Another Newbie needs help..

[nranka]

Need Urgent Help

Hi I'm a COMPLETE Newbie to Asterix and also IP Telephony itself.
Am trying to setup a small home office and am experimenting with Asterix.
I'm based out of India and have a 20 number PRI line from Airtel.
I've set-up an Elastix 1.6 box and have the following questions now:

1) Under trunks - how many trunks must I add / create? Just one for the main PRI number (40ABC200) or one for each number in the series (40ABC200 - 40ABC299) - all of which apparently reach my elastix box? This I tested by having just one elastix trunk and setting ZAP identifier as g0.

2) If I should only one trunk setup - then how do I ensure that different departments of my SOHO business can dial out to their respective customers using a different callerid? For instance, I want the gifting department to dial out its customer who should get the caller id is 40ABC210. When the branding departments calls out to their customers the customers must see the callerid as 40ABC250. How do I achieve this using only 1 trunk?

3) Please help me understand the concept of ZAP Channel DID? Do I need it? What I want to achieve for my SOHO is also DID numbers (40ABC210 for Gifting department and 40ABC250 for branding department, etc). Somehow I feel I need to setup ZAP Channel DID to enable the call routing based on number dialled (40ABC210 or 40ABC250). Please help.

4) Btw, on my currently installation (i.e single Zap g0 trunk) I'm facing a wierd problem. If I dial any india cell phone number (10 digits) the call goes thru. If I dial a local number (8 digits) I get a "All circuits are busy now" error messsage. The same local number preceeded by 2 digits of the local code (80) allows the calls to go thru. Some randon 2 digit code preceeded a valid local number also produced the "All Circuits are busy now " message. Dialing a long distance mobile number (11 digits - basically 0 preceeding the 10 digit number) also generates the "All Circuits are busy now" message. Any idea what I'm doing wrong? My dial pattern for ourbound route is just '.' and Dial Rule for Trunk is blank/empty.

Feedback and help in the above regard would be highly appreciated.

Btw, I'm currently using Elastix 1.6 as it was recommended on another forum for its superior / easy sugarcrm integration. I however, did now have the issue 4 mentioned above when I used Elastix 2.3.3. Should I revert back to the latest version?

regards
nranka

 

[jgutierrez]

I will recommend you to start reading:

* Elastix Without Tears
* Unified Communications With Elastix

They are available on the book/manual section of this forum.

 

[nranka]

Hi, first off thanks for the prompt response..

I've read elastix w/o tears atleast 2 times. Unfortunately that did not give me any clarity on my above posted queries. I'll try to read the other recommended book. Will post back if i cant get any clarity from that either.

regards
Nranka

 

[nranka]

Hi!

I did not get a chance to read "Unified Communications With Elastix" but did end-up browsing thru *Elastix Without Tears once again. However, I rebuilt the system with Elastix 2.0.3 and my issue no 4 (in my original post above) is sorted. However, I'm still clueless with regards to the other issues / questions that I had. If the book *Unified Communications with Elastix will address my questions above, then I'd take the advise and buy it.

I'm however facing another problem now with Echo:

When making calls using a softphone (3CX) to an external number via PSTN I'm facing a very poor quality of sound (with background nosie) and the called party is getting a very strong Echo. Have tried this with multiple softphones and calling different parties (landline & mobile) and this problem still persists. Would sincerely appreciate assistance in this matter. Details required to help me troubleshoot this are as below:

1) The System itself if an AMD Quad Core processor system, with 8GB RAM
2) The card I'm using is Digium Wildcard TE121 w/o the hardware echo canceller module (as I was recommended against it)

