[asterisk-users] AGI: Not getting answers from get_data in a call-file call

it to 1.4 it gives this strange behaviour. Is there anybody with a hint on how to resolve this? Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://www.mavetju.org/weblog

Re: [asterisk-users] AGI: Not getting answers from get_data in a call-file call

and then drops into the callback2.agi script which is the one I had in the earlier email. As I said earlier: nothing strange, nothing spectacular. Just strange that it doesn't work for calls initiated by Asterisk. -- Edwin Groothuis |Personal website: http://www.mavetju.org

Re: [asterisk-users] AGI: Not getting answers from get_data in a call-file call

-- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://www.mavetju.org/weblog/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing

[asterisk-users] Maximum retries for seqno 102 when re-inviting.

for it: http://bugs.digium.com/view.php?id=10449 The output of sip debug is funny from line 366, where it is transmitting and re-transmitting a lot of re-invites back to the softswitch with CSeq 102. Has somebody else seen this behaviour, and know how to resolve it? Edwin -- Edwin Groothuis

[asterisk-users] RAD IPmux8

-- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://www.mavetju.org/weblog/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE

[asterisk-users] Tipping Point IPS blocking Asterisk SIP quaility messages

://.eee.oulu.fi/research/ouspq/protos/ It seems to be default to block, which will cause a couple of issues for people today :-) Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin

[asterisk-users] Re: asterisk-users Digest, Vol 30, Issue 4

on app_rxfax.so you can confirm that is is being linked. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ --Bandwidth and Colocation provided

[asterisk-users] HP DL380 and the TE4xxP cards

Hello, Does anybody have experience with the Quad T1/E1 PRI cards in an HP DL380? Just a yes it works fine or a never again is enough :-) Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au

[Asterisk-Users] Remember the incoming context?

. And you can use Dial(Local/number/initialcontext) for that. Oops, this initial context is lost somewhere on the line. Unless I'm very mistaken, there is no way to find out what the original context was, is there? Edwin -- Edwin Groothuis |Personal website: http

Re: [Asterisk-Users] Remember the incoming context?

, but then require the right context. Looks like I have to submit a patch! Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ --Bandwidth

[Asterisk-Users] Blocked channels, according to our telco... leading to CONGESTION status

! Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list

[Asterisk-Users] Re: Incoming Calls Getting Crossed - Weird

though. I have something similar with PRI - PRI and PRI - SIP calls http://bugs.digium.com/view.php?id=6502 Nothing heard from Digium Support yet. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http

[Asterisk-Users] Update to the latest zaptel driver - Congestion gone, but scary write errors replaced it

:03:08 DEBUG[19274] chan_zap.c: Write returned -1 (Resource temporarily unavailable) on channel 216 Feb 17 10:03:34 DEBUG[19334] chan_zap.c: Write returned -1 (Resource temporarily unavailable) on channel 2 That is on zap-channels. What is going on? Edwin -- Edwin Groothuis

[Asterisk-Users] Re: dual TE410, both span 3 is broken (Josh Krueger)

some new' quad PRI cards (2nd generation) and there it all works fine with... Really really really makes me wonder about what went wrong with our setup. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http

[Asterisk-Users] dual TE410, both span 3 is broken

and clockslip-ticks on the line. Juggling the cables between other interfaces, and the then-non-span-3 is clear and the new-span-3 is full of HDLC errors and clockslips. Myself I'm going with a this must be hardware problem, unless somebody else has an idea. Edwin -- Edwin Groothuis

[Asterisk-Users] Vegastream clockslip problems

different ports etc. It all lead to no resolution for it. Is there somebody on the list who has a Vegastream 400 working fine? Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin

[Asterisk-Users] Channel juggling, what is it good for?

