Re: [asterisk-users] Block outbound calls based on IP address
Thanks for the reply however it is not possible to get the public IP address using the SIP_HEADER function (see my original post). We have many devices connecting from hundreds of dynamic external IPs. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Block outbound calls based on IP address
Thanks.
exten = s,n,Set(foo=${CHANNEL(peerip)}) ; Doesn't return anything
exten = s,n,Set(foo=${CHANNEL(recvip)}) ; Doesn't return anything
exten = s,n,Set(foo=${SIPCHANINFO(peerip)}) ; Returns public IP when
calling from a SIP device
Strange that CHANNEL doesn't return anything.
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Re: [asterisk-users] Block outbound calls based on IP address
Hi,
How many Public IPs connect to you ? If they are less than 15 or 10 , I
suggest you make sip.conf peers for them with host=Publicip and then decide
if you want that to be blocked or rerouted to some other direction !
If that isn't doable then try extracting/parsing some IP using the
SIP_HEADER function. Collect some header on incoming call and extract your
required IP field and then do some DB operation etc
Regards,
Sammy
On Tue, Aug 7, 2012 at 2:40 PM, CB kj...@xnet.co.nz wrote:
Thanks.
exten = s,n,Set(foo=${CHANNEL(peerip)}) ; Doesn't return anything
exten = s,n,Set(foo=${CHANNEL(recvip)}) ; Doesn't return anything
exten = s,n,Set(foo=${SIPCHANINFO(peerip)}) ; Returns public IP when
calling from a SIP device
Strange that CHANNEL doesn't return anything.
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[asterisk-users] Block outbound calls based on IP address
We are looking to further secure our Asterisk installation by inspecting the IP address that a SIP INVITE comes from and performing some logic to determine whether the call should proceed. The purpose of this is to prevent calls to certain expensive destinations if the SIP message is coming from a foreign IP that we don't expect. I can see that it's possible to use the SIP_HEADER function however that may not contain the public IP address. For example here is an invite from the external IP address 58.28.1.1 but that information is not contained in the SIP header: U 58.28.1.1:5060 - 203.89.1.1:5060 INVITE sip:1...@domain.com SIP/2.0..Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK-d8754z-fc116e03a80ef774-1---d8754z-;rport. .Max-Forwards: 70 ..Contact: sip:000333082261336@192.168.1.103:5060..To: sip:1...@domain.com..From: sip:000333082261...@domain.com;tag=7 dcb1e4d..Call-ID: NDMyZmRhY2Q4ZjNhMjAxMDJhOTA3OTU0MzMyNTkzNjI...CSeq: 1 INVITE..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INF O..Content-Type: application/sdp..Supported: replaces..User-Agent: X-Lite release 5.0.0 stamp 67284..Content-Length: 217v=0..o=- 12988751314362048 1 IN IP4 192.168.1.103..s=CounterPath X-Lite 5.0.0..c=IN IP4 192.168.1.103..b=AS:1638..t=0 0..m=audio 5062 RTP/AVP 0 8 3 101..a=rtpmap:101 telephone-event/8000..a=fmtp:1 01 0-15..a=sendrecv.. Is it possible to determine the public IP address from the dialplan? Any advice appreciated. attachment: winmail.dat-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Agents outbound calls to be recorded
Hi All; I know that incoming calls for the agent can be recorded, but how I can let the outbound calls for the agents to be recorded? I can determine the directory to store the outbound calls of the agents to be other than the directory to store the incoming calls of the agents? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Agents outbound calls to be recorded
Hi All; I know that incoming calls for the agent can be recorded, but how I can let the outbound calls for the agents to be recorded? I can determine the directory to store the outbound calls of the agents to be other than the directory to store the incoming calls of the agents? Regards Bilal Hi Bilal, use MixMonitor() in the outgoing extension for your agents. Regards, Ruben -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Agents outbound calls to be recorded
On 7/6/2011 4:36 AM, bilal ghayyad wrote:
Hi All;
I know that incoming calls for the agent can be recorded, but how I can let the
outbound calls for the agents to be recorded? I can determine the directory to
store the outbound calls of the agents to be other than the directory to store
the incoming calls of the agents?
Regards
Bilal
This is an example of what we do.
MixMonitor(crm/${STRFTIME(${EPOCH},,%B)}/${STRFTIME(${EPOCH},,%d-%m-%Y)}/${STRFTIME(${EPOCH},,%Y%m%d)}-${EXTEN:3}N-${UNIQUEID}-${CALLERID(NUM)}.wav,v(-1)V(2)b,)
What this does is save the recording in:
/var/spool/asterisk/monitor/crm/July/06-07-2011/ (Date in Euro format)
with name:
mmdd-dialednumber-uniqueid-extensionthatdialed.wav
Warning:
I've seen 1.8 create the directory if it does not exist. Asterisk 1.4
will NOT create it. Don't know what 1.6 does with it.
