I know I asked this some time back, and I got no response then, and
neither did someone who asked at the start of 2009 either by the looks of
it (other than a reply from me to use a PCI card!!!)

However I now have a client who’d bought one of these boxes and installed
ISDN2e – against my advice (because I got no feedback here, therefore
they’re junk with asterisk, right?)

However I’m currently stuck with it and it’s a right royal PITA to program
up by the look of it – I’ve done all the obvious stuff, and am now wading
through their 600+page manual, but ugh…

So if anyone has used one with asterisk as a bridge to ISDN2e in the UK,
would you mind sharing a config with me?

I have it registering (it pretends to be a SIP phone, it registers with
Asterisk), but when I try to send a call to it, it times out – I do see
debug on it’s console though – same when I send a call into it from the
PSTN side – I see ISDN debug messages, but no SIP activity.

And I thought Mediatrix units were bad – but at least they are relatively
easy to program…

Bah.

Any clues appreciated,

Thanks,

Gordon

hi:
our current connection is below:

sip phone< --->asterisk< ---->alcatel PBX< ---->PSTN

asterisk and alcatel PBX is connected via E1 isdn-pri.

when I use sip phone to dial outside PSTN world:
1. with 1.4 it is fine.
2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip
phone can not hear the ring and the beginning of the PSTN voice.
3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN
voice. I try to play options with “prematuremedia” and
“progressinband”. but I can not find working settings.

I don’t know what other options I can try.
thank a lot for information!!

I have call-limit=1 at sip.conf

From: danny@debsinc.com
To: asterisk-users@lists.digium.com
Date: Mon, 2 May 2011 12:20:40 -0500
Subject: Re: [asterisk-users] asterisk call completion issue

From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of satish patel

Sent: Monday, May 02, 2011 12:19
PM

To: asterisk-users

Subject: [asterisk-users] asterisk
call completion issue

Hi All,

I am testing CC feature with asterisk 1.8 but i am having some issue. We have
polycom 501 SIP phone and those are configured with two line with same
extensions. When i am requesting for CC i am not getting call back from
asterisk but it works if i reboot my polycom phone ( In short when phone get
register )

Is this because of two line configured ? or some configuration issue ?

[Danny Nicholas]

I would check call-limit and see what reducing that would do
for you.

Dan et al;

This looks like a perfect solution.

However, I have one issue. If I initiate the macro manually (put it in
the proper context/dialplan) it works. I see the *.wav file being
created and growing in the /var/spool/asterisk/monitor directory.

If I try to implement it adding the MixMonApp =>
*1,self/both,Macro,mixmon to the [applicationmap] in features.conf, I
cannot get it to work.

Steps.

1. added the example macro to the dialplan in extensions.conf
2. added the line MixMonApp => *1,self/both,Macro,mixmon to the
features.conf file under [applicationmap]
3. sip reload / dialplan reload / reload res_features
4. see the message that ‘Mapping Feature ‘apps’ to app ‘Macro(callrec)’
5. make incoming call – answer with SIP phone
6. I press *1 on the keypad, I hear the tones, but it does not begin
recording
7. see nothing in the CLI and no new files get created in
/var/spool/asterisk/monitor directory.

What am I missing? Probably something simple.

Any words of wisdom?

Glen

Synopsis

Transfer caller to remote extension.

Description

Requests the remote caller be transferred to a given destination. If TECH (SIP, IAX2, LOCAL etc) is used, only an incoming call with the same channel technology will be transfered. Note that for SIP, if you transfer before call is setup, a 302 redirect SIP message will be returned to the caller.

The result of the application will be reported in the The result of the application will be reported in the None - TRANSFERSTATUS channel variable:

• TRANSFERSTATUS -
  • SUCCESS - Transfer succeeded.
  • FAILURE - Transfer failed.
  • UNSUPPORTED - Transfer unsupported by channel driver.

Syntax

Transfer([Tech]destination)

Arguments

• dest
  • Tech
  • destination