Elastix dials wrong extensions on outbound calls

gquinlan

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#1
Hello,

I have an ongoing issue that I REALLY need to resolve. When placing outbound calls that require the user to dial an extension, 9 times out of 10 they end up either getting 'invalid extension error', or a completely different active extension. It is intermittent and there is no pattern or apparent event that precipitates this issue. I hope someone has some ideas. Thank you in advance! :S

Greg

Elastix 1.6-14
Aastra Phones
 

dicko

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#2
My interpretation of your post is that after the call is completed successfully, the far end system has trouble resolving your manual key presses,.

If so check your DTMF settings and make sure they are compatible with the codecs you are using and the VSP itself handleing of it. There are plenty of posts here and elsewhere to help, search for DTMF.

dicko
 

gquinlan

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#3
Thanks for the reply! I think I found the culprit though. For some reason Elastix loses communication with the Sangoma cards. If I go to the Hardware Detection tab and have it replace the chan_dahdi.conf file and detect Sangoma hardware, the problem goes away. I plan to shut the server down tomorrow and make sure the cards are seated firmly. To be continued I'm sure!!

Greg
 

jgutierrez

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#4

gquinlan

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#5
Good morning,

Thanks for the link. I don't think the issues are the same. My problem involves once you're already connected and need to dial an extension. It doesn't always dial the correct extension.

Best,

Greg
 

gquinlan

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#6
I checked that post and believe my problem is different. We don't have an issue with connecting to the phone on the other end. It's when they have an auto-attendant/ivr and we have to dial an extension. Sometimes the system on the other end responds after the first or second key press(in other words, before you finish entering the entire extension). That results in an "invalid extension" message from their end. This is very frustrating to me, and my supervisor's patience is wearing thin. Btw, I just put in a new Elastix 025 Server running 2.0.
 

jgutierrez

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#7
Try updating the last packages of your elastix executing:
yum update
 

gquinlan

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#8
I'm not sure I understand what you mean by 'updating the last packages'? Since my original post I have installed an Elastix 025 appliance running 2.0. Do I just need to run yum update?

Thanks,

Greg
 

andyshawn

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#9
Hey Greg,

Do you have this problem with every remote system that has an IVR or is it just one system?
Have you checked your DTMF settings as dicko suggested?
 

gquinlan

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#10
Hi Andy,

It doesn't appear to happen with every remote ivr, nor does it happen every time. I had one user report last week that when she placed a call that required her to dial an extension, she got the "invalid extension" error. She then hung up and immediately tried again, and the call went through fine. Needless to say it is an intermittent problem.

Greg
 

andyshawn

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#11
Hey Greg,

This are my thoughts.

Maybe the user did enter the wrong extension or maybe the network connection from that user to the pbx was bad, hence the packets containing her entry did not arrive on time or in the correct order. We all know that VoIP is very sensitive, hence the slightest glitch will affect everything.
I would suggest you take a closer look at the persons who complain connection, latency and also their handset.
It could also be a problem on the remote system....u never know!
 

gquinlan

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#12
I wish it were as simple as dialing the wrong extension, but I've experienced it myself on occasion. I will look at the network theory this coming week. I may also need to find out more info from the folks on the other end to see what they are using for phone systems. Do you have any opinion on jgutierrez' suggestion of updating packages?
 

dicko

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#13
It will be very hard for anybody to help you until you actually post the type of trunk you are using and if they are SIP trunks the type of dtmf signaling you are using and the allowed codecs, until then everyone here is "whistling in the wind"

If you are using the Sangoma trunking then

http://wiki.sangoma.com/wanpipe-linux-a ... ing#swdtmf

but also

http://www.voip-info.org/wiki/view/Asterisk+DTMF

particularly the

toneduration=

(in /etc/asterisk/chan_dahdi.conf now)



If analog or tdm trunks almost always tx/rx gains need tuning for reliable DTMF generation/detection (use dahdi_monitor and milliwatt testing, (other posts here and elsewhere) ) as does the EC need to be functional (or at least non-intrusive) for reliable dtmf pass through.
 

gquinlan

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#14
Thanks dicko, I'll check those links out today. We are connected to 4 standard phonelines(Fairpoint) and DTMF is set to rfc2833. I hope that was the correct info you asked about. As you can tell, I'm new at this, but am learning all the time!
 

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