Problems with Sangoma a104d and euroisdn e1

Asterisk Asterisk Support
1

Hello guys,

i’ve got a problem with calling out with my asterisk 1.2 system over my zap channels:

what i made
install wanpipe rpm
run wancfg
configured port 1 with euroisdn e1
wanrouter start
asterisk -r works

run wancfg_zaptel
configured everything

everything looks fine, ztcfg -vv shows me 31 chans configured

here my configuration, i think it’s an error in my config

extensions.conf

[globals]
…
…
TRUNK=Zap/g1
…
…
[default]
…
exten => _0.,1,Dial(Zap/g1/${EXTEN:1})
…

in asterisk CLI
vici*CLI> zap show channels

Chan Extension Context Language MusicOnHold pseudo from-pstn 1 from-pstn 2 from-pstn 3 from-pstn 4 from-pstn 5 from-pstn 6 from-pstn 7 from-pstn 8 from-pstn 9 from-pstn 10 from-pstn 11 from-pstn 12 from-pstn 13 from-pstn 14 from-pstn 15 from-pstn 17 from-pstn 18 from-pstn 19 from-pstn 20 from-pstn 21 from-pstn 22 from-pstn 23 from-pstn 24 from-pstn 25 from-pstn 26 from-pstn 27 from-pstn 28 from-pstn 29 from-pstn 30 from-pstn 31 from-pstn

zapata conf:

[trunkgroups]

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1

immediate=no

;Sangoma A104 port 1 [slot:1 bus:12 span:1]
switchtype=euroisdn
context=from-pstn
group=0
signalling=pri_cpe
channel =>1-15,17-31

Ouboundcall in CLI:

-- Executing Dial("SIP/cc100-0989d768", "Zap/g1/017681050342") in new stack Sep 18 05:00:38 NOTICE[5045]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Timeout on SIP/cc100-0989d768 == CDR updated on SIP/cc100-0989d768 -- Executing Goto("SIP/cc100-0989d768", "#|1") in new stack -- Goto (default,#,1) -- Executing Playback("SIP/cc100-0989d768", "invalid") in new stack -- Playing 'invalid' (language 'en') == Spawn extension (default, #, 1) exited non-zero on 'SIP/cc100-0989d768' -- Executing DeadAGI("SIP/cc100-0989d768", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CHANUNAVAIL----------") in new stack -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CHANUNAVAIL---------- completed, returning 0

anyone an idea? perhaps a wrong configuration with the group?

regards
philip

2

Why are you using an Asterisk version that is past its End of Life?

3

because it’s vicidial. it’s a full callcenter suite and the newest releast from vicidial only works stable with asterisk 1.2, so i can’t update.

but many other people work with asterisk 1.2 and sangoma cards…

i found a stupid mistake…i used group 0 and zap/g1…

now i get a new error :frowning:

-- Executing Dial("SIP/cc100-0946af08", "Zap/g1/017681050342") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/017681050342 -- Channel 0/2, span 1 got hangup, cause 6 -- Hungup 'Zap/2-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing DeadAGI("SIP/cc100-0946af08", "agi://127.0.0.1:4577/call_log--H Vcauses--PRI-----NODEBUG-----6-----CHANUNAVAIL----------") in new stack -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----6

