Goautodial and E1, inbound and outbound calls problem

[sherkan]

Hi everybody
I' using a simple PC as a server for a mini-callcenter for my office, I installed Goautodial-ce-2.1
I reinstalled asterisk dahdi and libpri, I used asterisk1.4.39 and the latest dahdi & libpri
I'm connected to my career using E1, connected to my TE405P digium card

I having a problem with inbound and outbound calls, I can see them on my CLI, but still it hangs up once I call or I tried to call the E1 number
Here's my conf files:
/etc/dahdi/system.conf ---::

# Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS/CRC4 ClockSource
# span=1,1,0,ccs,hdb3,crc4
span=1,0,0,ccs,hdb3,crc4
# termtype: te
bchan=1-15,17-31
dchan=16
echocanceller=mg2,1-15,17-31
# ***************************************************************************
# Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" HDB3/CCS/CRC4
span=2,0,0,ccs,hdb3,crc4
# termtype: te
bchan=32-46,48-62
dchan=47
echocanceller=mg2,32-46,48-62
----------------------------------------------------------------------------------
My /etc/asterisk/chan_dahdi is

[channels]
#include /etc/asterisk/dahdi-channels.conf
context=trunkinbound
hidecallerid=no
;hidecalleridname=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
transfer=yes
canpark=no
cancallforward=yes
callreturn=no
echocancel=yes
echocancelwhenbridged=no
rxgain=0.0
txgain=0.0
group=0
callgroup=0
pickupgroup=0
immediate=no
busydetect=yes
busycount=8

----------------------------------------------------------------------
My /etc/asterisk/dahdi_chanenels is

; Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER)
group=0
context=default
switchtype = euroisdn
signalling = pri_cpe
channel => 1-15,17-31
overlapdial=yes
usecallerid=yes

; Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2"
group=1
context=default
switchtype = euroisdn
signalling = pri_cpe
channel => 32-46,48-62
overlapdial=yes
usecallerid=yes

-------------------------------------------------------------------------------
And my extensions.conf is like this

[globals]
CONSOLE=Console/dsp
TRUNK=Zap/r1
TRUNKX=zap/r2
TRUNKXYZ=DAHDI/G0

[trunkinbound]
; DID call routing process
exten => _X.,1,AGI(agi-DID_route.agi)

[default]
include => vicidial-auto
exten => _078XXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _078XXXXXXX,2,Dial(${TRUNKXYZ}/${EXTEN:1},,To)
exten => _078XXXXXXX,3,Hangup

[sherkan]

Here's what happen in my CLI when I try to call the E1 line, then it hangs up:


go*CLI>
-- Accepting call from '788537919' to '0788196500' on channel 0/1, span 1
-- Executing [0788196500@default:1] AGI("DAHDI/1-1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [0788196500@default:2] Dial("DAHDI/1-1", "DAHDI/G0/788196500||To") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G0/788196500
-- DAHDI/31-1 is proceeding passing it to DAHDI/1-1
-- DAHDI/31-1 is making progress passing it to DAHDI/1-1
-- DAHDI/31-1 is making progress passing it to DAHDI/1-1
-- Channel 0/31, span 1 got hangup request, cause 31
[Sep 18 05:20:43] WARNING[27523]: app_dial.c:801 wait_for_answer: Unable to forward voice or dtmf
-- Hungup 'DAHDI/31-1'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [0788196500@default:3] Hangup("DAHDI/1-1", "") in new stack
== Spawn extension (default, 0788196500, 3) exited non-zero on 'DAHDI/1-1'
-- Executing [h@default:1] DeadAGI("DAHDI/1-1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----31-----CHANUNAVAIL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Hungup 'DAHDI/1-1'


