Re: [Asterisk-Users] Config TE110P and TDM400 with 2 FXS modules
Hi Dennis,
You need to add immediate=no before your channel assignment; asterisk
will then give you a dialtone when you pick up the handset. Also the
context uniware_sendfax must have the pattern match and associated zap
dial that you need.
HTH.
Cheers,
Paul
www.austechpartnerships.com
t) +61 (0)3 9221 0888
SIP) [EMAIL PROTECTED]
IAX) [EMAIL PROTECTED]
IAXtel) 1700-482-8273
ATP Centrex)
Dennis wrote:
signalling=fxs_ks
context=uniware_sendfax
channel =33-34
-=-=-=-
What we want to be able to do is plug a standard telephone into an fxs
port on the tdm card, and have it get a line from the E1 when the
handset is picked up.
exten = _NXXX,1,Dial(Zap/g1/${EXTEN}|20,t)
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Config TE110P and TDM400 with 2 FXS modules
Hi Dennis, Yes, you've got it correct. Glad its working for you. Cheers, Paul www.austechpartnerships.com t) +61 (0)3 9221 0888 SIP) [EMAIL PROTECTED] IAX) [EMAIL PROTECTED] IAXtel) 1700-482-8273 ATP Centrex) Dennis wrote: Hi, Thanks for that Paul, it has solved that problem at least. In doing so however it broke the incoming calls on the other card. To fix this we need both immediate=yes set before the TE110P channels, and then immediate=no after that but before the TDM channel assignment. It all works for now, but is this how the configuration for these files is meant to be? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Skilled API consultant required - preferably with Salesforce.com intergration
Hi Dean, We at ATP have a range of resellers/integrators on our system to provide solutions around Australia. Get them to contact us, and we'll put them in touch with the nearest integrator with the correct skillset. Cheers, Paul www.austechpartnerships.com t) +61 (0)3 9221 0888 SIP) [EMAIL PROTECTED] IAX) [EMAIL PROTECTED] IAXtel) 1700-482-8273 ATP Centrex) Dean Collins wrote: Hi all, I was just on the phone with a B2C company in Australia who are looking to integrate an Asterisk solution with their Salesforce.com CRM platform. They are looking for a consultant/team to provide the following functionality * Complex IVR Eg can interface via API into Salesforce for customer service interaction and product initiation. * Call centre Eg basic queues, fallovers etc, reports, * Salesforce.com CTI integration, screen pop, outbound calling etc. If anyone on this list has extensive API experience with ivr and preferably with some salesforce.com screen pop experience then please email me and I’ll pass along their details. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Driver not configuring correctly on TE210P for CCS
Hi Alex, Have you checked that your jumper setting on the card has been shorted for E1. Its open by default to T1 - which overrides your zaptel.conf settings. Cheers, Paul Alex Barnes wrote: Dec 28 12:51:55 caudi_apx1 kernel: TE2XXP: Launching card: 0 Dec 28 12:51:55 caudi_apx1 kernel: TE2XXP: Setting up global serial parameters Dec 28 12:51:56 caudi_apx1 kernel: Found a Wildcard: Wildcard TE210P Dec 28 12:51:56 caudi_apx1 kernel: About to enter spanconfig! Dec 28 12:51:56 caudi_apx1 kernel: Done with spanconfig! Dec 28 12:51:56 caudi_apx1 kernel: Registered tone zone 4 (United Kingdom) Dec 28 12:51:56 caudi_apx1 kernel: About to enter startup! Dec 28 12:51:56 caudi_apx1 kernel: TE2XXP: Span 1 configured for ESF/B8ZS Dec 28 12:51:56 caudi_apx1 kernel: wct2xxp: Setting yellow alarm on span 1 etc/zaptel.conf # Define the E210P span=1,1,0,ccs,hdb3,crc4 bchan=1-8 dchan=16 unused=9-15,17-31 ztcfg: [EMAIL PROTECTED] ~]# ztcfg -vvv -d 2 Line 154: loadzone=uk Line 162: defaultzone=uk Line 224: span=1,1,0,ccs,hdb3,crc4 Line 225: bchan=1-8 Line 226: dchan=16 Line 227: unused=9-15,17-31 End of File -- www.austechpartnerships.com t) +61 (0)3 9221 0888 SIP) [EMAIL PROTECTED] IAX) [EMAIL PROTECTED] IAXtel) 1700-482-8273 ATP Centrex) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] recall button using tdm400 Australia
I am sorry, you lost me here? You mean set rxflash to the max and flash to the min time? What times should I use? Currently I have: pulsedial=no flash=100 rxflash=100 Hi Brian, flash=80 rxflash=120 Cheers, Paul -- www.austechpartnerships.com t) +61 (0)3 9221 0888 SIP) [EMAIL PROTECTED] IAX) [EMAIL PROTECTED] IAXtel) 1700-482-8273 ATP Centrex) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] recall button using tdm400 Australia
Brian May wrote: Hello, How do I get the recall button working on a phone attached to a TDM400 FXS port using Asterisk? I did a web search, and found people with exactly the same problem, but no solution. I suspect the timing is set for American standards, is it possible to get it to work with Australian phones? Thanks. Hi, Yes you can - change ZT_DEFAULT_FLASHTIME from 750ms to 100ms in zaptel.h Cheers, Paul -- www.austechpartnerships.com t) +61 (0)3 9221 0888 SIP) [EMAIL PROTECTED] IAX) [EMAIL PROTECTED] IAXtel) 1700-482-8273 ATP Centrex) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] recall button using tdm400 Australia
Hi Brian Ideally I would like to continue using the pre-built binary for Debian. If possible. Unfortunately - the only way is to submit a patch Also, I would assume that rxflash and/or flash in zapata.conf does the same thing, but so far I haven't had any luck. As such, I am not entirely convinced changing the source code would help either... You are correct - rxflash and flash in zapata does the equivalent, but I should also have said in my earlier post that you need to drop the max pulse time (for pulse dialling) to be less than the hook flash timing. Default settings for max pulse is 150ms, which inteferes with Australian hook flash of 100ms. - It does work, as it is running in our setup here. When I push recall, I get an interruption in the call, but either the call is disconnected (no, it isn't on hold) and I get the dialtone, or (more likely) I get the same call back again. The fact I get both behaviours seems weird. Perhaps it is a bug in asterisk 1.0.9? You need to set rxflash and flash as max and min times for the hookflash to work. Cheers, Paul -- www.austechpartnerships.com t) +61 (0)3 9221 0888 SIP) [EMAIL PROTECTED] IAX) [EMAIL PROTECTED] IAXtel) 1700-482-8273 ATP Centrex) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Atick Certification on FXO Modules (Australia)
