Re: [asterisk-users] Transfer Call to Cell Phone
We do have full features on our lines so both lines are free once the
transfer is complete. We also have toll calls on our lines so it would
not be a problem, so I do not have to worry about ATT dropping the
The issue really isn't whether you have the ability to make toll calls on
your line. The concern here is in what the regulatory agencies call toll
bridging which is using a system to relay a call from one local calling are
to another local calling area to avoid a toll charge. This is one of those
gray areas that can become a problem if your not careful. The problem comes
up if you have customers that can call you as a local call and you are
forwarding them on to another party that is a local call for you but would
be a toll call for the customer. This is essentially what toll bridging is
about. Now your not likely to have to worry about the legal ramifications of
this since your merely connecting the customer with an extension of your
company, namely your salesman. Where this could become a problem for you
would be in transferring the customer using the same pots line. The reason
is that ATT is handling the transfer. When you transfer the call, it
essentially becomes a new call. The main difference is that you have
provided the called number. So the software in the Class 5 (End office)
switch, takes the number you provide and runs the call through its routing
translations (similar to the Asterisk dialing plan) and if it determines
that the destination number is outside the originators Local Area Transport
Area or LATA, then it will either drop the originator to a message that
says, You must first dial a 0 or 1 before calling this number or it may
deny the transfer allowing you to stay connected to the customer. Neither
one looks very professional. The only way around this would be to provide
another line or trunk to pass the call down. Now if your not in an
overlapping LATA this probably isn't an issue.
The only way I can get it to work is by have the call on the 1st
line then transfer it out on the 2nd line. After that is complete both
lines are free.
Are you saying that you are able to route a call from line 1 to line 2 and
have the call transfer, thus freeing the lines or that once the call
completes the lines are freed? I've never seen the first scenario. The
second scenario is the normal behavior.
Can you give an example of creating an extension which points to a cell
phone. Secondly how can you have if no one answers an extension it dials
the cell number next. That maybe answered in the example.
In extensions.conf use something like this.
[global]
SIP-PROV = sip.urprovider.com
; Now set the call forward numbers
CFN21 = 551234 ; These are normally set in an external file
[internal]
exten = 21,1,Macro(stdext,${SIP/21},${CFN${EXTEN}})
[macro-stdext];
; Standard extension macro:
; ${ARG1} - Device(s) to ring
; ${ARG2} - Our call forward number
exten = s,1,Dial(${ARG1},10)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,GotoIf($[${LEN(${ARG2})}0]?s-CFWD,1)
exten = s-NOANSWER,2,Voicemail(${MACRO_EXTEN},u)
exten = s-BUSY,1,Voicemail(${MACRO_EXTEN},b)
exten = s-CFWD,1,Dial(SIP/[EMAIL PROTECTED],20)
exten = s-CFWD,2,Goto(s-NOANSWER,2)
exten = _s-.,1,Goto(s-NOANSWER,2)
exten = a,1,VoicemailMain(${MACRO_EXTEN})
There is more to this but this should show the basics of what we use. I
store my Call Forward Numbers (CFN) in an external file. This allow me to
update the file externally (currently with a web interface but as soon as I
get the prompts recorded it will be done with an IVR) and then just reload
the extensions to activate the new numbers. Also I using SIP for pretty much
everything. Our TDM400 doesn't even have modules, it's just there for
timing. However you should be able to convert the SIP calls to ZAP calls for
you use. The internal context is included in our default context. Dialing
extension 21 calls the stdext macro. This dials the local extension first.
If not answered after 10 seconds, we check to make sure we have a phone
number to send the call out with. If not we send it on to voice mail.
Otherwise we send it to the s-CFWD. The check listed here is a very
rudimentary check but again I hope you get the idea. Next we try the call to
the CFN. If not answered in 20 seconds, then we send it to voice mail.
Finally if the user presses the star button during the attempt, we send them
on to Voicemail mail so they can check their messages.
Hopefully this helps.
John
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Re: [asterisk-users] Urgent. When the peer returned a 301 forwarded, asterisk thinks it's a local extension.
When making an outbound call, the outbound peer return a 301 forwarded with URI to other domain, but asterisk think it's a local domain and try to look it up from extension.conf. What phones are you using? This sounds a lot like a problem, I have using Grandstream phones. John ___ --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] Modification of Caller ID based on context
Matthew Brothers wrote:
Hi,
I have been looking for an example of accomplishing this, but
I've been unable to locate something similar to what I'm trying
to do.
Here's the scenario:
Users caller ID is set to their internal extension (200-250).
This is set in sip.conf for each user. Each user has a local DID
as well (hosted through Vitelity, for example (555)111-). The
problem is that this extension was being passed to the outside
world. I currently have a SetCallerID command changing the
CallerID to our main office number, but some users want their DID
sent, not the general number.
The problem is that if their caller ID is set to their DID, when
users hit redial on their phones internally they dial out and
back in. I corrected this by putting each DID in extensions.conf
under their three digit extension, but that seems a bit like a
kludge obviously.
I'm looking for a method of sending the internal three digit
extension only when a user is dialing another user internally,
otherwise it will send their DID. Is their a method to do this in
the dial plan? Anyone have an example of how to accomplish this?
Thanks in advance.
Mike,
I have a similar setup (I even use Vitel) and the easiest and
cleanest method that I have found to accomplish this is with the
AstDB. You can simply create a cross-reference of DIDs and Internal
extensions similar to extdid/200 = 555111 ... extdid/250 =
5551112272 in the AstDB. Then you can change your outgoing dialplan
to change the caller id based upon this cross reference. Example:
exten = NXXNXX,n,Set(outgoingCID=MAINNUMBER)
exten = NXXNXX,n,
GotoIf($[ ${DB_EXISTS(extdid/${CALLERID(num)})} = 0 ]?makecall)
exten = NXXNXX,n,
Set(outgoingCID=${DB(extdid/${CALLERID(num)})})
exten = NXXNXX,n(makecall),Set(CALLERID(num)=${outgoingCID})
...
You could even simplify your incoming context by cross-referencing
in the other direction. That is didext/555111 = 200 ...
didext/5551112272 = 250.
exten = NXXNXX,n,
Goto(internal-extensions,${DB(didext/${EXTEN})},1)
OR you could do something similar with LOCAL channels or with a Dial
command.
Here is my solution. I've stripped out most of the unimportant stuff.
Because our carrier charges for PICs on a per-DID basis, we set the
Caller*ID number for long distance calls to be the main number,
regardless of what the person's DID is. It also allows use of more
than one main number, depending on the device making the call.
The macro-dial-result is not important for this. It is a macro we use
to figure out what happened to the call based on HANGUPCAUSE and what,
if any tone or message to send the caller, as well as decide if the call
failed and should be sent out a different route.
In sip.conf set up the device like this:
[0004f201e570-a]
callerid=Room, Computer 3726
setvar=DID=9852463726
setvar=BTN=9858982022
accountcode=3726
type=friend
host=dynamic
secret=S
context=toll-access
My extensions.conf looks like this:
[toll-access]
;
; 9-1-nxx-nxx-
exten = _91NXXNXX,1,Set(USE_BTN=yes)
exten = _91NXXNXX,n,Gosub(outgoing-call-fixup,${EXTEN},1)
exten = _91NXXNXX,n,Dial(${PSTN}/${EXTEN:1},,g)
exten = _91NXXNXX,n,Macro(dial-result,SIP/[EMAIL PROTECTED])
;
; 9-1-985-nxx-
exten = _91985NXX,1,Gosub(outgoing-call-fixup,${EXTEN},1)
exten = _91985NXX,n,Dial(${PSTN}/${EXTEN:1},,g)
exten = _91985NXX,n,Macro(dial-result,SIP/[EMAIL PROTECTED])
[outgoing-call-fixup]
;
exten = _X.,1,GotoIf($[${LEN(${CALLERID(num)})} != 10]?check-btn)
exten = _X.,n,Return
exten = _X.,n(check-btn),GotoIf($[${USE_BTN} = yes]?set-btn)
exten = _X.,n,Set(CALLERID(num)=${DID})
exten = _X.,n,Return
exten = _X.,n(set-btn),Set(CALLERID(num)=${BTN})
exten = _X.,n,Return
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Re: [asterisk-users] AudioCodes Gateway and Asterisk
Sent it to AudioCodes (in a text file). I will let you guys know what the issue was. - Original Message - From: Shanon Swafford [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, June 27, 2007 1:22 AM Subject: Re: [asterisk-users] AudioCodes Gateway and Asterisk When you see [ERROR] in the Message Log, either the MP firmware is buggy or the far end is sending something out of spec in the SIP Message. You'll need to upgrade to the latest MP firmware then report this to whomever you bought it from. Or fix the far end to send the message in spec or form that doesn't cause the [ERROR]. Also, do your supporter a favor and don't paste those logs directly into emails. The wrap makes them horrible to read and they can't send them on to Audiocodes like that. Put them in a text file which preserves the line length. Regards, Shanon http://www.abptech.com/support/qa/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B Sent: Sunday, June 24, 2007 2:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AudioCodes Gateway and Asterisk - Original Message - From: Shanon Swafford [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, June 21, 2007 6:27 PM Subject: Re: [asterisk-users] AudioCodes Gateway and Asterisk On 6/21/07, Dovid B [EMAIL PROTECTED] wrote: Hi List, I am trying to call from my asterisk box (1.2.18) to and audiocodes MP114. I keep getting an error from asterisk of -- Got SIP response 415 Unsupported Media Type back from XXX.XXX.XX.XX. Both box's are set up to use G729. Anyone have a hint as to what it may be ? Are you sure, your asterisk supports G729? It isn't supported by default, you need additional modules or hardware cards for G729 support. If it is - what are you using for G729 - that might help to identify the problem. Regards, Atis If the AudioCodes is sending back that 415, the Message Log in the AudioCodes is invaluable. Set your debug level to 5/6 and watch it while you make test calls. Once you learn how to interpret this output, you'll be well on your way with AudioCodes. If G729 is active on the MP, but still giving back that error, G729 might not be in a profile if you are using them. Also, firmware that comes on the MPs is normally sorta buggy, ask your reseller for the latest version. http://www.abptech.com/support/faqs/ Regards, Shanon ABP Technology Shanon, The audiocodes were preftctly with other providers using G729. It's just having an issue with asterisk. Here is the output from the AudioCodes: Log is Activated 12d:23h:36m:17s ( lgr_flow)(828 ) Incoming SIP Message from XXX.XXX.XX.XXX:5060 [File: Line:-1] 12d:23h:36m:17s INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Record-Route: lt;sip:XXX.XXX.XX.XXX;ftag=as4a537e63;lr=ongt; Via: SIP/2.0/UDP XXX.XXX.XX.XXX;branch=z9hG4bK1db7.ecac6e86.0 Via: SIP/2.0/UDP XXX.XXX.XX.XXX:5060;branch=z9hG4bK1ef72aa9;rport=5060 From: 55560888 lt;sip:[EMAIL PROTECTED]gt;;tag=as4a537e63 To: lt;sip:[EMAIL PROTECTED]gt; Contact: lt;sip:[EMAIL PROTECTED]:5060gt; Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Enswitch Max-Forwards: 16 Date: Wed, 20 Jun 2007 19:44:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 490 v=0 o=root 29170 29170 IN IP4 XXX.XXX.XX.XXX s=session c=IN IP4 XXX.XXX.XX.XXX t=0 0 m=audio 14878 RTP/AVP 18 0 8 10 3 111 5 7 110 97 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no;mode-change-period=0 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/80 12d:23h:36m:17s ( sip_stack)(830 ) ?? [WARNING] AcSIPParser: Unrecognized Header was detected at line: 12 [File: Line:-1] 12d:23h:36m:17s ( lgr_flow)(831 ) | | new GetNewSIPCall created - #8 [File: Line:-1] 12d:23h:36m:17s ( sip_stack)(832 ) new AcSIPCallAPI created - #5 [File: Line:-1] 12d:23h:36m:17s ( lgr_stk_mngr)(833 ) Resource StackSession lt;#5gt; Allocated [File: Line:-1] 12d:23h:36m:17s ( lgr_flow)(834 ) | |(SIPTU#8)INVITE State:Idle() [File: Line:-1] 12d:23h:36m:17s ( sip_stack)(835 ) SIPCall(#8) changes state from Idle to Invited [File: Line:-1] 12d:23h:36m:17s ( sip_stack)(836 ) AcSIPParser: Problem in AcSIPCallAPI::ParseSDP [File: Line:-1] 12d:23h:36m:17s ( sip_stack)(837 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol ' [File: Line:-1] 12d:23h:36m:17s ( sip_stack)(838 ) !! [ERROR] Message type: INVITE [File: Line:-1] 12d:23h:36m:17s ( sip_stack)(839 ) !! [ERROR] Source header: [File: Line:-1] 12d:23h:36m:17s ( sip_stack)(840 ) !! [ERROR] Line: 20. Column: 27 [File: Line:-1] 12d:23h:36m:17s ( lgr_flow)(841 ) | | | #5:SIP_SETUP_EV([EMAIL PROTECTED]) [File: Line:-1] 12d:23h:36m:17s ( lgr_stk_ses)(842 )
[asterisk-users] Fwd: problem with one way audio
-- Forwarded message -- From: Vidura Senadeera [EMAIL PROTECTED] Date: Jun 27, 2007 1:56 PM Subject: Re: problem with one way audio To: asterisk-users@lists.digium.com Hi, If you have analog or digital cards installed. make sure to configure cards with proper signalling in /etc/zapel.conf. Hope you will be eliminate the issue using this hint. Regards, Vidura. ___ --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] Zap dialling issues
I'm having problems getting an Xorcom USB Bri 4 dialling out in
Australia.
