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× Questions about D110/210/410 T1/E1/J1 PRI Cards

Dahdi without ringing tone

14 years 11 months ago #3260 by pemer
Hi there,

we have a asterisk 1.6 and the dahdi channel drivers and a secound PBX from alcatel.
we have a problem with the ringing tone in the sip channel. we didn't hear anyone.

If we comment out in the config file /etc/asterisk/moduels the line with the Dahdi channel dirver, the sip channel playes an ring tone.
Someone knows the problem?

thx in advance for any answear.
14 years 11 months ago #3261 by james.zhu
hello:
please give a picture how your system works, does your second pbx connects with asterisk through SIP or PRI?
do you use dahdi channels? if so, what is real problem with SIP? what is the firewall, Suse and logs? please give more info.
james.zhu

14 years 11 months ago #3267 by pemer
Hi,

the hardware constelation is in the attachted file Tel-plan.jpeg.
Please look at the file.

So the Problem is the ring tone in the SIP channel.
If a call from Alcatel system telephone to a sip-phone is initialised, the alcatel system telephone processing a ring tone.
If a sip phone is processing a call to a sip phone or a alcatel system phone, the sip phone didn't process a ring tone.
Now, if i am disabling the Asterisk module chan_dahdi.so, the call from sip to sip is playing a ring tone, without chan_dahdi.so is the alcatel system phone not accessible from SIP.

So, i could not understand what is the problem in asterisk of the cbannel driver dahdi.

thx in advance for a answear

i] Last edited by pemer at 2009-5-27 16:01 [/i
Attachments:
14 years 11 months ago #3270 by james.zhu

Original posted by pemer at 2009-5-27 15:59


Hi,

the hardware constelation is in the attachted file Tel-plan.jpeg.
Please look at the file.

So the Problem is the ring tone in the SIP channel.
If a call from Alcatel system telephone to a sip-p ...


hello:
I try to give a answer. if you can not make a sip-to-sip call in asterisk, i think you have something wrong in your dialplan. for sip ring tone, you have to make sure the call
route reaches the sip and set your sip.conf correctly. please give logs for that or give me ssh for that!
regards!
James.zhu

14 years 11 months ago #3273 by pemer
Hi James,

the Problem is not the dialplan. from SIP to SIP working the dialing but not the inband ringtone of the call.
So if you did a call with a SIP Phone from our Asterisk to any other SIP Phone, the asterisk pbx did't progress a ringing tone. But the call still working.
So no ring tone in SIP.

And if i am disableing the dahdi channel driver from asterisk, the ring tone working.

The Dialplan is not the Problem.

SSH Account - Please send me your ssh-public key per email. You can create it with
ssh-keygen and is in your folder ~.ssh/id_rsa.pub

regards
Patrick
14 years 11 months ago #3275 by james.zhu
hello,
i am still confusing the setting for your system, the pri line is working or not? if it works, the hardware and channels should not be a problem. i do not think the chan_dahdi will effect the sip to sip calls in asterisk. it is totally different things. you must check the sip setting(dtmfmode) and your dialplan or even the version of asterisk and dahdi.
james.zhu

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