[asterisk-users] T1, FoneBRIDGE, and dropped D-Channel
I hope someone can help me out with this issue. It has been dogging me for months and I can't seem to get it to go away. I have a Rhino Ceros box running Asterisk 1.4.21.2 connected via eth0 (nVidia MCP61 Ethernet) to a RedFone FoneBRIDGE2 dual-port with EC. The FB is the latest hardware rev and the latest firmware. I'm running the latest fonulator version and I'm running Zap-1.4.11 sourced from RedFone. Nothing else is on eth0. It is currently connected thru a dedicated switch to the FB and the secondary server although I've observed this problem when connected directly. There are two other eth cards, one for the internal network and one for the DMZ. My problem is that every now and then the D-Channel will drop which will terminate all calls in process. The D-channel will immediately come back up (usually within a second) but that doesn't do any good because the calls are gone by then and users are mad. The log entry at one of these events looks like this: [Feb 3 08:14:02] ERROR[26063] chan_zap.c: Write to 65 failed: Unknown error 500 [Feb 3 08:14:02] ERROR[26063] chan_zap.c: Short write: 0/15 (Unknown error 500) [Feb 3 08:14:02] WARNING[26063] chan_zap.c: Detected alarm on channel 1: Yellow Alarm (same message for other 22 channels) [Feb 3 08:14:02] NOTICE[2660] chan_zap.c: PRI got event: Alarm (4) on Primary D-channel of span 1 [Feb 3 08:14:02] WARNING[2660] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [Feb 3 08:14:02] NOTICE[2662] chan_zap.c: Alarm cleared on channel 1 (same message for other 22 channels) [Feb 3 08:14:02] NOTICE[2660] chan_zap.c: PRI got event: No more alarm (5) on Primary D-channel of span 1 [Feb 3 08:14:02] NOTICE[2660] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Feb 3 08:14:02] NOTICE[2660] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 You'll notice the timestamps are all within the same 1-second interval which makes me think it is literally missing one packet and causing the drop. I'm sure the configs are fine as they've been reviewed by about 20 people and the system works most of the time. If the machine has been freshly started up, this happens about once every other day. The machine has currently been running for over 36 days and I'm seeing several per day now. ATT has run a stress test on the line from the CO to the smartjack and found no problems. The cable from the smartjack to the FoneBRIDGE is about 18 and I've tried a couple with no difference. I'm convinced this is interrupt related. When I initially commissioned this machine, the FB was connected to eth2 and I couldn't get it to link up with the CO at all. The D-Channel was flapping like crazy. I switched it to eth0 and it worked. You can see from my interrupts that the on-board and the add-in cards are clearly on different busses. CPU0 0: 3266196236IO-APIC-edge timer 1: 2IO-APIC-edge i8042 8: 3IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 169: 207230361 IO-APIC-level ohci_hcd:usb1 177:5080313 IO-APIC-level sata_nv 185: 0 IO-APIC-level sata_nv 193:1632824 IO-APIC-level eth1 201: 39823124 IO-APIC-level eth2 225: 2565938694 PCI-MSI eth0 NMI: 0 LOC: 3266207768 ERR: 1 MIS: 0 So the fact that I couldn't link up when I was on one card and I could when I am on another (with no config changes... other than re-directing ztdynamic) leads me directly to this interrupt issue. Can anyone shed some light here? Has someone seen this before? If so, how did you solve it? Thanks! Jason -- This e-mail message, including any attachments, is only for the use of the intended recipient (s). The information contained may be confidential, in which case its disclosure or reproduction is strictly prohibited. If you are not the intended recipient, please return it immediately to its sender at the above address and delete it. ___ -- 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] Authorize Microsoft SQL
I'm doing something similar to validate employees for DISA access. I
built Asterisk with ODBC support by installing unixODBC and FreeTDS
before I built Asterisk. I have a couple of stored procedures on the MS
SQL box that do the heavy lifting and hide the database details from the
Asterisk system. Really, the backend could be any ODBC compliant
datasource that supports stored procs. (I use the stored procedure to
expose a consistent interface regardless of the database schema behind
it)
Here is the relevant portion of my dialplan: (You can also see I use
ODBC to push CDR records back to the database for logging purposes)
exten = s,1,NoOp() ; Validate the employee's id number. Give
them MAX_ID_TRIES to get it right.
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Set(ID_TRIES=0) ; Set the max number of login attempts
exten = s,n,Set(MAX_ID_TRIES=3)
exten = s,n(get_id),NoOp()
exten = s,n,Set(ID_TRIES=$[${ID_TRIES} + 1])
exten =
s,n,Read(ID_ENTERED,/var/lib/asterisk/sounds/custom/disa_greet1,5)
exten = s,n,Set(ID_RESULT=${ODBC_INFO(ClockID,${ID_ENTERED})})
exten = s,n,GotoIf($[${ISNULL(${ID_RESULT})}]?:valid_id,1)
exten = s,n,Playback(/var/lib/asterisk/sounds/custom/disa_badempnum)
exten = s,n,GotoIf($[${ID_TRIES}
${MAX_ID_TRIES}]?get_id:login_fail,1)
exten = valid_id,1,NoOp() ; Validate the employee's pin number. Give
them MAX_PIN_TRIES to get it right.
exten = valid_id,n,Set(PIN_TRIES=0) ; Set the max number of login
attempts
exten = valid_id,n,Set(MAX_PIN_TRIES=3)
exten = valid_id,n(get_pin),NoOp()
exten = valid_id,n,Set(PIN_TRIES=$[${PIN_TRIES} + 1])
exten =
valid_id,n,Read(PIN_ENTERED,/var/lib/asterisk/sounds/custom/disa_greet2,
4)
exten =
valid_id,n,Set(PIN_RESULT=${ODBC_PIN(ClockID,${ID_ENTERED},${PIN_ENTERED
})})
exten = valid_id,n,GotoIf($[${ISNULL(${PIN_RESULT})}]?:valid_login,1)
exten =
valid_id,n,Playback(/var/lib/asterisk/sounds/custom/disa_badpincode)
exten = valid_id,n,GotoIf($[${PIN_TRIES}
${MAX_PIN_TRIES}]?get_pin:login_fail,1)
exten = login_fail,1,NoOp() ; They suck. They couldn't get either the
pin number or the emp id right.
exten =
login_fail,n,Playback(/var/lib/asterisk/sounds/custom/disa_faillogin)
exten = login_fail,n,Hangup()
exten = valid_login,1,NoOp()
exten = valid_login,n,Set(CALLDATE=${STRFTIME(${EPOCH},GMT+5,%x %X)})
exten = valid_login,n,Set(CLID=${CALLERID(num)})
exten = valid_login,n,Set(UNID=${CDR(uniqueid)})
exten = valid_login,n,Set(DBINS =
${ODBC_DISA(${CALLDATE},${CLID},${ID_ENTERED},${UNID})})
exten =
valid_login,n,Playback(/var/lib/asterisk/sounds/custom/disa_greet3)
exten = valid_login,n,DISA(no-password,from-disa,CID Name
xx)
exten = valid_login,n(end),Goto(valid_login,s,1)
With unixODBC you need a couple of config files...
