[Asterisk-Users] Re: quadBRI in bri_net mode - t3 timer expired
* Paul Hewlett <[EMAIL PROTECTED]> wrote: > On Thursday 29 June 2006 20:08, Sebastian Kayser wrote: > > i successfully connected our old PBX to an asterisk server with a > > junghanns quadBRI, the quadBRI ports running in bri_cpe_ptmp mode > > connected to the interal PBX ISDN ports. > > > > Now i tried to turn it round as our PBX depends on it for some features > > and changed one of the quadBRI ports to bri_net signalling and connected > > it to one of the external PBX ISDN ports (how do you name that in telco > > jargon?). > >did you change the jumpers on the card ? Yes, i did =) Port 3 is jumpered as NT. asterisk*CLI> zap show status Description Alarms IRQbpviol CRC4 quadBRI PCI ISDN Card 1 Span 1 [TE] (ca· OK 0 0 0 quadBRI PCI ISDN Card 1 Span 2 [TE] (ca· OK 0 0 0 quadBRI PCI ISDN Card 1 Span 3 [NT] (ca· OK 0 0 0 quadBRI PCI ISDN Card 1 Span 4 [TE] (ca· OK 0 0 0 > > The card led goes green (indicates an ISDN link). Strange thing here is, the led stays green even if i unplug the cable ... - Sebastian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] quadBRI in bri_net mode - t3 timer expired
Hi all, i successfully connected our old PBX to an asterisk server with a junghanns quadBRI, the quadBRI ports running in bri_cpe_ptmp mode connected to the interal PBX ISDN ports. Now i tried to turn it round as our PBX depends on it for some features and changed one of the quadBRI ports to bri_net signalling and connected it to one of the external PBX ISDN ports (how do you name that in telco jargon?). The card led goes green (indicates an ISDN link), but nothing happens when i try to place a call from our PBX using the new connection (no incoming call on the *-console). Instead qozap complains every few seconds Jun 29 19:39:55 asterisk kernel: qozap: activating layer 1, span 3 Jun 29 19:39:58 asterisk kernel: qozap: t3 timer expired for span 3 Jun 29 19:39:58 asterisk kernel: qozap: not re-activating layer1 span 3 Jun 29 19:39:58 asterisk kernel: qozap: clearing alarms on span 3 What is it trying to tell me? My quadBRI doesn't do any powerfeeding, might that be a problem? - Sebastian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: E1 hardware for asterisk
* Christian Victor <[EMAIL PROTECTED]> wrote: > > Regarding echo cancel. Is there someone with hands-on experience > > regarding the echo canceller performance of the Junghanns E1 cards > > compared to for example the Sangoma ones? > > Well - the Junghanns does the echocancel in software and the Sangoma > A104d does it in hardware. Alright, i just had a look at their product lineup. It seems as not only the A104d but also the low end of their E1 cards (i.e. A101) comes with this onbard echo canceller (EDAC), right? > So on the Sangoma echo cancel does not affect CPU performance while on > the Junghanns it does. And in terms of quality? Does one of them perform noticable better than the other. - Sebastian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: E1 hardware for asterisk
* Christian Victor <[EMAIL PROTECTED]> wrote: > Tristan schrieb: > > What would you recommend ? > > > > Digium TE411P, Sangoma A104D, Eicon Diva Cards ? > > Ah - I should have read this bevor my last answer. ;-) > > I personally prefer the Sangoma E1 cards. The work in almost every PCI > system and the echo cancel - if you really need it - is far better than > the one provided by the Digium cards. Don't know about Eicons echo > cancel as I never used one. Regarding echo cancel. Is there someone with hands-on experience regarding the echo canceller performance of the Junghanns E1 cards compared to for example the Sangoma ones? http://www.junghanns.net/en/singleE1_produkt.html http://www.junghanns.net/en/doubleE1_produkt.html - Sebastian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Got reject for frame 0, but we only have others!
Hi all, what could be the cause for the following messages? May 22 12:44:53 WARNING[3791]: chan_zap.c:8498 zt_pri_error: 2 !! Got reject for frame 0, but we only have others! May 22 12:44:54 WARNING[3791]: chan_zap.c:8498 zt_pri_error: 2 !! Got reject for frame 0, but we only have others! May 22 12:44:55 WARNING[3791]: chan_zap.c:8498 zt_pri_error: 2 !! Got reject for frame 0, but we only have others! May 22 12:44:56 WARNING[3791]: chan_zap.c:8498 zt_pri_error: 2 !! Got reject for frame 0, but we only have others! May 22 12:44:57 WARNING[3791]: chan_zap.c:8498 zt_pri_error: 2 !! Got reject for frame 0, but we only have others! They flood my asterisk log since some days and i can't relate the date they started to a specific configuration change. I haven't yet noticed any bad side-effects, but i tend to feel better without any WARNINGs in my * log. I have already enabled "bri intense debug span 2", but i have to admit that i am not able to recognize any valuable pattern not to mention any sense at all :( (maybe that's because i am a systems administrator and not a telco guy). At least there is some kind of BRI traffic each time such a message is logged. The BRI debug can be found at http://skayser.de/mls/au/reject-bri-intense-debug.txt Maybe some of you are more capable to interpret those cryptic BRI messages. asterisk*CLI> show version Asterisk 1.2.7.1-BRIstuffed-0.3.0-PRE-1p Cheers - Sebastian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Ringing indication not working as expected
* Mimmus <[EMAIL PROTECTED]> wrote: > > Thanks for the advice. indications.conf is now existent and > > Asterisk is reloaded but the problem still persists. > Reloaded? Peraphs restarted is better... The reload messages informed about the re-reading of indications.conf. However, even restart doesn't change anything about the ringing indication problem. See my other reply for further debug information i have gathered. - Sebastian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Ringing indication not working as expected
