similar to: problem with dahdi and meetme

i am experiencing problem with asterisk 1.2.18 I've downloaded and installed pwlib and openh323 with the following commands: cd /path/to/pwlib ./configure make clean opt cd /path/to/openh323 ./configure make clean opt then 'ive set the corresponding PATH PWLIBDIR=/data/programmi/asterisk_1.2.18/pwlib_v1_10_0/ export PWLIBDIR OPENH323DIR=/data/programmi/asterisk_1.2.18/openh323_v1_18_0/

I'm experiencing some problem using the transfer() application,expecially when a call in received from a queue. I'm using Asterisk 1.4.22.1 This is my scenario: ; this is the piece of code in extensions.conf that place the call in the queue when 1111 is called exten => 1111,1,Answer exten => 1111,n,Queue(2000|t) ;this is the piece of code that calls the user test when 2222 is

I updated Asterisk to 1.6.2.7 and now the user introduction in the meetme application is no longer working: [May 27 09:26:51] WARNING[2407]: channel.c:4034 ast_request: No channel type registered for 'DAHDI' -- Created MeetMe conference 1023 for conference '800' [May 27 09:26:51] WARNING[2407]: app_meetme.c:3640 find_conf: No DAHDI channel available for conference, user

Hello there: I think I have a silly kernel configuration problem. I'm running: * vanilla 2.6.27.10 kernel built from source * dahdi-2.1.0.4 built from source So far so good, dahdi module loads just fine: dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.1.0.4 when I try to: hal04 dahdi # modprobe dahdi_dummy FATAL: Error inserting dahdi_dummy

Hi to all the busy-level / busy-limit setting in sip.conf is available for Asterisk 1.4.22 ? This is a piece of my sip.conf: [202] type=friend secret=202 host=dynamic ; This device registers with us username=202 ; Username to use when calling this device before registration limitonpeers = yes call-limit = 2 busy-level = 1 The directive busy-level is ignored.... I've also tried

Hi., do you think that sbr policy in queue strategy will be useful? Bye ---------- Forwarded message ---------- From: nik600 <nik600 at gmail.com> Date: Sat, 7 Mar 2009 15:21:14 +0100 Subject: add a new queue strategy: SBR To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com> Hi to all isn't there any plan to add the Skills Based Routing strategy in

sorry for top-posting, the two topics started with 2 different reason subject, but then we finished on the same problem. btw,the 2 show channel are reported above: the channel with DTMF working kcenter*CLI> core show channel SIP/pbx2-000004b9 -- General -- Name: SIP/pbx2-000004b9 Type: SIP UniqueID: 1467323106.1275 Caller ID: xxxx Caller ID Name: xxxx

I have a TDM400. Just updated Fedora 12 to kernel 2.6.32. Rebuilt and installed dahdi-2.2.1. kernel modules loaded. lsmod | grep wctdm wctdm 37233 0 dahdi 194985 1 wctdm lsmod | grep dahdi dahdi 194985 1 wctdm crc_ccitt 1549 2 dahdi,isdnhdlc dmesg: dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.2.1

thanks John yeah, your approach is much siple, i've tried it but i'm not able do detect DTMF tones. it seems that on calls that i receive DTMF tones are handled correctly, but on calls generated from Asterisk to the world when the called side sends some DTMF digits they are not detected: -- Executing [s at macro-myconnector:1] NoOp("SIP/pbx2-000004b2", "") in new

Hi. I have a Debian Leni system with asterisk-1.8. I was trying to get meetme to work and it depends on dahdi, so I compiled dahdi-trunk and dahdi-tools-trunk, however, when trying to insert dahdi_dummy, it complained about symbol crc_ccitt_table, although the module was actually there in the kernel tree. So, I took the Debian source, and I had the config and I did make Bzimage, make modules

Howdy, I am playing around with asterisk within an LXC container on Ubuntu 11.04. I have asterisk (1.4.42) running fine, but want access to dahdi_dummy for timing (meetme). I have dahdi installed on the "host", and dahdi_dummy is loaded: root at astnorth:/# ls -ltr /dev/dahdi total 0 crw-rw---- 1 root root 196, 250 2011-06-08 13:59 transcode crw-rw---- 1 root root 196, 253

Dear all i'm using an "old" Asterisk 1.6.2.9-2+squeeze12, and want to know if is possible to configure a scenario like this: 1) receive a call and put it on-hold in a queue (OK) 2) monitor the queue and trigger an outbound call to a remote number using AMI, setting the channel of the on-hold on a specific var named channel2Link (OK) 3) when the remote number answer, trigger an

Hi to all i'm using PlayDTMF with AJAM, after the authentication, i make a request like this: host:8088/asterisk/mxml?action=PlayDTMF&Channel=SIP/200-sdadsadkioah&Digit=1 the result is: <ajax-response> <response type='object' id='unknown'><generic response='Success' message='DTMF successfully queued' /></response>

hi is it possible to set up in the dialplan (on in sip.conf, or something else) the hostname of the outgoing uri call? This is my scenario: - CCM integrated with Asterisk via h323 - SIP user registerd to Asterisk - Asterisk is behind NAT - Asterisk ip is 10.10.10.2 - SIP user view Asterisk as 10.10.15.2 (Asterisk is behind NAT) When the CCM calls the SIP user the call works perfectly. The

Hi to all i'm using Asterisk 1.4 and need to announce something like 'The operator answering to you call is XXX' to the caller, is it possible to do that using an AGI script ? The syntax in Asterisk 1.4 is Queue(queuename[|options][|URL][|announceoverride][|timeout][|AGI]) So, setting up an appropriate AGI script can i play an audio file (or create it with some tts) to the

Dear all i'm planning a migration to asterisk for a high volume IVR service (from 1000 to 1500 concurrent call) The IVR service is based only on DTMF tones so the features required is - play feature - dtmf detection Asterisk will receive calls via VOIP (SIP with g711 codec) The IVR service wil be a static service based on Asterisk dialplan with some prompt (from 0 to 5, play of files in

I am installing dahdi on a machine lspci 00:00.0 Host bridge: Advanced Micro Devices [AMD] RS780 Host Bridge 00:01.0 PCI bridge: Hewlett-Packard Company Unknown device 9602 00:04.0 PCI bridge: Advanced Micro Devices [AMD] RS780 PCI to PCI bridge (PCIE port 0) 00:05.0 PCI bridge: Advanced Micro Devices [AMD] RS780 PCI to PCI bridge (PCIE port 1) 00:06.0 PCI bridge: Advanced Micro Devices [AMD]

Hi to all. What can i do if a customer needs to log in the CDR all the dialpan actions related to a call? I mean, not only the lastapp e the lastdata but all the dialpan actions! I know that the actual CDR system store one record for each call (and for billing purposes this can be correct) but in some cases the approach needed is something similar to the queue_log. I know that exists ResetCDR

Hi, I have some servers that are dedicated to do meetme conferencing. From some previous test i concluded that I need to use dahdi_dummy as it is more accurate. If I did use the core timers in dahdi (not loading dahdi_dummy) I got bad quality in the conferences and dahdi_test showed 99.6% as worst. I thought maybe the issue as bad hardware for the timing or something else. But today I

Dear all i'm planning a transparent migration from a courier server that provides both IMAP and POP3 access to users to a remote dovecot server with both IMAP and POP3 access. I have to migrate about 2500 users for 250 GB of space. I'm using dovecot 2.2.5.4 on debian6 squeeze. To make a transparent migration i have to maintain old IMAP UIDs and POP3 UIDs, so i've read