Re: [Asterisk-Users] Asterisk, QSIG and Tenovis PBX?
On 03/31/06 23:29 Jim Houser said the following: Looking at the TE100P I don't see it listed Q.SIG as supported. We have it working great as PRI. Am I wrong about the Q.SIG support? Q.SIG and the like are supported from libpri. we got it working with a TE410P, but i'm sure getting to work with the single span cards shouldnt be much different. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] Asterisk, QSIG and Tenovis PBX?
Hi, we are still trying to properly connect a Tenovis PBX to an Asterisk server (asterisk 1.2.6, libpri 1.2.2, zaptel 1.2.5, Digium Wildcard TE110P), this time with QSIG. Calling from a Tenovis phone to a SIP phone (i.e. traditional phone - Tenovis PBX - QSIG - Asterisk - SIP phone) works with the following messages: --- Don't know what to do if second ROSE component is of type 0x6 Don't know what to do if second ROSE component is of type 0x6 Don't know what to do if second ROSE component is of type 0x6 Don't know what to do if second ROSE component is of type 0x6 !! Unknown IE 49 (cs5, Unknown Information Element) !! Unknown IE 50 (cs5, Unknown Information Element) -- Accepting call from '1311' to '03' on channel 0/1, span 1 -- Executing Goto(Zap/1-1, default|8403|1) in new stack -- Goto (default,8403,1) -- Executing NoOp(Zap/1-1, 8403) in new stack -- Executing Dial(Zap/1-1, SIP/8403) in new stack -- Called 8403 -- SIP/8403-af88 is ringing -- SIP/8403-af88 is ringing -- SIP/8403-af88 is ringing -- SIP/8403-af88 answered Zap/1-1 == Spawn extension (default, 8403, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' --- However, the opposite way (i.e. SIP phone - Asterisk - QSIG - Tenovis PBX - traditional phone) doesn't work at all. I get the following messages: --- -- Executing Dial(SIP/8403-5b0f, Zap/g1/1311) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/1311 Mar 31 12:45:34 WARNING[23193]: chan_zap.c:7792 pri_fixup_principle: Call specified, but not found? Mar 31 12:45:34 WARNING[23193]: chan_zap.c:9046 pri_dchannel: Unable to move channel 1! Don't know what to do if second ROSE component is of type 0x6 XXX Invalid Progress indicator value received: 14 -- Zap/1-1 is ringing Don't know what to do if second ROSE component is of type 0x6 XXX Invalid Progress indicator value received: 14 -- Zap/1-1 answered SIP/8403-5b0f -- Hungup 'Zap/1-1' == Spawn extension (default, 1311, 1) exited non-zero on 'SIP/8403-5b0f' --- The called phone does NOT ring and I get some kind of busy tone on the SIP phone. As Asterisk says answered and the SIP phone counts the ellapsed time, it seems like the call has succeeded from the SIP phone's and Asterisk's perspective, i.e. the Tenovis PBX generates the busy tone?! The Tenovis service guy told me that I need to tell him the correct QSIG settings for the PBX: --- QBC QSIG B-Kanal zyklisch *QBS QSIG Leistungsmerkmale sperr. QBADI QSIG barr. suppl. serv. addit. indication QBANI QSIG barr. suppl. serv. ani QBCCC Rueckruf komplett sperren QBCFA senden (De)Aktivier. sperren *QBCFC RUL Pruefung sperren QBCFF senden RUL-Facility sperren QBCFL RWL spaetes Ausloesen sperren QBCHN QSIG Gebuehren Anforderung Netz *QBCHR Anfordern Gebuehren sperren QBCII Call Intrusion Invoke sperren QBCLI QSIG barr. suppl. serv. call linkage QBCMN QSIG barr. suppl. serv. CoMmon info extension QBCMS QSIG barr. suppl. serv. CoMmon info solic.serv. QBCMU QSIG barr. suppl. serv. CoMmon info unsolic.serv. QBCNF QSIG barr. suppl. serv. conference *QBCOI Anklopfen sperren QBCPI QSIG barr. suppl. serv. call park QBCPR QSIG barr. suppl. serv. call park retrieve QBCST QSIG barr. suppl. serv. csta *QBCTF senden Umlege-Facility sperr. QBCTM TLC line code QBDAS Sperren der Distinctive-Alerting Signalisierung QBDCH QSIG barr. suppl. d channel supervision QBDMI QSIG barr. suppl. serv. DSS module invoke QBDNW QSIG barr. suppl. serv. csta QBDSP QSIG barr. suppl. serv. display QBMMI SS minimail invoke barring QBMWI QSIG barr. suppl. serv. messg. wait. invoke QBNIA Namensanz. geruf. Tln sperren QBNIB Namensanz. bes. Tln sperren QBNIC Namensanz. verbu. Tln sperren QBNIO Namensanz. ruf. Tln sperren QBNMW NWR Message Waiting im Netzwerk sperren QBNWP QSIG netzweite Partner sperren QBPDI QSIG barr. suppl. serv. post dial info QBPRI Ersatzwege-Suche sperren *QBPUP QSIG barr. suppl. serv. pick-up QBRCI QSIG barr. suppl. serv. recall invoke QBRPE QSIG barr. suppl. serv. radio paging equip. QBSEA QSIG Dienstkennung erweiterte Adressierung QBSME QSIG Dienstkennung Herstel. Erweiterung QBSOM QSIG
Re: [Asterisk-Users] Asterisk, QSIG and Tenovis PBX?
