[Asterisk-Users] Calls not cleared down if extra destinations or dial commands added to extension
We have a weird situation where if the external called hangs up the call before it is answered asterisk seems not to handle it if the original dial command is replaced following a timeout. We are trying to pass the call to the main reception, but if there is no answer then it should ring another extension in addition to the first extension the idea being that we don't end up with people chasing the call from desk to desk. In the example below if the caller hangs up during step 4 asterisk will continue on to step 5 and start recording a voicemail. I can't see that we we are trying to do anything unusual here, anyone able to shed any light? exten = 470,1,SetCIDName(Tech Support) exten = 470,2,Dial(SIP/1,10,tr) exten = 470,3,Dial(SIP/1SIP/2,10,tr) exten = 470,4,Dial(SIP/1SIP/2SIP/23,10,tr) exten = 470,5,Voicemail(sb000) ___ 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] Group dial, first phone cannot pickup call if included in subsequent steps.
I've tried it on another system and can reproduce the fault at will. It would seem that the first Dial command is not terminated when the second Dial includes the first extension. e.g. exten = 100,1,Dial(SIP/1001,10,tr) exten = 100,2,Dial(SIP/1002,10,tr) exten = 100,3,Dial(SIP/1003,10,tr) Will do exactly what you'd expect, dials exten 1001 for 10 secs, then 1002 for 10 secs, and finally 1003 for 10 secs. However once you add extra extensions it all goes horribly wrong. e.g. exten = 100,1,Dial(SIP/1001,10,tr) exten = 100,2,Dial(SIP/1002SIP/1003,10,tr) This does what is expected, after ringing extension 1001 for 10 secs extensions 1002 1003 start ringing and the first one to answer gets the call where upon the second one stops ringing. However the following is nastier: exten = 100,1,Dial(SIP/1001,10,tr) exten = 100,2,Dial(SIP/1001SIP/1003,10,tr) After ringing extension 1001 for 10 secs extensions 1001 and 1003 start ringing but only extension 1003 can pickup the call. If extension 1001 is picked up after extension 1003 has started ringing it is dead as a dodo. Is a bug or a feature? On Tue, 2005-05-10 at 09:46, bam wrote: I have a simple dial plan to cascade calls when the first phone does not answer: exten = 100,1,Dial(SIP/1000,10,tr) exten = 100,2,Dial(SIP/1000SIP/1001,10,tr) exten = 100,3,Dial(SIP/1000SIP/1001SIP/1002,10,tr) exten = 100,4,Voicemail(u100) Problem is that the once the call goes onto the second and subsequent steps exten 1000 cannot answer the call. When the user picks up the phone it is just dead, no dial tone, nothing. Occasionally the handset will hang and need to be power-cycled. I've swapped out the phone, the power supply, and even the cabling, but no joy. As long as exten 1000 picks up the call at step one everything works fine. Apart from that everything else seems tickety-boo. Cisco 7905s with latest SIP build and Asterisk CVS-HEAD-03/02/05 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Group dial, first phone cannot pickup call. Cisco 7905 hangs.