Alex
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Re: [asterisk-users] no outbound calls
You have to check and verify the SIP trunk details, as ext to ext works once the pbx is up, but to call out, it should go through your provider.so just recheck your provider's details. Regards Sandesh On Wed, Oct 14, 2009 at 8:24 AM, Ott Rose sixfourimp...@hotmail.com wrote: here is the debug from the CLI. I think I know where the problem is I just can figure out how to fix it. The IP in the From and To i think is where the problem is. When I make an outbound call. i get the message the call cannot be completed as dialed. if i call another ext it works. I posted the debug for both calls. ==outbound call=== --- Transmitting (NAT) to 10.0.0.46:5060 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.0.46:5060 ;branch=z9hG4bKfd2143ede5319cf9b.273e4b904b0cc3101;received=10.0.0.46 From: ext sip:1...@10.0.0.8 sip%3a...@10.0.0.8;tag=9d9e3944ba To: 93214545 sip:93214...@10.0.0.8 sip%3a93214...@10.0.0.8 ;tag=as290bd498 Call-ID: 401d30b0a1893e80 CSeq: 13401 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:99676...@10.0.0.8 sip%3a99676...@10.0.0.8 Content-Type: application/sdp Content-Length: 254 v=0 o=root 3609 3609 IN IP4 10.0.0.8 s=session c=IN IP4 10.0.0.8 t=0 0 m=audio 14398 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv = ext to ext=== SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.46:5060 ;branch=z9hG4bK280378de3608b5bf1.2576ced30c6198b74;received=10.0.0.46 From: ext sip:1...@10.0.0.8 sip%3a...@10.0.0.8;tag=d729237fcc To: 111 sip:1...@10.0.0.8 sip%3a...@10.0.0.8;tag=as553ab5e9 Call-ID: c7cc32657c620790 CSeq: 8007 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:1...@10.0.0.8 sip%3a...@10.0.0.8 Content-Type: application/sdp Content-Length: 254 v=0 o=root 3609 3609 IN IP4 10.0.0.8 s=session c=IN IP4 10.0.0.8 t=0 0 m=audio 10414 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv -- Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. http://clk.atdmt.com/GBL/go/177141664/direct/01/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] no outbound calls
here is the debug from the CLI. I think I know where the problem is I just can figure out how to fix it. The IP in the From and To i think is where the problem is. When I make an outbound call. i get the message the call cannot be completed as dialed. if i call another ext it works. I posted the debug for both calls. ==outbound call=== --- Transmitting (NAT) to 10.0.0.46:5060 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.0.46:5060;branch=z9hG4bKfd2143ede5319cf9b.273e4b904b0cc3101;received=10.0.0.46 From: ext sip:1...@10.0.0.8;tag=9d9e3944ba To: 93214545 sip:93214...@10.0.0.8;tag=as290bd498 Call-ID: 401d30b0a1893e80 CSeq: 13401 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:99676...@10.0.0.8 Content-Type: application/sdp Content-Length: 254 v=0 o=root 3609 3609 IN IP4 10.0.0.8 s=session c=IN IP4 10.0.0.8 t=0 0 m=audio 14398 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv = ext to ext=== SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.46:5060;branch=z9hG4bK280378de3608b5bf1.2576ced30c6198b74;received=10.0.0.46 From: ext sip:1...@10.0.0.8;tag=d729237fcc To: 111 sip:1...@10.0.0.8;tag=as553ab5e9 Call-ID: c7cc32657c620790 CSeq: 8007 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:1...@10.0.0.8 Content-Type: application/sdp Content-Length: 254 v=0 o=root 3609 3609 IN IP4 10.0.0.8 s=session c=IN IP4 10.0.0.8 t=0 0 m=audio 10414 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141664/direct/01/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transferring Outbound Calls
Incoming calls ring SIP users who have |Ttr in their dial plan, but outgoing
calls are done through a macro as follows:
[macro-diallink2voip]
exten = s,1,Dial(SIP/[EMAIL PROTECTED],120)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-ANSWER,1,Hangup
exten = s-CONGESTION,1,Dial(SIP/[EMAIL PROTECTED],120)
exten = s-CONGESTION,2,Goto(ss-${DIALSTATUS},1)
exten = s-CANCEL,1,Hangup
exten = s-BUSY,1,Busy(30)
exten = s-CHANUNAVAIL,1,Dial(SIP/[EMAIL PROTECTED],120)
exten = s-CHANUNAVAIL,2,Goto(ss-${DIALSTATUS},1)
exten = ss-ANSWER,1,Hangup
exten = ss-CONGESTION,1,Congestion(30)
exten = ss-CANCEL,1,Hangup
exten = ss-BUSY,1,Busy(30)
exten = ss-CHANUNAVAIL,1,Congestion(30)
When a user presses # both callers hear the keytone instead of getting a
transfer prompt on
outbound calls. Would I be correct in assuming that I could add ,Ttr after
the 120 on all
the Dial lines? I am remote and need to direct a user to make this change who
isn't very
technical so getting it right the first time would be great :)
Thanks!
jlc
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[Asterisk-Users] Re: Outbound calls through Broadvoice
Thanks Ronald!
Works like a charm. What my brain sees and what my hands type are not always
the same thing! doh!
MR
maybe just a typo, br_OA_dvoice
I gone away from broadvoice, since they admitted to have troubles and I
had still to pay for NO phone call !!! (multiple lines)
bye
Ronald Wiplinger
Hi all, a noob here, I am trying to get outbound calls through
asterisk working with Broadvoice.
I have consulted the following two online tutorials:
http://www.broadvoice.com/support_install_asterisk.html
http://www.voip-info.org/wiki/view/Asterisk+settings+Broadvoice
in an effort to make outbound calls.
My current settings are as follows:
sip.conf
register =
[EMAIL PROTECTED]:SECRET:[EMAIL PROTECTED]/XX
where XX = our phone number including area code
and secret is our broadvoice defined secret
[sip.braodvoice.com]
type=peer
dynamic=yes
username=XX
fromuser=XX
authname=XX
user=phone
secret=SECRET
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
outboundproxy=sip.broadvoice.com
insecure=very
dtmfmode=inband
dtmf=inband
canreinvite=no
context=incoming
I receive the following error through asterisk when attempting a call:
Apr 8 13:08:43 WARNING[17425]: chan_sip.c:9634
handle_response_invite: Forbidden - wrong password on authentication
for INVITE to 'My Name sip:[EMAIL PROTECTED];tag=as23aa39db'
Now, we can receive incoming calls perfectly fine, but I just can't
wrap my head around what is wrong with the outgoing. I figure it's
got to be the way I am passing the phone number to call to Broadvoice:
exten = _3XNXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _3XNXXNXX,2,Congestion
or possibly, the fact that my name is showing up in the outbound call,
but the account isn't registered to my name, but someone else where I
work.
or my conf files are wrong somehow?
Otherwise, I got nothing.
Any help would be greatly appreciated by one frustrated noob!
oh, please CC me at mraley [at] syndiolemurgroup.com [remove the
mammal species]
Thanks!
Mike
--
\***/
~ Mike Raley ~
Asterisk / VoIP Programmer
Syndio Group, LLC.