----------------------------------------------
Here's what happen when I try to call my number for example, using my softphone:


go*CLI>
-- Executing [0788537919@default:1] AGI("SIP/cc100-00000005", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [0788537919@default:2] Dial("SIP/cc100-00000005", "DAHDI/G0/788537919||To") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G0/788537919
-- DAHDI/31-1 is proceeding passing it to SIP/cc100-00000005
-- DAHDI/31-1 is making progress passing it to SIP/cc100-00000005
-- DAHDI/31-1 is making progress passing it to SIP/cc100-00000005
-- Channel 0/31, span 1 got hangup request, cause 31
[Sep 18 05:22:49] WARNING[28511]: app_dial.c:801 wait_for_answer: Unable to forward voice or dtmf
-- Hungup 'DAHDI/31-1'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [0788537919@default:3] Hangup("SIP/cc100-00000005", "") in new stack
== Spawn extension (default, 0788537919, 3) exited non-zero on 'SIP/cc100-00000005'
-- Executing [h@default:1] DeadAGI("SIP/cc100-00000005", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----31-----CHANUNAVAIL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
go*CLI>

------------------------------------------------------------

I tried to follow some links of striker and also viewed many other similar posts on this forum, but still, cant figure out why the inbound and outbound still refusing to work
Thanks a lot for the help and your time, God bless

[sherkan]

Anybody tell me where does the below warning comes from?

[Sep 19 03:45:24] WARNING[10919]: app_dial.c:801 wait_for_answer: Unable to forward voice or dtmf
-- Hungup 'DAHDI/31-1'

Is it codecs problem? It definitely looks like it's asterisk problem, nothing to do with vicidial, I'm sorry... anybody who have eve faced this issue before?

[DomeDan]

no experience with E1 but some google for "wait_for_answer: Unable to forward voice or dtmf" brought me to this:

http://www.mail-archive.com/asterisk-us ... 89463.html

I get that error too (my Asterisk version is 1.2.x but should be the
same) when that Zap channel is not available and you are trying to use it.
You should get a CHANUNAVAIL from Asterisk channel status.

viewtopic.php?t=17845

It was a dial plan issue. Our telco confirmed that they were seeing 12 digit numbers coming through with a 9 in front (instead of the 11 we'd normally expect). So our dial plan was incorrect, I'm guessing that the EXTEN:1 was actually just removing the _ and our Telco doesn't require a 9 to be passed to them:

http://forums.digium.com/viewtopic.php? ... 6dfbff45f6

When reconnect trunk or restart asterisk it works again.
How can i diagnose it?
...
I have the same problem when calling to 09xx numbers in holland !
But not with all numbers.

btw, hangup cause 31 is a strange one:
http://wiki.freeswitch.org/wiki/Hangup_causes

NORMAL_UNSPECIFIED - This cause is used to report a normal event only when no other cause in the normal class applies.

ask your provider what they see, if it reach them

[schernit]

I have the same 31 error. How did you solve it ? Was it a carrier bug ? Please tell me something because I have this issue for the last 2 months and nobody knows what the bug is.

regards,

Sebastián

[williamconley]

I have the same 31 error. How did you solve it ? Was it a carrier bug ? Please tell me something because I have this issue for the last 2 months and nobody knows what the bug is.

regards,

Sebastián

1) Welcome to the Party!

2) As you are obviously new here, I have some suggestions to help us all help you:

When you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

This IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3) got hangup request, cause 31 is a generic "i'm gonna hang up and have no idea why" response. Which means it could be due to a fairly large number of reasons, in fact it could even be considered a normal hangup (ie: no error, just received a hangup request). You should use debug on the channel and/or ask your carrier if they saw the call at all and if so was there an error of some sort.

4) Do note that this is note likely in any way related to Vicidial, but to Asterisk (the PBX engine which Vicidial controls ...). The Asterisk community is much larger than the Vicidial community and you may find more results/help there (possibly, certainly worth a try).

5) Please verify that your asterisk version in Admin->Servers->(click on your server) is the same as the asterisk version reported when you enter "asterisk -V" at the command line and that you are using the latest version of Vicidial (perhaps even consider a fresh install at Vicibox.com ...).

6) Happy Hunting!