trixbox - The Open Platform For Business Telephony

Simple Call centre Message taking screen popping script

phillip198 — Sat, 21 Nov 2009 17:53:25 +0000

Hi All,

not sure where to post this, but i have written a very simple script to pop the DDI and CLI on screen and send it to a webpage which then populates some simple fields and can enable a call center agent or message taking company to take calls and pass them to there clients.

like i said its only basic but i have provided it here for you to download...

http://www.p-web.co.uk/downloads/Simple_Call_Scripter.zip

Simple video demonstration here
http://www.youtube.com/watch?v=4D3XHaFH9RM

it uses the ASTassitent to get the DDI and CLI passes it on to the script i have written and then passes it to a formtoemail form to email the client you are taking calls for.

i have already deployed it to a small call center, but this script can be adapted to work with a MySQL database to produce billing etc....

i have provided some instructions in the read me, if you get stuck please post.....

To many files in /var/spool/asterisk/monitor

j333 — Sat, 21 Nov 2009 15:00:44 +0000

many files in /var/spool/asterisk/monitor

I'm getting this error message when I try to listen to voice recording
I believe to money wave file in a system
did anybody know how to delete all the recording file in simple command
or if there is any window interface that somebody recommend to go this task

voicemail to Email problem

maayan1 — Sat, 21 Nov 2009 14:48:38 +0000

Hi all
I am truly sorry for bringing up that old-debated problem but I cant seem to get it to work for me and I dont have any more hair to pull out

I run TB 2.2.4
I have a static IP with a DYNDNS domain to point to it.
I am trying to send voicemail to my Email address at Gmail through the smtp of my domain (maayan-XXXX.com) hosted at dreamhost.com

in sendmail.cf:
define(`SMART_HOST',`mail.maayan-XXXX.com')dnl
FEATURE(`authinfo',`hash -o /etc/mail/authinfo.db')dnl

in authinfo
AuthInfo: "U:user@maayan-XXXX.com" "P:password" "M:PLAIN"

All according to the explanation here:
http://www.trixbox.org/node/28002

this is the tail /var/log/maillog after leaving a VM:

Nov 21 16:16:37 asterisk1 sm-msp-queue[3690]: starting daemon (8.13.1): queueing@01:00:00
Nov 21 16:45:12 asterisk1 sendmail[3765]: nALEjCvp003765: from=asterisk, size=215165, class=0, nrcpts=1, msgid=<Asterisk-1-1816048588-16-3060@asterisk1.local>, relay=asterisk@localhost
Nov 21 16:45:13 asterisk1 sendmail[3767]: nALEjCo5003767: from=<asterisk@asterisk1.local>, size=215296, class=0, nrcpts=1, msgid=<Asterisk-1-1816048588-16-3060@asterisk1.local>, proto=ESMTP, daemon=MTA, relay=localhost [127.0.0.1]
Nov 21 16:45:13 asterisk1 sendmail[3765]: nALEjCvp003765: to="Yafa Office" <ez.gvaot@gmail.com>, ctladdr=asterisk (100/101), delay=00:00:01, xdelay=00:00:01, mailer=relay, pri=245165, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (nALEjCo5003767 Message accepted for delivery)
Nov 21 16:45:16 asterisk1 sendmail[3769]: STARTTLS=client, relay=mx1.balanced.frisky.mail.dreamhost.com., version=TLSv1/SSLv3, verify=FAIL, cipher=DHE-RSA-AES256-SHA, bits=256/256
Nov 21 16:45:17 asterisk1 sendmail[3769]: nALEjCo5003767: to=<ez.gvaot@gmail.com>, ctladdr=<asterisk@asterisk1.local> (100/101), delay=00:00:04, xdelay=00:00:04, mailer=relay, pri=335296, relay=mx1.balanced.frisky.mail.dreamhost.com. [208.113.200.12], dsn=5.0.0, stat=Service unavailable
Nov 21 16:45:17 asterisk1 sendmail[3769]: nALEjCo5003767: forward /var/lib/asterisk/.forward.asterisk1: Group writable directory
Nov 21 16:45:17 asterisk1 sendmail[3769]: nALEjCo5003767: forward /var/lib/asterisk/.forward: Group writable directory
Nov 21 16:45:17 asterisk1 sendmail[3769]: nALEjCo5003767: nALEjHo5003769: DSN: Service unavailable
Nov 21 16:45:18 asterisk1 sendmail[3769]: nALEjHo5003769: to=<asterisk@asterisk1.local>, delay=00:00:01, xdelay=00:00:00, mailer=local, pri=246536, dsn=2.0.0, stat=Sent