I can receive calls into the system without an issue, but I can not for
the life of me dial out of the system. Below are my configs, I'm hoping
its something simple that I just can't see as I've been looking at it
for to long. Can any one point me in the right direction.
P.S. Yes it is meant to be in TE PTP mode as the current Digium B410P
works fine in that mode
/etc/asterisk/extensions.conf Extract
[internal]
include=features
include=speeddial
;Extention number for main Q
exten = 700,1,Goto(mainq,q,1)
;-
;Calling a local extensions mailbox
exten = _*7XX,1,Set(Extension=${EXTEN:1})
exten = _*7XX,n,Goto(directtovoicemail,s,1)
;--
;Static externaly accessable Conference room with recording
exten =
599,1,Set(MEETME_RECORDINGFILE=/var/spool/asterisk/monitor/conference-50
0-${EPOCH});
exten = 599,n,MeetMe(500,cMr,4081)
exten = 599,n,Hangup
;Dynamic Conference rooms for internal users to transfer callers to
exten = _5XX,1,MeetMe(${EXTEN},cMd)
exten = _5XX,n,Hangup
exten = 6000,1,Dial(zap/0418608609)
/etc/asterisk/zapata.conf
[channels]
;echocancel = yes
;transfer = yes
callgroup=1
pickupgroup=1
; Span 1: XBUS-00/XPD-00 Xorcom XPD #0/0: BRI_TE
group=0,11
context=zapin
switchtype = euroisdn
signalling = bri_cpe
channel = 1-2
; Span 2: XBUS-00/XPD-08 Xorcom XPD #0/8: BRI_TE
group=0,12
context=zapin
switchtype = euroisdn
signalling = bri_cpe
channel = 4-5
; Span 3: XBUS-00/XPD-10 Xorcom XPD #0/16: BRI_TE
;group=0,13
;context=zapin
;switchtype = euroisdn
;signalling = bri_cpe
;channel = 7-8
; Span 4: XBUS-00/XPD-18 Xorcom XPD #0/24: BRI_TE
;group=0,14
;context=zapin
;switchtype = euroisdn
;signalling = bri_cpe
;channel = 10-11
/etc/zaptel.conf
# Autogenerated by /usr/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: XBUS-00/XPD-00 Xorcom XPD #0/0: BRI_TE
span=1,1,1,ccs,ami
# termtype: te
bchan=1-2
dchan=3
# Span 2: XBUS-00/XPD-08 Xorcom XPD #0/8: BRI_TE
span=2,2,1,ccs,ami
# termtype: te
bchan=4-5
dchan=6
# Span 3: XBUS-00/XPD-10 Xorcom XPD #0/16: BRI_TE
span=3,3,1,ccs,ami
# termtype: te
bchan=7-8
dchan=9
# Span 4: XBUS-00/XPD-18 Xorcom XPD #0/24: BRI_TE
span=4,4,1,ccs,ami
# termtype: te
bchan=10-11
dchan=12
# Global data
loadzone= au
defaultzone = au
Error recieved in console without group
-- Executing Dial(SIP/701-09f0fc18, zap/0418608609) in new stack
Jun 27 19:27:13 NOTICE[4011]: app_dial.c:1089 dial_exec_full: Unable to
create channel of type 'zap' (cause 34 - Circuit/channel congestion)
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/701-09f0fc18' status is 'CONGESTION'
Error recieved in console with g0 in the dial string
-- Executing Dial(SIP/701-08d76e98, zap/g0/0418608609) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g0/0418608609
-- Zap/1-1 is proceeding passing it to SIP/701-08d76e98
-- Channel 0/1, span 1 got hangup request
-- Channel 0/1, span 1 received AOC-E charging 0 units
Jun 27 19:28:35 WARNING[4046]: app_dial.c:738 wait_for_answer: Unable to
forward voice
-- Hungup 'Zap/1-1'
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/701-08d76e98' status is
'CHANUNAVAIL'
Nathan Dennis
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Direct: +61 (7) 4044
0302
124 Spence Street Fax:+61 (7) 4041 6600
CAIRNS QLD 4868Mobile: 0418 608609
Australia
E-mail: [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
Web Site: www.i-solutions.net.au http://www.i-solutions.net.au/
Offices and agents in Cairns - Brisbane - Melbourne -- Adelaide --
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[asterisk-users] Missing 'init keys' command
Hi, I have two new Asterisk installations (1.4.4 and 1.4.5) and I have created rsa keys and they can now see each other as online peers: moe*CLI iax2 show peers Name/UsernameHost Mask Port Status bart 192.168.2.201 (S) 255.255.255.255 4569 OK (48 ms) 1 iax2 peers [1 online, 0 offline, 0 unmonitored] but on the 1.4.5 instllation I cannot execute 'show keys' neither 'init keys' moe*CLI init keys No such command 'init keys' (type 'help' for help) This may be the reason that I cannot place calls from one Asterisk to the other. chan_iax2.c:7285 socket_process: I don't know how to authenticate moe to 192.168.2.201 thanks in advance, Jonathan. ___ --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] IAX trunking using a different port
Hi all, Is it possible having a trunk using, for exemple, UDP port 4570 and all the other IAX (not trunk) connection using the standard UDP port 4569? Thanks. Ronaldo. ___ --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] minibrowser for each snom phone
Hello, each snom phone is able to use services from standard web servers since firmware release 7.1.7. For further information go to http://snom.com/wiki/index.php/Xmlobjects cheers, Hirosh Dabui -- Hirosh Dabui snom technology AG Computer Engineering ___ --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] ISDN data-call question
Hi list, When this question came up, I realised how little I know about ISDN data calls (the sort used for ISDN video-conferencing etc), so I thought I would solicit pointers here. I have a requirement for an Asterisk-based system to connect to an ISDN30 line (using Sangoma hardware), and to present ISDN2 sockets, probably using a Xorcom BRI unit in NTE mode. So far so good, I believe I can build that. Assuming I do no transcoding, and disable echo cancelling, will an ISDN data connection survive being passed through asterisk? Both the ISDN30, and ISDN2 use Zaptel drivers, so the passthrough should in theory be fairly painless. Thanks for any feedback. Regards, Steve ___ --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] Self Calling test
I've had slew of problems with my Bell Canada Single Number Reach (SNR) dropping in the past couple of months. Another outage Monday for several hours has me wondering if there's a way to 1. Make a call out of my system via a PSTN back to my SNR line, say every 30 minutes (this I'm sure is easy enough via the call file...however...) 2. Track the outgoing call and match to an incoming call...if there's no incoming call...it means my Bell circuit or VoIP provider or something is down...send me an email that the service is down such that I can reroute my SNR to cellular. The whole point of this SNR was to give me mobility...though that came at a cost...Add the Voip off Asterisk and it's a near perfect solution except when this fails. From a network perspective, I've got dual hosted solution now to resolve network outages and recent tests have shown that works well, albeit the switch takes about 20 minutes to propagate the dns updates but otherwise flawless. It's embarrassing and I'm losing credibility when clients are asking if I'm still in business as the phone has dropped way to often in the past few month. Interesting enough all outages to date have been Fridays or Mondays. Does anyone else do anything like this. Anyone else using the Bell SNR service? Suggestions welcome. Thanks in advance Dave Bour Desktop Solution Center 905.381.0077 ___ --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] Zap dialling issues
On Wed, Jun 27, 2007 at 07:41:48PM +1000, Nathan Dennis wrote:
I'm having problems getting an Xorcom USB Bri 4 dialling out in
Australia.
First off, is layer 1 and 2 up? If you use a recent version of our
driver (see my previous message) this should be observed by a steady
single blink to the port's led (green, as the first two ports are TE).
If you're not sure, then what's the output of:
egrep 'Layer 1|D-Channel' /proc/xpp/XBUS-*/XPD-*/bri_info
I can receive calls into the system without an issue, but I can not for
the life of me dial out of the system.
Hmmm... this suggests that the problem is with chan_zap's configuration
(zapata.conf).
Below are my configs, I'm hoping
its something simple that I just can't see as I've been looking at it
for to long. Can any one point me in the right direction.
P.S. Yes it is meant to be in TE PTP mode as the current Digium B410P
works fine in that mode
/etc/asterisk/extensions.conf Extract
[internal]
include=features
include=speeddial
;Extention number for main Q
exten = 700,1,Goto(mainq,q,1)
;-
;Calling a local extensions mailbox
exten = _*7XX,1,Set(Extension=${EXTEN:1})
exten = _*7XX,n,Goto(directtovoicemail,s,1)
;--
;Static externaly accessable Conference room with recording
exten =
599,1,Set(MEETME_RECORDINGFILE=/var/spool/asterisk/monitor/conference-50
0-${EPOCH});
exten = 599,n,MeetMe(500,cMr,4081)
exten = 599,n,Hangup
;Dynamic Conference rooms for internal users to transfer callers to
exten = _5XX,1,MeetMe(${EXTEN},cMd)
exten = _5XX,n,Hangup
exten = 6000,1,Dial(zap/0418608609)
/etc/asterisk/zapata.conf
[channels]
Add here:
pridialplan=unknown
;echocancel = yes
;transfer = yes
callgroup=1
pickupgroup=1
; Span 1: XBUS-00/XPD-00 Xorcom XPD #0/0: BRI_TE
group=0,11
context=zapin
switchtype = euroisdn
signalling = bri_cpe
channel = 1-2
; Span 2: XBUS-00/XPD-08 Xorcom XPD #0/8: BRI_TE
group=0,12
context=zapin
switchtype = euroisdn
signalling = bri_cpe
channel = 4-5
; Span 3: XBUS-00/XPD-10 Xorcom XPD #0/16: BRI_TE
;group=0,13
;context=zapin
;switchtype = euroisdn
;signalling = bri_cpe
;channel = 7-8
; Span 4: XBUS-00/XPD-18 Xorcom XPD #0/24: BRI_TE
;group=0,14
;context=zapin
;switchtype = euroisdn
;signalling = bri_cpe
;channel = 10-11
/etc/zaptel.conf
# Autogenerated by /usr/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: XBUS-00/XPD-00 Xorcom XPD #0/0: BRI_TE
span=1,1,1,ccs,ami
# termtype: te
bchan=1-2
dchan=3
# Span 2: XBUS-00/XPD-08 Xorcom XPD #0/8: BRI_TE
span=2,2,1,ccs,ami
# termtype: te
bchan=4-5
dchan=6
# Span 3: XBUS-00/XPD-10 Xorcom XPD #0/16: BRI_TE
span=3,3,1,ccs,ami
# termtype: te
bchan=7-8
dchan=9
# Span 4: XBUS-00/XPD-18 Xorcom XPD #0/24: BRI_TE
span=4,4,1,ccs,ami
# termtype: te
bchan=10-11
dchan=12
# Global data
loadzone= au
defaultzone = au
Error recieved in console without group
-- Executing Dial(SIP/701-09f0fc18, zap/0418608609) in new stack
Jun 27 19:27:13 NOTICE[4011]: app_dial.c:1089 dial_exec_full: Unable to
create channel of type 'zap' (cause 34 - Circuit/channel congestion)
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/701-09f0fc18' status is 'CONGESTION'
Error recieved in console with g0 in the dial string
-- Executing Dial(SIP/701-08d76e98, zap/g0/0418608609) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g0/0418608609
-- Zap/1-1 is proceeding passing it to SIP/701-08d76e98
-- Channel 0/1, span 1 got hangup request
-- Channel 0/1, span 1 received AOC-E charging 0 units
Jun 27 19:28:35 WARNING[4046]: app_dial.c:738 wait_for_answer: Unable to
forward voice
-- Hungup 'Zap/1-1'
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/701-08d76e98' status is
'CHANUNAVAIL'
Nathan Dennis
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CAIRNS QLD 4868Mobile: 0418 608609
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Offices and agents in Cairns - Brisbane - Melbourne -- Adelaide --
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Re: [asterisk-users] Missing 'init keys' command
On 6/27/07, Jonathan Unai Marquez [EMAIL PROTECTED] wrote: but on the 1.4.5 instllation I cannot execute 'show keys' neither 'init keys' As of Asterisk 1.4, these commands have been standardized to the module verb format like most of the other CLI commands. In other words, the commands are now keys init and keys show. -Jared ___ --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] Wait to numbers
Hello everybody. I have a problem with my dialplan. That my extensions.conf: [incoming] exten = 943712666,1,Wait(2) exten = 943712666,2,Answer() exten = 943712666,3,Background(/home/lazkano/welcom) exten = 943712666,4,Wait(1) exten = 943712666,5,Background(/home/lazkano/extension) exten = 943712666,6,Wait(4) exten = 943712666,7,Dial(SIP/104|30|tm) exten = 943712666,8,Hangup() exten = 101,1,Dial(SIP/101|30|tm) exten = 102,1,Dial(SIP/102|30|tm) exten = 103,1,Dial(SIP/103|30|tm) exten = 104,1,Dial(SIP/104|30|tm) When someone call to the office the a recorded voice tell welcom, them an other record says if you know the extension, press it and wait 4 seconds. The problem is that in exten = 943712666,6,Wait(4) it doesn't take any naumber you must enter the extension in the exten = 943712666,5,Background(/home/lazkano/extension). There is an other command to wait 4 seconds and wait for numbers? Thanks for all. Enjoy your day. ___ --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] Help with IAX Trunk
Hi I've two servers : 1. UK 2. Pakistan Pakistan * server has ISDN30. Pakistan(ISDN30) UK === User Im planning to setup an IAX2 trunk between these two server ? so , how much bandwidth I need for 30 simul. calls ? Im planning to use G729 on both my server ? to support 30 calls over IAX2 do I've to change some setting during compile time or not ? pls suggest. thanks arun ___ --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] Round Robin SIP peers?
