 1000. Another test is to put a standard phone into the PBX port 1 of Tenor, set the pbxtg and pstntg passthru to 1 for yes and try a call from the phone to the pstn. Also, try putting the tenor in bypass. Finally, make sure that your phone line is set for dtmf and not rotary dial.
Other than that, the only way I can help is if I were to make some test calls from my unit to your unit and see what I hear as I may be able to tell about the dial tone, and other tones from my experience.
If you would like me to do this, please contact me at email@example.com.
Mike30th March 2006 at 08:51 #30823Bert-Guest
I found the cause of the issue : i used a standard phone wire to link pstn #1 and the phone catch. I tried with another wire from quintum one with a “double end”, and it works … 🙂30th March 2006 at 14:14 #30824MikeM to BertGuest
Glad to hear that you got that straightened out. I did see your IM yesterday as well, but I was quite busy.17th May 2006 at 10:24 #30825Bert-Guest
As I’m unable to send new message (it says that msg is ok and I just have to refresh, but in fact my msg never appeared), I post my question here :
About test cmd (in A800):
I know I have to make a test o. I want to use the line 5.
So I do that :
test o 5 v. -> dialtone detected.
test d 061210xxx
All digits sent.
And the called phone doesn’t ring.
But as I remember, I must have Received Digit ‘0’
Received Digit ‘6’
Received Digit ‘1’
All digit sent
Someone knows why I’m don’t have the Received digit ‘x’ plz ?
Another thing, what about if the pstn line is RNIS ??? does the quintum still detect tone ?17th May 2006 at 13:29 #30826Mikem to BertGuest
yes, there seems to be a problem with MSN last night. I think it is cleared up now.
As for your question, typically you would see digit received for each one. I cannot say why you are not seeing this for all the digits. Additionally, when you use the test o command, you do not need to put a v at the end.
For RNIS, I am sorry to say that I do not know what that means. If you could give me the description, I could tell you whether it is supported.
thanks18th May 2006 at 10:15 #30827Bert-Guest
Sorry RNIS is digital lines in France (ISDN).
About msg I can’t send, it is about this forum, not Msn 🙂
Well as I can see, in France, ISDN boxes provide analog lines via their ISDN network.
And I’m pretty sure my pb comes from that. Does Quintum able to use this kind of “analogic lines” ??18th May 2006 at 13:30 #30828MikeM to BertGuest
so we are talking about BRI lines in France, no the analog tenor does not support BRI lines, only analog, so you must provide a isdn terminal adapter to change the isdn to analog or you can purchase a Tenor BX unit which is a BRI supported Tenor.18th May 2006 at 13:40 #30829Bert-Guest
Well, I plugged the Quintum on a *real* analog line (as I don’t know about analog lines provided by an ISDN equipment).
I test and get dialtone. I have the correct num log (Received digit …)
But called phone still not ring.
So my question is can we use Quintum A800 on the Z* interfaces of a ISDN PSTN access ?
Is some specific settings (e.g. tone frequency) ?
Thanx for any help
Bert-18th May 2006 at 14:52 #30830MikeM to BertGuest
Again, you need to get an adapter to change the ISDN line to standard analog. If you have one of these adapters, they should provide you with 2 analog ports on the back side since each BRI line has 2 B channels for voice. If you have this, you should be able to connect the tenor to the analog port of the adapter and treat it like a standard analog line.
Now I do not know if there is any configuration that you will need to do on the adapter.
Mike29th June 2006 at 14:07 #30831Bert-Guest
Hmmm… Still unable to post new thread in this forum 🙁
So I’ll use this one.
I would like to know if there is a way to block caller ID on a quintum DX2060 ? If yes, how ?
thanx for any help!