www.syndiogroup.com
[EMAIL PROTECTED]
/***\
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Re: [Asterisk-Users] TDM04B Outbound calls
I just install a Digium TDM04B card. I created 4 separate Zap channels and one outbound routing containing zap channels from 1 to 4. If a phone line is plug in Zap/1 then works fine, but if I unplug phone line from Zap/1 (simulating fail) the system keep dialing out on Zap/1, even with no dial tone; Only if Zap/1 is busy it will use Zap/2. There is any work around or different setting to avoid this situation? I don't believe asterisk has any code to detect whether a pstn line is plugged in or not. The chipset on the TDM-fxo modules do support that function, but the drivers don't do anything with it right now. Mark added code for that about a year ago, but commented it out within a day or two as it caused problems for some folks. Don't know if that code remains in the drivers as yet or not. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM04B Outbound calls
If that is the case, then why on one port FXO (WCFXO) it work different. If there is no dial tone on this card system will play All circuit are busy now and if a second card is installed the call will rollover to the second card automatically. My concern is on a 4 line system if the first line loose dial tone nobody can make outgoing calls unless first channel is busy. - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, July 10, 2005 1:42 PM Subject: Re: [Asterisk-Users] TDM04B Outbound calls I just install a Digium TDM04B card. I created 4 separate Zap channels and one outbound routing containing zap channels from 1 to 4. If a phone line is plug in Zap/1 then works fine, but if I unplug phone line from Zap/1 (simulating fail) the system keep dialing out on Zap/1, even with no dial tone; Only if Zap/1 is busy it will use Zap/2. There is any work around or different setting to avoid this situation? I don't believe asterisk has any code to detect whether a pstn line is plugged in or not. The chipset on the TDM-fxo modules do support that function, but the drivers don't do anything with it right now. Mark added code for that about a year ago, but commented it out within a day or two as it caused problems for some folks. Don't know if that code remains in the drivers as yet or not. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM04B Outbound calls
I just install a Digium TDM04B card. I created 4 separate Zap channels and one outbound routing containing zap channels from 1 to 4. If a phone line is plug in Zap/1 then works fine, but if I unplug phone line from Zap/1 (simulating fail) the system keep dialing out on Zap/1, even with no dial tone; Only if Zap/1 is busy it will use Zap/2.There is any work around or different setting to avoid this situation? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Limiting outbound calls
My VoIP provider allows me to have more than one call outbound on the same line simultaneously, for some reason. I am pretty sure that they do not want this to happen, so I'd like instead to limit each line to one call. I do not want the users to have to dial another prefix to go out on another line. Is there any way to add multiple accounts for my _9. extension and have Asterisk automatically find one that is not being used? I know * keeps track of which lines are in use, so I do not think it should be hard to do. I have looked at SetGroup and CheckGroup(1) commands but I can't figure out how to have it try the next line then the next and so on in case of failure. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limiting outbound calls
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Adrian Avramescu wrote:
My VoIP provider allows me to have more than one call outbound on the
same line simultaneously, for some reason. I am pretty sure that they
do not want this to happen, so I'd like instead to limit each line to
one call.
I do not want the users to have to dial another prefix to go out on
another line. Is there any way to add multiple accounts for my _9.
extension and have Asterisk automatically find one that is not being
used?
I know * keeps track of which lines are in use, so I do not think it
should be hard to do. I have looked at SetGroup and CheckGroup(1)
commands but I can't figure out how to have it try the next line then
the next and so on in case of failure.
try:
exten = _9.,1,SetGroup(voip1)
exten = _9.,2,CheckGroup(1)
exten = _9.,3,Dial(${VOIP1}...)
exten = _9.,4,Hangup()
exten = _9.,103,NoOp()
exten = _9.,104,SetGroup(voip2)
exten = _9.,105,Dial(${VOIP2}...)
exten = _9.,106,Hangup()
exten = _9.,205,NoOp()
exten = _9.,206,Playtones(congestion)
exten = _9.,207,Congestion()
exten = _9.,208,Hangup()
HTH
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961 Gossiptel:9309811
N 52.567623, W 2.137621
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[Asterisk-Users] distribute outbound calls
Any ideas on how to rotate (evenly distribute) outbound calls over a number of 'trunks' or contexts? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] distribute outbound calls
On 4/14/05, jltaylor [EMAIL PROTECTED] wrote: Any ideas on how to rotate (evenly distribute) outbound calls over a number of 'trunks' or contexts? Funny I was just looking at the following thread, looks like it might work. http://lists.digium.com/pipermail/asterisk-users/2003-May/011484.html Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] distribute outbound calls
Rotate or make sure the line you are dialling out on isn't in use before you try and use it? Lookup setgroup/checkgroup on the Wiki if it's the latter - http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup Allows you to create logical groups for just about anything, check whether the groups are full perform conditional steps, etc. Can post sample syntax if you're interested. D. FFF Managed Technology Ltd 60 Cook St P.O. 6368 Wellesley St Auckland t +64 9 356 2911 f +64 9 358 9070 m +64 21 415 297 w www.fff.co.nz jltaylor wrote: Any ideas on how to rotate (evenly distribute) outbound calls over a number of 'trunks' or contexts? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] distribute outbound calls
Groups for each trunk and check the dial plan groupcount and cycle thru the trunks or keep a list of trunks in a DB and just loop thru that first call route 1 second route 2 etc. I'll give it some more thought when I wake up but I think you would have to track concurrent channels per trunk to balance it properly. -- William ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Restrict outbound calls on Broadvoice
Hello, I have been researching Asterisk and have a few questions. Is it possible to allow users to only call certain area codes? Reason for this is that the plan from BroadVoice allows unlimited calling to certain countries. Any country outside of this plan would be charged on a per minute basis and I am trying to avoid that. Another question is if there is a way to distinguish aland line phone number from a cell phone number? Some plans charge more for cell phone calls. Last question I have is if I subscribe to Broadvoice, is it possible to have 3 SIP phones connected to the Asterisk PBX and have them all make simultaneous calls under 1 account with Broadvoice? Main reason why I'm trying to setup an Asterisk server is so that some hotel guests can dial long distance but I wouldn't be allowing incoming calls. The hotelhas regular phones for this. Regards, Jimmy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Restrict outbound calls on Broadvoice
Jimmy,
allow users to only call certain area codes?
Create a dial plan that requires the area code to match, and send all other
non-matching calls into an extension that plays an 'unauthorized' message,
or whatever you want.
I have this for the world plan
[broadvoice-toll-free]
; US calls (7 and 11 digits)
exten = _NXX,1,Macro(outcall,${EXTEN})
exten = _1NXXNXX,1,Macro(outcall,${EXTEN:1})
; Broadvoice International free
exten = _01131.,1,Macro(outcall,${EXTEN})
exten = _01132.,1,Macro(outcall,${EXTEN})
exten = _01133.,1,Macro(outcall,${EXTEN})
exten = _01134.,1,Macro(outcall,${EXTEN})
exten = _011353.,1,Macro(outcall,${EXTEN})
exten = _01139.,1,Macro(outcall,${EXTEN})
exten = _01141.,1,Macro(outcall,${EXTEN})
exten = _01143.,1,Macro(outcall,${EXTEN})
exten = _01144.,1,Macro(outcall,${EXTEN})
exten = _01145.,1,Macro(outcall,${EXTEN})
exten = _01146.,1,Macro(outcall,${EXTEN})
exten = _01147.,1,Macro(outcall,${EXTEN})
exten = _01149[2-9].,1,Macro(outcall,${EXTEN})
exten = _01156.,1,Macro(outcall,${EXTEN})
exten = _01161.,1,Macro(outcall,${EXTEN})
exten = _01165.,1,Macro(outcall,${EXTEN})
exten = _011852.,1,Macro(outcall,${EXTEN})
exten = _01186.,1,Macro(outcall,${EXTEN})
exten = _011886.,1,Macro(outcall,${EXTEN})
The outcall macro sets a few more things up (namely determine the outgoing
line to use) and Dials([EMAIL PROTECTED])
way to distinguish a land line phone number from a cell phone number?
