[asterisk-users] asterisk linkedin group
asterisk linkedin group I have created an asterisk linkedin group for anyone interested. http://www.linkedin.com/e/gis/45252/66270A773F53 Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems HIROTEC AMERICA Please visit us on the web at www.hirotecamerica.com HIROTEC AMERICA Ph. 248-836-5100 Fx. 248-836-5101 Please only print this email if it is necessary. Help spread environmental awareness. This e-mail and any files transmitted with it are intended only for the person or entity to which it is addressed and may contain confidential material and/or material protected by law. Any retransmission or use of this information may be a violation of that law. If you received this in error, please contact the sender and delete the material. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] FYI about my Mona Vie business venture
Asterisk work does not pay all of my bills, so I have joined up with a company that has a very good payment plan. I have recently become a Mona Vie Independent Distributor. I am not going to go into a sales pitch. This is just an FYI to this opportunity. The company has grown into a Billion dollar company in just 2 years. This company's compensation plan is the best and quickest that I have seen. My brother-in-law has only been in the business for a month and is already making a profit. The first thing that I noticed when researching the opportunity, was that I could find no negative statements about it. The product itself has many health benefits. So far: My knees no longer ache. We are both sleeping better. I literally do not stir once asleep. My restless leg syndrome has not been noticed. I seem to have more energy. The main ingredient is the acai berry. Here is a list of what it is supposed to do: Boosts energy levels Improves digestive function Improves mental clarity/focus Promotes sound sleep Provides all vital vitamins Contains several important minerals Is an extremely powerful free radical fighter Acai has very high levels of fibers Cleanses and Detoxifies the body of infectious toxins Strengthens your immune system Enhances sexual desire and performance Fights cancerous cells Slows down the aging process Promotes healthier and younger-looking skin Alleviates diabetes Normalizes and regulates cholesterol levels Helps maintain healthy heart function Minimizes inflammation Improves circulation Prevents artherosclerosis Enhances visual acuity The income can be made in two ways (actually more, but two primary ways) 1. Reselling the product at a marked up price. This is something that I have no interest in, and do not personally know anyone doing this. 2. Team Commissions. a. You make back 5 percent of the sales that occur below you in your tree. b. You only have to personally sign 2 people. Other people above you will be adding to your tree. c. They call it a binary system, where you only have 2 people directly under you, and any other people that you add go down to the bottom and benefit others as well as yourself. d. I already have two people underneath me and have not personally signed anyone yet, so it is a quick growing tree, even for people that may not be as motivated. e. After a month, My brother-in-law has NO more out of pocket expenses to stay in this system. The money he is earning is paying for his Minimum requirements. The rest is profit. To sign up to be a distributor , which is required to make money, is $54 A case of Mona Vie is $120. A case will last 2 people a month. (you only take 2 ounces a day) This may seem like a lot, but: 1. You will not need to buy any vitamins. 2. My brother-in-law is already making $200 a month, after being in the system for a month, So his cost for the Mona Vie is covered and he is making $80 a month. 3. As more people sign up, the amount he gets back will increase. Anyway, I am not intending this to be pushy or salesy, I just wanted to let my associates, that may be looking for additional income, know about this. Here is the Website, if you are interested in researching this: http://teamvie.blogspot.com/ http://www.monavie.com Also, feel free to Google it. I am very excited with this, both in the health benefits I am already seeing, and the income potential. Please feel free to let me know if this is something that you may be interested in, and I can get you more information. Thank You, Steven B [EMAIL PROTECTED] Please visit us on the web at www.hirotecamerica.com HIROTEC AMERICA Ph. 248-836-5100 Fx. 248-836-5101 Please only print this email if it is necessary. Help spread environmental awareness. This e-mail and any files transmitted with it are intended only for the person or entity to which it is addressed and may contain confidential material and/or material protected by law. Any retransmission or use of this information may be a violation of that law. If you received this in error, please contact the sender and delete the material. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FYI about my Mona Vie business venture - apology and rethink
I feel the need to apologize for my previous email to the list. I was thinking that I was sharing something that I am currently exited with, with my associates. I now realize that I was sooo... off-topic, it's ridiculous. Sorry for the improper post to the list. Feel free to keep any further comments, concerning my improper use of the list, off-list. Thank You, Steven B [EMAIL PROTECTED] http://teamvie.blogspot.com/ Please visit us on the web at www.hirotecamerica.com HIROTEC AMERICA Ph. 248-836-5100 Fx. 248-836-5101 Please only print this email if it is necessary. Help spread environmental awareness. This e-mail and any files transmitted with it are intended only for the person or entity to which it is addressed and may contain confidential material and/or material protected by law. Any retransmission or use of this information may be a violation of that law. If you received this in error, please contact the sender and delete the material. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] best way for night ringer??
