Re: [Asterisk-Users] Insert pause in SIP String
Eric Wieling wrote: Erick Weber V. wrote: I'll Like to now how to insert a pause on a SIP string. I have a ATA 186 and a FXS => FXO converter so I will like to program a extension that can be dialed and it will dial the ATA extention, wait for dial tone and then dial the phone number. You cannot put pauses in any dial string in Asterisk except calls using ANALOG Zap or ANALOG Voicetronix ports. This isn't really an Asterisk problem, it's a protocol problem. You could hack something into Asterisk to work around the problem, but that's Non-Trivial Well SIP just forwards user name parts, it is not really aware that a user name you forward to a PSTN gateway really is a dial string. There's some work in the tel: url name space to standardize dial strings, and there's the good old set of Hayes commands, but I guess you should check the documentation for the FSX-FXO-converter to find out how to insert a pause. For the record, there's a difference betweeen dial strings and e.164 phone numbers. Dial strings are instructions on how to dial a phone number in a certain environment - "dial 9 and wait for dialtone for outside calls". /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP: missing "180 ringing"
On Wed, 2004-04-14 at 07:55, I wrote: > Did any of you ever experience missing "180 ringing" messages when > dialing from a sip agent through an asterisk box? Reading up some more on this it now appears to me this is RFC compliant behaviour as a 183 Session Progress is being sent instead. So, my question really is: Is there a way to force asterisk to answer with 180 ringing? Thilo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP: missing "180 ringing"
Did any of you ever experience missing "180 ringing" messages when dialing from a sip agent through an asterisk box? I tried both a Zap and another SIP channel for dialout. Using a ISDN zap channel the call setup would look like SIP-UA * PSTN | || |INVITE->>>|| | || |<<<-Trying|| | || | |SETUP->>| | || |<-Sess Prog| | || | |<<-SETUP ACK| | || | |<<-CALL PROC| | || | |<<-ALERTING | | || | |<<-CONNECT | | || | |CONNECT ACK>| | || |<<<-200 OK|| | || I would expect the incoming ALERTING message to trigger a 180 ringing message on the sip side. Some user agents appear to not indicate "Ringing" to the caller unless the see this message. So it would be very desirable to have those. This behaviour occurs with 0.7.2, the lastest stable and the last weeks unstable checkout. Has anybody seen this before? Thilo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Voicemail storage in DB
Matt White wrote: James H. Thompson wrote: Would it make any sense to store the voice mail formatted as a email msg in a Maildir directory structure. Then you could also retreive them with an email client. As an extension of this thought, how about going one step further and storing the voicemail on an imap server directly? It would remove the whole storage question and allow storing on a remote system. The tools in the WU-IMAP c-client package would be pretty useful... Two of my students are working on this very project right now. Some thoughts/caveats: One design constraint that is mostly enforced in the asterisk code is that it run standalone (I think the mpg123 dependency is the sole exception) and there was very strong sentiment that anything we do NOT require the installation of a whole IMAP suite. So that complicates things somewhere between a little and a lot. Basically the task is to design a maildir type entity that can be completely manipulated within the asterisk application itself. We're still not coding it heavily, but my sense is that the real gotcha is going to lie in the IVR access routines. They'll have to be mapped into IMAP-space, as I see things right now, and in looking at the code that's already there, that isn't going to be a trivial thing. There is also the split between the UW orientation (keep the files in the maildir owned by the user they're sent to) and the cyrus orientation (lock down the IMAP store and require all access to pass through a server agent). I think the maildir approach is the Correct One, but the path thither appears to be a least minorly studded with complexity. My HO, of course. Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Voicemail storage in DB
James H. Thompson wrote: Would it make any sense to store the voice mail formatted as a email msg in a Maildir directory structure. Then you could also retreive them with an email client. As an extension of this thought, how about going one step further and storing the voicemail on an imap server directly? It would remove the whole storage question and allow storing on a remote system. The tools in the WU-IMAP c-client package would be pretty useful... -- Matt White [EMAIL PROTECTED] Arts and Science Computer Labs University of Saskatchewan It sure is Monday... Ain't it a sin I've gotta work my way thru the week again. - Mark Chesnutt..."Sure Is Monday" ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Voicemail storage in DB
On Mon, Apr 12, 2004 at 02:19:23PM -0400, James H. Cloos Jr. spake thusly: > Yes, there would be. This is the same issue as using nfs mail spools > with maildir style storage. W/o locking there is no way to guarantee > that two servers do not create the same vm file on top of one another. The Maildir algorithm ensures unique filenames thus no locking is needed. It is safe to use Maildir over NFS. However, I don't understand why one would want to store the voicemails in a db anyway. A filesystem is a special case of a database designed to do just this. Store a pointer to the file in the db if you have to. -- Tracy Reed The attachment is a digital signature. http://copilotconsulting.com More info: http://copilotconsulting.com/sig pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] tcp/ip stack tweaks
On Mon, 12 Apr 2004, Scott Laird wrote: > I mean, even G.711 is only ~80 kbps (including overhead), so you > should be able to run hundreds of simultaneous conversations on 100 > Mbps Ethernet without running out of bandwidth. Disclaimer: I do not build networks for a living -- the following comes from years of working near people who do, as well as a more-than-passing interest in making networks work well. Usually the LAN is not the problem. Wide-area links (such as your Internet connection) are where attention must be paid. ~80kbps is not much traffic, sure, but making sure that the packets that comprise that flow are delivered in a timely manner -- and not swamped by someone streaming video or doing a Windows Update -- is the trick. Even a T1/E1-sized IP link to head-office will give you grief if you're trying to conduct low-latency traffic at the same time as the VP's assistant sends the 50MB all-staff Xmas-Party-Invite.doc via e-mail... :) The problem can be in buffer sizes. One trick that router admins used to use to keep WAN links busy and happy (particularly links with a high round-trip-time such as satellite) was to increase the length of the transmission queues and the transmit and receive windows -- increasing the number of packets that get lined up for transmission, and making it more likely that we've continued transmitting instead of waiting for acknowledgements. This will utterly destroy anything real-time, as any large transfer will fill up the longer queue all at once (you'll know what I mean if you've ever tried to use telnet or SSH over a link that's simultaneously filled up with a big FTP or HTTP download, or the above example e-mail). So, the trick was to keep the queue short enough to maintain some degree of interactiveness, yet long enough that the link is kept active. Nowadays, we can use QoS and priority queueing to put interactive stuff straight to the front of the queue -- but we still have to hope that our ISPs and carriers treat that the same as we do. (Speaking of window sizes, I've heard that for extreme performance it is not wise to mix TCP and UDP on the same interface because of the different ways that windows and buffers are sized with TCP and UDP... I don't know enough to confirm or refute this. I can imagine that TCP's method of setting the window size -- blast it out until it doesn't fit any more -- might cause a problem for a UDP-based RTP stream already in-flight... Comments?) Back to LANs: it is possible to see problems on very busy Ethernets. In unswitched networks, if utilisation is high, collisions are more likely (we don't call it Crashernet for nothing). This might be sufficient to give you some loss of quality, or at least start to exercise your jitter buffering. In switched networks, if your backbone is being overworked you may get packet loss etc. Neither of these situations are directly helped by QoS -- if your packet can't get on the wire, or gets thrown off later, a few bits in a header won't change that much (although some routers and switches may make traffic with identified QoS less likely to be thrown out -- kind-of like how the Discard Eligible bit is used in Frame Relay). Cheers, Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP->h323 problem DTMF
I've configured Asterisk 0.7.2 to work together with Cisco ATA186 (SIP,G.711. RFC2833)
and OpenPhone (H.323, G.711).
But there is an issue while calling from ATA186 to OpenPhone via Astrisk - when I
press any key on analogue phone connected to ATA, Asterisk shows following message:
-- Executing Dial("SIP/519-3781", "OH323/62.213.36.100|20|Tt") in new stack
-- Called 62.213.36.100
-- OH323/L4366 answered SIP/519-3781
1:36.475LogChanTx:8130bc0 PWLib Assertion fail: Invalid parameter,
file rtp.cxx, line 385, Error=22
bort, ore dump, gnore?
┘and connection becomes one-way style - voice transmits from OpenPhone only.
This problem doesn't appear while calling from OpenPhone to ATA186.
extensions.conf
-
[general]
static=yes
writeprotect=no
[globals]
[demo]
exten => s,1,Wait,1
exten => s,2,Answer
exten => s,3,Dial(SIP/519,20,Tt)
exten => s,4,Hangup
exten => s,104,Hangup
[default]
include => demo
[extensions]
exten => 100,1,Dial(OH323/xx.xx.xx.xx,xx,Tt)
exten => 100,2,Hangup
exten => 100,102,Hangup
exten => 102,1,Dial(SIP/519,20,Tt)
exten => 102,2,Hangup
exten => 102,102,Hangup
[local-access]
include => extensions
-
h323.conf
---
[general]
listenAddress=xx.xx.xx.xx,xx
listenPort=1720
connectPort=1720
tcpStart=1
tcpEnd=2
udpStart=1
udpEnd=2
fastStart=no
h245Tunnelling=no
h245inSetup=no
inBandDTMF=no
silenceSuppression=no
jitterMin=20
jitterMax=100
ipTos=none
outboundMax=10
inboundMax=10
simultaneousMax=10
wrapLibTraceLevel=1
libTraceLevel=0
libTraceFile=stdout
gatekeeper=DISCOVER
gatekeeperTTL=600
userInputMode=RFC2833
amaFlags=default
accountCode=H323
context=voip-h323
[register]
;
alias=asterisk
alias=123
;
; Aliases/prefixes routed in "all-aliases" context.
;
context=all-aliases
alias=ASTERISK
alias=666
;
; Aliases/prefixes routed in "more-aliases" context.
;
context=more-aliases
alias=665
;
; Aliases/prefixes routed in "all-prefixes" context.
;
context=all-prefixes
gwprefix=00
gwprefix=01
;
; Aliases/prefixes routed in "more-stuff" context.
;
context=more-stuff
alias=664
gwprefix=02
;-
; Specify and configure CODEC related
; options
;-
[codecs]
codec=G711A
frames=20
;codec=G711U
;frames=20
;codec=GSM0610
;frames=4
codec=G7231
;frames=2
;codec=G729
;frames=2
;codec=G7231
;frames=6
---
sip.conf
---
[general]
port = 5060 ; Port to bind to
bindaddr = xx.xx.xx.xx,xx ; Address to bind to
context=INVALID
tos=lowdelay
;disallow=all ; Disallow all codecs
;allow=ulaw ; Allow codecs in order of preference
trancfer=yes
threewaycalling=yes
[519]
type=friend
host=xx.xx.xx.xx,xx
context=local-access
reinvite=no
canreinvite=no
dtmfmode=RFC2833
qualify=300
callerid="ATA186" <519>
;mailbox=21
nat=no
[520]
type=friend
host=xx.xx.xx.xx,xx
context=local-access
reinvite=no
canreinvite=no
;dtmfmode=inband
qualify=300
callerid="x-lite" <520>
;mailbox=21
nat=yes
---
Pavel Riko
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RE: [Asterisk-Users] Vonage goes to .ca
Hopefully they won't mandate a specific amount of "Canadian content" per call. ;) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of TC Sent: Tuesday, April 13, 2004 7:40 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Vonage goes to .ca FYI http://www.itbusiness.ca/index.asp?theaction=61&sid=55298 did not like this by line in the story t"he CRTC has said it will likely regulate voice over IP the same as other phone services." ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T100P E&M Wink Trunk
Keep in mind that if this is a Local T1 you CAN have people calling you with
no callerid. Make sure to include that possibility in your dialplan:
With CallerID:
exten => _*XX*916222,1,Dial(
Without CallerID:
exten => _**916222,1,Dial(
MATT---
-Original Message-
From: Mike Machado [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 13, 2004 7:39 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] T100P E&M Wink Trunk
I am setting up a box with a T100P. Everything is going well. The
company I am working with has their one phone switch gear. They
provisioned me a E&M Wink T1. Cannot do PRI unfortunately. We chose E&M
so we could pass an unlimited number of DIDs to the trunk as apposed to
FXS loopstart signaling. I can make outbound calls no problem, but I am
having problems with the dial plan for inbound calls. The way they setup
the trunk inbound calls have a dialed number as
"**". I do not know how to parse this out and
map it in the dial plan. Are there substr functions I can use? Can I
just call SetCIDNum on an INBOUND call to get the callerid functions
working?
