Incoming Sound Choppy Asterisk(Trixbox) to Uplink

[dcbour]

I just installed Uplink yesterday but am having consistently choppy sound on the incoming sound. According to the people at the other end, it sounds fine to them. I see a couple of posts regarding this, but no replies.

In my case, it's a Trixbox install.

 

I've got a VTech portable Skype phone running off the same system where the Uplink and Skype are installed. Sound quality is fine there.

 

The Asterisk is a Trixbox 2.2 installation, no zaptel devices (ie, for timing).

 

I've only configured it for outgoing calls at this point. My other trunk on the system runs flawlessly.

 

Any suggestions as this is unusable in this state.

 

Thanks in advance

Dave

[ghannelais]

In my case, it's a Trixbox install.

 

The Asterisk is a Trixbox 2.2 installation, no zaptel devices (ie, for timing).

 

Dave

 

Hi Dave

Trixbox here too (v. 1.2.3) with Zaptel board and no problem with choppy sound.

Asterisk on a Linux server

Uplink on a Windows server

SIP phones are Linksys SPA942 on all work stations

 

This is my Trixbox trunk settings in case it could help:

allow=ulaw

canreinvite=yes

context=from-trunk

disallow=all

host=dynamic

nat=yes ;very important. uplink won't work without this

secret=xxxx ;define your own password as in uplink

type=friend

username=xxxx ;make sure this is same to trunk name

 

Good luck,

Georges

[dcbour]

Had type = peer, changed to friend, no change.

Checked if I had ztdummy module loaded, shows in the lsmod list. Can't think of anything else. Any help / suggestions appreciated.

 

Of note...10-20% of calls from internal sip softphones are fine...balance are choppy...any call routing through via disa ALWAYS choppy.

 

Thanks in advance

D.

[cvega99]

username=xxxx ;make sure this is same to trunk name

 

The above did it for me. As soon as I made the user name the same as the trunk it worked well. I have one problem though. When I use express talk I can dial out because I dial

 

9+1(AREACODE)NUMBER

 

However if I use a regular line or a DISA from outside I don't have a "+" to prefix the number.

 

Is there a way to prefix a + for this trunk in Trixbox? Or can this be done from the dial section in the UPLINK software?

 

Thanks in advance.

[nmsf]

I've managed to configure uplink with trixbox. The sound is ok. I can receive skype calls on my extension. But i don't know how to make skype call (not skypeout). Anyone?

[z_smurf]

I just installed Uplink yesterday but am having consistently choppy sound on the incoming sound. According to the people at the other end, it sounds fine to them. I see a couple of posts regarding this, but no replies.

In my case, it's a Trixbox install.

 

I've got a VTech portable Skype phone running off the same system where the Uplink and Skype are installed. Sound quality is fine there.

 

The Asterisk is a Trixbox 2.2 installation, no zaptel devices (ie, for timing).

 

I've only configured it for outgoing calls at this point. My other trunk on the system runs flawlessly.

 

Any suggestions as this is unusable in this state.

 

Thanks in advance

Dave

 

Hi!

I am also having very choppy sound with The Uplink with the following setup:

 

Skype (windows) -> Uplink (windows) -> Asterisk 1.2 (Linux) -> Zaptel -> Old ISDN PBX -> Old Digital office phone

 

The following setup gives perfect sound though:

 

Skype (windows) -> Uplink (windows) -> Sipura adaptor -> Normal analog phone

 

Anyone have any idea why sound is choppy through asterisk/Zaptel? Bad timing or something?

 

//z_smurf

[itjumper]

Hi!

I am also having very choppy sound with The Uplink with the following setup:

 

Skype (windows) -> Uplink (windows) -> Asterisk 1.2 (Linux) -> Zaptel -> Old ISDN PBX -> Old Digital office phone

 

The following setup gives perfect sound though:

 

Skype (windows) -> Uplink (windows) -> Sipura adaptor -> Normal analog phone

 

Anyone have any idea why sound is choppy through asterisk/Zaptel? Bad timing or something?

 

//z_smurf

 

 

Still have this problem with calls coming from zaptel. Anyone has a fix ??? Sip is perfect

[markosjal]

Still have this problem with calls coming from zaptel. Anyone has a fix ??? Sip is perfect

 

More than likely this is because asterisk 1.2 has no real jitter buffers. This may affect your connections are far from the internet backbone.

 

You may be able to minimize this effect by setting up reinvites. canreinvite = yes, among other prerequisites. This depends much on your configuration however which I know little about.