[asterisk-users] Upgrading to 11.4.0 and ast_channel_make_compatible_helper: No path to translate

/fixedline-0004'. == MixMonitor close filestream (mixed) And when I try to try to initiate a call with a manager script, I receive an authentication error from the script. How might I find more info to help diagnose either or both of these issues? -- Eric Smith

Re: [asterisk-users] Upgrading from 1.4 to 11.4.0 - solved (trivial)

Issues below solved by configuring iaxmodem in extensions.conf and in correcting the context of the initiating telnet script. ... shiny new asterisk -- Eric Smith Eric Smith wrote on Sun-23-Jun 13 9:24PM Hi After upgrading from 1.4 to 11.4.0, I am able to receive calls and direct them

[asterisk-users] Upgrading from 1.4 to 11.4.0

an authentication error from the script. How might I find more info to help diagnose either or both of these issues? -- Eric Smith -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

[asterisk-users] dahdi 2.6.1+2.6.1 compile fails

..7e534b7 100644 |--- a/xpp/oct612x/include/octdef.h |+++ b/xpp/oct612x/include/octdef.h -- File to patch: -- Eric Smith -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] dahdi 2.6.1+2.6.1 compile fails

+++-===-==-==- ii linux-headers-3.5.0-17-gene 3.5.0-17.28i386 Linux kernel headers for version 3.5.0 on 32 bit x86 SMP [eric@pepper ~] $ uname -r Linux pepper 3.5.0-17-generic -- Eric Smith Adolphe Cher-Aime wrote on Sat-03-Nov 12 10:26PM Hi Eric, Make sure that you have the proper kernel

[asterisk-users] deny=0.0.0.0.0/0.0.0.0.0 does not seem to block external access

' rejected because extension not found. I am running an ancient Asterisk 1.4.26.2 (yes I know what I should be doing). Why might the deny not be working? -- Eric Smith -- _ -- Bandwidth and Colocation Provided by http://www.api

[asterisk-users] dropped calls on android voip connection

nexusone/nexusone (Unspecified)D 0Unmonitored How could I prevent these calls from dropping? --- Eric Smith -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] Notify me when the call is answered

. I want the call to stay connected to the original extension. How would I achieve a notification this way or another way? And is it possible to Dial() and only connect to your extension when someone answers the call? -- - Eric Smith

Re: [asterisk-users] Notify me when the call is answered

] WARNING[14288]: file.c:1273 waitstream_core: Unexpected control subclass '-1' [Mar 17 16:46:06] == Begin MixMonitor Recording IAX2/4506-6141 [Mar 17 16:46:06] -- SIP/nohacker-09747d30 is ringing Any ideas? -- - Eric Smith Jeremy Kister said: On 3/17/2011 8:52 AM, Eric Smith wrote: How would

[asterisk-users] Thailand DID

Hi Could someone point me to a provider of DID's in Thailand. Thanks for any response. -- - Eric Smith -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

[asterisk-users] colored CLI with reattach

Using Asterisk 1.4.26.2 I can get a nice colored CLI if I run asterisk -c But I cannot achieve this when I reattach to an existing instance (as i want to do) with asterisk -r. Is there a way to reattach and have color? Thanks -- - Eric Smith

[asterisk-users] settings for soft phones

is very good and much better than I achieve with a sip client pointing to my own server. Are there some things that I could be missing on the asterisk configuration that could improve the call quality for sip clients on mobile device. -- - Eric Smith

[asterisk-users] DID in Cape Town South Africa

Are there any service providers offering Cape Town DID's? -- - Eric Smith ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman

[asterisk-users] retrieve last number dialled

What is the easiest (simplest) way to do this? -- - Eric Smith ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo

[asterisk-users] gui for conferencing

Is there something simple like gastman that provides functionality to establishing conferencing? -- Eric Smith ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit

[asterisk-users] asterisk log analyzer

Please point me to a simple analyzer tool that parses asterisk log files. I do not want a web based or java application, Just a script that parses the logs and extracts highest priority information, ideally something I can put in a pipe (not to smoke). Thanks -- Eric Smith Fruitcom Amsterdam

[Asterisk-Users] Only single channel recorded with Monitor

-7214 Aug 15 18:31:45 DEBUG[9995]: Stopping retransmission on '[EMAIL PROTECTED]' of Response 2: Found Any ideas how to fix this? Thanks -- Eric Smith ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman

[Asterisk-Users] run a script on completion of call

How do I run an external script on completing a call? Like if I want to send email to the caller. Thansk Eric ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or

[Asterisk-Users] gsm call-hunting [OT]

Hi Has anyone heard of solutions for implementing call-hunting over a bank of gsm lines / sim cards? We wan to have a single gsm dial in number for access to asterisk. Only solution I know about is HP's opencall which is very expensive. Apologies if this is somewhat off-topic. Thanks Eric

[Asterisk-Users] ... - FoIP HOWTO

is imperfect but would be very interested in a HOWTO or FAQ on implimenting this with Asterisk. Thanks for any pointers. Eric Smith According to Alexander Lopez on Tue, May 03, 2005 at 03:02:08PM -0400: } Funny thing is that Faxes over IP (SIP ATA186) and Fax Over Public } Internet (FOPI