Here is what I see in the log when a call comes in:
-- Starting simple switch on 'Zap/24-1'
== Unknown extension '*916111*916222' in context 'default'
requested
-- Playing 'ss-noservice' (language 'en')
-- Hungup 'Zap/24-1'
916111 is the calling number (Caller ID)
916222 is the called number
What would be the best way to convert this so I can use just ${EXTEN}?
I can get it to work if I do something like:
exten => _*XX*916222,1,Dial(
but that seems like a hack, plus that does not set callerid. I also dont
know how that work react to callerid being blocked.
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[Asterisk-Users] DNID Digits - Australia
Hi, Yet another question, now that I have callerid working correctly, I'm trying to work out how to utilise the different numbers I have. I have a 100 number range allocated to my E1/PRI/OnRamp service. My incoming calls are handled like this: Advertised/published number is an analogue line terminating on a X101P. If the analog line is busy, it has a call diversion to the PRI on a TE405P Whenever I look at a channel while a call is in progress I see something like this: Call on the analog line: -- General -- Name: Zap/125-1 Type: Zap UniqueID: 1081902956.34 Caller ID: (N/A) DNID Digits: (N/A) State: Up (6) Rings: 1 NativeFormat: 68 WriteFormat: 4 ReadFormat: 4 1st File Descriptor: 28 Frames in: 1252 Frames out: 2500 Time to Hangup: 0 For a call on the PRI: -- General -- Name: Zap/3-1 Type: Zap UniqueID: 1081902207.29 Caller ID: 28304 DNID Digits: (N/A) State: Up (6) Rings: 1 NativeFormat: 72 WriteFormat: 4 ReadFormat: 4 1st File Descriptor: 20 Frames in: 5346 Frames out: 1734 Time to Hangup: 0 So, from the above I can see the callerid of the call (nice) but I don't know which one of the 100 possible numbers was dialled. Is this what *should* appear as the DNID? Does anyone know why I don't get this information showing up? BTW, when enabling pri intense debug span 1 I do see this info: < Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) < Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) < Ext: 1 User information layer 1: A-Law (35) < Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0
Re: [Asterisk-Users] CallerID in Australia
On Wed, 14 Apr 2004, Gary wrote: > they actually send the caller-id info after the SECOND ring. That depends. ;) If you define ring as 'single burst of ring voltage', then it does come after the second ring. That's how * will have to look for it, after all. It changes if you have Distinctive Ring, though; the first ring signal becomes a single burst (not sure the duration). The callerid is presented after that, and *then* the ring changes to the Distinctive Ring cadence. The patch that Duane mentioned was to try and get * to recognise the presentation of distinctive ring cadence *after* the CID data is detected. I don't doubt that there are problems with that patch -- it was my first attempt at *-hacking. > Now of course if the au indications were changed to combine the first > and second ring to appear as one ring, no other changes would be needed > ?? But then it would not sound like Australian ring any more ;) From what I can see, the ring cadence you define cannot have 'non-repeating' sections like you can do with tones in indications.conf (the 'bang' sections). Cheers, Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID in Australia
On Wed, 2004-04-14 at 00:52, Vic Cross wrote: > G'day Adam, > > This drove me nuts for a few days just recently (only fixed it yesterday > in fact, and I've not had a chance to update any doco anywhere yet). > > On Wed, 14 Apr 2004, Adam Goryachev wrote: > > > Actually, now that I look at the file again, I can also see: > > Line: 80 > > /* Typically, how many rings before we should send Caller*ID */ > > #define DEFAULT_CIDRINGS 1 > > Yes, that's one of the changes. Change this to > > #define DEFAULT_CIDRINGS 2 Great, that is the only thing I have changed, and it is now working nicely (except that I don't receive callerid on my analog line, as if I would want to pay extra for it!)... > > I am in Australia, which I think expects callerid at a different time to > > other countries Although other people have told me callerid is > > working correctly for them > > You're right, our callerid must wait a little longer after the first ring > burst (but otherwise is US Bellcore FSK). > > In callerid.c, on or about line 467, change the value 4000 to 5600 [1]. > Recompile, reinstall, shutdown and restart. > I have one phone here that works with standard Asterisk, so when another > phone did not work I thought it was the phone. Then a third phone did not > work, so I started looking for the problem. I think that early phones > were built to the US standard, and so will work with US callerid, but the > Australian standard was changed later so newer phones need the callerid > data to arrive later. I didn't need to do this for the phones that I have, but I'll try and remember that in case I buy different phones later... Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T100P E&M Wink Trunk
On Wed, 2004-04-14 at 10:32, Mike Machado wrote:
> I found ways to do substrings. So this is what I did. I changed the zap
> channels to come into a context called 'fixup' and then jump into
> default after doing the parsing and setting of CID.
>
> [fixup]
>
> exten => _*NXXNXX*NXXNXX,1,SetCIDNum(${EXTEN:1:10})
> exten => _*NXXNXX*NXXNXX,2,Goto(default|${EXTEN:-10:10}|1)
>
>
> So far things seem to be working well. Does this sound like a good way
> to deal with this problem?
This assumes that the strings are fixed lengths. If you can be sure that
you will always get the same length strings, then it will be 'good
enough'. Personally, I feel the since you are given special characters
to use as a seperator, you might consider using them. Using cut would do
this.
The other option is to use a regular expression, I'm not too sure how
that would work out in asterisk, but the variables and tiki have some
info on this.
Regards,
Adam
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Re: [Asterisk-Users] VoicePulse Connect Problems
It was probably because I emailed everyone in the company every day until it got fixed... My next step was to start calling executives on their cell phones. ;-) To be fair, it took them a while to get back to my original email, but after I emailed the CEO, they were very responsive and sent me updates almost every day. I can't have been the only one getting those, since they told me they were aware of the problems even before I started bitching. Chris. On Tue, 13 Apr 2004, Isaac McDonald wrote: > It works now! I did nothing on my end either. VP must monitor this list. > > Isaac > > Robert Jackson wrote: > > >Just a quick couple of questions for ya'll. > > > >1) Does anyone know if VoicePulse Connect will be supporting dtmf tones? > >I have had a terrible time getting a hold of anyone over there, and I > >need this functionality before I can migrate to * completely. > > > >2) Are there currently any problems with inbound DID's? Everything is > >setup properly in *, but I am not able to receive inbound calls, through > >VoicePulse of course. It was working properly yesterday, and without > >changing anything it stopped working. > > > >Thanks in advance, > > > >Robert Jackson > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > -- chris maresca senior partner - www.olliancegroup.com linux, up 8 days ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T100P E&M Wink Trunk
I found ways to do substrings. So this is what I did. I changed the zap
channels to come into a context called 'fixup' and then jump into
default after doing the parsing and setting of CID.
[fixup]
exten => _*NXXNXX*NXXNXX,1,SetCIDNum(${EXTEN:1:10})
exten => _*NXXNXX*NXXNXX,2,Goto(default|${EXTEN:-10:10}|1)
So far things seem to be working well. Does this sound like a good way
to deal with this problem?
On Tue, 2004-04-13 at 17:19, Adam Goryachev wrote:
> > map it in the dial plan. Are there substr functions I can use? Can I
>
> Look at the wiki/tiki www.voip-info.org and search for cmd cut or at
> your console do a show application cut.
>
> If you still don't know what to do, do a search on the wiki for
> variables or read the README.variables in the asterisk source code (docs
> directory).
>
> After all that, if you still can't get it working, tell us what you have
> tried, and someone is sure to adjust your attempts to a working config.
>
> Good luck
>
> Adam
>
>
>
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Re: [Asterisk-Users] T100P E&M Wink Trunk
> map it in the dial plan. Are there substr functions I can use? Can I Look at the wiki/tiki www.voip-info.org and search for cmd cut or at your console do a show application cut. If you still don't know what to do, do a search on the wiki for variables or read the README.variables in the asterisk source code (docs directory). After all that, if you still can't get it working, tell us what you have tried, and someone is sure to adjust your attempts to a working config. Good luck Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T100P E&M Wink Trunk
I am setting up a box with a T100P. Everything is going well. The
company I am working with has their one phone switch gear. They
provisioned me a E&M Wink T1. Cannot do PRI unfortunately. We chose E&M
so we could pass an unlimited number of DIDs to the trunk as apposed to
FXS loopstart signaling. I can make outbound calls no problem, but I am
having problems with the dial plan for inbound calls. The way they setup
the trunk inbound calls have a dialed number as
"**". I do not know how to parse this out and
map it in the dial plan. Are there substr functions I can use? Can I
just call SetCIDNum on an INBOUND call to get the callerid functions
working?
Here is what I see in the log when a call comes in:
-- Starting simple switch on 'Zap/24-1'
== Unknown extension '*916111*916222' in context 'default'
requested
-- Playing 'ss-noservice' (language 'en')
-- Hungup 'Zap/24-1'
916111 is the calling number (Caller ID)
916222 is the called number
What would be the best way to convert this so I can use just ${EXTEN}?
I can get it to work if I do something like:
exten => _*XX*916222,1,Dial(
but that seems like a hack, plus that does not set callerid. I also dont
know how that work react to callerid being blocked.
