[thameema]

I configured my xten pro work with my asterisk. If i call another xten the voice quality is clear. But if i make calls to pstn lines using my broadvoice the quality is ok for a minute and start fading out. I tried with SJPhone too and same problem.

I am sure most people must have the same configuration. Please help me configure the xten to work without any noise problems.

Another problem i have is, most of the times i m getting the following lines in my asterisk log.

Aug  2 20:22:28 NOTICE[2772]: rtp.c:515 ast_rtp_read: Unknown RTP codec 72 received
Aug  2 20:22:29 NOTICE[2772]: rtp.c:515 ast_rtp_read: Unknown RTP codec 72 received
Aug  2 20:22:34 NOTICE[2772]: rtp.c:515 ast_rtp_read: Unknown RTP codec 72 received

[mberlant]

What device are you using for your PSTN interface? What CODECs is it capable of supporting? What CODECs have you allowed for use with your Asterisk on this interface?

[thameema]

I am using sipura 3k for the interface. I am using ulaw in my asterisk for the pstn extension. If u need more details, please let me know

[mberlant]

So far, so good. What country are you in? Is your SPA's port's impedance set to match the requirements of your country's phone network?

[thameema]

i m in US. I think you didnt get my full question. I am calling from my XTEN to any landline number or mobile number, the voice quality is not that good and i m hearing noise.

[mberlant]

Yes, I was asking about the physical interface between your SPA3000's FXO port and your telephone company. If something is amiss here, like an impedance mismatch, it will adversely affect your outbound calling as you describe. Since you just noted that you are in the US, you should confirm that your FXO impedance is set for 600 ohms.

If it's not your impedance it might be the PSTN Line gain. Reduce the gain a little at a time, reboot the SPA, and try again until you find the best balance between voice level and noise level.

Now, this may not be your problem. Your original posting says that your Asterisk is receiving a bad CODEC request. It would be a good idea to confirm that the Asterisk and the SPA are both constrained to use the same CODEC.

By the way, what CODEC are your X-Lite computers using to talk to Asterisk? To each other?

[thameema]

Thanks mberlant. Xten clients are using ulaw and in my sip.conf i allow ulaw, alaw and gsm in the same order i described here. I will check the impedence in SPA for the pstn problem.

I think the problem is only with xten and if i fix the codec issue, then the noise problem get solved automatically...what you think?

[thameema]

One more update on this: As I was reading in voip-info.org i came across the following about G711 codec (ulaw or alaw) i am using with xten.

There are two versions: A-law and U-law. U-law is indigenous to the T1 standard used in North America and Japan. [b]The A-law is indigenous to the E1 standard used in the rest of the world[/b]. The difference is in the method the analog signal being sampled. In both schemes, the signal is not sampled linearly, but in a logarithmic fashion. A-law provides more dynamic range as opposed to U-law. The result is a less 'fuzzy' sound as sampling artifacts are better supressed.

So, if a person calling from gulf or asia (not japan) should use alaw codec? because i m using ulaw and i m in US, the quality is not a problem.

[mberlant]

It is possible that your X-Lite clients are ending up using GSM, despite the fact that there is sufficient bandwidth within your LAN to support G.711. You can confirm this by looking at the bottom of the X-Lite status window while the conversation is in progress. You will see the active CODEC in bold (not greyed out) and in a little box. Please confirm which CODEC is in use when connected to the other X-Lite and which is in use when connected via the SPA to the PSTN.

The SPA is incapable of GSM, so you should not have it allowed in sip.conf for this device. The problem may be with Asterisk trying to negotiate for the use of GSM and failing to do so (hence the error message). Independent of this, your Asterisk may be having trouble transcoding between GSM (on the X-Lite side) and G.711 (on the SPA side), giving you the poor voice quality you are complaining about.

[thameema]

Yes you are right. I had the gsm allowed in sip.conf for xten clients and removed it. But eventhough the quality is same and keep getting the error.
I searched the net and this forum for this kind of codec error messages associated to xten but there were no posts other than mine. Does it mean that everyone who uses xten didn't have any problem with codec....?

I want to tell you about another thing which is banging my head. I have my asterisk server running in datacenter in my dedicated server. I don't have any TDM or zaptel cards. So, when I installed asterisk i didn't install zaptel and libpri packages as I don't have any cards installed. Is this acceptable or do i need to install all the packages even though i don't have the cards installed? Does it do anything with poor voice quality?

Sorry for too many confusing thoughts...:-)