Asterisk 1.2.13 I am trying to figure out the best way for a night bell at work. Note: I have no spare buttons available on the phones. But I do have two lines and two park positions as buttons. Option 1 (easiest and the one I just implemented) When asterisk is in night mode, Connect to IVR, List all options and then if they dial 0 or timeout, ring every phone in the building. Option 2 (This would require user training) When asterisk is in night mode, List all options and then if they dial 0 or timeout, ring an analog ringer and have a pickup code to grab the line. I am not sure that I can even do directed pickup on asterisk 1.2.13 Option 3 (I believe this is best, but am not sure where to start) When asterisk is in night mode, List all options and then if they dial 0 or timeout, park the call, then overhead page a record stating that there is a call that needs to be picked up on line 5401. They already have a button (with BLF) for 5401. How are others handling night calls when there is no receptionist available. Thank You, Steven BerkHolz Board member of Connectech Greater Detroit www.connectech.org Please visit us on the web at www.hirotecamerica.com HIROTEC AMERICA Ph. 248-836-5100 Fx. 248-836-5101 This e-mail and any files transmitted with it are intended only for the person or entity to which it is addressed and may contain confidential material and/or material protected by law. Any retransmission or use of this information may be a violation of that law. If you received this in error, please contact the sender and delete the material. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk linkedin group
asterisk linkedin group I have created an asterisk linkedin group for anyone interested. http://www.linkedin.com/e/gis/45252/66270A773F53 Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems HIROTEC AMERICA Board member of Connectech Greater Detroit www.connectech.org Please visit us on the web at www.hirotecamerica.com HIROTEC AMERICA Ph. 248-836-5100 Fx. 248-836-5101 This e-mail and any files transmitted with it are intended only for the person or entity to which it is addressed and may contain confidential material and/or material protected by law. Any retransmission or use of this information may be a violation of that law. If you received this in error, please contact the sender and delete the material. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Split asterisk in two ?? One TDM and One IP only??
I have built an asterisk server with a TE412P card on a Dell 2950. It does incoming calls via DID over PRI, our IVR, SIP/IAX extensions, Fax/Analog extensions via an old PBX via PRI, voicemail, etc. My issue now is that I find it difficult to test/upgrade to new versions. This is what I am thinking of doing. Server1 Keep one physical server just for TDM functions. PRI to Telco PRI to old PBX for Fax. (basically using it as a mux) Keep meetme here for Digium card timing. Server2 Build a new asterisk install within Xen VM with data stored on an iSCSI SAN. This would be all IP. IAX and SIP extensions. IAX and SIP providers. IVR Voicemail Web access to voicemail CDR This way I can test different versions of the features of Server2 (clone with different IP) without affecting production. I assume that I just use an IAX or SIP trunk between the two asterisk servers. Does this make sense? Are others doing similar? Are there any other features that require the TDM card besides PRI, Fax and Meetme? I have heard of people using Xen for IP only asterisk, but are there any known gotchas? Thanks, Thank You, Steven BerkHolz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Announce option for meetme - is it used?
Announce option for meetme - is it used? It makes a caller record their name, but I do not see where this name recording is ever used. Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems HIROTEC AMERICA Fax. 248-836-5101 www.hirotecamerica.com Board member of www.glimasoutheast.org Our company name has changed to HIROTEC AMERICA www.hirotecamerica.com Please update any contact info with my new email address [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] meetme list (unmonitored)?
What does unmonitored mean in the below reference? Ref: CLI meetme list User #: 01 5665 zzz Channel: SIP/5665-9f8038a0 (unmonitored) User #: 02 5664 no nameChannel: SIP/5664-0096b660 (unmonitored) 2 users in that conference. Also, is there a way to see durations via meetme CLI? Thank You, Steven Our company name has changed to HIROTEC AMERICA www.hirotecamerica.com Please update any contact info with my new email address [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vonage SIP access via asterisk?
Does anyone have a working connection to Vonage via asterisk? (SIP, not ATA) I just signed up to test their service and they sent me a Number, Proxy, port and password. Every reference I have tried leaves me with a 404 error coming from Vonage. If you have a working setup, please post some config references. Thank You, Steven BerkHolz Soon to be known as HIROTEC AMERICA www.hirotecamerica.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] metermaid and 1.2.13?
