Re: [asterisk-users] Multi-tenant with receptionist features for managed service
For MT check out Thirdlane's MT PBX: http://www.thirdlane.com/products/thirdlane-pbx-mte I use the PBX Manager which it's based on and it works very well. ___ -- 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
Re: [asterisk-users] DTMF does not work
On Mon, Dec 29, 2008 at 1:55 PM, Brent Vrieze bvri...@cimsoftware.comwrote: I got no resonses to this and some funny bounces so I'm trying again. First of all Merry Christmas. Second, my first problem with my provider not staying registered with our server was my fault. We moved our server room and I restarted the test system and the production system causing them to ping-pong back and forth registering with our provider causing random problems, they are both set to register with the same account right now. I shut Asterisk down on the one and now we don't drop any longer. doh!!! Last, We are having DTMF problems with our provider (via:talk). Does anyone have any experience with them and if so can you share it? via:talk does have a sample sip.conf and extensions.conf file to use but the dial plan they set up does not require any DTMF so they may never have tested it. We have tried inband, auto, rfc2833 for our DTMF and nothing works. I have submitted a ticket with them but the last time I did that they never responded so that is why I am posting here. I signed up with another SIP provider for a test account and the DTMF passes no problem from them so I must conclude there is some setting that via:talk has that is causing the problem. via:talk will not confirm this but they must be using Asterisk as all the menus and such they have feel very Asteriskish. Is there something I can tell via:talk to try on their end to make this work? As a side symptem every time our system registers with via:talk it seams to jump from server to server on their end. They must have some sort of load balancing going on that is causing that. In the past we could get the DTMF to pass when we were on the initial server we registered with but when we got pushed to another server the DTMF would fail till I did a sip reload or restarted Astersk. Now we get no DTMF ever. System set up. Asterisk 1.4.22 Asterisk GUI 2.0 users.conf [trunk_1] context = DID_trunk_1 host = galvatron.vtnoc.net username = user name secret = password trunkname = via:talk - galvatron ; GUI metadata hasiax = no registeriax = no hassip = yes registersip = yes trunkstyle = voip hasexten = no fromuser = user name authuser = user name insecure = port,invite dtmf = rfc2833 dtmfmode = rfc2833 relaxdtmf = yes rfc2833compensate = yes port = 5060 canreinvite = no fromdomain = galvatron.vtnoc.net disallow = all allow = ulaw,gsm If you need to see more of the setup info I can provide. Thanks Brent I have the same problems with Viatalk. The problem is with their new servers. You are pointed to galvatron.vtnoc.net which is one of those. I currently have mine working by using their old servers. Try calling support, changing your account to rfc2833 if you haven't already and then point to chicago-1e.vtnoc.net with your same settings . You will have DTMF working, but I am not sure when the old servers are going away. Good Luck, Sean ___ -- 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
Re: [asterisk-users] Convert CallerID name to uppercase
In TRUNK versions of Asterisk, there is a function called TOUPPER, which converts strings to upper case. I don't know when, exactly, it appeared but I expect if it's not in the version you're using it may be portable backwards without too much difficulty if the version you're using supports functions. JT *CLI core show function TOUPPER This looks like exactly what I need. I see that it's available in 1.6 so I will upgrade and let you know how it goes. Thank You. -Sean Dennis ___ -- 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] Convert CallerID name to uppercase