Certain countries have designated area codes for cell phones. For Germany,
it's 17... so I simply disallow any area code that starts with 1 in the plan
above. That works for me for now. Filtering cell phones in other
countries is harder...
3 SIP phones connected to the Asterisk PBX and have them
all make simultaneous calls under 1 account with Broadvoice?
Yes. See the Dial command, you can specify multiple extensions to ring:
ex: Dial(SIP/EXT1SIP/EXT2SIP/EXT3)
--Luki
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[Asterisk-Users] Local Outbound Calls on PRI
I'm in the process of turning up a PRI in one of my markets and have run into a problem I have never seen before. I am unable to place a local outgoing call. Long Distance over the same PRI works fine. When I attempt to place a local call using the PRI I see Asterisk attempt to dial, and am greeted with a busy signal. This signal appears to originate on the telco's switch. I have had a central office tech from the CLEC insert a monitor in the distribution point on the switch and observe call flow. According to the tech call flow appears proper, and he was able to tell me the number I was calling from and the number I attempted to dial. He then placed a PRI test-set at the distribution point in the switch and successfully made and terminated a variety of calls from that point. He then took the PRI test set out to my physical location and did the same test, made and received local and long distance calls on one of my trunks. After thinking about this last night I decided to re-update my Asterisk installation to the latest bleeding edge CVS version and re-test from my test extension with the exact same results. I successfully make long distance calls, successfully receive any calls, but the local calls originated from the SIP phone (SNOM200 and Mediatrix2102) fail with a busy signal that seems to originate from the CLEC's switch. Any suggestions? Thanks in advance. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Local Outbound Calls on PRI
Some switches are fussy about you getting the NPI and TON (sometimes jointly known as the dial plan) right. That is usually the cause of the problem you see. Regards, Steve Paul Oster wrote: I'm in the process of turning up a PRI in one of my markets and have run into a problem I have never seen before. I am unable to place a local outgoing call. Long Distance over the same PRI works fine. When I attempt to place a local call using the PRI I see Asterisk attempt to dial, and am greeted with a busy signal. This signal appears to originate on the telco's switch. I have had a central office tech from the CLEC insert a monitor in the distribution point on the switch and observe call flow. According to the tech call flow appears proper, and he was able to tell me the number I was calling from and the number I attempted to dial. He then placed a PRI test-set at the distribution point in the switch and successfully made and terminated a variety of calls from that point. He then took the PRI test set out to my physical location and did the same test, made and received local and long distance calls on one of my trunks. After thinking about this last night I decided to re-update my Asterisk installation to the latest bleeding edge CVS version and re-test from my test extension with the exact same results. I successfully make long distance calls, successfully receive any calls, but the local calls originated from the SIP phone (SNOM200 and Mediatrix2102) fail with a busy signal that seems to originate from the CLEC's switch. Any suggestions? Thanks in advance. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Local Outbound Calls on PRI
On Fri, 24 Sep 2004 10:06:06 -0500, Paul Oster [EMAIL PROTECTED] wrote: I'm in the process of turning up a PRI in one of my markets and have run into a problem I have never seen before. I am unable to place a local outgoing call. Long Distance over the same PRI works fine. When I attempt to place a local call using the PRI I see Asterisk attempt to dial, and am greeted with a busy signal. This signal appears to originate on the telco's switch. Any chance you're sending more or fewer digits than the CLEC expects to see from you? I would expect that the tech would have picked up on it when he was watching your call attempts but just in case... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Local Outbound Calls on PRI
Just tried it with 7,10, and 11 digit dialing, and got the expected error from the switch, the number you have dialed is not a long distance number, there is no need to dial the digit one before the number... Good suggestion, but that doesn't appear to be the problem. On Fri, 24 Sep 2004 11:25:12 -0400, Scott Lykens [EMAIL PROTECTED] wrote: On Fri, 24 Sep 2004 10:06:06 -0500, Paul Oster [EMAIL PROTECTED] wrote: I'm in the process of turning up a PRI in one of my markets and have run into a problem I have never seen before. I am unable to place a local outgoing call. Long Distance over the same PRI works fine. When I attempt to place a local call using the PRI I see Asterisk attempt to dial, and am greeted with a busy signal. This signal appears to originate on the telco's switch. Any chance you're sending more or fewer digits than the CLEC expects to see from you? I would expect that the tech would have picked up on it when he was watching your call attempts but just in case... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Local Outbound Calls on PRI
What are you sending for the CSID? Dialing LD goes through the CLEC and may be excepting your call no matter what the CSID is. The local switch may be rejecting you because the CSID you are sending is not what they are expecting. I had a the same experience on a legacy phone system. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Oster Sent: Friday, September 24, 2004 11:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Local Outbound Calls on PRI Just tried it with 7,10, and 11 digit dialing, and got the expected error from the switch, the number you have dialed is not a long distance number, there is no need to dial the digit one before the number... Good suggestion, but that doesn't appear to be the problem. On Fri, 24 Sep 2004 11:25:12 -0400, Scott Lykens [EMAIL PROTECTED] wrote: On Fri, 24 Sep 2004 10:06:06 -0500, Paul Oster [EMAIL PROTECTED] wrote: I'm in the process of turning up a PRI in one of my markets and have run into a problem I have never seen before. I am unable to place a local outgoing call. Long Distance over the same PRI works fine. When I attempt to place a local call using the PRI I see Asterisk attempt to dial, and am greeted with a busy signal. This signal appears to originate on the telco's switch. Any chance you're sending more or fewer digits than the CLEC expects to see from you? I would expect that the tech would have picked up on it when he was watching your call attempts but just in case... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Local Outbound Calls on PRI