Here is my /etc/odbc.ini:
[OHSQL_ELABOR]
Driver = FreeTDS
Description = Connection to eLabor database on OHSQL - LIVE
Trace = No
Server = ohsql.ohio..xxx
Database= eLabor
Port= 1870
TDS_Version = 8.0
ReadOnly= Yes
[OHSQL_ASTERISK]
Driver = FreeTDS
Description = Connection to Asterisk Database
Trace = No
Server = ohsql.ohio.x.xxx
Database= Asterisk
Port= 1870
TDS_Version = 8.0
Here is my /etc/odbcinst.ini:
(The FileUsage=1 is important when working against MS SQL... the driver
doesn't support multiple connections)
[FreeTDS]
Description = FreeTDS Driver (MS-SQL access)
Driver = /usr/local/freetds/lib/libtdsodbc.so
Setup = /usr/local/freetds/lib/libtdsS.so
FileUsage = 1
Here is /etc/asterisk/func_odbc.conf
; We define two DSNs for database function access:
; - eLaborSQL which provides access the eLabor database
;(Could be testing or live... depends on res_odbc.conf)
; - AsteriskSQL which provides access to the Asterisk database
[INFO]
; This is a general grab statement to allow us to access any column in
the employee table
; by clock ID
dsn=eLaborSQL
read=SELECT ${ARG1} FROM Employee WHERE ClockID = ${ARG2} and Terminated
= 0
[PIN]
; This will return a given column based on the clock ID PIN passed in
dsn=eLaborSQL
read=SELECT ${ARG1} FROM Employee WHERE ClockID = ${ARG2} and PIN =
${ARG3} and Terminated = 0
[DISA]
;This will insert a new record into the DISA database to allow for cdr
match-ups
dsn=AsteriskSQL
read=INSERT INTO Asterisk_DISA (calldate, src, empID, uniqueid) VALUES
('${ARG1}','${ARG2}','${ARG3}','${ARG4}')
And finally... here is /etc/asterisk/res_odbc.conf
[eLaborSQL]
enabled = yes
dsn = OHSQL_ELABOR
pooling = yes
limit = 1
username = x
password = xx
pre-connect = yes
; Many databases have a default of '\' to escape special characters. MS
SQL
; Server does not.
backslash_is_escape = no
[AsteriskSQL]
enabled = yes
dsn = OHSQL_ASTERISK
pooling = yes
limit = 1
username =
Re: [asterisk-users] anoyingly answers already in use pstn line
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jack Bates Sent: Friday, October 17, 2008 4:48 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] anoyingly answers already in use pstn line I am using Asterisk and an X101P card as a glorified answering machine. We have a residential PSTN line with about six phones connected to it. Like an answering machine, I want Asterisk answer the line *only* when an incoming call is not answered after four rings. This mostly works. My extensions.conf is at the end of this message. The problem is that Asterisk will sometimes answer the line when someone is already talking on one of the six phones connected to it. Sometimes Asterisk will answer the line and start playing the greeting in the middle of a conversation! This is especially a problem when I am talking on the phone to an automated system, because although I hang up the phone I am talking on, neither the automated system nor Asterisk will hang up. I have not yet discovered a pattern to when Asterisk answers the line. It always answers after four rings, but it sometimes answers when someone is already talking on one of the phones connected to the line. In a perfect world, Asterisk would be the only thing connected to the line, and all our phones would be Asterisk extensions. Unfortunately we do not currently have the required VoIP phones or FXS interface... Is there any way to make Asterisk less flaky, and answer the line *only* when an incoming call is not answered after four rings? --- [default] exten = s,1,Wait(20) exten = s,n,Answer exten = s,n,Background(recordings/coop-greeting) exten = s,n(instruct),Background(recordings/leave-message) exten = s,n,Background(recordings/enter-extension) exten = s,n,Background(recordings/dial-by-name) exten = s,n,Background(recordings/visit-website) exten = s,n,WaitExten ; General delivery mailbox exten = #,1,Voicemail(6000) exten = #,n,Goto(s,instruct) ; Dial by name exten = a,1,Directory(default) ; Entering an invalid extension replays the instructions exten = i,1,Playback(invalid) exten = i,n,Goto(s,instruct) ; Timeout goes to voicemail exten = t,1,Goto(#,1) exten = 6003,1,Macro(stdexten,6003,SIP/cstewart) exten = 6004,1,Macro(stdexten,6004,SIP/mhockley) exten = 6005,1,Macro(stdexten,6005,SIP/jbates) [...] ___ -- 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 Others may wish to chime in and confirm or deny this but the card is probably getting confused by you loading the line with the other phones. I know most of the analog cards I've worked with (which does not include the X101P) really get cranky if there is anything else hanging off that line. The only solution I've seen to the problem is to change things around so that the card is the only thing on the line. In know you said you haven't switched to IP or FXS but is there a reason why? Your problem would go away and you would be able to leverage all the features of Asterisk if you just got a single ATA. Something like a Linksys PAP2T-NA can be had for around $55 USD. Disconnect your PSTN line at the entrance bridge, run it into the X101P, and plug the PAP2T into the house. It is convenient and doesn't require any changes in internal wiring. (You might have to run a few wires if the bridge is on the back of your house.) No need for new phones or anything. Granted, all the internal phones would be on one extension but you have that situation now... And with the ATA you've solved your problem. As the need arises, get more ATAs or IP phones or whatever and build out your internal phone network. Jason ___ -- 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+heartbeat
How did you define the secondary IP address? Did you actually set that up in the network scripts and bind it to eth0 or did you just define it in /etc/ha.d/haresources? You should only have the virtual IP defined in haresources along with the primary server and what you want to do on node up/down. My haresources file has a single line: ohasterisk01 10.191.32.31 MailTo::user@domain.com::Asterisk fonulator asterisk We're obviously using the redFone FoneBRIDGE for our T1 connection as you can see we're firing the fonulator script and then asterisk. HTH! Jason -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Nhadie Sent: Tuesday, October 14, 2008 12:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk+heartbeat Hi, I'm using heartbeat as a failover for my asterisk server. on the active server 1 i have 10.10.10.1 eth0 10.10.10.3 secondary eth0 asterisk listens to the secondary ip, so that if server 1 fails, server 2 will then get that IP. so if server 1 fails, server 2 will have the IP 10.10.10.2 eth0 10.10.10.3 secondary eth0 problem is i have to bind asterisk to the secondary IP if dont, i cant make calls. but if server 2 is inactive, asterisk does not run, as on the config it is binded on the secondary ip. anyone uses heartbeat for failover? tia. regards, nhadie ___ -- 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 ___ -- 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] func_odbc creating records or best practice
Robert,
You can access CDR information within the dialplan using the CDR
variable. I'm doing something very similar with a DISA feature for our
employees. We use ODBC to validate them against an existing MSSQL server
(check their employee ID pin number) then when all is well, I write
some information about the call (including the uniqueid field) out to a
'tracking' table I setup. Then I can join the tracking table and the cdr
table on the uniqueid column and associate employees with calls.