4

in the asterisk log
perhaps some more information about the problem

Sep 18 14:30:31 DEBUG[3231] chan_sip.c: Setting NAT on RTP to 524288 Sep 18 14:30:31 DEBUG[3231] chan_sip.c: Checking SIP call limits for device cc100 Sep 18 14:30:31 DEBUG[3231] chan_sip.c: build_route: Contact hop: <sip:cc100@192.168.254.6:5060;transport=UDP> Sep 18 14:30:31 DEBUG[3144] channel.c: Avoiding initial deadlock for 'SIP/cc100-0946af08' Sep 18 14:30:31 VERBOSE[27531] logger.c: -- Executing Dial("SIP/cc100-0946af08", "Zap/g1/017681050342") in new stack Sep 18 14:30:31 VERBOSE[27531] logger.c: -- Requested transfer capability: 0x00 - SPEECH Sep 18 14:30:31 VERBOSE[27531] logger.c: -- Called g1/017681050342 Sep 18 14:30:31 VERBOSE[3211] logger.c: -- Channel 0/2, span 1 got hangup, cause 6 Sep 18 14:30:31 DEBUG[27531] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/2-1 Sep 18 14:30:31 DEBUG[27531] chan_zap.c: Hangup: channel: 2 index = 0, normal = 17, callwait = -1, thirdcall = -1 Sep 18 14:30:31 DEBUG[27531] chan_zap.c: Already hungup... Calling hangup once, and clearing call Sep 18 14:30:31 DEBUG[27531] chan_zap.c: disabled echo cancellation on channel 2 Sep 18 14:30:31 DEBUG[27531] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/2-1 Sep 18 14:30:31 DEBUG[27531] chan_zap.c: Updated conferencing on 2, with 0 conference users Sep 18 14:30:31 DEBUG[27531] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on Zap/2-1 Sep 18 14:30:31 DEBUG[27531] chan_zap.c: disabled echo cancellation on channel 2 Sep 18 14:30:31 VERBOSE[27531] logger.c: -- Hungup 'Zap/2-1' Sep 18 14:30:31 VERBOSE[27531] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Sep 18 14:30:31 DEBUG[27531] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL.

i found sth in google

cause code 6, which translates to “the service quality of the specified channel is insufficient to accept the connection.”

but it can’t be the cable, because it works fine on our other tksystem

5

another log from pri debug span 1

[code]vici*CLI> pri debug span 1
Enabled debugging on span 1
– Executing Dial(“SIP/cc100-09b50eb0”, “Zap/g1/017681050342”) in new stack
– Making new call for cr 32772
– Requested transfer capability: 0x00 - SPEECH

Protocol Discriminator: Q.931 (8) len=40
Call Ref: len= 2 (reference 4/0x4) (Originator)
Message type: SETUP (5)
[04 03 80 90 a3]
Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0)
Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16)
Ext: 1 User information layer 1: A-Law (35)
[18 03 a9 83 84]
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type: 3
Ext: 1 Channel: 4 ]
[6c 07 21 80 63 63 31 30 30]
Calling Number (len= 9) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user number not screened (0) ‘cc100’ ]
[70 0d a1 30 31 37 36 38 31 30 35 30 33 34 32]
Called Number (len=15) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) ‘017681050342’ ]
[a1]LI>
Sending Complete (len= 1)
– Called g1/017681050342
< Protocol Discriminator: Q.931 (8) len=9
< Call Ref: len= 2 (reference 4/0x4) (Terminator)
< Message type: RELEASE COMPLETE (90)
< [08 02 82 86]
< Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2)
< Ext: 1 Cause: Channel unacceptable (6), class = Normal Event (0) ]
– Processing IE 8 (cs0, Cause)
– Channel 0/4, span 1 got hangup, cause 6
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
– Hungup ‘Zap/4-1’
== Everyone is busy/congested at this time (1:0/0/1)[/code]

6

ViciDial runs fine under Asterisk 1.4 (most versions of 1.4 Asterisk above 1.4.18 are stable). All new installations that we are doing of ViciDial are using a patched version of 1.4.21.2.

7

Ok i will try it!
What will be better?

first install asterisk 1.4 and then vicidial.
or to take the vicidialnow installation and then upgrade to 1.4?

8

This is the build that we made of our patched version of 1.4.21.2:
download.vicidial.com/required-a … ici.tar.gz

Not sure how easy it is to upgrade on ViciDialNOX because not all of the utilities to recompile are on there. I would recommend trying ViciBox instead if you need an ISO.

9

do u have a good link with a howto?
because i’m really new in asterisk and vicidial.

Thanks!!

10

our carrier made a mistake.
they configured channel 1 to 7 for data…i don’t why…
and the second problem was, i have to dial 72162xx instead of 072162xx

Thanks for help!