CSID is caller sending ID. This is what number you are sending from the PBX to the local carrier. -Original Message- From: Paul Oster [mailto:[EMAIL PROTECTED]] Sent: Friday, September 24, 2004 12:02 PM To: Henry Devito Subject: Re: [Asterisk-Users] Local Outbound Calls on PRI Could you expand that acronym for me? CSID Calling station ID Called Station ID Not quite sure which you are referrring to in your response. TIA Paul On Fri, 24 Sep 2004 11:20:26 -0500, Henry Devito [EMAIL PROTECTED] wrote: What are you sending for the CSID? Dialing LD goes through the CLEC and may be excepting your call no matter what the CSID is. The local switch may be rejecting you because the CSID you are sending is not what they are expecting. I had a the same experience on a legacy phone system. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Paul Oster Sent: Friday, September 24, 2004 11:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Local Outbound Calls on PRI Just tried it with 7,10, and 11 digit dialing, and got the expected error from the switch, the number you have dialed is not a long distance number, there is no need to dial the digit one before the number... Good suggestion, but that doesn't appear to be the problem. On Fri, 24 Sep 2004 11:25:12 -0400, Scott Lykens [EMAIL PROTECTED] wrote: On Fri, 24 Sep 2004 10:06:06 -0500, Paul Oster [EMAIL PROTECTED] wrote: I'm in the process of turning up a PRI in one of my markets and have run into a problem I have never seen before. I am unable to place a local outgoing call. Long Distance over the same PRI works fine. When I attempt to place a local call using the PRI I see Asterisk attempt to dial, and am greeted with a busy signal. This signal appears to originate on the telco's switch. Any chance you're sending more or fewer digits than the CLEC expects to see from you? I would expect that the tech would have picked up on it when he was watching your call attempts but just in case... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Local Outbound Calls on PRI
Your extensions.conf and console output might help us out quite a bit. Thank you, Steve Maroney On Fri, 24 Sep 2004, Paul Oster wrote: I'm in the process of turning up a PRI in one of my markets and have run into a problem I have never seen before. I am unable to place a local outgoing call. Long Distance over the same PRI works fine. When I attempt to place a local call using the PRI I see Asterisk attempt to dial, and am greeted with a busy signal. This signal appears to originate on the telco's switch. I have had a central office tech from the CLEC insert a monitor in the distribution point on the switch and observe call flow. According to the tech call flow appears proper, and he was able to tell me the number I was calling from and the number I attempted to dial. He then placed a PRI test-set at the distribution point in the switch and successfully made and terminated a variety of calls from that point. He then took the PRI test set out to my physical location and did the same test, made and received local and long distance calls on one of my trunks. After thinking about this last night I decided to re-update my Asterisk installation to the latest bleeding edge CVS version and re-test from my test extension with the exact same results. I successfully make long distance calls, successfully receive any calls, but the local calls originated from the SIP phone (SNOM200 and Mediatrix2102) fail with a busy signal that seems to originate from the CLEC's switch. Any suggestions? Thanks in advance. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Two outbound calls at once
Actually, if I place 2 calls at once, one from the Vodavi and one from a single line on a channel bank, it all works correctly. If I place 2 calls at once from the Vodavi (attached to * via a T1, not PRI) I have the problem. It doesn't matter what numbers I call. There is no problem placing 2 calls at once from the Vodavi when it is connected directly to the CO, but only when I run it through *. Thanks, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C. Maj Sent: Friday, July 09, 2004 8:43 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Two outbound calls at once On Thu, 8 Jul 2004, David Goldfein waxed: Hello, I am having an issue with making two simultaneous outbound calls. When I dial, both phones try to take the same channel and it causes an error. Anyone have any suggestions. My set up is as follows: CO - PRI - ASTERISK - VODAVI(pbx). Thanks, Dave *CLI 8's It doesn't look like you have a channel collision problem, other than the same far end number being dialed. Are you able to place at least one call with success ? --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Two outbound calls at once
On Thu, 8 Jul 2004, David Goldfein waxed: Hello, I am having an issue with making two simultaneous outbound calls. When I dial, both phones try to take the same channel and it causes an error. Anyone have any suggestions. My set up is as follows: CO - PRI - ASTERISK - VODAVI(pbx). Thanks, Dave *CLI 8's It doesn't look like you have a channel collision problem, other than the same far end number being dialed. Are you able to place at least one call with success ? --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Two outbound calls at once
Hello, I am having an issue with making two simultaneous outbound calls. When I dial, both phones try to take the same channel and it causes an error. Anyone have any suggestions. My set up is as follows: CO PRI ASTERISK VODAVI(pbx). Thanks, Dave *CLI -- Starting simple switch on 'Zap/69-1' -- Executing Wait(Zap/69-1, .1) in new stack -- Executing DISA(Zap/69-1, no-password|local) in new stack -- Starting simple switch on 'Zap/68-1' -- Executing Wait(Zap/68-1, .1) in new stack -- Executing DISA(Zap/68-1, no-password|local) in new stack Jul 8 20:44:20 WARNING[-1394406480]: cdr.c:286 ast_cdr_init: CDR already initialized on 'Zap/69-1' -- Executing Dial(Zap/69-1, Zap/g2/6022831234) in new stack -- Called g2/6022831234 Jul 8 20:44:20 WARNING[-1416709200]: cdr.c:286 ast_cdr_init: CDR already initialized on 'Zap/68-1' -- Executing Dial(Zap/68-1, Zap/g2/6022831234) in new stack -- Called g2/6022831234 -- Channel 0/2, span 2 got hangup -- Forcing restart of channel 0/2 on span 2 since channel reported in use -- Hungup 'Zap/26-1' == No one is available to answer at this time -- Executing Congestion(Zap/68-1, ) in new stack -- Channel 0/1, span 2 got hangup -- B-channel 0/2 successfully restarted on span 2 -- Hungup 'Zap/25-1' == No one is available to answer at this time -- Executing Congestion(Zap/69-1, ) in new stack == Spawn extension (local, 2831234, 2) exited non-zero on 'Zap/69-1' -- Hungup 'Zap/69-1' == Spawn extension (local, 2831234, 2) exited non-zero on 'Zap/68-1' -- Hungup 'Zap/68-1'
RE: [Asterisk-Users] No outbound calls at a PRI possible
Hello Darren,
The error messages that you reported in your last e-mail
are actually outbound Q.931 call setup messages that are
being sent to DTAG from your Asterisk machine. The direction
of the message is indicated in the first column of the trace
output in the form of or . Although these are not error
messages I am surprised to see those particular messages
being generated with your current zapata.conf settings; with
pridialplan=local I would have expected something similar to
the following messages during call
setup:
Calling Number (len=14) [ Ext: 0 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user
number not screened (0) 'X58777' ]
Called Number (len=14) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) 'X986600' ]
(I have inserted X in the PSTN numbers above to protect
the innocent Calling and Called parties.)
Please retry pridialplan=local and pridialplan=unknown in
zapata.conf and post the trace results so we compare results.
With pridialplan=local in zapata.conf the outbound call setup
from Asterisk to DTAG should look ideal.
I will try again in the late evening (the pri is in production use in
another Detewe...)
On a different subject, how are your results with telephony
calls from the Asterisk machine to your Hicom PBX? I would
have expected the zaptel.conf entry to have been:
#hicom (siemens)
span=2,0,0,ccs,hdb3,crc4
...so that your Asterisk provides clocking/timing information
for the Hicom.
