similar to: record call transfered from IAX softphone to SIP hardphone

Hello List, i need to be able to record the call transferred from iax extension to sip extension when i call the sip extension from the IAX extension i can record the call without any issue but when i receive a call from customer in IAX and i transfer this call to SIP client the conversation between customer and IAX client is recorded but the conversation between customer and sip extension is

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

Hi group I'm newbie on Asterisk so I followed the Linux Networking CookBook by Carla Schroder to make my first call. My asterisk box is on a Debian box with an public static IP. The clients (2) are with dynamic private IP's I'm using SJphone on a PC and a Linksys PAP2-NA to make calls between them. Both of them register well on my Asterisk server but when I call from the SJPhone to

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 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

Hi, I just installed OH323 Plugin and im now tryin to make simple Configuration to connect Openphone and Xlite to my Asterisk-Server. All works fine, i just wanna know if there's a better way to do it? Is there anything wrong with my Config? OH323.conf [general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=8000 udpEnd=8005 fastStart=no

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! Problem: I can't hear what the people at Location B i saying, they hear me but I do not hear them. They can call, I can call. Just no sound. My current setup is: Softphones/Hardphones(Location A) <-> Asterisk <-> Firewall/Nat <-> Internet <-> Firewall/Nat <-> Softphone/hardphone(Location B) I am having problems with sound, I have opened the

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

Can anyone shed any light on an issue with agent penalties? I have 2 queues set up with agents working both queues, but where agent 1 should have a penalty for queue 2 and agent 2 should have a penalty for queue 1. When a call is sent to either queue, it rings agents with and without penalties at the same time. I set up a second system and cannot replicate the issue on the test system. I

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

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

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

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 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 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 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

hello list, i have installed 2 diguim cards in my server using asterisk 1.4 (i use the old version with zapata.conf and zaptel.conf) i want to use the span 1 for group 1 and span 2-span 6 for the group 2 (i want to active the round-robin for span 2 and 6) in order to activate the WIMAX and FH please see the configuration below and tell me if there is anything wrong question 2: what is

tnaks for your response but the number dialed exist and i can call this number when i configure the trunk directly in x-lite and i call call also this number from my cell phone . any help thanks and regards 2015-03-25 12:59 GMT+00:00 Matthew Jordan <mjordan at digium.com>: > On Wed, Mar 25, 2015 at 7:35 AM, Salaheddine Elharit > <salah.elharit200 at gmail.com> wrote: > >