I have a simple dial plan to cascade calls when the first phone does not answer: exten = 100,1,Dial(SIP/1000,10,tr) exten = 100,2,Dial(SIP/1000SIP/1001,10,tr) exten = 100,3,Dial(SIP/1000SIP/1001SIP/1002,10,tr) exten = 100,4,Voicemail(u100) Problem is that the once the call goes onto the second and subsequent steps exten 1000 cannot answer the call. When the user picks up the phone it is just dead, no dial tone, nothing. Occasionally the handset will hang and need to be power-cycled. I've swapped out the phone, the power supply, and even the cabling, but no joy. As long as exten 1000 picks up the call at step one everything works fine. Apart from that everything else seems tickety-boo. Cisco 7905s with latest SIP build and Asterisk CVS-HEAD-03/02/05 ___ 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] G729 Phones and G711ulaw Voicemail
Either disable G729 or splash out $20 and get a couple of licenses, it is hardly a King's ransom after all. On Tue, 2005-05-10 at 01:04, Ren Mayorga wrote: Hi, I have an asterisk server without any G729 licenses, and a couple of BT-100 phones that actually works already with G729 passtrought (*) conf. My problem, is when the BT-100 try to call to the voicemail application, It first try G729, and then the call hang up. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No music on hold when transferring call
MOH is working in that a defined extension works just fine:exten = 6000,1,Answerexten = 6000,2,MusicOnHold()musiconhold.conf is as per the default:[classes]default = quietmp3:/var/lib/asterisk/mohmp3,-zand zapata.conf and sip.conf havemusiconhold=default and musicclass=default respectively.However when I put a call on hold for transfer or just pressing the hold button there is no music. Normally I would expect to see something like the following, but nothing appears in the trace. --Started music on hold, class 'default', on SIP/4101-5ea9If a blind transfer is initiated the original caller gets hold music while the blind transfer is setup so I fear that something is back to front. All help gratefully received. Message sent using UebiMiau 2.7.2 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cannot make outgoing calls on Mediatrix 1204 from Asterisk
Mediatrix has been setup with automatic calling enabled as suggested elsewhere with the four ports forwarding calls to extensions 1001, 1002, 1003, 1004 respectively. Inbound traffic pretty much does what is expected albiet it takes a few rings and some warnings before the call is passed to Asterisk. Outbound calls head off to the 1204 before they loop back to asterisk appearing as an imbound call which is not really what is expected. Automatic calling is supposed to be inbound only. Any ideas? The trace looks like this: -- Executing Dial(SIP/212-acc5, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- Executing Dial(SIP/1001-7f99, SIP/210|20|tr) in new stack -- Called 210 -- SIP/210-b655 is ringing extensions.conf [BT_PSTN] ; Inbound calls exten = _100X,1,Dial(SIP/210,20,tr) [LOCAL_SIP] ; All internal extensions exten = _0.,1,Dial(SIP/[EMAIL PROTECTED]) Mediatrix is version 4.4.13.88 ___ 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] One touch voicemail on Cisco 7940/60
On Thu, 2005-04-21 at 21:36, Ron Wellsted wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Morris, Simon wrote:
Hello,
I'd like to program my Cisco phones to authenticate themselves to
voicemail upon hitting the right button on my 7940/60's
Ideally the voicemail app will detect which extension the call is coming
from and drop the user straight into the menu.
Is this possible?
Many thanks
~sm
Yes this is possible.
In your extensions.conf:
exten = _8501,1,Answer()
exten = _8501,2,VoicemailMain(s${CALLERIDNUM})
exten = _8501,3,Hangup()
then program the messages button to dial 8501 either via settings, SIP
Settings, Messages URI (set to 8501) or in the SIPDefault.cnf file
However if you are busy callers will immediately be redirected to this
extension and get your voicemail menu unless you have call waiting
enabled on the phone. Suggest you try this:
; Assuming your extension is 2034 and 8501 is your voicemail extension.
;
exten = 8501/2034,1,VoicemailMain(s2034)
exten = 8501,1,Voicemail(b2034)
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P delayed ring on incoming calls?
I have setup an asterisk box with 3off X100P cards and hooked them up to the PSTN. So far so good, everything does what it is supposed to do for the msot part. Incoming calls seem to ring three or four times before asterisk then skips to do what it is supposed to do. If the caller drops the call before the extensions have started ringing asterisk seems not to pick this up and carries on regardless. Anyone got any ideas? This was built from CVS. == Spawn extension (BT_PSTN, s, 1) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1' -- Starting simple switch on 'Zap/3-1' Apr 22 09:20:11 NOTICE[20851]: chan_zap.c:5585 ss_thread: Got event 2 (Ring/Answered)... Apr 22 09:20:13 NOTICE[20851]: chan_zap.c:5585 ss_thread: Got event 2 (Ring/Answered)... Apr 22 09:20:14 NOTICE[20851]: chan_zap.c:5585 ss_thread: Got event 2 (Ring/Answered)... Apr 22 09:20:16 NOTICE[20851]: chan_zap.c:5585 ss_thread: Got event 2 (Ring/Answered)... -- Executing Dial(Zap/3-1, SIP/110SIP/112|20|tr) in new stack -- Called 110 -- Called 112 -- SIP/110-ff2f is ringing -- SIP/112-4713 is ringing -- SIP/110-ff2f answered Zap/3-1 extensions.conf [BT_PSTN] exten = s,1,Answer exten = s,2,Dial(SIP/110SIP/112,20,tr) exten = s,3,Voicemail(u000) zapata.conf context=BT_PSTN callerid=Inbound Call 01774987987 signalling = fxs_ks channel=1-3 group = 1 ___ 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] X100P delayed ring on incoming calls?