If this configuration is not set correctly you could find
that the systems seem to communicate well at first but after
a while you might see strange PRI errors (every hour or so)
that relate to clock synchronisation problems.
The Hicom has been switched to secondary clocking... We had some problems
with the cables, so we tried everything possible..
I guess we will change it back later on, so that we could use the Hicom
without * if asterisk stops (could that be?:)
But there is also another problem, if I try to dial out via Hicom to DTAG,
the Hicom sends digit after digit.
My dial line is:
exten = _0.,1,Dial(Zap/g1/${EXTEN:1},60)
and that works fine with SIP and IAX. But with the Hicom I get only the
first two digits and then it trys to dial out: error.
Does I have to use schemes like exten = _0XXX
But I guess that the german numbers have differnt lengths.
Thanke you.
Felix
Hi Felix,
on some UK public switches I have seen similar bad call setup
problems with a release cause of 28 (Invalid number format)
when using:
pridialplan=national
Have you tried:
pridialplan=unknown
in zapata.conf?
It seems as though the omission of the pridialplan= statement
in zapata.conf is treated by Asterisk as pridialplan=national.
We could probably give you more relevant suggestions if you
would enable a more verbose level of output and post the call
setup trace results here. Try the following command from the
Asterisk CLI before making your next call:
pri debug span x
Where x = single integer digit for the PRI span that will be
used to make the outgoing call. (Eg. 1)
Please drop a note to the list (either way) with your results.
HTH
Darren
--
Comgate
TelcoInternetBroadcast
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
ePyron Felix Deierlein
Sent: 09 May 2004 20:32
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No outbound calls at a PRI possible
Hello,
i guess the problem ist pridialplan from zapata.conf
with
pridialplan = local
it works :-). But I still get the error messages:
Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI:
Unknown Number Plan (0)
Presentation: Unknown (67) '' ] Called
Number (len= 9) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '899312' ]
What pridialplan should I use with an
E1 with Euroisdn from the German Telekom (DTAG or T-Com).
Thanks
Felix
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ePyron
Felix Deierlein
Sent: Sunday, May 09, 2004 6:48 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] No outbound calls at a PRI possible
Hello all,
the scenario:
Carrier S2M-- * -S2M--Siemens
|
|
SIP Clients
and many other features
With much help from the list, the PRI links are without alarms and
inbound calls are working fine (from both: Carrier and Siemens).
But I am not able to dial wether outbound nor to the Siemens PBX.
I allways get the message:
== Everyone is busy at this time
After hours of googling and reading and trying I seek
[Asterisk-Users] No outbound calls at a PRI possible
Hello all,
the scenario:
Carrier S2M-- * -S2M--Siemens
|
|
SIP Clients
and many other features
With much help from the list, the PRI links are without alarms and inbound
calls are working fine (from both: Carrier and Siemens).
But I am not able to dial wether outbound nor to the Siemens PBX.
I allways get the message:
== Everyone is busy at this time
After hours of googling and reading and trying I seek help...
Thank you very much.
Felix Deierlein
My extension.conf (only important parts):
[AtInternal]
;exten = 402,1,Macro(stdexten,402,Zap/g2/402)
exten = 402,1,Dial(Zap/g2/595402)
[ePInternal]
include=system
include=test
include=AtInternal
exten = 812,1,Macro(stdexten,812,${ePFfd})
exten = 814,1,Macro(stdexten,814,${ePFjw})
exten = 854,1,Macro(stdexten,854,${ePFch})
exten = 5950,1,Macro(stdexten,812,${ePFfd})
exten = _0.,1,Dial(Zap/g1/${EXTEN:1},60)
[zapata.conf]
[channels]
language=en
context=default
switchtype=euroisdn
;pridialplan=national
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
;pridialplan=national
switchtype = euroisdn
signalling = pri_cpe
group = 1
channel = 1-15
channel = 17-31
immediate=no
switchtype = euroisdn
signalling = pri_net
group = 2
callgroup=2
pickupgroup=2
channel = 32-46
my zaptel.conf
#amt (carrier)
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
#hicom (siemens)
span=2,1,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-62
loadzone=uk
defaultzone=uk
channel = 48-62
PRI Debugging Infos:
Call to Carrier: (Destination was 899312)
-- Executing Dial(SIP/ePfd-b455, Zap/1/899312|60) in new stack
-- Making new call for cr 32774
Protocol Discriminator: Q.931 (8) len=40
Call Ref: len= 2 (reference 6/0x6) (Originator)
Message type: SETUP (5)
Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Speech (0)
Ext: 1 Trans mode/rate: 64kbps, circuit-mode
(16)
Ext: 1 User information layer 1: A-Law (35)
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type:
3
Ext: 1 Channel: 1 ]
Display (len= 6) [ 1Felix ]
Calling Number (len= 7) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user
number not screened (0) '812' ]
Called Number (len= 9) [ Ext: 1 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '899312' ]
Sending Complete (len= 0)
-- Called 1/899312
Protocol Discriminator: Q.931 (8) len=14
Call Ref: len= 2 (reference 32774/0x8006) (Terminator)
Message type: STATUS (125)
Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
Public network serving the local user (2)
Ext: 1 Cause: Info. element nonexist or not implemented
(99), class = Protocol Error (6) ]
Cause data 0: 14 (20)
Cause data 1: 01 (1)
Call State (len= 1) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state:
Call Initiated (1)
-- Processing IE 8 (Cause)
-- Processing IE 20 (Call State)
Protocol Discriminator: Q.931 (8) len=10
Call Ref: len= 2 (reference 32774/0x8006) (Terminator)
Message type: CALL PROCEEDING (2)
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type:
3
Ext: 1 Channel: 1 ]
-- Processing IE 24 (Channel Identification)
Protocol Discriminator: Q.931 (8) len=13
Call Ref: len= 2 (reference 32774/0x8006) (Terminator)
Message type: DISCONNECT (69)
Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
Public network serving the local user (2)
Ext: 1 Cause: Invalid number format (28), class = Normal
Event (1) ]
Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0:
0 Location: Public network serving the local user (2)
Ext: 1 Progress Description: Inband
information or appropriate pattern now available. (8) ]
-- Processing IE 8 (Cause)
-- Processing IE 30 (Progress Indicator)
-- Channel 1, span 1 got hangup
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication,
peerstate Disconnect Request
Protocol Discriminator: Q.931 (8) len=9
Call Ref: len= 2 (reference 6/0x6) (Originator)
Message type: RELEASE (77)
Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
Private network serving the local user
RE: [Asterisk-Users] No outbound calls at a PRI possible
Hello,
i guess the problem ist pridialplan from zapata.conf
with
pridialplan = local
it works :-). But I still get the error messages:
Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI:
Unknown Number Plan (0)
Presentation: Unknown (67) '' ]
Called Number (len= 9) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '899312' ]
What pridialplan should I use with an
E1 with Euroisdn from the German Telekom (DTAG or T-Com).