In my dialplan, I use the following snippet for setting the values in
the tracking table: (The DBNIS= line is where I do the insert)
exten = valid_login,1,NoOp()
exten = valid_login,n,Set(CALLDATE=${STRFTIME(${EPOCH},GMT+5,%x %X)})
exten = valid_login,n,Set(CLID=${CALLERID(num)})
exten = valid_login,n,Set(UNID=${CDR(uniqueid)})
exten = valid_login,n,Set(DBINS =
${ODBC_DISA(${CALLDATE},${CLID},${ID_ENTERED},${UNID})})
exten =
valid_login,n,Playback(/var/lib/asterisk/sounds/custom/disa_greet3)
exten = valid_login,n,DISA(no-password,from-disa,XXX
614)
exten = valid_login,n(end),Hangup
HTH!
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
McNaught
Sent: Monday, April 28, 2008 6:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] func_odbc creating records or best practice
Hi,
I am trying to write a custom application which will integrate with an
existing MSSQL crm system.
We need to get ahold of the CDR(uniqueid) field in during call-time -
I see from doing a DumpChan(), the CDR unique ID is available as soon
as the call is created. CDRs usind odbc are only written once the
call is completed. Does anyone know if it is possible to use
func_odbc to create a temporary record then delete it so that this
information is available to MSSQL. I was not sure if func_odbc was
limited to just using UPDATE/SELECT queries.
Would there be a better way to do this using the AMI or AGI? It just
seems a little strange to use a database for storing temporary data
such as this?
Thanks in Advance
Robert McNaught
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[asterisk-users] Problem between Asterisk and an Aastra 57i
Stefan, There are a couple of things you'll want to look at but I would seriously recommend setting it back to factory default and starting over. This ensures the settings I don't mention below (a couple you have changed) are reset to factory. If there is a specific reason to have different extensions appear on different buttons, use the Line configuration for each button. However, if the phone is only handling a single extension, put the settings in the Global SIP section. In that section (or in the Line X sections) the only things you need are: - Screen name: What will show on the phone when it is idle - Authentication Name: This is what you have setup in Asterisk as the extension. Usually this is the numeric extension number. - Password: The secret for this extension in Asterisk - Proxy server: Asterisk's IP address - Proxy Port: 5060 - Registrar Server: Asterisk's IP - Registrar Port: 5060 You can obviously change the IP if you don't want to use DHCP... I assume you can handle that. Everything else should work with factory defaults. Also, I have noticed that if the phone must go thru a firewall to reach the Asterisk server, you may need to have 'Send MAC Address in REGISTER Message' turned ON in the Global SIP settings. A router I was using at home prevented a 57i from connecting until I turned that option on. Finally, the guys at NerdVittles (http://nerdvittles.com/) just named the 57i their favorite Asterisk phone. (I agree with them) They have some great config scripts that plug into TrixBox and PBX in a Flash systems that will download the proper config to the phone via TFTP. You just set the TFTP address in the phone and put the MAC address into the server and go... the scripts do the rest and config the phone with the most common settings. You may want to check that out as well. Hope that helps! Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Guenther Sent: Wednesday, March 05, 2008 1:18 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Problem between Asterisk and an Aastra 57i Hi, I'm currently trying to connect an Aastra 57i to our Asterisk Server. The strange thing is, that altough I have definitely entered the correct IP address of the server, the phone doesn't even attempt to register. Here is the configuration file (local.cfg) of the phone: firmware md5: dee6e938b469e217a87138076f47fe41 boot count: 1 tone set: Germany language 1: German time server1: 192.53.103.108 time server2: 192.53.103.104 time format: 1 date format: 5 sip line1 auth name: AWirth sip line1 password: AWirth493 sip line1 user name: 47 sip line1 display name: AWirth sip line1 screen name: Alfred Wirth sip line1 proxy port: 5060 sip line1 registrar ip: 192.168.8.7 sip line1 registrar port: 5060 sip line1 outbound proxy port: 5060 sip line1 registration period: 84400 sip line1 dtmf method: 1 sip line1 missed call summary subscription: 1 sip line1 backup proxy port: 5060 sip line1 backup registrar port: 5060 sip use basic codecs: 1 sip out-of-band dtmf: 0 log module sip: 100 log module net: 100 log server ip: 192.168.8.7 log server port: 514 Has anyone successfully connected an Aastra 57i to an Asterisk server and could give me a hint what maybe wrong? I have used tcpdump to monitor the connection - the phone doesn't send any packages on port 5060. Thanks for any hints, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ -- 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 ___ -- 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] T1, Rhino, Nortel
Steve, Thanks for your input but please allow me to clarify that I'm not a noob. I'm perfectly capable of looking up the definition of a protocol error just the same as you... And I had already done that before I posted the question. I think you misunderstood what I posted. The switchtype in the Nortel is set to DMS100 and it is successfully communicating with a real DMS100. Why would I turn around and change the switchtype in Asterisk to National to see if it's going to work? I already know it won't work unless I change the IFC type for my DCH in the Option 11 as well. And that, if you have any Nortel Meridian experience, is not a job for the faint of heart... especially on a live system. I'm also not convinced that setting localdialplan to unknown or pridialplan to unknown is going to do much of anything either. The debug trace showed the call made it all the way to CONNECT before the Option 11 dumped it. Dialplan format is dealt with much earlier during call setup. The Option 11, in fact, would have rejected the call with a numbering plan error if the dialplan format was wrong. However, I will try these two settings because I'm not using them now and it won't hurt to give them a