On Fri, 2005-04-22 at 10:45, Dave Cotton wrote: On Fri, 2005-04-22 at 10:22 +0100, bam wrote: I have setup an asterisk box with 3off X100P cards and hooked them up to the PSTN. So far so good, everything does what it is supposed to do for the most part. Incoming calls seem to ring three or four times before asterisk then skips to do what it is supposed to do. If the caller drops the call before the extensions have started ringing asterisk seems not to pick this up and carries on regardless. Anyone got any ideas? This sounds like the CallerID problem. * is trying to get the ID, but the UK's method is different to the default, so it does not get an ID it finally gives up and processes the call. Look for UKCaller ID settings in the archives or Wiki. (I left the UK 12 years ago so I've never looked at it). What a hero, problem solved. ;-) zapata.conf usecallerid=no ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail name (greet.wav) is not played
How or when is the voicemail name actually played? I've recorded my name message and can see that the voicemail directory now has two new greet files and the original greet.gsm has been overwritten. # ls /var/spool/asterisk/voicemail/default/4100/INBOX/ -l -rw-r--r-- 1 root root 8943 Feb 10 17:22 busy.gsm -rw-r--r-- 1 root root 3993 Apr 14 17:31 greet.gsm -rwx-- 1 root root 38764 Apr 14 17:31 greet.wav -rwx-- 1 root root 3960 Apr 14 17:31 greet.WAV drwxr-xr-x 2 root root 4096 Feb 24 12:15 INBOX -rw-r--r-- 1 root root 8943 Feb 10 17:22 unavail.gsm There is no mention in Wiki or Google and I've even resorted to scouring the source code, but all I can find are the options to record the name. How do I use the name option? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Voicemail name (greet.wav) is not played
Sorry, I've not bee clear enough, when does the greet file get played at all? I can see how to record the greet.* sound file, and the documentation for that, but so far can only see the busy and unavail messages being played. There are no error messages assocated with this, just users asking what happened to the name that they recorded in response to the prompts. many thanks, Brian On Thu Apr 14 12:22:02 CDT 2005, Roderick A. Anderson wrote: On Thu, 2005-04-14 at 18:05, bam wrote: How or when is the voicemail name actually played? I've recorded my name message and can see that the voicemail directory now has two new greet files and the original greet.gsm has been overwritten. # ls /var/spool/asterisk/voicemail/default/4100/INBOX/ -l -rw-r--r-- 1 root root 8943 Feb 10 17:22 busy.gsm -rw-r--r-- 1 root root 3993 Apr 14 17:31 greet.gsm -rwx-- 1 root root 38764 Apr 14 17:31 greet.wav -rwx-- 1 root root 3960 Apr 14 17:31 greet.WAV drwxr-xr-x 2 root root 4096 Feb 24 12:15 INBOX -rw-r--r-- 1 root root 8943 Feb 10 17:22 unavail.gsm There is no mention in Wiki or Google and I've even resorted to scouring the source code, but all I can find are the options to record the name. How do I use the name option? Could this be a permissions issue? Should greet.(wav,WAV) be the same as greet.gsm? Rod -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Missing digits on TDM400P incomplete dial string
We are experiencing problems on a FXS interface where the client is dialing
numbers, but digits are being dropped somewhere from the dial string.
Typically one or two digits are not being presented. We've tried different
handsets to no avail, and I am assuming that it is some sort of timing problem.