Our legacy PBX will not accept the callerID name in anything but capital letters. (Harris 20-20) When I send a call to the legacy PBX from asterisk I would like to have asterisk convert the callerID name to uppercase letters. Is there a way to do this? Thanks for any input. -Sean ___ -- 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
Re: [asterisk-users] Unable to run make menuselect for asterisk-addons
Jonn R Taylor wrote: Hi all, I am unable to run make menuselect for asterisk-addons. Works fine for zaptel and asterisk. Here is the output. Jonn [EMAIL PROTECTED] asterisk-addons]# make menuselect CC=gcc CXX=g++ LD= AR= RANLIB= CFLAGS= make -C menuselect CONFIGURE_SILENT=--silent makeopts \make[1]: Entering directory `/usr/src/asterisk-addons/menuselect' Package gtk+-2.0 was not found in the pkg-config search path. Perhaps you should add the directory containing `gtk+-2.0.pc' to the PKG_CONFIG_PATH environment variable No package 'gtk+-2.0' found make[1]: Leaving directory `/usr/src/asterisk-addons/menuselect' make[1]: Entering directory `/usr/src/asterisk-addons/menuselect' make[1]: `makeopts' is up to date. make[1]: Leaving directory `/usr/src/asterisk-addons/menuselect' CC=gcc CXX=g++ LD= AR= RANLIB= CFLAGS= make -C menuselect CONFIGURE_SILENT=--silent make[1]: Entering directory `/usr/src/asterisk-addons/menuselect' gcc -g -c -D_GNU_SOURCE -Wall -c -o menuselect.o menuselect.c gcc -g -c -D_GNU_SOURCE -Wall -c -o strcompat.o strcompat.c gcc -g -c -D_GNU_SOURCE -Wall-c -o menuselect_curses.o menuselect_curses.c make[2]: Entering directory `/usr/src/asterisk-addons/menuselect/mxml' gcc -O -Wall -c mxml-attr.c gcc -O -Wall -c mxml-entity.c gcc -O -Wall -c mxml-file.c gcc -O -Wall -c mxml-index.c gcc -O -Wall -c mxml-node.c gcc -O -Wall -c mxml-search.c gcc -O -Wall -c mxml-set.c gcc -O -Wall -c mxml-private.c gcc -O -Wall -c mxml-string.c /bin/rm -f libmxml.a /usr/bin/ar crvs libmxml.a mxml-attr.o mxml-entity.o mxml-file.o mxml-index.o mxml-node.o mxml-search.o mxml-set.o mxml-private.o mxml-string.o a - mxml-attr.o a - mxml-entity.o a - mxml-file.o a - mxml-index.o a - mxml-node.o a - mxml-search.o a - mxml-set.o a - mxml-private.o a - mxml-string.o ranlib libmxml.a make[2]: Leaving directory `/usr/src/asterisk-addons/menuselect/mxml' gcc -o cmenuselect menuselect.o strcompat.o menuselect_curses.o mxml/libmxml.a -lncurses gcc -g -c -D_GNU_SOURCE -Wall -c -o menuselect_stub.o menuselect_stub.c gcc -o menuselect menuselect.o strcompat.o menuselect_stub.o mxml/libmxml.a make[1]: Leaving directory `/usr/src/asterisk-addons/menuselect' Generating input for menuselect ... ** *** Install ncurses to use the menu interface! *** ** menuselect changes NOT saved! [EMAIL PROTECTED] asterisk-addons]# rpm -qa | grep ncurses ncurses-5.5-24.20060715 ncurses-devel-5.5-24.20060715 [EMAIL PROTECTED] asterisk-addons]# rpm -qa | grep gtk gtk2-2.10.4-20.el5 [EMAIL PROTECTED] asterisk-addons]# ___ I had the same problem with asterisk 1.4.18. I switched to 1.4.21 and it worked great. ___ -- 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
Re: [asterisk-users] BLF functionality
I find it amazing how often I find myself stuck on a problem and then someone else posts a question about it to the list. I am in the same boat with the OP (although I never thought to test incoming calls until I read his message). If I call a phone it will show busy, however if I make a call from that phone it still shows as idle. I've set call-limit and limitonpeers and restarted asterisk but still no joy. What am I missing? I'm running 1.4.21.2 Relevant sip.conf: [lan-soundpointip](!) type=friend host=dynamic disallow=all allow=ulaw dtmfmode=rfc2833 qualify=no call-limit=10 limitonpeers=yes [3900](lan-soundpointip) username=3900 secret=sdjghdfkjhgdf context=phone-operator callerid=Operator 3900 [3917](lan-soundpointip) username=3917 secret=dfkghdjfhdkfd context=phone-isdept callerid=Dave Fullerton 3917 mailbox=3117 In my general section of my sip.conf I have: allowsubscribe=yes notifyringing=yes limitonpeer=yes notifyhold=yes and it works both ways. ___ -- 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
Re: [asterisk-users] Asterisk end-user GUI?
