search for: outgoings

hello every body, i have big problem to configure h323 trunk between cisco router and asterisk 11.13.1 which uses ooh323 module. any body knows if ooh323 module can work with cisco routers or not???? (in gateway mode, it is ok and register in cisco gatekeeper but i can not configure trunk h323) any comments or hints are really appreciated. SAM -------------- next part -------------- An HTML

hello Dmitry thank you for your reply. Ok, you are right. i want to configure trunk h323 between asterisk 11.13.1 and 2800 cisco router. this is my scenario: PBX(100)--->cisco--->asterisk11.13.1---->PBX(200) when i call from 100 to 200, everything is ok but when i call from 200 to 100, phone rings but after i answer it, i have no voice and call terminates after 5 seconds. this is

Hi Gurus, We have configured asterisk to trunk with avaya with ooh323 channel driver. The sip phone registered on asterisk can dial the extensions registered on avaya via this trunk , and vice versa works too. Even we can make the avaya branch to dial asterisk?s extension and then this extension dial back to another avaya?s extension. But if we dial the external DID number via this trunk from

Hi, i have an asterisk box running with E100P (E1) line as PSTN gw. Sometimes zap channels hang and i couldn't make any PSTN calls but SIP calls are still fine. When this happens I also couldn't restart/reload asterisk from the CLI. I have to kill the asterisk process and run safe_asterisk again. any ideas? asterisk*CLI> show channels Channel (Context

HI I am trying to use the outcall going by the wiki.( http://www.voip-info.org/wiki-Asterisk+auto-dial+out) But I keep getting the errors below. Here is a sample of a callout file. What am I doing wrong? ////Begin Outgoing.call//// Channel: sip/2075 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: managers Extension: 2184 Priority: 1 ////End outgoing.call//// Nov 9 20:32:02

Hello, I am new on asterisk and do some tests on freepbx. I have 2 SIP provider: Provider1: In-/Out- Flatrate, only 1 Number Provider2: Incoming Flatrate, Outgoing Cost depend on destination, 3 numbers On Asterisk site i have 3 phones (branch ??, don't know how its called in asterisk) Is it possible to do something like: Phone 1: Incoming Call: Number1/Provider1 Outgoing Call:

I am hoping to get some help here with building custom PHP to manage a 'wake up call' system. I have the script where the user can set the schedule for an extension wake up call. It appears to write to the /var/spool/asterisk/outgoing/ directory. My two issues: 1 - when the files do get moved over to outgoing/ directory via a cron job, the permissions show "-rw-r--r-- 1 apache

    I am having a strange problem with a new provider.  We already have a couple SIP trunks working fine.  We are trying a new provider but we are having one way audio problems with outgoing calls.  Incoming calls do have two way audio, only outgoing calls have this problem.  I do not see anything odd with a packet capture and using PJSIP history to check.  The provider says that on outgoing

Hello Everybody, I have a genuine problem in Asterisk setup. I have three inbound trunks in my asterisk box, everything is working fine but the only problem is when any user make an out- going call through his/her extension it goes with same number labeled on this. Can we set each of these lines to have fixed outgoing numbers like if extn: 201 make an outgoing call the recipient should

Hi all, I was looking at a way to add the caller id to the outgoing calls (which are made using .call files) using asterisk. Any ideas how to do this ? Currently I get 'Unknown' number displayed on my phone when asterisk makes an outgoing call. Also using something like this is not working as it still displays unknown number. I want set the callerid on the 1.call which is made. exten

I have created a call file which has been moved into the outgoing directory. However the log file displays the following message: Unable to open /var/spool/asterisk/outgoing/1.call: Permission denied, deleting I have executed chmod 777 1.call on the file prior to moving it to the outgoing directory but is there something else I need to do before the file can be used by Asterisk? Any help

??? Usually phone companies set the outgoing CallerID for you but recently we got control over that and are now setting the outgoing Calleir ID ourselves.? My problem now is that the CDR will put the outgoing CID in the CDR instead of the extension that dialed and that causes problems for reports.? What is the proper way to set outgoing CID and keeping the original extension number in the

Everyone, I have been trying to connect a PBX with H323 IP trunks with g711 codec to my Asterisk server running ooh323 service. I can place calls to and from either the Asterisk, or PBX with no problem, but when I try to pickup the call on either end, the phone hangs up immediately. Debug shows normal to me but at the last few lines of data there is an error shown that I have not been able to

Hi Stef, Reffering to page < http://www.docum.org/stef.coene/qos/faq/cache/7.html > I found a para as follows, " CBQ is not always as accurate as it should be. See docum.org on the test page for some tests with bounded classes. The algorithm used by cbq is very "link sensitive". It calculates the link idle time and for that it needs the real link bandwidth, average

Hello list. We just upgraded to 1.6.1.11. We are using real time information stored on mysql databases. That is all running fine. Now, since we upgraded, some member don't get calls from queues. In CLI: "queue show" shows something like: 611 (Local/611 at agents) with penalty 20 (realtime) (*In use*) has taken no calls yet We use the extension 611 in different computers, in the

I have a linux server with 3 public IPs, and I use SSH tunnelling to connect to each of them. Let's call them: 1.1.1.1 (venet0:0), 1.1.1.2 (venet0:1), 1.1.1.3 (venet0:2). When I tunnel using 1.1.1.1, outgoing IP for the public is: 1.1.1.1. But when I tunnel using 1.1.1.2 or 1.1.1.3, the outgoing IP for the public is still 1.1.1.1. I've been googling for days, and tried relevant channels

Hi all, I have a Wildcard TE110P connected to a E1 line an I want to reserve channels in the following way: channels 1-15 and 17-21 for incoming calls channels 22-28 for outgoing calls channels 29-31 for emergency calls My zaptel.conf looks like this: ; incoming group = 1 signalling=pri_cpe context=from-zaptel channel => 1-15 channel => 17-21 ; outgoing group = 2 signalling=pri_cpe

Hi, how can I make asterisk do something when an outgoing call is picked up? The background is that I would like to record incoming and outgoing phone calls. In order to do this, I need to play an announcement telling the person calling or being called that the call will be recorded. Here's what I'm trying to do: call comes in: if(I pick up) { play announcement to

Hi, I am developing a call recording application on Asterisk 11.2 and have this configuration in my dialplan: [macro-ccdev2-rec] exten => s,1,MixMonitor(${ARG1},b) [outgoing-originate] exten => _X.,1,NoOp(Will send call to ${EXTEN}) exten => _X.,n,Dial(SIP/${EXTEN}@x.y.z) [outgoing-originate-rec] exten => h,1,Agi(agi://localhost/ajpbx.agi?path=uploadrec&callid=${CC_CALLID})

Hello all: I am looking at the documentation of the new firewalld service in CentOS 7. It looks like no matter what I configure with it, outgoing connections are still going to be allowed. That does not seem very secure. I always set my servers to default policy of DROP for everything incoming and outgoing and then add rules to allow very specific traffic through. Is this possible using the