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Summary:ASTERISK-07288: One way audio, zap->*->IAXY Dropping incompatible voice frame
Reporter:Jonathan Galpin (jlgdeveloper)Labels:
Date Opened:2006-07-05 09:11:54Date Closed:2006-09-12 14:47:46
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_iax2
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:When using an IAXY modem and receiving a call from the pstn, audio from the iaxy to the pstn is dropped.

The issue only affects incoming PSTN calls. pstn(ulaw) --> *(slin) --> iaxy(iax2) --> (ulaw)analog handset. Outgoing calls the same route are fine as well as sip phones --> * --> iaxy --> handset.

I did a fresh install of Libpri 1.2.3, Zaptel 1.2.6 and Asterisk 1.2.9.1, Linux 2.6 kernel. I am using a Digium TMDXXB Card.

The * server is in one location, port 4569 open, and the iaxy device is behind nat. Not a firewall issue since opening port 4569 to the iaxy or putting the iaxy on the dmz does not change it.

Thanks, Jonathan

Call log follows:

   -- Starting simple switch on 'Zap/1-1'
   -- Executing NoOp("Zap/1-1", "Incoming call --> 813909XXXX") in new stack
   -- Executing Wait("Zap/1-1", "1") in new stack
   -- Executing Set("Zap/1-1", "TIMEOUT(digit)=5") in new stack
   -- Digit timeout set to 5
   -- Executing Set("Zap/1-1", "TIMEOUT(response)=15") in new stack
   -- Response timeout set to 15
   -- Executing BackGround("Zap/1-1", "custom/business-hours-menu") in new stack
   -- Playing 'custom/business-hours-menu' (language 'en')
 == CDR updated on Zap/1-1
   -- Executing Dial("Zap/1-1", "SIP/sip-carmen-desk&SIP/sip-annie-desk&IAX2/annie-home&SIP/sip-jon-desk|30|mt") in new stack
   -- Called sip-carmen-desk
   -- Called sip-annie-desk
   -- Called annie-home
   -- Called sip-jon-desk
   -- Started music on hold, class 'native', on Zap/1-1
   -- SIP/sip-carmen-desk-6b8b is ringing
   -- SIP/sip-jon-desk-f9f6 is ringing
   -- Call accepted by 71.100.16.XXX (format ulaw)
   -- Format for call is ulaw
   -- IAX2/annie-home-6 is ringing
   -- SIP/sip-annie-desk-bb83 is ringing
   -- IAX2/annie-home-6 answered Zap/1-1
   -- Stopped music on hold on Zap/1-1
Jul 3 15:23:49 NOTICE[3058]: channel.c:1904 ast_read: Dropping incompatible voice frame on IAX2/annie-home-6 of format slin since our native format has changed to ulaw
Jul 3 15:23:49 NOTICE[3058]: channel.c:1904 ast_read: Dropping incompatible voice frame on IAX2/annie-home-6 of format slin since our native format has changed to ulaw
Jul 3 15:23:49 NOTICE[3058]: channel.c:1904 ast_read: Dropping incompatible voice frame on IAX2/annie-home-6 of format slin since our native format has changed to ulaw
Jul 3 15:23:49 NOTICE[3058]: channel.c:1904 ast_read: Dropping incompatible voice frame on IAX2/annie-home-6 of format slin since our native format has changed to ulaw
Jul 3 15:23:50 NOTICE[3058]: channel.c:1904 ast_read: Dropping incompatible voice frame on IAX2/annie-home-6 of format slin since our native format has changed to ulaw
   -- Hungup 'IAX2/annie-home-6'
 == Spawn extension (default, 1, 1) exited non-zero on 'Zap/1-1'
   -- Hungup 'Zap/1-1'


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

Setting transcode_via_sln=no and restarting * has no effect.