shot. I suspect though that the real reason is something is not implemented correctly in libpri. Things go downhill in the call trace when the Option 11 responds with a RELEASE message and a cause code of 100. That code is 'Invalid Information Element' which usually means something is not quite right with the last message. In this case, the previous packet is the CONNECT message. This would lead me to believe libpri may not be implementing everything the Option 11 is looking for from a DMS100. There is a bug (see http://bugs.digium.com/view.php?id=9058) involving something similar and I wonder if this isn't along the same lines. I originally asked if anyone else had come across this and if so, what did they do to solve it. I'll ask it again and qualify that I would like to get input from someone that has specific experience with this. I'm trying to sort out if this is possibly a bug that needs reported or something I've got configured wrong. If anyone has successfully connected an Option 11 with Asterisk using a switchtype of DMS100, please e-mail me off-list. I would like to compare notes. Thanks! Jason CLASS 1.6 (Protocol error; e.g. unknown message)Cause No. 96 - Mandatory Info missingThis cause indicates that the equipment sending this cause has received amessage which is missing an information element which must be present in themessage before that message can be processed. This is your first clue. Don't mess with the part that works. Start by trying different values on span 3 (or is it 2? you call it both). I would try switchtype=national first. I would also try things like localdialplan=unknown pridialplan=unknown and other signalling related settings and flip the possible settings around until it works. Look at the example files on your box for ideas. I bet you will have a holy cow! It worked moment and nobody will have to spoon feed you the answer. Thanks, Steve Totaro ___ -- 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 ___ -- 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] T1, Rhino, Nortel
Hi all, I'm trying to insert a Rhino Ceros box equipped with a Rhino R2T1 dual-T1 card and running the latest version of Trixbox (2.4.2) between the central office and a Nortel Option 11. The switch at the CO is a DMS100. Basically, I'm taking the T1, connecting it to port 0 on the R2T1 card, and then connecting port 1 to the Nortel. (Actually a CSU and then the Nortel) We're running PRI over T1... Channels 1-23 are B and channel 24 is D. So I configured the system ahead of time with line encoding, line length, switchtype, timing source, etc. The timing source on port 0 in Zaptel.conf is '1' so I get timing from the CO and it is '0' on port 1 so I send timing to the Nortel. When I hooked it up over the weekend, the spans came up as expected with no errors or anything. Calls between the Asterisk box and the CO work like a charm. The CO doesn't know it's talking to a different box and I get everything I need, call ID, DID, etc with no problems at all. But the calls between the Asterisk box and the Nortel will not go through. I enabled debug on that span and placed calls both ways. When I call from the Nortel to the Asterisk box, the PRI debug shows the call failed with cause code 100. Based on what I can find, this looks like the Nortel is mad about the formatting of something in the messages. When I reverse that and call from the Asterisk box to the Nortel, those calls fail with a cause code of 54. Best I can tell that means 'incoming call barred' but how could it be barred? The Asterisk box should look like the DMS100 to the Nortel. I duplicated the calling information I was seeing from the CO when I tried to call the Nortel plus I tried a couple of variants... no dice. Am I missing something here? I don't understand how I can be talking to a real DMS100 on one T1 and it works perfect but when I act like a DMS100 on the other T1, the Nortel is getting mad. Can anyone offer some ideas? Maybe a clarification on these cause codes? My depth of knowledge in this area isn't that deep... a wading pool at best... so I'm hoping one of you guys that has worked with this stuff a long time might be able to give me some direction. I'm posting below Zaptel, Zapata, and a CLI dump of a call from the Nortel into the Asterisk system with pri debug span turned on. TIA! Jason # Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 2: R2T1/0/1 R2T1 (PCI) Card 0 Span 1 span=2,1,0,esf,b8zs # termtype: cpe bchan=1-23 dchan=24 # Span 3: R2T1/0/2 R2T1 (PCI) Card 0 Span 2 span=3,0,0,esf,b8zs # termtype: net bchan=25-47 dchan=48 # Span 4: Rhino RCB8FXX/1 Rhino RCB8FXX/1 fxsks=49 fxsks=50 fxoks=51 fxoks=52 # ??: 53 ---/1/4 # ??: 54 ---/1/5 # ??: 55 ---/1/6 # ??: 56 ---/1/7 # Global data loadzone= us defaultzone = us ; Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit ; Zaptel Channels Configurations (zapata.conf) ; ; This is not intended to be a complete zapata.conf. Rather, it is intended ; to be #include-d by /etc/zapata.conf that will include the global settings ; ; Span 2: R2T1/0/1 R2T1 (PCI) Card 0 Span 1 group=0 context=from-trunk switchtype = dms100 signalling = pri_cpe channel = 1-23 ; Span 3: R2T1/0/2 R2T1 (PCI) Card 0 Span 2 group=1 context=from-trunk switchtype = dms100 signalling = pri_net channel = 25-47 ; Span 4: Rhino RCB8FXX/1 Rhino RCB8FXX/1 ;;; line=49 FXO/1/0 signalling=fxs_ks callerid=asreceived group=3 context=from-pstn channel = 49 context=default ;;; line=50 FXO/1/1 signalling=fxs_ks callerid=asreceived group=3 context=from-pstn channel = 50 context=default ;;; line=51 FXS/1/2 signalling=fxo_ks callerid=Channel 51 6051 mailbox=6051 group=5 context=from-internal channel = 51 callerid= mailbox= group= context=default ;;; line=52 FXS/1/3 signalling=fxo_ks callerid=Channel 52 6052 mailbox=6052 group=5 context=from-internal channel = 52 callerid= mailbox= group= context=default ; ??: 53 ---/1/4 ; ??: 54 ---/1/5 ; ??: 55 ---/1/6 ; ??: 56 ---/1/7 Protocol Discriminator: Q.931 (8) len=39 Call Ref: len= 1 (reference 21/0x15) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 04 e9 80 83 15] Channel ID (len= 6) [ Ext: 1 IntID: Explicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 DS1 Identifier: 0
[asterisk-users] Need some dialplan help
I'm hoping someone can give me a little dialplan assistance. Here is my
scenario...