Are there any parameters I can tweak to try and rectify this?
zapata.conf
context=hardwire
group=3
signalling=fxo_ks
mailbox=8765
callerid=Acme 8765
channel=32
extensions.conf
[hardwire]
;
exten = _NXX,1,SetCallerID(0141411${CALLERIDNUM})
exten = _NXX,2,CallingPres(3)
exten = _NXX,3,Dial(Zap/g1/0141${EXTEN})
exten = _0.,1,SetCallerID(0141411${CALLERIDNUM})
exten = _0.,2,CallingPres(3)
exten = _0.,3,Dial(Zap/g1/${EXTEN})
exten = t,1,Hangup ; If they take too long, give up.
exten = i,1,Hangup ; If they get it wrong, give up
___
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] Missing digits on TDM400P incomplete dial string - Email found in subject
I've had a quick fiddle to little avail, the readings looked prey good to
be honest before I started fiddling. Looking a little closer it appears
that it is the digit 1 that is being lost more that any other.
At 15:25 07/05/04, you wrote:
Run /usr/src/zaptel/ztmonitor 32 -v
And adjust your gains in /etc/asterisk/zapata.conf accordingly.
Gregory P. Scasny
Golden Technologies Inc.
http://www.golden-tech.com
219-462-7200
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bam
Sent: Friday, May 07, 2004 3:35 AM
To: [EMAIL PROTECTED]
Subject: [SPAM] - [Asterisk-Users] Missing digits on TDM400P incomplete
dial string - Email found in subject
We are experiencing problems on a FXS interface where the client is
dialing
numbers, but digits are being dropped somewhere from the dial string.
Typically one or two digits are not being presented. We've tried
different
handsets to no avail, and I am assuming that it is some sort of timing
problem.
Are there any parameters I can tweak to try and rectify this?
zapata.conf
context=hardwire
group=3
signalling=fxo_ks
mailbox=8765
callerid=Acme 8765
channel=32
extensions.conf
[hardwire]
;
exten = _NXX,1,SetCallerID(0141411${CALLERIDNUM})
exten = _NXX,2,CallingPres(3)
exten = _NXX,3,Dial(Zap/g1/0141${EXTEN})
exten = _0.,1,SetCallerID(0141411${CALLERIDNUM})
exten = _0.,2,CallingPres(3)
exten = _0.,3,Dial(Zap/g1/${EXTEN})
exten = t,1,Hangup ; If they take too long, give
up.
exten = i,1,Hangup ; If they get it wrong, give up
___
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] Missing digits on TDM400P incomplete dial string - DTMF problem?
I turned down the rxgain and txgain to -22.0 and -16.0 respectively and
things started to look a whole lot more acceptable. Then the client sticks
on his BT DECT phone and I start losing all the 1s from the dial string.
Does anyone know if BT DECT phones have dodgy DTMF tones?
At 17:19 07/05/04, you wrote:
I've had a quick fiddle to little avail, the readings looked prey good to
be honest before I started fiddling. Looking a little closer it appears
that it is the digit 1 that is being lost more that any other.
At 15:25 07/05/04, you wrote:
Run /usr/src/zaptel/ztmonitor 32 -v
And adjust your gains in /etc/asterisk/zapata.conf accordingly.
Gregory P. Scasny
Golden Technologies Inc.
http://www.golden-tech.com
219-462-7200
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bam
Sent: Friday, May 07, 2004 3:35 AM
To: [EMAIL PROTECTED]
Subject: [SPAM] - [Asterisk-Users] Missing digits on TDM400P incomplete
dial string - Email found in subject
We are experiencing problems on a FXS interface where the client is
dialing
numbers, but digits are being dropped somewhere from the dial string.
Typically one or two digits are not being presented. We've tried
different
handsets to no avail, and I am assuming that it is some sort of timing
problem.