why do I get stat=Service unavailable?
Can some1 please help me figure this out??!?
Thanx

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Zap gone after power outage

Jos — Sat, 21 Nov 2009 09:27:48 +0000

We have been running trixbox fine for years now. today after a power outage the outbound/inbpund calls (Zap) are gone.

Is exactly the same configuration files everything.

Even the card led is green, I have another PBX running the same (trixbox) and the only difference I found was that when doing a "cat" to /proc/zaptel/1
the one that works says (in use), while the other is just there.

The point is, how can I get my zap back the way it was before the power outage?

MONDO does not work after clean installation

mst — Sat, 21 Nov 2009 05:29:07 +0000

I have just installed MONDO on 2.6.2 TRIX and rebooted server.

Simply typed: mondoarchive and looks like mondoarchive: command not found

I have fallowed guide with installation script. Am I missing something?

MST

Logrotate bug

Underlord — Sat, 21 Nov 2009 02:21:37 +0000

I have an issue that is exactly the same as this one http://www.trixbox.org/forums/trixbox-forums/help/logrotate-not-r... though it occurs on 2.6.2.3.

Basically there seems to be two logrotate files for asterisk in /etc/lodrotate.d, asterisk and asterisk.logrotate, and the two of them conflict.

If I run logrotate manually I get an error:
error: asterisk.logrotate:1 duplicate log entry for /var/log/asterisk/event_log

This prevents the asterisk logs from being rotated. As a work around I've deleted /etc/logrotate.d/asterisk

Can anyone shed some light on this issue?

Caller ID

bnichols — Sat, 21 Nov 2009 01:22:42 +0000

I haven't been able to find this type of setup in any of the other posts. We aren't able to get the caller id to work from outside calls. We currently have a t1 going into Cisco 2811>>>siptrunk>>trixbox. We only have 2 of the 23 phone numbers going to the trixbox because the 2811 is also connected to a Call Manager server. I know that the Cisco 2811 can send caller id because it works just fine through the call manager setup. Once we get the caller id setup we will redirect everything to the trixbox and take out the call manager server. The caller id on the trixbox phones only shows the server ip 192168XXXXXX as the caller.

Does anyone have any ideas on where I should start?

Thanks

First time setup - HELP!

gregorywest — Sat, 21 Nov 2009 00:42:12 +0000

I am trying to get my TB to dial out. No matter what I do all I seem to get is the trixbox message saying "You call can not be completed as dialed....."

What am I missing

Device and User mode, with custom contexts.

rjefferis — Sat, 21 Nov 2009 00:04:52 +0000

Hi everyone, hope you are all well.

I have a query that has been bugging me for the last couple of days.

Basically i would like to be able to set up a system in device and user mode. This of course is easy. However i would like to disable any calls unless there is a user logged into the device. So far i have a custom context for the devices that only gives access to a couple of dial plans including log on and off.

The issue being, that when a user logs in to the device, the context stays the same and they can still not dial out. I assumed that when the user logged in, it would use the user context and clid but this is obviously not the case.

Does anyone have a solution/workaround for this.

Thanks

Rob

Trixbox Basics

hbrown8517 — Fri, 20 Nov 2009 22:01:38 +0000

Is there some service or documentation that can help me with a special project for my business? I will list the tasks that are on my agenda:

1. I need to configuration an IP phone that needs a name and extension changed, the phone is connected to my trixbox.

2. Is it possible to setup the Asterisk sever to allow received Fax to multiple email addresses, I have a designated fax number on the trixbox? Although, the fax option will be ideal if one email address can receive fax messages.

