Re: [Asterisk-Users] PRI got event: HDLC Bad FCS (8) on PrimaryD-channel of span
The Problem occurs only on the second card in the System, so i really have no idea.Thanks for your helpNico-- -Ursprüngliche Nachricht-Von: Anthony Rodgers <[EMAIL PROTECTED]>Gesendet: Friday, 28. Apr 2006 0:24 +0200An: "Asterisk Users Mailing List - Non-Commercial Discussion" Betreff: Re: [Asterisk-Users] PRI got event: HDLC Bad FCS (8) on PrimaryD-channel of spanLooks like a timing problem - zaptel.conf and zapata.conf, please. A. On Apr 25, 2006, at 3:05 AM, Nico Giefing wrote: > > Hello, > > I get an Error every minute on the second card of two installed TE410P > Cards in our System. > > The error is: > PRI got event. HDLC Abort (6) on Primary D-channel of span 5(-8) > PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 5(-8) > > Is it possible that there are known problems with 2 cards in one > system? > > I'm running Asterisk/Libpri/zaptel from SVN branch-1.2-16008 > > I was running Debian Stable with Kernel 2.4.25 > > Since Yesterday i'm running Kernel 2.6.8 > > The Interrupte of the cards are: 16 and 28 > > > Do anybody have any idea how i can solve this Problem? > > > > > > -- > > > --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 --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 zapata.conf Description: Zip archive zaptel.conf Description: Zip archive ___ --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] PRI got event: HDLC Bad FCS (8) on Primary D-channel of span
Hello,I get an Error every minute on the second card of two installed TE410P Cards in our System.The error is: PRI got event. HDLC Abort (6) on Primary D-channel of span 5(-8)PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 5(-8)Is it possible that there are known problems with 2 cards in one system?I'm running Asterisk/Libpri/zaptel from SVN branch-1.2-16008I was running Debian Stable with Kernel 2.4.25Since Yesterday i'm running Kernel 2.6.8The Interrupte of the cards are: 16 and 28Do anybody have any idea how i can solve this Problem? -- ___ --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
Re: [Asterisk-Users] ISDN and Analog DIAL UP Connection Through Asterisk and Digium TE405P
We tried to give a MAX4000 behind a Asterisk with TE 405, but the connection is very slow (max of 28.8) and we have also a problem with a Fax Server behind the Asterisk, We loose lines and so on.Did anyonehave an idea?ThanksNico-- -Ursprüngliche Nachricht-Von: Don Pobanz <[EMAIL PROTECTED]>Gesendet: Tuesday, 28. Mar 2006 19:19 +0200An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] ISDN and Analog DIAL UP Connection Through Asterisk and Digium TE405PNico Giefing wrote: > > Is it possible to establish a ISDN DIAL up Connection and Analog Dial up > Connection (V90) trough asterisk with Digium TE405? I do the v90 dial up. The modem is connected to an Adtran 750 channel bank. Our DID trunks are on a T1 line to the phone company. If you have analog lines to the phone company it will not work since only 1 A/D conversion is allowed! We aren't doing any IDSN. It ?may? be possible. Don Pobanz > Nico Giefing > --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___ --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] ISDN and Analog DIAL UP Connection Through Asterisk and Digium TE405P
Did anybody know, Is it possible to establish a ISDN DIAL up Connection and Analog Dial up Connection (V90) trough asterisk with Digium TE405? Thanks a lot for help. Nico Giefing -- ___ --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
Re: [Asterisk-Users] Ring more than two isdn phones simultaneously