Are there any parameters I can tweak to try and rectify this?
zapata.conf
context=hardwire
group=3
signalling=fxo_ks
mailbox=8765
callerid=Acme 8765
channel=32
extensions.conf
[hardwire]
;
exten = _NXX,1,SetCallerID(0141411${CALLERIDNUM})
exten = _NXX,2,CallingPres(3)
exten = _NXX,3,Dial(Zap/g1/0141${EXTEN})
exten = _0.,1,SetCallerID(0141411${CALLERIDNUM})
exten = _0.,2,CallingPres(3)
exten = _0.,3,Dial(Zap/g1/${EXTEN})
exten = t,1,Hangup ; If they take too long, give
up.
exten = i,1,Hangup ; If they get it wrong, give up
___
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
[Asterisk-Users] Jump to extension from voice menu
Is there a way to allow a caller to enter an extension number that is more than one digit long in a voice menu? I want to have a menu that allows something like If you know the extension number of the person please enter it otherwise 1 for sales, 2 for...etc many thanks in advance, 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] manipulating with numbers - StripMSD, Prefix
Looks like you are shy a zero
Try exten = _50.,Prefix,001051
At 12:06 07/01/04, you wrote:
Hello,
I can not seem to be able to get StripMSD and Prefix to work for me in
extensions.conf. This is an example of what I have:
exten = _050.,1,StripMSD,1
exten = _50.,Prefix,01051
exten = _001051.,1,Dial(${TRUNK2}/${EXTEN})
exten = _001051.,2,Busy
exten = _001051.,102,Busy
What I want to achieve is to call 001051501657887 on TRUNK2 when dialing
0501657887.
___
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] One way h323 to Cisco 7905?
I've acquired a Cisco 7905 with H323 s/w that I have connected to *. It can make calls happily enough to H323 SIP extensions and out to the PSTN, however when ever I try to call it from any destination the call fails with H323:0 Could not call 192.168.9.23 Hungup 'H323:0' Everyone is busy at this time. TCPDUMP shows a short but spirited exchange between the 7905 and *, but nothing on the console to give me a hint. Anyone got any ideas? ___ 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] Address Separator hex b causes callerid rejection
I've now twigged that this is an SS7 flag and is being set in our switch as a result of * passing the Network provided screening indicator to a value that is interpreted as untrusted. Is there a simple way of changing the default value for this? At 16:22 30/01/04, you wrote: I am having a little bit of a problem with BT rejecting my callerid values as they are prefixed by hex b. This indicates that the caller id is user provided and not verified. Does anyone know how I can control where this appears in the cli? The purpose of the separator is described below: 1 - PNO 006 section 2.4.19 c note states that the hex b denotes an address separator, to separate the part which is network provided from that which is user provided - This means that it separates the extension number from the rest of the number. 2 - PNO008 section 22.1.3.3 states that the hex b dependant on its position, denotes whether the screening indicator is user provided not verified, network provided or user provided verified and passed. ___ 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] Address Separator hex b causes callerid rejection
The problem is that the ISDN call to our switch has the screening indicator set to untrusted, to the switch sticks 0xb on the front of the CLI. BT then drop the CLI altogether. So I need to find a way of fiddling with the * presentation. At 12:29 02/02/04, you wrote: I missed your earlier message, but to try and help: a) You are correct, the hex 'b' usage is only valid in the UK specific BT IUP SS#7 interconnect protocol and therefore is nothing to do with PRI usage whatsoever (or indeed ISUP SS#7). b) In q931.c these various flags can be set for outbound CLI (caller id), from user provided not verified, user provider verified, user provided verfication failed, and network provided. c) BT however will only accept CLI that you are authorised to send - whatever the state of the flags, this means that you can only send CLI that matches numbers that have been allocated to your PRI. If you are trying to do otherwise this will always fail. Linus ___ 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] Address Separator hex b causes callerid rejection
We do indeed, no one to blame :-( CallerPres(3) does the business, thank you very much. Is there a list of all the applications somewhere? I've been looking in asterisk/apps/ and can't find CallerPres. all the best, Brian At 13:41 02/02/04, you wrote: Ahhh. so you have an interconnect switch then I take it! You need to set the q931.c value - PRES_ALLOWED_NETWORK_NUMBER which has a value of '3'. I think, although I've never tried this, you can actually call the application CallingPres with the value of 3 before making an outbound call. CallingPres(3) I think should do it - someone else might be able to advise better. Linus ___ 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: Asterisk and gnugk (bam)