It is unclear to me if the metermaid patch should be in 1.2.13 or not. Please advise. Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems TESCO Group Companies Fax. 248-836-5101 www.TESCOGroup.com Board member of www.glimasoutheast.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VM notification to pager and phone
I looked for a reference to do this for some time to replace the callout
feature in my old AVT voicemail.
I never found one, so I decided to dig in.
Here is my first run. It is in production, so unless I find a problem,
I am done.
Script set to run every 5 min. via cron.
This sets a lock file to prevent 2 scripts from running.
Check for a VM in our Emergency after hours support mailbox.
If found, it sends a numeric page to our rotating pager.
If no one has listened to the mail in 7 minutes, it calls a cell phone.
On this call, it connects directly to a prompt, then VoicemailMain with
the ext. already included.
If no one has listened to the mail in 7 minutes, it calls a second cell
phone.
On this call, it connects directly to a prompt, then VoicemailMain with
the ext. already included.
If no one has listened to the mail in 7 minutes, it calls the rotating
pager again.
This continues to loop until the VM is listened to.
isnotify.sh:
--
LOCKFILE=/tmp/5134outdial.lock
MESSAGEFILE=/var/spool/asterisk/voicemail/default/5134/INBOX/msg.txt
CALLFILE1=/tmp/5134outdial1.call
CALLFILE2=/tmp/5134outdial2.call
CALLFILE3=/tmp/5134outdial3.call
CALLUSER=asterisk
OUTGOING=/var/spool/asterisk/outgoing/
date
# echo lock file check
[ -f $LOCKFILE ] echo $LOCKFILE exists exit 0
touch $LOCKFILE
function recip1 {
if [ -f $MESSAGEFILE ]
then
echo $MESSAGEFILE exists!
echo calling IS pager
echo Channel: ZAP/g0/1XXX892 $CALLFILE1
echo MaxRetries: 2 $CALLFILE1
echo RetryTime: 60 $CALLFILE1
echo WaitTime: 30 $CALLFILE1
echo Context: ext-local $CALLFILE1
echo Extension: 5681 $CALLFILE1
echo Priority: 1 $CALLFILE1
echo CallerID: IT VoiceMail XX5682 $CALLFILE1
chown $CALLUSER:$CALLUSER $CALLFILE1
chmod 664 $CALLFILE1
echo move
echo moving $CALLFILE1 to $OUTGOING
mv $CALLFILE1 $OUTGOING
else echo No MV
rm -f $LOCKFILE
exit
fi
sleep 10m
recip2
}
function recip2 {
if [ -f $MESSAGEFILE ]
then
echo $MESSAGEFILE exists!
echo calling BerkHolz
echo Channel: ZAP/g0/1XXX083 $CALLFILE2
echo MaxRetries: 2 $CALLFILE2
echo RetryTime: 60 $CALLFILE2
echo WaitTime: 30 $CALLFILE2
echo Context: ext-local $CALLFILE2
echo Extension: 5682 $CALLFILE2
echo Priority: 1 $CALLFILE2
echo CallerID: IT VoiceMail XX5682 $CALLFILE2
chown $CALLUSER:$CALLUSER $CALLFILE2
chmod 664 $CALLFILE2
echo moving $CALLFILE2 to $OUTGOING
mv $CALLFILE2 $OUTGOING
else echo No MV
rm -f $LOCKFILE
exit
fi
sleep 10m
recip3
}
function recip3 {
if [ -f $MESSAGEFILE ]
then
echo $MESSAGEFILE exists!
echo calling Gibson
echo Channel: ZAP/g0/1XXX061 $CALLFILE3
echo MaxRetries: 2 $CALLFILE3
echo RetryTime: 60 $CALLFILE3
echo WaitTime: 30 $CALLFILE3
echo Context: ext-local $CALLFILE3
echo Extension: 5682 $CALLFILE3
echo Priority: 1 $CALLFILE3
echo CallerID: IT VoiceMail XX5682 $CALLFILE3
chown $CALLUSER:$CALLUSER $CALLFILE3
chmod 664 $CALLFILE3
echo moving $CALLFILE3 to $OUTGOING
mv $CALLFILE3 $OUTGOING
else echo No MV
rm -f $LOCKFILE
exit
fi
sleep 10m
recip1
}
recip1
rm -f $LOCKFILE
--
Dial Plan:
--
exten = 5681,1,Answer
exten = 5681,n,Wait(3)
exten = 5681,n,SendDTMF(X)
exten = 5681,n,SendDTMF(X)
exten = 5681,n,SendDTMF(X)
exten = 5681,n,SendDTMF(X)
exten = 5681,n,SendDTMF(X)
exten = 5681,n,SendDTMF(X)
exten = 5681,n,SendDTMF(X)
exten = 5681,n,SendDTMF(X)
exten = 5681,n,SendDTMF(X)
exten = 5681,n,SendDTMF(X)
exten = 5681,n,SendDTMF(#)
exten = 5681,n,Macro(hangupcall)
exten = 5682,1,Answer
exten = 5682,n,Wait(1)
exten = 5682,n,Macro(user-callerid)
exten = 5682,n,Playback(it-services)
exten = 5682,n,Macro(get-vmcontext,5134)
exten = 5682,n,VoiceMailMain([EMAIL PROTECTED])
exten = 5682,n,Macro(hangupcall)
--
Thank You,
Steven BerkHolz
- MCSA - MCSE -
Board member of
www.glimasoutheast.org
___
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[asterisk-users] dial D option with w for wait?
