[asterisk-users] Best cheap card to use for home Asterisk system???
Hi all - I'm building an Asterisk system (Trix2.2) for the house- I'd like to do the following things: I have a single phone line (happens to be Charter Communications VOIP, but I have their ATA and they've connected to red/green pair in the house wiring) What I'd like to do is this: Get some low-end but reliable card/external adapter which would connect to their ATA and tie into Asterisk to take calls and faxes I'm assuming this should be something with one FXO and one FXS port to connect the incoming line to and to connect the red/green wiring in the house to. I don't mind if all the house phones ring at one time for the moment, as line 2 on them are the Asterisk extensions. Whatever I use must also have failover capability, such that when Asterisk is not working right (server down completely OR just not responding) then the unit fails over and cross connects and makes things work just as is normally the case with Charter only. Unfortunately, I don't have a budget of hundreds of dollars for a true Digium multiport card - I've already built out Asterisk and have a Cisco ATA supporting line 2 on a couple of cordless phones, but I'd like to have the failover piece so that if * starts failing, the home phones still work.. Thanks, Tim ___ --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 and ATA-186 question-- calling one port from the other port..
Hi - I need to configure Asterisk (Trixbox 2.2) and my ATA-186 with both ports. I need to be able to call one port from the other-- the idea is to have two phones in two different locations that _can_ call each other. So, in reading the Asterisk Wiki and other sites, the best documentation I found was this: http://www.loligo.com/asterisk/Cisco/ATA-186-guide.v20030628.txt http://www.loligo.com/asterisk/Cisco/ATA-186-guide.v20030628.txt Note that each line must have it's own distinct and complete configuration, and if you use both lines on the ATA-186, it will REGISTER twice. Further note that you cannot call one line from the other on the same device using the direct extension numbers, so you will have to be clever about naming and aliases within Asterisk. That is outside the scope of this document. However, that _specifically_ says that in the provided config, you cannot call one port from the other port using direct lines-- It does indicate that you CAN in fact work that out, using naming and aliases within Asterisk. Therefore, I assume that it IS possible to use an ATA like this--- but that the author of this particular doc either doesn't know how (but does know it can be done) or just didn't want to go into it in a low-level howto. So --- Does anyone know how to do this? thanks, Tim Most days, there are several fires burning at once. Some days, what's burning is your fire extinguisher. To err is human; to truly screw it up requires the root password. ___ --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 - Cisco Callmanager System Prompts
On my CallManagers here at work, this appears to be what you're looking for: C:\Program Files\Cisco\TFTPPath\CCC\2_0_1_0\en_US This is on CCM 4.1 Tim Reimers Asheville City Schools www.asheville.k12.nc.us desk- 828-350-6195 mobile-828-545-3104 fax- 828-255-5454 Most days, there are several fires burning at once. Some days, what's burning is your fire extinguisher. To err is human; to truly screw it up requires the root password. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Monday, July 16, 2007 2:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT - Cisco Callmanager System Prompts On 7/16/07, Cory Andrews [EMAIL PROTECTED] wrote: Off topic, but involves an Asterisk deployment in a roundabout way. Anyone here intimately familiar with Cisco Callmanager (Version 4-5), that can tell me where a directory of the standard system voice prompts for Callmanager might be obtained? I am looking for the text and filenames of the standard prompt set that ships with Callmanager, have been all over the Cisco site and I can't find it. Thanks in advance! Cory J Andrews Cory, You could try asking on the cisco-voip mailing list: http://puck.nether.net/mailman/listinfo/cisco-voip -- Kristian Kielhofner ___ --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] OT - Cisco Callmanager System Prompts