Hi all, I have a cheapskate customer whom wants to leverage some cheap all-you-can-eat VoIP connections rather than pay for a per minute provider. On the inbound side I think I have a solution in that I can activate the call forward on busy option with his provider (some noname white label house) but how do I balance his outgoing minutes? Is there some way that I can set up a round robin where each outgoing call goes out over a different line? If not is there some way that I can create a pool of lines such that when 2 folks make a call they get separate lines? Thanks Mark ___ --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] Problems compiling Asterisk 1.4.5
Hi! I have this errors compiling Asterisk 1.4.5 cdr_tds.c:86:2: warning: #warning You have older TDS, you should upgrade! cdr_tds.c: In function `tds_log': cdr_tds.c:213: error: too many arguments to function `tds_process_simple_query' cdr_tds.c: In function `mssql_connect': cdr_tds.c:326: error: `TDSCONNECTINFO' undeclared (first use in this function) cdr_tds.c:326: error: (Each undeclared identifier is reported only once cdr_tds.c:326: error: for each function it appears in.) cdr_tds.c:326: error: `connection' undeclared (first use in this function) cdr_tds.c:379: warning: implicit declaration of function `tds_free_connect' cdr_tds.c:393: error: `result_type' undeclared (first use in this function) cdr_tds.c:393: error: too many arguments to function `tds_process_simple_query' cdr_tds.c: In function `tds_load_module': cdr_tds.c:434: warning: unused variable `result_type' make[1]: *** [cdr_tds.o] Error 1 make: *** [cdr] Error 2 Any Ideas? ___ --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] Multi port IAX Gateway
On 6/26/07, Mike Hammett [EMAIL PROTECTED] wrote: I am looking for a gateway that has several FXS ports and uses IAX. I have a need for 16 ports, but will accept 6 or 8 port gateways as well. Here is an 8 port gateway that should suit your purposes: http://www.digium.com/en/products/hardware/asteriskappliance.php Unfortunately, I think they're only selling the developer's kits at the moment. I don't know when the retail version will be out. ___ --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] Problems compiling Asterisk 1.4.5
equis software wrote: Hi! I have this errors compiling Asterisk 1.4.5 cdr_tds.c:86:2: warning: #warning You have older TDS, you should upgrade! From yum search tds on FC5 freetds.i386 0.64-4.lvn5livna Matched from: freetds Implementation of the Sybase/Microsoft TDS (Tabular DataStream) protocol FreeTDS is a project to document and implement the TDS (Tabular DataStream) protocol. TDS is used by Sybase(TM) and Microsoft(TM) for client to database server communications. FreeTDS includes call level interfaces for DB-Lib, CT-Lib, and ODBC. http://www.freetds.org/ freetds-devel.i386 0.64-4.lvn5livna Matched from: freetds-devel Header files, libraries and development documentation for freetds This package contains the header files, static libraries and development documentation for freetds. If you like to develop programs using freetds, you will need to install freetds-devel. http://www.freetds.org/ make[1]: *** [cdr_tds.o] Error 1 make: *** [cdr] Error 2 cdr_tds.o ? looks like Asterisk is picking up an installed, but ancient, freetds and trying to build the cdr module for it. Update or remove freetds regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems compiling Asterisk 1.4.5
On 6/27/07, equis software [EMAIL PROTECTED] wrote: cdr_tds.c:86:2: warning: #warning You have older TDS, you should upgrade! This is trying to tell you that your TDS code is too old, and should be upgraded. In looking at the Asterisk code, it seems to need version 0.62 or newer of the TDS libraries on your system. -Jared ___ --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] kore dump
What is a god Windows application to read core dump files?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of J. Oquendo
Sent: Tuesday, June 26, 2007 4:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] kore dump
Vadim Berezniker wrote:
use the safe_asterisk script
it will restart asterisk if it crashes and it enables core dumps (your
core size limit is probably set to 0 when you start asterisk).
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of *Ed
Nuñez
*Sent:* Tuesday, June 26, 2007 2:22 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL PROTECTED]
*Subject:* [asterisk-users] kore dump
I am running Asterisk 1.4.5 and addons 1.4.1 in a CentOS 5 Server.
My PBX has experienced several core dumps the last couple of days and
I am not sure if this is whats causing it, but it always seems to
happen when a particular extension on a grandstream phone uses ChanSpy
SIP group.
I have not been able to locate where the core dump file is being
saved. I cant find it in my TMP directory.
I would also like to know if Asterisk can be setup to automatically re
start if there is a core dump. I was thinking of setting up a cron job
to launch Asterisk every minute. If its running, no harm done, and if
it crashes, the cron job will make sure that its started every 60
seconds.
Any suggestions?
Thank you
Ed Nuñez
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If that fails you could always try something like:
*/2 * * * * /bin/ps -C /usr/bin/asterisk || { /usr/bin/asterisk }
or so...
--
J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743
echo infiltrated.net|sed 's/^/sil@/g'
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to say something. -- Plato
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Re: [asterisk-users] Help with IAX Trunk
On 6/27/07, Arun Kumar [EMAIL PROTECTED] wrote: so , how much bandwidth I need for 30 simul. calls ? If you're using IAX2 trunking, the bandwidth requirements will be much less than if you're not using IAX2 trunking. Make sure you have trunk=yes in the peer definition in iax.conf. Off the top of my head (without actually running the numbers), I would guess that 30 simultaneous calls using the g.729 codec and using IAX2 trunking would take less than 512kbit/sec in each direction. to support 30 calls over IAX2 do I've to change some setting during compile time or not ? No, just make sure you have a suitable timing source (Digium card, ztdummy, etc.) for the IAX2 trunk. -Jared ___ --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] Wait to numbers
On 6/27/07, Josu Lazkano [EMAIL PROTECTED] wrote: There is an other command to wait 4 seconds and wait for numbers? Use the WaitExten() application instead of Wait(). -Jared ___ --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] Problems compiling Asterisk 1.4.5
Thanks Yes, I need an TDS upgrade. Right now I have installed freetds 0.64 and the errors change... [CC] cdr_tds.c - cdr_tds.o cdr_tds.c: In function `mssql_connect': cdr_tds.c:350: error: too many arguments to function `tds_alloc_context' make[1]: *** [cdr_tds.o] Error 1 make: *** [cdr] Error 2 On 6/27/07, Jared Smith [EMAIL PROTECTED] wrote: On 6/27/07, equis software [EMAIL PROTECTED] wrote: cdr_tds.c:86:2: warning: #warning You have older TDS, you should upgrade! This is trying to tell you that your TDS code is too old, and should be upgraded. In looking at the Asterisk code, it seems to need version 0.62 or newer of the TDS libraries on your system. -Jared ___ --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] kore dump
On 6/27/07, Ed Nuñez [EMAIL PROTECTED] wrote: What is a god Windows application to read core dump files? No. Core files must be examined on the same system that created them. ___ --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] Fax Throughput
You fixed your clocking then. That was what I was thinking of. Make sure that your Dialogic card is also pulling timing from the Digium card as well. What version of zaptel drivers are you running? --- Matthew Fredrickson Software Engineer Digium, Inc. On Jun 26, 2007, at 2:54 PM, Don Kelly wrote: Matt, thanks for pointing me to zaptel.conf; I'm new to Asterisk and can use all the help I get! Here are the non-comment lines from zaptel.conf (not set up by me): span=1,1,0,esf,b8zs span=2,1,0,esf,b8zs bchan=1-23 dchan=24 bchan=25-47 dchan=48 loadzone = us defaultzone=us The first span is connected to the PSTN. The second is connected to a Windows-based server using Dialogic hardware and custom software. The second span has a clock priority equal to the first one. I'm guessing that this has the effect of ignoring clock from the first span (same as '0') and using clock from the second. Not good. I've changed the clock priority for span 2 to '0'; if we lose the PSTN we'll rely on the Digium card for clock. Fax throughput seems fine with this change. In zapata.conf I find: ; Network Side signalling = pri_cpe group = 1 context = pstn-inbound channel = 1-23 ; IVR Side signalling= pri_net group = 2 context = ivr-inbound channel = 25-47 The default would be switchtype=national, which is correct. I see that for 'signalling', 'group' and 'context' = has been used, rather than the = that I see in documentation. Does this matter? --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: Matthew Fredrickson [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 26, 2007 9:22 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax Throughput Can you post your zaptel.conf so we can verify your timing settings? --- Matthew Fredrickson Software Engineer Digium, Inc. On Jun 25, 2007, at 11:10 PM, Don Kelly wrote: I've tried timing faxes two ways: From a fax machine on a station port of an AltiGen PC/PBX served by an MCI PRI calling back into the same PRI and reaching a RightFax server on a station port behind the AltiGen. From the same fax machine on the same station port of the AltiGen PC/PBX served by the same MCI PRI calling a number on an XO PRI connected to an Asterisk system (Digium TE410P), dialing out on another channel on the same PRI back into the MCI PRI and reaching the RightFax server on the station port behind the AltiGen. extensions.conf includes: exten = 6122353002,1,dial(zap/g1/6122590773) Sending a one-page fax with moderate density (no graphics) takes almost five minutes longer going through the Asterisk server. Any suggestions? ___ --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] kore dump
On 6/27/07, Ed Nuñez [EMAIL PROTECTED] wrote: What is a god Windows application to read core dump files? The core files are meant to be read by the gdb debugger on the machine in which the crash happened, so that gdb can look at the debugging symbols in the code and the system libraries. A core file by itself is pretty useless, so I doubt anyone has written a Windows application to read core dump files. -Jared ___ --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] Wait to numbers
Thankyou Jared, that it! it works! 2007/6/27, Jared Smith [EMAIL PROTECTED]: On 6/27/07, Josu Lazkano [EMAIL PROTECTED] wrote: There is an other command to wait 4 seconds and wait for numbers? Use the WaitExten() application instead of Wait(). -Jared ___ --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] problem with one way audio
Jason Backshall wrote: Do you have CallProgress=yes in your zapata.conf? This one just bit me in the arse this morning. I set it to no and one-way audio went away. Have heard of issues similar to this - and whilst disabling callprogress may make that symptom disappear, it probably shouldn't be seen as a 'solution', as callprogress has it's place (disconnection detection, etc). Don, have any changed been made to your zapata.conf immediately before this issue started occuring? Jason. I thought that callprogress was highly experiemental according to the wiki. Not sure how recent that information is though. -- Warm Regards, Lee ___ --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] Problems compiling Asterisk 1.4.5
equis software wrote: Thanks Yes, I need an TDS upgrade. Right now I have installed freetds 0.64 and the errors change... [CC] cdr_tds.c - cdr_tds.o cdr_tds.c: In function `mssql_connect': cdr_tds.c:350: error: too many arguments to function `tds_alloc_context' make[1]: *** [cdr_tds.o] Error 1 make: *** [cdr] Error 2 On 6/27/07, *Jared Smith* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On 6/27/07, equis software [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: cdr_tds.c:86:2: warning: #warning You have older TDS, you should upgrade! This is trying to tell you that your TDS code is too old, and should be upgraded. In looking at the Asterisk code, it seems to need version 0.62 or newer of the TDS libraries on your system. -Jared ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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 The unfortunate thing is that freetds is not a package in CentOS, that being said, the source based install is fairly straight forward. ___ --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+squid
Hi, I've installed Asterisk 1.2.13, and it works ok, but I have some voip clients behind a squid proxy server, and this clients can't connect to the Asterisk server. I added the access lists which permit the voip ports through the proxy, but the clients can't connect. This access lists in squid.conf are: acl safe_ports port 5060 acl safe_ports port 4569 acl safe_ports port 5036 acl safe_ports port 2727 acl safe_ports port -20001 Have you any idea how can I solve this problem? rs ___ --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] Asterisk+squid
rozsa wrote: Hi, I've installed Asterisk 1.2.13, and it works ok, but I have some voip clients behind a squid proxy server, and this clients can't connect to the Asterisk server. I added the access lists which permit the voip ports through the proxy, but the clients can't connect. This access lists in squid.conf are: acl safe_ports port 5060 acl safe_ports port 4569 acl safe_ports port 5036 acl safe_ports port 2727 acl safe_ports port -20001 Have you any idea how can I solve this problem? rs ___ --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 You need to do 1 to 1 NAT on the ports to get them through. I would suggest using a sip proxy on the squid server. ___ --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] Asterisk to Cisco 2600 GW DTMF Not Working, Working now
On 6/26/07, JR Richardson [EMAIL PROTECTED] wrote:
Hi All,
I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router
with a PRI card in it, handing off to a PBX and vise verse. Calls in
and out are working fine except for DTMF from Asterisk to the 2600.
DTMF from the 2600 to Asterisk is fine.