Christopher Lee wrote: Out of interest is there any estimated date for the TDM400 FXO modules receiving A-tick certification? And has anyone compared the FXO modules with the X100P on Australian exchanges/equipment? Do they perform any better than the X100? Cheers, Chris Lee ___ 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 Hi Chris, We at ATP www.austechpartneships.com are Digium's distributor in Australia/NZ, and are responsible for the A-tick process. Its currently in the testhouse, but we have no completion date as yet. We will be advising the community when that would be done here on the list, on our website and of course on Digium's website. Cheers Paul ___ 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] Limiting incoming SIP calls Original CallerID on transfer
From: Erik Barker [EMAIL PROTECTED] We are currently using Polycom IP600 VOIP phones for our office which are capable of handling 2 calls per SIP registration. What we're finding is when staff are on the phone, Asterisk will pass them a second call which will show up on their display, and an audible beep is heard over the phone (regular call waiting). I would like to limit the number of calls sent to each phone to 1 call only; otherwise respond as being busy. I have looked at trying to accomplish this in the sip.conf by using the 'incominglimit' and 'outgoinglimit' parameters, however, the only one that *seems* to work is the 'incominglimit'. This prevents further calls from reaching the phones, rings busy, but does not allow our phones to initiate a 2nd call OR transfer their existing call. The 'outgoinglimit' parameter does not seem to have any effect on limiting whatsoever. Is there a way to limit calls passed to the phones from Asterisk and also allow each phone to initiate 2 calls or transfer calls (disable call waiting)?? I have also looked at the WIKI for the parameters listed above and it *appears* that 'outgoinglimit' should do what I want, however it also states that this function has been disabled?? The _outgoinglimit__ is currently disabled in the source code of the SIP channel. http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20incominglimit Hi to all, Sorry for not responding a lot earlier to the issue of incominglimit as I have been absolutely flat chat, no excuses however, hope I can shed some light on this issue. When I first worked on this, incominglimit and outgoinglimit was already in place, but it didn't work in preventing call waiting at all. Customers of ours were on GS phones, and all of you know the problem with that on call waiting. The trouble is, with call waiting you can't really separate incoming and outgoing cleanly. The main aim was to prevent call waiting whenever the phone was in use (it should be changed to inuse as in the CLI command sip show inuse) whether on an incoming or outgoing call. Once you separate the two functions, call waiting will fail, ie when on outgoing call, the incoming counter will not increment, therefore you WILL receive an incoming call and vice-versa. I know that what I've done is not pretty (ie to incorporate the two into one), but it does fix the problem with call waiting on SIP phones that can't or won't handle it. I am open to ideas or suggestions and will put in the time to fix this properly once an for all. Please reply either on or off list. Paul ___ 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] Asterisk + BudgeTone (behind NAT)
From: Owen Kelso [EMAIL PROTECTED] Sent: Sunday, January 11, 2004 10:07 AM Subject: [Asterisk-Users] Asterisk + BudgeTone (behind NAT) On the NAT'ed side I have the BudgetTone set up to use STUN and ports 5060 for SIP and 19000 for RTP. The firewall that performs NAT forwards ports 5060 and 19000-19100 UDP to the phone. Hi Owen, Even though your GS is behind a NAT, it shouldn't be set for STUN unless you're actually using a STUN server. I have GS installed in many locations behind a NAT but without the STUN option set. Paul ___ 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] Fwd: new cvs build failure
- Original Message - From: Martin [EMAIL PROTECTED] mkdir -p /sbin install -m 755 ztcfg /sbin make: install: Command not found make: *** [install] Error 127 [EMAIL PROTECTED] zaptel]# Why won't zaptel make install ? Martin Martin, Looks like somewhere along the line, you lost the install binary, which is part of your standard installation. Doing a whereis install - /usr/bin/install. Basically it copies the file to the target directory and sets the attributes. Paul ___ 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] Strange Call waiting problems - SNOM 200 Grandstream Budgetone
- Original Message - From: Michael [EMAIL PROTECTED] Sent: Thursday, January 08, 2004 10:40 PM if you are on a call on the Budgetone 101 and a 2nd call is received, instead of a call waiting beep being played, it rings on the handset speaker! which makes it almost impossible to speak to the 1st caller, but if you hang up the 1st call, the phone rings and it is possible to answer the 2nd call normally. Anyone got any ideas on what could be causing these problems ? Hi Michael, GS does not really support callwaiting, and you don't have the ability to disable from the phone configuration. You can however disable in '*', by adding incominglimit=1 for each GS phone in sip.conf. Also ensure that you a username=blah where blah is the same as your phone definition. HTH. Paul ___ 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] Newbie - MWI
- Original Message -
From: John Coll [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 05, 2004 9:07 AM
Subject: [Asterisk-Users] Newbie - MWI
Sorry for the partial post a moment ago
With help I got two phones communicating - PCMA/PCMU was the problem.