3. I would like to change default Voice Mail message instructions on the Trixbox, also. How would I go about doing this?

Can anyone give me some insight on the current projects?

Sincerely,

Private Business Owner

Inbound Caller ID

crbender — Fri, 20 Nov 2009 21:53:08 +0000

I have installed CallerID Superfecta recently. When I use the internal debug tool it works as advertised. When I get the same number actually calling into my system the CNAM is not being sent to the extension. The phone number comes through ok just not the name. I have been scouring the forums for two days trying to find an answer to this without luck. Any help would be greatly appreciated.
Thanks
Chuck

Backup Choices

MetalSmith — Fri, 20 Nov 2009 21:23:19 +0000

Hi all,

I use tbm-backup to run my backups and have used it to successfully restore a trixbox. What does the Complete SQL Dump, trixbox Dashboard, and Operator Panel back up?

I have a fully working trixbox only using System Recordings, Config Files, HudLite Config Files, Phone Provisioning.

So I am not sure what information is being backed up from using Complete SQL Dump, trixbox Dashboard, and Operator Panel?

Thanks.

Rotating ring group

bschmer — Fri, 20 Nov 2009 20:29:45 +0000

Good afternoon.

I have set up a ring group for our after-hours support folks so that if the first person doesn't answer, the second person gets called and so on. However the primary after-hours responsibility changes from week to week, so I'd like to be able to automatically change the order of the numbers listed in the ring group so that the person that the numbers get cycled to a 'fresh' person. Something like:

Week 1 order: 101 -> 102 -> 103
Week 2 order: 102 -> 103 -> 101
Week 3 order: 103 -> 101 -> 102
Week 4 order = Week 1 order
....

What's the easiest way to accomplish this via a script that I can run from cron?

Thanks,
Bryan

Outbound Calls fail with "All circuits are busy"

kevincurrey — Fri, 20 Nov 2009 18:16:00 +0000

Hi all.

I have a new Trixbox 2.6.2.3 setup running in a virtual machine w/ 1 GB ram. The box is registered to Callcentric and properly recieving incoming calls. Outbound calls fail with "All circuits are busy now" message. The message is being generated by Trixbox, NOT Callcentric.

SIP Trunk Settings (all others blank):
Peer Details:
username=177(number)
type=peer
secret=(password)
insecure=very
host=callcentric.com
fromuser=177(number)
fromdomain=callcentric.com
context=from-pstn
address=callcentric.com

Trunk Name: home

This is somewhat different from the settings listed on the callcentric support page, but follows the format given by datu503 in this post: http://www.trixbox.org/forums/trixbox-forums/sip-and-iax-trunks-a....

In the Dial Rules box I have tried ".", "XXXXXXXXXXX" and leaving it blank. (Callcentric requires 11 digits for all calls. I'll setup a dial plan after I get this working)

I have a single outgoing route defined:
Route Name: to-home
Dial Patterns: XXXXXXXXXXX (that's 11 digits)
Trunk Sequence: 0 - SIP/home

I should point out that I have a 3cx box on the same network that has been working fine, but I'm trying Trixbox for some features I can't get in 3cx.

As a side note, that "All ciruits are busy" message and any other computer generated messages play very slowly, and I'm not sure why since that system is showing plenty of available memory and hardly using any processor capacity.