Just realised that that's the files... and you were asking for the text read out in each one. I don't know if there is any text of them available anywhere-- but the files are there and their names are there. Tim Reimers Asheville City Schools www.asheville.k12.nc.us desk- 828-350-6195 mobile-828-545-3104 fax- 828-255-5454 Most days, there are several fires burning at once. Some days, what's burning is your fire extinguisher. To err is human; to truly screw it up requires the root password. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Monday, July 16, 2007 2:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT - Cisco Callmanager System Prompts On 7/16/07, Cory Andrews [EMAIL PROTECTED] wrote: Off topic, but involves an Asterisk deployment in a roundabout way. Anyone here intimately familiar with Cisco Callmanager (Version 4-5), that can tell me where a directory of the standard system voice prompts for Callmanager might be obtained? I am looking for the text and filenames of the standard prompt set that ships with Callmanager, have been all over the Cisco site and I can't find it. Thanks in advance! Cory J Andrews Cory, You could try asking on the cisco-voip mailing list: http://puck.nether.net/mailman/listinfo/cisco-voip -- Kristian Kielhofner ___ --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] RE: Fatpipe Support - Authorization to open Box - fwrps2001101288
thanks! www.cacti.net - Open source application for handling SNMP manageable devices. There is already a FatPipe host/device template and graph template that someone has built! I haven't loaded it as yet-- note his comments about his RRA being included in his templates. http://forums.cacti.net/about10840.htmlhighlight=fatpipe Your Asterisk people should consider the mailing list: asterisk-users@lists.digium.com Here's the page where those lists come from- I'm only on the users list, but there may be other lists your folks would be interested in. http://lists.digium.com/mailman/listinfo/ It's quite an active list with a fairly large number of people who know what they're doing with Asterisk (quite a few posts are beyond my experience, and I'm a Cisco certified AVVID engineer supporting their VOIP system) thanks for all your help! Tim From: Wyatt Fowler [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 11, 2006 3:58 PMTo: Tim ReimersSubject: Fatpipe Support - Authorization to open Box - fwrps2001101288 *** WYATT *** 4/5/2006 2:30:37 PM I gave permission to Tim Reimer at Asheville City School District to open up the box of the Fatpipe with serial number fwrps2001101288 and reseat the WAN cards due to a possible bad NIC. I checked about the syslog and as of now, the Fatpipe does not support that - I know it is being talked about in development. Wyatt FowlerSupport EngineerFatpipe Networks4455 South 700 East, Suite 100Salt Lake City, Utah 84107Ph:(800)724-8521 xtn 3Fax: (801)281-0317Email: [EMAIL PROTECTED]Website: www.fatpipeinc.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
RE: [Asterisk-Users] Cisco 2620 as PRI gateway
That worked! thanks Kurt... On a side note... Somehow, ALL of the phone calls ended up coming to my SIP phone--- I need to figure out the appropriate 'inbound routing' such that all calls coming from the PRI router (extensions 6350 through 6399) get sent directly to the right extension... right now, the incoming routing is set to 'use incoming calls' settings... I don't know what I need to configure such that calls for all 50 DIDs arriving 'from' the router are handled by (hopefully one route) that says send the call to the extension that matches the digits Thanks, Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of kurt x Sent: Tuesday, February 14, 2006 9:50 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Cisco 2620 as PRI gateway A 488 can mean a codec miss match. Check that your Asterisk box is configured for g729. Kurt ___ --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] Cisco 2620 as PRI gateway
Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of kurt x Sent: Friday, February 10, 2006 10:51 AM To: Asterisk Subject: RE: [Asterisk-Users] Cisco 2620 as PRI gateway debug ccsip message Kurt ___ --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] Cisco 2620 as PRI gateway