Next stpe is to try voicemail. VM works fine, I can leave a mesage and
then
retrieve it - but no MWI on the phone and no stutter dialtone.
I promise I've spent the requisite 4 hours + on google etc. but have
really
no further ideas.
The setup is 2 Grandstream phones (latest firmware) and an asterisk on a
LAN. The cofig files I am using are shown below. Any suggestions would be
appreciated.
john
-
;
; liza:/etc/asterisk/sip.conf
;
[general]
port = 5060
bindaddr = 0.0.0.0
externip = 10.0.1.198
[5702]
type=friend
host=dynamic
context=johnhome
reinvite=no
canreinvite=no
qualify=300
callerid=John workroom #1 5702
mailbox=5702
disallow=all
allow=ulaw
allow=alaw
; dtmfmode=rfc2834
dtmfmode=info
username=5702 ; not convinced this is needed
nat=yes
[5703]
same as above in effect
-
;
; liza:/etc/asterisk/extensions.conf
;
[general]
static=yes
writeprotect=no
;
[globals]
CONSOLE=Console/dsp
[johnhome]
exten = 5702,1,Dial(SIP/5702,20,Ttr)
exten = 5702,2,Voicemail(u5702)
exten = 5702,102,Voicemail(b5702)
exten = 5702,103,Hangup
exten = 5703,1,Dial(SIP/5703,20,Ttr)
exten = 5703,2,Voicemail(u5703)
exten = 5703,102,Voicemail(b5703)
exten = 5703,103,Hangup
exten = 88,1,VoicemailMain(${CALLERIDNUM})
-
;
; /etc/asterisk/voicemail.conf
;
[general]
format=wav49|gsm|wav
[johnhome]
5702 = 5702,John Coll,john
5703 = 5703,John Coll,john
John,
You have your voicemail within the johnhome context, so for your sip
config, your phone entry for voicemail should be [EMAIL PROTECTED]
Paul
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Callwaiting / limits?
- Original Message - From: Stephen J. Wilcox [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, December 22, 2003 12:54 AM Subject: [Asterisk-Users] Callwaiting / limits? Hi, I'm using grandstream phones, when on a call and a second call comes in the call waiting indication is to play ringing which means you cant actually hear your original call. I want to stop this but cant, heres my options 1. Change the callwaiting indication, I assume this is produced by the phone and in the case of grandstream there seems to be no way to control this. 2. Use of incoming/outgoing limit in sip.conf. This works okay except there is no 'absolute limit' type option, meaning that if i place an outbound call from my grandstream it is possible to send a new incoming call in and we have the call waiting again. I assume others have found this, whats the solution? Steve Hi Steve, The incominglimit applies to both incoming and outgoing calls, so long as I'm on the phone, any incoming call gets sent to voicemail. Use the sip show inuse on the CLI to check the inuse counter is being incremented when on a call, whether receiving or outgoing. Is anybody else having this problem ? Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best SIP PHones to buy ?
