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 Dienstkennung andere Herstel. APDUs
QBSTC QSIG Dienstken. fuer Pruef. Uebertrg. Zaehler
QBSYI QSIG barr. suppl. serv. synchron. invoke
QBTDL QSIG barr. suppl. serv. terminal download
QBUUS QSIG barr. suppl. serv. user to user
QBZAV QSIG barr. suppl. serv. zarvt
QCA Gespraechsaufnahmemeldung QUe
QCM QSIQ mit ACM
QDT QSIG Ueberwachung Wahlzeitablauf
QEI QSIG ETSI-ISO Anpassung
QF0 QSIG Reserve Erweiterung 0
QF1 QSIG Reserve Erweiterung 1
QF2 QSIG Reserve Erweiterung 2
QF3 QSIG Reserve Erweiterung 3
QF4 QSIG Reserve Erweiterung 4
QF5 QSIG Reserve Erweiterung 5
QF6 QSIG Reserve Erweiterung 6
QF7 QSIG Reserve Erweiterung 7
QF8 QSIG Reserve Erweiterung 8
QF9 QSIG Reserve Erweiterung 9
QFR QSIG in Frankreich
QIS QSIG ISO
QJI Anpassung QSIG f?r jistel IS
QLC QSIG Leitungs Code
QLCE1 QLC Erweiterung 1
QLCE2 QLC Erweiterung 2
QLCEC QLC Leitg. Code extern Rufe
QLCEO QLC extern nur PABX Nummer
QLCP1 QLC private Rufnr. nach Level 1
QLCP2 QLC private Rufnr. nach Level 2
QLCPN QLC priv. NP durch Knotennummer
QLCPP QLC privater NP bevorzugt
QLCRI QLC oefftl Rufnr. zu internat
QLCRN QLC oefftl Rufnr. zu national
QLCUC QLC ohne Einschraenkungen
QLCUO QLC o. Einschr. nur PABX Nr.
*QMS QSIG Meldungs Segmentierung
QNA Anzlize name priorit?r
QPR QSIG path replacement
QTD QSIG Type-Of-Number Wahl
QUL QSIG u-Law
---
However I can't find anything about these settings in conjunction with
Asterisk on the web...
Any ideas??
Cheers, Johann
/proc/zaptel/1:
---
Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" HDB3/CCS/CRC4
1 WCT1/0/1 Clear (In use)
2 WCT1/0/2 Clear (In use)
3 WCT1/0/3 Clear (In use)
4 WCT1/0/4 Clear (In use)
5 WCT1/0/5 Clear (In use)
6 WCT1/0/6 Clear (In use)
7 WCT1/0/7 Clear (In use)
8 WCT1/0/8 Clear (In use)
9 WCT1/0/9 Clear (In use)
10 WCT1/0/10 Clear (In use)
11 WCT1/0/11 Clear (In use)
12 WCT1/0/12 Clear (In use)
13 WCT1/0/13 Clear (In use)
14 WCT1/0/14 Clear (In use)
15 WCT1/0/15 Clear (In use)
16 WCT1/0/16 HDLCFCS (In use)
17 WCT1/0/17 Clear (In use)
18 WCT1/0/18 Clear (In use)
19 WCT1/0/19 Clear (In use)
20 WCT1/0/20 Clear (In use)
21 WCT1/0/21 Clear (In use)
22 WCT1/0/22 Clear (In use)
23 WCT1/0/23 Clear (In use)
24 WCT1/0/24 Clear (In use)
25 WCT1/0/25 Clear (In use)
26 WCT1/0/26 Clear (In use)
27 WCT1/0/27 Clear (In use)
28 WCT1/0/28 Clear (In use)
29 WCT1/0/29 Clear (In use)
30 WCT1/0/30 Clear (In use)
31 WCT1/0/31 Clear (In use)
---
Wolfgang Zweimueller
2006-Mar-31 04:49 UTC
[Asterisk-Users] Asterisk, QSIG and Tenovis PBX?
Hi Johann, "Johann Hanne" <jhml@gmx.net> 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