Thanks
Felix
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
ePyron Felix Deierlein
Sent: Sunday, May 09, 2004 6:48 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] No outbound calls at a PRI possible
Hello all,
the scenario:
Carrier S2M-- * -S2M--Siemens
|
|
SIP Clients
and many other features
With much help from the list, the PRI links are without
alarms and inbound calls are working fine (from both: Carrier
and Siemens).
But I am not able to dial wether outbound nor to the Siemens PBX.
I allways get the message:
== Everyone is busy at this time
After hours of googling and reading and trying I seek help...
Thank you very much.
Felix Deierlein
My extension.conf (only important parts):
[AtInternal]
;exten = 402,1,Macro(stdexten,402,Zap/g2/402)
exten = 402,1,Dial(Zap/g2/595402)
[ePInternal]
include=system
include=test
include=AtInternal
exten = 812,1,Macro(stdexten,812,${ePFfd})
exten = 814,1,Macro(stdexten,814,${ePFjw})
exten = 854,1,Macro(stdexten,854,${ePFch})
exten = 5950,1,Macro(stdexten,812,${ePFfd})
exten = _0.,1,Dial(Zap/g1/${EXTEN:1},60)
[zapata.conf]
[channels]
language=en
context=default
switchtype=euroisdn
;pridialplan=national
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
;pridialplan=national
switchtype = euroisdn
signalling = pri_cpe
group = 1
channel = 1-15
channel = 17-31
immediate=no
switchtype = euroisdn
signalling = pri_net
group = 2
callgroup=2
pickupgroup=2
channel = 32-46
my zaptel.conf
#amt (carrier)
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
#hicom (siemens)
span=2,1,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-62
loadzone=uk
defaultzone=uk
channel = 48-62
PRI Debugging Infos:
Call to Carrier: (Destination was 899312)
-- Executing Dial(SIP/ePfd-b455, Zap/1/899312|60) in new stack
-- Making new call for cr 32774
Protocol Discriminator: Q.931 (8) len=40 Call Ref: len= 2
(reference
6/0x6) (Originator) Message type: SETUP (5) Bearer
Capability (len= 3)
[ Ext: 1 Q.931 Std: 0 Info transfer
capability: Speech (0)
Ext: 1 Trans mode/rate: 64kbps,
circuit-mode
(16)
Ext: 1 User information layer
1: A-Law
(35) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0,
Exclusive
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified
Channel Type:
3
Ext: 1 Channel: 1 ] Display (len= 6)
[ 1Felix ]
Calling Number (len= 7) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user
number not screened (0) '812' ]
Called Number (len= 9) [ Ext: 1 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '899312' ]
Sending Complete (len= 0)
-- Called 1/899312
Protocol Discriminator: Q.931 (8) len=14 Call Ref: len=
2 (reference 32774/0x8006) (Terminator) Message type: STATUS (125)
Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0)
0: 0 Location:
Public network serving the local user (2)
Ext: 1 Cause: Info. element nonexist or
not implemented
(99), class = Protocol Error (6) ]
Cause data 0: 14 (20)
Cause data 1: 01 (1)
Call State (len= 1) [ Ext: 0 Coding: CCITT (ITU) standard
(0) Call state:
Call Initiated (1)
-- Processing IE 8 (Cause)
-- Processing IE 20 (Call State)
Protocol Discriminator: Q.931 (8) len=10 Call Ref: len=
2 (reference 32774/0x8006) (Terminator) Message type: CALL
PROCEEDING (2) Channel ID (len= 5) [ Ext: 1 IntID:
Implicit, PRI Spare: 0, Exclusive
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified
Channel Type:
3
Ext: 1 Channel: 1 ]
-- Processing IE 24 (Channel Identification) Protocol
Discriminator: Q.931 (8) len=13 Call Ref: len= 2
(reference
RE: [Asterisk-Users] No outbound calls at a PRI possible
Hi Felix,
on some UK public switches I have seen similar bad call setup problems with
a release cause of 28 (Invalid number format) when using:
pridialplan=national
Have you tried:
pridialplan=unknown
in zapata.conf?
It seems as though the omission of the pridialplan= statement in zapata.conf
is treated by Asterisk as pridialplan=national.
We could probably give you more relevant suggestions if you would enable a
more verbose level of output and post the call setup trace results here. Try
the following command from the Asterisk CLI before making your next call:
pri debug span x
Where x = single integer digit for the PRI span that will be used to make
the outgoing call. (Eg. 1)
Please drop a note to the list (either way) with your results.
HTH
Darren
--
Comgate
TelcoInternetBroadcast
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of ePyron Felix
Deierlein
Sent: 09 May 2004 20:32
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No outbound calls at a PRI possible
Hello,
i guess the problem ist pridialplan from zapata.conf
with
pridialplan = local
it works :-). But I still get the error messages:
Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI:
Unknown Number Plan (0)
Presentation: Unknown (67) '' ]
Called Number (len= 9) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '899312' ]
What pridialplan should I use with an
E1 with Euroisdn from the German Telekom (DTAG or T-Com).
Thanks
Felix
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
ePyron Felix Deierlein
Sent: Sunday, May 09, 2004 6:48 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] No outbound calls at a PRI possible
Hello all,
the scenario:
Carrier S2M-- * -S2M--Siemens
|
|
SIP Clients
and many other features
With much help from the list, the PRI links are without
alarms and inbound calls are working fine (from both: Carrier
and Siemens).
But I am not able to dial wether outbound nor to the Siemens PBX.
I allways get the message:
== Everyone is busy at this time
After hours of googling and reading and trying I seek help...
Thank you very much.