- Original Message - From: Michael T Farnworth [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, December 21, 2003 4:31 AM Subject: Re: [Asterisk-Users] Best SIP PHones to buy ? We have bought around 30 Grandstream phones, both BT101 and BT102. In general the phone is reasonable, but it does have limitations. Notable issues for users tend to be the lack of any sort of consultative transfer or easy access to conference calling. Also its 'call waiting' facility is a rather annoying and loud normal ringing noise, rather than the usual 'beep beep' that people are probably used to and you can't disable the call waiting feature. Entering numbers also has problems as if you dial too quickly you tend to lose digits, even if you heard the tones and saw them appear on the display. Michael, I've put in a patch to fix call waiting for SIP phones, which is now part of the CVS. For each UA in sip.conf, insert the line incominglimit=1 Also, ensure that you have a usename=blah. This will stop the call waiting for the Grandstream phones. HTH. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] unable to configure my Grandstream phone
- Original Message - From: Balaji NJL To: [EMAIL PROTECTED] Sent: Monday, December 15, 2003 8:47 AM Subject: [Asterisk-Users] unable to configure my Grandstream phone snip Attempting native bridge of SIP/2003-b895 and SIP/2000-53e2 WARNING[5126]: File chan_sip.c, Line 1954 (process_sdp): No compatible codecs! -- Got SIP response 481 Call Leg/Transaction Does Not Exist back from 192.168.0.58 and then the call drops. When i am making a call using Grandstream ph, it rings the other side when they pick up the phone the call then drops. then i get the above error message. the follwoing us sip and Grandstream conf [general] port = 5060 bindaddr = 0.0.0.0 context = bogon-calls ;context = default disallow=all allow=gsm Balaji, Grandstreams do not support GSM. Options available can be seen on the GS config page. Unless you purchase G729s (for low bandwidth), your only choice is G711. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] incominglimit stuck in app_queue
- Original Message - From: Paul Lambert [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: Paul Liew [EMAIL PROTECTED] Sent: Wednesday, December 03, 2003 4:16 AM Subject: Re: [Asterisk-Users] incominglimit stuck in app_queue I've seen this same thing. But it doesn't happen only for phones using the queue I believe it is a bug in the chan_sip driver. What I have found is that when a phone sip phone is unplugged/not registered and a call comes in it increments the counter and doesn't reset the counter when the phone reregisters. I submitted a bug report on this http://bugs.digium.com/bug_view_page.php?bug_id=601 I think this bug should be reopened: http://bugs.digium.com/bug_view_page.php?bug_id=408 Hi All, I've put up the latest patch to fix this. Basically its caused by a call coming on a SIP channel when it becomes unregistered in between registrations and timeouts. Please test and provide some feedback. Thanks. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call waiting disable in sip
- Original Message - From: Anton Yurchenko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, November 29, 2003 3:34 AM Subject: Re: [Asterisk-Users] call waiting disable in sip what would happend if all operators are busy? would app_queue exit? would it schedule the call to wait and until the number of them reaches the maxlen ( it is defined in queues.conf) ? Hi Anton, Before I submitted the patch to bugtracker to fix this problem, I tested this for both the Dial and Queue apps, and it works as per other channels, ie when all the queue operators are busy, the calling party will stay in the queue until an agent becomes free. All parameters within the queue.conf apply. The only parameter you need to specify in sip.conf is the incominglimit for this to work. For GS phones, set this to 1. By the way, this is no longer a patch as it has been incorporated into the CVS as of 26/11/03. Let me know if you encounter any problems. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] double-dial in SIP Grandstream
callwaiting=no is not supported by chan_sip. Call waiting enabling/disabling is a function of SIP phones. Unfortunately, GS does not support disabling call waiting as yet, so I've had to put in a patch to overcome the problem. Look under http://bugs.digium.com/bug_view_page.php?bug_id=408. You need incominglimit=1 to stop the ringing caused by callwaiting when you are on the phone. Paul - Original Message - From: Bisker, Scott (7805) [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 19, 2003 12:57 AM Subject: RE: [Asterisk-Users] double-dial in SIP Grandstream Marc, This is the typical behavior for call waiting. While you are initiating a call, people who call your number will get a busy signal until your first call connects. Once the call connects, the number 2 caller will hear a ring until you pickup. If you want to disable callwaiting then put callwaiting=no in sip.conf for that particular alias. [alias] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer
Hi Mick, It's going to be hard for anybody here on the list to help you, unless you are more specific, ie, what you did exactly to get a crash, and console output (with verbose set) debugs, logs (under /var/log/asterisk) and some configuration files. We'll be in a better position to help you then without trying to be mind readers. Paul - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 17, 2003 6:49 PM Subject: RE: [Asterisk-Users] Call transfer WARNING[1242952640]: File app_dial.c, Line 318 (wait_for_answer): Unable to forward voice This is what I get And a crash Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, 17 November 2003 5:14 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Call transfer On Mon, 17 Nov 2003 [EMAIL PROTECTED] wrote: Does anyone know how to make Calls auto transfer to a mobile if no one answers ?? suppose your mobile number is +923008508070 exten = 15,1,Dial(IAX/farfon|30) ; try for 30 seconds on IAX exten = 15,2,Dial(Zap/1/03008508070|45) ; then try for 45 on my cell - wasim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hunt groups and SIP?
Also, check my patch http://bugs.digium.com/bug_view_page.php?bug_id=408 which does fix incominglimit/outgoinglimit. Stops callwaiting on sip phones. Hope that helps. Paul - Original Message - From: David Gomillion [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, November 18, 2003 7:29 AM Subject: RE: [Asterisk-Users] Hunt groups and SIP? Look at the Queue application... it will probably fulfill your needs. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Sizemore Sent: Monday, November 17, 2003 2:25 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Hunt groups and SIP? I would like to setup a hunt group, not a group ring, using sip phones. Anyone done this with sip devices? Comments suggestions? I have not had much luck with the outgoinglimit=1, incominglimit=1 stuff that I would need to get busy extinctions to work right, which is why I'm asking on the list. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX version 0.9.2 available for download
Is it possible to incorporate iLBC codec, some hotels only allow 28.8 dial-up links, and then your product will be really useful on the road How _does_ * work on dialup? I have never tried. I know you have an immediate 200-300ms lag but how is it otherwise? I've used a GS plugged into a notebook over a 33.6 link on G729 - works very well. However G729/GSM uses about 31kbps, makes it hard to push it down a 28.8 link, iLBC uses about 24-25kbps and would be ideal. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] contact
Sorry to do this to the list, but I have no choice . Walker, I've been trying to send you an email off-list for the last couple of weeks, but one of my mail-hops is failing, do you have alternative address that I can try ??? Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] First AGI help..