I currently have an ATT T1 connected to a Nortel Optn 11. I recently
purchased a Rhino system with a Rhino dual T1 card. What I want to do is
insert the Rhino box between the CO and the Nortel on the T1 so I can
start migrating users over to the Asterisk system in the near future.
But, in the meantime, I basically need to take all the calls that come
into the Rhino box on the first T1 (Zap/g0) just go back out on the
second T1 (Zap/g1). We have a number of DIDs that come in on that T1 and
I need them all transparently bridged for the time being.
We are running the latest Trix install on the Rhino box and I've setup
an inbound route for Zap/g0 that will dump into a custom context
(custom-nortel,s,1) Where I need the help is on the custom context part.
I'm not that strong with the Dial command and I want to make sure I get
it right because I have a very limited time to cut in the physical
insertion and I can't spend the time debugging it when it goes live.
Right now, I literally started with this:
[custom-nortel]
;This custom extension will take calls and put them on the outbound
trunk to the Nortel
exten = s,1,NoOp()
exten = s,n,Set(DIALDIGITS = ${EXTEN}) ;This should put the DID info
into DIALDIGITS
exten = s,n,Dial(Zap/1/${DIALDIGITS},,gjo) ;Dial the Nortel with
the same DID that we were called with
I know I've still got to handle voicemail and such but my main question
is if this will do what I'm looking for? Anyone done something like this
before that would have some insight?
Thanks,
Jason
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[asterisk-users] How are you using Asterisk at Home ?
I setup Trixbox on an Dell Precision 360. I ported my old POTS line over to a pay-as-you-go through Teliax because we weren't using more than 500 minutes a month on the home line. When a caller rings in, I screen the call with time-of-day routing. In general, if the call comes before 7:30 AM or after 10:30 PM, it isn't going to ring through (we had 'problems' with my father-in-law calling us at 7:00 on Saturday to see what we were doing). Instead, they get a voice menu with me politely telling the caller we're not accepting calls at that time. But, I added a code of '111' to that menu and gave it to the family. If they are calling with an emergency, they enter that code and it rings all the extensions in the house plus both of our cell phones. The first one to pickup grabs the call. If calls aren't restricted by TOD, they have to get past privacy manager and blacklist before they will ring some of the extensions (did this with a ring group). If nobody picks up, they are dropped into a voice menu that allows them to leave either of us messages or transfer to our cell phones. This way we can just give everyone a single number and not worry about letting out our cell phone numbers. Of course, calls to the cell phones are confirmed so when one comes in, we have to hit 1 on the cell if we want to accept the call... otherwise its back into VM for the caller. Of course, voicemails are sent via e-mail to my wife and I and I also setup an Aastra 57i on my desk at work that connects to the company server on line 1 and to the home box on line 2. I even got a second line from Teliax in August and set it up to only ring the phone at work. I used this line while I was setting up the wife's surprise 30th birthday party. It was brilliant because guests could call me and there was no trace of the call on my cell phone where she might see it and it didn't ring the home phones. I'm not doing anything really cool like pausing the TV but the setup has worked very well and has given us control over the phone. Instead of us being slaves to when people call, they get through at our pleasure now. It has been a big improvement. (plus it has impressed some of my friends!) Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of D4rk F1ber Sent: Monday, October 08, 2007 6:54 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How are you using Asterisk at Home ? I am very new to Asterisk, it was a weekend project of mine that I jumped into this weekend. I have it up and working on a box at home, and I am nearly half way through the book I purchased friday Asterisk: The Future of Telephony 2nd Edition. Anyway, I started this out so I could help a friend who wanted a VoIP PBX solution for his small business. I have been working with Cisco Callmanager for about 6 years now, and prior to that did help manage other PBXs as well as work on various Motorola VoFR projects as well. My friend came to me and well everything I deal with is really for larger businesses, and since I had heard about Asterisk in the past I thought it would be a good reason to finally jump into it. And what a jump it has been. Only scratching the surface with this thing and well I am very impressed with what I have seen so far. The main point for me writting others is to find out how others are using Asterisk for the home? Bit of over kill for most I am sure, and to be honest we (Wife, kid and I) don't even have a home phone anymore. After playing with this though, shesh I could have fun with it at home. :-) Thinking about getting a SIP line or trunk or something to tie into this for home usage. One of the next projects for me personally is to get a SIP client for my Cingular/ATT 8525, it has wifi and hsdpa running Windows Mobile 6 and I am certain I have run across SIP clients before for these things. Be fun to play with and get working. So yes I am asking because I am unimaginative and need ideas on selling this to the wife. :-) That and I am just curious about what others feel are useful uses for it within the home, and what others get excited about regarding it all. ___ --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 ___ --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] Terrible clicking on T1
Hey All, I have an Asterisk box connected to a Nortel Option 11C via a T1. In the Asterisk box we have a Sangoma A101C and in the Nortel we have a TMDI card. The Nortel is also hooked to the PSTN via a T1 on a different NTAK09 PRI card. I've included the Zapata.conf and zaptel.conf files below. Our issue is that when a call is sent over the tie line between the two systems, the audio on the Asterisk side is terrible. There are rapid 'clicks' on it similar to when you have a cell phone close to an analog phone or a set of computer speakers. The clicks start as soon as the audio channel is opened (when I start to get rings) and it only affects the Asterisk side of the call. But, it affects both inbound and outbound audio on that side. On the Nortel side, the audio they hear is soft and distorted. On the Asterisk side, the audio they hear is full of the clicking but broken thru when the caller speaks. It's almost like the 'silence packets' are being interpreted wrong by Asterisk. If I put the Asterisk box on the T1 for the PSTN, it works perfect. The best part of all this... if we disable the TMDI card in the Nortel and then re-enable it, the audio is pristine... until the Nortel runs it's nightly maintenance routines. Then the noise is back the next day. We can always clear the problem with the disable/re-enable trick but it always come back after maintenance. We've been through tech support with Sangoma and we are confident it isn't the Sangoma card. We've had the TMDI card replaced in the Nortel and we still have the problem. Pure IP calling on the Asterisk box works fine so it isn't between the phones and Asterisk. I'm now completely out of ideas and I'm looking for some direction to go here. Does anybody have any ideas? I desperately need some help. TIA, Jason Asterisk 1.2.18 built by root @ build.trixbox.org on a i686 running Linux on 2007-05-08 22:33:23 UTC # Zaptel Channels Configurations (zaptel.conf) # loadzone=us defaultzone=us #Sangoma A101 port 1 [slot:14 bus:0 span: 1] span=1,0,0,esf,b8zs bchan=1-23 dchan=24 ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-zaptel signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include genzaptelconf configs #include zapata-auto.conf group=1 ;Include AMP configs #include zapata_additional.conf ___ --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] Terrible clicking on T1