Here are the Asterisk console warnings I get when I send DTMF from
Asterisk to the 2600:
== Forcing Marker bit, because SSRC has changed
Jun 26 17:53:52 WARNING[14248]: channel.c:2328 set_format: Unable to
find a codec translation path from ilbc to ulaw
Jun 26 17:53:52 WARNING[14248]: channel.c:2328 set_format: Unable to
find a codec translation path from ilbc to ulaw
Jun 26 17:53:52 WARNING[14248]: chan_sip.c:2555 sip_write: Asked to
transmit frame type 1024, while native formats is 4 (read/write = 4/4)
Jun 26 17:53:52 WARNING[14248]: channel.c:2693
ast_channel_make_compatible: No path to translate from
SIP/53061-92e0(4) to SIP/10.10.10.10-78fa(1024)
Jun 26 17:53:52 WARNING[14248]: channel.c:3520 ast_channel_bridge:
Can't make SIP/53061-92e0 and SIP/10.10.10.10-78fa compatible
Jun 26 17:53:52 WARNING[14248]: res_features.c:1381 ast_bridge_call:
Bridge failed on channels SIP/53061-92e0 and SIP/10.10.10.10-78fa
== Spawn extension (iaxtest, 2144466715, 3) exited non-zero on
'SIP/53061-92e0'
The call drops.
If I enable ILBC codec with Asterisk, here is what I get:
== Forcing Marker bit, because SSRC has changed
Jun 26 17:56:28 WARNING[14332]: codec_ilbc.c:175 ilbctolin_framein:
Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP
(160)?
Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
codec 122 received
Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
codec 122 received
Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
codec 122 received
Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
codec 122 received
Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
codec 122 received
Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
codec 122 received
Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
codec 122 received
Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
codec 122 received
Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
codec 122 received
The call continues with this error until I hang up.
I have been adjusting the dial-peer dtmf settings in the 2600 and have
tried all the dtmf settings in Asterisk.
Any guidance will be appreciated.
Thanks.
JR
--
JR Richardson
Engineering for the Masses
This was a self induced problem, after mocking up in the lab it seemed
to work fine, but my production system didn't. I debugged the RTP and
captured the DTMF tones between the working and not working setup and
noticed the production system was sending DTMF codec number [96] and
the lab system was sending DTMF codec number [101].
This was a result of adding [96] = {0, AST_RTP_DTMF}, to rtp.c in
effort to resolve errors I was getting when passing calls to a cisco
call manager. The errors went away, but now sends an invalid codec
number to the 2600 gateway, which drops the call. I took out that
codec number in rtp.c, recompiled and DTMF works fine now. I'm sure
my codec errors will come back but at least DTMF will work. I'd
rather purge error logs than not have DTMF.
JR
--
JR Richardson
Engineering for the Masses
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Re: [asterisk-users] Asterisk+squid
On 6/27/07, rozsa [EMAIL PROTECTED] wrote: Hi, I've installed Asterisk 1.2.13, and it works ok, but I have some voip clients behind a squid proxy server, and this clients can't connect to the Asterisk server. I added the access lists which permit the voip ports through the proxy, but the clients can't connect. This access lists in squid.conf are: acl safe_ports port 5060 acl safe_ports port 4569 acl safe_ports port 5036 acl safe_ports port 2727 acl safe_ports port -20001 Have you any idea how can I solve this problem? I usually pass VoIP traffic without it going through the proxy. It can be dangerous, but if you set up your rules right, it should be OK. The only real exposure is that other things can hop on those ports. But then again, the safe_ports has the same challenge... ___ --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] Problems compiling Asterisk 1.4.5
I fix the problem deleting cdr_tds.c I know this is not the solution but, It woks. Thanks On 6/27/07, Anthony Francis [EMAIL PROTECTED] wrote: equis software wrote: Thanks Yes, I need an TDS upgrade. Right now I have installed freetds 0.64 and the errors change... [CC] cdr_tds.c - cdr_tds.o cdr_tds.c: In function `mssql_connect': cdr_tds.c:350: error: too many arguments to function `tds_alloc_context' make[1]: *** [cdr_tds.o] Error 1 make: *** [cdr] Error 2 On 6/27/07, *Jared Smith* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On 6/27/07, equis software [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: cdr_tds.c:86:2: warning: #warning You have older TDS, you should upgrade! This is trying to tell you that your TDS code is too old, and should be upgraded. In looking at the Asterisk code, it seems to need version 0.62 or newer of the TDS libraries on your system. -Jared ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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 The unfortunate thing is that freetds is not a package in CentOS, that being said, the source based install is fairly straight forward. ___ --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] Nuance Buys Tegic from AOL for $265m
Dean Collins wrote: Nuance Communications has agreed to buy Tegic Communications, the developer of the T9 predictive text input software for mobile phones, from AOL for $265 million in cash. http://www.wirelessweek.com/article.aspx?id=149702 Article goes on to say T9 is in use on over 2.5billion phones – wow now that’s a patent worth filing. I've never really used the English version of T9, but the Chinese version sucks. There are several other similar input schemes which do a far better job. Steve ___ --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] inband DTMF for g729
Gary Chen wrote: Does anybody know why Asterisk does not support inband DTMF for G.729? Our SIP carrier use inband dtmf for G.729. This causes problem for us to use it for our Asterisk IVR system. Any suggestion to solve this problem? I supposed the basic why is nobody has done it. G.729 spoils the quality of DTMF, and detection reliability degrades a bit, but not that much. I can put a DTMF test stream through G.729 and my decoder gets almost the same results as feeding the test stream direct to the decoder. I've never tried this, but it looks like you could do reasonable DTMF decoding from the G.729 parameters directly, without decoding to linear data at all. That might avoid patent licencing for this task. So, with some effort, and possibly with some patent licencing it could kinda work. On the other hand, the industry standard approach is to avoid this completely, and use RFC4733 (used to be RFC2833), which avoids any degradation in performance, and is simple to implement. Steve ___ --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] kore dump
What is a god Windows application to read core dump files? Microsoft jokes aside, I would seriously doubt there could be a good Windows application for analyzing core dumps. Due to the OS specific nature of core dumps, the need to have the source files, debugger and more, would make it difficult. I'm not saying there isn't one, I've just never heard of one. When developing software modules, I've had some success using crash on Fedora systems. Though as a whole system, a review of the logs to see what was changed just prior to getting the core dumps has been more effective at isolating the problem than the analysis of the core dump. John ___ --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] Using MSAccess to dial on a Zap line
Hello, We use MS Access 2000 (I know, we're migrating away from it) as an application to automatically dial phone numbers. The old phone system we have allowed the call representative using the application to take their phone off hook, push a button in the app, and the app would send the phone number to the phone system and dial the number. We are moving to Asterisk for our main phone system. A middleware program has been written to watch for dial events in a database, then the program calls the Zap station the call rep is at using the manager interface. Is there a better way to do this? The complaint we are getting now is the call rep doesn't want their phone to ring when making a call. Can the manager interface give a phone number to dial on an off hook Zap line? Thanks! -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road, Building 1, Rochester, NY 14624 Office: 888-865-0065 Ext. 202 Mobile: (585) 721-8679 ___ --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] Self Calling test
I see three parts to this if I was doing it. 1) set up an extension that, when dialed, requests a huge pin number. upon successfull pin number entry, it 'touch'es a file on the server to update its modification time [internal] ; could be extension to update heartbeat, asks for pin next exten = ,1,WaitExten(60) exten = ,2,Hangup exten = 12345679,1,System(touch /etc/such_and_such) exten = 12345679,2,Hangup 2) a cron entry that checks the modification time on that file every 10 minutes or so and sends an email if the modification time is more than 30 or so minutes old. You could use the 'find' command to find files matching that name and the criteria of less than 30 minutes age. If a file matching this criteria isn't found (it's there but too old), generate the email you desire. 3) a call file every 29 minutes that dials out and back into the above extension you created. could be as simple as using 'w' in the dial string to dial the pin at the right time. Notice I put WaitExten(60) up there, that should accommodate just about anything :) Dial(ZAP/g1/18005551212www12345679 The reason I put both AND 12345679 as extensions in the same context might seem redundant, as you could simply dial 12345679 from the main menu, but you don't necessarily want callers to the IVR to have a 60-second pause before repeating the IVR for example, so pressing would extend the wait to 60 seconds... If your main IVR uses Background() application solely, though, and lasts for a while, you might just skip the part. Anyway, just an idea. Let me know if I can clarify.. :) Moj P.S. Just thought of a cleaner way, skipping the filesystem overhead. Every time the pin number extension is dialed, set an asterisk db entry that contains the current second in the epoch. then, regularly use cron to check the stored astdb epoch value and compare it to the current one. Might be cleaner, and we've got that new STRFTIME() or whatever-it-is to help us (memory failing now) Moj some more Dave Bour wrote: I've had slew of problems with my Bell Canada Single Number Reach (SNR) dropping in the past couple of months. Another outage Monday for several hours has me wondering if there's a way to 1. Make a call out of my system via a PSTN back to my SNR line, say every 30 minutes (this I'm sure is easy enough via the call file...however...) 2. Track the outgoing call and match to an incoming call...if there's no incoming call...it means my Bell circuit or VoIP provider or something is down...send me an email that the service is down such that I can reroute my SNR to cellular. The whole point of this SNR was to give me mobility...though that came at a cost...Add the Voip off Asterisk and it's a near perfect solution except when this fails. From a network perspective, I've got dual hosted solution now to resolve network outages and recent tests have shown that works well, albeit the switch takes about 20 minutes to propagate the dns updates but otherwise flawless. It's embarrassing and I'm losing credibility when clients are asking if I'm still in business as the phone has dropped way to often in the past few month. Interesting enough all outages to date have been Fridays or Mondays. Does anyone else do anything like this. Anyone else using the Bell SNR service? Suggestions welcome. Thanks in advance Dave Bour Desktop Solution Center 905.381.0077 ___ --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] Round Robin SIP peers?
On Wed, 27 Jun 2007, Mark Phillips wrote: Is there some way that I can set up a round robin where each outgoing call goes out over a different line? If not is there some way that I can create a pool of lines such that when 2 folks make a call they get separate lines? This might be possible to do with DUNDi, but I don't know a lot about it. It seems to me that by far the most promising option is to build a persistent in-memory or AstDB variable into your dial plan and rotate it on each outbound call leg using some conditional logic that checks for proceeding calls originated from that particular customer, and increments/resets as needed. And, of course, the most elegant option at your disposal is probably to employ AEL: http://www.voip-info.org/wiki/view/Asterisk+AEL ... or AGI, which both will allow you to build all the outboard logic you want into the dial plan execution process. Cheers, -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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] inband DTMF for g729
On Thu, 28 Jun 2007, Steve Underwood wrote: So, with some effort, and possibly with some patent licencing it could kinda work. On the other hand, the industry standard approach is to avoid this completely, and use RFC4733 (used to be RFC2833), which avoids any degradation in performance, and is simple to implement. I would also be curious to know when and where there is ever a compelling reason to use inband DTMF representation with any codec, in any set of circumstances, barring one in which some intermediate element does not support out-of-band RFC 4733 events. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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] Customized Ring Tone
Hello all,
I'm running Asterisk 1.4.5 and Zaptel 1.4.3 on Debian Etch i386 with the
Digium's Dev Kit that comes with 1 FXO and 1 FXS. How do I configure my
home PBX in such a way that whenever someone calls on my trunkline (PSTN)
number, he/she will hear a customized ring tone, probably playing an MP3
file, instead of a boring standard ring tone while the extension number that
is forwarded the call is still ringing? My current
/etc/asterisk/extensions.conf file looks like this:
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
[pstn]
exten = s,1,NoOp(Caller ID is ${CALLERID(num)})
exten = s,2,Dial(Zap/1,15,g2)
exten = s,n,Congestion
[local]
ignorepat = 9
exten = _9.,1,Dial(Zap/g1/${EXTEN:1})
exten = _9.,n,Congestion
exten = 11,1,Dial(Zap/1,20,rt)
Thank you in advance.
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Re: [asterisk-users] Using MSAccess to dial on a Zap line
On Wed, 27 Jun 2007, Jason Martin wrote: Is there a better way to do this? The complaint we are getting now is the call rep doesn't want their phone to ring when making a call. Can the manager interface give a phone number to dial on an off hook Zap line? You would need to somehow associate the off-hook Zap channel with the particular user in the middleware layer. In principle, there is not a good way to reimplement what you are suggesting, because what you are suggesting appears to carry the logical implication that Zap can somehow dial on *behalf* of the phone station while it is off-hook, or for that matter on-hook, without calling back to the station in order to bridge the call after there is progress on the far end. I know of no way to do this. What you might be able to do to ease your complaints, though--assuming this is not already how it's implemented--is to call the destination number first, and only call the originating Zap station back once progress/ringing on the far end has begun. The events in the Manager API should provide for this, I think. Otherwise, this type of click-to-call functionality doesn't work so well with analog phones. If you had IP handsets you might be able to find a way to cleverly RPC a call origination request into an actual handset, or at least exert finer control (at the SIP level) over how and when the agent phone is put into the loop of the call scenario with ringback suppression, custom indicator lights, and things like that. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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] Customized Ring Tone
Add an Answer and add a m option to your dial command. They will hear
your music on hold until you answer.