Hi ?php // From Kapjod's sample.. ob_implicit_flush(true); set_time_limit(0); $err = fopen(php://stderr,w); $in = fopen(php://stdin,r); $out = fopen(php://stdout,w); //This works.. fputs($out, Verbose \Calling phone\n); // This doesn't fputs($out, exec(Dial(sip/2012)\n); fclose($in); fclose($out); fclose($err); ? You'll find its to do with your syntax - show agi exec produces Usage: EXEC application options Executes application with given options. Returns whatever the application returns, or -2 on failure to find application ie use spaces not (. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snatching calls
- Original Message - From: Billy Huddleston [EMAIL PROTECTED] how could you do this with sip and VOIP? From: Steven Critchfield [EMAIL PROTECTED] You want to look into call groups and pickup groups. To pickup the call you use *8#. from /usr/src/asterisk/configs/zapata.conf.sample ; ; Ring groups (a.k.a. call groups) and pickup groups. If a phone is ringing ; and it is a member of a group which is one of your pickup groups, then ; you can answer it by picking up and dialing *8#. For simple offices, just ; make these both the same ; callgroup=1 pickupgroup=1 As per what Steve says, the same applies for SIP, check /usr/src/asterisk/configs/sip.conf.sample Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX Soft phone v0.9.1 is available for downlaod...
Hi Dan, Nice goingsome testing feedback. Testing your new client with voicemail - after entering password (4 digits), 1 for new messages, then no further digits can be sent. So far everything else OK. Regards, Paul - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, November 04, 2003 8:19 AM Subject: Re: [Asterisk-Users] DIAX Soft phone v0.9.1 is available for downlaod... Hi, - Original Message - From: Florian Overkamp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 03, 2003 10:18 PM Subject: Re: [Asterisk-Users] DIAX Soft phone v0.9.1 is available for downlaod... ... Cool, that would pretty much do it. Now, in earlier mails and also this morning, even though the post got caucht because it wasnt sent from my usual mailbox :- I inquired about future 'skinability' of the application: I'll put this on the wish list. I think that the ringtones (even speach as ring) will come first Also, do you think its possible to block access to the user/server-config part (maybe via a flag in the .ini file) ? It might confuse the less computer-able users :-) I even think to a way to protect the config file to an unauthorized external editing. As you can see, the credentials are a little bit crypted (not difficult to guess, but not for the normal user). I can build a special release where you can edit credentials only through a separate tool. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Campon feature
Hi Walker,
I've put that up on
http://bugs.digium.com/bug_view_page.php?bug_id=464
Paul
- Original Message -
From: Walker Haddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, October 30, 2003 11:50 AM
Subject: Re: [Asterisk-Users] Campon feature
On Thu, Oct 30, 2003 at 10:28:32AM +1100, Paul Liew wrote:
Hi all,
Having fixed my problems with the call waiting ringing on the GS phones,
I needed to extend that with a campon facility (available on some legacy
systems - sort of callwaiting without phone ringing). I've managed to
implement that by adding/modifying app_queue.c. Basically, when calling the
SIP phone, and if busy, I can camp the call onto that extension, with MOH,
allowing the caller to drop out to voicemail or other priority, if they wish
to. You just need to record an additional voice file as instructions for the
caller in the campon function. Sample of extensions.conf
Paul, this looks great. I'd like to try it.
--
DataCrest, Inc. -- Technically Superior **
Walker Haddock http://www.datacrest.com
DataCrest, Inc.e-mail: [EMAIL PROTECTED]
1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216 fax: 1-205-823-7838
***
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: call waiting beep
I am thinking of coding a solution using variables, Cut, and ChanIsAvail. here is what i'm thinking of doing Create a variable that contains the string SIP/gs1SIP/gs2SIP/gs3 ... etc check each phone with ChanIsAvail, and use Cut to remove its representation in the string (if its not avail) then do a dial( variable ) If that doesn't work for some reason, i will try the patch. Thanks for the info. -Sean R. I don't think that will work, its been tried before, ChanIsAvail seems to work only for Zap devices. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call waiting beep
Sean Rodger wrote: Is there anyway to turn off the call waiting beep in the grandstream and/or cisco ata186? I have a dial statement in my extensions.conf that rings 5 phones at the same time by combining them with the in the dial statement. i.e.) exten = blah,blah,Dial(SIP/GS1SIP/GS2SIP/GS3SIP/ata186aSIP/ata186b,25,t) If one of those lines is being used, then the user gets a really loud call waiting beep, and on the ata186, also an inband callerid noise (perhaps changeable on the ata186). thanks for any info. Sean Rodger The loud call waiting noise from the GS is supposed to be resolved in the next firmware release.. Hopefully it will be out soon.. Later.. Until GS comes out with their release, you can try the patch I put up on http://bugs.digium.com/bug_view_page.php?bug_id=408 There is a problem with that Michael pointed out to me, re holding and using ChanIsAvail which I'm looking into right now, but it works for normal incoming/outgoing calls via Dial as per your scenario. Check it out and let me know how you go. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Already on the phone?