Here's a debug--- Longish, but I'm not sure what info in this might be useful to anyone-- I have a zyxel SIP phone configured as ext '6351' on Asterisk-- I can successfully call the SIP phone from an Sjphone client on my PC and talk between the two- so the SIP phone is in fact registered with * correctly... The Cisco router has matching for 63[5-9]x configured- Here's a debug from the router: ACS-GW# *May 26 13:02:42: ISDN Se1/1:23 Q931: pak_private_number: Invalid type/plan 0x0 0x1 may be overriden; sw-type 13 *May 26 13:02:42: ISDN Se1/1:23 Q931: pak_private_number: Invalid type/plan 0x0 0x0 may be overriden; sw-type 13 *May 26 13:02:42: ISDN Se1/1:23 Q931: Applying typeplan for sw-type 0xD is 0x4 0 x1, Called num 3506351 *May 26 13:02:42: ISDN Se1/1:23 Q931: TX - SETUP pd = 8 callref = 0x7A54 Bearer Capability i = 0x8090A2 Standard = CCITT Transer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA18387 Preferred, Channel 7 Calling Party Number i = 0x2181, '8283506180' Plan:ISDN, Type:National Called Party Number i = 0xC1, '3506351' Plan:ISDN, Type:Subscriber(local) *May 26 13:02:42: ISDN Se1/1:23 Q931: RX - CALL_PROC pd = 8 callref = 0xFA54 Channel ID i = 0xA98387 Exclusive, Channel 7 *May 26 13:02:43: ISDN Se1/0:23 Q931: RX - SETUP pd = 8 callref = 0x0014 Bearer Capability i = 0x8090A2 Standard = CCITT Transer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98383 Exclusive, Channel 3 Calling Party Number i = 0x2181, '8283506180' Plan:ISDN, Type:National Called Party Number i = 0x80, '6351' Plan:Unknown, Type:Unknown *May 26 13:02:43: //-1//SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.12.1.252:5060;branch=z9hG4bK12A7 From: sip:[EMAIL PROTECTED];tag=3FB70415-7B4 To: sip:[EMAIL PROTECTED] Date: Sun, 26 May 2002 18:02:43 gmt Call-ID: [EMAIL PROTECTED] Supported: 100rel,timer Min-SE: 1800 Cisco-Guid: 2619306202-1879642582-3189047311-607696096 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY , INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Remote-Party-ID: sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=o ff Timestamp: 1022436163 Contact: sip:[EMAIL PROTECTED]:5060 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 211 v=0 o=CiscoSystemsSIP-GW-UserAgent 7318 2409 IN IP4 10.12.1.252 s=SIP Call c=IN IP4 10.12.1.252 t=0 0 m=audio 16396 RTP/AVP 18 c=IN IP4 10.12.1.252 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 *May 26 13:02:43: //-1//SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 10.12.1.252:5060;branch=z9hG4bK12A7 From: sip:[EMAIL PROTECTED];tag=3FB70415-7B4 To: sip:[EMAIL PROTECTED];tag=as21b1fb71 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 *May 26 13:02:43: ISDN Se1/0:23 Q931: TX - CALL_PROC pd = 8 callref = 0x8014 Channel ID i = 0xA18383 Preferred, Channel 3 ACS-GW# *May 26 13:02:43: //-1//SIP/Msg/ccsipDisplayMsg: Sent: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.12.1.252:5060;branch=z9hG4bK12A7 From: sip:[EMAIL PROTECTED];tag=3FB70415-7B4 To: sip:[EMAIL PROTECTED];tag=as21b1fb71 Date: Sun, 26 May 2002 18:02:43 gmt Call-ID: [EMAIL PROTECTED] Max-Forwards: 70 CSeq: 101 ACK Content-Length: 0 *May 26 13:02:43: ISDN Se1/0:23 Q931: TX - DISCONNECT pd = 8 callref = 0x8014 Cause i = 0x8081 - Unallocated/unassigned number ACS-GW# *May 26 13:02:43: ISDN Se1/1:23 Q931: RX - PROGRESS pd = 8 callref = 0xFA54 Cause i = 0x8281 - Unallocated/unassigned number Progress Ind i = 0x8288 - In-band info or appropriate now available *May 26 13:02:43: ISDN Se1/0:23 Q931: RX - RELEASE pd = 8 callref = 0x0014 *May 26 13:02:43: ISDN Se1/0:23 Q931: TX - RELEASE_COMP pd = 8 callref = 0x801 4 ACS-GW# *May 26 13:02:49: ISDN Se1/1:23 Q931: TX - DISCONNECT pd = 8 callref = 0x7A54 Cause i = 0x8090 - Normal call clearing *May 26 13:02:49: ISDN Se1/1:23 Q931: RX - RELEASE pd = 8 callref = 0xFA54 *May 26 13:02:49: ISDN Se1/1:23 Q931: TX - RELEASE_COMP pd = 8 callref = 0x7A5 4 ACS-GW# *May 26 13:03:07: //-1//SIP/Msg/ccsipDisplayMsg: Received: OPTIONS sip:10.12.1.252 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.28:5060;branch=z9hG4bK3741dad8 From: Unknown sip:[EMAIL PROTECTED];tag=as6479f479 To: sip:10.12.1.252 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102
RE: [Asterisk-Users] Cisco 2620 as PRI gateway
Yeah-- sorry...
"
dial-peer voice 635099 voipdescription calls sent
to Asteriskpreference 1destination-pattern
[635-9]..progress_ind setup enable 3session target
ipv4:10.10.1.28dtmf-relay h245-alphanumeric
"
I had been trying to do this with H.323 -- the Call Manager
uses H.323
There are some sip commands available in that dial-peer
ACS-GW(config-dial-peer)#voice-class sip ?
rel1xx Type of reliable provisional response
support transport Configure transport related
parameters url url type in
request line of outgoing INVITE
Not sure how I set those---
This:
voice-class codec 1voice-class h323
1
is what is in there for the Call Manager h.323 dial-peer
That's obviously NOT what I want for the Asterisk-SIP
connection...
but I don't know what Ineed to do regarding the 'sip
url' or 'sip transport' or 'sip rel1xx' commands, if
anything...