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Re: [Asterisk-Users] Woodpeckers Revisited
Hi Michael, I tried to reply to your prive e-mail, but it seems like your mail service blacklists the whole of Hong Kong. :-\ Regards, Steve Michael Welter wrote: Just when I thought I couldn't be wrong, I was wrong. We have woodpeckers that drill into the arial telephone cables, and water seeps through the holes and partially grounds the tip and/or ring wires causing hum. I thought the hum/buz on my lines was a telco problem. The Qwest HQ noise team assures me that my lines are within spec. Sure enough, when I listen on the test set the lines are clear. The lines terminate at an Adtran 750 channel bank on my * system. When I reconnect the lines to the channel bank and make a call, I get the hum/buz noise. I have replaced every Adtran component (even the chassis), but the hum/buz stays with the lines. From the CO we have a digital fibre optic system which terminates at a neighborhood cabinet. From there, analog copper cables distribute service to the houses. I'm suspecting that the digital-to-analog process doesn't give a smooth analog signal but rather a "stair-stepped" signal, with each step 1/8000 sec in duration (I wish I had a 'scope to confirm this.) The human ear can't hear this stair-stepped signal, so it's ok for POTS use. However, when I put this stair-stepped signal to the channel bank, it converts it back into a digital signal. I'm thinking that, because it's not a smoothed signal, the analog-to-digital process injects hum and buz. Does _anyone_ have more information on this? In the meantime I've had an ISDN circuit installed so as to have digital all the way to the * box. However, I can't get the ASUSCOM ISDNLink card to work with ISDN4Linux :-( Cheers, Mike P.S. The woodpeckers are still eating my house. There is a nest is an exterior wall which is driving my cats nuts! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vonage goes to .ca
On Tue, Apr 13, 2004 at 07:09:57PM -0400, Jon Pounder wrote: > > FYI > > http://www.itbusiness.ca/index.asp?theaction=61&sid=55298 > > did not like this by line in the story > > t"he CRTC has said it will likely regulate voice over IP the same as other > > phone services." > I noticed exactly the same thing and thought the same when I read it earlier. > CRTC knows nothing about ip and should stay the hell out of it. > You should not expect the same things out of a voip circuit as you do a > land line. > What's next ? Are they going to regulate ftp ? > Are they going to poke their noses into private voip networks and try to > regulate that as well ? Where is the dividing line ? In the UK, Oftel (the telecoms regulator) has already stated that offering VoIP services is already covered by existing voice regulation, just because you're transporting voice over a different medium doesn't make any difference. Of course Ofcom (the new super regulator that has replaced Oftel and 4 other regulators) has blurred the issue trying to make allowances specifically for VoB ... Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P Issues
In case anyone was wondering, I managed to solve the issue. Turned out to be a problem with one of the X100P cards conflicting IRQ's with my ethernet card. This was causing zaptel to see only one X100P and then TDM400P, so the second channel was the TDM400P, which is why I was getting the errors about the FXS signalling. It's all good now! :) Thanks to all who assisted. Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Slaves: 04) 4 channels configured. Regards, Jeremy -- jeremy bogan[ [EMAIL PROTECTED] ] segment publishing - design.develop.host ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vonage goes to .ca
> FYI > http://www.itbusiness.ca/index.asp?theaction=61&sid=55298 > > did not like this by line in the story > t"he CRTC has said it will likely regulate voice over IP the same as other > phone services." > I noticed exactly the same thing and thought the same when I read it earlier. CRTC knows nothing about ip and should stay the hell out of it. You should not expect the same things out of a voip circuit as you do a land line. What's next ? Are they going to regulate ftp ? Are they going to poke their noses into private voip networks and try to regulate that as well ? Where is the dividing line ? > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > Jon Pounder Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID in Australia
they actually send the caller-id info after the SECOND ring. Now of course if the au indications were changed to combine the first and second ring to appear as one ring, no other changes would be needed ?? Gary On Wed, 14 Apr 2004 00:36:16 +1000, Duane wrote: >Adam Goryachev wrote: >> I am in Australia, which I think expects callerid at a different time to >> other countries Although other people have told me callerid is >> working correctly for them > > From what I've been able to guess at Telstra sends a short ~50ms chirp >to the phone, the caller id and then the first full ring, other >countries such as the US get the first full ring and then caller ID, and > at present caller ID seems to work but different ring candacies don't... > >-- >Best regards, > Duane > >http://www.cacert.org - Free Security Certificates >http://www.nodedb.com - Think globally, network locally >http://www.sydneywireless.com - Telecommunications Freedom >http://happysnapper.com.au - Sell your photos over the net! >http://e164.org - Using Enum.164 to interconnect asterisk servers >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom phones noise cancellation
Sean Garland wrote: In almost all my calls now, I am getting beeps and loud and soft parts of a conversation. It is getting very irritating. Has anyone had this happen? How do I get rid of it? Set relaxdtmf=no in /etc/asterisk/zapata.conf. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Vonage goes to .ca
FYI http://www.itbusiness.ca/index.asp?theaction=61&sid=55298 did not like this by line in the story t"he CRTC has said it will likely regulate voice over IP the same as other phone services." ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Upgraded to latest CVS, now no IAX1?
Brian Cuthie wrote: Tor Houghton wrote: On Tue, Apr 13, 2004 at 02:56:48PM -0400, Jeremy McNamara wrote: Use IAX2, it is a better IAX protocol. Jeremy McNamara P.S. If you really must have it, dig thru the channels/Makefile, but there is zero reason to use it any longer. Well, I use IAX1 between the clients on the inside of the NAT to my local Asterisk, and IAX2 between the local Asterisk and my remote Asterisk. Previously (I have not tried yet with current version), when both clients and Asterisk used IAX2, the clients would communicate directly with remote Asterisk and so confuse my NAT firewall. Tor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Probably a port collision on your NAT box. I believe that IAX and IAX2 use different ports. Or you can deactivate transfers at iax.conf: notransfer=yes Daniel -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom phones noise cancellation
In almost all my calls now, I am getting beeps and loud and soft parts of a conversation. It is getting very irritating. Has anyone had this happen? How do I get rid of it? Thanks Sean Garland, MCP+I, A+ Siskiyou Technology Consultants 205 N. Mt. Shasta Blvd. Suite 100 Mt. Shasta, CA 96067 Phone: (530)926-1489 FAX: (530)926-6296 [EMAIL PROTECTED] http://www.siskiyoutech.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TAPI driver
Could you post a link? Thanks! On Apr 13, 2004, at 2:59 PM, Nick Knight wrote: Just a quick note, I have been putting together a TAPI driver for Asterisk, this enables the user to perform things like click to dial from any TAPI enabled app (such as outlook or ACT etc). At the moment it is very basic and can only perform click to dial but further functionality will be coming. It uses the Asterisk manager to place calls. Please feel free to use it - not much documentation as yet but will be coming, can be found on sourceforge project name asttapi. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Insert pause in SIP String
Erick Weber V. wrote: I'll Like to now how to insert a pause on a SIP string. I have a ATA 186 and a FXS => FXO converter so I will like to program a extension that can be dialed and it will dial the ATA extention, wait for dial tone and then dial the phone number. You cannot put pauses in any dial string in Asterisk except calls using ANALOG Zap or ANALOG Voicetronix ports. This isn't really an Asterisk problem, it's a protocol problem. You could hack something into Asterisk to work around the problem, but that's Non-Trivial ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse Connect Problems
They just updated their software and that seems to have resolved the DTMF issues, at least for me. Chris. On Tue, 13 Apr 2004, Andrew Kohlsmith wrote: > > 1) Does anyone know if VoicePulse Connect will be supporting dtmf tones? > > I have had a terrible time getting a hold of anyone over there, and I > > need this functionality before I can migrate to * completely. > > Works just fine for me. Don't send in-band DTMF if you're not using the > alaw/ulaw/slinear codecs. It won't work. > > Regards, > Andrew > > > -- chris maresca senior partner - www.olliancegroup.com linux, up 8 days ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Upgraded to latest CVS, now no IAX1?
On Tue, Apr 13, 2004 at 04:58:19PM -0400, James Golovich wrote: > > # If you really want IAX1 uncomment the following, but it is > # unmaintained > # > #CHANNEL_LIBS+=chan_iax.so > Thanks all, I'll move to IAX2 after I've tested the notransfer option. Tor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zapateller issues
Yeah, tried this. Seems that the Zapateller code is not written correctly. The problem is that if one does this exten => s,1,Zapateller(answer|nocallerid) Then the call is answered by Zapateller regardless of the callerID state. The tones are played if there is no caller id. The problem with this is that the call gets answered regardless of state and so the caller gets charged. If my dial plan forwards the call to my phone and the caller wants to hangup before I get to it he has to pay anyway. This is expensive especiall if calling from abroad. When the Zapateller has answered there is no longer any ringing heard whilst the calling party is waiting for a SIP extension to pick up either. I can see some advantages of having the above situation howevever. eg, one could answer the call and then dump the caller onto hold whilst the extensions are rung. If I do this exten => s,1,Zapateller(nocallerid) The call gets answered when there is no callerid info but DOES NOT play the tones which is kinda useless. I have also noticed that when the tones are played they are often trunkated; the beginning gets cut off. Could a few seconds pause be written in so that the connection has a chance to sort itself out before the tones play? > On Mon, 12 Apr 2004, Mark Phillips wrote: >> I tried, >> >> exten => s,1,Zapateller(answer|nocallerid) >> exten => s,2,Privacymanager >> exten => s,3,Dial(a bunch of SIP extensions) >> >> But then every call was answered regardless of CID and the tones were >> heard. > > I tried the sample I found at: > http://www.loligo.com/asterisk/current/extensions.conf > and it worked well. Look for the "inbound-analog" context. > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > G7LTT/KC2ENI Mark Phillips ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Upcoming 1.0 Release Suggestions
Since Asterisk 1.0 will be released soon I am wondering if Digium runs CVS stable on IAXtel and Digium's own PBX. If they are, then great! It will get a good workout. If not, then WHY? A great way for a product to get bugs fixed are for the group that codes the product to run it in a production enviroment. I don't think I'd trust a release enough to run it myself if the people that wrote it doesn't even use it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Upgraded to latest CVS, now no IAX1?
On Tue, 13 Apr 2004, Eric Wieling wrote: > Tor Houghton wrote: > > > Well, I use IAX1 between the clients on the inside of the NAT to my local > > Asterisk, and IAX2 between the local Asterisk and my remote Asterisk. > > Previously (I have not tried yet with current version), when both clients > > and Asterisk used IAX2, the clients would communicate directly with remote > > Asterisk and so confuse my NAT firewall. > > Are you using cvs latest or cvs stable? I thought IAX1 was still in cvs > stable, but I could be wrong. To enable IAX1, the following line in channels/Makefile needs to be uncommented. # If you really want IAX1 uncomment the following, but it is # unmaintained # #CHANNEL_LIBS+=chan_iax.so James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sphinx voice recognisation
Has anyone had any luck with voice recognisation using sphinx, if yes then could u please send some pointers. does the eagi app for sphinx really work cause i'v tierd it and sphix dosent seem to do anything -- regards Vikram (http://www.vicramresearch.com) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Upgraded to latest CVS, now no IAX1?
Tor Houghton wrote: Well, I use IAX1 between the clients on the inside of the NAT to my local Asterisk, and IAX2 between the local Asterisk and my remote Asterisk. Previously (I have not tried yet with current version), when both clients and Asterisk used IAX2, the clients would communicate directly with remote Asterisk and so confuse my NAT firewall. Are you using cvs latest or cvs stable? I thought IAX1 was still in cvs stable, but I could be wrong. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoicePulse Connect Problems
Mine, too, are fixed...I was in much the same boat as the original poster...an old DID in 212 worked with DTMF, two much newer ones in 213 and 818 (new markets, apparently) didn't until this morning. On Tue, 13 Apr 2004 16:02:37 -0400, "Robert Jackson" <[EMAIL PROTECTED]> said: > Very cool. I am just glad they got it fixed. > > -Original Message- > From: Isaac McDonald [mailto:[EMAIL PROTECTED] > Sent: Tuesday, April 13, 2004 3:56 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] VoicePulse Connect Problems > > > It works now! I did nothing on my end either. VP must monitor this list. > > Isaac > > Robert Jackson wrote: > > >Just a quick couple of questions for ya'll. > > > >1) Does anyone know if VoicePulse Connect will be supporting dtmf > >tones? I have had a terrible time getting a hold of anyone over there, > >and I need this functionality before I can migrate to * completely. > > > >2) Are there currently any problems with inbound DID's? Everything is > >setup properly in *, but I am not able to receive inbound calls, > >through VoicePulse of course. It was working properly yesterday, and > >without changing anything it stopped working. > > > >Thanks in advance, > > > >Robert Jackson > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoicePulse Connect Problems
Very cool. I am just glad they got it fixed. -Original Message- From: Isaac McDonald [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 13, 2004 3:56 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoicePulse Connect Problems It works now! I did nothing on my end either. VP must monitor this list. Isaac Robert Jackson wrote: >Just a quick couple of questions for ya'll. > >1) Does anyone know if VoicePulse Connect will be supporting dtmf >tones? I have had a terrible time getting a hold of anyone over there, >and I need this functionality before I can migrate to * completely. > >2) Are there currently any problems with inbound DID's? Everything is >setup properly in *, but I am not able to receive inbound calls, >through VoicePulse of course. It was working properly yesterday, and >without changing anything it stopped working. > >Thanks in advance, > >Robert Jackson >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TAPI driver
Hello all, Just a quick note, I have been putting together a TAPI driver for Asterisk, this enables the user to perform things like click to dial from any TAPI enabled app (such as outlook or ACT etc). At the moment it is very basic and can only perform click to dial but further functionality will be coming. It uses the Asterisk manager to place calls. Please feel free to use it - not much documentation as yet but will be coming, can be found on sourceforge project name asttapi. Regards Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse Connect Problems
It works now! I did nothing on my end either. VP must monitor this list. Isaac Robert Jackson wrote: Just a quick couple of questions for ya'll. 1) Does anyone know if VoicePulse Connect will be supporting dtmf tones? I have had a terrible time getting a hold of anyone over there, and I need this functionality before I can migrate to * completely. 2) Are there currently any problems with inbound DID's? Everything is setup properly in *, but I am not able to receive inbound calls, through VoicePulse of course. It was working properly yesterday, and without changing anything it stopped working. Thanks in advance, Robert Jackson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Upgraded to latest CVS, now no IAX1?