From WIKI: D(digits): After the called party answers, send digits as a DTMF stream, then connect the call to the originating channel. (You can also use 'w' to produce .5 second pauses.) When I use the D option to send a call to my paging system and pick a zone, the Tone is too early. I have tried the 'w' option, but it does not appear to work. No matter how many 'w's I use, the tone is still immediately on answer. Is this a known issue? Is there a work around? My current workaround is to send both channel to a meetme that runs a macro to play the tone. This is way to much overhead to play a single tone after .5 or 1 seconds. Please advise. Thank You, Steven BerkHolz Board member of www.glimasoutheast.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] some transfers dropped.
We are having an issue with transferred calls being dropped. Looking at the asterisk 1.2.10 logs, it appears that when it is dropped, the SIP unit send a CANCEL message to the server. On successful transfers this is not seen. The errors logged in the SIP Unit error log, I believe are from the second attempt to transfer the call, after it has actually been disconnected. Nothing is deferent in the logs above the CANCEL request for successful or failed transfers. So, I am not sure why the CANCEL is being sent. I can not discern what may be different when it fails. Thank You, Steven BerkHolz Board member of www.glimasoutheast.org ref: from SIP Phone (I think these are the second invite after it is hung up) 2006-OCT-20 17:49:52 GMT +++ Current Timestamp +++ 2006-OCT-20 17:19:47 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-20 15:56:37 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-20 15:50:00 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-20 15:45:38 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-20 15:11:28 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-20 15:10:58 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-20 14:59:26 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-20 12:45:30 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 19:53:25 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 18:40:52 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 18:03:45 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 17:55:55 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 15:09:13 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 15:04:33 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 14:52:12 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 14:34:35 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 14:20:17 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER 2006-OCT-19 13:45:33 GMT NLPA ERROR: sipFailureResponseToRefer: received 603 response to REFER ref. from asterisk 1.2.10 logs: Oct 20 13:19:45 VERBOSE[10652] logger.c: -- Requested transfer capability: 0x00 - SPEECH Oct 20 13:19:45 DEBUG[8159] channel.c: Avoiding initial deadlock for 'Zap/25-1' Oct 20 13:19:45 VERBOSE[10652] logger.c: -- Called g2/5155 Oct 20 13:19:45 VERBOSE[10652] logger.c: Transmitting (no NAT) to 172.16.8.200:5065: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.8.200:5065;branch=z9hG4bKline0-2425957956;received=172.16.8.200 From: From Desksip:[EMAIL PROTECTED];tag=2425948795 To: sip:[EMAIL PROTECTED];tag=as279eb184 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Oct 20 13:19:45 DEBUG[10658] app_queue.c: Device 'Zap/25' changed to state '2' (In use) but we don't care because they're not a member of any queue. Oct 20 13:19:45 DEBUG[8159] devicestate.c: Changing state for Zap/25 - state 2 (In use) Oct 20 13:19:45 DEBUG[10659] app_queue.c: Device 'Zap/25' changed to state '2' (In use) but we don't care because they're not a member of any queue. Oct 20 13:19:45 DEBUG[8167] chan_zap.c: Enabled echo cancellation on channel 25 Oct 20 13:19:45 VERBOSE[10652] logger.c: -- Zap/25-1 is ringing Oct 20 13:19:45 DEBUG[8159] devicestate.c: Changing state for Zap/25 - state 6 (Ringing) Oct 20 13:19:45 DEBUG[10660] app_queue.c: Device 'Zap/25' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. Oct 20 13:19:45 DEBUG[8171] chan_sip.c: Header 0: (0) Oct 20 13:19:46 VERBOSE[8171] logger.c: -- SIP read from 172.16.8.200:5065: CANCEL sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 172.16.8.200:5065;branch=z9hG4bKline0-2425957956 To: sip:[EMAIL PROTECTED] From: From Desksip:[EMAIL PROTECTED];tag=2425948795 Call-Id: [EMAIL PROTECTED] Max-Forwards: 70 CSeq: 2 CANCEL Content-Length: 0 Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 0: CANCEL sip:[EMAIL PROTECTED] SIP/2.0 (36) Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.16.8.200:5065;branch=z9hG4bKline0-2425957956 (65) Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 2: To: sip:[EMAIL PROTECTED] (27) Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 3: From: From Desksip:[EMAIL PROTECTED];tag=2425948795 (55) Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 4: Call-Id: [EMAIL PROTECTED] (43) Oct 20 13:19:46 DEBUG[8171] chan_sip.c: Header 5: Max-Forwards: 70 (16) Oct 20 13:19:46
[asterisk-users] Inhouse SIP to ZAP has echo sometimes.