results of the cat /procs/interrupts are as follows:
[root@DRAELASTIX ~]# cat /proc/interrupts
CPU0 CPU1 CPU2 CPU3
0: 68623698 0 0 0 IO-APIC-edge timer
1: 120 0 0 0 IO-APIC-edge i8042
7: 0 0 0 0 IO-APIC-edge parport0
8: 1 0 0 0 IO-APIC-edge rtc
9: 0 0 0 0 IO-APIC-level acpi
12: 4 0 0 0 IO-APIC-edge i8042
50: 56284 0 0 0 PCI-MSI ahci
58: 407093 0 0 0 PCI-MSI eth0
66: 18 0 0 0 PCI-MSI HDA Intel
169: 68557317 0 0 0 IO-APIC-level ohci_hcd:usb4, ohci_hcd:usb5, ohci_hcd:usb6, ohci_hcd:usb7, wcte12xp0
177: 2 0 0 0 IO-APIC-level ehci_hcd:usb1, ehci_hcd:usb2, ehci_hcd:usb3
185: 616206 0 0 0 IO-APIC-level ide2, HDA Intel
NMI: 11991 4078 3374 2633
LOC: 68621864 68621792 68621719 68621646
ERR: 0
MIS: 0


contents of chan_dahdi.conf as as follows:

chan_dahdi.conf
[channels]
context=from-pstn
signalling=fxs_ks
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
faxdetect=incoming
echotraining=800
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1

;Uncomment these lines if you have problems with the disconection of your analog lines
;busydetect=yes
;busycount=3


immediate=no

#include dahdi-channels.conf
#include chan_dahdi_additional.conf



contents of dahdi-channels.conf as as follows:

; Autogenerated by /usr/sbin/dahdi_genconf on Mon Oct 17 10:52:52 2011
; If you edit this file and execute /usr/sbin/dahdi_genconf again,
; your manual changes will be LOST.
; Dahdi Channels Configurations (chan_dahdi.conf)
;
; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended
; to be #include-d by /etc/chan_dahdi.conf that will include the global settings
;

; Span 1: WCT1/0 "Wildcard TE121 Card 0" (MASTER)
group=0,11
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 1-15,17-31
context = default
group = 63



Please let me know if any other information is needed. Unfortunately I don't even know where to being to work at resolving this echo & noise issue.

regards
nranka

 

[nranka]

bump

 

[danardf]

Hmmm. I've no experience with this kind of card. But you could try to adjust the rxgain value, and maybe txgain value.

The values are expressed in dB, so, you could have a negative value and/or a decimal value.

Remeber to implement a QoS on your LAN too (side router, or switch)

 

[nranka]

Hi Franck, thanks for much for the help. More newbie question - where will I find the rx & fx values?
Also, this echo I get, is not always there. This is what is making it harder for me to troubleshoot. I've got a telecaller at our office who works thru a database making about 100 calls per day. Some of the called parties complain of the echo. Do you think this is due to network issues?

I don't have any QoS on the network, but the system Im using (on which I run the 3cx softphone) is connected via rj45 to a switch and the elastix server is also directly connected to the same switch. So between the softphone and elastix there should not be any network issues at all.

Do you think there is something else I need to try? Do you think it could be an issue with the soundcard on the client system (where the softphone is installed)? I'm using the soundcard that come built into the Main Boards (Mercury or any other desktop intelboard).

regards
nranka

 

[danardf]

Hi Franck, thanks for much for the help

.

No problem

More newbie question - where will I find the rx & fx values?

Are you kidding????? :blink:

It's you which give us your config...

contents of chan_dahdi.conf as as follows:

chan_dahdi.conf
[channels]
context=from-pstn
signalling=fxs_ks
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
faxdetect=incoming
echotraining=800
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1

;Uncomment these lines if you have problems with the disconection of your analog lines
;busydetect=yes
;busycount=3


immediate=no

#include dahdi-channels.conf
#include chan_dahdi_additional.conf

Read Elastix Without Tears before and try to find a way using the search engine from this forum.
There's lots of threads about that.

For every VoIP system, you need absolutly to implement a QoS on your LAN.
3CX, SIPX, Asterisk, Alcatel OXE, Avaya, ..or else.
It's same thing for all.

 

[vishalanand]

Hi Nishant,

I am in the same state as you were 9 months back. I am trying to seek resolutions to my issues but I guess have not been sucessful in cracking the monster.

Was wondering if its ok with you u, if I can sek your advice over phone as we both are based in India and will not be an issue for me to call you.

Looking forward to your reply.
Thnxs.