Hi Johann, Johann Hanne [EMAIL PROTECTED] writes: Hi, we are still trying to properly connect a Tenovis PBX to an Asterisk server (asterisk 1.2.6, libpri 1.2.2, zaptel 1.2.5, Digium Wildcard TE110P), this time with QSIG. I tried with asterisk 1.2.2 against Alcactel 4400 on Monday. We had partially success. But at a specific config on the Alcatel side, the called number was not set by the SETUP message but via INFORMATION messages. Well, libpri doesn't like it this way. AFAIR, libpri does Q.SIG basic call, so you should set the Tenovis also to basic call. If this doesn't help, please run a pri debug span 1 while you make calls and post the output. My conclusion with Q.SIG: do not use it at this implementation level. YMMV. cu, Wolfgang ___ --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] Asterisk, QSIG and Tenovis PBX?
On 03/31/06 19:49 Wolfgang Zweimueller said the following: My conclusion with Q.SIG: do not use it at this implementation level. YMMV. i'll beg to differ. we've used Q.SIG successfully with an Ericsson MD110 for a customer in thailand. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] Asterisk, QSIG and Tenovis PBX?
Hello Dinesh I got a Panasonic KX-TDA100, can you tell me please how can you configure the PBX side? Qsig slave? master? and the other side of the asterisk? I got TE100P Regards, Daniel Dinesh Nair wrote: On 03/31/06 19:49 Wolfgang Zweimueller said the following: My conclusion with Q.SIG: do not use it at this implementation level. YMMV. i'll beg to differ. we've used Q.SIG successfully with an Ericsson MD110 for a customer in thailand. ___ --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] Asterisk, QSIG and Tenovis PBX?
Looking at the TE100P I don't see it listed Q.SIG as supported. We have it working great as PRI. Am I wrong about the Q.SIG support? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Sent: Friday, March 31, 2006 9:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk, QSIG and Tenovis PBX? Hello Dinesh I got a Panasonic KX-TDA100, can you tell me please how can you configure the PBX side? Qsig slave? master? and the other side of the asterisk? I got TE100P Regards, Daniel Dinesh Nair wrote: On 03/31/06 19:49 Wolfgang Zweimueller said the following: My conclusion with Q.SIG: do not use it at this implementation level. YMMV. i'll beg to differ. we've used Q.SIG successfully with an Ericsson MD110 for a customer in thailand. ___ --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 This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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] Asterisk, QSIG and Tenovis PBX?
Dinesh Nair [EMAIL PROTECTED] writes: On 03/31/06 19:49 Wolfgang Zweimueller said the following: My conclusion with Q.SIG: do not use it at this implementation level. YMMV. i'll beg to differ. we've used Q.SIG successfully with an Ericsson MD110 for a customer in thailand. Well, that's the YMMV. I have it also running with an Alcatel 4200. But my last experience with the 4400 showed me that there is something missing in the Q.SIG implementation. I also have seen some weird things with Q.SIG on BRI. And as long as I don't know what will happen when I connect * to some PBX, I won't tell my customers about Q.SIG. cu, Wolfgang ___ --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