On Tue, 2004-04-13 at 20:13, Tor Houghton wrote: > Well, I use IAX1 between the clients on the inside of the NAT to my local > Asterisk, and IAX2 between the local Asterisk and my remote Asterisk. > Previously (I have not tried yet with current version), when both clients > and Asterisk used IAX2, the clients would communicate directly with remote > Asterisk and so confuse my NAT firewall. In iax.conf, set: notransfer=yes That prevents IAX from transferring call to remote Asterisk, & so it will stay in path. F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Upgraded to latest CVS, now no IAX1?
Tor Houghton wrote: On Tue, Apr 13, 2004 at 02:56:48PM -0400, Jeremy McNamara wrote: Use IAX2, it is a better IAX protocol. Jeremy McNamara P.S. If you really must have it, dig thru the channels/Makefile, but there is zero reason to use it any longer. Well, I use IAX1 between the clients on the inside of the NAT to my local Asterisk, and IAX2 between the local Asterisk and my remote Asterisk. Previously (I have not tried yet with current version), when both clients and Asterisk used IAX2, the clients would communicate directly with remote Asterisk and so confuse my NAT firewall. Tor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Probably a port collision on your NAT box. I believe that IAX and IAX2 use different ports. -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Internationalisation/Internationalization
On Tue, 2004-04-13 at 16:11, Benjamin Wakefield wrote: > On the tiki it says for international digits, I can dump them in the > "digits/au" directory. > I tried that -- just because, I also made a copy in "au/digits". > When the queue announces the position I it says: > -- Started music on hold, class 'default', on SIP/11-5324 > -- Stopped music on hold on SIP/11-5324 > -- Playing 'dcsi/queue-thereare' (language 'au') > -- Playing 'digits/2' (language 'en') > -- Playing 'dcsi/queue-callswaiting' (language 'au') > See that? The digits are 'en'! I can't work out why. Bug: http://bugs.digium.com/bug_view_page.php?bug_id=0001097 Patch listed there doesn't work for me, I'd be very happy to see a fix... F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoicePulse Connect Problems
Isaac: The DID is in Ocala, FL. I am not sure if it is a new market or not. I have not heard anything from their support folks either, but I just checked the line again and it is working. I did nothing to fix it. I just don't understand. If you don't mind give yours a try again and let me know if it is working as well. Thanks, Robert -Original Message- From: Isaac McDonald [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 13, 2004 2:03 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoicePulse Connect Problems Robert Jackson wrote: >Just a quick couple of questions for ya'll. > >1) Does anyone know if VoicePulse Connect will be supporting dtmf >tones? I have had a terrible time getting a hold of anyone over there, >and I need this functionality before I can migrate to * completely. > >2) Are there currently any problems with inbound DID's? Everything is >setup properly in *, but I am not able to receive inbound calls, >through VoicePulse of course. It was working properly yesterday, and >without changing anything it stopped working. > >Thanks in advance, > >Robert Jackson >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > I have two incoming DID's from Voicepulse. The first one I set up over 3 months ago and has been working fine, including incoming DTMF. The second one I ordered last week in the Birmingham, AL market, its a new rate center, DTMF does not work on this DID. I e-mailed support on this issue and have yet to get a response. Is your DID in a of of their new markets by chance? Isaac ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] controlling call duration
Hi! > On Tue, 13 Apr 2004, Dmitry Mishchenko waxed: > > > In other words can I receive information which we are usually getting in CDRs > > during the time when the call is still active? > > Yes, via the manager interface. Check manager.conf, it > lets * talk on port 5038. The other option is that you write your own AGI that triggers at the beginning of the call and records the info that you want. Then you can add another script that checks for "show channels" or something similar and removes the temporary Db entry the first script created in case the call has been completed. Downside: You don't get to trace what happened in between, i.e. you are missing all the transfer and parking games. And all the queue fun. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Upgraded to latest CVS, now no IAX1?
On Tue, Apr 13, 2004 at 02:56:48PM -0400, Jeremy McNamara wrote: > > Use IAX2, it is a better IAX protocol. > > > Jeremy McNamara > > > P.S. If you really must have it, dig thru the channels/Makefile, but > there is zero reason to use it any longer. > Well, I use IAX1 between the clients on the inside of the NAT to my local Asterisk, and IAX2 between the local Asterisk and my remote Asterisk. Previously (I have not tried yet with current version), when both clients and Asterisk used IAX2, the clients would communicate directly with remote Asterisk and so confuse my NAT firewall. Tor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Upgraded to latest CVS, now no IAX1?
Tor Houghton wrote: Hi, I just upgraded to the recent CVS, and IAX1 no longer seems to be available. Is there a way to reenable it? Use IAX2, it is a better IAX protocol. Jeremy McNamara P.S. If you really must have it, dig thru the channels/Makefile, but there is zero reason to use it any longer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hunting S(n)IPs
> Another thing to try is to disable call waiting on the > [EMAIL PROTECTED] phone (if call waiting is enabled, it's doing > what you've asked it to)... > Yep, except on the Polycom, we have found no way to disable call-waiting. WW Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hunting S(n)IPs
> 2) I think you should have been looking at incominglimit, > not outgoinglimit, or possibly both of them together in > some combination. > Another perspective issue. Apparantly 'incoming' means into the [*] box, and outgoing is leaving the [*]. In any case, I tried both, but 'outgoing' is confirmed broken. WW Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Upgraded to latest CVS, now no IAX1?
Hi, I just upgraded to the recent CVS, and IAX1 no longer seems to be available. Is there a way to reenable it? Tor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tcp/ip stack tweaks
Scott Laird wrote: There shouldn't be much that needs tuned, unless your network is overloaded and dropping packets. If that's happening, then you're going to need to dig in and take a look at QoS on Linux *and* on your switches and routers, but odds are that won't be a problem on most LANs. I mean, even G.711 is only ~80 kbps (including overhead), so you should be able to run hundreds of simultaneous conversations on 100 Mbps Ethernet without running out of bandwidth. Are you having problems? What are you seeing? I just asked this question out of curiosity. We're thinking about deploying this network wide, and besides QoS on routers and switches I've been looking into performance tweaks on the pbx box itself to get the most life out of it. Once you get into enterprise situations the default settings are never good enough. -- Rock River Internet Roger Grunkemeyer 202 W. State St, 8th Floor[EMAIL PROTECTED] Rockford, IL 61101815-968-3888 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXS => FXO Converter Problem
Andrew:
It didn't work, the problem is that * stays on priority 1 until you hangup
and the it pass to priority 2 so what I think is that it has to be all in
the priority 1 line
Hope we can figure it out
Erick
- Original Message -
From: "Andrew Thompson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, April 13, 2004 12:48 PM
Subject: RE: [Asterisk-Users] FXS => FXO Converter Problem
> Erick Weber V. wrote:
> > Hello:
> >
> > I have a ATA 186 and a FXS => FXO converter so I will like to program
> > a extension that can be dialed and it will dial the ATA extention #,
> > wait for dial tone and then dial the phone number.
>
> Unfortunately I don't believe there is a concept of "wait for dial tone."
>
> You'll just need to test it, and see how long it takes to do the answer,
> pickup, get dialtone. Time it a few times, then add a second or so onto
> that.
>
> I tried to write up an example of what you should put into your
> extensions.conf, but It's a little over my head in this case. My thoughts
> are:
>
> exten => _91NX,1,Dial(sip/yourata)
> exten => _91NX,1,SendDigits(${EXTEN:1})
>
> NOTE: I've not tested this...
>
> -
> Andrew Thompson
> http://aktzero.com/
>
>
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Re: [Asterisk-Users] FXS => FXO Converter Problem
Andrew
Thanks for your answer
I'll test this conf an I'll post it so you know if it works
Thanks
Erick
- Original Message -
From: "Andrew Thompson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, April 13, 2004 12:48 PM
Subject: RE: [Asterisk-Users] FXS => FXO Converter Problem
> Erick Weber V. wrote:
> > Hello:
> >
> > I have a ATA 186 and a FXS => FXO converter so I will like to program
> > a extension that can be dialed and it will dial the ATA extention #,
> > wait for dial tone and then dial the phone number.
>
> Unfortunately I don't believe there is a concept of "wait for dial tone."
>
> You'll just need to test it, and see how long it takes to do the answer,
> pickup, get dialtone. Time it a few times, then add a second or so onto
> that.
>
> I tried to write up an example of what you should put into your
> extensions.conf, but It's a little over my head in this case. My thoughts
> are:
>
> exten => _91NX,1,Dial(sip/yourata)
> exten => _91NX,1,SendDigits(${EXTEN:1})
>
> NOTE: I've not tested this...
>
> -
> Andrew Thompson
> http://aktzero.com/
>
>
> ___
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>http://lists.digium.com/mailman/listinfo/asterisk-users
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>
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Re: [Asterisk-Users] VoicePulse Connect Problems
Robert Jackson wrote: Just a quick couple of questions for ya'll. 1) Does anyone know if VoicePulse Connect will be supporting dtmf tones? I have had a terrible time getting a hold of anyone over there, and I need this functionality before I can migrate to * completely. 2) Are there currently any problems with inbound DID's? Everything is setup properly in *, but I am not able to receive inbound calls, through VoicePulse of course. It was working properly yesterday, and without changing anything it stopped working. Thanks in advance, Robert Jackson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have two incoming DID's from Voicepulse. The first one I set up over 3 months ago and has been working fine, including incoming DTMF. The second one I ordered last week in the Birmingham, AL market, its a new rate center, DTMF does not work on this DID. I e-mailed support on this issue and have yet to get a response. Is your DID in a of of their new markets by chance? Isaac ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call parking on central asterisk system
I have 2 asterisk systems connected with an iax2 trunk. The first has SIP phones and x100 line cards, the second at a remote location has a TDM with zap extensions. When calls are parked by the zap extensions at the second system, the calls are parked on the second system so users at the first server cannot access them. I will try to create an extn to connect to the second system to collect the call but would prefer to have the second system park the call at the first. Does anyone know how I can do this? TIA Stuart. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Lucent Phones
I am confused as well. They also made it clear that the contract terms included recouping the cost of the ConnectReach for them, so I doubt that TWTC is offering it at no extra cost. My contention with that, of course, is, "why not take my ConnectReach and give it to someone else... you already have had 18 months of me paying for it". As for the refund window, well, Mark's a nice guy, but I don't know if he's _that_ nice. ;) Greg On Tue, 2004-04-13 at 10:50 -0400, Troy Settle wrote: > > -Original Message- > > From: Gregory Junker > > > > TWTC has examined the T100P and informed me that it's > > impossible, since > > their IBL uses proprietary formatting and signalling. Also, I ought to > > be able to use the data channels directly, according to > > Digium, since my > > proposed Asterisk box is also our router. > > > > If they are lying to me (which I doubt...they have a vested > > interest in > > using a proprietary method), then as for finding out which > > channels are > > used for what is as simple as trial-and-error and a cell > > phone. ;) (and > > of course, the $400 or so to pick up a T100P to try it out...) I am > > guessing first four are voice and next 12 are data. > > I'm rather confused by this. It was my understanding that the connectreach > was nothing more than a glorified channel bank with IP routing capabilities > (I have 2 customers with 6 voice lines, and 384k of data that's handed off > as ethernet by the connectreach. The voice lines come off a 50pin telco > connector. > > If TWTC, like my CLEC, offers the connectreach at no additional cost, then I > seriously doubt that they would lie to you. Returning the connectreach > would save them some small amount of money at the end of the day. If their > solution is propriatary, that's fine, but I don't see how/why they wouldn't > be able to reprovision the T1 as a normal circuit. > > If you can get Digium to give you a 30 day refund window, then I'd say that > it's well worth it to give this a try. > > -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FXS => FXO Converter Problem
Erick Weber V. wrote:
> Hello:
>
> I have a ATA 186 and a FXS => FXO converter so I will like to program
> a extension that can be dialed and it will dial the ATA extention #,
> wait for dial tone and then dial the phone number.
