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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

...hone rings but after i answer it, i have no voice and call terminates after 5 seconds. this is ooh323 debug(in asterisk11.13.1 system): ooh323_get_rtp_peer OOH323/peer-2-5 -> (null):0, 1 this ia h322_log: 10:42:10:835 Processing MakeCall command ooh323c_o_3 10:42:10:835 Created a new call (outgoing, ooh323c_o_3) 10:42:10:835 Enabled RTP/CISCO DTMF capability for (outgoing, ooh323c_o_3) 10:42:10:835 Enabled RFC2833 DTMF capability for (outgoing, ooh323c_o_3) 10:42:10:835 Dtmf mode set to H.245(alphanumeric) for (outgoing, ooh323c_o_3) 10:42:10:835 Dtmf mode set to H.245(signal) for (out...

...-------------------- [avaya] type=peer context=default ip=192.168.0.14 ; UPDATE with appropriate ip address port=1720 ; UPDATE with appropriate port e164=101 disallow=all allow=ulaw allow=alaw /var/log/asterisk/h323_log ----------------------------- 18:44:37:802 Using configured media info (outgoing, ooh323c_o_10) 18:44:37:802 Created new logical channel entry (outgoing, ooh323c_o_10) 18:44:37:802 Building reverse olc. (outgoing, ooh323c_o_10) 18:44:37:802 Receive channel of type audio started (outgoing, ooh323c_o_10) 18:44:37:802 FastStart Element = { 18:44:37:802 forwardLogicalCha...

...load asterisk from the CLI. I have to kill the asterisk process and run safe_asterisk again. any ideas? asterisk*CLI> show channels Channel (Context Extension Pri ) State Appl. Data Zap/31-1 (default 9388 1 ) Dialing AppDial (Outgoing Line) SIP/1024-1330 (network 968290897 2 ) Ring Dial Zap/g2/68290897 Zap/30-1 (default s 1 ) Dialing AppDial (Outgoing Line) SIP/1004-bca1 (network 993841544 2 ) Ring Dial Zap/g2/93841544 Zap/29-1 (default...

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 WARNING[1102211008]: pbx_spool.c:184 apply_outgoing: Unknown keyword ' Channel' at line 1 of /var/spool/asterisk/outg...

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: Number1/Provider1 Phone 2: Incoming Call: Number1/Provider2 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 apache 100 Jan 1 2016 5680a312a28b2.call" and the calls get sent when the date comes to pass. But my question is, if I mv 3 files from my...

    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 calls the get random characters instead of the media port for RTP.     We are using As...

...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 get different no and if extn: 202 make an outgoing call the recipient should get different one. Please can someone help me in this. Thanks Shantanu -------------- next part -------------- An HTML attachment was scrub...

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...

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...

??? 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 outgoin...

...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 find any information to help with troubleshooting. Could someone assist with the fix? **Last few line from logfile** 12:48:19:257 Queued H245 messages 1. (outgoing, ooh323c_o_2) 12:48:19:257 msgCtxt Reset? Done (outgoing, ooh323c_o_2) 12:48:19:257 MasterSlaveDetermination done - Slave(outgoing, ooh323c_o_2) 12:48:19:257 Not opening logical channels as Cap exchange remaining 12:48:19:257 Finished handling H245 message. (outgoing, ooh323c_o_2) 12:48...

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

...(pid = 12955) Channel Location State Application(Data) DAHDI/9-1 main at FnacSACLojas:2 Up Playback(audio/FnacSAC/qualida DAHDI/31-1 s at ZON:7 Up BackGround(audio/ZON/prima1) SIP/209-00000570 main at agents:1 Up AppQueue((Outgoing Line)) SIP/604-0000056e t at agents:1 Up AppQueue((Outgoing Line)) DAHDI/5-1 main at FnacSACBilhetei Up Queue(FnacSACBilheteira,t,,,18 SIP/206-0000056c main at agents:1 Up AppQueue((Outgoing Line)) SIP/234-0000056b 1 at agents:1 Up...

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 on freenode. I tried to read the manual but I'm not experienced enough to fully grasp it. According to th...

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 channel => 22-28 ; emergency group = 3 signalling=pri_cpe channel => 29-31...

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...

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}) exten => _X,1,NoOp(Will send call to ${EXTEN}, CC_CALLID is ${CC_CALLID}, C...

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 new firewalld service or should I disable it...