* Sebastian Kayser <[EMAIL PROTECTED]> wrote: > * Sebastian Kayser <[EMAIL PROTECTED]> wrote: > > are there any caveats regarding ringing indication with Asterisk? > PSTN <-- 3 x BRI --> POTS (NEC) <-- 3 x BRI --> Asterisk > ^ ^ > | | > POTS telephone sets snom SIP phones > > With no options set in the Dial command, i.e. Dial(Zap/g2/,60), > the ringing behaviour is as follows. > > - snom -> snom - OK > - snom -> POTS telephone set - OK > - snom -> PSTN - NOK > > OK = ringing is signalled to the calling party as soon as Asterisk > indicates it on the console (... is ringing). > > NOK = no ringing is signalled to the calling party _although_ Asterisk > indicates it on the console. I gave "brig debug span 1" a try and it led to the following obvious difference (sorry for the long lines). snom -> POTS telephone set: 1 < Message type: ALERTING (1) 1 < [1 181 011 891 ] 1 < Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 1 <ChanSel: B1 channel 1 ] 1 < [1 1e1 021 811 811 ] 1 < Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) 1 < Ext: 1 Progress Description: Call is not end-to-end ISDN; further call progress information may be available inband. (1) ] snom -> PSTN telephone: 1 < Message type: ALERTING (1) 1 < [1 181 011 891 ] 1 < Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 1 <ChanSel: B1 channel 1 ] 1 < [1 1e1 021 821 881 ] 1 < Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) 1 < Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] So its: Location: Private network serving the local user Call is not end-to-end ISDN; further call progress information may be available inband. which works vs. Location: Public network serving the local user Progress Description: Inband information or appropriate pattern now available. which doesn't work. What's causing Asterisk to indicate ringing to the caller in the first place but not in the second place? Is this regular behaviour? Is there any way to also indicate ringing for snom -> PSTN telephone calls? - Sebastian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Ringing indication not working as expected
* Sebastian Kayser <[EMAIL PROTECTED]> wrote: > are there any caveats regarding ringing indication with Asterisk? > > I have got an asterisk installation with a quadBRI driven by BRIstuff. > Internal phones are various snoms (320 / 360) connected via SIP and > Idefisk softphones connected via IAX2. Outgoing calls are "routed" > through the Zap interfaces. > > When i set up the action for an external extension as > > Dial(Zap/g2/,60,R) > > or > > Dial(Zap/g2/,60) > > and initiate an outgoing call, Asterisk tells me that the called party > is ringing (Zap/4-1 is ringing) but there is no ringing indicated to the > calling party. No matter whether the calling party is a snom hardphone > or an idefisk softphone. I tried to narrow down the problem. My installation looks like this PSTN <-- 3 x BRI --> POTS (NEC) <-- 3 x BRI --> Asterisk ^ ^ | | POTS telephone sets snom SIP phones With no options set in the Dial command, i.e. Dial(Zap/g2/,60), the ringing behaviour is as follows. - snom -> snom - OK - snom -> POTS telephone set - OK - snom -> PSTN - NOK OK = ringing is signalled to the calling party as soon as Asterisk indicates it on the console (... is ringing). NOK = no ringing is signalled to the calling party _although_ Asterisk indicates it on the console. So although the Zap interface is used for both types of "external" calls (snom -> POST, snom -> PSTN) the ringing indication to my snoms fails for calls to the PSTN. Any ideas on how to further debug / troubleshoot this behaviour? - Sebastian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Ringing indication not working as expected
* Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote: > "R" is not a valid Dial option. Sure about that? My Asterisk installation lists it as a valid option. asterisk*CLI> show application Dial [...] R- indicate ringing to the calling party when the called party indicates [...] > "r" is the option you wanted. HOWEVER, if you are not hearing ringback, > "r" will almost never fixes the issue. "r" leads to the ringing being indicated right from the start. So if i call my cellphone from a SIP-connected snom, ringing is indicated to me immediately, whereas the cellphone starts to ring not until about 3-5 seconds later. =/ > Make sure you have a /etc/asterisk/indications.conf In some situations > if you do not have that file you will not hear ringback. Thanks for the advice. indications.conf is now existent and Asterisk is reloaded but the problem still persists. /etc/asterisk/indiciations.conf: [general] country=de [de] description = Germany ringcadance = 1000,4000 dial = 425 ring = 425/1000,0/4000 busy = 425/480,0/480 congestion = 425/480,0/480 callwaiting = 425/2000,0/6000 dialrecall = 425/500,0/500,425/500,0/500,425/500,0/500,1600/100,0/900 record = 1400/500,0/15000 info = 950/330,0/200,1400/330,0/200,1800/330,0/1000 - Sebastian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ringing indication not working as expected
Hi all, are there any caveats regarding ringing indication with Asterisk? I have got an asterisk installation with a quadBRI driven by BRIstuff. Internal phones are various snoms (320 / 360) connected via SIP and Idefisk softphones connected via IAX2. Outgoing calls are "routed" through the Zap interfaces. When i set up the action for an external extension as Dial(Zap/g2/,60,R) or Dial(Zap/g2/,60) and initiate an outgoing call, Asterisk tells me that the called party is ringing (Zap/4-1 is ringing) but there is no ringing indicated to the calling party. No matter whether the calling party is a snom hardphone or an idefisk softphone. Am i missing something? asterisk*CLI> show version Asterisk 1.2.7.1-BRIstuffed-0.3.0-PRE-1p - Sebastian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