Thanks for any help you can offer,
Kevin

Caller Id

uricohen — Fri, 20 Nov 2009 18:11:29 +0000

If i mak e a extension using CUSTOM Extension and i put my cell phone number in the dial i want it that when it calls my cell it will say the caller id of the caller not of my office.

thank you

Trixbox doesn't answer an incomming call.

stefan24 — Fri, 20 Nov 2009 18:07:13 +0000

Hello,

I have set up trixbox to the point that I can call internally also make out bound calls, but it won't pick up an incomming phone call.
I have set up inbound routes, but can't get it to work, could someone give me a hand along with this? I am not sure what files I should submit to help with the process, as I'm a newbie to all of this.

kind regards
Stefan

creating zap trunk with fusion100

woronti — Fri, 20 Nov 2009 18:05:59 +0000

i have an asterisk machine that has a four port TDM400P card. three ports on it are connected to my channel banks, how do i setup my sim adapter (psitek fusion 100) to talk to asterisk, by connecting to one of the channel banks. presently my calls are routed internally. thank you

Emergency message notification system - help how to?

boeingpilot — Fri, 20 Nov 2009 17:19:50 +0000

This will be a little long winded, so please bear with me.

I am in process of designing the new phone system for our church. They currently have a Toshiba system which works ok, but they want to go to the next level. Trixbox should do everything they want, but I am having a problem with handling their after hours emergency notification system. Perhaps someone can help.

Currently (on the Toshiba) if an after hours emergency occurs, the calling party selects an option from the IVR, and this puts them into a voicemail box for emergencies. After the message is left the system then calls the first clergy on duty, and informs them there is a message. If the phone call is not answered, it repeats this 3 times over 10 minutes time. To stop the system from progressing, the email must be listened to. If not, after the first cycle of 10 minutes, the system goes to the next notification person on the list and repeats the three attempts over 10 minutes. Again, if no one listens to the message, it continues to go down a notification list until someone calls the system back and checks messages. It move down the list to ensure that someone is notified, checks the message and acts on emergency. (They don't want message to hang overnight)

The reasoning behind this (and why follow-me is not used) is that occasionally you will get someone who will call the emergency box with a routine message. This keeps the clergy on duty from taking a non-emergency call at that time. (basically phone screening).

So... can anyone suggest a way that I could replicate this work flow on a trixbox, or suggest another solution that would give me similar functionality? (e.g. notification to a list in order over time, and message receipt acknowledgment or continue notifications)

If someone thinks they could code the functionality, I will put a bounty out for it.

Thanks

Scott

Which phones to buy for these requirements? Thoughts please..

efx456 — Fri, 20 Nov 2009 16:55:53 +0000

I need to buy about 20 phones for a small office. They have to support:

- Paging (press a button to say an announcement to a group of extensions: "John, call waiting for you on line 1" - John doesn't have a fixed extension, he will pick up from whatever desk he's sitting at)
- Parking a call with the press of a button (4-8 parking spots: we call them line 1, line 2, line 3, ...etc.)
- Have blinking lights for parking spots that have calls waiting in them
- The usual basic functions (transfer, hold, conference)

I hope to pay about $100-130 per phone (wholesale prices).

I've done some research, here are my findings:

- Cisco 7940/7960 phones are out (they don't support paging).
- Grandstream phones are out too? (not the greatest quality)
- Polycom's are out? too expensive, hard to set up
- Linksys SPA94x/5xx are a possibility, but I'm not sure if they have enough blinking lights and soft keys to support paging and parking.
- Snom 320? Seems like an option but may be a bit expensive. Would it do everything I need?
- Aastra? Which models would do what I need but are not too expensive?

Questions I'm asking myself: What should I look for exactly? How many BLF lights should I look for? How many soft keys?

I'd greatly appreciate any thoughts/suggestions/comments. Thanks in advance.

Echo cancellation/Volume control Guide - Digium TDM400

nicmac — Fri, 20 Nov 2009 16:16:05 +0000

I've seen some references to echo cancellation guides but I am unable to find them. I'm having an issue with echo and very low volume with my digium card. Can someone please point me to a guide?

Can VoIP and analog phones/trunks be used concurrently?

ah_clem — Fri, 20 Nov 2009 15:56:47 +0000

This may be a "dumb" question but I don't want to just assume...

Will Trixbox support the use of both analog trunks (via a Sangoma card) and SIP trunks on the same box? Also can both analog and VoIP phones/endpoints as extensions on the same system? If so are there any special configuration issues that are required?