Unfortunately I don't believe there is a concept of "wait for dial tone."
You'll just need to test it, and see how long it takes to do the answer,
pickup, get dialtone. Time it a few times, then add a second or so onto
that.
I tried to write up an example of what you should put into your
extensions.conf, but It's a little over my head in this case. My thoughts
are:
exten => _91NX,1,Dial(sip/yourata)
exten => _91NX,1,SendDigits(${EXTEN:1})
NOTE: I've not tested this...
-
Andrew Thompson
http://aktzero.com/
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[Asterisk-Users] Dialout from SIP to PSTN
Hi,
i install the Asterisk PBX on a linux machine with i4l to connect to PSTN
(EuroISDN). And i configure a very simple dial plan in extension.conf.
After this, i connect with a SIP program to asterisk and would call my
cellular phone, but got this error:
-- Executing Ringing("SIP/ACzerniak-0904", "") in new stack
-- Executing Dial("SIP/ACzerniak-0904", "Modem/g1/01x") in new
stack
chan_modem.c:181 modem_call: Destination g1/01x requres a
real destination (device:destination)
-- Couldn't call g1/01x
-- Hungup 'Modem[i4l]/ttyI1'
== Everyone is busy at this time
-- Executing Congestion("SIP/ACzerniak-0904", "") in new stack
== Spawn extension (default, 901x, 3) exited non-zero on
'SIP/ACzerniak-0904'
I change the TRUNK variable from Modem/g1 to Modem/ttyI[0|1], but this have
the same effect.
What means the "Destination g1/01x requires a _real_ destination" ?
Thanks in advanced.
Regards,
Andreas.
The modem.conf:
[interfaces]
context=remote
driver=i4l
dialtype=tone
mode=immediate
group=1
msn=85xx
device => /dev/ttyI0
device => /dev/ttyI1
The extentsion.conf
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=Modem/g1
TRUNKMSD=1 ; MSD digits to strip
(usually
PHONE1=SIP/ACzerniak
The [default] section includes:
exten => _90ZX,1,Ringing ; read it from
exten => _90ZX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _90ZX,3,Congestion
--
"If you want to pray. Go to the sea."
Andreas Czerniak <[EMAIL PROTECTED]>
PGPkey http://pgp5.ai.mit.edu:11371/pks/lookup?op=get&search=0xEDB224EC
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Re: [Asterisk-Users] VoicePulse Connect Problems
> 1) Does anyone know if VoicePulse Connect will be supporting dtmf tones? > I have had a terrible time getting a hold of anyone over there, and I > need this functionality before I can migrate to * completely. Works just fine for me. Don't send in-band DTMF if you're not using the alaw/ulaw/slinear codecs. It won't work. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T100P / ZAP / PRI errors
That would be a valid configuration, and yes yellow is an option for setting a yellow alarm when no channels are open. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel Sent: Tuesday, April 13, 2004 10:57 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] T100P / ZAP / PRI errors That makes sense. I've always found the following wording (from the sample file) confusing, not clear about whether "this" and "this span" referred to the span connection on the card, or the span itself: # The timing parameter determines the selection of primary, secondary, and # so on sync sources. If this span should be considered a primary sync # source, then give it a value of "1". For a secondary, use "2", and so on. # To not use this as a sync source, just use "0" So just to clarify some more, I assume if you're using a 4-span card like a TE410P, you would have something like this (in a T1 environment with all connected to a telco switch): span=1,1,0,esf,b8zs,yellow(primary time source) span=2,2,0,esf,b8zs,yellow (secondary time source) span=3,0,0,esf,b8zs,yellow span=4,0,0,esf,b8zs,yellow Do you agree with this? (the "yellow" is an optional parameter) Cheers Scott Stingel www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bisker, Scott (7805) Sent: Tuesday, April 13, 2004 4:59 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] T100P / ZAP / PRI errors In laymans terms. To use your telco's T-1 as the timing source span=1,1,0,esf,b8zs,yelllow To use the internal clock of the card you would use (I'm pretty sure that this would only be used for channel banks, or connections to other PBX hardware. I don't think a telco is going to use your PBX as a timing source) span=1,0.0,esf,b8zs,yellow If you have multiple telco connections on multiple spans you would have something like this span=1,2,0,esf,b8zs,yellow(secondary time source) span=2,1,0,esf,b8zs,yellow (primary time source) span=3,0,0,esf,b8zs,yellow (provide the time source, i.e. channel bank) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Monday, April 12, 2004 9:39 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] T100P / ZAP / PRI errors Now you've got me utterly confused ... So, in layman's terms, if I connect a T100P to a circuit provided by the Telco, and the Telco says that they will provide timing, I have to put WHAT? span=1,0,0,esf,b8zs,yellow this means '0' this span is not a sync source, i.e. the Telco will provide my 8kHz. Could one use '2' with impunity (span=1,2,0,...)? I am still not clear under which circumstances one should use '0' versus '2'. WW - Original Message Follows - > Holy crap people, trim your replies! > > > You didn't say what's at the other end of your PRI line, but you > > might try having the other end be the timing sync source. Try: > > span=1,0,0,esf,b8zs instead. Maybe > that will help. > > We need to get this documented *clearly* once and for all. > > Zaptel T1/E1 hardware either free-runs to its own internal 8kHz time > source, or it tries to lock to the recovered clock from the line. > > Zapata.conf says that timing of 0 means "do not use this span for > timing." Zero does not mean "slave timing", it means not to use this > span as a recovered clock source for timing at all. Timing values of > 1 or 2 mean try to lock the internal clock to the recovered clock > from the span. > > A value of 0 means that this span's recovered clock never gets used as > a timing source. A value of 1 means that this span is the primary > clock source -- If the span is up, try to lock the internal clock to > the clock recovered from this span. A value of 2 means to use this > span for timing only if the primary span is down. > > To reiterate: a value of 0 means that the other end must be locking to > the zaptel's clock or else clock slips will occur. > > Feel free to correct me if I'm wrong, but I am pretty sure I have this > right. :-) > > Regards, > Andrew > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___
[Asterisk-Users] Bug with 'r' in dial
The lastest CVS's versions (both stable and head), the 'r' option in app_dial doesn't work with SIP and Re-invites. I've heard reports that it's not working with IAX2 either.. I'm using Cisco gateway and cisco ATA's and I am doing re-invites, and it's worked up till this point.. What's going on? Thanks, Billy +--+ | Billy Huddleston Senior Systems Administrator | | Net-Express http://www.nxs.net | | 114 Sherway Rd. Voice: 865-691-2011 | | Knoxville, TN 37922 Fax: 865-691-9894 | | [EMAIL PROTECTED]| +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] small question 3 way calling
The GS phones do not currently support conferencing on the phones using the conference button. You'll probably have better luck setting up a conference room, help with which I'm absolutely worthless... The on-phone conferencing should be addressed in a future GS firmware revision. HTH, Ryan On Apr 13, 2004, at 10:46 AM, Anthony Law wrote: According to voip-info.org, "3 way calling: Normally implemented by the phone" I am using a Grand Stream 100 and not able to make this work. I can dial out to 1st number then with the flash button I am able to dial out again to a 2nd number. I am not able to bind them together into 1 conversation. Is there something I have to set on the phone config or in sip.conf?? Anyone knows? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quality Problem
Il 17:28, martedì 13 aprile 2004, Robert Siedl ha scritto: > Hi List, > > I have asterisk running on my server and work with 2 cisco ata und 1x > snom device. I can intern call it´s fine. But wenn i make a extern call, > I have many quality troubles. The extern user hear me good, but I hear > him bad (robotics). I work with SIP an ALAW protocol. > > Where can i look this error? I am a new asterisk user. > > Thank you, > > best regards, > > Robert Siedl Hello, I've also had the same problem. As I know, this issue is related to chan_capi and the new lock features of asterisk (that are used by chan_mgcp) until new releases, the only solution (I've found) is to roll back to cvs version prior 12nd march 2004. Hope this help ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Immix C3-FXO gateway
Assuming that it is a Welltech gw, the setting is peer-to-peer mode. Jorge John Bittner wrote: Hi, Anyone get the Immix C3-FXO Sip gateway to work with asterisk. I have it working for outbound calls but cant get it to work for inbound calls. The unit has an built-in greeting and it keeps picking up the call. Cant find the command to turn it off and set it to forward the calls to asterisk. Any help on this would be appreciated. Thanks John Bittner Simlab.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 and dtmf
On Fri, 2004-04-09 at 02:56, Alessio Focardi wrote: > HI, > > quick and simple question: is it possible to use inband dtmf with g729? Absolutely not. > What I would like to do is to have sip clients connected to asterisk and a zaptel > card to make pstn phone calls. > > My concern is to allow sip users to use digits for call destinations that > do require menu actions while retaining low bandwith occupation. If asterisk couldn't decode reliably a DTMF signal on the end of a network link, what makes you think it would survive a analog/digital/analog conversion on top of the lossy codec. If you search the archives, you will find that asterisk will convert the oob DTMF to inband when it goes to PSTN, or at least it is supposed to. I think there may have been some problems lately with the length of a DTMF tone played, but it is supposed to work. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXS => FXO Converter Problem
Hello: I have a ATA 186 and a FXS => FXO converter so I will like to program a extension that can be dialed and it will dial the ATA extention #, wait for dial tone and then dial the phone number. Thanks Erick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoicePulse Connect Problems
Just a quick couple of questions for ya'll. 1) Does anyone know if VoicePulse Connect will be supporting dtmf tones? I have had a terrible time getting a hold of anyone over there, and I need this functionality before I can migrate to * completely. 2) Are there currently any problems with inbound DID's? Everything is setup properly in *, but I am not able to receive inbound calls, through VoicePulse of course. It was working properly yesterday, and without changing anything it stopped working. Thanks in advance, Robert Jackson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VideoMail
Since * does video over sip has anyone tried to configure voicemail2 to be able to leave a video message? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P and NTL (ex Cable + Wireless)
Stephen Davies [EMAIL PROTECTED] wrote: > > Has anybody got any experience using an X100P on an NTL phone line in > > the UK (I'm in an ex Cable & Wireless area if that makes any > > difference). > > > Indeed the call end termination doesn't work on an NTL line. I'm not > so sure it works too well on other lines either. > > The main practical issue is with voicemail, as you say. > > My final solution was to switch to ISDN. > I have no such problems with BT. Perhaps that's another (cheaper) option. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] controlling call duration
On Tue, 13 Apr 2004, Dmitry Mishchenko waxed: > In other words can I receive information which we are usually getting in CDRs > during the time when the call is still active? Yes, via the manager interface. Check manager.conf, it lets * talk on port 5038. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] small question 3 way calling
According to voip-info.org, "3 way calling: Normally implemented by the phone" I am using a Grand Stream 100 and not able to make this work. I can dial out to 1st number then with the flash button I am able to dial out again to a 2nd number. I am not able to bind them together into 1 conversation. Is there something I have to set on the phone config or in sip.conf?? Anyone knows? Regards, Anthony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call queue list members using sql query
On Mon, 12 Apr 2004, Dragan Mickovic waxed: > Is it possible for asterisk to do an sql query in order to > get the member list of a call queue? No, you will have to write code besides SQL in order to do it. To go the C route, try modifiying app_queue. To use a different language, you could code something over the manager interface that interacts with AddQueueMember and RemoveQueueMember in extensions.conf. That would even give you some more dynamic control of the members. There's lots more ways to do it, tho. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID in Australia
Duane wrote: Adam Goryachev wrote: I am in Australia, which I think expects callerid at a different time to other countries Although other people have told me callerid is working correctly for them Forgot to mention there is a patch for this, but it won't patch cleanly against current CVS... http://bugs.digium.com/bug_view_page.php?bug_id=0001007 That patch is not very efficient, since it leaves the detector running at all times. It would be more efficient to do a simple energy test all the time, and enable the CLI decoder only when some energy is seen. However, it looks like it should work for the UK as well as Australia. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 and dtmf