Check out www.thirdlane.com they have a excellent end user portal. Ken D'Ambrosio wrote: I badly want to roll out Asterisk at my job. Unfortunately, my boss is dazzled by shiny objects. We had a vendor in today who showed us their system which, honestly, didn't suck -- but boy, is it going to be expensive! One major component of the eye candy was an end-user interface that allowed the user to initiate calls to a contact list, check for presence, create conferences, etc. Is there anything like that, aimed at end-users (as opposed to admins) for Asterisk? I'd even be willing to go with proprietary; I just don't want a wholly-proprietary, hobbled, licensed-to-Heck-and-back system, which is where it looks like my boss is leaning. Thanks! -Ken ___ -- 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 ___ -- 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
Re: [asterisk-users] Toll Free International Number
Try www.tollfreeforwarding.com, they do just that. Larry Costigan wrote: Hello All, I am looking to find a way to provide international toll free access to our Knoxville, TN (USA) office from our customers in the UK and in Australia, and when I talked with ATT I was surprised to find out how expensive they are... Surely, other businesses are not paying this much - are they?!?! Can someone in this good group please help me with some advice as to who can provide affordable and reliable international toll free service for a better price than ATT? Thanks in advance, Larry Costigan Food Donation Connection (Asterisk fan and ABE user) ___ -- 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 ___ -- 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
Re: [asterisk-users] Problem with cisco 7970G [EMAIL PROTECTED]
Jorge Munoz wrote: Hi everyone This is the first time I post something here so I’m sorry about my English, I don’t know how to write properly. Well, I’ve been working with Cisco 7960 telephones and my boss bought new ones , 7970G with SIP70.8-2-2SR3S firmware version, those work perfectly, but one of them has the SIP70.8.3.5S version, and this one doesn’t connect to the server , I wanted to install the SIP70.8.2.2SR3S version, but I couldn’t, is there anyone who knows how to do it? Many thanks. When I updated to SIP70.8.3.5S on my 7970 I had to change natEnabled1/natEnabled to natEnabled/natEnabled in the XML file to make the phone register. I believe it is a bug in the new firmware. ___ -- 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
Re: [asterisk-users] Call manager using Asterisk as voicemail server (SIP) not working ...
Steve Hickel wrote:
I have sip set up on Callmanager 4.x. When others call my ext of 2016 on
ccm after a busy or no answer, asterisk voice mail answers by saying,
Mailbox password. I want it to put them into my mailbox so they
can leave a message. Somehow I must be missing something... Please
help!
I have spent 19 hours easy on trying to figure this one out.
SIP DN is on CCM
VOICEMAIL on Asterisk is .