How does one debug SIP activity? I see the debugs for
calls--- but I don't know the related debugs for actively
watching--
like you would 'debug isdn q931' -- that's the
outgoing side of the router--
what would be the debug for a SIP call 'arriving' at the
router??
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Juan
SalasSent: Wednesday, February 08, 2006 2:17 PMTo:
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE:
[Asterisk-Users] Cisco 2620 as PRI gateway
Did
you create the dial-peers in the2651?
-Mensaje original-----De: Tim Reimers
[mailto:[EMAIL PROTECTED]Enviado el: Wednesday,
February 08, 2006 1:41 PMPara: Asterisk Users Mailing List -
Non-Commercial DiscussionAsunto: RE: [Asterisk-Users] Cisco 2620 as
PRI gateway
sip-ua sip-server ipv4:asterisk server
ip address
OK -
So I added those lines to my 2651 with the IP of my
asterisk box...
How would I set this up as a SIP trunk in
Asterisk?
I have done this, in building a SIP trunk in
AMP.
host=10.12.1.252type=friend
I
don't know if/how to specify a username/password (as was the defaults in
there- the router didn't support having that configured..)
So I
picked friend..
Then, in call routing, I picked my "Outbound
Routing"
the
"9_outside" route of "9|."
Set
that to use the new 'gw-rtr' I'd created...
no
go...
Debug ISDN q931 doesn't show anything going to the
router...
In
Asterisk-
" -- Got SIP response 481 "Call Leg/Transaction Does Not Exist"
back from 10.12.1.252"
snipped from below
The
router doesn't show anything...
the below shows up in
Asterisk - mode
-- Executing Macro("SIP/6351-cc18",
"dialout-trunk|3|2439499|") in new stack -- Executing
GotoIf("SIP/6351-cc18", "1?3:2)") in new stack -- Goto
(macro-dialout-trunk,s,3) -- Executing
Macro("SIP/6351-cc18", "user-callerid") in new stack --
Executing DBget("SIP/6351-cc18", "AMPUSER=DEVICE/6351/user") in new
stack -- DBget: varname=AMPUSER, family=DEVICE,
key=6351/user -- DBget: set variable AMPUSER to
6351 -- Executing DBget("SIP/6351-cc18",
"AMPUSERCIDNAME=AMPUSER/6351/cidname") in new stack --
DBget: varname=AMPUSERCIDNAME, family=AMPUSER,
key=6351/cidname -- DBget: set variable AMPUSERCIDNAME
to Tim-Zyxel -- Executing GotoIf("SIP/6351-cc18", "0?5")
in new stack -- Executing SetCallerID("SIP/6351-cc18",
"Tim-Zyxel 6351") in new stack -- Executing
NoOp("SIP/6351-cc18", "Using CallerID "Tim-Zyxel" 6351") in new
stack -- Executing Macro("SIP/6351-cc18",
"record-enable|6351|OUT") in new stack -- Executing
GotoIf("SIP/6351-cc18", "0 0?2:4") in new stack --
Goto (macro-record-enable,s,4) -- Executing
AGI("SIP/6351-cc18", "recordingcheck|20060208-115748|1139417868.14") in new
stack -- Launched AGI Script
/var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060208-115748|1139417868.14: Outbound recording not
enabled -- AGI Script recordingcheck completed,
returning 0 -- Executing NoOp("SIP/6351-cc18", "No
recording needed") in new stack -- Executing
Macro("SIP/6351-cc18", "outbound-callerid|3") in new
stack -- Executing GotoIf("SIP/6351-cc18", "1?3") in new
stack -- Goto
(macro-outbound-callerid,s,3) -- Executing
DBget("SIP/6351-cc18", "USEROUTCID=AMPUSER/6351/outboundcid") in new
stack -- DBget: varname=USEROUTCID, family=AMPUSER,
key=6351/outboundcid -- DBget: set variable USEROUTCID
to 6351 -- Executing GotoIf("SIP/6351-cc18", "0?6") in
new stack -- Executing SetCallerID("SIP/6351-cc18",
"6351") in new stack -- Executing NoOp("SIP/635
RE: [Asterisk-Users] Cisco 2620 as PRI gateway