how many connection do you have from your asterisk to the old pbx? i think on 1 ISDN connection its only possible to let 2 phones ring, because 1 ISDN 2 channels... Nico - Original Message - From: "Arik Funke" <[EMAIL PROTECTED]> To: Sent: Saturday, August 20, 2005 7:44 PM Subject: [Asterisk-Users] Ring more than two isdn phones simultaneously > I am using a HFC-S card in nt mode with zaphfc driver to connect an > internal isdn bus. I would like to signal an incoming call on, let's > say, 4 phones. Right now I use: > > Dial(Zap/g1/21&Zap/g1/22&Zap/g1/24&Zap/g1/23&Zap/g1/29,,t) > > where g1 are my two isdn channels provided by HFC-S card an the > 21,22,etc my internal numbers. > > When the command is executed however, only the first two specified > phones ring. Etc. with the first channel 21 ist called, with the second > 22. How can I get asterisk to signal to all phones with just one isdn > channel? I am trying to duplicate the setup I had with my old isdn pbx > with did above trick just fine... Maybe somebody can help me configure > asterisk appropriately? > > Cheers, > Arik > > > PS: I gave following a try but without success: > Dial(Zap/g1/21-29,,t) > Dial(Zap/g1/21+29,,t) > ___ > 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
Re: [Asterisk-Users] [OT] Looking for Web based SIP endpoint
you need a sip-provider? - Original Message - From: "Bruce Ferrell" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Saturday, August 20, 2005 12:38 AM Subject: [Asterisk-Users] [OT] Looking for Web based SIP endpoint > I think the title more or less says it all. > > Is there any such animal? > > TIA > ___ > 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
Re: [Asterisk-Users] CVS-HEAD Compile Problem
but the non head version is not working with realtime configuration? hm, i think its a problem with app_expr.c but i will try now to copy the app_expr.c from cvs-version i will let you know Nico - Original Message - From: "Trey Scarborough" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, August 19, 2005 8:31 PM Subject: Re: [Asterisk-Users] CVS-HEAD Compile Problem > I ran into the same problem the other day and just went back to non head > version It would be nice to figure out why it does this. > > - Original Message - > From: Nico Giefing > To: Asterisk Users Mailing List - Non-Commercial Discussion > Sent: Friday, August 19, 2005 9:20 AM > Subject: [Asterisk-Users] CVS-HEAD Compile Problem > > > I have a little Problem, > > I will compile asterisk CVS-HEAD but after 20 second of compiling i get the > message as shown at http://pastebin.com/340654 about 1000 times. > > Do anybody know a solution for this? > > Thanks a lot > > Nico > > > > ___ > 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
[Asterisk-Users] Overlap digits...
Hello, I'm again there I have also a Problem with Overlap Digits... I'm getting a Call from my Telco to the extension 1234 and i will forward it with exten => 1234,1,Dial(Zap/g1/987654), but asterisk is not dialing 987654, asterisk is dialing 987654 and as overlap digits 1234. so i see on the CDR's from my telco as dialed number 9876541234 and thats not what i want. Do anybody know a solution for this? Nico ___ 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] CVS-HEAD Compile Problem
I have a little Problem, I will compile asterisk CVS-HEAD but after 20 second of compiling i get the message as shown at http://pastebin.com/340654 about 1000 times. Do anybody know a solution for this? Thanks a lot Nico ___ 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
Re: [Asterisk-Users] Compilation Problem with asterisk-addons
- Original Message - From: "Juan Luis Moyano" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, June 21, 2005 6:09 AM Subject: Re: [Asterisk-Users] Compilation Problem with asterisk-addons > On Lun, 20 de Junio de 2005, 6:49 pm, Nico Giefing dijo: > > Hello, i have a little Problem with compiling asterisk-addons > > > > > > the failure is: > > > > app_addon_sql_mysql.c:164:64: macro "AST_LIST_REMOVE" requires 4 > > arguments, but only 3 given > > app_addon_sql_mysql.c: In function `del_identifier': > > app_addon_sql_mysql.c:164: error: ÀSR_LIST_REMOVE' undecalred (first use > > in this function) > > app_addon_sql_mysql.c:164: error: (Each undeclared identifier is reported > > only once > > app_addon_sql_mysql.c:164: error: for each function it appears in.) > > > > > > Does anybody know anything about this problem? > > > > > > Thank you for your help. > > > > > > Nico > > ___ > > 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 > > Nico, I'm having the same issue while compiling asterisk-addons. Here I > post the error I get: > > app_addon_sql_mysql.c:164:64: macro "AST_LIST_REMOVE" requires 4 > arguments, but only 3 given > app_addon_sql_mysql.c: In function `del_identifier': > app_addon_sql_mysql.c:164: error: `AST_LIST_REMOVE' undeclared (first use > in this function) > app_addon_sql_mysql.c:164: error: (Each undeclared identifier is reported > only once > app_addon_sql_mysql.c:164: error: for each function it appears in.) > make: *** [app_addon_sql_mysql.o] Error 1 > > Also it's important to mention that I'm running asterisk-1.0.7 compiled > from the ebuild on a Gentoo (kernel 2.6), also I've merged latest mysql, > perl and DBD-mysql. I don't know what is the best way to compile > asterisk-addons on a gentoo system so if someone had accomplished this, > please let me know. Thanks in advance. > > -- > Juan Luis Moyano > [EMAIL PROTECTED] > > ___ > 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 > Hi, my configuration is debian stable (3.1) with 2.6 kernel and the cvs version of asterisk and asterisk-addons. does anybody know a solution for this problem? Nico ___ 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] Compilation Problem with asterisk-addons
Hello, i have a little Problem with compiling asterisk-addons the failure is: app_addon_sql_mysql.c:164:64: macro "AST_LIST_REMOVE" requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: error: ÀSR_LIST_REMOVE' undecalred (first use in this function) app_addon_sql_mysql.c:164: error: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:164: error: for each function it appears in.) Does anybody know anything about this problem? Thank you for your help. Nico ___ 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
Re: [Asterisk-Users] Caller ID on TelaSIP SIP Channel
is there no number showed on your mobile phone only or does asterisk rewrite the cli with nothing? Do you have your own connection to POTS? Nico - Original Message - From: "Doug Eubanks" <[EMAIL PROTECTED]> To: Sent: Wednesday, June 15, 2005 10:45 PM Subject: Re: [Asterisk-Users] Caller ID on TelaSIP SIP Channel > I tried it now that you mentioned it > > SetCallerPres(allowed_passed_screen) > > It had no effect... > > Doug > > > do you set anything like callingpres? > > if not, try this parameters the documentation you will find on > > www.voip-info.org > > > > > > Nico > > > > - Original Message - > > From: "Doug Eubanks" <[EMAIL PROTECTED]> > > To: > > Sent: Wednesday, June 15, 2005 10:18 PM > > Subject: [Asterisk-Users] Caller ID on TelaSIP SIP Channel > > > > > >> > >> > >> I can't seem to get consistant outbound caller ID working correctly. I > >> have set the fromuser and callerid field in my sip.conf for my TelaSIP > >> peer, but half the time it shows up as "No Caller ID" on my cell phone, > >> other times it shows it correctly. > >> > >> Using asterisk CVS. Any ideas? > >> > >> Doug > >> > >> > >> ___ > >> 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 > ___ 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
Re: [Asterisk-Users] Caller ID on TelaSIP SIP Channel
do you set anything like callingpres? if not, try this parameters the documentation you will find on www.voip-info.org Nico - Original Message - From: "Doug Eubanks" <[EMAIL PROTECTED]> To: Sent: Wednesday, June 15, 2005 10:18 PM Subject: [Asterisk-Users] Caller ID on TelaSIP SIP Channel > > > I can't seem to get consistant outbound caller ID working correctly. I > have set the fromuser and callerid field in my sip.conf for my TelaSIP > peer, but half the time it shows up as "No Caller ID" on my cell phone, > other times it shows it correctly. > > Using asterisk CVS. Any ideas? > > Doug > > > ___ > 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
[Asterisk-Users] Problem with overlap dialing
Hello, I have a Problem with overlap dialing. If i get an Call to extension xyz wit overlap digits and asterisk have to forward this extension to another PRI-Port Asterisk dial the number as in extension.conf described (Zap/g1/abcdefg) i get this plus the complete extension as overlap digit to the outgoing PBX. Did anyone know, is there a command to kill all overlap digits on this extension? Thanks a lot Nico ___ 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