Here is my sip.conf:
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
allowexternaldomains=yes
allowexternalinvites=no
allowguest=yes
allowsubscribe=no
allowtransfer=yes
alwaysauthreject=no
autodomain=no
callevents=no
compactheaders=no
dumphistory=no
g726nonstandard=no
ignoreregexpire=no
jbenable=no
jbforce=no
jblog=no
maxcallbitrate=384
maxexpiry=3600
minexpiry=60
nat=no
notifyringing=no
pedantic=no
promiscredir=no
recordhistory=no
relaxdtmf=no
rtcachefriends=no
rtsavesysname=no
rtupdate=no
sendrpid=yes
sipdebug=no
t1min=100
t38pt_udptl=no
[authentication]
[sip]
type=friend
context=incoming
host=172.20.1.57
ipaddr=172.20.1.57
allow=ulaw
allow=alaw
nat=no
canreinvite=yes
qualify=yes
Here is my voicemail.conf
[zonemessages]
eastern=America/New_York|'vm-received' Q 'digits/at' IMp
central=America/Chicago|'vm-received' Q 'digits/at' IMp
central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours'
military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
european=Europe/Copenhagen|'vm-received' a d b 'digits/at' HM
[other]
[general]
format=wav49|gsm|wav
serveremail=asterisk
attach=yes
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
emaildateformat=%A, %B %d, %Y at %r
sendvoicemail=yes
attachfmt=wav
deletevoicemail=no
envelope=no
maxgreet=60
maxmessage=120
maxmsg=100
minmessage=1
operator=yes
review=yes
saycid=no
sayduration=yes
mailcmd=/usr/sbin/sendmail -t
externotify=/var/libasterisk/scripts/vm.sh
[default]
2016=1234,Steve,[EMAIL PROTECTED]
Here is the relevant parts of my extensions.conf:
[macro-dialout-callmanager]
exten=s,1,ChanIsAvail(SIP/sip)
exten=s,2,Cut(AVAILCHAN=AVAILCHAN,,1)
exten=s,3,Dial(${AVAILCHAN}/${ARG1})
exten=s,4,Hangup
exten=s,102,Congestion
[incoming]
exten=,1,GotoIf($[${RDNIS}]?2:400)
exten=,2,MailboxExists([EMAIL PROTECTED]
exten=,3,Congestion
exten=,103,Voicemail(su${RDNIS})
exten=,104,Playback(vm-goodbye)
exten=,105,Hangup
exten=,400,VoicemailMain
[general]
static=yes
writeprotect=no
clearglobalvars=no
autofallthrough=yes
priorityjumping=no
[default]
exten=_230,1,SetCallerID(${EXTEN:3})
exten=_230,2,Dial(SIP/[EMAIL PROTECTED])
exten=_230,3,Answer
exten=_230,4,Wait,1
exten=_230,5,Hangup
exten=_231,1,SetCallerID(${EXTEN:3})
exten=_231,2,Dial(SIP/[EMAIL PROTECTED])
exten=_231,3,Answer
exten=_231,4,Wait,1
exten=_231,5,Hangup
I am using users.conf, but don't know how that ties in or whether I even
need it...???
thanks,
Steve
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You didn't mention what version of asterisk, but if you are using
version 1.4.x, in extensions.conf you need to use:
CALLERID(rdnis) instead of just RDNIS
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Re: [asterisk-users] Switch recommendation?
Hilary Miller wrote: This will be my first major asterisk experiment and I'm trying to choose a PoE switch for 15-24 phones. I was going to spend $400 on this: http://www.newegg.com/product/product.asp?item=N82E16833124053 but then I see this on ebay: http://cgi.ebay.com/WS-C3524-PWR-XL-EN-Cisco-3524-24-FE-Switch-W-PoE-VoIP_W0QQitemZ370043264927QQihZ024QQcategoryZ51268QQssPageNameZWDVWQQrdZ1QQcmdZViewItem and I'm thinking, hey, thats a lot cheaper and it is PoE. Will the Cisco IP phone's proprietary wizardry be a problem for my flock on Linksys IP phones? Because as long as it can do vlan qos and poe I think I can scrape by for half the price, right? Thanks for reading! The Cisco 3524 switch doesn't support 802.3af which is what your Linksys phones are going to want. If you have just Cisco phones this would work. To have 802.3af you have to have at least a Cisco 3560 series switch. See: http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a00801189b5.shtml#powerover for reference ___ -- 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
Re: [asterisk-users] SPA-962+ SPA-932- blf function
John Meksavan wrote: Asterisk Users, I am running Asterisk 1.4.11 on Debian Etch system with the TDM03B wildcard. I recently purchased a SPA-962 and SPA-932- the sidecar for our receptionist. After reading many forum postings on how to configure the side car, I uprgraded the SPA-962 software to 5.1.18(SC) version. I got the sidecar to subscribed to an extension on the Asterisk server, but the LED state on the SPA-932 never changes even when I am a call with that extension on another VOIP phone- SPA-941. I got the speed dial function to work, but the blf function does not appear to work. Did anybody get the blf function to work? What I am doing wrong? Any input would be greatly appreciated. Thanks in advance. Regards, John How well do you know your celebrity gossip? Talk celebrity smackdowns here. http://originals.msn.com/thebigdebate?ocid=T002MSN03N0707A ___ -- 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 To make it work properly I had to add the following to sip.conf: allowsubscribe=yes notifyringing=yes limitonpeer=yes notifyhold=yes See if that helps. -Sean ___ -- 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