The phone works fine with oh323, its just the need to authenticate the endpoint and match a non-fixed ip to a number that has sent me off in the direction of gnugk. If I could do it all in * I would. thanks, brian At 18:05 29/01/04, Roger wrote: Hi, I also had some problems using chan_oh323 together with gnugk. * - gnugk - h323-phone When I called the phone and hang up, befor the phone was picked up, the h323-phone continued ringing. The same, when the h323- and some sip-phones were called, and the sip-phone picked up the call first. (It is annoying, when you are talking to someone at the phone and the phone on the neighbour desk does not stop ringing!) Now, I switched to chan_h323 and the h323-phone works better. The only problem what remained, is that the phone and * sometimes don't manage to negotiate a codec both are supporting. But when gnugk is not in routed mode, everything is fine! Roger. ___ 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] Address Separator hex b causes callerid rejection
I am having a little bit of a problem with BT rejecting my callerid values as they are prefixed by hex b. This indicates that the caller id is user provided and not verified. Does anyone know how I can control where this appears in the cli? The purpose of the separator is described below: 1 - PNO 006 section 2.4.19 c note states that the hex b denotes an address separator, to separate the part which is network provided from that which is user provided - This means that it separates the extension number from the rest of the number. 2 - PNO008 section 22.1.3.3 states that the hex b dependant on its position, denotes whether the screening indicator is user provided not verified, network provided or user provided verified and passed. ___ 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 and gnugk
I've tried every variation I can think of and always seem to end up with one of the two servers frantically trying to authenticate itself. I guess it it just he different terminology in the config files that is confusing me. Could I beg a hint? -- oh323.conf listenport=1725 #connectport=1720 gatekeeper=195.206.192.194 gatekeeperPassword=OurSecret gnugk.ini [RoutedMode] GKRouted=1 H245Routed=1 CallSignalPort=1725 AcceptUNregisteredCalls=0 At 20:53 22/01/04, Lubomir Christov wrote: yes :) bam wrote: This is quite possibly a daft question, but it is possible to run * and gnugk on the same system with gnugk acting as a proxy for netmeeting endpoints and feeding everything for PSTN and SIP out through *? ___ 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] Asterisk and gnugk
We need to be able to support Netmeeting users and in doing so we need to authenticate them to ensure that we don't get unauthorised users. I'd love to stick to SIP, but it is not an option unfortunately. regards, Brian At 17:14 29/01/04, Jeremy McNamara wrote: Then at the risk of being flamed (again) why do you need H.323? Most real IP Phones out there have some other firmware option than H.323 and there are certianly carriers out there that have seen the light and offer some other VoIP signalling protocol. ___ 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] Asterisk and gnugk
This is quite possibly a daft question, but it is possible to run * and gnugk on the same system with gnugk acting as a proxy for netmeeting endpoints and feeding everything for PSTN and SIP out through *? ___ 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] grandstream asterisk configuration
Make sure that udp packets can get from the server back to the grandstream. At 12:40 14/01/04, you wrote: hi, I have the following configuration: Grandstream -- NAT (Netgear RP614)--Internet--Asterisk(public IP) i can register fine and call ringing is working as good. The problem is = i cant hear audio both ways and i get this error: WARNING[22544]: File rtp.c, Line 375 (ast_rtp_read): RTP Read error: Resource temporarily unavailable my sip.conf file is as follows: [general] port =3D 5060 ; Port to bind to bindaddr =3D 0.0.0.0 ; Address to bind to ;externip =3D 200.201.202.203 ; Address that we're going to put in = SIP messages if we're behind a NAT tos=3Dlowdelay disallow=3Dall; Disallow all codecs allow=3Dulaw ; Allow codecs in order of preference dtmfmode=3Dinfo [grandstream1] type=3Dfriend host=3Ddynamic secret=3Dmysecret context=3Doutgoing nat=3Dyes reinvite=3Dno canreinvite=3Dno qualify=3D2000 has anyone done this before? chandra ___ 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] E100P pinouts anyone?