On Thu, Aug 09, 2007 at 11:39:38AM -0400, Gleim, Jason wrote: I have an Asterisk box connected to a Nortel Option 11C via a T1. In the Asterisk box we have a Sangoma A101C and in the Nortel we have a TMDI card. The Nortel is also hooked to the PSTN via a T1 on a different NTAK09 PRI card. I've included the Zapata.conf and zaptel.conf files below. Our issue is that when a call is sent over the tie line between the two systems, the audio on the Asterisk side is terrible. There are rapid 'clicks' on it similar to when you have a cell phone close to an analog phone or a set of computer speakers. The clicks start as soon as the audio channel is opened (when I start to get rings) and it only affects the Asterisk side of the call. But, it affects both inbound and outbound audio on that side. On the Nortel side, the audio they hear is soft and distorted. On the Asterisk side, the audio they hear is full of the clicking but broken thru when the caller speaks. It's almost like the 'silence packets' are being interpreted wrong by Asterisk. If I put the Asterisk box on the T1 for the PSTN, it works perfect. The best part of all this... if we disable the TMDI card in the Nortel and then re-enable it, the audio is pristine... until the Nortel runs it's nightly maintenance routines. Then the noise is back the next day. We can always clear the problem with the disable/re-enable trick but it always come back after maintenance. My bet is clock-slip due to a fight over who's clocking the line. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 I thought that might be an issue too... and it was originally. When we started out, I had the Sangoma card generating the timing for the span but we could never get the d-channel to come up. Turns out that since we were connected to the PSTN, we had to let the Nortel set the timing on the span because it was receiving the timing from the CO. (Essentially the timing needed to 'flow' away from the CO) But, since we got that fixed and the span started working, I felt that timing wasn't the source of the problem. Plus, if we dump the error counters on both ends, they are not incrementing... even if the span is up for several days and we clearly have the audio problems. The slip counters, framing error, etc all stay at 0 and you would figure that if it was timing slip, those would be incrementing on at least one of the sides. J. ___ --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 ___ --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] pri call by call trunking?
We spent a considerable amount of time getting an A101 up and running. Try to find out what type of switch you are connecting to. In our case, we were working against a Nortel. For some reason, if we used ni2, it would not work. Finally setting the switchtype to 5ess or DMS100 would work and now everything sings. Hope that helps. Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Wednesday, August 01, 2007 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] pri call by call trunking? Call Sangoma On 8/1/07, Erik Anderson [EMAIL PROTECTED] wrote: On 8/1/07, John covici [EMAIL PROTECTED] wrote: I had some troubles -- try setting the timing parameter to 0 (second one in your span) and see if that helps. If I'm reading the docs correctly, this param should only be set to 0 if you *never* want to use the T1 connected to this port for timing. That's not the case in my situation, as I need to be syncing with the telco's clock. That said, in the interest of troubleshooting, I did try setting it to zero - this didn't fix the problem. -erik ___ --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 ___ --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 ___ --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] Bad Line Noise over T1
I've got a system where I'm integrating a Nortel Option 11c with a Trixbox 2.0.0 system using a Sangoma A101 T1 card. (Running on a Dell PowerEdge 350) We've got things mostly up and running and all seems well... except... If I call from a SIP extension (X-lite soft phone) dialing 9 where is an extension on the Opt 11, the call goes through to the Opt 11 but I have terrible line noise in the earpiece of the softphone and low/distored audio back out of it to the hard phone on the Opt 11. The noise starts the moment the softphone goes 'off-hook' and the hard extension starts ringing and only gets worse once the call is picked up. (essentially white noise by then) Initially, during ringing, the noise is pulsed... like when a cell phone is next to speakers and there is no conversation on the line. Once the call is connected, the noise essentially fills the gaps in the audio... almost like comfort noise gone psycho. The noise reminds me of a modem when the carrier has been established between two endpoints... that static sound. Except that it clearly stops when there is audio on the channel. I've checked the error counters on the A101 card before and after a call and they look fine so it doesn't seem to be jitter or slip or anything like that on the T1. I also tried the calls with 'echocancel' 'echocancelwhenbridged' set to both yes no in Zapata.conf. I've also turned echo cancellation on and off on the A101 card using wancfg. I've tried txgain and rxgain values from 0.0 to -10.0 with no affect. Now, to really mess with things, if I dial another SIP softphone extension on the Asterisk box (or IVR or VM), the audio is pristine so I can rule out softphone problems and issues with the audio hardware on the PC. (Plus, I've tested this from several softphones and they all exhibit the problem.) It is only when I'm routing a call over that T1 that I get the noise. And to add to the mystery, the hard extension on the Opt 11 has no noise on the line. If I talk into the hard phone, I can hear it on the softphone perfectly but the noise fills all the gaps in the audio. If I talk into the softphone, I can hear it on the hard phone but the audio is a bit soft and distorted. I'm stumped on this. I've never ran into this type of audio problem before. Has anyone seen this before and found a solution? Below is Zapata.conf and Zaptel.conf Thanks! Jason ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en group = 0 context=from-zaptel signalling=pri_net switchtype = 5ess callerid = asreceived channel = 1-23 rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=no echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both ;faxdetect=incoming ;faxdetect=outgoing faxdetect=no ;Include genzaptelconf configs #include zapata-auto.conf ;group=1 ;Include AMP configs #include zapata_additional.conf # Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: ZTDUMMY/1 ZTDUMMY/1 1 # Global data loadzone= us defaultzone = us # PRI to Nortel span=1,0,0,esf,b8zs bchan=1-23 dchan=24 ___ --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] Newbie Planning Help
