[Help] Gaps in playback in asterisk test

Asterisk Asterisk Support
1

Hello,

I have installed the latest version of Asterisk (1.2.9.1) and zaptel (1.2.6) from source on a machine running Debian Sarge (3.1, kernel 2.6)

The machine is a Dell Dimension 3000 2.4Ghz P4 with 256mb of ram.

All of this takes place on my local area network, both the asterisk and my other machine are connected directly to my D-Link DL-614+ router. The network setup is really as simple as this. It’s DHCP also but I don’t think it might affect anything.

The problem is that when I test on extension 1000, I get huge, gigantic gaps in playback.

“Congr …[gap]… tulations you ha …[gap]… ve success …[gap]… fully installed as …[gap]… terisk” at which point I hangup because I can’t take it anymore.

The gaps vary in length of time, and vary in place, for example it might say “Congratulations” completely then there would be a gap. Sometimes it could be considered halfway decent until you hear a gigantic gap again… and again… and again.

I connect to the asterisk from a windows 2000 machine running the latest xlite.

I have no FXO in the machine, though I do have an X100P right next to me on my desk right now.

I originally thought it was a timing issue, and perhaps it has been in the past, but I modprobed zaptel and ztdummy successfully as far as I can tell.

However if I run ztcfg, I get that ztdummy is UNCONFIGURED.

Or if I “asterisk -r”, and “zap show status”, I get the UNCONFIGURED again.

I can post here what asterisk says as it’s playing back the message:

Asterisk Ready.
    -- Unregistered SIP '1234'
    -- Registered SIP '1234' at 192.168.0.102 port 52966 expires 3600
    -- Saved useragent "X-Lite release 1002tx stamp 29712" for peer 1234
    -- Executing Goto("SIP/1234-cdb5", "default|s|1") in new stack
    -- Goto (default,s,1)
    -- Executing Wait("SIP/1234-cdb5", "1") in new stack
    -- Executing Answer("SIP/1234-cdb5", "") in new stack
    -- Executing Set("SIP/1234-cdb5", "TIMEOUT(digit)=5") in new stack
    -- Digit timeout set to 5
    -- Executing Set("SIP/1234-cdb5", "TIMEOUT(response)=10") in new stack
    -- Response timeout set to 10
    -- Executing BackGround("SIP/1234-cdb5", "demo-congrats") in new stack
    -- Playing 'demo-congrats' (language 'en')
  == Spawn extension (default, s, 5) exited non-zero on 'SIP/1234-cdb5'
    -- Unregistered SIP '1234'
exit

Seems pretty good I think. At the end I shut down the phone and asterisk.

The xlite log goes like this:

[06-06-11]13:55:15.986 | Info (min) Utilities | "Detected CPU [              Intel(R) Pentium(R) 4 CPU 2.60GHz]" | sua::CProcessorInformation::Detecti386CPU
[06-06-11]13:55:16.971 | Info (min) Jitter | "Adding latency to offset lack of data" | sua::CJitterBuffer::HandleOneSecondTick
[06-06-11]13:55:19.049 | Info (min) RTP | "RTCP calculation 37 37 37 37 0" | sua::CRTPDestination::AppendAndNotifyAboutReceiverReport
[06-06-11]13:55:19.971 | Info (min) Jitter | "Adding latency to offset lack of data" | sua::CJitterBuffer::HandleOneSecondTick
[06-06-11]13:55:23.876 | Info (min) RTP | "RTCP calculation 134 134 97 97 0" | sua::CRTPDestination::AppendAndNotifyAboutReceiverReport
[06-06-11]13:55:23.971 | Info (min) Jitter | "Adding latency to offset lack of data" | sua::CJitterBuffer::HandleOneSecondTick
[06-06-11]13:55:24.971 | Info (min) Jitter | "Adding latency to offset lack of data" | sua::CJitterBuffer::HandleOneSecondTick
[06-06-11]13:55:31.103 | Info (min) RTP | "RTCP calculation 166 166 32 32 0" | sua::CRTPDestination::AppendAndNotifyAboutReceiverReport
[06-06-11]13:55:31.972 | Info (min) Jitter | "Adding latency to offset lack of data" | sua::CJitterBuffer::HandleOneSecondTick
[06-06-11]13:55:32.972 | Info (min) Jitter | "Adding latency to offset lack of data" | sua::CJitterBuffer::HandleOneSecondTick
[06-06-11]13:55:38.070 | Info (min) RTP | "RTCP calculation 314 314 148 148 0" | sua::CRTPDestination::AppendAndNotifyAboutReceiverReport

Seems pretty bad this time. What’s going on?

Here’s a part of my sip.conf file (the only part that I modified to add my extension 1234:

[1234]
type=friend
auth=md5
username=1234
secret=obviouslynotmypassword
callerid="First Extension" <1234>
host=dynamic
nreinvite=no
canreinvite=no
disallow=all
allow=gsm
allow=ulaw

Maybe it is a timing issue, but I do not want to put in the X100P at this point as the whole point is to learn asterisk and how it works, not to shortcut every problem I encounter.

But I guess I could put it in to test and see if it does go away.

Otherwise, anyone knows what is going on, and how to remedy this? I’ve searched quite a bit and the others that had this didn’t have ztdummy.

I’ve tried to make this as clean/neat and detailed as I could, and not make a horrible wall of text, if I somehow did something wrong in terms of board etiquette please tell me and I’ll fix it.

Thanks to all.