similar to: how to avoid call waiting? Or check DIALSTATUS before Dial()?

I'm trying to connect a sip call from sipgate to Asterisk A to Asterisk B. Both are behind NAT, but port forwarded. I get the connection, but no voice - either in or out. I can call on SIP from A to B (and from B to A). Do it all the time. Asterisk A receives SIP calls from Junction and Teliax. CLI on A looks right: == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 ==

Upgraded from 16.2.14 to 1.6.2.15 on Fedora 13, with aastra 9133i and 57i. On 9133i and 57i: #<extension># works for a blind transfer. Xfer<extension>Xfer doesn't! All this worked on 1.6.2.14. Nothing useful on cli, verbose 3, DEBUG. Here extension 169 answers an outside call, and tries to transfer it to 145 using the Xfer button: -- SIP/169-0000009c answered

Hi, I'm having problem with call parking. When I park call, either via transfer to xten or park digit sequence from features.conf, I hear the parking lot number read to me and the user gets transferred. However, MOH stops for the caller the moment user is transferred. The user can be retrieved by dialing the parked extension and voice resumes. If the parked user hangs up, the channel state

On 1.6.0.18-rc3 using app_fax.so, spandsp-0.0.5, anytime I use SendFAX asterisk restarts: [Nov 15 19:00:36] VERBOSE[17013] logger.c: -- Executing [s at fax-tx-test:1] ESC[1;36;40mNoOpESC[0;37;40m("ESC[1;35;40mSIP/nhi-rive rside-sip-00000000ESC[0;37;40m", "ESC[1;35;40mContext fax-tx-testESC[0;37;40m") in new stack [Nov 15 19:00:36] VERBOSE[17013] logger.c: --

hello list i have an issue related to outbound calls i can contact all the number except on number given by our provider in trunk the issue just when i configure my trunk in our server but when i configure the trunk directly in x-lite i can contact this number without issue below the cli == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [0149xxxxxx at

Hi List, we're calling a sip phone from our Asterisk Server, and try to add logic depending on the dialstatus Stripped down example; exten = 494XXXXXXXXX,n,Dial(SIP/4120089,15,w) exten = 494XXXXXXXXX,n,Goto(98-${DIALSTATUS},1) exten = 494XXXXXXXXX,n,Hangup() ..... exten = 98-BUSY,1,NoOp(Busy) exten = 98-BUSY,n,ExecIf($["${Voicemail}" =

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 >

Hi All, I'm confused! My Asterisk system has a Zap trunk and three SIP trunks. I'd like to configure the dialplan to route via the first trunk in a list and if that's not available or it's busy, fall over to the second, then to the third, etc. AIUI Dial(Zap/1&SIP/out1&SIP/out2/${EXTEN}) rings all the trunks in the list and bridges to the first to answer. Unfortunately,

Hi list, I need your help ,I have an incoming call x the ivr and the operator takes the call. ext "101" , If a second call reenters and the operator is talking, I want to send to the extension 102 I use the Variable DIALSTATUS , but not working check IVR [IVRINMA] exten => s,1,Wait(1) exten => s,n,Set(CHANNEL(language)=es) same=> n,Set(TIMEOUT(digit)=4) same=>

Hi All, I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO card and installed in my asterisk server. My freepbx detected the x100p FXO card and i can see the card specific details in command line. I have configured the following things. 1. OUTBOUND caller id and Dialing rules in Freepbx. 2. INBOUND route When i call to the PSTN number before

Hi this message give me when I calling a number than actually not busy: "Dial failed due to trunk reporting BUSY - giving up" max channel is unlimited and sometimes it dial number ok but most of the time it gives me this error. Please inform me how can solve this problem. thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I'm trying to resolve a weird issue with SIP routing. I have a number of SIP trunks, from a selection of providers, all of which are registered in sip.conf: [general] context=default allowguest=no allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=yes tcpbindaddr=0.0.0.0 transport=udp bindport=15060 srvlookup=yes allowsubscribe=yes

Hello, I am trying to alter the outbound callerID for extensions within a context I have created. I wrote the following: exten => _9.,2,ExecIf($[$["${REALCALLERIDNUM}" = "360"] | $["$ {REALCALLERIDNUM}" = "670"]]|Set|CALLERID(num)=581560) exten => _9.,3,ExecIf($[$["${REALCALLERIDNUM}" = "361"] | $["$

I've been working on writing a subroutine to page groups of phones at once and I'm having some difficulty. My goal is to have a user call an extension, I record the page they wish to play, I then page out that recorded file to the phones in groups. [sub-masspage] exten => s,1,NoOP same => n,Answer same => n,Set(filename=$PAGE) same => n,Wait(1) same =>

I'm using BLACKLIST() to check numbers, which does not like leading + signs. I want to test if there is a plus sign, and then remove it. I tried: ; strip leading plus sign same => n, Verbose( callerid 0:1 is ${CALLERID(num):0:1} ) same => n,ExecIf($["${CALLERID(num):0:1}" = "+"]?Set(CALLERID(num) = ${CALLERID(num):1})

Is there a way skip / ignore the member whose status is busy in the Queue. I have two channel member in queue and i have set the peer limit 2 for these members. I want to skip those member who are currently on the call (answered to calls) and now their status is busy, if Queue see the busy status caller will not enter in the Queue and go to the next priority. [test-queue] strategy = rrmemory

I am trying to record calls on demand both inbound and outbound calls. I can record outbound calls just fine but not inbound calls or calls from an internally between extensions. I am using the latest asterisk 1.8.x certified version. On an outbound call I see: == Using SIP RTP CoS mark 5 -- Called SIP/ BVTrunk /7190000000 -- SIP/BVTrunk-00000163 is making progress passing it to

Hi everyone! I'm quite a newbie at this Asterisk stuff so please bear with me. We've recently decided to start training in Asterisk via AsteriskNow! Asterisk version is 1.4.18.1 through AsteriskNow! 1.02 The box we have is paired with a Digium TE110P and we've managed to get it to the point where incoming calls via a single DID (from NTT Japan) can be received and answered

On Tue, 30 Aug 2016 17:56:35 +0200 Administrator TOOTAI <admin at tootai.net> wrote: > Something like > > exten => 5555551111,1,Verbose(Door buzzer calling) > same => n,Set(toRing=) > same => n,ExecIf($["${DEVICE_STATE(SIP/user1)}" = "NOT IN > USE"]?Set(toRing=${toRing}&SIP/user1) Failed. I checked the online docs and the syntax seems to

Imagine that you have this code: exten => _X!,n,Set(foo=${QUEUE_WAITING_COUNT(${QueueName})})) If ${QueueName} happens to be unset, this will cause a warning: [Jul 22 14:26:17] ERROR[8114]: app_queue.c:5187 queue_function_queuewaitingcount: QUEUE_WAITING_COUNT requires an argument: queuename The obvious solution: exten => _X!,n,ExecIf($["${QueueName}" !=