Alex
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of GNUbie
Sent: Wednesday, June 27, 2007 12:18 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Customized Ring Tone
Hello all,
I'm running Asterisk 1.4.5 and Zaptel 1.4.3 on Debian Etch i386 with the
Digium's Dev Kit that comes with 1 FXO and 1 FXS. How do I configure my
home PBX in such a way that whenever someone calls on my trunkline
(PSTN) number, he/she will hear a customized ring tone, probably playing
an MP3 file, instead of a boring standard ring tone while the extension
number that is forwarded the call is still ringing? My current
/etc/asterisk/extensions.conf file looks like this:
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
[pstn]
exten = s,1,NoOp(Caller ID is ${CALLERID(num)})
exten = s,2,Dial(Zap/1,15,g2)
exten = s,n,Congestion
[local]
ignorepat = 9
exten = _9.,1,Dial(Zap/g1/${EXTEN:1})
exten = _9.,n,Congestion
exten = 11,1,Dial(Zap/1,20,rt)
Thank you in advance.
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[asterisk-users] Has anyone sucessful Asterisk to an Avaya IP phone
I have as large customer that would like to repalce all their Avaya PBXs with a OpenSer/Asterisk solution. Now the plan is to replace their 12,000 Avaya desk sets with Polycoms. My question is has anyone successfully used with OpenSer and or Asterisk, if so I would like to talk with you about hiring you to build this in our lab envirnment. Bob G. [EMAIL PROTECTED] -- We've Got Your Name at http://www.mail.com! Get a FREE E-mail Account Today - Choose From 100+ Domains ___ --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] Has anyone sucessful Asterisk to an Avaya IP phone
I have as large customer that would like to repalce all their Avaya PBXs with a OpenSer/Asterisk solution. Now the plan is to replace their 12,000 Avaya desk sets with Polycoms. My question is has anyone successfully used with OpenSer and or Asterisk, if so I would like to talk with you about hiring you to build this in our lab envirnment. Bob G. [EMAIL PROTECTED] -- We've Got Your Name at http://www.mail.com! Get a FREE E-mail Account Today - Choose From 100+ Domains ___ --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] QueueMetrics 1.4 released today
Hello list, After a few months of developement, we are proud to release QueueMetrics 1.4. This release adds a very large number of new features and bug fixes, for example: - New master engine! It should be 4x faster and 2x as memory efficient as QM 1.3, though it's tracking much more information. It's 100% compatible witth the old configuration switches. - New clustering engine! clustering is now fully supported for historical, live and agent's page data. The configuration is a bit different from version 1.3.3 - New Agent's page: an agent can log on, log off, go to pause and terminate pauses. - Call codes tracking: your agent can associate a call code to each incoming/outgoing call and QM will report on it. - Pause codes: an agent can mark WHY he goes on pause from the ACD, and QM will report on it. - Multi-stint calls: if a call has been handled by multiple queues, eg it has been passed to an overflow queue, it is now possible to track its progression ...plus over 50 other improvements, bugs fixed and little improvements - see the changelog file. In total, this release produces over 150 different results! You can download the latest version immediately from the downloads page at http://queuemetrics.com/download.jsp together with the updated 130-page User manual. As an alternative, if you run RHEL/CentOS/TrixBox/AAH, you can install it automatically using yum - see the installation page at http://queuemetrics.com/install.jsp As you probably know, QueueMetrics is a commercial software but it is available free of charge for smaller systems / SOHOs / interested hackers. It is possible to request a free unlimited demo licence from http://queuemetrics.com/sendDemoLicence.jsp I am looking forward to your comments and feedback. Thanks l. -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk+squid
On 6/27/07, rozsa [EMAIL PROTECTED] wrote: Hi, I've installed Asterisk 1.2.13, and it works ok, but I have some voip clients behind a squid proxy server, and this clients can't connect to the Asterisk server. I added the access lists which permit the voip ports through the proxy, but the clients can't connect. This access lists in squid.conf are: acl safe_ports port 5060 acl safe_ports port 4569 acl safe_ports port 5036 acl safe_ports port 2727 acl safe_ports port -20001 Have you any idea how can I solve this problem? rs rs, Squid is an HTTP/HTTPS caching proxy server. It has nothing to do with any of the protocols used for Asterisk (except maybe for the new HTTP manager interface). I'm not really sure what you are trying to do but UDP SIP on port 5060 (the only one I recognize out of your list) will never pass through Squid. I think you need to setup IP masquerading... -- 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
[asterisk-users] OpenSer/Asterisk PBX solution
We have been working a OpenSer/Asterisk solution to replace our Avaya PBXs.The OpenSer is to provide scalability and the Asterisk to provide rich features.I know this has been many times for calling card platforms but I'm not sure if anyone has a good scalable solution they are using on their virtual PBX or in a CPE PBX environment?If so I would like to talk to them about buy their deploying, testing and buying their solution? Bob [EMAIL PROTECTED] -- We've Got Your Name at http://www.mail.com! Get a FREE E-mail Account Today - Choose From 100+ Domains ___ --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] Asterisk+squid
Squid (or any other HTTP proxy) cannot proxy VoIP traffic. rozsa wrote: Hi, I've installed Asterisk 1.2.13, and it works ok, but I have some voip clients behind a squid proxy server, and this clients can't connect to the Asterisk server. I added the access lists which permit the voip ports through the proxy, but the clients can't connect. This access lists in squid.conf are: acl safe_ports port 5060 acl safe_ports port 4569 acl safe_ports port 5036 acl safe_ports port 2727 acl safe_ports port -20001 Have you any idea how can I solve this problem? rs ___ --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] .call file
Hello All, Is there any way to pass additional parameters while calling AGI from *.call file? Channel: Local/[EMAIL PROTECTED] MaxRetries: 0 RetryTime: 15 WaitTime: 15 Application: AGI Data: recordvoice.php Something like Data: recordvoice.php?id=3453name=asterisk Cheers, Nitesh ___ --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] Error While Calling AGI
Hello All, Found some strange problem while Asterisk trying to call the AGI file. If I pick up the call on the first attempt, it will execute my AGI file properly. But if I don't pick up the call and let Asterisk call me again, it adds StartRetry next to my AGI file name. Which will cause the AGI to fail to execute. -- Attempting call on SIP/5181 for application AGI(recordvoice.php) (Retry 1) -- Attempting call on SIP/5181 for application AGI(*recordvoice.phpStartRetry: 3700 1 (1182971439)*) (Retry 2) -- Attempting call on SIP/5181 for application AGI(*recordvoice.phpStartRetry: 3700 1 (1182971439)*) (Retry 3) Channel SIP/08f39360 was answered. Launching AGI(*recordvoice.phpStartRetry*: 3700 1 (1182971439)) on SIP/08f39360 -- Launched AGI Script /var/lib/asterisk/agi-bin/*recordvoice.phpStartRetry: 3700 1 (1182971439)* -- AGI Script *recordvoice.phpStartRetry: 3700 1 (1182971439)* completed, returning 0 Can anyone help? By the way I am executing using *.call file. File make.call: - Channel: SIP/5181 MaxRetries: 3 RetryTime: 30 WaitTime: 15 Application: AGI Data: recordvoice.php Cheers, Nitesh ___ --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] .call file
You can certainly use variables in the call file that get passed to the AGI. SetVar: MyVar=44 jerry ___ --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] Has anyone sucessful Asterisk to an Avaya IP phone
Bob, We are on a similar assignment right now. Please contact me off-list if you would like to discuss how we might be helpful. Thanks, Bryan M. Johns Partner Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 http://www.sheltonjohns.com - Original Message - From: Bob Gibson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, June 27, 2007 12:25:15 PM (GMT-0500) America/New_York Subject: [asterisk-users] Has anyone sucessful Asterisk to an Avaya IP phone I have as large customer that would like to repalce all their Avaya PBXs with a OpenSer/Asterisk solution. Now the plan is to replace their 12,000 Avaya desk sets with Polycoms. My question is has anyone successfully used with OpenSer and or Asterisk, if so I would like to talk with you about hiring you to build this in our lab envirnment. Bob G. [EMAIL PROTECTED] - -- We've Got Your Name at Mail.com Get a FREE E-mail Account Today - Choose From 100+ Domains ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7941 localized menus with SIP firmware
On Tue, 2007-06-26 at 21:45 +0200, Olivier wrote: Hi, Has anyone met any success, installing localized (ie non-english) menus within SIP firmware enabled Cisco 7941 ? Those phones seem to be trying to download localized menus from Cisco Call Manager but as they are managed by an Asterisk server, I'm looking for a workaround. Any advice ? Regards Actually Cisco only sendx xml for certain things. It uses a modified SIP stack and it's native SCCP stack to provision button templates, softkeys, etc.. I did hours of packet captures to try and get the info, but it is embedded into the call control stack of their phones. If you read the chan_sccp code a bit, it has a few different button layout options, that are encoded in the SCCP driver and not xml files. I wish they would go to all config files, but I doubt they will... -Greg ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] .call file
Thanks Jerry, But how can I access the Set variable in my AGI file? Like I do for callerId $cidnum = $agi-request['agi_callerid']; Is there any for Set? Cheers, Nitesh Jerry Geis wrote: You can certainly use variables in the call file that get passed to the AGI. SetVar: MyVar=44 jerry ___ --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] .call file
Thanks Jerry, But how can I access the Set variable in my AGI file? Like I do for callerId $cidnum = $agi-request['agi_callerid']; Is there any for Set? Cheers, Nitesh I dont use that programming (php) - I use C. I ask the AGI printf(Get variable name\n\r); and if gives it back to me. use voip-info.org search for setvar and agi. Jerry ___ --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] North American voice BRI - Informal survey
Hi, folks: I remain intrigued by the gap in BRI implementation between North America and Europe, and I wanted to get feedback from the list members on the matter. I'm seriously considering making the leap in our office. In Europe, the idea that an office that does not have enough lines to justify PRI would use analog lines is perceived as technologically backwards, and yet that's what happens in offices all over North America all the time. Finding BRI interfaces for many North American key systems is difficult. And all this is in spite of the fact that carriers providing PRI can also provide BRI. The minimum partial PRI offered here is 10 channels. What if an office has only 4 lines? Voice BRI is scarcely advertised. In these parts, Telus does indeed offer it. (I had to know what I was looking for, though.) I did some inquiries about monthly fees. Here's what I was quoted for 2B+D voice service (all these prices are in Canadian dollars; 1 USD buys 1.05 CAD): 1 Year Contract $91.75 3 Years Contract $82.50 5 Years Contract $79.85 They are not keen on month-to-month, but I squeezed a price out of them. It was something like $110 a month (it was not in the formal quote ;) ). The calling features are packaged as one (for both channels). You can't mix and match. If I only want caller ID, I'm stuck with everything else, too. 1 Year contract $27.90 3 Years contract $27.30 5 Years contract $25.75 I think the month-to-month for this was $29.90. So, say we take a 1 year contract, with calling features: $119.65, before taxes (we'll ignore the installation fees for the sake of this analysis). Now, comparing this with our current arrangement for two lines, forward on busy on one and caller ID on both, it comes to $114.17 before taxes. If one were to go head with the 1 year contract, it's hardly worth the difference to do analog. Thoughts? Who here has used BRI in North America? And when you did, what interface hardware did you use? -Stephen- ___ --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] North American voice BRI - Informal survey
On Wed, 2007-06-27 at 14:32 -0600, Stephen Bosch wrote: Hi, folks: Snip Thoughts? Who here has used BRI in North America? And when you did, what interface hardware did you use? -Stephen- I grew up on BRI when the internet first started taking off here. All terminated into Ascend Pipeline 50 or 25 routers. Gave 2 B and dynamic 128Kb/s bandwidth. With that said, the equipment to provision BRI on a class 5 switch here is another story. If the building they are delivering to does not have the right DLC cards, etc - it is usually chaeper for them to send a DS1 and pull 2 analog channels from it, and that is why you see BRI more exxpensive. With fiber being deployed to most buildings (or at least RTs) nowadays, the line cards do not play a factor since the DLC has to already be there. At the telco I worked, it was our philosophy to put in a mux and split out analog before going BRI. Equipment was cheaper to maintain, and provisioners were not burdened with 2 channel isdn. Now we did sell a lot of DS1 and DS3 PRIs for modem service, etc ___ --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] North American voice BRI - Informal survey
Greg Oliver wrote: On Wed, 2007-06-27 at 14:32 -0600, Stephen Bosch wrote: Hi, folks: Snip Thoughts? Who here has used BRI in North America? And when you did, what interface hardware did you use? -Stephen- I grew up on BRI when the internet first started taking off here. All terminated into Ascend Pipeline 50 or 25 routers. Gave 2 B and dynamic 128Kb/s bandwidth. With that said, the equipment to provision BRI on a class 5 switch here is another story. If the building they are delivering to does not have the right DLC cards, etc - it is usually chaeper for them to send a DS1 and pull 2 analog channels from it, and that is why you see BRI more exxpensive. That's kind of a chicken-egg problem. If BRI isn't advertised/offered/encouraged, then who's going to buy the right cards? As for the cost: in the example I provided, the difference is barely there -- if I get two calling features for each line, I'm better off with BRI on a one year contract. (Of course, I haven't taken into account analog bundle pricing, which would bring the cost for features on analog lines down some. But again, I don't see the difference being very big. It all ends up concentrating on the same telco equipment anyway.) With fiber being deployed to most buildings (or at least RTs) nowadays, the line cards do not play a factor since the DLC has to already be there. At the telco I worked, it was our philosophy to put in a mux and split out analog before going BRI. Equipment was cheaper to maintain, and provisioners were not burdened with 2 channel isdn. Now we did sell a lot of DS1 and DS3 PRIs for modem service, etc Most businesses are using relatively modern PBXs now, so -- provided the appropriate module is installed in the system -- you should be able to run the BRI right to the customer premises. The other stuff you describe just sounds like a way of getting more line density out of older infrastructure. -Stephen- ___ --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] North American voice BRI - Informal survey
