Hangup detection, AC Terminal Impedance

[beesknieg]

Hi Guys

First of all please don't tell me to get ISDN or port he numbers to a SIP trunk as I know this is the best option but boss man doesn't agree

I have a GXE5028 (Piece of *** Grandstream) acting as a Analog gateway for my Asterisk VM
The problem i have is that when users call and as the phone starts to ring they hangup for whatever reason the Grandstream doesn't drop the call to Asterisk just keeps ringing until you answer the "ghost"

I know the problem is with the Tones set on Gransdream but I can not find the correct values, Please if someone can help me with this I have been struggling for months now !

I will attach n snip of my current settings and the options i have for AC Terminal Impedance


Thank You in advance

 

[envo]

Your boss sounds like a tool

 

[Clarotech]

You already know the answer is ISDN or SIP.

If you add up the lost hours you've sunk into this along with frustrations experienced by staff and customers it really shouldn't be that hard to motivate.

 

[beesknieg]

You already know the answer is ISDN or SIP.

If you add up the lost hours you've sunk into this along with frustrations experienced by staff and customers it really shouldn't be that hard to motivate.

Could you maybe suggest any good ISDN gateway so i can go with proper motivasion on why and what we need to get here, really need something that qould provide the best quality

 

[Clarotech]

Could you maybe suggest any good ISDN gateway so i can go with proper motivasion on why and what we need to get here, really need something that qould provide the best quality

Take a look at Patton. There are also other units on the market, but we know that unit works well.

Any reason you're not looking at VoIP? Needs a stable internet connection, but at least you don't have to spend money on a gateway.

 

[beesknieg]

Take a look at Patton. There are also other units on the market, but we know that unit works well.

Any reason you're not looking at VoIP? Needs a stable internet connection, but at least you don't have to spend money on a gateway.

we have a 0861XXXXXX number registered at telkom that redirects to a hunting group of 7 analog lines, Telkom says they can't port a 0861XXXXXX number to a SIP trunk so i guess my only proper answer would be ISDN

 

[Clarotech]

we have a 0861XXXXXX number registered at telkom that redirects to a hunting group of 7 analog lines, Telkom says they can't port a 0861XXXXXX number to a SIP trunk so i guess my only proper answer would be ISDN

Yup unfortunately you are stuck with that one, but there are ways around that. We have a PRI service which we use for customers to point their 0861 numbers at which we then redirect into VoIP.

Remember that if you convert to ISDN you get 2 channels per line, so for 7 analogue lines you need 4 ISDN which will handle 8 concurrent calls.

 

[beesknieg]

Yup unfortunately you are stuck with that one, but there are ways around that. We have a PRI service which we use for customers to point their 0861 numbers at which we then redirect into VoIP.

Remember that if you convert to ISDN you get 2 channels per line, so for 7 analogue lines you need 4 ISDN which will handle 8 concurrent calls.

Thank you for the reply helped me alot, one last question, what debugging software would you recommend i get alot of calls dropping or people cant hear each other for a few seconds or calls drop exactly after 45 seconds if i phone another Vodacom 087XXXX number (very weird)

I know it is a issue with RTP and would like to debug more (it is not a NAT issue)

 

[Clarotech]

Thank you for the reply helped me alot, one last question, what debugging software would you recommend i get alot of calls dropping or people cant hear each other for a few seconds or calls drop exactly after 45 seconds if i phone another Vodacom 087XXXX number (very weird)

I know it is a issue with RTP and would like to debug more (it is not a NAT issue)

Are these calls out over your analogue gateway or over pure SIP?

 

[beesknieg]

Are these calls out over your analogue gateway or over pure SIP?

All our outgoing calls are routed over a pure SIP from Vodacom the analog lines are incoming calls from the 0861XXXXXX I used to have the SIP over a dedicated ADSL 10Mb (shoot me now) but i am currently running over a ME to test if the asynchronous wasn't my problem but the dropping calls still persist


Everyone are using Eyebeam 1.5 with the Ulaw and Alaw codecs to the PBX and the PBX translates to Vodacom with the G729a
I have seen RTP timeout errors in my Asterisk CLI after enabling RTP debug but with so many calls it is humanly impossible to go through the whole of the log

Therefore im looking for some advice on debugging software

 

[Clarotech]

All our outgoing calls are routed over a pure SIP from Vodacom the analog lines are incoming calls from the 0861XXXXXX I used to have the SIP over a dedicated ADSL 10Mb (shoot me now) but i am currently running over a ME to test if the asynchronous wasn't my problem but the dropping calls still persist

Everyone are using Eyebeam 1.5 with the Ulaw and Alaw codecs to the PBX and the PBX translates to Vodacom with the G729a
I have seen RTP timeout errors in my Asterisk CLI after enabling RTP debug but with so many calls it is humanly impossible to go through the whole of the log

Therefore im looking for some advice on debugging software

Sounds like NAT/Firewall issue to me. Take a look at this post: http://stackoverflow.com/questions/23050268/asterisk-call-drop-after-30-seconds it details SIP debugging which should show you more detail about what's happening. You should also be able to ask Vodacom to send you a SIP trace from their side. In our client base we use VoIPmonitor do similar monitoring and SIP traces.

It's very difficult to troubleshoot these things on a busy box, so might just be best to arrange a quiet time or do it after-hours.

You can run calls over ADSL, providing the line is stable and you are not exceeding the max concurrent calls you can manage with your codec of choice. We have good and bad experiences with ADSL just depends on the area.