Answers in-line... Hope this helps! Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alan Chandler Sent: Wednesday, February 28, 2007 3:46 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Newbie Planning Help snip -- --- a) to what extent Asterisk can manage everything necessary to allow machines A and B to communicate if they were SIP phones. Is it possible to go for a setup with the firewalls/NAT devices as shown - Asterisk can register and manage both A B even though they are behind NAT devices. NAT=yes is required, of course, for Asterisk and the endpoint to properly communicate. You probably know but just in case, SIP endpoints maintain a signaling channel through port 5060. When a call comes in, they open a RTP media stream somewhere between port 1 and port 2. NAT can sometimes mess this up and it usually shows itself as one-way audio. IAX endpoints send signaling and media over the same port so there is less risk in NAT problems. b) if I go with IAX softphones, does communication between A and B have to go through S, or can Asterisk hand-off the IAX conversation so that A and B talk directly. - I do not believe IAX allows for a hand-off between the two endpoints. Most people don't want the hand-off anyway as it prevents the parties from using in-call feature codes. This is why most everyone sets canreinvite=no for SIP endpoints. c) the example documentation shows seperate entries in iax.conf for incoming and outgoing calls. In my case (assuming IAX softphones) would I just have entries for A and B of type friend? - yes. 'friend' is you friend for IAX softphones! Can someone give me some advice about how to proceed. Thanks -- Alan Chandler http://www.chandlerfamily.org.uk ___ --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 ___ --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
RE: [asterisk-users] Options for moving to * friendly Business VSP
Al, If you want the most flexibility you can get and you want to (or can) use a purely IP solution, then I would recommend looking at the pay-as-you-go plans a lot of VoIP service providers offers. (Lots of recommendations on this list) Most of them will allow you to pay a small monthly fee (~ $5) for a public number and then $0.02/min for usage. The nice thing is that there is no limit to the number of channels. So, if you have a single 'public' number and someone calls it, the service provider sends it to your box. While that call is in progress, if someone else calls, the service provider just opens another channel to your box and it rings. You don't have to maintain a block of numbers and a hunt group which costs money and limits your max simultaneous calls. Same thing with the other company that is merging... just port their main number(s) over and go. You can have several numbers (including 800 TF) which all run over the same connection. The service provider will set the call information during the call setup period. Asterisk can read this and determine which 'line' has been called so you can route appropriately. (Basically DID) To smooth the transition, you would get a temp number from your VoIP service provider and do all the testing. (Since you already have Asterisk setup, this would be as easy as adding a new SIP or IAX trunk.) Then, when you are ready, set your current lines to forward to that temp number and order the number ports. When the ports go through, the numbers will move which will drop the forwards and you should be left with uninterrupted service. You might also find that doing it this way saves you money. A pure IP solution doesn't make you pay for hard-lines that are there strictly for capacity purposes. How often do those last few lines get used in that hunt group versus how much they cost? The real cost per call is much higher on those lines but businesses keep them anyway because they have to be ready for that one time a month when all the lines are busy. I think you'll find with this solution that it scales automatically and as long as you keep the account refilled, you can make and take as many calls as you want. (I believe a number of providers support an account threshold below which they will automatically refill your account with a specific amount.) In regards to your number portability problem... I would make your first call to the public utility commission to find out if CableVision is even allowed to hold that number. I believe a lot of the rules that opened the markets to the CLECs required that a number be portable from the ILEC to any CLEC and vice-versa. Your area may have regulations that require your CLEC to make the number portable between service providers and the person at CableVision you spoke with may either be unaware of it or deliberately misleading you. In general, I find the phone companies suddenly become very cooperative when you call them back with someone from the PUC backing you up. HTH! Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Al Stery Sent: Saturday, October 07, 2006 12:51 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Options for moving to * friendly Business VSP previuos post mangled. Hi all, I have a client whose business is currently running on [EMAIL PROTECTED] 2.6 with Cablevision' s (CV) Optimum Voice (OV) and 3 lines. There are going to be 4 additional trunks needed and I'd like to move/migrate them off of OV, to a better more flexible/open/supportive VSP. OV does not share SIP credentials and operates a closed system which required the use of digium tdm-400b card in order to get the trunks into * and limits what we can achieve. There are two parts to this plan. Here are some of the requirements for the first part. The current 3 lines are setup as a hunt group so there's only one published number. My client needs to (at least for the time being) retain that phone number (business continuity) and CV does NOT allow number's in exchange blocks they own to be ported out. Due to this fact, I was pondering keeping one of the OV trunks open (the main number from the hunt group), and set it to forward all calls to the new hunt group number on the new VSP. This would be done until such time as the majority of customers are updated with the new phone number. I'm not sure how something like this would function but my concern would be how the hand-off on the forward would behave. For example, can this scenario handle multiple incoming calls simultaneously or would one call be dumped off into OV's voicemail system? Also, once a call is forwarded to the new number, is the original OV trunk freed up to accept/forward more incoming calls? or is it tied to that call? Part two. Another business is merging in, bringing with it 4 lines of their own, one of which is an 800 TF number, all currently configured via Verizon POTS serivce. Ideally, I'd like to get those 4 trunks
RE: [Asterisk-Users] Asterisk with Vonage
Brian Deep posted this to the list back in August. I still haven't tried it
myself but he said it worked. If you try it out and it works, please post back
your success. (or failure if that is the case)
Jason
sip.conf:
[general]
externip=X.X.X.X
port=5060
bindaddr=X.X.X.X
context=vonage-out
disallow=all
allow=ulaw
allow=alaw
nat=yes
register=:[EMAIL PROTECTED]:5060/201
[vonage]
username=
type=peer
secret=PASSWORD
port=5060
nat=yes
host=atlas-east.vonage.net
fromdomain=vonage.net
canreinvite=no
fromuser=
dtmfmode=rfc2833
context=vonage-out
[201]
type=friend
username=201
secret=PASSWORD
host=dynamic
dtmfmode=rfc2833
defaultip=X.X.X.X
mailbox=201
callerid=NAME
progressinband=no
context=from-sip
extension.conf:
[vonage-out]
exten = ,1,Goto(from-sip,201,1)
[from-sip]
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Jones
Sent: Wednesday, March 29, 2006 8:59 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk with Vonage
I know Vonage doesn't officially have a bring your own device type program,
but they do offer a softphone. Has anyone gotten Asterisk to connect directly
to Vonage? This would be a great help!!