Sometimes we get echo heard on SIP phone when dialing out. Zap channel is on aTE411 Card. It is using a PRI to XO. As far as I know echo is created on the far side. Could the Zaptel card be the far side as far as the SIP phone is concerned? Calls from our soon to retire legacy PBX do not have this problem. Those calls are Legacy - PRI - asterisk -PRI - XO - Dest. Any suggestions? Zaptel.conf: context=from-pstnswitchtype=nationalpridialplan=unknown prilocaldialplan=unknownpriindication=inbandsignalling=pri_cpeusecallerid=yeshidecallerid=no usecallingpres=yesechocancel=yesechocancelwhenbridged=yesechotraining=yesgroup=0callgroup=1pickupgroup=1useincomingcalleridonzaptransfer=yescallerid=asreceivedaccountcode=Imusiconhold=defaultoverlapdial=nofacilityenable=yesnsf=nonechannel = 1-23 I also set "static int vpmdtmfsupport = 0;" in wct4xxp.c to remove sporotic DTMF tones. Thank You, Steven BerkHolz- MCSA - MCSE -Manager of Information SystemsTESCO Group CompaniesFax. 248-836-5101www.TESCOGroup.com Board member ofwww.glimasoutheast.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Odd echo issue with speaker phone
I assume that this is from the echo canceller, but I am not sure. A call is started via SIP speakerphone. When the handset is picked up, there is a slight echo of your own voice after you speak.(duh, is there any other kind of echo) If the call is made without the speaker phone, there is no echo. I am stumped by this one. Thank You, Board member ofwww.glimasoutheast.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Odd echo issue with speaker phone
I assume that this is from the echo canceller, but I am not sure. A call is started via SIP speakerphone. When the handset is picked up, there is a slight echo of your own voice after you speak.(duh, is there any other kind of echo) If the call is made without the speaker phone, there is no echo. I am stumped by this one. Thank You, Board member ofwww.glimasoutheast.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hint status from dialplan?
Can I get the hint status from the dialplan? I am intending to add lit buttons for the parking slots. currently I am using 1.2.11 with 1 parking button and several pickup buttons (speed dials to the parking slots) since 1.4 allows park() to specify a parking slot, I figure that I can remove the park button and just have several buttons for the slots. plan: button assigned to a virtual extension (we will call it 2001) the hint for 2001 will point to parking slot 701. There fore the button will be lit if a call is parked. If the button pressed will call 2001, check the hint status of 701 and either park(701) or ParkedCall(701) depending on the status of the slot. So, Can I get the hint status from the dialplan? Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems TESCO Group Companies Fax. 248-836-5101 www.TESCOGroup.com Board member of www.glimasoutheast.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting QOS settings in asterisk and/or CentOS?
How would I go about setting the TOS bit to "RTP IP TOS Byte: 18 (hex)" for SIP and IAX traffic at the asterisk server? Also, Do you have a quick reference on how to configure a Cisco switch to prioritize SIP traffic? I check in various Cisco docs, and there are so many references, and none of them seem to relate directly to using the TOS bit for QOS. I am looking into using the TOS bit because that is the only method that my SIP devices use. (Citel Handset Gateway) ref: QOS settings from Citel Handset Gateway: Handset Gateway - QoS Configuration IP Type of Service RTP IP TOS Byte: 18 (hex) Silence Suppression Mute Mode: On, UDP keep-alive every 10 secondsG.711 Voice Activity Detection: Off Codec Preferences G.711u: 1 (Highest priority) G.711a: 2 Thank You, Steven BerkHolz- MCSA - MCSE -Manager of Information SystemsTESCO Group CompaniesFax. 248-836-5101www.TESCOGroup.com Board member ofwww.glimasoutheast.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] All circuits are busy now???
