search for: outbound

...this thread "Setting up Security Groups" because I'm trying to set up some sip user groups with certain calling rights, e.g., one group of sip phones have the right to make any type of call. another group of sip phones have the right to make local & toll free calls only. The outbound dialing is broken down into 4 contexts in extensions.conf; outbound-local, outbound-toll, outbound-tollfree, and outbound-longdistance. The sip users are set up in 3 contexts in extensions.conf; sip-superuser, sip-operator and sip-basic. Under default context in extensions.conf, I have an inc...

Hello All, Looking for a little guidance on Real Time Pattern Matching. We are attempting to block outbound 411 via when someone dials NXX-555-XXXX, The must common being NXX-555-1212. However, We have some outbound providers that consider any call to NXX-555-XXXX a directory assistance call. So simply making my pattern _NXX5551212 doesn't work. So as you can see from the lines below. If I Dia...

...ication. [global] type = global debug = yes [transport1] type = transport bind = 0.0.0.0 protocol = udp *local_net=<yourlocalnet I.E. 10.10.10.10/24 <http://10.10.10.10/24>>external_media_address=<your public ip address>external_signaling_address=<your public address>* [outbound.vitelity.net] type = aor remove_existing = yes qualify_frequency = 60 contact = sip:64.2.142.93 [outbound.vitelity.net] type = endpoint context = TestApp transport = transport1 aors = outbound.vitelity.net dtmf_mode = rfc4733 force_rport = yes rtp_symmetric = yes rewrite_contact = yes send_rpid =...

...ly in trixbox. I am hoping somebody who has expertise in trixbox could help me debug this problem. I can post an asterisk log snippet if anybody is interested. Here is the extension. The callout file is below too. ----------------Extension_custom.conf-------------------- [dialout] exten => outbound,1,Answer ; switches to outbound-handler exten => outbound,2,Wait(60) exten => outbound,3,Hangup exten => outbound-handler,1,Dial(${CallInitiate_phonenumber},50,gM(outbound-connect^${AGI_SERVER}${AGI_URL}^${CallInitiate_hashdata}^${MACHINE_STATUS_UNKNOWN})) exten => outbound-handler,2,G...

.... Here's the new sip.conf settings they sent me. type=friend dtmfmode=auto host=64.2.142.93 allow=all nat=yes canreinvite=no trustrpid=yes sendrpid=yes When I use these settings to originate calls using the sip.conf they sent me, everything works. Action: Originate ActionID: S8 Channel: SIP/outbound.vitelity.net/8005555555 Exten: createcall Context: TestApp Priority: 1 Timeout: 60000 CallerID: John Doe <1234> Variable: CALLERID(num-pres)=allowed_passed_screened Async: true I translated those settings to the following for pjsip.conf... [transport1] type = transport bind = 0.0.0.0 proto...

Hi All, I setup auto dial on my asterisk server. The problem is asterisk does not wait for called party to answer the call but proceed to process the extension specifed in my .call file My sample call file : hannel: local/0124787924@outbound-reminder MaxRetries: 5 RetryTime: 300 WaitTime: 40 Account: Reminder context: remindem extension: s priority: 1 Set: MSG=0135.20070601.0124787924 Set: APPTDT=20070601 Set: APPTTIME=0135 Set: APPTPHONE=0124787924 Set: CALLATTEMPTS=5 Set: CALLDELAY=300 My outbound-reminder context: [outbound-remind...

....c:696 dial_exec: Unable to create channel of type 'SIP)' Jan 18 10:10:05 WARNING[-1115923536]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call MGC12320050118100959228289-68802635075-54444D000001F742219B3341ED26C644@66.33.1 for seqno 1 (Non-critical Response) My extension.conf outbound dial peer: [outbound] exten => _124XX,1,Dial(SIP)/${EXTEN:1}@outbound) exten => _124XX,2,Playback(invalid) exten => _124XX,3,Hangup My sip.conf [outbound] type=peer host=192.168.1.1 What the * needs to do is receive the call via SIP and then send it out dialed extension via SIP to an a...

Well it appears i have no clue what im doing. I thought i had the below script working to shape outbound ftp traffic....however, it is shaping inbound traffic too. I have NO clue why. Please comment if anyone has any ideas why this doesnt work. I want to shape only outbound ftp traffic and not inbound or lan traffic. #!/bin/bash #shaping passive and active outbound ftp traffic on an internal c...

Hello, I want to use Asterisk to use Kamailio as an outbound proxy for routing calls to remote SIP end points, one option could be to use a default peer, but in my case, my outbound proxy can change based on the remote end point, so this option doesn't work. And another problem is that I don't know how to configure Asterisk to prepare the Request-URI...

