similar to: set musiconhold only for caller

hello list i need your help please regarding an issue with snom300 and aastra6731i using asterisk 11.13.0 asterisk snom 300 8.7.3.25 astra 6731i 2.6.0.2019 i have configured the trunks like below 100 in snom300 200 in snom300 300 in aastra6731i 400 in x-lite the calls between x-lite and aastra ====ok inbound and outbound the calls between x-lite and snom300====> ok inbound and

Hello, We are working on a new Asterisk installation and have run into some problems related to playing MusicOnHold for a caller when they have been placed on hold by an agent, that took the call from a queue. A. When pressing the HOLD button on SNOM 190 and Grandstream BudgeTone SIP phones, MusicOnHold works fine when making inbound or outbound direct calls by extension. Music starts to play

Hello, I have a problem in Asterisk 1.4 each day I need to restart *asterisk service asterisk* restart in order to unblock the calls My question how can I do in order to check the issue, and if there is any tool or log? Thanks and regards. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list i have configured astersik 1.4 with sip i have a question when i put in dial plan.conf exten => _0678922645.,1,Set(CALLERID(number)=520460587) exten => _0678922645 .,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0)) exten => _0678922645 .,n,Dial(Zap/g1/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)) exten => _067892264*5*,2,Hangup() i can not call my

Hello list, i have one question related to call forwarding. i have 2 number for the inbound and i want to configure asterisk like that. When the customer call the first number 0522XXXXXX the call will be forwarding automatically to anther number 0520xxxxxx Does anybody have a solution to this problem. Thanks and Regards. -------------- next part -------------- An HTML attachment was

Hi, Le 12/03/2015 17:28, Salaheddine Elharit a ?crit : > hello list, > > i use the code below > > [macro-chanspy] > exten => s,1,Authenticate(${ARG1}) > exten => s,n,ChanSpy(SIP/${EXTEN:3},__dqs) Here you have a problem: ${EXTEN} value is s [...] Daniel

Hi all! My problem is that calls being placed in the queue, and are waiting while the agents are busy, when an agents is then free they gets connected to the agent but there is silence (no voice). If a caller has not to wait in the queue, there is no problem. My agents have an iax2 client, and imcoming calls are over SIP. queue.conf: persistentmembers=yes autofill=yes ringinuse=no

i noticed that when i active the voicemail in the IP-phone where the number 0033149xxxxxx is configured i can call this number without issue Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/FD/0033149xxxxxx == Begin MixMonitor Recording SIP/101-0000010d -- SIP/FD-0000010e is making progress passing it to SIP/101-0000010d > 0x2b393cfc2610 -- Probation passed

thank you so much it work you must add 1 like below [app-chanspy] exten => _0071XX,*1,*Macro(chanspy,1234) exten => _0072XX,*1,*Macro(chanspy,5678) exten => _0073XX,*1,*Macro(chanspy,8910) best regards. 2015-03-11 19:48 GMT+00:00 Carlos Chavez <cursor at telecomabmex.com>: > On 3/11/15 12:48 PM, Salaheddine Elharit wrote: > >> hello list, >> >> i use

You would need to give more information really. Your sip.conf file listing the entries for the phones especially which codecs are permitted. A copy of the 'asterisk -rvvv' console output when you make the call. On 27/03/15 17:05, Salaheddine Elharit wrote: > please no body has som with aastra can help me in this issue > > 2015-03-26 11:02 GMT+00:00 Salaheddine Elharit >

hello list, i use the code below [macro-chanspy] exten => s,1,Authenticate(${ARG1}) exten => s,n,ChanSpy(SIP/${EXTEN:3},dqs) exten => s,n,Hangup app-chanspy] exten => _0071XX,*1,*Macro(chanspy,1234) exten => _0072XX,*1,*Macro(chanspy,5678) exten => _0073XX,*1,*Macro(chanspy,8910) but when i do 007100 for exemple i spy another agnet 102 or 103 any help please thanks and

hello list, i have created a callfiles with my asterisk 1.4.43 like: Channel: SIP/watara/06xxxxxxxx MaxRetries: 10 RetryTime: 5 WaitTime: 20 Context: mycontext Extension: s Priority: 1 extensions.conf mycontext exten => s,1,Ringing() exten => s,n,Playback(hello-world) exten => s,n,Dial(SIP/105) exten => s,n,Hangup() it works with one number how can i do in order to create a

hello list i have an IVR menu in asterisk 1.4 like below exten => 600,1,Ringing() exten => 600,n,Wait(2) exten => 600,n,Goto(home,s,1) [home] exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/) exten => s,n,Background(${sounds_path}music1) exten => s,n,Background(${sounds_path}music2) exten => s,n,Background(${sounds_path}music3) exten =>

Hello list i have an issue with my dahdi_channels.conf in span 1 when i use it like below i can do my outband calls without issue ; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) group=0,11 context=from-pstn switchtype = euroisdn signalling = pri_cpe channel => 17-31 context = default group = 63 but when i add in channel 1-15 like: channel => 1-15,17-31 i receive all

hello list, i need your help about cdr ,i have installed the module cdr in my asterisk 1.4 . for the inbound calls when i call my sip exten like below : exten => 506,1,Dial(SIP/223, 10) exten => 506,n,Dial(SIP/276, 10) in CDR i have just one line with SIP /276 the last line but there is no historic for the first SIP 223 recid Record ID | calldate |clid |src

I am having a problem with the Call Queue Feature. If I let a user into the Queue prior to an agent being available for them to take the call, they experience the following: 1) they hear that they are the first in line 2) when the agent finally logs in (the caller on hold in the queue is sent to the users phone) 3) the AGENT is still in the login phase hearing that they are "successfully

I was playing with the agent application to see if I could get it to work. Everything works fine, except that Asterisk plays musiconhold while an agent is logged in and is not taking a call. Is there a way to disable the music in this situation? Imagine working tech support where you had to listen to hold music when you weren't taking a call. Now think of your company's choice of hold

Hello! I've upgraded from 1.2.9 to 1.2.14 recently but experience an unexpected behaviour with musiconhold: While in 1.2.9 musiconhold was playing continuous on sequential extensions after a timeout, it is restarted for every extension in 1.2.14: ;music starts exten => 902,1,Dial(SIP/phone1@proxy.com|5|m(mymusic)) ;music starts again exten =>

Hi, I have the same problem with the queue configuration When I receive 2 calls only 1 phone ring even if more agent's phone are free. The second call will go to an other agent only if the first call is pickup. Somebody have a solution ? This is my config file : Queue.conf [general] ; ; Global settings for call queues ; ; Persistent Members ; Store each dynamic agent in each queue

Tzafrir Cohen wrote: > > The agent picks up the phone but neither the agent nor the caller > > > here anything. >So please provide a more complte trace and a the relevant partt of your >dialplan. > Here is the relevant part of the dialplan: [local] exten => 98,1,Dial(SIP/sguenther,20,tr) exten => 98,2,VoiceMail(98|u) exten => 98,3,hangup exten =>