Iax.conf
========
[annie-home]
type=friend
accountcode=iaxyanniehome
host=dynamic
context=internal-staff
auth=md5
secret=xxxxxxxxxxxxx
dissalow=all
allow=ulaw
callerid="xxxxxxx" <(813) 000 0000>
qualify=yes

IAXY provisioning
=================
; IAXY Provisioning description
;
dhcp
;ip: 216.207.244.xxx
;netmask: 255.255.255.xxx
;gateway: 216.207.244.xxx
codec: ulaw
;codec: adpcm
server: 70.xxx.xxx.xxx
altserver: 192.168.0.18
user: annie-home
pass: xxxxxxxxxxx
register
;heartbeat
;debug

This is a similar issue:
http://bugs.digium.com/view.php?id=4101

load log re chan zap
====================
   -- Reloading module 'chan_zap.so' (Zapata Telephony w/PRI)
 == Parsing '/etc/asterisk/zapata.conf': Found
Jul 3 15:58:52 WARNING[3561]: chan_zap.c:10886 setup_zap: Ignoring signalling
   -- Reconfigured channel 1, FXS Kewlstart signalling
   -- Reconfigured channel 2, FXS Kewlstart signalling
   -- Reloading module 'chan_mgcp.so' (Media Gateway Control Protocol (MGCP))
   -- Reloading module 'app_queue.so' (True Call Queueing)
 == Parsing '/etc/asterisk/queues.conf': Found
   -- Reloading module 'cdr_pgsql.so' (PostgreSQL CDR Backend)
 == Parsing '/etc/asterisk/cdr_pgsql.conf': Found
   -- Reloading module 'cdr_custom.so' (Customizable Comma Separated Values CDR Backend)
 == Parsing '/etc/asterisk/cdr_custom.conf': Found
   -- Reloading module 'codec_adpcm.so' (Adaptive Differential PCM Coder/Decoder)
 == Parsing '/etc/asterisk/codecs.conf': Found
   -- codec_adpcm: using generic PLC
   -- Reloading module 'codec_ulaw.so' (Mu-law Coder/Decoder)
Reloading SIP
 == Parsing '/etc/asterisk/sip.conf': Found
Reloading MGCP
 == Parsing '/etc/asterisk/mgcp.conf': Found
   -- Registered IAX2 to '64.61.93.90', who sees us as 70.124.132.xxx:4569 with no messages waiting

   -- Registered IAX2 to '64.61.93.87', who sees us as 70.124.132.xxx:3405 with no messages waiting

 == Parsing '/etc/asterisk/sip_notify.conf': Found
Jul 3 15:58:52 WARNING[2168]: chan_mgcp.c:4209 reload_config: Unable to get our IP address, MGCP disabled

==================================================
Comments:By: Jonathan Galpin (jlgdeveloper) 2006-07-14 19:37:29

Changing the codec to adpcm does not solve the issue either. The error message is the same except the reference to the codec ulaw is now adpcm.
By: Serge Vecher (serge-v) 2006-07-17 08:20:10

hmmm, I don't see chan_local involved -- are you sure that's where the problem is? Perhaps chan_iax2?
By: Jonathan Galpin (jlgdeveloper) 2006-07-28 03:06:22

IAX2 will suit me if it suits you. I am not knowledgable enough yet to characterize it, and was following that similar issue above. I assumed chan-local is the slin? Am presently travelling in Europe, using the iaxy. Back next week.
By: Clod Patry (junky) 2006-08-08 22:33:24

what about if you allow=slin for that annie-home?

just to confirm,
annie-home is:
Registered IAX2 to '64.61.93.90', who sees us as 70.124.132.xxx:4569 or the other registration?
By: Serge Vecher (serge-v) 2006-08-25 11:10:10

ok, what happened here? Is this still an issue in 1.2.11?
By: Clod Patry (junky) 2006-08-25 12:10:19

closing, since no responde.
If you can give us more infos, feel free to re-open.
By: Jonathan Galpin (jlgdeveloper) 2006-08-25 17:50:35

I will be testing based on the above comments. Just allow a little more time.
By: Jonathan Galpin (jlgdeveloper) 2006-09-11 12:35:48

I tested the issue on fresh installs of both asterisk 1.2.8 and 1.2.12, and have found that either it has been resolved, or for some other reason I am no longer experiencing the issue.

For some reason, I could not get revision 42669 and the zaptel trunk to recognize my tmd card....but that is a separate issue.

Lets close the item, thanks for your attention.
By: Joshua C. Colp (jcolp) 2006-09-11 12:51:33

Fixed (perhaps from my set_format change).