OK, it’s been a while since I drank from the pool of wisdom hear on the list.

After cracking my head against the wall for a few days trying to figure this out, I have decided to swallow my pride and take the drink.

So, on to my question:

I have some agents/operators setup in sip.conf which point to a context where I have just about disabled outbound calls (only specific numbers can be dialed).

The purpose of this is to allow the inbound calls to come in, then if the operator has a need, they transfer the call to a pre-defined extension which lives in the limited context defined in sip.conf.

This has worked for some time to restrict outbound calling and where calls can be transferred to.

Now I would like to open up the numbers the inbound calls can be transferred to. So, easy enough I thought and I went on my merry way adding the regular patterns to the context such as NXXNXXXXXX and so on.

Hooray, now the operators can transfer anywhere.

New Problem, now operators can pick up the previous inbound only line and dial out to anything that matches the patterns I have defined in the context for their extension in sip.conf.

What I really need to make work here is Attended-Transfer since that is what is desired by those using the system.

It seems that any variables I try to set on the way in don’t carry through too well during an attended transfer.

Basically, I need the ability to know for sure at the point the call ends up in the outbound context (defined in sip.conf) if the call is actually a transfer from an inbound call or if it’s a direct dial outbound call with no incoming call attached. If I can figure out how to know this for sure, I
can just do a GoToIf type of thing in the outbound context that just kills the call if there is no proof that it’s a transfer.

I hope this makes sense, please let me know if more info is needed.

Running Asterisk 1.8.8.0.

A huge thanks in advance to the list for any help with this, it’s driving me batty.

Regards, Todd R.

Hello,
I have noticed some occasional one way audio on a specific sub set of calls in my system. First, let me be more specific about what I mean about occasional one way audio. Unlike most of the posts I’ve seen (where the end fix was either NAT’ing or RTP issues) the calls in question will begin and progress just fine, and then in one direction (always the same direction) audio stops for a second or two. When this occurs, it seems like every other, or every third word is cut out for 5-15 seconds (estimate, I have not timed it), and then the call returns to normal. I experienced this probably 5 separate times during a 1.5 hour call.

A bit about the setup:
The inbound calls in question come in to an Asterisk system over an E1 line in Scotland, and are forwarded to another Asterisk system in the US over an IAX2 connection. The call is then delivered via SIP to either a handset (various models) or soft phone (various packages). After the call has progressed for some time, audio from me to the caller drops every other word for a few seconds.

On the Asterisk server with the E1 connection I see a fairly steady stream of the following error when watching the console (asterisk -cvvvr):
WARNING[1175660480]: chan_zap.c:6147 zt_pri_error: PRI: Read on 81 failed: Unknown error 500
NOTICE[1175660480]: chan_zap.c:6873 pri_dchannel: PRI got event: 8 on span 1
WARNING[1150495040]: chan_iax2.c:516 iax_error_output: Ignoring unknown information element ‘Unknown IE

The last of those is the most common.

I have checked the following:
Server/Network port speed/duplex match both sides allow the following codecs: ulaw (one side also has alaw/gsm)
jitterbuffer settings match each side has the other as an IAX friend, and registers to its peer


The UK server is running an old SVN version of Asterisk, and the US server is running 1.4.31, and performing an upgrade on either of these servers is not possible at this time.

Pleas advise on what else I can check to correct this issue.

Converting this customer from a MiTel system to asterisk. Discovered that the inbound calls from the T1 are going to extension 366. (This was mapped in the MiTel for some arcane purpose.) The dial plan I am currently using is shown below. When loading the dial plan, I get this warning:

WARNING[5004]: pbx_config.c:1561 pbx_load_config: The use of ‘_.’ for an extension is strongly discouraged and can have unexpected behavior.
Please use ‘_X.’ instead at line 331 of extensions.conf

Question: Do I need to worry about this warning?

I’m a little leery of just using 366 in the dialplan, since the company we are dealing with is a little flaky.


[from-pstn]
exten =>s,1,NoOp(pstn call from ${CALLERID(all)} exten=${EXTEN})
same =>n,Goto(MainMenu,s,1)
exten =>_.,1,NoOp(pstn call from ${CALLERID(all)} exten=${EXTEN})
same =>n,Goto(MainMenu,s,1)

If this is the wrong place to post this I’m sure someone will let me know.

I’m looking for a reliable, inexpensive call termination service
(SIP). The one I am presently with does not seem to know what IPs they send inbound calls from, and it is maddening to keep up with the FW
changes necessary, not to mention the calls which are not connected.

Kind regards, Chris

I know that I should know this. But I’m having serious brain farts at the
moment.

I want to have a call be counted in a number of ways

outbound
inbound
potential

so, for example, a call comes into my dialplan, I want to add it to

TotalCalls (all calls inbound + outbound)
InboundCalls (all inbound calls)
Potenial (all inbound calls that may potentially enter a conference)
Potenial (all inbound calls that may potentially enter a conference by
conference name)

the potential category needs to be decremented when the call actually
enters a conference

so, I have

exten => _[0-9A-Za-z].,n,Set(GROUP(conference)=TotalCalls)
exten => _[0-9A-Za-z].,n,Set(GROUP(conference)=Inbound)
exten => _[0-9A-Za-z].,n,Set(GROUP(conference)= Potenial)
exten => _[0-9A-Za-z].,n,Set(GROUP()= Potenial_${CONFNAME})

group_count(inbound)
group_count(inbound@conference)
group_count(potential@conference)
group_count(potential_${CONFNAME} )

however, when the call enters the conference, I need to unset the potential
calls groups

how do I do this ?

/me feels very very stupid

Julian