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Delay in Audio while picking up calls

illizit
Posts: 55
 Member Since:
2007-01-17
Submitted by illizit on Wed, 10/15/2008 - 1:55am.

Hello.

We have a situation where picking up calls that are on hold results in our voice not being heard for abount 2-5 seconds. We can however hear the other party during this time. It also ocurrs when transfering calls; it takes a few seconds before the party can hear us. We have verified this ocurring in other trixbox installs as well, there might be something we are setting up wrong. I have tried recompiling zap drivers and zttest shows 99.9%. Both boxes are using bandwidth.com for incoming and outgoing.

Any suggestions??

Thanks!

‹ Need Dial Plan Help Severe security warning - Extensions with matching secret ›

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tyardley
Posts: 209
 Member Since:
2007-09-09
I have seen this too
Wed, 10/15/2008 - 2:00am

I have noticed this with soft phones. I have a customer using Vitelity and Teliax and this same thing happens on a soft phone on both Windows and OS X. Does not happen on their AASTRA or Polycom phones. I don't know why or what is causing it at this moment. But I wanted to let you know that I have noticed it too.

Thank you,
Engineer Tim

Trixbox Engineer
http://engineertim.com

--

http://engineertim.com
http://xenforge.org
http://twitter.com/engineertim


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illizit
Posts: 55
 Member Since:
2007-01-17
Well right now it's pretty
Wed, 10/15/2008 - 2:55pm

Well right now it's pretty bad and it happens with all phones. This customer has Grandstream phones (GXP-2000) and Aastra 57i phones.


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illizit
Posts: 55
 Member Since:
2007-01-17
Ok, I created a custom
Thu, 10/16/2008 - 11:12am

Ok, I created a custom dialplan in extensions_custom.conf in order to manually handle the calls coming from bandwidth.com

It basically detects if an incoming call's DID matches and then calls an extension. Doing it like this, it works PERFECT. No audio delays, etc. So there is clearly something wrong inside freepbx, unless of course I am configuring something wrong...


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SkykingOH
Posts: 9678
 Member Since:
2007-12-17
Some call traces with the
Thu, 10/16/2008 - 8:17pm

Some call traces with the delayed audio would be helpful.

Also posting your custom code and the peer from sip_additional.conf that FreePBX created. Please use the code tag options so we can wade through the posts easier.

--

Scott

aka "Skyking"


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illizit
Posts: 55
 Member Since:
2007-01-17
Hello, I have a context
Fri, 10/17/2008 - 9:02pm

Hello,

I have a context inside the USER section of the trunk called bandwidth-custom.
I only have this for testing purposes and it works perfect.

Inside extensions_custom.conf I have:

[bandwidth-custom]
exten => +13052222222,1,Answer
exten => +13052222222,n,Dial(SIP/100,30)
exten => +13052222222,n,Hangup

When I call my DID, it simply calls my extension. Using this dial plan creates no issues, everything works perfect, however; in freepbx even if I point an incoming route directly to my extension I have the same issue.
Trace logs reveal no information (as far as I can tell) They only show when I place the caller on hold (MOH STARTING) and when I take the caller of hold (MOH STOPING). The caller on the other end hears when the MOH stops and I can even hear them! They cannot however hear me for about 2-4 seconds; I just suddenly "pop in". The issue also occurs when I delete the trunk completly and just have the call come in as an anonymous call. (The provider sends us the call directly to our IP)


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joshpatten
Posts: 733
 Member Since:
2007-01-20
This may be a long shot, but
Fri, 10/17/2008 - 10:46pm

This may be a long shot, but give it a try:

try removing the 't' option from you dial string under general settings, then set all of your SIP extensions and SIP trunks with the option:

canreinvite=yes

the extensions will have a specific area you can just put yes in, but the trunks you will have to input that line.

If this clears your issue up then something is going haywire with asterisk's rtp processing engine.

(I don't know if it will even do anything at all, but it won't hurt anything to try)


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illizit
Posts: 55
 Member Since:
2007-01-17
Wow!!! Thank you!! Removing
Sat, 10/18/2008 - 12:42am

Wow!!! Thank you!!
Removing the t option worked! Why would it cause this anyways? I did a complete reinstall from scratch and I had the same issue until I removed the t option.

Thanks!


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joshpatten
Posts: 733
 Member Since:
2007-01-20
removing the t option pretty
Sat, 10/18/2008 - 7:26am

removing the t option pretty much takes asterisk out of the media stream, so if you were using the "press pound to transfer" feature and wish to keep using it, they you will need to re-enable it. Also, if you set canreinvite=yes then it's possible you will lose call recording functionality.

I don't have a use for either so I simply remove the t option and set canreinvite=yes on everything and save some CPU horsepower.


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illizit
Posts: 55
 Member Since:
2007-01-17
I didn't touch the
Sat, 10/18/2008 - 11:48am

I didn't touch the canreinvite options, all I did was remove the t, which shows is just for removing asterisk out of the media stream. I do not use any features that require this option so it's fine. What I am really interested in is why its causing an issue; especially on a brand new box.


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illizit
Posts: 55
 Member Since:
2007-01-17
My box specs are 2x Dual
Sat, 10/18/2008 - 11:51am

My box specs are 2x Dual Core Xeon Procs, (2.0ghz), 2gb RAM and two 160GB SCSI 15k hd's.


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msetton
Posts: 9
 Member Since:
2008-09-23
What if i want to use call
Fri, 06/04/2010 - 11:05am

What if i want to use call recording? Is there another way around it?


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