A system I'm building right now has a Sangoma A200 with a FXO and a FXS module installed, and that seems to be working (sort of). I have an extension set up for the FXO, but I cannot seem to register a VoIP phone to a different extension. Before I beat my head on the monitor troubleshooting this I just want to make sure I'm not trying to do something impossible.

Thanks in advance...

7 Offices & new MPLS & internet failover

MillsapsPE — Fri, 20 Nov 2009 15:41:42 +0000

We have 7 offices that are already running TB setups. Each office currently has its own internet service. Each TB connects to each TB via IAX trunks.

We are looking at implementing AT&T MPLS service to the 7 offices. We are also looking at maintaining internet connections at Office A & Office B that will serve as internet access points for the whole MPLS network. Office A (T1) would be primary and Office B (cablemodem) would be the failover.

My question is related to how to handle the failover. My AT&T technical rep recommended Fatpipe IPVPN's in Office A&B. But, these are $$ ($6k+ each). What other options are available? Even if they are a little more "manual" in making the switch from Office A to Office B.

Thanks.

Fallow me missing in TIME CONDITION

mst — Fri, 20 Nov 2009 15:22:00 +0000

Fallow me is missing in TIME CONDITION on TRIXBOX 2.6.2.3

In ver 2.4 under TIME CONDITIONS we can see DESTINATION IF TIME MATHCES: FALLOW ME

IN 2.6.2 there is no FALLOW ME as an available option under DESTINATION IF TIME MATCHES.

Repeating DTMF problem.

ah_clem — Fri, 20 Nov 2009 15:20:25 +0000

Just started trying to implement a Sangoma A200 card configured with FX0 and FXS (one each) modules. My issue is that I am getting inconsistent and intermittent results when trying to make an outbound call. Sometimes it's "All circuits are busy" from the Trixbox, sometimes it's "This call cannot be completed as dialed, you must dial "1" and the area code." from A.T.&T (PTSN provider) and other times it works fine.

In the "Must dial 1" scenario when I view the full.log file I see the complete dial string appearing in the log, but the PTSN acts as though it's incomplete.

"Sent deffered digit string: T1(area code and number)w"

I'm new to the whole Sangoma implementation, so please cut me some slack if I'm missing something...

trixbox CE current release is 2.8.0
Asterisk 1.4.20-1 RPM by vc-rpms@voipconsulting.nl built by root @ trixbuild-2.5
Not sure where to access the Sangoma, Zaptel, and Wanpipe versioning, sorry will supply if you'll tell me how to determine...

Thanks in advance.

Need to upgrade Phython 2.3 to 2.4

djsullie — Fri, 20 Nov 2009 15:01:18 +0000

I have searched around the net and this forum I just dont see a way to upgrade to python 2.4 I guess the easiest way would be a yum update just not sure on the command. any ideas how to do this?

running free pbx 2.6
centos 4.8

thank you
Dan

International Dialing over its own trunk

bravonoj — Fri, 20 Nov 2009 14:00:28 +0000

hello all,

I have been searching and trying to figure something out. I have two providers on my trixbox server that I use. I use Vitelity (switched from Broadvoice...best thing I ever did) for NANPA numbers, and I have also a VoipJet account that I want to use for International calls. The trunks are set up and work fine for NANPA dialing, but when I tried to set up another outbound route to use for WORLD dialing, I kept seeing the calls try to go out over the Vitelity trunk instead of the Voipjet trunk.

I set up (deleted now, so I cant post configs) the outbound route with a password for International out VoipJet and issued amportal restart after, and it still tried to go out Vitelity.

Any ideas? Suggestions? Best Practices? Thanksgiving Recipies? I'm all ears.

thanks

How to change 5060 sip port in trixbox

jalokim — Fri, 20 Nov 2009 11:35:46 +0000

Im trying to change trixbox listen port but without any resoults.

Putting bindport to sip.conf gave no resoults.

Could you help me with this ?