Hello all, Here is my problem, I try to place a call to FWD (free world dialup) trough my asterisk PBX. my config is as follow: extensions.conf ---------------- [internal] exten => 613,1,Dial(IAX2/iaxfwd-outbound/613) (service echo de FWD) exten => xxxxxx,1,Dial(IAX2/iaxfwd-outbound/xxxxxx) mon numero FWD exten => yyyyyy,1,Dial(IAX2/iaxfwd-outbound/yyyyyy) celui d'un ami FWD iax.conf -------- [general] context=default bandwidth=low disallow=lpc10 jitterbuffer=no forcejitterbuffer=no tos=lowdel...

Hi I need a small PBX for use on the move. This means that outbound calls will need to be made over the cell phone network. Assuming a small hardware PBX with a spare mini-PCI slot or a USB slot then what hardware options do I have to get an outbound cellular channel? Options need to be rock solid, so no bluetooth to a cell phone kind of solutions need apply....

...ers with 1.6 as something of a testbed, but I'm encountering a few problems, and wondering if anyone can help... In extensions.conf, there are a number of contexts defined for each group of users, along the lines of: [groupa] [groupb] etc. In each of those, there's a command include => outbound [outbound] has entries similar to the following: exten => _0[123]XXXXXXXX,1,Macro(outbound,${EXTEN}, provider1, provider2) the macro "outbound" is defined in extensions.ael as follows: macro outbound (number, route1, route2) { dosomestuff; } This has worked fine in 1.2 and 1.4, but...

Recently, I've been having issues with the URIs returned from e.164.org and toll free calls. It seems that the URIs that are returned from ENUMQUERY and ENUMRESULT are no longer the proper numbering schemes that the poviders use. I've been using the following [enum] template in my outbound route for quite some time with great success until recently. [enum](!) exten => _X.,n,Set(ARRAY(i,id)=1,${ENUMQUERY(+${EXTEN},ALL,e164.org)}) exten => _X.,n,Set(max=${ENUMRESULT(${id},getnum)}) exten => _X.,n,While($["${i}" <= "${max}"]) exten => _X.,n,Set(uri=$...

I have two asterisk servers. One is at location 1 and the other is at location 2. What I am trying to do seems straightforward. I want the Asterisk server at location 2 to send all it outbound calls to the Asterisk Server at location 1. Both asterisk servers can dial each other using extensions without a problem, but when users on Asterisk server 2, dial 9XXX-XXX-XXXX the call never reaches the zap channel on Asterisk server 1. I have a workaround working right now using switch =>...

Hi There, We have single PRI with multiple DID numbers and its working fine in receiving call. And if you make outbound call it will send main-line CallerID (company name). Now we want individual caller id for per extensions on outbound calls. like if i call someone he will get my extension as callerid ( 617-838-XXXX) XXXX is my sip extension something like this so next time i direct get call from users. How to do...

...e's an update... My network admin would not turn off the ALG because it would cause several other problems to other phone systems we have. He looked at the sip trace. What he found is the PJSIP trace showed a different IP address than the older chan_sip so he had me change the aor contact to outbound.vitelity.net At this point, it seems to be working (and this is going through a Cisco ALG). I will run more tests, but here is the pjsip.conf I have. [global] type = global debug = yes [transport1] type = transport bind = 0.0.0.0 protocol = udp [outbound.vitelity.net] type = aor remove_existi...

...e being shaped for some reason. >Could this be shaping on the ISP side?? What >happens when the tc rules >are shut off?? No, everything works fine >Can you determine what ports are being used for >inbound data transfers? >What makes you select those ports you defined as >the outbound?? Same ports, 50000-51000 and 65437. I choose these ports because they are the ports that I have passive ftp traffic on and 65437 is the active ftp port. I just dont understand why the inbound traffic is being limited. The outbound shaping works fine. Script: I am using the following script...

Hello! I've run into a problem where a user is making an outbound call at the same time that an inbound call is being made on the same analog line. It appears that as the zap channel is opened for the outbound call, it is simply answering the inbound call. Obviously, both parties involved in the calling get a bit confused. Previously, it happened only on an occas...

...For the application I?m working on, the only endpoint is the endpoint to > Vitelity. > > We use AMI to Originate calls from Asterisk endpoint through Vitelity to > phones. > > After that, we control the call through AMI to perform the work we need. > > > > And it's outbound calls that aren't working right? > > > > > *From:* asterisk-users-bounces at lists.digium.com [mailto: > asterisk-users-bounces at lists.digium.com] *On Behalf Of *George Joseph > *Sent:* Monday, December 15, 2014 4:42 PM > *To:* Asterisk Users Mailing List - Non-Comme...

...way. Is there anything special I should to to make this work? Note my soft phone does not have any issues using the same dialing rules and extension information. Here is some of my config stuff: ns1*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.116 5060 Unmonitored 101/101 68.156.63.118 D N 1038 Unmonitored 3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 0 offline] Inboun...