[Asterisk-Users] Unable to forward frame/voice

[Christian M. Watts]

Hi,

We've exhausted our internal capabilities as well as Sangoma tech support and
were hoping someone with some expertise could help us with a pointer. Briefly,
our issue is as follows.

Periodically (several times an hour), we get either of the following error
messages in our asterisk messages log. These correspond with dropped outbound
calls on a one-to-one basis when the second error happens. The first error
sometimes causes a dropped call and sometimes does not:

Jun 30 16:40:27 WARNING[5395] app_dial.c: Unable to forward frame
Jun 30 16:45:07 WARNING[5455] app_dial.c: Unable to forward voice


Our hardware is as follows:

Compaq DL380 Dual PIII 1Ghz, 1.2 GB RAM, Onboard SmartArray for SCSI RAID
Sangoma A102U dual-port T1 card
Digi Datafire T1 fax/modem board


Our software is as follows:

Linux 2.4.30
Asterisk, Zaptel and Libpri from CVS HEAD as of 6/28/05
Sangoma wanpipe 2.3.3-beta11 (latest as of this post)
Patton electronic's latest drivers and firmware for our Digi Datafire board
(still no 2.6 Linux support, which is why we're on 2.4)
Hylafax 4.2.1 driving the Digi Datafire


The path (for the problem calls) looks like this:

Digi Datafire -> Sangoma Port B -> Sangoma Port A -> Telco

Basically, sending a fax over a PRI with asterisk doing TDM bridging in the
middle.


We have confirmed the following (based on similar posts to this list related to
the same problem with Digium boards as well as Sangoma tech support
assistance):

1. Sangoma Port A takes clocking from the telco
2. Sangoma Port B retransmits A's clocking and acts as master
3. Sangoma tech support says our configs are correct
4. Zaptel.conf is set up with Sangoma Port A as the primary clock source, and
Port B to not be used as a clock source
5. LBO, switch options, etc. are correct for the environment (since 98% of
outbound calls are fine, this seems fairly obvious)
6. ISDN Transfer Capability gets properly set to 3K1AUDIO for calls
7. No IRQ sharing on the system
8. IDE DMA mode is irrelevant, since there are no IDE disks in the system (other
than the CDROM)


We have tried the following:

1. Asterisk, libpri and zaptel versions from 6/1/2005, 6/15/2005 and 6/28/2005 -
no change in behavior
2. Wanpipe drivers 2.3.3-beta8 and 2.3.3-beta11 - no change in behavior
3. Wanpipe configured both with and without the D-Channel hardware HDLC - no
change in behavior
4. Firmware versions 24 (shipped version) and 25 (latest version) on the Sangoma
card - no change in behavior
5. callprogress and busydetect both 'yes' and 'no' in zapata.conf (currently
'no') - no change in behavior
6. Added SetTransferCapability(3K1AUDIO) to our dialplan, just to be sure - no
change in behavior


General environment:

1. We run TDM only, no VoIP protocols are in use. SIP, IAX2, MGCP are all noload
in modules.conf.
2. This problem occurs with as few as one simultaneous channel active and as
many as 15 simultaneous channels active with equal frequency (i.e.: not load
related). The load on the box is negligible in any case, plenty of RAM is free,
etc.
3. Restarting asterisk does seem to cause the problem not to re-present itself
for 30 minutes to 2 hours. When asterisk is restarted, the Sangoma and Zaptel
kernel modules are also unloaded and reloaded.


Again, any pointers or help would be greatly appreciated.

Thanks,
Christian