Toll Free Directory Assistance Listings

command007 — Fri, 20 Nov 2009 11:15:09 +0000

Hi everyone.

I have been searching for a SIP trunking provider that offers the ability to list toll free numbers with Toll Free Directory Assistance (http://www.tollfreeda.com). This is the same thing as calling 1-800-555-1212. I've contacted a few directly such as Flowroute, Vitelity and Voip.ms and none of them offer the ability to list in this directory. The directory requires providers to list the numbers and you cannot register yourself.

I need this feature since I am going to operate a technical support call center and need to list each toll free number under the company name in which we are doing support for. I am sure a large provider would offer this, but the rates with those providers for low volume companies are not as good.

Does anyone have any ideas?

I appreciate the help!

- Guy

Enum Outbound Not working

vbhoj74 — Fri, 20 Nov 2009 09:33:34 +0000

I looked all over the forum for this particular issue but did not found anything specific so had to post this. I'm new to trixbox and am running the v 2.8.0.2. I've gone through the tutorial on setting up ENUM with FreePBX at http://asterisktutorials.com and also read the tribox2 without fears. I have successfully registered two of my PSTN lines with enum, so my SIP was working since they called back to verify.

My issue is here with outgoing calls, whenever I call a enum number, it tells me that the person is unavailable, the log though shows successful lookup but hangup the call later. I suspect maybe the enum number is incorrect, but whatever number I surfed gave me the same result, so can someone provide me with a valid working enum test number ?

-thanks in advance.
---------------------------------------------------------------------------------------------------