"All circuits are busy now" makes perfect sense in my PRI trunk is full. How do I stop asterisk from playing this recording when it is a wrong/bad number? I gat a call today that a user was trying "all day" to call a number in Mexico and kept getting the above recording. I said, try in on your cell phone, and they received a "this number is not is service". I would like to either hear the far recording (I think I will get billed for this), or internally play a different message. I think the issue is that I am using a PRI and am receive the cause code that is triggering the above recording. Can asterisk play a different message for this? and only play the above message if "MY" circuit is busy? Thank You, Steven BerkHolz- MCSA - MCSE -Manager of Information SystemsTESCO Group CompaniesFax. 248-836-5101www.TESCOGroup.com Board member ofwww.glimasoutheast.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iax vs. sip?
I have no NAT issues. My PBX is multihomed and the outside IP is locked down for all except IAX and SIP ports. With the current version of asterisk, which transport is better right now? I am looking at 6-10 simultaneous calls over a half T1. I am not asking about codecs here, I am asking about SIP vs. IAX if the provider does either. (we are looking at testing Teliax next) I have seen posts about jitter in IAX, so I am not sure if SIPmight bebetter to use right now. Also, since IAX uses the same port for all of the calls, the call separation has to be done higher in the OSI stack. I do not know if this is better or worse or neither. Thank You, Steven BerkHolz- MCSA - MCSE -Manager of Information SystemsTESCO Group CompaniesFax. 248-836-5101www.TESCOGroup.com Board member ofwww.glimasoutheast.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Iaxy and SendDTMF??
I have an Iaxy that I am using to access our overhead paging system.
It is ext 5480 and required a 1 (office), 2 (Shop), or 3 (all) DTMF tone
after it answers.
If I dial 5480, I hear a tone to let me know that it is ready for the
digit.
I made an extension 5481 that using a macro and sendDTMF to send the
digit.
; exten = 5481,1,Answer()
exten = 5481,1,NoOp(${TIMESTAMP} paging Group 1 Office Page)
exten = 5481,2,DIAL(IAX2/5480,,M(MYDTMF1))
; exten = 5481,2,DIAL(IAX2/5480,,D(w1))
[macro-MYDTMF1]
; exten = s,1,SIPDtmfMode(inband)
exten = s,1,Wait(2)
exten = s,2,SendDTMF(1)
The problem is that I never hear any tone from the Paging Unit.
It pages, but there appears to be a 3-4 second delay before the audio
comes out of the speakers.
If I Dial 5480 and then the 1, the audio is immediate.
I did a test to my SIP phone first and I could hear the digit if I set
the SIPDtmfMode(inband).
But I am trying to do this with an Iaxy.
Worst case, I could buy an ATA, but I wouild like to get this working
with the iaxy if possible.
Please advise.
Thank You,
Steven BerkHolz
- MCSA - MCSE -
Manager of Information Systems
TESCO Group Companies
Fax. 248-836-5101
www.TESCOGroup.com
boardmember of
www.glimasoutheast.org
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[asterisk-users] nat and qualify questions
Are there any problems with always having nat=yes and qualify=yes? We just opened up our server to be accessible to SIP from the internet. (used to require VPN) I had to set the SIP setting for my test softphone to nat=yes and qualify=yes. This makes sense. Some of these phone will never leave our building. Some of these phone will come and go. (laptops) Is the any negatives to just have all phones set to nat=yes and qualify=yes? If not, why is it not the default? Thank You, Steven BerkHolz- MCSA - MCSE -Manager of Information SystemsTESCO Group CompaniesFax. 248-836-5101www.TESCOGroup.com Board member ofwww.glimasoutheast.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IVR with LDAP query for phone number and mobile number??