Gary- You have a Cisco 2600 acting as both a SIP gateway to the Asterisk box -and- as an H.323 gateway for your CCM? Can you provide me with config details? 12.3(8)T3,c2600-ipvoice-mz.123-8.T3 with T1 (2 Port) Multi-Flex Trunk I tried to set up h.323 to the Asterisk box-- didn't know there was a possibility of running SIP and H.323 at the same time... thanks, Tim From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary RichardsonSent: Tuesday, February 07, 2006 9:09 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Cisco 2620 as PRI gateway I have a 2811 working as a SIP gateway. My IOS version is 12.3(11)T5. Looking through my config I notice:sip-ua sip-server ipv4:asterisk server ip addressEverything else in the config file is for our h323 call manager gear. I can't remember if I needed to add the above line to make a sip server run on the router. In order to place a call to the PSTN, I Dial(SIP/9XX@ip address of my 2811) and everything works. As for how much of this applies to a 2600.. you'll have to see. On 2/6/06, Schochet, Wes [EMAIL PROTECTED] wrote: I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it.Can I make this thing into MGCP gateway or even a SIP gateway for asterisk?Seems likeit should bee useful for something!I'm perfectly happy to do my homework, but also don't feel thee need toreinvent the wheel!So, links with relevant info would be appreciated.If there is a config for a 2621 being used as a gateway out there somewhere, Iwouldn't be too proud to take a look at that either!Asterisk configs wouldbe great too!Thanks,Wes___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Cisco 2620 as PRI gateway
sip-ua sip-server ipv4:asterisk server ip
address
OK -
So I added those lines to my 2651 with the IP of my
asterisk box...
How would I set this up as a SIP trunk in
Asterisk?
I have done this, in building a SIP trunk in
AMP.
host=10.12.1.252type=friend
I
don't know if/how to specify a username/password (as was the defaults in there-
the router didn't support having that configured..)
So I
picked friend..
Then,
in call routing, I picked my "Outbound Routing"
the
"9_outside" route of "9|."
Set
that to use the new 'gw-rtr' I'd created...
no
go...
Debug
ISDN q931 doesn't show anything going to the router...
In
Asterisk-
" -- Got SIP response 481 "Call Leg/Transaction Does Not Exist"
back from 10.12.1.252"
snipped from below
The
router doesn't show anything...
the below shows up in
Asterisk - mode
-- Executing Macro("SIP/6351-cc18",
"dialout-trunk|3|2439499|") in new stack -- Executing
GotoIf("SIP/6351-cc18", "1?3:2)") in new stack -- Goto
(macro-dialout-trunk,s,3) -- Executing
Macro("SIP/6351-cc18", "user-callerid") in new stack --
Executing DBget("SIP/6351-cc18", "AMPUSER=DEVICE/6351/user") in new
stack -- DBget: varname=AMPUSER, family=DEVICE,
key=6351/user -- DBget: set variable AMPUSER to
6351 -- Executing DBget("SIP/6351-cc18",
"AMPUSERCIDNAME=AMPUSER/6351/cidname") in new stack --
DBget: varname=AMPUSERCIDNAME, family=AMPUSER,
key=6351/cidname -- DBget: set variable AMPUSERCIDNAME to
Tim-Zyxel -- Executing GotoIf("SIP/6351-cc18", "0?5") in
new stack -- Executing SetCallerID("SIP/6351-cc18",
"Tim-Zyxel 6351") in new stack -- Executing
NoOp("SIP/6351-cc18", "Using CallerID "Tim-Zyxel" 6351") in new
stack -- Executing Macro("SIP/6351-cc18",
"record-enable|6351|OUT") in new stack -- Executing
GotoIf("SIP/6351-cc18", "0 0?2:4") in new stack --
Goto (macro-record-enable,s,4) -- Executing
AGI("SIP/6351-cc18", "recordingcheck|20060208-115748|1139417868.14") in new
stack -- Launched AGI Script
/var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060208-115748|1139417868.14: Outbound recording not
enabled -- AGI Script recordingcheck completed, returning
0 -- Executing NoOp("SIP/6351-cc18", "No recording
needed") in new stack -- Executing Macro("SIP/6351-cc18",
"outbound-callerid|3") in new stack -- Executing
GotoIf("SIP/6351-cc18", "1?3") in new stack -- Goto
(macro-outbound-callerid,s,3) -- Executing
DBget("SIP/6351-cc18", "USEROUTCID=AMPUSER/6351/outboundcid") in new