Alessio Focardi wrote: HI, quick and simple question: is it possible to use inband dtmf with g729? What I would like to do is to have sip clients connected to asterisk and a zaptel card to make pstn phone calls. My concern is to allow sip users to use digits for call destinations that do require menu actions while retaining low bandwith occupation. You can only do INBAND DTMF with the G711 ULAW or G711 ALAW. Other codecs distort DTMF. This is not an Asterisk issue, it's a codec issue. That's why there is Out of Band DTMF. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] *** List etiquette - digest readers
If you're reading the digest of the Asterisk-users mailing list: * Please always strip the parts of the message you're not replying to - do not resend the whole digest! * Please always change the subject so it reflects your message - Do not send a message with a subject of RE: Asterisk-Users digest, Vol 1 #3413 - 14 msgs It does *not* reflect the content of your message There are so many readers of this mailing list out there, so resending a whole digest wastes bandwidth and storage space for us all. If you're not changing the subject, people like me will not read your message and you will not get many answers, if any at all. Thank you for helping us managing the mailing list and making it easy to follow the stream of messages! ...and yes, sending a digest back raises risks of you getting flamed by Critch or Bkw. Beware :-) /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Quality Problem
Hi List, I have asterisk running on my server and work with 2 cisco ata und 1x snom device. I can intern call it´s fine. But wenn i make a extern call, I have many quality troubles. The extern user hear me good, but I hear him bad (robotics). I work with SIP an ALAW protocol. Where can i look this error? I am a new asterisk user. Thank you, best regards, Robert Siedl Ikarus GuardNT hat dieses eMail auf Viren und Trojaner untersucht. Nichts Verdächtiges gefunden. keine Anlagen gefunden --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Internationalisation/Internationalization
Good Morning, I'm working with a queue at the moment and I having trouble with my digits. Australia is my example. On the tiki it says for international digits, I can dump them in the "digits/au" directory. I tried that -- just because, I also made a copy in "au/digits". When the queue announces the position I it says: -- Started music on hold, class 'default', on SIP/11-5324 -- Stopped music on hold on SIP/11-5324 -- Playing 'dcsi/queue-thereare' (language 'au') -- Playing 'digits/2' (language 'en') -- Playing 'dcsi/queue-callswaiting' (language 'au') See that? The digits are 'en'! I can't work out why. I set "language = au" in my sip.conf and I also used SetLanguage(au) before I dumped the call in the queue. Didn't make any difference though. I'm using the latest CVS. I've just copied over the files in the digits directory for now (sorry Allison), but that isn't the way it should work. Am I missing something, or is something not quite right? :) Ben Benjamin Wakefield [EMAIL PROTECTED] http://www.dcsi.net.au/ DCSI - We do Internet. 64 Queen Street Warragul, VIC 3820 AU Ph: (+61) 1300 665 575 Fx: (+61) 1300 556 595 -BEGIN GEEK CODE BLOCK- Version: 3.12 G! d- s: a-- C+ UL++ P+ L++ E W+ N+ o- K- w+$ O--- M-- V? PS !PE Y-- PGP- t 5 X+ R- tv b- DI-- D--- G-- e* h* r- z++ --END GEEK CODE BLOCK-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hunting S(n)IPs
Andrew Thompson wrote: [EMAIL PROTECTED] wrote: Another observation of something which doesn't work: exten => 3200,1,Dial(SIP/3200,20,tTr) exten => 3200,2,Playback(tt-weasels) exten => 3200,3,Hangup exten => 3200,102,Dial(SIP/3201,20,tTr) exten => 3200,103,Playback(tt-weasels) exten => 3200,104,Hangup exten => 3200,203,Dial(SIP/3202,20,tTr) exten => 3200,204,Playback(tt-weasels) exten => 3200,205,Hangup The [EMAIL PROTECTED] phone does NOT give a BUSY indication even afer the first call has been answered. Therefore, Call#2 happily dials 3200 again, although 3200 is currently talking. I also tried to limit the number of calls going to the phone with outgoinglimit=1 in the sip.conf, but that makes no difference either. According to the wiki that functionality is broken. Two things: 1) Have you looked at call queue's? 2) I think you should have been looking at incominglimit, not outgoinglimit, or possibly both of them together in some combination. - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I may be missing something here, but I'll make this suggestion just in case you haven't already considered it. Have your phone register multiple call appearances with the same DN. For instance, my 7960 has three appearances of "2205". Calls are automatically offered to the first available appearance, kind of like what you'd expect. I think this is the behavior you're looking for, but you may be trying to do it he hard way. Cheers, Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hunting S(n)IPs
On Tue, 13 Apr 2004, Andrew Thompson wrote: > Two things: > > 1) Have you looked at call queue's? > > 2) I think you should have been looking at incominglimit, not outgoinglimit, > or possibly both of them together in some combination. > In response to [EMAIL PROTECTED], who wrote: > > > > The [EMAIL PROTECTED] phone does NOT give a BUSY indication even afer the > > first call has been answered. Therefore, Call#2 happily dials 3200 > > again, although 3200 is currently talking. I also tried to limit the > > number of calls going to the phone with outgoinglimit=1 in the > > sip.conf, but that makes no difference either. According to the wiki > > that functionality is broken. Another thing to try is to disable call waiting on the [EMAIL PROTECTED] phone (if call waiting is enabled, it's doing what you've asked it to)... Cheers, Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T100P / ZAP / PRI errors
That makes sense. I've always found the following wording (from the sample file) confusing, not clear about whether "this" and "this span" referred to the span connection on the card, or the span itself: # The timing parameter determines the selection of primary, secondary, and # so on sync sources. If this span should be considered a primary sync # source, then give it a value of "1". For a secondary, use "2", and so on. # To not use this as a sync source, just use "0" So just to clarify some more, I assume if you're using a 4-span card like a TE410P, you would have something like this (in a T1 environment with all connected to a telco switch): span=1,1,0,esf,b8zs,yellow(primary time source) span=2,2,0,esf,b8zs,yellow (secondary time source) span=3,0,0,esf,b8zs,yellow span=4,0,0,esf,b8zs,yellow Do you agree with this? (the "yellow" is an optional parameter) Cheers Scott Stingel www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bisker, Scott (7805) Sent: Tuesday, April 13, 2004 4:59 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] T100P / ZAP / PRI errors In laymans terms. To use your telco's T-1 as the timing source span=1,1,0,esf,b8zs,yelllow To use the internal clock of the card you would use (I'm pretty sure that this would only be used for channel banks, or connections to other PBX hardware. I don't think a telco is going to use your PBX as a timing source) span=1,0.0,esf,b8zs,yellow If you have multiple telco connections on multiple spans you would have something like this span=1,2,0,esf,b8zs,yellow(secondary time source) span=2,1,0,esf,b8zs,yellow (primary time source) span=3,0,0,esf,b8zs,yellow (provide the time source, i.e. channel bank) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Monday, April 12, 2004 9:39 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] T100P / ZAP / PRI errors Now you've got me utterly confused ... So, in layman's terms, if I connect a T100P to a circuit provided by the Telco, and the Telco says that they will provide timing, I have to put WHAT? span=1,0,0,esf,b8zs,yellow this means '0' this span is not a sync source, i.e. the Telco will provide my 8kHz. Could one use '2' with impunity (span=1,2,0,...)? I am still not clear under which circumstances one should use '0' versus '2'. WW - Original Message Follows - > Holy crap people, trim your replies! > > > You didn't say what's at the other end of your PRI line, but you > > might try having the other end be the timing sync source. Try: > > span=1,0,0,esf,b8zs instead. Maybe > that will help. > > We need to get this documented *clearly* once and for all. > > Zaptel T1/E1 hardware either free-runs to its own internal 8kHz time > source, or it tries to lock to the recovered clock from the line. > > Zapata.conf says that timing of 0 means "do not use this span for > timing." Zero does not mean "slave timing", it means not to use this > span as a recovered clock source for timing at all. Timing values of > 1 or 2 mean try to lock the internal clock to the recovered clock > from the span. > > A value of 0 means that this span's recovered clock never gets used as > a timing source. A value of 1 means that this span is the primary > clock source -- If the span is up, try to lock the internal clock to > the clock recovered from this span. A value of 2 means to use this > span for timing only if the primary span is down. > > To reiterate: a value of 0 means that the other end must be locking to > the zaptel's clock or else clock slips will occur. > > Feel free to correct me if I'm wrong, but I am pretty sure I have this > right. :-) > > Regards, > Andrew > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID in Australia
G'day Adam, This drove me nuts for a few days just recently (only fixed it yesterday in fact, and I've not had a chance to update any doco anywhere yet). On Wed, 14 Apr 2004, Adam Goryachev wrote: > Actually, now that I look at the file again, I can also see: > Line: 80 > /* Typically, how many rings before we should send Caller*ID */ > #define DEFAULT_CIDRINGS 1 Yes, that's one of the changes. Change this to #define DEFAULT_CIDRINGS 2 > I am in Australia, which I think expects callerid at a different time to > other countries Although other people have told me callerid is > working correctly for them You're right, our callerid must wait a little longer after the first ring burst (but otherwise is US Bellcore FSK). In callerid.c, on or about line 467, change the value 4000 to 5600 [1]. Recompile, reinstall, shutdown and restart. I have one phone here that works with standard Asterisk, so when another phone did not work I thought it was the phone. Then a third phone did not work, so I started looking for the problem. I think that early phones were built to the US standard, and so will work with US callerid, but the Australian standard was changed later so newer phones need the callerid data to arrive later. Cheers, Vic Cross [1] 5600 gives you 700ms, which is the time implied by a callerid test program I found. I did not want to pay $40 to get a copy of the Australian standard (especially since I can get commentary about the Bellcore standard off the Net), so I don't know exactly what it's supposed to be, but 700ms works for me. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Lucent Phones
> -Original Message- > From: Gregory Junker > > On Mon, 2004-04-12 at 11:28 -0400, Troy Settle wrote: > > At this point, I'm using straight Asterisk, with a a PSTN > gateway at a data > > POP passing calls via IAX to my PBX here in the office. > > Who is the PSTN gateway provider? > > The only CLEC around here that is seriously considering any > sort of VoIP > commercial service is Time Warner Telecom (TWTC), our current telecom > provider, and I have no details on what they are considering. If > VoicePulse had a reason to offer PSTN local exchange service in this > area I'd drop TWTC like a bad habit...I'd then settle for the > Cincinnati > Bell DSL or some other form of lower-cost business-class broadband for > IP data access. KMC Telecom is my CLEC. I'm colocated with them at their central office. I have a DS3 for bringing PRI into my Lucent TNT. The TNT can function as a rudimentary switch and has the ability to generate T1/PRI that plug right into my * box. So, in essense, I'm my own PSTN gateway provider. > > > > > FWIW, you should be able to completely eliminate the > Connectreach and bring > > your T1 directly into *. You just need to find out what > channels on the T1 > > are used for voice, and which are used for data. Using a > T400 or TE405, you > > can cross connect the data channels out to another T1 to go > into your > > router. > > TWTC has examined the T100P and informed me that it's > impossible, since > their IBL uses proprietary formatting and signalling. Also, I ought to > be able to use the data channels directly, according to > Digium, since my > proposed Asterisk box is also our router. > > If they are lying to me (which I doubt...they have a vested > interest in > using a proprietary method), then as for finding out which > channels are > used for what is as simple as trial-and-error and a cell > phone. ;) (and > of course, the $400 or so to pick up a T100P to try it out...) I am > guessing first four are voice and next 12 are data. I'm rather confused by this. It was my understanding that the connectreach was nothing more than a glorified channel bank with IP routing capabilities (I have 2 customers with 6 voice lines, and 384k of data that's handed off as ethernet by the connectreach. The voice lines come off a 50pin telco connector. If TWTC, like my CLEC, offers the connectreach at no additional cost, then I seriously doubt that they would lie to you. Returning the connectreach would save them some small amount of money at the end of the day. If their solution is propriatary, that's fine, but I don't see how/why they wouldn't be able to reprovision the T1 as a normal circuit. If you can get Digium to give you a 30 day refund window, then I'd say that it's well worth it to give this a try. -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial Plan Format Strings
I had exactly the same question. There is some really useful documentation
on voip-info.org regarding the extensions.conf syntax.
