[asterisk-users] Random call drops on IAX2

[Sebastian]

anybody?

On 11/10/2010 06:51 PM, Sebastian wrote:
> Hello list,
>
> I have an Asterisk setup with the following details:
>
> 1. 3 x internal extensions / sip hardphones - Grandstream 2000
> 2. 2 x internal extensions / dahdi cordless phone
> 3. 1 x 2 FSX ports OpenVOX pci card
> 4. 1 x internal sip extension / sip softphone (linphone)
> 5. 1 x 800Mhz Asterisk + Linux server
> 6. Asterisk version is 1.6.2.13
> 7. 1 x IAX2 incoming trunk from phone provider for 1 phone number (2
> channels).
> 8. 1 x IAX2 outgoing trunk (theoretically unlimited channels) to phone
> provider.
> 9. Internet connection used is ADSL 8mbs/350kbs.
> 10. All internet traffic goes through the Linux/Asterisk box (2 NIC's)
> and I use basic traffic shaping/priority using tos_sip/tos_audio/tos/cos
> directives in sip.conf and iax.conf.
> 11. The server doesn't have much load - some basic samba file sharing
> for 5 computers, some dovecot IMAP email for same machines.
>
>
> The setup has worked fine for it's first 8 months. After that, users
> started to report dropped calls. Most of the time (but doesn't seem to
> be always) it seems to be the following scenario:
>
> 1. A member of staff places an outgoing call.
> 2. While in conversation - an external call is coming in.
> 3. The original outgoing call is cut off.
>
> However, this is not consistent. I've tried to trigger it when on site -
> and didn't manage. It tends to happen randomly, about 1-3 times per day.
>
> What I've done:
>
> 1. Measured the uplink bandwidth - it works out consistently at about
> 350kbs - enough for (theoretically) up to 4 simultaneous ulaw/alaw calls
> (80kbs each).
> 2. There isn't much internet traffic generated by the local network -
> and the pattern of call drops doesn't match any spike in Internet
> activity from/to local network.
> 3. I've managed to put through 4 concurrent incoming and outgoing calls
> - and left them on for about 30 minutes - none of them cut off during
> this time.
> 4. I've tested the RAM on the server - couldn't discover any fault.
> 5. I've called the IAX provider and so far they are adamant that the
> problem is not at their end.
> 6. I've upgraded Asterisk to latest 1.6 trunk - 1.6.2.13 (use to be
> version 1.6.1.6)
>
> I have kept and eye on the Asterisk logs - but just cannot find what is
> happening. I have asked users to let me know the time of call drops - so
> that I can look at specific portions of the Asterisk log. I attach below
> the log section concerned. I've been told that the user placed an
> outgoing call  - and they were cut off. I personally don't see anything
> in the log - it looks like a clean hang-up to me. I don't even know
> where to narrow things down. Is it the server hardware, is it the
> internet connection, is it the ADSL modem/router upstream of the server,
> is it the Grandstream phones, or is it the IAX provider?
>
> As you can see below:
>
> 1. The outgoing call starts at 11:22:45
> 2. At 11:22:50 another previous incoming call ends.
> 3. Then the original outgoing call gets answered at 11:22:53
> 4. Then about a minute later, at 11:23:51, the outgoing call gets hanged up.
>
> The user claims the outgoing call got cut off - but I don't see any
> error in the logs:
>
> [Nov 10 11:22:22] VERBOSE[4365] pbx.c:   == Spawn extension
> (from_internal_phones, 23, 1) exited non-zero on 'SIP/25-00000087'
> [Nov 10 11:22:45] VERBOSE[3242] netsock.c:   == Using SIP RTP TOS bits 184
> [Nov 10 11:22:45] VERBOSE[3242] netsock.c:   == Using SIP RTP CoS mark 5
> [Nov 10 11:22:45] VERBOSE[4366] pbx.c:     -- Executing
> [901512541010 at from_internal_phones:1] NoOp("SIP/23-00000089", "Adding
> call to trunk group") in new stack
> [Nov 10 11:22:45] VERBOSE[4366] pbx.c:     -- Executing
> [901512541010 at from_internal_phones:2] Set("SIP/23-00000089",
> "GROUP()=TRUNK_GROUP") in new stack
> [Nov 10 11:22:45] VERBOSE[4366] pbx.c:     -- Executing
> [901512541010 at from_internal_phones:3] Dial("SIP/23-00000089",
> "IAX2/ihs_trunk_out/01512541010") in new stack
> [Nov 10 11:22:45] VERBOSE[4366] app_dial.c:     -- Called
> ihs_trunk_out/01512541010
> [Nov 10 11:22:45] VERBOSE[3237] chan_iax2.c:     -- Call accepted by
> ip.address.rem.oved (format ulaw)
> [Nov 10 11:22:45] VERBOSE[3237] chan_iax2.c:     -- Format for call is ulaw
> [Nov 10 11:22:50] VERBOSE[4364] pbx.c:   == Spawn extension
> (from_trunks, proceed, 3) exited non-zero on 'IAX2/01519226927_in-2238'
> [Nov 10 11:22:50] VERBOSE[4364] chan_iax2.c:     -- Hungup
> 'IAX2/01519226927_in-2238'
> [Nov 10 11:22:52] VERBOSE[4366] app_dial.c:     --
> IAX2/ihs_trunk_out-1579 is ringing
> [Nov 10 11:22:52] VERBOSE[4366] app_dial.c:     --
> IAX2/ihs_trunk_out-1579 is making progress passing it to SIP/23-00000089
> [Nov 10 11:22:53] VERBOSE[4366] app_dial.c:     --
> IAX2/ihs_trunk_out-1579 stopped sounds
> [Nov 10 11:22:53] VERBOSE[4366] app_dial.c:     --
> IAX2/ihs_trunk_out-1579 answered SIP/23-00000089
> [Nov 10 11:24:51] VERBOSE[4366] chan_iax2.c:     -- Hungup
> 'IAX2/ihs_trunk_out-1579'
> [Nov 10 11:24:51] VERBOSE[4366] pbx.c:   == Spawn extension
> (from_internal_phones, 901512541010, 3) exited non-zero on 'SIP/23-00000089'
> [Nov 10 11:27:43] VERBOSE[3242] chan_sip.c:     -- Registered SIP '23'
> at 192.168.12.16 port 5064
>
>
> Any ideas or comments on this one are much appreciated. I'm not sure
> where else to look to get more relevant information.
>
> Sebastian
>