On Wed, Jun 27, 2007 at 03:49:57PM -0500, Greg Oliver wrote: With that said, the equipment to provision BRI on a class 5 switch here is another story. If the building they are delivering to does not have the right DLC cards, etc - it is usually chaeper for them to send a DS1 and pull 2 analog channels from it, and that is why you see BRI more exxpensive. I am told that AE (now AGmumble, I think) never *has* to date gotten a reliably working BRI card for the GTD-5, and there are *lots* of those out there still... 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 ___ --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] North American voice BRI - Informal survey
Voice BRI is scarcely advertised. In these parts, Telus does indeed offer it. (I had to know what I was looking for, though.) BRI is a service the telcos would like to forget about here in the US. We ordered it at the house because we're sufficiently near a radio station that we tend to get POTS interference, and I wanted the flexibility to do virtually anything with the lines, including X2 dialup inbound (remember X2? ;-) ). That was around the peak of the BRI craze here in the US. I did some inquiries about monthly fees. Here's what I was quoted for 2B+D voice service (all these prices are in Canadian dollars; 1 USD buys 1.05 CAD): 1 Year Contract $91.75 3 Years Contract $82.50 5 Years Contract $79.85 They are not keen on month-to-month, but I squeezed a price out of them. It was something like $110 a month (it was not in the formal quote ;) ). We're at something around $50 on M2M, but there was a fairly steep install (maybe $250?). It ends up being around $115/mo for the 2 BRI lines (4 channels total). The calling features are packaged as one (for both channels). You can't mix and match. If I only want caller ID, I'm stuck with everything else, too. 1 Year contract $27.90 3 Years contract $27.30 5 Years contract $25.75 I think the month-to-month for this was $29.90. Ick. Around here, SBC/Ameritech/ATT prefers you to order by package code. You can order a-la-carte but it is damn expensive. The package we selected included Caller-ID. Cheaper packages were also available, but did not include Caller-ID, or only included 1B, or only data service, or whatever. So, say we take a 1 year contract, with calling features: $119.65, before taxes (we'll ignore the installation fees for the sake of this analysis). Now, comparing this with our current arrangement for two lines, forward on busy on one and caller ID on both, it comes to $114.17 before taxes. If one were to go head with the 1 year contract, it's hardly worth the difference to do analog. Right, but you also have to ask yourself, do I like to punish myself? Do you want to be on the wrong end of the support equation when the line fails? You can't just call SBC repair. They'll say that you don't have SBC service. You then have to make sure you keep track of the ISDN group's number, and call them, and be prepared to wait an hour a shot to talk to someone. Do you want to be stuck with a service where you can't just plug in a normal test set to check for dialtone? Do you want to have to figure out what combination of service adapters is needed to make it all work? Do you want to deal with oddities and irregularities in how the service works and interfaces to your PBX? These are just *some* of the questions that pop to mind. You *do* get a gorgeous crisp clean signal like nothing you've ever heard before. But it is a lot of work. Thoughts? Who here has used BRI in North America? And when you did, what interface hardware did you use? Well, at the time, there was pretty much nothing that was considered to be reliably supported by Asterisk for NA BRI. I picked up an Adtran Atlas 550 with a 4BRI-U interface and an octal FXS, and I use the unit's built-in T1 network port to connect to an Asterisk box. This works nicely, except for the things for which it doesn't work nicely. The box is fundamentally being used as a BRI-PRI translator, but gives me some neat extras. The BRI ports can be configured to work as user or network, so I've got some of my legacy ISDN devices (Courier I-Modem, and some other various stuff) that I can have switched through the Asterisk box and have them work - all digital signal path :-) The Adtran, however, has some limitations. The nastiest has to do with the way it handles DN's. It always grabs the first DN on a BRI for the outbound caller-ID. Adtran says no plans to fix. There are also problems getting it to register correctly to handle more than one call per DN; I have had it working in the past, but now it is pretty reliably broken. It's really too damn bad because the Adtran seems to have so many nice capabilities. We don't use special calling features (aside from Caller-ID, which I do not really consider to be a calling feature) so no idea about any of the other stuff like 3way, etc. We do that on the Asterisk box. I wouldn't buy the Adtran solution again. It cost about $2500 total to get up and running, IIRC, with used eBay equipment, but the idea behind it is extremely attractive. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To
[asterisk-users] Bypass local dialplan and redirect INVITE
Hi, I think this maybe impossible, but still want to try. If an INVITE with different host go throught my Asterisk, I want it not look into local dialplan and forward the request. Some of you may suggest I use real sip proxy, but I need a stateful proxy doing this signal proxess. Incoming Asterisk or whateverdef.com abc.com | INVITE([EMAIL PROTECTED]) | | | || INVITE | | | |--| | || 301 (Call forwarding | | | to abc.com) | | |--| | |INVITE | |-| a regular stateless SIP proxy will forward this 301 all the way back to incoming, which I don't want to. Asterisk will treat first INVITE as local dial plan, even it's not in local domain. Still cann't satisfied me. ___ --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] North American voice BRI - Informal survey
A few years ago I had a Qwest BRI 2b+d because I could not get DSL (I was surprised to get this). I had it on a Cisco 800 series router and ppp-multilinked the two channels together to get whopping 128k plus two phones numbers. It was kinda neat in that the D channel would drop one of the links when I had an incoming phone call. It cost me about $170 a month and I was really happy when they dropped in a Lucent Stinger and I moved to ADSL. One other note: the line was rock solid and support happened to be really good. I suspect this is because there were only a few businesses with strange needs and other then that the equipment was left alone. ;-) -James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Greco Sent: Wednesday, June 27, 2007 4:43 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] North American voice BRI - Informal survey Voice BRI is scarcely advertised. In these parts, Telus does indeed offer it. (I had to know what I was looking for, though.) BRI is a service the telcos would like to forget about here in the US. We ordered it at the house because we're sufficiently near a radio station that we tend to get POTS interference, and I wanted the flexibility to do virtually anything with the lines, including X2 dialup inbound (remember X2? ;-) ). That was around the peak of the BRI craze here in the US. I did some inquiries about monthly fees. Here's what I was quoted for 2B+D voice service (all these prices are in Canadian dollars; 1 USD buys 1.05 CAD): 1 Year Contract $91.75 3 Years Contract $82.50 5 Years Contract $79.85 They are not keen on month-to-month, but I squeezed a price out of them. It was something like $110 a month (it was not in the formal quote ;) ). We're at something around $50 on M2M, but there was a fairly steep install (maybe $250?). It ends up being around $115/mo for the 2 BRI lines (4 channels total). The calling features are packaged as one (for both channels). You can't mix and match. If I only want caller ID, I'm stuck with everything else, too. 1 Year contract $27.90 3 Years contract $27.30 5 Years contract $25.75 I think the month-to-month for this was $29.90. Ick. Around here, SBC/Ameritech/ATT prefers you to order by package code. You can order a-la-carte but it is damn expensive. The package we selected included Caller-ID. Cheaper packages were also available, but did not include Caller-ID, or only included 1B, or only data service, or whatever. So, say we take a 1 year contract, with calling features: $119.65, before taxes (we'll ignore the installation fees for the sake of this analysis). Now, comparing this with our current arrangement for two lines, forward on busy on one and caller ID on both, it comes to $114.17 before taxes. If one were to go head with the 1 year contract, it's hardly worth the difference to do analog. Right, but you also have to ask yourself, do I like to punish myself? Do you want to be on the wrong end of the support equation when the line fails? You can't just call SBC repair. They'll say that you don't have SBC service. You then have to make sure you keep track of the ISDN group's number, and call them, and be prepared to wait an hour a shot to talk to someone. Do you want to be stuck with a service where you can't just plug in a normal test set to check for dialtone? Do you want to have to figure out what combination of service adapters is needed to make it all work? Do you want to deal with oddities and irregularities in how the service works and interfaces to your PBX? These are just *some* of the questions that pop to mind. You *do* get a gorgeous crisp clean signal like nothing you've ever heard before. But it is a lot of work. Thoughts? Who here has used BRI in North America? And when you did, what interface hardware did you use? Well, at the time, there was pretty much nothing that was considered to be reliably supported by Asterisk for NA BRI. I picked up an Adtran Atlas 550 with a 4BRI-U interface and an octal FXS, and I use the unit's built-in T1 network port to connect to an Asterisk box. This works nicely, except for the things for which it doesn't work nicely. The box is fundamentally being used as a BRI-PRI translator, but gives me some neat extras. The BRI ports can be configured to work as user or network, so I've got some of my legacy ISDN devices (Courier I-Modem, and some other various stuff) that I can have switched through the Asterisk box and have them work - all digital signal path :-) The Adtran, however, has some limitations. The nastiest has to do with the way it handles DN's. It always grabs the first DN on a BRI for the outbound caller-ID. Adtran says no plans to fix. There are also problems getting it to register correctly to handle more than one call per DN; I have had it working in the past, but now it is pretty reliably broken. It's really too damn
[asterisk-users] Voicestick / i2telecom.com
Hello, I have been using Voicestick inbound (no outbound) successfully for the last few months. Noticed in my logfile that sip registration failed on 6/27/07 at 3am EDT and no successful registration since. Calls to my number eventually timeout as I don't have voicemail setup - as the first step in trouble shooting I tried to enable voicemail on the voicestick website but this fails also Transaction Failed. Please try again later. Nothing changed in my config. Asterisk 1.2.18. Can anyone confirm that there's an outage with Voicestick inbound? Huw ___ --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] Using MSAccess to dial on a Zap line
Look at TAPI driver for asterisk, here are some: http://www.snapanumber.com/ http://www.voip-info.org/wiki/view/Asterisk+TAPI http://www.thirdlane.com/outlookdialer.htm The last one being specific for Outlook, so I'm not sure if it's real TAPI or just an outlook add in. On 6/27/07, Jason Martin [EMAIL PROTECTED] wrote: Hello, We use MS Access 2000 (I know, we're migrating away from it) as an application to automatically dial phone numbers. The old phone system we have allowed the call representative using the application to take their phone off hook, push a button in the app, and the app would send the phone number to the phone system and dial the number. We are moving to Asterisk for our main phone system. A middleware program has been written to watch for dial events in a database, then the program calls the Zap station the call rep is at using the manager interface. Is there a better way to do this? The complaint we are getting now is the call rep doesn't want their phone to ring when making a call. Can the manager interface give a phone number to dial on an off hook Zap line? Thanks! -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road, Building 1, Rochester, NY 14624 Office: 888-865-0065 Ext. 202 Mobile: (585) 721-8679 ___ --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] Voicestick / i2telecom.com
The most obvious question first. Your account is paid up to date? Dave Bour Desktop Solution Center 905.381.0077 [EMAIL PROTECTED] For those who just want it to work... Giving you complete IT peace of mind. (Sent via Blackberry - hence message may be shorter than my usual verbose responses) PIN 4cc364db (as of March 24, 2007) - Original Message - From: [EMAIL PROTECTED] [EMAIL PROTECTED] To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com Sent: Wed Jun 27 19:21:40 2007 Subject: [asterisk-users] Voicestick / i2telecom.com Hello, I have been using Voicestick inbound (no outbound) successfully for the last few months. Noticed in my logfile that sip registration failed on 6/27/07 at 3am EDT and no successful registration since. Calls to my number eventually timeout as I don't have voicemail setup - as the first step in trouble shooting I tried to enable voicemail on the voicestick website but this fails also Transaction Failed. Please try again later. Nothing changed in my config. Asterisk 1.2.18. Can anyone confirm that there's an outage with Voicestick inbound? Huw ___ --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] North American voice BRI - Informal survey
