similar to: how to join 2 channels using AGI/AMI

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

this is the point, and the strange thing: DTMF is set to rfc2833, but is working both on incoming and outgoing calls, it is not working only on calls generated with the Originate AMI command, or with the queue member that point to Local dialplan, as you suggested 2016-06-30 22:53 GMT+02:00 John Kiniston <johnkiniston at gmail.com>: > Looking at your logs it looks like you may need to

Dear all i'm creating an outgoing call to number xxx with this command: http://host:port/mxml?action=Originate&Channel=Local/xxx at to-external &Exten=testDTMF&Context=cRETEUNICA&Priority=1 wich points correctly to this portion of dialplan: [cRETEUNICA] exten => testDTMF,1,Answer exten => testDTMF,n,Read(digito,,1) exten => testDTMF,n,SayDigits(${digito}) The

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

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

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

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

Dear all is it possible to handle a queue using a programmed IVR? As i understood, is possible to: - do some IVR in the dialplan BEFORE to queue the call - put a timeout to exit from the call and then do some IVR in the dialplan - intercept a single dialtone to exit the queue and performe some IVR in the dialplan (context setting in the queue) I've tested these things but in each case if i

Hi to all. I'm trying to use meetme on asterisk 1.4.22.1. On a debian i've compiled (as i need h323 support) openh323_v1_18_0 pwlib_v1_10_0 dahdi-linux-2.1.0.3 dahdi-tools-2.1.0.2 asterisk-1.4.22.1 All works fine, dahdi status is: asterik:/data/programmi# /etc/init.d/dahdi status ### Span 1: DAHDI_DUMMY/1 "DAHDI_DUMMY/1 (source: RTC) 1" (MASTER) asterik:/data/programmi#

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

Dear all i'm trying to originate an outgoing call with the command originate, from Asterisk's CLI i'm typing: CLI> originate IAX2/my-iax-provider/number2call application wait 10 [Apr 18 19:31:12] DEBUG[32331]: chan_iax2.c:4000 create_addr: prepending 40 to prefs -- Call accepted by 62.149.202.150 (format ilbc) -- Format for call is ilbc -- Hungup

Hi to all i'm planning the migration of a company on Asterisk, i have planned this scenario: 2 server with * 4 GB RAM * 2 CPU 64 bit dual core * RAID 1 * 2 network interfaces 1000 Mbit/s Each server will have a virtual interface that will be switched from one to the other in case of hardware problem. The question is: can one server with those settings manage up to 200 simultaneous call?

Hi to all. Is possible with the Asterisk 1.4 cli view the current calls and their codec? Thanks to all -- /*************/ nik600 http://www.kumbe.it

Dear all i have a TE205P connected to an Asterisk 1.2.18. Yes i know, the version is old but since now the system was stable and i don't have the necessity of an upgrade. The system provide an IVR service that: 1) receive the call 2) verify the queue length 3) hangup if queue length is > 1 4) put the call in the queue othervise Then, there is an AGI php script that 1) verify the queue

Dear all is it possible to rsync in a master-slave scenario saving to slave gzip content? i'm not talking about compression during transfer, i'm talking about -saving- the destination in a compressed format. Example: FROM: -folder_A --file_A --file_B TO: -folder_A --file_A.gz --file_B.gz I know that this won't be a "real" rsync between two folder, but it will be an

Hi to all in a scenario where: - the bandwith is shared with other traffic (HTTP,VPN,ecc) - the PBX is on a remote VPN peer - due to many reasons Qos is not usable There is a IAX trunk between 2 Asterisk 1.4 i've tried different codecs (ulaw,alaw,gsm) but the main problem still remain the same: too many voice packet get lost. The main problem is surely on the network, but the strange thing