I'm trying to hookup an E100P to an E1 PRI that I know is working, it drives an Ascend Max happily. The E100P has a red LED flashing away slowly and the telco switch reports that there is no level one signalling link. So that I can rule out the obvious things does anyone have the pinouts for the RJ45 on the E100P? I've tried straight through and crossover to no avail. I get a couple of errors on modprobing the driver, but if I reverse the order it seems to load OK. ztcfg is happy and zttool seems to see the card so I'm pretty certain it is something silly. any help gratefully received. Brian [EMAIL PROTECTED] root]# modprobe zaptel [EMAIL PROTECTED] root]# modprobe wct1xxp ZT_CHANCONFIG failed on channel 32: No such device or address (6) /lib/modules/2.4.20-24.9/misc/wct1xxp.o: post-install wct1xxp failed /lib/modules/2.4.20-24.9/misc/wct1xxp.o: insmod wct1xxp failed [EMAIL PROTECTED] root]# modprobe wcfxs [EMAIL PROTECTED] root]# modprobe wct1xxp [EMAIL PROTECTED] root]# ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream 102 flashing display
The phone powers up and I can make calls through my Asterisk gateway to other endpoints. However the four leds under the keypad are permanently illuminated and the backlight slowly flashes on and off. When I pick up the handset there is a repeated tone before I get a dial tone. I know it's trying to tell me something, but the manual does not give anything away. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem - installing TDM400P module
You could try $ modprobe zaptel $ modprobe wcfxs You need the zaptel bits first. At 09:52 23/12/03, you wrote: $insmod wcfxs Using /lib/modules/2.4.20-8/misc/wcfxs.o /lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_ec_chunk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk to H.323 without gatekeeper
I'm missing something here. I've put the following in extensions.conf and a
few variations thereof. I've taken the sample configs and added to them, so
when I dial 2200 from netmeeting * answers and runs me through the demo
announcements.
The pots extensions 2200 and 2107 (TDM400) work fine calling each other and
cause netmeeting to ring when I dial 3100, but the audio is one way
pots-netmeeting when I answer in netmeeting.
If it is an RTFM situation please give me a URL, pretty postcard to anyone
than can help me.
extensions.conf
[incoming-h323]
exten = 3001,1,Dial,OH323/192.153.153.64
exten = 3001,2,Busy
exten = 3001,102,Busy
[default]
include = incoming-h323
include = demo
exten = 2107,1,Dial(Zap/32,20)
exten = 2107,2,Voicemail(u2107)
exten = 2107,102,Voicemail(b2107)
exten = 2200,1,Dial(Zap/33,20)
exten = 2200,2,Voicemail(u2200)
exten = 2200,102,Voicemail(b2200)
At 13:42 19/12/03, you wrote:
bam wrote:
I've read through the archives and have picked up that * does not need a
gatekeeper to talk directly with an H323 handset to send and receive calls.
I'm trying to go PSTN*-H323 and all the examples that I can find
use a gatekeeper. Are there any examples or hints for doing it without
the gatekeeper?
many thanks in advance
Brian
[your_context]
exten = _9XX,1,Dial,H323/78632${EXTEN:[EMAIL PROTECTED]|30
exten = _9XX,2,Busy
exten = _9XX,102,Busy
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
--
-
Best Regards,
Pavel Litvinenko.
ICQ: 16224754
Ph: (8632) 923962, 923640
sip:[EMAIL PROTECTED]
___
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] Asterisk to H.323 without gatekeeper
I cracked the concept of how to handle incoming calls and route them to the right context, apologies for being a little slow on the uptake. I can now call between pots end points and netmeeting endpoints. Still having problems with sound despite having set everything to use G711A. POT to POT via * fine and netmeeting to netmeeting direct is OK. NM to NM via * is silent in both directions. NM to POT via * is silent NM to POT, but OK POT to NM i.e. the NM user can hear the POT user but not the other way. any pointers gratefully accepted. At 14:07 22/12/03, you wrote: I'm missing something here. I've put the following in extensions.conf and a few variations thereof. I've taken the sample configs and added to them, so when I dial 2200 from netmeeting * answers and runs me through the demo announcements. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk to H.323 without gatekeeper
I've read through the archives and have picked up that * does not need a gatekeeper to talk directly with an H323 handset to send and receive calls. I'm trying to go PSTN*-H323 and all the examples that I can find use a gatekeeper. Are there any examples or hints for doing it without the gatekeeper? many thanks in advance Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