Michael, A couple of things - having a quick look at the app_ChanIsAvail code - it seems that it is designed for Zap devices, so using them on any SIP phones would not provide the expected result. Secondly, which SIP phone are you using, I can't put calls on hold and make further calls without parking them. In either case, I suspect the call has been palmed off to asterisk, otherwise you wouldn't be able to make further outgoing calls (the incoming limit would block it). The inuse limit would apply while you are actually in a call. Does it work when you take the original call back off hold ?? I think having the ability to change the incominglimit from the dialplan might be a good idea, but I think prior to any discussion on that, this patch would have to be proven to work reliably and if approved by Digium - put into the CVS. Paul I put it on hold and placed a few other calls. Then I see: pbx1*CLI sip show inuse UsernameincomingLimit outgoingLimit 12125550011 0 N/A 0 N/A 1212555 0 N/A 0 N/A 1212555 0 N/A 0 N/A 12125550029 0 N/A 0 N/A 12125550012 0 N/A 0 N/A 1212555 0 1 0 N/A 12125550028 0 N/A 0 N/A 12125550014 0 N/A 0 N/A So it looses status of existing call somehow. Now callwaiting is there again. It seems that the status is lost after calling chanisavail application, although I'm not sure about that. Also if I can make a suggestion it would be great not to have incominglimit set statically per client, but have an application to change it from dialplan (have no idea how hard it is to implement). If there are other ways to check if the line is already in use or turn on/off callwaiting on SIP clients, that would also be very nice and desirable feature. Thanks. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Campon feature
Hi all,
Having fixed my problems with the call waiting
ringing on the GS phones, I needed to extend that with a campon facility
(available on some legacy systems - sort of callwaiting without phone ringing).
I've managed to implement that by adding/modifying app_queue.c. Basically, when
calling the SIP phone, and if busy, I can camp the call onto that extension,
with MOH, allowing the caller to drop out to voicemail or other
priority,if they wish to.You just need to record an additional voice
file as instructions for the caller in the campon function. Sample of
extensions.conf
[macro-ext];; Standard extension
macro:; ${ARG1} - Technology/Number;exten =
s,1,Dial(${ARG1},30|tr)exten = s,2,Voicemail(u${MACRO_EXTEN})exten
= s,102,Campon(${ARG1}) ; phone busy
camp the caller onexten = s,103,Voicemail(b${MACRO_EXTEN}) ; caller
decides to leave voicemailexten =
s,203,Directory(Default) ; caller decides
to call another extension
[extensions]
; our extensionsexten =
2001,1,Macro(ext,SIP/2001)
If there is any interest, I'll post it up to the
bugtracker as a feature ...
Paul
Re: [Asterisk-Users] Already on the phone?
Michael, I've added a patch a week ago on to bugtracker to fix this - feel free to try it and let me know http://bugs.digium.com/bug_view_page.php?bug_id=408 Paul - Original Message - From: Michael Ulitskiy [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, October 29, 2003 10:31 AM Subject: [Asterisk-Users] Already on the phone? Hi, I'm wondering if there's a way within a dialplan or AGI to find out if an extension (SIP client) is already in use and the person is already on the phone? By default the channel is assumed available and callwaiting tone is transmitted to the called extension. AFAIK there's no way to turn off callwaiting from within the dialplan. I need to avoid the callwaiting behavior in some cases and pass the call to another extension if called extension is already in use. Is this possible with asterisk? I've tried chanisavail application, but since callwaiting is enabled it always returns true. Thanks. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Waiting on SIP phones
- Original Message - From: Walker Haddock [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, October 22, 2003 11:10 PM Subject: Re: [Asterisk-Users] Call Waiting on SIP phones Subject: Re: [Asterisk-Users] Call Waiting on SIP phones Paul, I applied the patch successfully last night on the CVS from last night. I set the incominglimit=1 in sip.conf. I am still getting the ring in the Grandstream phones. I posted a bug report on the bugtracker. If you would like more system information, please let me know. thanks, Walker Hi Walker, I tried to duplicate your problem, but I couln't, could you please let me know the exact calling situation, so I could try and duplicate it. Also noted that you have a repeat SIP number in your Dial command for extension 203 (don't think that could impact it). Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Waiting on SIP phones
- Original Message - From: Walker Haddock [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, October 22, 2003 5:50 AM Subject: Re: [Asterisk-Users] Call Waiting on SIP phones Paul, I'm getting a patch error when I diff to the chan_sip.c that I just got from CVS this morning. It looks like this morning's version hasn't changed from the version I had from 9/24/03. Here's the .rej file output: Walker, in case I did something wrong while posting the patch on to the list - try the patch I've put up on bugtracker http://bugs.digium.com/bug_view_page.php?bug_id=408 I've applied that patch on to the latest cvs and its ok. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Waiting on SIP phones
Hi All,
This is the first time I'm submitting a patch, and I hope it fixes more than
it breaks. I'm putting it here, since John Todd mentioned a while ago about
the heavy load Mark and crew have at Digium (doing such good work), so I
thought all of us could test this first, and if ok submit for inclusion in
CVS later if appropriate.
