I have latest asterisk running on redhat 9; I use mediatrix gateways running SIP protocol. I have installed hisax compatible passive adapter on my asterisk box (HFC-S PCI Active chip). Problem is following; when I dial through my ISDN adapter and run echo test I got excellent response (clear sound, no breaks), when I connect my SIP gateways between each other users hear each other perfectly, no jitters, errors or breaks. But; when I try to call from ISDN to SIP gateway I can hear perfectly what is said to me from SIP side, but my voice "recorded" on isdn adapters appears jittered or broken to the other party, and if I speak to loud it is cut completely. I use ulaw/alaw on my SIP gateways. this is my modem.conf file (this is channel one, I have one more running); msn=340 driver=i4l type=autodetect incomingmsn=340 device => /dev/ttyI0 this is my sip.conf file (sample; I have 7 more identical ports); [201] type=friend username=201 host=dynamic defaultip=192.168.3.210 dtmfmode=inband any clues, ideas what to check? p.s. also, when SIP user calls my ttyI0 then I do not hear ringing tone ---- I don't believe you can do today's job with yesterday's methods and be in business tomorrow. mailto:marko@printel.hr http://printel.hr
one more thing I have just configured so that I enter asterisk through ttyI0 and then exit back to PSTN (or in my case ISDN) thru ttyI1 (second B channel on the same adapter) and zero problems, sound is perfect, no jittering, breaks or any problem whatsoever so something happens in between asetrisk box and my SIP gateway and I really do not have a clue what ---- Sometimes you're the bug, sometimes you're the windshield. mailto:marko@printel.hr http://printel.hr -----Original Message----- From: Marko Rakar Sent: Monday, March 22, 2004 4:58 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] jittered voice over hisax passive card I have latest asterisk running on redhat 9; I use mediatrix gateways running SIP protocol. I have installed hisax compatible passive adapter on my asterisk box (HFC-S PCI Active chip). Problem is following; when I dial through my ISDN adapter and run echo test I got excellent response (clear sound, no breaks), when I connect my SIP gateways between each other users hear each other perfectly, no jitters, errors or breaks. But; when I try to call from ISDN to SIP gateway I can hear perfectly what is said to me from SIP side, but my voice "recorded" on isdn adapters appears jittered or broken to the other party, and if I speak to loud it is cut completely. I use ulaw/alaw on my SIP gateways. this is my modem.conf file (this is channel one, I have one more running); msn=340 driver=i4l type=autodetect incomingmsn=340 device => /dev/ttyI0 this is my sip.conf file (sample; I have 7 more identical ports); [201] type=friend username=201 host=dynamic defaultip=192.168.3.210 dtmfmode=inband any clues, ideas what to check? p.s. also, when SIP user calls my ttyI0 then I do not hear ringing tone
Rich Adamson
2004-Mar-22 09:40 UTC
[Asterisk-Users] jittered voice over hisax passive card
> I use mediatrix gateways running SIP protocol. > > I have installed hisax compatible passive adapter on my asterisk box > (HFC-S PCI Active chip). > > Problem is following; when I dial through my ISDN adapter and run echo > test I got excellent response (clear sound, no breaks), when I connect > my SIP gateways between each other users hear each other perfectly, no > jitters, errors or breaks. > > But; when I try to call from ISDN to SIP gateway I can hear perfectly > what is said to me from SIP side, but my voice "recorded" on isdn > adapters appears jittered or broken to the other party, and if I speak > to loud it is cut completely.There is an option in the Mediatrix called Voice Detection (or something like that) that is set to Auto. Turn that "off". The problem relates to asterisk needs a constant flow of rtp traffic (not just traffic when you are speaking), and with the voice detection feature turned on, asterisk does not get that constant flow. Rich
Hi,
I have a problem with a Eicon Diva Server 4 BRI. I have 4 BRI ISDN and
11 number for these 4 ISDN. At first I have connected one of these 4 ISDN.
When I try to call I receive the next trace:
-- Executing ChanIsAvail("SIP/716-b0cd",
"CAPI/971844367&CAPI/971846015&CAPI/971846034&CAPI/971846036&CAPI/971846094&
CAPI/971846141&CAPI/971846142&CAPI/971846143&CAPI/971846146&CAPI/971846147&C
API/971846148") in new stack
-- data = 971844367
-- capi request omsn = 971844367
== found capi with omsn = 971844367
-- CAPI Hangingup
-- Executing SubString("SIP/716-b0cd",
"CANAL=CAPI[contr1/971844367]/0|12|9") in new stack
Mar 22 17:51:00 WARNING[262161]: app_substring.c:63 substring_exec: The use
of Substring application is deprecated. Please use ${variable:a:b} instead
-- Executing Dial("SIP/716-b0cd",
"CAPI/@971844367:687754642|17") in new
stack
-- data = @971844367:687754642
-- capi request omsn = @971844367
== found capi with omsn = 971844367
== CAPI Call CAPI[contr1/971844367]/1 == CAPI Call
CAPI[contr1/971844367]/1 -- creating pipe for PLCI=-1
> sent CONNECT_REQ MN =0x7
-- Called @971844367:687754642
-- CONNECT_CONF ID=001 #0x0007 LEN=0014
Controller/PLCI/NCCI = 0x301
Info = 0x0
== received CONNECT_CONF PLCI = 0x301 INFO = 0
== DISCONNECT_IND PLCI=0x301 REASON=0x3301
> sent DISCONNECT_RESP PLCI=0x301
-- CAPI Hangingup
-- removed pipe for PLCI = 0x301
== No one is available to answer at this time
My extensions.conf is the next:
exten=>_XXXXXXXXX,1,ChanIsAvail(CAPI/971844367&CAPI/971846015&CAPI/971846034
&CAPI/971846036&CAPI/971846094&CAPI/971846141&CAPI/971846142&CAPI/
971846143&CAPI/971846146&CAPI/971846147&CAPI/971846148)
;exten=>_XXXXXXXXX,1,ChanIsAvail(CAPI/971844367)
exten=>_XXXXXXXXX,2,SubString,CANAL=${AVAILCHAN}|12|9
exten=>_XXXXXXXXX,3,Dial(CAPI/@${CANAL}:${EXTEN}|17)
exten=>_XXXXXXXXX,104,Playback(invalid)
exten=>_XXXXXXXXX,105,Hangup()
and my capi.conf is the next:
[global]
mode=immediate
isdnmode=multipoint
txgain=0.8
rxgain=0.8
[interfaces]
msn=971844367,971846015,971846034,971846036,971846094,971846141,971846142,97
1846143,971846146,971846147,971846148
;msn=971844367
incomingmsn=*
controller=1
context=default
echocancel=1
echotail=64
devices=8
any idea?