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
-- Executing [16416312405061@from-internal:1] Macro("SIP/2001-b7e918b0", "user-callerid,SKIPTTL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/2001-b7e918b0", "AMPUSER=2001") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/2001-b7e918b0", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/2001-b7e918b0", "1?Set(REALCALLERIDNUM=2001)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/2001-b7e918b0", "AMPUSER=2001") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/2001-b7e918b0", "AMPUSERCIDNAME=Vineet Bhojnagarwala") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/2001-b7e918b0", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/2001-b7e918b0", "AMPUSERCID=2001") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/2001-b7e918b0", "CALLERID(all)="Vineet Bhojnagarwala" <2001>") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/2001-b7e918b0", "REALCALLERIDNUM=2001") in new stack
-- Executing [s@macro-user-callerid:10] ExecIf("SIP/2001-b7e918b0", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:11] GotoIf("SIP/2001-b7e918b0", "1?continue") in new stack
-- Goto (macro-user-callerid,s,20)
-- Executing [s@macro-user-callerid:20] NoOp("SIP/2001-b7e918b0", "Using CallerID "Vineet Bhojnagarwala" <2001>") in new stack
-- Executing [16416312405061@from-internal:2] Set("SIP/2001-b7e918b0", "_NODEST=") in new stack
-- Executing [16416312405061@from-internal:3] Macro("SIP/2001-b7e918b0", "record-enable,2001,OUT,") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/2001-b7e918b0", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/2001-b7e918b0", "recordingcheck,20091120-145308,1258708988.91") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck,20091120-145308,1258708988.91: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] MacroExit("SIP/2001-b7e918b0", "") in new stack
-- Executing [16416312405061@from-internal:4] Macro("SIP/2001-b7e918b0", "dialout-enum,3,16312405061,,") in new stack
-- Executing [s@macro-dialout-enum:1] GosubIf("SIP/2001-b7e918b0", "0?sub-pincheck,s,1") in new stack
-- Executing [s@macro-dialout-enum:2] Macro("SIP/2001-b7e918b0", "outbound-callerid,3") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/2001-b7e918b0", "0?Set(CALLERPRES()=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/2001-b7e918b0", "0?Set(REALCALLERIDNUM=2001)") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/2001-b7e918b0", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/2001-b7e918b0", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/2001-b7e918b0", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/2001-b7e918b0", "TRUNKOUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/2001-b7e918b0", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/2001-b7e918b0", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/2001-b7e918b0", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/2001-b7e918b0", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
-- Executing [s@macro-dialout-enum:3] Set("SIP/2001-b7e918b0", "OUTBOUND_GROUP=OUT_3") in new stack
-- Executing [s@macro-dialout-enum:4] GotoIf("SIP/2001-b7e918b0", "1?nomax") in new stack
-- Goto (macro-dialout-enum,s,6)
-- Executing [s@macro-dialout-enum:6] Set("SIP/2001-b7e918b0", "DIAL_NUMBER=16312405061") in new stack
-- Executing [s@macro-dialout-enum:7] Set("SIP/2001-b7e918b0", "DIAL_TRUNK=3") in new stack
-- Executing [s@macro-dialout-enum:8] ExecIf("SIP/2001-b7e918b0", "1?AGI(fixlocalprefix)") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
-- AGI Script fixlocalprefix completed, returning 0
-- Executing [s@macro-dialout-enum:9] AGI("SIP/2001-b7e918b0", "enumlookup.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/enumlookup.agi
-- enumlookup.agi: Looking up 16312405061 on e164.org via dns_get_record
-- enumlookup.agi: Setting DIALARR to sip/16312405061@voip1.kayfamily.net|
-- AGI Script enumlookup.agi completed, returning 0
-- Executing [s@macro-dialout-enum:10] GotoIf("SIP/2001-b7e918b0", "0?end") in new stack
-- Executing [s@macro-dialout-enum:11] Set("SIP/2001-b7e918b0", "TRYDIAL=sip/16312405061@voip1.kayfamily.net|") in new stack
-- Executing [s@macro-dialout-enum:12] Set("SIP/2001-b7e918b0", "DIALARR=") in new stack
-- Executing [s@macro-dialout-enum:13] Dial("SIP/2001-b7e918b0", "sip/16312405061@voip1.kayfamily.net|,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-enum:14] NoOp("SIP/2001-b7e918b0", "Dial exited in macro-enum-dialout with CHANUNAVAIL") in new stack
-- Executing [s@macro-dialout-enum:15] GotoIf("SIP/2001-b7e918b0", "1?dialloop") in new stack
-- Goto (macro-dialout-enum,s,10)
-- Executing [s@macro-dialout-enum:10] GotoIf("SIP/2001-b7e918b0", "1?end") in new stack
-- Goto (macro-dialout-enum,s,18)
-- Executing [s@macro-dialout-enum:18] NoOp("SIP/2001-b7e918b0", "Exiting macro-dialout-enum") in new stack
-- Executing [16416312405061@from-internal:5] Macro("SIP/2001-b7e918b0", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Playback("SIP/2001-b7e918b0", "all-circuits-busy-now,noanswer") in new stack
-- Playing 'all-circuits-busy-now.ulaw' (language 'en')
-- Executing [s@macro-outisbusy:2] Playback("SIP/2001-b7e918b0", "pls-try-call-later,noanswer") in new stack
-- Playing 'pls-try-call-later.ulaw' (language 'en')
-- Executing [s@macro-outisbusy:3] Macro("SIP/2001-b7e918b0", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/2001-b7e918b0", "vw") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/2001-b7e918b0", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/2001-b7e918b0", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/2001-b7e918b0", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/2001-b7e918b0", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/2001-b7e918b0", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/2001-b7e918b0' in macro 'hangupcall'
== Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/2001-b7e918b0' in macro 'outisbusy'
== Spawn extension (from-internal, 16416312405061, 5) exited non-zero on 'SIP/2001-b7e918b0'
-- Executing [h@from-internal:1] Macro("SIP/2001-b7e918b0", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/2001-b7e918b0", "vw") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/2001-b7e918b0", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/2001-b7e918b0", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/2001-b7e918b0", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/2001-b7e918b0", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/2001-b7e918b0", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/2001-b7e918b0' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/2001-b7e918b0'
-- ast_get_srv: SRV lookup for '_sip._UDP.sip.flowroute.com' mapped to host sip.flowroute.com, port 5060