This is an extension to work done earlier (sorry I can't remember your name)
with regard to incominglimit and outgoinglimit to prevent that horrible
call waiting in your ear for Grand Stream phones. It worked only if you
received a call and only once.
I've tested this on my system between GS and X-ten, using normal extension
and queue calling, and it seems to work ok for me. No call waiting at all,
whether I originate or receive calls. You only need to set incominglimit=1
for each sip phone that you wish to block call waiting.
Anway, enough blurb, please test and let me know how you go .
Paul
--- chan_sip.c.save 2003-10-20 21:51:52.0 +1000
+++ chan_sip.c 2003-10-21 09:02:33.0 +1000
@@ -959,7 +959,9 @@
return 0;
}
switch(event) {
+ /* Incoming and outging affects the inUse counter */
case DEC_IN_USE:
+ case DEC_OUT_USE:
if ( u-inUse 0 ) {
u-inUse--;
} else {
@@ -967,6 +969,7 @@
}
break;
case INC_IN_USE:
+ case INC_OUT_USE:
if (u-incominglimit 0 ) {
if (u-inUse = u-incominglimit) {
ast_log(LOG_ERROR, Call from user
'%s'
rejected due to usage limit of %d\n, u-name, u-incominglimit);
@@ -977,6 +980,7 @@
u-inUse++;
ast_log(LOG_DEBUG, Call from user '%s' is %d out of
%d\
n, u-name, u-inUse, u-incominglimit);
break;
+ /* Commented out - don't want to limit outgoing */
case DEC_OUT_USE:
if ( u-outUse 0 ) {
u-outUse--;
@@ -994,6 +998,7 @@
}
u-outUse++;
break;
+ */
default:
ast_log(LOG_ERROR, find_user(%s,%d) called with no
even
t!\n,u-name,event);
}
@@ -1086,6 +1091,12 @@
INVITE, but do set an autodestruct just
in ca
se. */
needdestroy = 0;
sip_scheddestroy(p, 15000);
+ /* channel still up - reverse dec of inuse
count
er */
+ if ( p-outgoing ) {
+ find_user(p, INC_OUT_USE);
+ } else {
+ find_user(p, INC_IN_USE);
+ }
} else {
char *res;
if (ast-hangupcause ((res =
hangup_cause2sip
(ast-hangupcause {
@@ -4708,6 +4719,14 @@
if (p-owner)
ast_queue_control(p-owner,
AST_
CONTROL_BUSY, 0);
break;
+ case 487:
+ /* channel now destroyed - dec the
inuse
counter */
+ if ( p-outgoing ) {
+ find_user(p, DEC_OUT_USE);
+ } else {
+ find_user(p, DEC_IN_USE);
+ }
+ break;
case 486: /* Busy here */
case 600: /* Busy everywhere */
if (p-owner)
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Waiting on SIP phones
Sorry, to repost - but I left a "/*" comment - here
it is again
Paul
--- chan_sip.c.save
2003-10-20 21:51:52.0 +1000+++ chan_sip.c 2003-10-21
09:26:41.0 +1000@@ -959,7 +959,9
@@
return 0;
} switch(event)
{+
/* Incoming and outging affects the inUse counter
*/
case
DEC_IN_USE:+
case
DEC_OUT_USE:
if ( u-inUse 0 )
{
u-inUse--;
} else {@@ -967,6 +969,7
@@
}
break;
case
INC_IN_USE:+
case
INC_OUT_USE:
if (u-incominglimit 0 )
{
if (u-inUse = u-incominglimit)
{
ast_log(LOG_ERROR, "Call from user '%s'rejected due to usage limit of %d\n",
u-name, u-incominglimit);@@ -977,6 +980,8
@@
u-inUse++;
ast_log(LOG_DEBUG, "Call from user '%s' is %d out of %d\n", u-name,
u-inUse,
u-incominglimit);
break;+
/* Commented out - don't want to limit outgoing
*/+
/*
case
DEC_OUT_USE:
if ( u-outUse 0 )
{
u-outUse--;@@ -994,6 +999,7
@@
}
u-outUse++;
break;+
*/
default:
ast_log(LOG_ERROR, "find_user(%s,%d) called with no
event!\n",u-name,event);
}@@ -1086,6 +1092,12
@@
INVITE, but do set an autodestruct just in case.