We still have a lot of users on a legacy PBX, so the Directory app is not sufficient. We also have users with mobile phones. Has anyone made an LDAP lookup that will pull this info from MS Active Directory? My thinking is to add this function to my main IVR. As long as my AD is accurate, it should contain all my info. Replace the current directory app in my main IVR with this function. When someone presses 9 in my IVR, do a lookup to AD with all of the possible number combinations. ex. 222 is aaa aab aac aba abb abc, etc. I am not sure how I would do the desk phone vs. mobile phone number options. Submit the accepted number to the dialplan so that 4 local digit extensions dial local, 4 digit legacy extensions dial out that PRI and mobiles dial out to PSTN or GSM gateway. Anyway, I haven't fully thought it through, but I figured I would ask if anyone else had done this yet. -- -- Steven http://www.glimasoutheast.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] metermaid patch
Olle, Will the metermaid patch help this issue?: http://bugs.digium.com/view.php?id=7435 I believe that the fix is in res_features.c , but I do not want to pursue it if it is already there. Also, thanks for your hard work on that patch. Our receptionist will really like the PARKINGEXTEN feature. Do you know if this patch will end up in 1.4? -- -- Steven http://www.glimasoutheast.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SE Michigan asterisk users group
I am thinking of getting an asterisk user group together for either SE Michigan or just Metro-Detroit. How much interest in asterisk in Michigan is there on this list? I am already on the board of glimasoutheast, with is a group for technology professionals. (very broad range) It is a spin-off from Automation Alley, which is SE Michigan's version of Silicone Valley. -- Steven http://www.glimasoutheast.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] me, voip.trxtel.com and early media
I was testing using trxtel for outbound toll free because I have an issue on my PRI where it will not handle early media. (IVRs that play as a ringback tone) There was a bug that was supposed to fix this Q4 of 2005, but I never saw any relief for it. voip.trxtel.com has the same issue, so at least I know that I am not alone. The number I am testing is 18663402763. On my cell phone, hitting any digit stops the recording. On my asterisk system, over PRI or trxtel, none of my DTMF goes through. If I wait for the timeout, the call gets answered and I can talk to an operator. I am using the latest stable of Libri, Zaptel and asterisk. If anyone has got a solution to this, please advise. If not, I suppose I should open a new bugs for this. -- Steven http://www.glimasoutheast.org where business and technology meet ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Backup Question?
This may be slightly off topic. I am using FreePBX, and using it's backup feature. Here is the question part: I would like to copy my backup off the asterisk server. From your experiences, which approach seems more resilient to failure: Push the backup from asterisk to another server using STFP or FTP? Pull the backups from asterisk from another server using SFTP or other? The destinationwould most likely be going to a Windows Server. I also have the option of installing Veritas Netbackup Client on the asterisk server, but I assume that this would not be good for the PBX's performance. steveb ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] calleridname.agi patch to only overwrite name if it is missing
I edited the calleridname.agi patch to only overwrite the name if it is
missing.
The asteridex option still overwrites the name since it is our master
list for known numbers.
--
Steven
calleridname.agi.patch:
--- C:\Documents and Settings\steveb\Desktop\calleridname.agi-orig Tue
Jun 13 14:37:09 2006
+++ C:\Documents and Settings\steveb\Desktop\calleridname.agi Tue Jun 13
14:37:09 2006
@@ -16,6 +16,7 @@
my $callerid = $input{'calleridnum'};
my $calleridfull = $input{'callerid'};
+my $calleridname = $input{'calleridname'};
if($callerid eq ''){
$callerid=$input{'callerid'};
@@ -31,7 +32,8 @@
$calleridfull =~ s/[\,\\']+/ /g;
-$AGI-verbose(CALLERID IS: $calleridfull\n);
+$AGI-verbose(CALLERID IS: $calleridfull\n);
+$AGI-verbose(CALLERID Name IS: $calleridname\n);
if ($callerid =~ /^(\d{3})(\d{3})(\d{4})$/) {
$npa = $1;
@@ -54,7 +56,7 @@
#$nxx='892';
#$station='8019';
-if ($Fonetastic '0') {
+if (($Fonetastic '0') ($calleridname != 'unknown')){
$AGI-verbose(Ready for Fonetastic.US lookup... \n);
if ($name = fonetastic_lookup ($npa, $nxx)) {
$newcallerid = \$name $npa$nxx$station\;
@@ -68,7 +70,7 @@
$AGI-verbose(Fonetastic.US lookup disabled.);
}
-if ($AnyWho '0') {
+if (($AnyWho '0') ($calleridname != 'unknown')){
$AGI-verbose(Ready for AnyWho lookup... \n);
if ($name = anywho_lookup ($npa, $nxx, $station)) {
$newcallerid = \$name $npa$nxx$station\;
@@ -82,7 +84,7 @@
$AGI-verbose(AnyWho lookup disabled.);
}
-if ($Google '0') {
+if (($Google '0') ($calleridname != 'unknown')){
$AGI-verbose(Ready for Google lookup... \n);
if ($name = google_lookup ($npa, $nxx, $station)) {
$newcallerid = \$name $npa$nxx$station\;
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[Asterisk-Users] Cell gateway for T-Mobile US??