srsergio
well I have found out this setting on my mediatrix unit but it did not solve my problem (I have solved many RTP warnings in asterisk command prompt though), since I do not have any other codecs to play with I have ordered single g729 licence and will play with that (if that solves my problem, although I think this is somehow hardware related problem) any other suggestions? ---- Sometimes you're the bug, sometimes you're the windshield. mailto:marko@printel.hr http://printel.hr -----Original Message----- From: Rich Adamson [mailto:radamson@routers.com] Sent: Monday, March 22, 2004 5:40 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] jittered voice over hisax passive card> I use mediatrix gateways running SIP protocol. > > I have installed hisax compatible passive adapter on my asterisk box > (HFC-S PCI Active chip). > > Problem is following; when I dial through my ISDN adapter and run echo> test I got excellent response (clear sound, no breaks), when I connect> my SIP gateways between each other users hear each other perfectly, no> jitters, errors or breaks. > > But; when I try to call from ISDN to SIP gateway I can hear perfectly > what is said to me from SIP side, but my voice "recorded" on isdn > adapters appears jittered or broken to the other party, and if I speak> to loud it is cut completely.There is an option in the Mediatrix called Voice Detection (or something like that) that is set to Auto. Turn that "off". The problem relates to asterisk needs a constant flow of rtp traffic (not just traffic when you are speaking), and with the voice detection feature turned on, asterisk does not get that constant flow. Rich _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
this is now getting interesting; when I do echo test from my mediatrix unit to asterisk it works correctly when I do echo test from my isdn4linux adapter it also works correctly when I connect two mediatrix units through asterisk they work correctly when I connect my isdn4linux adapter to public ISDN network it also runs fine but when I try to call from isdn4linux passive adapter to mediatrix then voice comming from mdiatrix is clear while my voice from isdn adapter to mediatrix is broken, cut off or completely garbgled I am completely baffled by this ---- Sometimes you're the bug, sometimes you're the windshield. mailto:marko@printel.hr http://printel.hr -----Original Message----- From: Marko Rakar Sent: Tuesday, March 23, 2004 9:51 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] jittered voice over hisax passive card well I have found out this setting on my mediatrix unit but it did not solve my problem (I have solved many RTP warnings in asterisk command prompt though), since I do not have any other codecs to play with I have ordered single g729 licence and will play with that (if that solves my problem, although I think this is somehow hardware related problem) any other suggestions? ---- Sometimes you're the bug, sometimes you're the windshield. mailto:marko@printel.hr http://printel.hr -----Original Message----- From: Rich Adamson [mailto:radamson@routers.com] Sent: Monday, March 22, 2004 5:40 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] jittered voice over hisax passive card> I use mediatrix gateways running SIP protocol. > > I have installed hisax compatible passive adapter on my asterisk box > (HFC-S PCI Active chip). > > Problem is following; when I dial through my ISDN adapter and run echo> test I got excellent response (clear sound, no breaks), when I connect> my SIP gateways between each other users hear each other perfectly, no> jitters, errors or breaks. > > But; when I try to call from ISDN to SIP gateway I can hear perfectly > what is said to me from SIP side, but my voice "recorded" on isdn > adapters appears jittered or broken to the other party, and if I speak> to loud it is cut completely.There is an option in the Mediatrix called Voice Detection (or something like that) that is set to Auto. Turn that "off". The problem relates to asterisk needs a constant flow of rtp traffic (not just traffic when you are speaking), and with the voice detection feature turned on, asterisk does not get that constant flow. Rich _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Jens P. Hansen
2004-Mar-24 14:44 UTC
[Asterisk-Users] jittered voice over hisax passive card
Marko Rakar wrote:>this is now getting interesting; > >when I do echo test from my mediatrix unit to asterisk it works >correctly >when I do echo test from my isdn4linux adapter it also works correctly >when I connect two mediatrix units through asterisk they work correctly >when I connect my isdn4linux adapter to public ISDN network it also runs >fine > >but when I try to call from isdn4linux passive adapter to mediatrix then >voice comming from mdiatrix is clear while my voice from isdn adapter to >mediatrix is broken, cut off or completely garbgled > >I am completely baffled by this > >Throw out I4L and the hisax drivers and use the zaphfc (found at http://www.junghanns.net), by using that you simply specify your faxboard as a zaptel device which makes the configuration much easier -and due to the excellent driver by Junghanns your jitter will disapear as mist in the sun. K.Rgds Jens,