Re: [asterisk-users] Conference problem

@lists.digium.com Date: Wednesday, April 22, 2009, 8:08 PM run a sip debug and check whether it's asterisk disconnecting the calls (usually a SIP BYE message) or whether Asterisk is getting the disconnect from your Cisco GW Martin On Wed, Apr 22, 2009 at 10:56 AM, Cristi Iconaru cristi_icon...@yahoo.com

[asterisk-users] Conference problem

Hello all,   I have some issues with the MeetMe application.   The working topology is as follows. The Asterisk (1.4.22-rc5) is connected through SIP trunk to a Call Manager (6.1.2) which is connected to a Cisco Voice Gateway. The Gateway is connected to PSTN through a PRI. The calls are

Re: [Asterisk-Users] h323 over NAT

Jeremy McNamara wrote: H.323 doesn't deal with NAT. Jeremy McNamara ayaz wrote: hello I am trying to configure h323 over NAT Can some one help me out Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED]

Re: [Asterisk-Users] Microsoft SQL

[EMAIL PROTECTED] wrote: I need to make a little IVR app and get/send the data into a MS-SQL database. As far as I know, it doesn't have driver for Linux. Anybody here already found here any workaround for this situation? Maybe, I can use an AGI interface to do that, maybe perl+ODBC? Isamar

Re: [Asterisk-Users] Channel Language

Mark Spencer wrote: Use setlanguage. Then organize the language files by directory e.g. /var/lib/asterisk/sounds/de /var/lib/asterisk/sounds/digits/de Also, say.c will have to be modified to support German style number handling. Mark On Sun, 27 Jul 2003, Peer Oliver schmidt wrote: Hi, I

[Asterisk-Users] error WARNING[28697]: File app_dial.c, Line 304 (wait_for_answer):Unable to forward voice

I am trying to put a call on a E1 ISDN : The configuration are simple: zapata.conf : [channels] context=inbound switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes ;echocancel=no

[Asterisk-Users] Alphanumerical digits

Sorry Martin to bother you again! I have an ISDN flux with 100 numbers. The local PSTN is sending now the DNIS/DID (so they said!!!) (I have set the immediate=no in zapata.conf) but I have the same problem as before : NOTE : the number is alphanumeric-DID alphanumeric (I will make tests with

[Asterisk-Users] Alphanumerical digits

I see the following line into debug (pri debug span 1): 1. Progress Description: Calling equipment is non-ISDN. (3) ] 2. Calling Number (len=14) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Into a ISDN that is working I got this : -- Making new call for cr 23274 --

Re: [Asterisk-Users] Drop the call in 10min

[EMAIL PROTECTED] wrote: There is a way to drop every running call every 10 min of conversation? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Check the whentohangup value from cdr

Re: [Asterisk-Users] Drop the call in 10min

[EMAIL PROTECTED] wrote: There is a way to drop every running call every 10 min of conversation? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ast_channel_setwhentohangup(chan,600);

Re: [Asterisk-Users] ISDN PRI E1 configuration with E100P

[EMAIL PROTECTED] wrote: hi Everyone, We are configuring an ISDN PRI E1 with an E100P card, when you load the drivers, and starts the asterisk, cards also starts fine, givin following output, *CLI == D-Channel on span 1 up -- B-channel 1 successfully restarted on span 1 -- B-channel

[Asterisk-Users] Error on PRI channel : Call specified but not found!

Anyone encountered this error on an PRI channel local PTSN: WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup on bad channel 18 WARNING[9226]: File chan_zap.c, Line 5275 (pri_fixup): Call specified, but not found? WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup

[Asterisk-Users] Error on PRI channel : Call specified but not found!

Anyone encountered this error on an PRI channel local PTSN: WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup on bad channel 18 WARNING[9226]: File chan_zap.c, Line 5275 (pri_fixup): Call specified, but not found? WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup

[Asterisk-Users] Answering on an zap device

How can I accept calls on a Wildcard E400P . Please include the zaptel.conf , zapata.conf and extension.conf to fully understand everything. Take a look firts to my configuration file. Where I did wrong? zaptel.conf: span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3 span=3,0,0,ccs,hdb3

Re: [Asterisk-Users] Answering on an zap device

:20, Cristi wrote: How can I accept calls on a Wildcard E400P . Please include the zaptel.conf , zapata.conf and extension.conf to fully understand everything. Take a look firts to my configuration file. Where I did wrong? zaptel.conf: span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3 span=3,0,0

Re: [Asterisk-Users] Answering on an zap device

Steven Critchfield wrote: Please trim unnecessary lines from the post. On Tue, 2003-07-08 at 09:23, Cristi wrote: What is the need for the callerid in the following configuration: (default config page) ;callerid=Green Phone(256) 428-6121 ;channel = 1 ;callerid=Black Phone(256) 428-6122

[Asterisk-Users] Debug PRI!

This indicate that the connection with the local provider PTSN it is ok? : -- Attempting call on Zap/10 for [EMAIL PROTECTED]:1 (Retry 2) -- Making new call for cr 32781 Protocol Discriminator: Q.931 (8) len=28 Call Ref: len= 2 (reference 13/0xD) (Originator) Message type: SETUP (5)