-Original Message-
From: Brian Cuthie [mailto:[EMAIL PROTECTED]
Sent: 13 April 2004 15:19
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Dial Plan Format Strings
Try something like this:
exten => _9NXX,1,Dial,IAX2/[EMAIL PROTECTED]/1${NPA}${EXTEN:1}
...
-brian
Nik Martin wrote:
>In the absence of "The Definitive Guide to Asterisk Dial Plans" book,
>I'd like to do something possibly unique with the formatting of
>extensions in my dial plan, and am having trouble. We use VoicePulse
>connect, which gives us local DID for inbound and outbound calls (even
>though DTMF tones are not working in Voice Pulse Connect at the
>moment). To dial local numbers, you have to dial the entire number,
>like 1 + area code + number. I'd like to eliminate this by having the
>user just dial 9 + 7 digit number, and have asterisk put the 1 + area
>code (which is in a variable in extensions.conf) in front of it prior
>to sending the request to Voice Pulse. Is this possible?
>
>
>Thanks,
>
>Nik Martin
>
>
>___
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>
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[Asterisk-Users] RE: T100P / ZAP / PRI errors
Hi Mike- It sounds like, from the discussion here, that your setup is already correct. Must be something else causing the occasional red alarm! Should not occur... Cheers Scott -Original Message- From: Mike Sturdee [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 13, 2004 7:08 AM To: Scott Stingel Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] T100P / ZAP / PRI errors On the other end of our PRI line is a telco switch. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] PC based Switchboard application
Hello Pertti, we would be interessted to, if you could send further informations... Thanks Regards Felix Deierlein [EMAIL PROTECTED] -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Pertti Pikkarainen Gesendet: Samstag, 10. April 2004 11:26 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] PC based Switchboard application We have switchboard application ( PC+browser+Java ) with quite a rich feature set. It talks to * via manager port. Works as a call center too. However, it is not open source. If you are interested in, please contact me directly. Best regards Pertti Keith D'Atrio wrote: > Hello All > I am looking for a PC based switchboard application. Cisco > CallManager has a web attendant console that allows you to use the PC > to transfer calls and the like and I was wondering if there was a > similar program compatible with *. > Thank you in advance > Keith D'Atrio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID in Australia
Adam Goryachev wrote: I am in Australia, which I think expects callerid at a different time to other countries Although other people have told me callerid is working correctly for them Forgot to mention there is a patch for this, but it won't patch cleanly against current CVS... http://bugs.digium.com/bug_view_page.php?bug_id=0001007 -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID in Australia
Adam Goryachev wrote: I am in Australia, which I think expects callerid at a different time to other countries Although other people have told me callerid is working correctly for them From what I've been able to guess at Telstra sends a short ~50ms chirp to the phone, the caller id and then the first full ring, other countries such as the US get the first full ring and then caller ID, and at present caller ID seems to work but different ring candacies don't... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #3413 - 14 msgs
I have two grandstream budtetone-100 and cisco 7960g phones. When I talk via speaker phone on either of the phones I get a lot of echo. Any suggestions? Also how do I turn on the mark echo canceller. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, April 13, 2004 4:41 AM To: [EMAIL PROTECTED] Subject:Asterisk-Users digest, Vol 1 #3413 - 14 msgs Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than "Re: Contents of Asterisk-Users digest..." Today's Topics: 1. VoiceMailBox wav file format in EMAIL. (James Gardiner) 2. TDM400P Issues (Jeremy Bogan) 3. Re: TDM400P Issues (Vic Cross) 4. Re: TDM400P Issues (Jeremy Bogan) 5. Re: TDM400P Issues (Christian Hoffmeyer) 6. Re: TDM400P Issues (Jeremy Bogan) 7. Re: ZAPRTC question(s) (Tony Mountifield) 8. Re: TDM400P Issues (Jeremy Bogan) 9. Re: TDM400P Issues (Jeremy Bogan) 10. Re: X100P and NTL (ex Cable + Wireless) (Stephen Davies) 11. Re: TDM400P Issues (Vic Cross) 12. Re: TDM400P Issues (Jeremy Bogan) 13. Re: Dial Outside SIP address from AGI (Ron McMillin) 14. Re: X100P and NTL (ex Cable + Wireless) (Vic Cross) --__--__-- Message: 1 From: "James Gardiner" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Date: Tue, 13 Apr 2004 16:12:15 +1000 Subject: [Asterisk-Users] VoiceMailBox wav file format in EMAIL. Reply-To: [EMAIL PROTECTED] Hi all, I am not sure if tis is a bug but.. Was learning about VM etc to see how it all worked, and I noticed the following.. In the default install, the VM system leaves 3 different copies of the Voice message. Sizefilename 13kbMsg.gsm 13kbMsg.wav 122kb Msg.WAV <- under UNIX we have case sensitive file names of course. I wanted to have a look at these files so loaded them into SOUND FORGE 6. This first thing I noticed was that the LARGER file is of much HIGHER volume. Like it had been normalised to 100% The smaller was file, when loaded into sound forge, did not load properly, only the first 2 seconds loads. Can anyone explain these issues and why they exist? All in all, I was wondering what would be the best format for best quality but with still great compression. I want to archive all calls for a period of time with self expire. (For example dedicate 5 gig disk space to the last number of calls that can fit in the 5gig.) I want to store the best quality possible but also make best use of disk space, so I can store for even longer periods. I was considering ogg but then is occurred to me that GSM or other codecs designed for audio with this frequency response may be better. (But the GSM file above is not as clear as the WAV ones produced.) I was also wondering if the VM system when emailing the audio can be setup to use something like ogg or MP3? Comments appreciated, James Gardiner --__--__-- Message: 2 To: [EMAIL PROTECTED] From: Jeremy Bogan <[EMAIL PROTECTED]> Date: Tue, 13 Apr 2004 16:14:43 +1000 Subject: [Asterisk-Users] TDM400P Issues Reply-To: [EMAIL PROTECTED] Hi, I just got my TDM400P card (2 modules) and i installed it no probs. The card is detected fine, but for some reason when I add the card to zaptel.conf i get the following error: --snip-- ZT_CHANCONFIG failed on channel 2: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? --snip-- My /etc/zaptel.conf looks like: --snip-- fxsks=1-2 fxoks=3-4 loadzone=au defaultzone=au --snip-- I currently have 2 x X100P cards that work no problem. Running a ztcfg - I get: Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Slaves: 04) 4 channels configured. ZT_CHANCONFIG failed on channel 2: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? Anyone have any ideas? I tried doing a search but couldn't really find anything. Thanks! -- jeremy bogan[ [EMAIL PROTECTED] ] segment publishing - design.develop.host --__--__-- Message: 3 Date: Tue, 13 Apr 2004 16:28:09 +1000 (EST) From: Vic Cross <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] TDM400P Issues Reply-To: [EMAIL PROTECTED] G'day Jeremy, On Tue, 13 Apr 2004, Jeremy Bogan wrote: > --snip-- > ZT_CHANCONFIG failed on channel 2: Invalid argument (22) > Did you forget that FXS interfaces
RE: [Asterisk-Users] Invalid module format in 2.6.5 after running make linux26
Scott Laird wrote: > Since the system clock ticks at 1 kHz in 2.6, is there any reason why > it can't be used (more or less) directly for timing in 2.6? That'd be > a lot easier then hooking into a 1 kHz USB interrupt source. Would someone who is familiar with the 2.6 series kernel please comment on this? I believe we(most of the geekworld) are going to migrate to 2.6 eventually, and not having a decent timing source will be an annoyance. - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T100P Timing Was:T100P/ ZAP / PRI errors
On Tue, 13 Apr 2004 [EMAIL PROTECTED] wrote: > I have no reason to doubt what you wrote, so I already > changed the timing parameter for my system ;). I did have > it set as span=1,0,0, ... Now, please, in what scenario > would one select option '0' and win what scenario would one > use option '2'? > I guess what gets me confused is the terminology 'primary > sync source' and 'span'. The way I read it is span === > digium card. If we are taking timing from the Telco, then Think priorities. 1 - First try to receive sync here. 2 - If that dosn't work, try here instead. 3 - Or if that dosent work, try here. 4 - Lowest priority, if those above won't work. And: 0 - Dont ever receive sync from this line. /Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P and NTL (ex Cable + Wireless)
Stephen Davies wrote: Hi Alex, Indeed the call end termination doesn't work on an NTL line. I'm not so sure it works too well on other lines either. I did some work a while back to add detection of the UK busy/hangup signal on the line, but I never got it working well enough to depend on it. The problem is that it is a single frequency tone. (The US one is dual-tone). Women's voices used to sometimes trigger my detector - causing hangups. The main practical issue is with voicemail, as you say. My final solution was to switch to ISDN. Steve If you look for the cadence as well as the tone, and look over a couple of cycles before deciding it should be robust. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial Plan Format Strings
have to dial the entire number, like 1 + area code + number. I'd like to
eliminate this by having the user just dial 9 + 7 digit number, and have
asterisk put the 1 + area code (which is in a variable in
extensions.conf)
in front of it prior to sending the request to Voice Pulse. Is this
possible?
Sure it's possible! Asterisk can do anything!
exten => _9XXX,1,Dial(Technology/123/1212${EXTEN:1})
See after the: "Technology/123/" there is a "1212" you can make that
your "1 + area code"
Then the: "${EXTEN:1}" dumps in the number that was dialled and chops
off the first (1) digit, which is the 9.
:)
Ben
Benjamin Wakefield
[EMAIL PROTECTED]
http://www.dcsi.net.au/
DCSI - We do Internet.
64 Queen Street
Warragul, VIC 3820 AU
Ph: (+61) 1300 665 575
Fx: (+61) 1300 556 595
-BEGIN GEEK CODE BLOCK-
Version: 3.12
G! d- s: a-- C+ UL++ P+ L++ E W+ N+ o- K- w+$ O--- M-- V? PS !PE Y--
PGP- t 5 X+ R- tv b- DI-- D--- G-- e* h* r- z++
--END GEEK CODE BLOCK--
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RE: [Asterisk-Users] Hunting S(n)IPs
[EMAIL PROTECTED] wrote: > Another observation of something which doesn't work: > > exten => 3200,1,Dial(SIP/3200,20,tTr) > exten => 3200,2,Playback(tt-weasels) > exten => 3200,3,Hangup > exten => 3200,102,Dial(SIP/3201,20,tTr) > exten => 3200,103,Playback(tt-weasels) > exten => 3200,104,Hangup > exten => 3200,203,Dial(SIP/3202,20,tTr) > exten => 3200,204,Playback(tt-weasels) > exten => 3200,205,Hangup > > The [EMAIL PROTECTED] phone does NOT give a BUSY indication even afer the > first call has been answered. Therefore, Call#2 happily dials 3200 > again, although 3200 is currently talking. I also tried to limit the > number of calls going to the phone with outgoinglimit=1 in the > sip.conf, but that makes no difference either. According to the wiki > that functionality is broken. > Two things: 1) Have you looked at call queue's? 2) I think you should have been looking at incominglimit, not outgoinglimit, or possibly both of them together in some combination. - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial Plan Format Strings
Try something like this:
exten => _9NXX,1,Dial,IAX2/[EMAIL PROTECTED]/1${NPA}${EXTEN:1}
...
