* You are viewing the archive for May 30th, 2012

IAX ATA can’t register

I have an ATCOM ATA that is trying to connect to an asterisk server using IAX. The ATA and Asterisk are on the same subnet, not firewall/nat etc.

Below is a a log excerpt, showing the REGREQ received, and then Asterisk goes on to send lots of REGAUTH…and this continues for a while, but the ATA is never registered (iax2 show peers shows not registered).

Any help would be appreciated. It sure LOOKS like a lot of TX for very few RX frames…so my first guess was network related but I’m not making any progress with that theory


Tx-Frame Retry[001] — OSeqno: 003 ISeqno: 001 Type: IAX Subclass: PING
Timestamp: 21003ms SCall: 10027 DCall: 18442 []
Tx-Frame Retry[001] — OSeqno: 003 ISeqno: 001 Type: IAX Subclass: PING
Timestamp: 21006ms SCall: 14940 DCall: 18442 []
Rx-Frame Retry[Yes] — OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ
Timestamp: 00003ms SCall: 18443 DCall: 00000 []
Tx-Frame Retry[003] — OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH
Timestamp: 00014ms SCall: 02267 DCall: 18443 []
CHALLENGE : 121149566
Tx-Frame Retry[002] — OSeqno: 001 ISeqno: 001 Type: IAX Subclass: LAGRQ
Timestamp: 10013ms SCall: 02267 DCall: 18443 []
Tx-Frame Retry[001] — OSeqno: 002 ISeqno: 001 Type: IAX Subclass: LAGRQ
Timestamp: 20013ms SCall: 02267 DCall: 18443 []
Tx-Frame Retry[000] — OSeqno: 003 ISeqno: 001 Type: IAX Subclass: PING
Timestamp: 21016ms SCall: 10572 DCall: 18443 []
Tx-Frame Retry[002] — OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH
Timestamp: 00015ms SCall: 14112 DCall: 18443 []
CHALLENGE : 530555480
Tx-Frame Retry[001] — OSeqno: 003 ISeqno: 001 Type: IAX Subclass: PING
Timestamp: 21015ms SCall: 12659 DCall: 18443 []
Tx-Frame Retry[001] — OSeqno: 001 ISeqno: 001 Type: IAX Subclass: LAGRQ
Timestamp: 10014ms SCall: 14112 DCall: 18443 []
Tx-Frame Retry[000] — OSeqno: 002 ISeqno: 001 Type: IAX Subclass: LAGRQ
Timestamp: 20016ms SCall: 00480 DCall: 18443 []
Tx-Frame Retry[000] — OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH
Timestamp: 00002ms SCall: 05489 DCall: 18443 []
CHALLENGE : 399007934

Sangoma Card Issue

On Wed, May 30, 2012 at 02:34:55PM -0400, Eric Wieling wrote:
> Has anyone experienced an issue with Sangoma analog cards where
> the card suddenly stops working? Trying to dial out shows the
> channel as busy, even though there is no active call on that port?
> This happened to us often when we used Digium cards (in fact this
> issue is why we stopped using Digium).

Odd, I’m not aware of any current issues with Digium’s cards or
drivers which would leave it in an alarm state that wasn’t
attributable to an intermittent cabling issue. I’ve seen a loss of
voltage between the tip and ring (assuming you’re talking about FXO
ports) due to cabling issues that would leave the card in alarm. If
it’s something marginal you could sometimes play with the battery

I had seen some issues with the newer analog cards and stuck alarm
states that were addressed in r7517 “voicebus: send ‘idle’ buffers
when the transmit descriptor underruns” [1] which was first released
in dahdi-linux 2.3.0.

[1] http://svnview.digium.com/svn/dahdi?view=revision&revision=7517

If you still have the cards with which this problem occurred often,
are able to reproduce it with the dahdi-linux 2.6.1, I would be
interested in determining what the problem is. If you are interested
and able as well, shoot me an email off-list.