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RE: [Asterisk-Users] Sprint Nextel sueing over VoIP patents
I wouldn't think anyone would consider Sprint a dying company. They just acquired Nextel so they've got money to spend. Maybe as an ILEC (which they are here in Ohio) they are viewing Vonage and Voiceglo as a force that needs to be stopped to prevent further eroding of their POTS network. I know that I cost SBC money when I dropped them for Vonage. They aren't getting the $$ for that line installed to my home anymore. (Which makes me downright giddy when I think about it.) Imagine if my whole neighborhood switched after SBC had built-out facilities... that would cost SBC a lot of money. They put those POTS lines in counting on them being active and producing income for a long time. Even if service is provided by a CLEC, the line makes money for the ILEC. But if the line is switched off before the payoff horizon, they lose money. Vonage just announced, with much fanfare, that they had hit 1 million lines. In the big scheme that isn't a lot, but that is still 1 million POTS lines that have been abandoned and are costing someone money. Maybe Sprint sees this as an opportunity to leverage their patents to stem the flow of people that are switching? It would explain the timing and why, if the patents have been around for 2, 3, or 4 years, that they are just now trying to enforce them. The lawsuit may be frivolous... Or maybe they are throwing the suit at Vonage Voiceglo thinking it may distract them enough to break some of their momentum... Or maybe Sprint has something, and if they get lucky, it goes their way and tosses the entire thing in the blender? Who knows? But I can't believe Sprint would pull a SCO and sue just to impress the investors. They don't really need to. Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: Friday, October 07, 2005 4:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sprint Nextel sueing over VoIP patents On Fri, 2005-10-07 at 01:34 -0700, John Todd wrote: To answer my own question: no, it doesn't seem like there is anything Asterisk-specific in the suit. It seems that Sprint is claiming that they own the rights to pretty much any VoIP technology. Carry on, everyone; this will be thrown out with the rest of the garbage after Vonage and others spend huge amounts of time and effort staving off the frivolity lawyers. sigh Is this just another dying company, like SCO, trying to give the impression it's still got something for investors. In SCO's case it appears nothing more than a pump and dump exercise. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sprint Nextel sueing over VoIP patents
I'll start with the disclaimer that I am not an attorney... nor do I play one on TV... But, a search of the US Patent Trademark Office reveals 13 patents assigned to Sprint that deal with VoIP. (http://www.uspto.gov/) 6947411 6944150 6937869 6909690 6870857 6868081 6865398 6741695 6731735 6697097 6681116 6556826 6373930 Of particular interest are the '9690, '4150, '1695, '3930 patents. '9690 is a patent on call admission control using silence suppression to better utilize network bandwidth. Specifically, it seems to deal with a method to apply adaptive silence suppression at the customer site... presumably in the ATA. '4150 is a patent on a 'gateway' layer to be implemented between a customer and the communications network as a means of offering and controlling services offered as well as optimizing the deliver of those services. '1695 is a patent on a method to interface packet-based and circuit-switched networks. It specifically mentions SIP and other protocols and how to interface them to signaling and voice paths in a circuit-switched network. Finally, '3930 is a patent on a method to 'redirect' call setup through a third party for the purposes of service restriction or authorization. Basically it's a method of implementing pre-paid service on a packet network. The only one that seems to me that would directly apply to the * community may be the '4150 or '1695 patents. But I don't know enough about patent law to know if it would be worth their time or if they would even have a case. There *maybe* something there too with some of the prepaid modules, like AstCC, if they could argue it was hosted on a separate system. Again, I don't know enough of the specifics to make an educated guess. OK... now that I did my part to add to the FUD, maybe somebody that knows more can build on what I found. Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: Wednesday, October 05, 2005 3:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: asterisk-biz@lists.digium.com Subject: Re: [Asterisk-Users] Sprint Nextel sueing over VoIP patents At 2:43 PM -0700 10/4/05, trixter http://www.0xdecafbad.com wrote: Sprint Nextel is sueing vonage, voiceglo and theglobe.com for infringing on VoIP patents. Sprint Nextel claims to have about 100 patents on VoIP technologies. Does anyone know which ones this article is talking about, and if so does asterisk have any of those features? The reason I am asking is that the article is vague, Vonage uses a fairly standard codec set, I dont know about the others. So if its not codecs I wonder if its something so generic that the patent would be tossed out upon challenge. Anyone thinking about doing a VoIP business may want to get more info before proceeding since they may not have the millinos vonage has to fight this. http://kansascity.bizjournals.com/kansascity/stories/2005/10/03/daily23 .html -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 This perhaps is quite relevant to the Asterisk community. While I don't know the specifics about Vonage, I do know that they have been rumored to have (in the past, or present) used Asterisk in their core for some services. (Voicemail? Conference? Messages?) This, however, is not confirmed. http://www.ilocus.com/ui_dataFiles/news18aug05.htm http://www.google.com/search?num=50hl=enlr=newwindow=1safe=offc2cof f=1q=%22vonage+uses+asterisk%22btnG=Search According to public information, Voiceglo uses IAX and Asterisk: http://lists.digium.com/pipermail/asterisk-users/2004-February/036311.ht ml http://www.business2.com/b2/web/articles/0,17863,1059204,00.html FYI: Voiceglo and theglobe.com are the same company for all intents and purposes. Therefore, I am very interested to see if this is merely co-incidental or if there is a reason that Sprint picked out two providers that use Asterisk in their core. Despite hysteria or misinformation on this (and other) lists, there is no direct information that I've seen that this is Sprint making a blanket patent lawsuit against anyone using VoIP. Perhaps this is just some specific feature that they have a legitimate patent on which has been infringed. I doubt this is a codec patent issue, nor an equipment patent issue (as previously discussed on -biz list.) Is there anyone with better detail on the lawsuit specifics able to comment? JT ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing
RE: [Asterisk-Users] Asterisk and Norstar MICS
I *believe* you can append '#' on the end of the dial string to tell Nortel you are done dialing. I know it works on the Option 11. Hope that helps! Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Thursday, July 21, 2005 10:38 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk and Norstar MICS On Friday 22 July 2005 10:15, Michael Di Martino wrote: My current issues is a 5 second delay for call that is being transferred from the Norstar units to the Asterisk servers VIA a PRI. Is their anything that can be done to speed up the transfer on the Norstar. Below is my current phone config. You need to tell the norstar that you are done dialing. It's waiting for more digits. Routing Service, Public DN Lengths and adjust the correct prefix. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