Thanks for the response, Joe. Joe Greco wrote: Voice BRI is scarcely advertised. In these parts, Telus does indeed offer it. (I had to know what I was looking for, though.) BRI is a service the telcos would like to forget about here in the US. We ordered it at the house because we're sufficiently near a radio station that we tend to get POTS interference, and I wanted the flexibility to do virtually anything with the lines, including X2 dialup inbound (remember X2? ;-) ). That was around the peak of the BRI craze here in the US. Yeah -- as I mentioned, it's not like the Canadian telcos are announcing it from the rooftops, either. I did some inquiries about monthly fees. Here's what I was quoted for 2B+D voice service (all these prices are in Canadian dollars; 1 USD buys 1.05 CAD): 1 Year Contract $91.75 3 Years Contract $82.50 5 Years Contract $79.85 They are not keen on month-to-month, but I squeezed a price out of them. It was something like $110 a month (it was not in the formal quote ;) ). We're at something around $50 on M2M, but there was a fairly steep install (maybe $250?). It ends up being around $115/mo for the 2 BRI lines (4 channels total). Wow, that's cheap. No wonder you don't get any customer service. I couldn't even get analog lines for that price. The calling features are packaged as one (for both channels). You can't mix and match. If I only want caller ID, I'm stuck with everything else, too. 1 Year contract $27.90 3 Years contract $27.30 5 Years contract $25.75 I think the month-to-month for this was $29.90. Ick. Around here, SBC/Ameritech/ATT prefers you to order by package code. You can order a-la-carte but it is damn expensive. The package we selected included Caller-ID. Cheaper packages were also available, but did not include Caller-ID, or only included 1B, or only data service, or whatever. Sorry -- I think I was wrong there. I think caller ID is always included -- but we need forward on busy, which is a calling feature, so it means we need the features package. On the regular analog lines, the caller ID is extra (nine bucks! crooks!). I suspect it's very difficult to configure this equipment, so they just throw the whole thing at you. So, say we take a 1 year contract, with calling features: $119.65, before taxes (we'll ignore the installation fees for the sake of this analysis). Now, comparing this with our current arrangement for two lines, forward on busy on one and caller ID on both, it comes to $114.17 before taxes. If one were to go head with the 1 year contract, it's hardly worth the difference to do analog. Right, but you also have to ask yourself, do I like to punish myself? Do you want to be on the wrong end of the support equation when the line fails? You can't just call SBC repair. They'll say that you don't have SBC service. You then have to make sure you keep track of the ISDN group's number, and call them, and be prepared to wait an hour a shot to talk to someone. I know what you mean. This is the kind of headache you get on fibre connections with Telus. However, the PRI and BRI are handled by the same advanced business services group here. I have no personal experience with BRI, but judging by the ubiquity of PRI, it shouldn't suck too horribly. Of course, that could just be my youthful optimism talking. How often have your lines failed? Do you want to be stuck with a service where you can't just plug in a normal test set to check for dialtone? Do you want to have to figure out what combination of service adapters is needed to make it all work? Do you want to deal with oddities and irregularities in how the service works and interfaces to your PBX? These are just *some* of the questions that pop to mind. Oof. You *do* get a gorgeous crisp clean signal like nothing you've ever heard before. But it is a lot of work. This is what is so tantalizing about it. I also like the call progress information. Well, at the time, there was pretty much nothing that was considered to be reliably supported by Asterisk for NA BRI. I picked up an Adtran Atlas 550 with a 4BRI-U interface and an octal FXS, and I use the unit's built-in T1 network port to connect to an Asterisk box. This works nicely, except for the things for which it doesn't work nicely. The box is fundamentally being used as a BRI-PRI translator, but gives me some neat extras. The BRI ports can be configured to work as user or network, so I've got some of my legacy ISDN devices (Courier I-Modem, and some other various stuff) that I can have switched through the Asterisk box and have them work - all digital signal path :-) The Adtran, however, has some limitations. The nastiest has to do with the way it handles DN's. It always grabs the first DN on a BRI for the outbound caller-ID. Adtran says no plans to fix. There are also problems getting it to register correctly to handle more
Re: [asterisk-users] Voicestick / i2telecom.com
I have one of their free pre-pay accounts i.e. no monthly charge. Still have the origianl $5 signup credit as I've never made an outbound call via voicestick. I only use the account for the inbound number. Maybe the inability to setup voicemail on the voicestick server is an indication that there is something wrong with my account. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Bour Sent: Wednesday, June 27, 2007 19:46 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Voicestick / i2telecom.com The most obvious question first. Your account is paid up to date? Dave Bour Desktop Solution Center 905.381.0077 [EMAIL PROTECTED] For those who just want it to work... Giving you complete IT peace of mind. (Sent via Blackberry - hence message may be shorter than my usual verbose responses) PIN 4cc364db (as of March 24, 2007) - Original Message - From: [EMAIL PROTECTED] [EMAIL PROTECTED] To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com Sent: Wed Jun 27 19:21:40 2007 Subject: [asterisk-users] Voicestick / i2telecom.com Hello, I have been using Voicestick inbound (no outbound) successfully for the last few months. Noticed in my logfile that sip registration failed on 6/27/07 at 3am EDT and no successful registration since. Calls to my number eventually timeout as I don't have voicemail setup - as the first step in trouble shooting I tried to enable voicemail on the voicestick website but this fails also Transaction Failed. Please try again later. Nothing changed in my config. Asterisk 1.2.18. Can anyone confirm that there's an outage with Voicestick inbound? Huw ___ --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] Transfer Call to Cell Phone
John Faubion wrote:
We do have full features on our lines so both lines are free once the
transfer is complete. We also have toll calls on our lines so it would
not be a problem, so I do not have to worry about ATT dropping the
The issue really isn't whether you have the ability to make toll calls on
your line. The concern here is in what the regulatory agencies call toll
bridging which is using a system to relay a call from one local calling are
to another local calling area to avoid a toll charge. This is one of those
gray areas that can become a problem if your not careful. The problem comes
up if you have customers that can call you as a local call and you are
forwarding them on to another party that is a local call for you but would
be a toll call for the customer. This is essentially what toll bridging is
about. Now your not likely to have to worry about the legal ramifications of
this since your merely connecting the customer with an extension of your
company, namely your salesman. Where this could become a problem for you
would be in transferring the customer using the same pots line. The reason
is that ATT is handling the transfer. When you transfer the call, it
essentially becomes a new call. The main difference is that you have
provided the called number. So the software in the Class 5 (End office)
switch, takes the number you provide and runs the call through its routing
translations (similar to the Asterisk dialing plan) and if it determines
that the destination number is outside the originators Local Area Transport
Area or LATA, then it will either drop the originator to a message that
says, You must first dial a 0 or 1 before calling this number or it may
deny the transfer allowing you to stay connected to the customer. Neither
one looks very professional. The only way around this would be to provide
another line or trunk to pass the call down. Now if your not in an
overlapping LATA this probably isn't an issue.
John you a right about the LATA I know I am in one LATA 536 or 538 for
eastern OK. But I do not know the LATA on the Wireless which is now ATT. So
I will keep a watch out for it. Thanks for the tip!
The only way I can get it to work is by have the call on the 1st
line then transfer it out on the 2nd line. After that is complete both
lines are free.
Are you saying that you are able to route a call from line 1 to line 2 and
have the call transfer, thus freeing the lines or that once the call
completes the lines are freed? I've never seen the first scenario. The
second scenario is the normal behavior.
I am saying here that I can transfer the call from line 1 to line 2 and once
I transfer off the asterisk box it frees the two phone lines. My whole
arguement was to find a solution for doing this automatically on the basises
of dial an extension which can just transfer it to the cell phone. So ext
4001 cell-1 ext 4002 cell-2 etc. I do not mid doing it manually. But thanks
for the help!
Can you give an example of creating an extension which points to a cell
phone. Secondly how can you have if no one answers an extension it dials
the cell number next. That maybe answered in the example.
In extensions.conf use something like this.
[global]
SIP-PROV = sip.urprovider.com
; Now set the call forward numbers
CFN21 = 551234 ; These are normally set in an external file
[internal]
exten = 21,1,Macro(stdext,${SIP/21},${CFN${EXTEN}})
[macro-stdext];
; Standard extension macro:
; ${ARG1} - Device(s) to ring
; ${ARG2} - Our call forward number
exten = s,1,Dial(${ARG1},10)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,GotoIf($[${LEN(${ARG2})}0]?s-CFWD,1)
exten = s-NOANSWER,2,Voicemail(${MACRO_EXTEN},u)
exten = s-BUSY,1,Voicemail(${MACRO_EXTEN},b)
exten = s-CFWD,1,Dial(SIP/[EMAIL PROTECTED],20)
exten = s-CFWD,2,Goto(s-NOANSWER,2)
exten = _s-.,1,Goto(s-NOANSWER,2)
exten = a,1,VoicemailMain(${MACRO_EXTEN})
There is more to this but this should show the basics of what we use. I
store my Call Forward Numbers (CFN) in an external file. This allow me to
update the file externally (currently with a web interface but as soon as I
get the prompts recorded it will be done with an IVR) and then just reload
the extensions to activate the new numbers. Also I using SIP for pretty much
everything. Our TDM400 doesn't even have modules, it's just there for
timing. However you should be able to convert the SIP calls to ZAP calls for
you use. The internal context is included in our default context. Dialing
extension 21 calls the stdext macro. This dials the local extension first.
If not answered after 10 seconds, we check to make sure we have a phone
number to send the call out with. If not we send it on to voice mail.
Otherwise we send it to the s-CFWD. The check listed here is a very
rudimentary check but again I hope you get the idea. Next we try the call to
the CFN. If not
Re: [asterisk-users] Transfer Call to Cell Phone
Ryan Goldberg wrote:
OCOSA ListAcc wrote:
Can you give an example of creating an extension which points to a cell
phone. Secondly how can you have if no one answers an extension it dials
the cell number next. That maybe answered in the example. I have the
system setup so it just dials out which ever line is not busy. Thanks!
I'm quite new to *, but I've got this in place in my first rendition, and
I'm pretty sure it does what you want:
exten = 101,1,Dial(SIP/${EXTEN},15,t)
exten = 101,n,Dial(Zap/4/12185551212,30,tpm)
exten = 101,n,VoiceMail([EMAIL PROTECTED])
exten = 101,n,Playback(vm-goodbye)
exten = 101,n,Hangup
caller dials extension 101. It first tries his desk for 15 seconds, then
it tries his cell over a zap channel (the 'p' turns on call screening),
then it finally hits voicemail. In our actual dialplan, the cell phone
call goes out over sip, so the line looks like this:
exten = 101,1,Dial(SIP/lesnet/12185551212,30,tpm)
Alternatively, the first line could be:
exten = 101,1,Dial(SIP/${EXTEN}Zap/4/12185551212,30,tpm)
which would dial both the desk and the cell at the same time...
See http://www.voip-info.org/wiki-Asterisk+cmd+Dial
Hope that helps.
Ryan
Great I will look it over this weekend and see if it works!!! Thanks!
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Re: [asterisk-users] North American voice BRI - Informal survey
Thanks for the response, Joe. n/p. I figure I'm probably one of a small number of people with such a taste for suffering at the hands of the telco. Yeah -- as I mentioned, it's not like the Canadian telcos are announcing it from the rooftops, either. We had some CLEC's offering it for a while. McLeod, I believe. Stopped. Wait, I think TDS still sells them. For business, at least. Competition. Ain't it grand. We're at something around $50 on M2M, but there was a fairly steep install (maybe $250?). It ends up being around $115/mo for the 2 BRI lines (4 channels total). Wow, that's cheap. No wonder you don't get any customer service. No, everyone else has problems with customer service too. The regulators periodically fine Ameritech for poor service, and then everything's fine for a little bit. Lather, rinse, repeat. I couldn't even get analog lines for that price. Heh. Ick. Around here, SBC/Ameritech/ATT prefers you to order by package code. You can order a-la-carte but it is damn expensive. The package we selected included Caller-ID. Cheaper packages were also available, but did not include Caller-ID, or only included 1B, or only data service, or whatever. Sorry -- I think I was wrong there. I think caller ID is always included Not here. -- but we need forward on busy, which is a calling feature, so it means we need the features package. On the regular analog lines, the caller ID is extra (nine bucks! crooks!). Right, that'd make it substantially more expensive here. I don't believe it doubles the cost, but something at least 50% higher, if my recollection serves. One of the secondary reasons for the BRI was that the cost of two phone lines worked out to be about the cost of the one BRI on this plan, until you noticed that the two phone lines still needed CID added on to them, making them a fair bit more expensive. I suspect it's very difficult to configure this equipment, so they just throw the whole thing at you. That's one of the problems with ISDN. So, say we take a 1 year contract, with calling features: $119.65, before taxes (we'll ignore the installation fees for the sake of this analysis). Now, comparing this with our current arrangement for two lines, forward on busy on one and caller ID on both, it comes to $114.17 before taxes. If one were to go head with the 1 year contract, it's hardly worth the difference to do analog. Right, but you also have to ask yourself, do I like to punish myself? Do you want to be on the wrong end of the support equation when the line fails? You can't just call SBC repair. They'll say that you don't have SBC service. You then have to make sure you keep track of the ISDN group's number, and call them, and be prepared to wait an hour a shot to talk to someone. I know what you mean. This is the kind of headache you get on fibre connections with Telus. However, the PRI and BRI are handled by the same advanced business services group here. I have no personal experience with BRI, but judging by the ubiquity of PRI, it shouldn't suck too horribly. Of course, that could just be my youthful optimism talking. How often have your lines failed? I think only once in well more than half a decade. Well, we've had times when the CO was unhappy and we needed to unplug the equipment for 10 minutes to get it back to a usable state. Three or four times. But only had to call once, I think. I should probably check. The problem is that when you need to make any changes, they want those run through the special services group too. So you want a PIC line freeze, eh, well, rot in phone hold hell. I think they stopped doing that. You *do* get a gorgeous crisp clean signal like nothing you've ever heard before. But it is a lot of work. This is what is so tantalizing about it. I also like the call progress information. Absolutely. There's no doubt that it has some great aspects. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ --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] Xorcom Bri 4 Port USB
Thanks Tzafrir, that did the trick.
But please note the that the bristuff patch from xorcom has broken links in it.
It can't download asterisk using the URL in the script. Easy enough to fix by
pointing to a known good URL.
From: [EMAIL PROTECTED] on behalf of Tzafrir Cohen
Sent: Mon 25/06/2007 7:09 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Xorcom Bri 4 Port USB
Hi
On Mon, Jun 25, 2007 at 06:38:37PM +1000, Nathan Dennis wrote:
Hi,
I'm having some trouble setting up a Xorcom Bri 4 port. I have compiled
asterisk and zaptel using the Bristuff bristuff-0.3.0-PRE-1y-g patches.
So I'm running zaptel-1.2.17.1 and asterisk-1.2.18.