stack -- DBget: varname=USEROUTCID, family=AMPUSER,
key=6351/outboundcid -- DBget: set variable USEROUTCID to
6351 -- Executing GotoIf("SIP/6351-cc18", "0?6") in new
stack -- Executing SetCallerID("SIP/6351-cc18", "6351") in
new stack -- Executing NoOp("SIP/6351-cc18", "CallerID set
to 6351") in new stack -- Executing
SetGroup("SIP/6351-cc18", "OUT_3") in new stack --
Executing CheckGroup("SIP/6351-cc18", "") in new stack --
Executing SetVar("SIP/6351-cc18", "DIAL_NUMBER=2439499") in new
stack -- Executing SetVar("SIP/6351-cc18", "DIAL_TRUNK=3")
in new stack -- Executing AGI("SIP/6351-cc18",
"fixlocalprefix") in new stack -- Launched AGI Script
/var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Could not
parse /etc/asterisk/localprefixes.conf -- AGI Script
fixlocalprefix completed, returning 0 -- Executing
SetVar("SIP/6351-cc18", "OUTNUM=2439499") in new stack --
Executing Cut("SIP/6351-cc18", "custom=OUT_3|:|1") in new
stack -- Executing GotoIf("SIP/6351-cc18", "0?16") in new
stack -- Executing Dial("SIP/6351-cc18",
"SIP/acs-gw-rtr/2439499") in new stack -- Called
acs-gw-rtr/2439499 -- SIP/acs-gw-rtr-b33f is
circuit-busy == Everyone is busy/congested at this time
(1:0/1/0) -- Executing Goto("SIP/6351-cc18",
"s-CONGESTION|1") in new stack -- Goto
(macro-dialout-trunk,s-CONGESTION,1) -- Executing
NoOp("SIP/6351-cc18", "Dial failed due to CONGESTION") in new
stack -- Executing Macro("SIP/6351-cc18", "outisbusy") in
new stack -- Executing Playback("SIP/6351-cc18",
"allison7/all-circuits-busy-now") in new stack -- Got SIP
response 481 "Call Leg/Transaction Does Not Exist" back from
10.12.1.252 -- Playing 'allison7/all-circuits-busy-now'
(language 'en') -- Executing Playback("SIP/6351-cc18",
"allison7/pls-try-call-later") in new stack -- Playing
'allison7/pls-try-call-later' (language 'en') -- Executing
Macro("SIP/6351-cc18", "hangupcall") in new stack --
Executing ResetCDR("SIP/6351-cc18", "w") in new stack --
Executing NoCDR("SIP/6351-cc18", "") in new stack --
Executing Wait("SIP/6351-cc18", "5") in new stack --
Executing Hangup("SIP/6351-cc18", "") in new stack == Spawn extension
(macro-hangupcall, s, 4) exited non-zero on 'SIP/6351-cc18' in macro
'hangupcall' == Spawn extension (macro-outisbusy, s, 3) exited
non-zero on 'SIP/6351-cc18' in macro 'outisbusy' == Spawn extension
(from-internal, 92439499, 2) exited non-zero on
'SIP/6351-cc18' -- Executing Macro("SIP/6351-cc18",
"hangupcall") in new stack -- Executing
RE: [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update ornot?
CentOS is what [EMAIL PROTECTED] is based on--- that should also inform of someone else's thinking about stable distros to use.. CentOS is still RPM based, so you'd be in familiar turf as far as that goes-- no 'apt' stuff to relearn to support a new distro.. t -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russ Price Sent: Tuesday, February 07, 2006 11:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update ornot? Zach A wrote: What is recommended for a production quality system, FC3 or FC4. Once installed, is it necessary to run yum update, does that make things any better or just take up more memory? I wouldn't recommend Fedora Core for a production system - at least not a server. For one thing, FC3 is now obsolescent, and FC updates in general have a very good chance of breaking things; I know from personal experience. Once support stops for a Fedora Core version, security updates via Fedora Legacy are few and far between. I'd go with CentOS 4.2 instead, or, if you have the bucks, the corresponding RHEL version. Updates are provided for a much longer period, and are far less likely to break things. Russ ___ --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
