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Summary:ASTERISK-08106: When the phone rings on incoming calls, there are a lot of error messages
Reporter:aslomp (aslomp)Labels:
Date Opened:2006-11-10 10:02:20.000-0600Date Closed:2007-01-15 23:30:19.000-0600
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/CodecHandling
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) sip.conf
( 1) verb.txt
( 2) verbose.txt
( 3) verbose1.txt
Description:Whenever the phone is picked up, there are a lot of errormessages.
Often is it
Nov 10 16:53:05 WARNING[24350]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 64/8) and is there one way audio.

Now I've changed the codecs, and are this the error messages:
Nov 10 16:53:05 WARNING[24350]: file.c:512 ast_openstream_full: File 90 does not exist in any format
Nov 10 16:53:05 WARNING[24350]: file.c:824 ast_streamfile: Unable to open 90 (format alaw): No such file or directory
Nov 10 16:53:05 WARNING[24350]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 8/8)
Nov 10 16:53:05 WARNING[24350]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 8/8)
Nov 10 16:53:05 WARNING[24350]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 8/8)
Nov 10 16:53:05 WARNING[24350]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 8/8)
Nov 10 16:53:05 WARNING[24350]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 8/8)
Nov 10 16:53:05 WARNING[24350]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 8/8)
Nov 10 16:53:05 WARNING[24350]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 8/8)
Nov 10 16:53:05 WARNING[24350]: chan_sip.c:2570 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 8/8)

****** ADDITIONAL INFORMATION ******

sip.conf

[general]
disallow=gsm
allow=all
disallow=gsm

[budgetphone] // is incoming context
disallow=all
allow=alaw
allow=ulaw
;allow=gsm
Comments:By: Olle Johansson (oej) 2006-11-10 13:04:46.000-0600

Please read the bug guidelines. We need a SIP debug according to those instructions to be able to help you.
By: Serge Vecher (serge-v) 2006-11-15 11:21:10.000-0600

1) Prepare test environment (reduce the amount of unrelated traffic on the server);
2) Make sure your logger.conf has the following line:
  console => notice,warning,error,debug
3) restart Asterisk with the following command:
  'asterisk -Tvvvvvdddddgc | tee /tmp/verbosedebug.txt'
4) Enable SIP transaction logging with the following CLI commands:
set debug 4
set verbose 4
sip debug
5) Attach verbosedebug.txt to the issue.
By: Olle Johansson (oej) 2006-11-15 14:37:37.000-0600

You have a problem with a missing file there.
By: Joshua C. Colp (jcolp) 2006-11-15 14:44:00.000-0600

Full console output would be great too. This might be a case where if a file isn't found then the translation path isn't restored correctly and funky stuff (like the above) happens... definitely need more information though.
By: aslomp (aslomp) 2006-11-16 10:45:48.000-0600

The error is the same, but now it wasn't 8/8 but 64/8 (line 501) (see attached file above)


By: Joshua C. Colp (jcolp) 2006-11-16 18:47:11.000-0600

What if you don't use the announceoverride capability or use a file that does indeed exist? Does it work as expected? I'm just trying to narrow down where the issue might be.
By: aslomp (aslomp) 2006-11-17 06:34:16.000-0600

The announceoverride is now disabled, but the errors remain from chan_sip
The problem isn't the missing file (which was misconfiguration by forgetting an | ).

but it's about the chan_sip. When the phone is picked up, there is no audio on either side.


By: Serge Vecher (serge-v) 2006-11-20 14:37:52.000-0600

ok, is there any difference if you upgrade to 1.2.13? Also, why is nat enabled?
By: aslomp (aslomp) 2006-11-22 10:59:34.000-0600

I'm at this moment upgrading to the latest 1.2 version (1.2.13)

NAT is enabled cause the phones are all on private IP's
By: aslomp (aslomp) 2006-11-22 11:12:17.000-0600

This is the latest version of Asterisk (1.2.13)
Still the same error remains
By: Joshua C. Colp (jcolp) 2006-11-22 11:15:28.000-0600

Okay, here is what I need to see:

1. sip.conf entries for relevant devices
2. COMPLETE verbose, not just the specific portion with the error on the screen over and over.

Hopefully we'll get to the bottom of this.
By: aslomp (aslomp) 2006-11-23 10:45:39.000-0600

complete verbose added (verb.txt) and sip.conf (still unchanged)
By: Joshua C. Colp (jcolp) 2006-12-06 10:19:06.000-0600

I'm back at. Can you please do a "show channel <insert channel name here>" on each channel on each side of the call? Thanks.
By: Serge Vecher (serge-v) 2007-01-09 13:17:41.000-0600

aslomp: we need your feedback.
By: Joshua C. Colp (jcolp) 2007-01-15 23:30:18.000-0600

I'm closing this out for now since it has been so long. If you get the needed information and it is still an issue please reopen this. Thanks!