The problem I'm having is that for one I get no LEDs showing if the unit
is in TE and NT mode (not a issue for me but may have some impact on
things) I have no errors in any logs I can see but once zaptel and
asterisk are started up I get a lots of warnings in asterisk such as
the following
What is the output of:
modinfo xpp | grep version
if this is something of the sort of 'r3495' then you indeed have an
older version of the driver where BRI support has not been matuire
enough and specifically leds display was not as it is today. In current
version (e.g: the one in zaptel 1.2.18/1.4.3) you will always see an
orange LED for NT or green led for TE on the port.
Please get the version of bristuff from:
http://updates.xorcom.com/astribank/bristuff/
http://updates.xorcom.com/astribank/bristuff/bristuff-0.3.0-PRE-1y-g-xr1.tar.gz
At least until we see a new version of bristuff.
and also see:
http://updates.xorcom.com/astribank/bristuff/INSTALL.html
Also, for the sake of those who will see the messages in a search:
Jun 25 18:18:07 WARNING[2833]: chan_zap.c:2662 pri_find_dchan: No D-channels
available! Using Primary channel 3 as D-channel anyway!
== Primary D-Channel on span 2 down
Jun 25 18:18:07 WARNING[2834]: chan_zap.c:2662 pri_find_dchan: No D-channels
available! Using Primary channel 6 as D-channel anyway!
== Primary D-Channel on span 3 down
Jun 25 18:18:07 WARNING[2835]: chan_zap.c:2662 pri_find_dchan: No D-channels
available! Using Primary channel 9 as D-channel anyway!
== Primary D-Channel on span 1 down
This message comes from chan_zap when a span is down. If a span has
pri_{cpe,net} signalling or bri_{cpe,net} signalling (bristuff BRI ptp)
then you'll get those messages for spans that are down. If the
signalling is bri_{cpe,net}_ptmp they'll be debug messages.
It errors for all for ports and makes no difference if I have the
ISDN cables connected or not. I want to run in ptp mode and
currently use a digium B410P card on the connections that work fine
so I know that the lines work and ptp is the correct mode.
Following are my configs. Any pointers you can give would be greatly
appreciated.
We are running Fedora 7.
Kernel Linux 2.6.21-1.3194.fc7 #1 SMP i686 i686 i386 GNU/Linux (Standard
Kernel with install)
Device has jumpers all set to TE mode.
/etc/init.d/zaptel.conf
# Span 1: XBUS-00/XPD-00 Xorcom XPD #0/0: BRI_TE
span=1,1,1,ccs,ami
# termtype: te
bchan=1-2
dchan=3
# Span 2: XBUS-00/XPD-08 Xorcom XPD #0/8: BRI_TE
span=2,2,1,ccs,ami
# termtype: te
bchan=4-5
dchan=6
# Span 3: XBUS-00/XPD-10 Xorcom XPD #0/16: BRI_TE
span=3,3,1,ccs,ami
# termtype: te
bchan=7-8
dchan=9
# Span 4: XBUS-00/XPD-18 Xorcom XPD #0/24: BRI_TE RED
#span=4,4,1,ccs,ami
# termtype: te
#bchan=10-11
#dchan=12
# Global data
loadzone= au
defaultzone = au
/etc/asterisk/zapata.conf
[channels]
; echocancel = yes
; transfer = yes
; threewaycalling = yes
#include zapata-channels.conf
/etc/asterisk/zapata-channels.conf
; Span 1: XBUS-00/XPD-00 Xorcom XPD #0/0: BRI_TE
callerid=asreceived
group=0
context=from-pstn
switchtype = euroisdn
signalling = bri_cpe
channel = 1-2
callerid=
group=
context=default
; Span 2: XBUS-00/XPD-08 Xorcom XPD #0/8: BRI_TE
callerid=asreceived
group=0
context=from-pstn
switchtype = euroisdn
signalling = bri_cpe
channel = 4-5
callerid=
group=
context=default
; Span 3: XBUS-00/XPD-10 Xorcom XPD #0/16: BRI_TE
callerid=asreceived
group=0
context=from-pstn
switchtype = euroisdn
signalling = bri_cpe
channel = 7-8
callerid=
group=
context=default
; Span 4: XBUS-00/XPD-18 Xorcom XPD #0/24: BRI_TE RED
;callerid=asreceived
;group=0
;context=from-pstn
;switchtype = euroisdn
;signalling = bri_cpe
;channel = 10-11
;callerid=
;group=
;context=default
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Using MSAccess to dial on a Zap line
On 6/27/07, Jason Martin [EMAIL PROTECTED] wrote: Hello, We use MS Access 2000 (I know, we're migrating away from it) as an application to automatically dial phone numbers. The old phone system we have allowed the call representative using the application to take their phone off hook, push a button in the app, and the app would send the phone number to the phone system and dial the number. We are moving to Asterisk for our main phone system. A middleware program has been written to watch for dial events in a database, then the program calls the Zap station the call rep is at using the manager interface. Is there a better way to do this? The complaint we are getting now is the call rep doesn't want their phone to ring when making a call. Can the manager interface give a phone number to dial on an off hook Zap line? Why not put the off hook zap lines into a meetme room and then as you're dialing lines out join them to the meetme room? That way the zap lines can stay off hook. That also leaves open the ability down the road to have a manager monitor/barge these conversations. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Customized Ring Tone
Hello Alex,
Does this mean that on my PSTN context, I will add the lines I inserted
below?
On 6/28/07, Alexander Lopez [EMAIL PROTECTED] wrote:
Add an Answer and add a m option to your dial command. They will hear
your music on hold until you answer.
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
[pstn]
exten = s,1,NoOp(Caller ID is ${CALLERID(num)})
exten = s,2,Answer()
exten = s,3,Dial(Zap/1,15,g2,m(music_file))
exten = s,n,Congestion
[local]
ignorepat = 9
exten = _9.,1,Dial(Zap/g1/${EXTEN:1})
exten = _9.,n,Congestion
exten = 11,1,Dial(Zap/1,20,rt)
Thank you.
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[asterisk-users] Any difference using * with Centos i386 and x86_64 ?
Hi Everyone, Im testing a ML100 G4 (Pentium D) Server from HP with a TDM400P from Digium. I just installed, with success, the following O.S. with Asterisk 1.4.5 1) Centos 4.4 2) Centos 4.5 3) Centos 5.0 Id like to receive a recommendation about whats S.O do you recommend install for Asterisk 1.4.5 a) Centos 4.x ou Centos 5.x b) I386 ou X86_64 Thanks in Advanced, André Lomonaco ___ --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] Ring/Off-hook in strange state 6
Digium says it is a problem with the telco line. And of course telco says problem is internal. I am going to call them back tomorrow and make sure that disconnect supervision is enabled. On 6/26/07, Alex Mcdowell [EMAIL PROTECTED] wrote: I don't have caller ID at all, not on the verizon side and usecallerid=no in zapata.conf. I do, however have the DSL on this line. I have a splitter and then I have a filter on the asterisk side. I am guessing this is the root of the problem. Thanks for any insight.-Alex On 6/26/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Daniel already pointed you in the right direction. I have seen this error many times, but it never causes a problem. Alex Mcdowell wrote: Can anybody at least point me in a direction?? On 6/25/07, Alex Mcdowell [EMAIL PROTECTED] wrote: I don't think my cards are bad, but maybe there is a problem with the one. It has been two weeks since I put my ticket in with Digium...and still no word. I am starting to get frustrated. On 6/22/07, Daniel Hazelbaker [EMAIL PROTECTED] wrote: Alex, I had this problem with a new TDM2400 card that we purchased. Specifically I would get that message and then it would pick up the ringing line AND the line next to it. Basically, lines 1 2 had been cross-linked somehow. After a few weeks of trouble-shooting with Digium tech support they cross-shipped me a new card and the problem (and that message) went away. Daniel Hazelbaker High Desert Church On Jun 22, 2007, at 1:22 PM, Alex Mcdowell wrote: HI I have two servers both of which get this message on one of the lines. Ring/Off-hook in strange state 6. The one server seems to be ok with it, but the other one when an extension picks up there is no one there and the incoming call keeps ringing. I tried to adjust the levels in wcfxo.c like someone had suggested, but it didn't do anything. I also upgraded zaptel to the latest. 1.2.18 and asterisk is running at 1.2.14. callprogess is set to no, as well as busydetect=no. This is a major problem since this box only has 1 other line, but at least it works. I can't seem to find much info on this issue. I can't believe others haven't run into it. I started a ticket with digium, but I guess they are pretty backed up. Here is what I am getting in the CLI: Thanks for any help -Alex -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 Jun 22 13:44:45 WARNING[24296]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 is ringing Jun 22 13:44:46 WARNING[24296]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 answered Zap/4-1 == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 Jun 22 13:44:57 WARNING[24303]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 is ringing Jun 22 13:44:58 WARNING[24303]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 answered Zap/4-1 == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 -- SIP/4125-09559118 is ringing ___ --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] voicemail.conf serveremail
Hello, I was wondering if there is a way to change the From address (not just the Return-Path) for voicemail notification emails in Asterisk. It looks like the serveremail directive in voicemail.conf just changes the Return-Path. I'm looking for something analogous to the -r option in mailx, for example. I need this since the mail server I'm using requires the sender to be on the system. Any advice would be appreciated. Thanks ___ --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] callback and bridge problem
Hi guys,
sorry for the long e-mail, i'm only trying to give as much information
as i think is relevant to my problem (console log, sip.conf and
extension.conf parts). I've sent this e-mail a couple of days ago, but
it bounced back today.
i've been practicing with callback for a while, but i'm at a dead end.
I hope somebody can help me to move on.
i have troubles getting two calls bridged together. Scenario is the
following:
- asterisk calls my cell via a SIP provider called neophone
- my cell rings, i pick up, and i find myself in:
[internal]
; callback is directed here
exten = s,1,WaitExten,50
include = voicemail-context
include = internal_extensions-context
include = dialout_prefix-context
because my call file looks like this:
Channel: SIP/[EMAIL PROTECTED]
Context: internal
Extension: s
Priority: 1
where 0620222 is my cell.
- after picking up, i dial 9520630111 where 952 is the dialing
prefix, 0630... is another cell. 952 is a prefix for another
registered account at the same provider (one account is allowed to place
one call at a time).
After this as you can see, the second number (..) is dialed.
However when i pick up the phone, the call hangs up.
This also happens when i use another prefix (another provider, even
PSTN) for the second call too.
The relevant part from asterisk console is at the end of this e-mail, i
don't really understand the warning messages.
- configs:
In sip.conf, the configuration for the two SIP accounts are:
register = 0621380:[EMAIL PROTECTED]
register = 0621381:[EMAIL PROTECTED]
[neophonex]
type=friend
host=sip.neophonex.hu
context=dialout_prefix-context
username=0621380
authname=0621380
fromuser=0621380
secret=password
callerid=0621380
fromdomain=sip.neophonex.hu
disallow=all
allow=alaw
allow=g723
dtmfmode=inband
nat=no
[neophonex-out]
type=friend
host=sip.neophonex.hu
context=dialout_prefix-context
username=0621381
authname=0621381
fromuser=0621381
secret=password
callerid=0621381
fromdomain=sip.neophonex.hu
disallow=all
allow=alaw
allow=g723
dtmfmode=inband
nat=no
extension.conf:
exten = _952.,1,Playback(kapcsolas,noanswer)
exten = _952.,n,Set(CALLERID(name)=0621380)
exten = _952.,n,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
I have tried every possible setting i know about, but still, when i call
outside, via 'turning around' in asterisk, both cells hung up when
answering the call. I have tried calling a regular landline phone
number but still hanging up.
Both accounts are valid, registered and have enough credit to dial
outside its voice network.
The only way the call does not hung up is when i dial extensions within
asterisk.
The asterisk log:
-- Called [EMAIL PROTECTED]
-- Call on SIP/neophonex-out-081a9cc0 left from hold
-- SIP/neophonex-out-081a9cc0 is making progress passing it to
SIP/neophonex-081ab240
[Jun 25 16:57:07] WARNING[18232]: chan_sip.c:11839
handle_response_invite: Re-invite to non-existing call leg on other UA.
SIP dialog '[EMAIL PROTECTED]'. Giving up.
-- Call on SIP/neophonex-out-081a9cc0 left from hold
-- SIP/neophonex-out-081a9cc0 answered SIP/neophonex-081ab240
-- Native bridging SIP/neophonex-081ab240 and
SIP/neophonex-out-081a9cc0
[Jun 25 16:57:10] WARNING[18232]: chan_sip.c:11839
handle_response_invite: Re-invite to non-existing call leg on other UA.
SIP dialog '[EMAIL PROTECTED]'. Giving up.
== Spawn extension (internal, 9520630111, 3) exited non-zero on
'SIP/neophonex-081ab240'
[Jun 25 16:57:10] NOTICE[18440]: pbx_spool.c:351 attempt_thread: Call
completed to SIP/[EMAIL PROTECTED]
Please help me to figure out why the calls are hung up.
Thanks
Adam
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[asterisk-users] Call transfer feature
Dear ALL I want to transfer call from one phone 2 another phone so this is asterisk feature or SIP Phone feature or endpoint feature how can i transfer phone call from to another phone Rgd Satish patel - Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase.___ --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] Updated Manual for Asterisk 1.4.x
Hello all, Anybody can point me to the right URL where I can read an updated manual for Asterisk 1.4.x? Thank you in advance. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