-brian
Nik Martin wrote:
In the absence of "The Definitive Guide to Asterisk Dial Plans" book, I'd
like to do something possibly unique with the formatting of extensions in my
dial plan, and am having trouble. We use VoicePulse connect, which gives us
local DID for inbound and outbound calls (even though DTMF tones are not
working in Voice Pulse Connect at the moment). To dial local numbers, you
have to dial the entire number, like 1 + area code + number. I'd like to
eliminate this by having the user just dial 9 + 7 digit number, and have
asterisk put the 1 + area code (which is in a variable in extensions.conf)
in front of it prior to sending the request to Voice Pulse. Is this
possible?
Thanks,
Nik Martin
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Re: [Asterisk-Users] Woodpeckers Revisited
Some people have some really wacky ideas about how sampled systems work :-) Regards, Steve Michael Welter wrote: Just when I thought I couldn't be wrong, I was wrong. We have woodpeckers that drill into the arial telephone cables, and water seeps through the holes and partially grounds the tip and/or ring wires causing hum. I thought the hum/buz on my lines was a telco problem. The Qwest HQ noise team assures me that my lines are within spec. Sure enough, when I listen on the test set the lines are clear. The lines terminate at an Adtran 750 channel bank on my * system. When I reconnect the lines to the channel bank and make a call, I get the hum/buz noise. I have replaced every Adtran component (even the chassis), but the hum/buz stays with the lines. From the CO we have a digital fibre optic system which terminates at a neighborhood cabinet. From there, analog copper cables distribute service to the houses. I'm suspecting that the digital-to-analog process doesn't give a smooth analog signal but rather a "stair-stepped" signal, with each step 1/8000 sec in duration (I wish I had a 'scope to confirm this.) The human ear can't hear this stair-stepped signal, so it's ok for POTS use. However, when I put this stair-stepped signal to the channel bank, it converts it back into a digital signal. I'm thinking that, because it's not a smoothed signal, the analog-to-digital process injects hum and buz. Does _anyone_ have more information on this? In the meantime I've had an ISDN circuit installed so as to have digital all the way to the * box. However, I can't get the ASUSCOM ISDNLink card to work with ISDN4Linux :-( Cheers, Mike P.S. The woodpeckers are still eating my house. There is a nest is an exterior wall which is driving my cats nuts! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial Plan Format Strings
Ahh, the {EXTEN:1} must serve to skip the 1st character of the extension,
stripping the 9 off
That's what I needed, thanks!
-Original Message-
From: Austin M. Brower [mailto:[EMAIL PROTECTED] On Behalf Of
Austin M. Brower
Sent: Tuesday, April 13, 2004 9:11 AM
To: Nik Martin
Subject: Re: [Asterisk-Users] Dial Plan Format Strings
Nik,
I use NuFone, but here's how I do it (my area code is 207):
exten => _9NXX,7,Dial(IAX2/[EMAIL PROTECTED]/1207${EXTEN:1})
Good luck,
Austin
On Tue, Apr 13, 2004 at 08:13:19AM -0500, Nik Martin wrote:
> In the absence of "The Definitive Guide to Asterisk Dial Plans" book,
> I'd like to do something possibly unique with the formatting of
> extensions in my dial plan, and am having trouble. We use VoicePulse
> connect, which gives us local DID for inbound and outbound calls (even
> though DTMF tones are not working in Voice Pulse Connect at the
> moment). To dial local numbers, you have to dial the entire number,
> like 1 + area code + number. I'd like to eliminate this by having the
> user just dial 9 + 7 digit number, and have asterisk put the 1 + area
> code (which is in a variable in extensions.conf) in front of it prior
> to sending the request to Voice Pulse. Is this possible?
>
>
> Thanks,
>
> Nik Martin
>
>
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Re: [Asterisk-Users] TAPI driver
Dear Nick Very usefull function driver how can i try it? Thanks in advance Dimitri On Monday 12 April 2004 21:00, Nick Knight wrote: > Hello all, > > > > Just a quick note, I have been putting together a TAPI driver for > Asterisk, this enables the user to perform things like click to dial > from any TAPI enabled app (such as outlook or ACT etc). At the moment it > is very basic and can only perform click to dial but further > functionality will be coming. It uses the Asterisk manager to place > calls. > > > > Please feel free to use it - not much documentation as yet but will be > coming, can be found on sourceforge project name asttapi. > > > > Regards > > > > Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T100P / ZAP / PRI errors
On the other end of our PRI line would be a telco switch. On Mon, 12 Apr 2004, Scott Stingel wrote: > Mike- > > You didn't say what's at the other end of your PRI line, but you might try > having the other end be the timing sync source. Try: span=1,0,0,esf,b8zs > instead. Maybe that will help. > > Regards > Scott Stingel > > www.evtmedia.com > > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sturdee > Sent: Monday, April 12, 2004 11:06 AM > To: Bisker, Scott (7805) > Cc: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] T100P / ZAP / PRI errors > > /etc/zaptel.conf: > > span=1,1,0,esf,b8zs > bchan=1-23 > dchan=24 > loadzone = us > defaultzone=us > > > > On Mon, 12 Apr 2004, Bisker, Scott (7805) wrote: > > > I'm running Zaptel CVS from April 8, LibPRI CVS April 8, and v-1.0 CVS > April 7. With dual T400P cards with no PRI errors at all. Possibly > something driver/config related? Are you timing from your PRI? I remember > getting some PRI errors when my timing config was hosed. Could you post > your zaptel.conf? > > > > > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] Behalf Of Scott > > Stingel > > Sent: Monday, April 12, 2004 12:09 PM > > To: [EMAIL PROTECTED] > > Subject: RE: [Asterisk-Users] T100P / ZAP / PRI errors > > > > > > Mike- > > > > Do you have access to any kind of PRI test set, like a T-Bird or > something. > > > > A red alarm would be easy to capture I imagine - it would be nice to > > confirm that it's a problem particular to your site. The reason I > > mention software is that I've noticed a lot of other messages > > regarding these spurious alarms that immediately clear. I've noticed > > changes in the code having to do with the timing source. > > > > But maybe you're right about the telco doing something odd! > > > > Cheers > > Scott > > > > www.evtmedia.com > > > > > > -Original Message- > > From: Mike Sturdee [mailto:[EMAIL PROTECTED] > > Sent: Monday, April 12, 2004 9:00 AM > > To: Scott Stingel > > Cc: [EMAIL PROTECTED] > > Subject: RE: [Asterisk-Users] T100P / ZAP / PRI errors > > > > I have been seeing this for over a month, and blaming it on our > > generally incompetent telco, so it's definately not a new issue. > > > > > > On Mon, 12 Apr 2004, Scott Stingel wrote: > > > > > This doesn't appear to be a load issue, since normally in that case > > > I would expect you would get a lot of (usually harmless) "frame reject" > > > messages in your /var/log/asterisk/messages file, and perhaps an > > > occasional "missed/double interrupt" message on the console. > > > > > > I wonder if there have been new bugs introduced in the PRI code. > > > I've seen a lot of changes in the timing section of the code at > > > least on the > > dev list. > > > > > > By all means, report it directly to them (a phone call is best). > > > > > > Cheers > > > Scott Stingel > > > > > > www.evt.media.com > > > > > > > > > -Original Message- > > > From: [EMAIL PROTECTED] > > > [mailto:[EMAIL PROTECTED] On Behalf Of Mike > > > Sturdee > > > Sent: Monday, April 12, 2004 8:30 AM > > > To: [EMAIL PROTECTED] > > > Subject: [Asterisk-Users] T100P / ZAP / PRI errors > > > > > > My PRI is being reset at least once a day with the following errors > > > in the logs. > > > > > > zaptel, zapata, libpri, and asterisk are from CVS this morning.. > > > This has been happening for weeks on all versions (including -stable). > > > > > > the T100P card appears to NOT be sharing an IRQ. > > > > > > xenon# cat /proc/interupts > > >CPU0 > > > 0:1203977 XT-PIC timer > > > 1: 3 XT-PIC keyboard > > > 2: 0 XT-PIC cascade > > > 5: 12004595 XT-PIC t1xxp > > > 8: 1 XT-PIC rtc > > > 9:1046347 XT-PIC eth0 > > > 14: 21317 XT-PIC ide0 > > > NMI: 0 > > > ERR: 0 > > > / > > > > > > Is this something I should be seeking support from Digium on being > > > their card? > > > > > > > > > Apr 12 11:04:59 WARNING[1226062640]: PRI: Short write: -1/15 > > > (Unknown error > > > 500) Apr 12 11:04:59 WARNING[1226062640]: Detected alarm on channel 1: > > > Red Alarm Apr 12 11:04:59 WARNING[1192491824]: Detected alarm on > > > channel 2: Red Alarm Apr 12 11:04:59 WARNING[1192491824]: Unable to > > > disable echo cancellation on channel 2 Apr 12 11:04:59 > > > WARNING[1192491824]: Detected alarm on channel 3: Red Alarm Apr 12 > > > 11:04:59 WARNING[1192491824]: Unable to disable echo cancellation on > > channel 3 Apr 12 11:04:59 WARNING[1192491824]: > > > Detected alarm on channel 4: Red Alarm Apr 12 11:04:59 > > WARNING[1192491824]: > > > Unable to disable echo cancellation on channel 4 Apr 12 11:04:59 > > > WARNING[1192491824]: Detected alarm on channel 5: Red Alarm Apr 12 > > > 11:04:59 > > > WARNING[1192491824]: Unable to disable echo cancellation on channel > > > 5 Apr 12 > > >
[Asterisk-Users] CallerID in Australia
Well, Once upon a time, I had problems receiving callerid, and then one
day, Mark was logged into my asterisk box helping with something else,
and I asked him about this, and he showed me a nice tweak to some source
file that made it work.
Some time later, I must have done hundreds of CVS updates, and along the
way, lost the above patch (one liner) and so callerID hasn't worked for
a long time. Silly me thought it must be something wrong with my
isdn4linux, or kernel hacks, or something.
Finally tonight, using a PRI into a TE405P (or maybe TE410P) with
current asterisk cvs stable version, and I called the ttsagi-sayani that
I found a long time ago, and it read out my correct callerid.
BUT, it still doesn't show up on my handset, so, this is where I
remember that I need the magic one line patch.
>From a very quick look at the source code, I think it will need to be
this that needs to be modified:
Line: 532
static int cidrings[] = {
2, /* Right after first long ring */
4, /* Right after long part */
3, /* After third chirp */
2, /* Second spell */
};
Actually, now that I look at the file again, I can also see:
Line: 80
/* Typically, how many rings before we should send Caller*ID */
#define DEFAULT_CIDRINGS 1
So, does anyone know if either of these are the things I need to change?
and what I need to change them to?
I am in Australia, which I think expects callerid at a different time to
other countries Although other people have told me callerid is
working correctly for them
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[Asterisk-Users] T100P Timing Was:T100P/ ZAP / PRI errors
Don & others, Thank you for your answer. The fog maybe lifting ;). The zaptel.conf file has the following in its comments: # # The timing parameter determines the selection of primary, secondary, and # so on sync sources. If this span should be considered a primary sync # source, then give it a value of "1". For a secondary, use "2", and so on. # To not use this as a sync source, just use "0" # You stated: > > In this case you want to use loop timing (use the T1 as a > sync source) so, > > span=1,1,0,esf,b8zs > I have no reason to doubt what you wrote, so I already changed the timing parameter for my system ;). I did have it set as span=1,0,0, ... Now, please, in what scenario would one select option '0' and win what scenario would one use option '2'? I guess what gets me confused is the terminology 'primary sync source' and 'span'. The way I read it is span === digium card. If we are taking timing from the Telco, then the digium card should slave from the Telco and cannot therefore be a primary sync source (for the loop), so I had it set at '0'. After your explanation, it now sounds like we are talking about timing for the BOX, not the loop. Therefore, the LOOP is the Primary Source for the BOX, hence use '1'. If we are generating the loop (fxs role) then there is no Telco to slave off, and the BOX should take its timing from the CARD hence select '0'. If this reasoning is correct, then when would one reasonably use '2' and even '3' etc. Sorry to be a pest about this. WW Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