Most gateways I have found are only sold overseas. Do these work in the US? My provider is T-Mobile (using their Blackberries). They support: GSM (I am pretty sure) GPRS EDGE We get unlimited Cell to Cell minutes and would like to leverage the possible savings. Does anyone know of a product that they have been happy with? SIP or Analog is fine although SIP (or IAX) is preferred for the asterisk side. Thanks. Steven Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems TESCO Group Companies Fax. 248-836-5101 www.TESCOGroup.com Board member of www.glimasoutheast.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: CallerID name inbound from PRI
XO fixed my caller ID name.
I am using FreePBX and I can include a wait to my custom extensions.
Is there a way to add a wait to the whole PRI?
I assume that if I set immediate to yes, I can then have a s extension
do the wait, but how would it get from the s to the DID extension?
(also, I would rather not answer every call)
Is there a magic spot in Free PBX's configs to add the wait for all
calls on that PRI, or do I need to alter the FreePBX code to add it when
creating the conf. Files?
Thank You,
Steven BerkHolz
- MCSA - MCSE -
Manager of Information Systems
TESCO Group Companies
Fax. 248-836-5101
www.TESCOGroup.com
Board member of
www.glimasoutheast.org
Steven [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Thanks for the info.
I went to add the Wait(2), but am unsure where to do it.
My context is from-pstn.
My [from-pstn] is:
[from-pstn]
exten = s,1,NoOp(${TIMESTAMP} PRI call in) ;I tried adding this
to see if s is used, but lothing was logged.
include = from-pstn-custom ; create this context
in extensions_custom.conf to include customizations
include = ext-did
exten = fax,1,Goto(ext-fax,in_fax,1)
My from-pstn-custom is non-existent and my ext-did is just an
include for ext-local, which is my inside extensions.
If I understand you correctly, I need the wait before I pick up the
line.
If I change the span to immediate=yes, I can use the s extension,
but It would also answer the line early.
I am drawing a blank where to put the wait.
Please advise.
--
--
Steven
http://www.glimasoutheast.org
Alexander Lopez [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
There is nothing you really need 'to do' if your PRI is working
already,
If you are able to receive and make calls your D-Channel is
functioning
properly. In the case of CallerID, some telcos provide this extra
function via the FACILITY messages instead of the SETUP messages, If
that is the case, you will get no Name but you will get a number. IT
simply means that Asterisk answered the call with the SETUP message
but
was unable to read in the CALLERID Name to pass on to your devices
because it comes later on in the call via the FACILITY.
Add a Wait(2) before you answer the call for your PRI, see if that
helps.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Sent: Monday, April 10, 2006 8:57 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] callerid name inboune from PRI
I switched PRI vendors recently, and one of my questions was
do you provide caller ID name in addition to number?
ATT Local did not, But XO communications said they did.
Before I call to complain, is there an setting to turn this
on in asterisk?
I want to make sure that I have my side covered before I call XO.
My current zaptel.conf is:
context=from-pstn
switchtype=national
pridialplan=unknown
prilocaldialplan=unknown
priindication = outofband
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
usecallingpres=yes
echocancel=yes
echocancelwhenbridged=no
group=0
callgroup=1
pickupgroup=1
accountcode=I
musiconhold=default
channel = 1-23
--
--
Steven
http://www.glimasoutheast.org
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[Asterisk-Users] revisit to legacy PBX and CID over PRI
My legacy PBX accepts CID number, but not name. My old PRI vendor never sent the name, so there was never an issue. I have wedged asterisk between the Legacy PBX and PSTN. PSTN - PRI - asterisk - PRI - Legacy. Any calls from asterisk (sip and iax extensions) which have callerID set, will not connect. The legacy PBX hangs up, but asterisk thinks that it is still ringing. I have added hidecallerid=yes to zapata.conf and no longer have the problem. But, now I have Legacy PBX users complaining about having no caller ID. I tried this, but it still would not complete the call. hidecallerid=no ;fix for no answer restrictcid=yes usecallingpres=no I have also tried to make the callerid name null, but asterisk still tries to send the data. I have also tried to dumb it down from NI2 to NI1, but asterisk still tries to send the callerID name in the PRI debug. Is there a way to send callerid number and not the name? ref. zapata.conf: context=panasonic swichtype=national pridialplan=unknown prilocaldialplan=unknown signalling=pri_net usecallerid=yes facilityenable=no hidecallerid=yes ;fix for no answer restrictcid=yes usecallingpres=no echocancel=no echocancelwhenbridged=no group=2 channel = 25-47 Steven ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