Asterisk 10.4.0 GotoIf to label problem when DUNDi active

I have a hotdesking environment at my main office, and up until today, the GotoIf that jumps straight to voicemail if a user isn’t log in was working just fine by label. Today, I deployed DUNDi to a satellite office, and now the GotoIf isn’t jumping to the right place. If I replace the label with a priority number, it jumps correctly. Alternatively, if I disable the switch statement for DUNDi, it jumps correctly. But with the DUNDi switch in service and the named label to jump to, it gives me this error:

[May 30 13:57:24] WARNING[6654]: pbx.c:10747 pbx_parseable_goto: Priority ‘not_logged_in’ must be a number > 0, or valid label

Dialplan snippets as follows:
[hotdesk] ;phones dial here
include => hotdesk_outbound

exten => _X.,1,NoOp()
same => n,Set(LOCATION=${CUT(CHANNEL,/,2)})
same => n,Set(LOCATION=${CUT(LOCATION,-,1)})
same => n,GotoIf($[${ISNULL(${WHO})}]?internal,${EXTEN},1)
same => n,Set(${WHO}_CID_NAME=${HOTDESK_INFO(cid_name,${WHO})})
same => n,Set(${WHO}_CID_NUMBER=${HOTDESK_INFO(cid_number,${WHO})})
same => n,Set(${WHO}_CONTEXT=${HOTDESK_INFO(defaultcontext,${WHO})})
same => n,Set(${WHO}_VMCONTEXT=${HOTDESK_INFO(vmcontext,${WHO})})
same => n,Set(GROUP(activecallers)=${WHO})
same => n,NoOp(Who: ${WHO} Calls: ${GROUP_COUNT(${WHO}@activecallers)})
same => n,Set(DEVICE_STATE(Custom:${WHO})=INUSE)
same => n,Set(CALLERID(name)=${${WHO}_CID_NAME})
same => n,Set(CALLERID(num)=${${WHO}_CID_NUMBER})
same => n,Goto(${${WHO}_CONTEXT},${EXTEN},1)

include => internal-privledged

include => internal
switch => DUNDi/peer

exten => _3XX,1,NoOp()
same => n,Set(E=${EXTEN})
same => n,Set(${E}_VMCONTEXT=${HOTDESK_INFO(vmcontext,${E})})
same => n,GotoIf($[${ODBCROWS} < 1]?not_logged_in)
same => n,Dial(SIP/${USER_LOCATION},20,wWU(answered^${E}))
same => n,ExecIf(${ISNULL(${E})}?NoOp(${HOTDESK_INFO(location,${E})}):ExecIf($[${GROUP_COUNT(${E}@activecalls)}>1]?Set(DEVICE_STATE(Custom:${E})=INUSE):Set(DEVICE_STATE(Custom:${E})=NOT_INUSE)))
same => n,Set(GROUP(activecalls)=${NULL})
same => n,Voicemail(${E}@${${E}_VMCONTEXT},b)
same => n,Hangup()
same => n(not_logged_in),Set(LOGGED_OFF=1)
same => n,Voicemail(${E}@${${E}_VMCONTEXT},u)
same => n,Hangup()

Any suggestions on other things to try? Or is this a bug I should file?

Thank you,

Noah Engelberth
MetaLINK Technologies

Introducing Limesco


Earlier this year I spoke to Malcolm Davenport from Digium about a mobile phone operator we’re starting in The Netherlands. He really liked the idea and proposed to send an email to this mailing list. Since this message is kind of spam and a little off-topic, I’ll limit myself to this single message and I also request any questions or replies to be kept of the list.

To the point: in The Netherlands we’re starting Limesco, a mobile phone operator aimed at IT specialists. What’s unique about Limesco, is that we want to bring more control to the end user, compared to what’s possible with existing operators.

Our most interesting offer from the start is usage of our gateway that translates signaling and voice streams between one of the existing mobile networks in +31 and the world of SIP and RTP. Essentially, this allows you to use any GSM or 3G compatible device, insert a Limesco SIM and use it as if it were an extension on an Asterisk server. All calls are routed over SIP without running special software on the mobile phone.

At this moment, we’re finishing up some organizational stuff and we want to start running with a closed pilot group soon. Normal usage charges apply during this period, but Limesco likes to reimburse part of the activation and monthly fee for pilot users that have nice ideas for experimentation and that also document their results on our wiki.

Since Limesco only operates in The Netherlands for now, most of our website and communication is in Dutch. However, if you’re interested in the project or the pilot, please become a member of “Vereniging Limesco” (with legal voting rights in the organization) at https://limesco.org/ or directly contact me by email.

Thank you,
Mark van Cuijk

Group call from DAHDI


I am trying to figure out how to make group call using DAHDI. I want to
make multiple call at once and conference among them. I know about meetme,
but that is for incoming conference call.

Please suggest.