Hi Asif! BTW: Posting that is the list for any others who interested in audiolost problem What I did now: I interconnected two asterisks with IAX2 - so the first one comes with SS7 and the second does VoIP - and IAX2 between them - Number of audiolosts semms much decreased (though some exists) - now I'm quite sure that asterisk REQUIRES internal jitter management to prevent TDM desyncronization. So desync always happens while using SIP as interconnection/OR any connection with any other SIP device. That's is a particular solution. Though - Now I see that with approx 20 simultaneous calls asterisk eats 15+ % of CPU - Seems jitterbuffer/and/or/PLC overhead. I'm sure that code could be tuned/replaced with better realization which will not require such a huge CPU power. Anton. On 12 April 2006 09:38, asif uddin wrote: > Hi Anton, > Sorry for replying you so late. Actually i am out of > station. I am travelling to some places now. Now > regarding the chan_ss7 problem. As of my experience with > audio lost problem, we have solved this problem by > changing and modifying the application. We have not done > any modification in chan_ss7. Hope u r problem is > solved. If not yet solved , i suggest u to see it from > application and Dial Plan side.Because I am confident > about ss7 part. And if U have solved it, please let me > know how u have solved and what was the problem and why > is ithappening with chan_ss7. Anton , Do u have T1.111 > Standard. Because i dont have it. Can u send it to me.It > will be a great help for me. > > > > > With Warm regards, > Asif. > > Anton <anton.vazir@xxxxxxxxx> wrote: > Hi Asif, > > Thanks for invitation for Yahoo, but I use Linux LICQ and > not yahoo messenger, I'll try to set it up, but doubt > that linux version supports voice :) > > Regarding the chan_ss7 - there is a data, and if look at > the message it said "written 0 of 160" - so data exists. > And audially such messages heared as clicks on the > channels. > > I've tried to interconnect 2 asterisk boxes via SS7 and > it does not shgw any losts - so it means that happen only > when there is RTP transmission involved. > > Any ideas? > Anton. > > On 9 April 2006 19:08, asif uddin wrote: > > Hi Anton, > > Sorry for the late reply. It was a weekend so iwent > > out. I am not using chan_ss7 with any VoIP interface. I > > am using it directly with the ss7 interface ( i got an > > E1 from telco). See the problem what i feel is u r > > trying to send something on say CIC=29 from chan_ss7. U > > have allocated the channel successful;ly , but u r > > unable to play the voice file (i.e there is nothing to > > play on that channel) . Therefore in this case chan_ss7 > > (ss7_write) is trying to play the audio file , but > > there is no audio file .and i think it is going into > > infinite loop also. > > > > Asif > > > > > > > > Anton wrote: > > Hi Asif! > > > > Did you use chan_ss7 terminated/originated calls with > > VoIP? SIP/H323 729/723 codecs? I thing problem is > > somewhere in desycronization... > > > > On 8 April 2006 21:17, asif uddin wrote: > > > Hi Anton , > > > I have used chan_ss7 , both for incoming as well as > > > outgoing w/o any problem. > > > > > > But once i got the similar kind of problem which u > > > got and then I broke my head behind the ss7 code, but > > > finally when we have modified the application (dial > > > out) , the problem was solved and the application is > > > running perfectly. can u tell me ? application u r > > > using. so that we can proceed further . > > > > > > Asif > > > > > > Anton wrote: Hi Asif! Thanks for > > > your mail. > > > > > > I use it as a gateway to cellular operator. I looked > > > in the code - the message is returned by ss7_write > > > function, which writes to a zaptel file descriptor. > > > So it just unable to write there, since it's getting > > > E_AGAIN from unix write(fd,...) > > > > > > than I looked at the zaptel code. That stream is > > > received by zt_read() function. I did not dig further > > > yet. May be you can. > > > > > > Now I'm going to do another test - 2 servers > > > interconnected by 2xE1, and I'll pass 60 simultaneous > > > calls with audio. And will see what will happen than. > > > > > > Regards, > > > Anton > > > > > > On 8 April 2006 20:44, asif uddin wrote: > > > > Hi Anton, > > > > Thank u for sending the file, i think it is only > > > > isup part (chan_ss7.c) , is there any progress in > > > > mtp. Because i am good at mtp level , any way i > > > > will see the isup part also. Now regarding the > > > > Audio lost problem. I have faced the similar kind > > > > of problem . Let me tell u that there is no problem > > > > in the stack. Can u tell me ? application u r > > > > using. > > > > > > > > >Did anyone resolved the issue of unability to > > > > > write to zaptel FD? It's still an issue. My > > > > > latest LIVE test showed that it happens with any > > > > > codecs, when asterisk call another asterisk with > > > > > g711 the only requirements - that there must be > > > > > several on-going live calls. And as more calls is > > > > > on - that happens more and more often. Audialy > > > > > that heared as ticks in the sound. I do use > > > > > Sangoma A102 card. > > > > > > > > > >Apr 8 19:39:38 NOTICE[21974]: chan_ss7.c:1884 > > > > > ss7_write: Write buffer full >on CIC=29 (wrote > > > > > only 0 of 160), audio lost. Apr 8 19:39:38 > > > > > NOTICE[22014]: chan_ss7.c:1884 ss7_write: Write > > > > > buffer full >on CIC=11 (wrote only 0 of 160), > > > > > audio lost. > > > > > > > > Asif > > > > > > > > > > > > --------------------------------- > > > > Yahoo! Messenger with Voice. Make PC-to-Phone Calls > > > > to the US (and 30+ countries) for 2?/min or less. > > > > > > __________________________________________________ > > > Do You Yahoo!? > > > Tired of spam? Yahoo! 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