*/
needdestroy =
0;
sip_scheddestroy(p,
15000);+
/* channel still up - reverse dec of inuse counter
*/+
if ( p-outgoing )
{+
find_user(p,
INC_OUT_USE);+
} else
{+
find_user(p,
INC_IN_USE);+
}
} else
{
char
*res;
if (ast-hangupcause ((res =
hangup_cause2sip(ast-hangupcause {@@ -4708,6 +4720,14
@@
if
(p-owner)
ast_queue_control(p-owner, AST_CONTROL_BUSY,
0);
break;+
case
487:+
/* channel now destroyed - dec the inusecounter
*/+
if ( p-outgoing )
{+
find_user(p,
DEC_OUT_USE);+
} else
{+
find_user(p,
DEC_IN_USE);+
}+
break;
case 486: /* Busy here
*/
case 600: /* Busy everywhere
*/
if (p-owner)
Re: [Asterisk-Users] Call Waiting on SIP phones
- Original Message - From: John Todd [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, October 21, 2003 10:29 AM Subject: Re: [Asterisk-Users] Call Waiting on SIP phones Paul - A few questions and comments: 1) So, does this also make incominglimit and outgoinglimit work as expected? The current method doesn't do quite what the average user thinks it would do. Yes it does John - however, I've made incominglimit and outgoinglimit to imply the same, so we only need incominglimit=1. 2) Your patch may be relevant to: http://bugs.digium.com/bug_view_page.php?bug_id=329 - I didn't realize that outgoinglimit=1 would only work for the first call, but fail subsequently. With type=peer, the incoming= and outgoing= modifiers really don't work at all for me, which makes them almost completely useless. If you have some method to fix that: great! The work once applies to incoming (never tested for outgoing) - Its now fixed for both - so let me know how you go. 3) If there is an existing bugtracker item, the source code diff's are best appended to that particular bug. If there isn't one open, go ahead and open one. Sending diff's to the mailing list is getting less common (and less desirable) as time goes on, since we have the bugtracker now. I've added to the bugtracker - http://bugs.digium.com/bug_view_page.php?bug_id=408 I've made it new report as it also fixes the incoming call waiting which 329 didn't refer to. Also thanks for your comments and also from Tilghman. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] use of SIP SHOW CHANNELS question
If you had a look under the help as the prompt said and entered help show - you would have found that it is show codecs Paul - Original Message - From: Aaron Martin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 20, 2003 11:56 AM Subject: Re: [Asterisk-Users] use of SIP SHOW CHANNELS question I dont think that is it: *CLI show codec 4 No such command 'show codec' (type 'help' for help) *CLI show audio codecs No such command 'show audio' (type 'help' for help) - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 20, 2003 12:40 PM Subject: Re: [Asterisk-Users] use of SIP SHOW CHANNELS question On Sunday 19 October 2003 18:01, Aaron Martin wrote: Also, what do the different 'format' numbers mean? Is there a table somewhere showing which format is which number? *CLI show codec 4 4 (1 2) G.711 u-law *CLI show audio codecs -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto Start
Dave, After the system boots up, you check to see which modules are loaded by doing a lsmod. zaptel and others that you need should be listed. If not you can manually add modprobe zaptel and the other drivers into your rc.local file. Paul Just before the Logon prompt appears on boot I see a message but it goes before I can read it and the prompt to logon appears. If I scroll back up the screen the message is not there. I now have only ./safe_asterisk in my rc.local file. Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] direct-inward-dialing (DID)
The number of digits that your telco sends to you is a configurable figure
(at least it is here in Aus). The example assumes that the telco is sending
you the last 4 digits.
Paul
Example: 456-7000 is your main number and you have 7001 to 7099 as DIDs:
exten = 7000,1,Goto(AutoAttendant|s|1)
exten = _7XXX,1,Macro(yourdialmacro|${EXTEN})
How are you dropping the 456 there? I thought extensions picked up what
either the SIP phone had dialled, or what DTMF detection picked up when *
answered the line...?
I'm looking at purchasing a PRI with 30 DIDs (can't get any fewer from
Bell
Canada) and routing the calls coming in to multiple remote * boxes based
on
the called number. I was going to ask a question similar to John's but
just didn't get around to it yet. :-)
If you could explain in a little more detail how you turn the CNID into an
extension I'd really appreciate it.
Andrew
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice detection
You can also use the AGI interface function RECORD FILE and specify a max record duration of 5s and silence detection of 1s. Time the duration of the call to asterisk - if its longer than 1 second you know you've got voice. If you need to check for voice over a longer period of time - repeat the call x times. This way you'll wait for approx 'x' seconds for voice or silence. I've done this using some C code and works very well as a grunt detector - timing out after 5 seconds of silence or returning immediately when voice is received. Paul - Original Message - From: Christian Hecimovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, October 04, 2003 10:39 AM Subject: Re: [Asterisk-Users] Voice detection dsp.c has silence detection that works quite well for detecting end-of-voice silence. It is used to allow only a certain amount of silence at the end of voicemails, for instance. See app_voicemail2.c on how to use it, specifically the function play_and_record(). Note that the silence threshold (how sensitive you are to silence) is read in from the voicemail.conf file. Since the silence detection stuff has a nice public API, you can use it for any app you write. See app_skel.c for a basic shell, and follow something like app_voicemail2.c. Read in the acceptable values for threshold and so forth from a configuration file (there is a nice Asterisk API for this, also), and you're set. For your purposes (playing a file after detecting silence in a remote voice stream), such an app should be quite simple. Christian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