I often see this happening: - ChanIsAvail returns Zap/94 - I dial out via it. - And then Moving call from channel 94 to channel 101 Why is it moving to channel 101? Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http

[Asterisk-Users] Re: Inter-Tel AXXESS failure: HDLC Bad FCS (8) on Primary D-channel of span 1

... and CVS-HEAD versions of everything. Try the 1.0.7 zaptel drivers with the 1.0.9 asterisk. Works fine here. Scary but true. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin

[Asterisk-Users] 1.0.9 - can't get link up, 1.0.7 works fine.

it's working again. Who has suggestions for me on what to do? Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ Asterisk-Users mailing

[Asterisk-Users] TDM400P REV I issues - ProSLIC vs TDM400P

make sense to anybody? Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

[Asterisk-Users] Re: TDM400P REV I issues - ProSLIC vs TDM400P

On Mon, Aug 01, 2005 at 11:56:26PM -0500, [EMAIL PROTECTED] wrote: On Tue, Aug 02, 2005 at 02:48:00PM +1000, Edwin Groothuis wrote: The REV I card shows up in the PCI table as: 02:05.0 Network controller: Tiger Jet Network Inc. Intel 537 (or 02:05.0 Class 0280: e159:0001

[Asterisk-Users] Caller ID, Called ID and Forwarded ID

of it? Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com

[Asterisk-Users] Re: can't make my PRI dial out

that there is a minor difference in the SETUP frame: In the Channel ID, Telstra hasn't set the exclusive bit, AAPT has. Solution for me was in zapata.conf: - spanmap = 1,1,1 + spanmap = 1,1 Edwin -- Edwin Groothuis |Personal website: http

[Asterisk-Users] Re: ISDN BRI + echo cancelling + Fax

/bug_view_page.php?bug_id=0003881 [patch] Add E option to the Dial command to explicitely disable echo cancellation Still have to make it available for HEAD. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http

[Asterisk-Users] Asterisk security problem: authorized SIP users can fake any callerid!

this problem - did I misunderstand something, what am I doing wrong? Besides setting it in the sip.conf (callerid=Foo Bar 911), no. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin

[Asterisk-Users] Digium E1/T1 card with mgetty+sendfax

of Quad E1 cards :-) and then just put it back to back against Asterisk server. And instead of letting mgetty+sendfax talk to /dev/ttyI[0-7], we use /dev/zap/[0-30]. Has anybody else ever tried this? Success- and horror stories are welcome! Edwin -- Edwin Groothuis |Personal

Re: [Asterisk-Users] Digium E1/T1 card with mgetty+sendfax

On Mon, Feb 28, 2005 at 04:33:05AM -0600, [EMAIL PROTECTED] wrote: On Mon, 28 Feb 2005, Edwin Groothuis wrote: For the project I've used the Eicon DIVA card. It has 8 BRI ports, and for about 25% of the time there are 7 or 8 in use. So we want to replace it with an E1 card. Only issue

[Asterisk-Users] /var/run/asterisk.ctl configuration

(=asterisk admin) to configure the ownership and permissions of /var/run/asterisk.ctl. If you need this feature, see http://bugs.digium.com/bug_view_page.php?bug_id=0003558 for the patches. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED

[Asterisk-Users] extensions.conf - redundancy removal

= 14140293353510,1,Macro(call-local-sip,edwin,${EXTEN:4}) Is there a way to do this in one line? -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin

Re: [Asterisk-Users] IPv6 and Asterisk?

will quickly find out why DNS was invented. I'm glad to test IPv6 internally, but my IPv6 tunnel (via FreeNet6) is too slow for VoIP. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin

[Asterisk-Users] Error WARNING[-150101888] when starting Asterisk.

?bug_id=0002745 Not that it helps you in this case, just explaining a little bit. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin

Re: [Asterisk-Users] Interrupting MusicOnHold while call in queue ?

configuration files. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com

Config files (was: Re: [Asterisk-Users] Interrupting MusicOnHold while call in queue ?)

On Fri, Nov 19, 2004 at 07:34:19PM +1100, Edwin Groothuis wrote: You can either do a make samples (which will overwrite your existing configuration) or look in the configs subdirectory in the asterisk source tree for queues.conf.sample. It would be nicer if make samples would install

[Asterisk-Users] FreeBSD Asterisk and G729 codec

version of Asterisk. You need to compile it on a linux machine and copy the libraries and binaries. Of course it would be much better if Digium would provide a FreeBSD binary of the library :-) Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED

[Asterisk-Users] TE410P - How many can I have?

need the bus which looks like the one marked with a 2: It has the nodge near the chassis. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin

[Asterisk-Users] SIP (or IAX) modem driver

to a virtual modem which then connects to a SIP server. And when a call comes in on that SIP server, it sets up a connection to the modem program which sends an RING to the terminal program. Does anybody know if such a driver / daemon / asterisk app exists (for Linux preferably). Edwin -- Edwin Groothuis