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Dial rules for Yeastar S20 PABX

crowbar2008

Well-Known Member
Joined
Nov 24, 2008
Messages
210
#1
Hi Guys,

Hopefully someone can help me out here. I have a client with a S20 PABX and 7 Extensions all running on Yeahlink T40P Sip phones and using a SIP provider for dialout and analoque lines for there incoming calls. I am having an isseu where they cannot dialout certian numbers like.

016 XXXXXX
079 XXXXXX
087 XXXXXX

My dialout rule is defined as just a . Is there a special pattern I need to specify for these numbers to work? Sorry I know this is probably a newbie question but I only got into selling VOIP to my customer last year.

Your help would be greatly appreciated.

Thanks,
Jan
 

jb30589

Active Member
Joined
Feb 6, 2013
Messages
35
#2
That rule should work fine.
Try X. to start with. Others might have better insight on this though.
 

Brandon

Well-Known Member
Joined
Feb 18, 2004
Messages
212
#3
Agree with the above, try X.

If that doesn't work, check the logs. Try SSH onto the S20 and run
Code:
asterisk -r  -vvvvv
to see what's going on, make a call to one of the numbers that doesn't work. (best to do this with nobody else making/receiving calls)
 

InfinityMVS

Well-Known Member
Joined
Apr 1, 2011
Messages
145
#4
Hi Jan,

Have you tested the numbers from another source like your mobile ? It makes no sense that only certain number are not dialed especially with the the outbound rule you have set, I usually set three rules XXXXXXXXXX, 00., and service numbers 1021[7-9]

With the . rule it could also be digit time out, so if the user is a slow dialer the call could be executed while they are still dialing the number
 

dillydog

Well-Known Member
Joined
Mar 18, 2010
Messages
106
#5
You can also force the calls over the incoming Telkom lines to see if they work...Perhaps your VOIP provider can assist with a log or a hangup cause. Are other number working? If nothing is routing I would be looking at codecs or nats.

Thanks
 

crowbar2008

Well-Known Member
Joined
Nov 24, 2008
Messages
210
#6
Thank you for the prompt reply. I am going to test all of this today and revert back. If I force them onto their analogue lines it seems to be working fine.
 

crowbar2008

Well-Known Member
Joined
Nov 24, 2008
Messages
210
#7
Hi everyone. Still battling with this issue. I have now tried a new SIP account and I can dial all numbers I want to. So I spoke to the current sip provider and they say I should “re instate” the G729 codec. So I asked them how must I do this and they say via the CLI interface on putty when I ssh into the pabx. Does anyone know how to do this as I have search the Internet high and low without any results.
 

Hamish McPanji

Honorary Master
Joined
Oct 29, 2009
Messages
34,825
#8
Hi everyone. Still battling with this issue. I have now tried a new SIP account and I can dial all numbers I want to. So I spoke to the current sip provider and they say I should “re instate” the G729 codec. So I asked them how must I do this and they say via the CLI interface on putty when I ssh into the pabx. Does anyone know how to do this as I have search the Internet high and low without any results.
New sip provider works, but old one doesn't? They must be having problems with their call routing on their end
 

crowbar2008

Well-Known Member
Joined
Nov 24, 2008
Messages
210
#9
That’s exactly what I said but they insist that it the codecs that doesn’t get loaded properly when a call get ittiated.
 

Hamish McPanji

Honorary Master
Joined
Oct 29, 2009
Messages
34,825
#10
That’s exactly what I said but they insist that it the codecs that doesn’t get loaded properly when a call get ittiated.
Load mizudroid or zoiper on or other sipphone on your smartphone. Add your VoIP provider settings. And test those numbers that are failing. (Am assuming android) . Also test numbers within the working range.

Don't waste a lot of time on the Yeastar side until it has been tested elsewhere. Eliminate issues with the VoIP provider first. First test on mobile network. Then test on your company network.
 

crowbar2008

Well-Known Member
Joined
Nov 24, 2008
Messages
210
#11
I have taken the pabx off site and tested at my house with both my LTE and ADSL. It does the same thing. So I took there sip details and added it to Zoiper on my phone. Called the number and it works. So everything is pointing to the pabx
 

Hamish McPanji

Honorary Master
Joined
Oct 29, 2009
Messages
34,825
#12
I have taken the pabx off site and tested at my house with both my LTE and ADSL. It does the same thing. So I took there sip details and added it to Zoiper on my phone. Called the number and it works. So everything is pointing to the pabx
Was G.729 enabled on your zoiper or not?
 

crowbar2008

Well-Known Member
Joined
Nov 24, 2008
Messages
210
#15
Hi Hamish. So eventually found out the most silly thing under general there is a SIP tab with the codecs the G729 codec was not even loaded in the list so stupid of me as soon as I loaded it straight away they could phone all numbers. I am not sure how I missed this. After 2 hard resets of the pabx it drove me mad. Thanks for all your help so far I really appreciate it.
 
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