Felix Deierlein
My extension.conf (only important parts):
[AtInternal]
;exten = 402,1,Macro(stdexten,402,Zap/g2/402)
exten = 402,1,Dial(Zap/g2/595402)
[ePInternal]
include=system
include=test
include=AtInternal
exten = 812,1,Macro(stdexten,812,${ePFfd})
exten = 814,1,Macro(stdexten,814,${ePFjw})
exten = 854,1,Macro(stdexten,854,${ePFch})
exten = 5950,1,Macro(stdexten,812,${ePFfd})
exten = _0.,1,Dial(Zap/g1/${EXTEN:1},60)
[zapata.conf]
[channels]
language=en
context=default
switchtype=euroisdn
;pridialplan=national
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
;pridialplan=national
switchtype = euroisdn
signalling = pri_cpe
group = 1
channel = 1-15
channel = 17-31
immediate=no
switchtype = euroisdn
signalling = pri_net
group = 2
callgroup=2
pickupgroup=2
channel = 32-46
my zaptel.conf
#amt (carrier)
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
#hicom (siemens)
span=2,1,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-62
loadzone=uk
defaultzone=uk
channel = 48-62
PRI Debugging Infos:
Call to Carrier: (Destination was 899312)
-- Executing Dial(SIP/ePfd-b455, Zap/1/899312|60) in new stack
-- Making new call for cr 32774
Protocol Discriminator: Q.931 (8) len=40 Call Ref: len= 2
(reference
6/0x6) (Originator) Message type: SETUP (5) Bearer
Capability (len= 3)
[ Ext: 1 Q.931 Std: 0 Info transfer
capability: Speech (0)
Ext: 1 Trans mode/rate: 64kbps,
circuit-mode
(16)
Ext: 1 User information layer
1: A-Law
(35) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0,
Exclusive
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified
Channel Type:
3
Ext: 1 Channel: 1 ] Display (len= 6)
[ 1Felix ]
Calling Number (len= 7) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user
number not screened (0) '812' ]
Called Number (len= 9) [ Ext: 1 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '899312' ]
Sending Complete (len= 0)
-- Called 1/899312
Protocol Discriminator: Q.931 (8) len=14 Call Ref: len=
2 (reference 32774/0x8006) (Terminator) Message type: STATUS
RE: [Asterisk-Users] No outbound calls at a PRI possible
Hello Again Felix,
first a quick apology: sorry, I re-read your e-mail and found the trace
information (lower down) that you had already posted. (It's late here, etc.)
The error messages that you reported in your last e-mail are actually
outbound Q.931 call setup messages that are being sent to DTAG from your
Asterisk machine. The direction of the message is indicated in the first
column of the trace output in the form of or . Although these are not
error messages I am surprised to see those particular messages being
generated with your current zapata.conf settings; with pridialplan=local I
would have expected something similar to the following messages during call
setup:
Calling Number (len=14) [ Ext: 0 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user
number not screened (0) 'X58777' ]
Called Number (len=14) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) 'X986600' ]
(I have inserted X in the PSTN numbers above to protect the innocent
Calling and Called parties.)
Please retry pridialplan=local and pridialplan=unknown in zapata.conf and
post the trace results so we compare results. With pridialplan=local in
zapata.conf the outbound call setup from Asterisk to DTAG should look ideal.
On a different subject, how are your results with telephony calls from the
Asterisk machine to your Hicom PBX? I would have expected the zaptel.conf
entry to have been:
#hicom (siemens)
span=2,0,0,ccs,hdb3,crc4
...so that your Asterisk provides clocking/timing information for the Hicom.
If this configuration is not set correctly you could find that the systems
seem to communicate well at first but after a while you might see strange
PRI errors (every hour or so) that relate to clock synchronisation problems.
MfG
Darren
--
Comgate
TelcoInternetBroadcast
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Storer,
Darren
Sent: 10 May 2004 01:29
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No outbound calls at a PRI possible
Hi Felix,
on some UK public switches I have seen similar bad call setup problems with
a release cause of 28 (Invalid number format) when using:
pridialplan=national
Have you tried:
pridialplan=unknown
in zapata.conf?
It seems as though the omission of the pridialplan= statement in zapata.conf
is treated by Asterisk as pridialplan=national.
We could probably give you more relevant suggestions if you would enable a
more verbose level of output and post the call setup trace results here. Try
the following command from the Asterisk CLI before making your next call:
pri debug span x
Where x = single integer digit for the PRI span that will be used to make
the outgoing call. (Eg. 1)
Please drop a note to the list (either way) with your results.
HTH
Darren
--
Comgate
TelcoInternetBroadcast
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of ePyron Felix
Deierlein
Sent: 09 May 2004 20:32
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No outbound calls at a PRI possible
Hello,
i guess the problem ist pridialplan from zapata.conf
with
pridialplan = local
it works :-). But I still get the error messages:
Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI:
Unknown Number Plan (0)
Presentation: Unknown (67) '' ]
Called Number (len= 9) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '899312' ]
What pridialplan should I use with an
E1 with Euroisdn from the German Telekom (DTAG or T-Com).
Thanks
Felix
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
ePyron Felix Deierlein
Sent: Sunday, May 09, 2004 6:48 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] No outbound calls at a PRI possible
Hello all,
the scenario:
Carrier S2M-- * -S2M--Siemens
|
|
SIP Clients
and many other features
With much help from the list, the PRI links are without
alarms and inbound calls are working fine (from both: Carrier
and Siemens).
But I am not able to dial wether outbound nor to the Siemens PBX.
I allways get the message:
== Everyone is busy at this time
After hours of googling and reading and trying I seek help...
Thank you very much.
Felix Deierlein
My extension.conf (only important parts):
[AtInternal]
;exten = 402,1,Macro(stdexten,402,Zap/g2/402)
exten = 402,1,Dial(Zap/g2/595402)
[ePInternal]
include=system
include=test
include=AtInternal
exten = 812,1,Macro(stdexten,812,${ePFfd})
exten = 814,1,Macro(stdexten,814,${ePFjw})
exten = 854,1,Macro(stdexten,854,${ePFch})
exten = 5950,1,Macro(stdexten,812,${ePFfd})
exten = _0.,1,Dial(Zap/g1/${EXTEN:1},60)
[zapata.conf
[Asterisk-Users] Make outbound calls only from certain hosts
Hi,
I'm testing outbound calls for the fist time, using isdn4linux and a cheap
20$ ISDN CARD: it works !
I have more problems restricting pstn calls can I allow inbound sip
access to ALL asterisk features ONLY from the requests sent by my Ser
proxy/registrar ?
In Ser I use this to rewrite authorized requests
if (uri=~[EMAIL PROTECTED]) {
rewritehostport(ipaddress:5090);
t_relay_to_udp(ipaddress, 5090);
break;
};
requests are then managed by asterisk running on the same machine and port 5090
What I want to do is to avoid access to someone calling directly
[EMAIL PROTECTED]:5090 prior of authenticating with ser.
Tnx for any help !
Alessio Focardi
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[Asterisk-Users] SIP: outbound calls
Hi all, Any advice on how to place a call from a SIP UA routed through *? Do I just place a sip call to [EMAIL PROTECTED]:5060 ? I am a little confused, since all of my Uas require registration for presence information. Thanks in advance, Tim -- Notice: This e-mail message, together with any attachments, contains information of Merck Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp Dohme or MSD and in Japan as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