Re: [asterisk-users] IAXy device
bilal ghayyad wrote: Hi All; I have been chocked just when I saw some posts talking about how much the IAXy is bad :) - So I would like to ask, did any one try it later and wether it is good or not? I am asking this because I need to use it as it is NAT Transparent (as I read also, and I did not try it to see how much it is transparent). What about codec? Why it is only support g711 and does not support compressed codec? And what about the IP address and the DNS usage and the DDNS usage? What main porblems contain and any advise? Regards Bilal The device has no echo cancellation and sounds horrible (lots of echo) on about half of the analog phones I tried it on. I wouldn't recommend it unless you absolutely need IAX. It's also very expensive for a 1 port ATA. ___ -- 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
Re: [asterisk-users] Cisco 79xx users/consultants, 7970G color in particular share information
Sigma Networks wrote: I would like to get in contact with users/consultants who are or have worked with the Cisco phones and Asterisk to trade information. Cisco has reluctantly made SIP available on their phones and most of the information on voip-info and other wiki's appears to be reverse engineered. There is a wealth of information out there which is terrific. I have a client with about 40 phones composed of 7970, 7960 and 7906 phones. I've upgraded all of these to SIP 8-3-3SR2S and the basic functions are working. My current questions are: 1. How to remotely reboot 7970s. I have both web access and SSH access to the phones. The instructions I have for SSH are to use (1) user/pass (or whatever is in the confg) and then (2) debug/debug. Surprisingly reset is not a valid command to restart the phone. There doesn't appear to be a reset on the web page, maybe there's a hidden URL? 2. BusyLampField? Thanks in advance. ___ -- 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 We have about 200 79x1's running SIP w/ asterisk and we are very pleased despite some of the non-standard things Cisco does. In answer to question 1 the only way we have found to reboot the phone remotely is shutdown the port on the POE switch. This will drop the PC's network as well if it is plugged into the phone. Question 2 I would like to know the answer to myself. I would be curious to know if it works with the SIP image in call manager. ___ -- 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
Re: [asterisk-users] OT: Is Cisco 7960 SIP firmware same as 7940 SIP firmware?
Mike Dent wrote: Hi, just wondered if it was the same firmware on both devices? thanks Mike ___ --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 Yes ___ --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
Re: [asterisk-users] What web GUI are people happy with?
Anciso, Roy wrote: Just wondering what web GUI people like for asterisk. I installed asterisk from source and I was looking at possibly installing web GUI for system management. So far freepbx.org looks promising anybody else have any suggestions. Thanks **Roy Anciso** Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] ___ --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 I recommend Thirdlanes PBX manager. We have several installations of it and it seems to work very well. The best thing about it is the end user portal. I believe there is a demo at thirdlane.com ___ --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] Echo Problems with IAXy
From what I have found the IAXy doesn't handle echo very well. About half of the analog phones I try on the adapter create an echo on the far end. The person I am talking to can hear themselves. I am using Asterisk 1.4 and have tried it with 1.2 as well with the same results. Is there is anything I can do in Asterisk to help solve the echo problem? Thanks, Sean Dennis ___ --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
