[Asterisk-Users] bug? Unterminated comment detected beginning on line 0
Hi, Using latest cvs. A comment-line begins with semicolon ; However - if the line contains ;-- or like this ; -- blabla bla -- You get this error and * stops reading that file: Feb 21 13:47:12 WARNING[17393]: config.c:664 config_text_file_load: Unterminated comment detected beginning on line 0 Shouldn't Asterisk skip any line beginning with a semicolon? Or should a comment now be terminated too? /Stig - N Y H E T E R! - IP-telefoni, spara tusenlappar om året! - Rikstäckande ADSL 0,25-24Mbit - Internetaccess (Modem/ISDN64+128 via Ymex - utan abonnemangskostnad! - Eposttjänster, även UUCP, Uppringd SMTP, MX fallback, DomänPOP - Surf24 - en billig bredbandstjänst från Ymex för kunder i Härnösand/Älandsbro. - Get your emailed Web-forms into a database of your choice!!! Checkout DBFORM V1.0, see details at http://www.ymex.se - Ymex AB| Alvägen 7 | 871 52 Härnösand | Sweden | http://www.ymex.se/ ___ 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] Any luck with attended transfer and ATA186?
Hi, Using latest cvs. I (as many otheres it seems) can't get Attended transfer to work with Cisco ATA186 (using SIP) Has anyone else had any luck? Same with 3-part calling, if one drops off, all are disconnected... /Stig ___ 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] Is this a bug or by design? Workaround?
Hi,
I need to use the trailing 5 digits of a callerid. callerid may be anything
from a length of 4 to 10 digits in this case.
Using this:
---
SubString,cid=${CALLERIDNUM}|-5|5
Works great, BUT shows this message:
The use of Substring application is deprecated. Please use ${variable:a:b}
instead
So, I try
-
SetVar(cid=${CALLERIDNUM:-5:5})
The result is a empty string if CALLERIDNUM is less than 5 digits long,
which is NOT the case of SubString. SubString command returns what remains of
the variable,
that is - if CALLERIDNUM is 4 digits in length, it returns 4 digits. If
CALLERIDNUM is 6 digits,
it returns 5 digits.
If this approach should replace Substring - it should behave identically,
shouldn't it?
If by design, is there a workaround?
/Stig
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Re: [Asterisk-Users] TDM400P Issues
Just a note regarding this issue. I'm using RH9, two X100p and one TDM400 Loading them in this order: zaptel wcfxo wcfxs zapata.conf like this: fxsks=1 fxsks=2 ;-- fxoks=3 fxoks=4 fxoks=5 fxoks=6 zaptel.conf like this: signalling=fxs_ks channel=1 channel=2 signalling=fxo_ks channel=3 channel=4 channel=5 channel=6 This is a fully working config. BUT, note that I will get same complains as you - during loading of modules. My solution for this was to remove the following from /etc/modules.conf post-install tor2 /sbin/ztcfg post-install wcfxo /sbin/ztcfg post-install wct1xxp /sbin/ztcfg post-install wct4xxp /sbin/ztcfg post-install wcfxs /sbin/ztcfg post-install wcfxsusb /sbin/ztcfg post-install torisa /sbin/ztcfg and instead running /sbin/ztcfg after modules was loaded. It seems as this ztcfg gets confused when running after each module as it is done when part part of post-install. Maybe helps... /Stig At 17:14 2004-04-13 +1000, you wrote: This looks wrong. What is the full output of ztcfg -vvv? Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. I'd be surprised if this worked as expected once you got * started... Well now that I just tried * it doesn't work... --snip-- [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, FXS Kewlstart signalling Apr 13 17:13:42 WARNING[16384]: chan_zap.c:665 zt_open: Unable to specify channel 4: No such device Apr 13 17:13:42 ERROR[16384]: chan_zap.c:5319 mkintf: Unable to open channel 4: No such device here = 0, tmp-channel = 4, channel = 4 Apr 13 17:13:42 ERROR[16384]: chan_zap.c:7355 setup_zap: Unable to register channel '4' Apr 13 17:13:42 WARNING[16384]: loader.c:313 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' -- Unregistered channel 1 -- Unregistered channel 2 Apr 13 17:13:42 WARNING[16384]: loader.c:408 load_modules: Loading module chan_zap.so failed! --snip-- Out of curiosity, what's the arrangement of the cards in the slots? Is the TDM card between the two X100Ps? The TDM is above the two X100P's, before it was below them. -- 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] PSTN calls do NOT hang up
Hi,
Asterisk either need to know when the remote caller ends his call,
or it must detect the silence.
Simplest solution is to activate silence detection, see
voicemail.conf.
You may need to do some testing to get the proper
silencethreshold setting.
Also search the archive, this is a often discussed issue...
http://mharc.lists.openservices.ca/archives/html/asterisk-users/
/Stig
At 17:25 2004-04-07 +0800, you wrote:
Hi all,
In my Asterisk setup, incoming calls
through Cisco PSTN gateway to Asterisk extensions sounds work fine. All
calls can be terminated properly after hangup. However, when calls were
forwarded to voicemail, after recording hangup the PSTN calls and
cisco FXO port remained connected unless cisco port was manually shut/no
shut. # key used to hang up the call did NOT help. Did anyone experience
the same problem??
--
sip*CLI
-- Executing
Answer(SIP/-0811b4b8, ) in new stack
-- Executing Wait(SIP/-0811b4b8,
1) in new stack
-- Executing VoiceMail(SIP/-0811b4b8,
u6917) in new stack
-- Playing 'voicemail/default/6917/unavail' (language
'en')
-- Playing 'vm-intro' (language 'en')
-- Playing 'beep' (language 'en')
-- x=0, open writing:
/var/spool/asterisk/voicemail/default/6917/INBOX/msg0003 format: gsm,
0x81254f8
-- x=1, open writing:
/var/spool/asterisk/voicemail/default/6917/INBOX/msg0003 format: wav49,
0x80fb178
-- x=2, open writing:
/var/spool/asterisk/voicemail/default/6917/INBOX/msg0003 format: wav,
0x811af70
-- Playing 'vm-msgsaved' (language 'en')
-- Executing Hangup(SIP/-0811b4b8,
) in new stack
== Spawn extension (sip, 6917, 4) exited non-zero on
'SIP/-0811b4b8'
sip*CLI
---
cisco#sh voice call
1/0/1
vtsp level 0 state = S_CONNECTvpm level 1
state = FXOLS_CONNECT vpm level 0 state = S_UP
--
dial-peer voice 999 voip
destination-pattern 8...
session protocol sipv2
session target ipv4:10.1.1.1:5065
session transport udp
codec g711ulaw
no vad
!
exten = 6917,1,Answer
exten = 6917,2,Wait(1)
exten = 6917,3,VoiceMail(u${EXTEN})
exten = 6917,4,Hangup
Thanks.
Ben
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[Asterisk-Users] Is Wildcard TDM400P capable of sending DTMF callerid?
Hi, Is Wildcard TDM400P capable of sending DTMF callerid? Does asterisk support it? I know X100P does not, but I have found no info as to TDM400P... /Stig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller entered digits ignored during wait....
Asterisk doesn't accept keys during wait, use Background
and play 1 sec silence instead.
/Stig
At 23:46 2004-03-30 -0600, you wrote:
On 2004 Mar 30, at 20:56, Gene Kochanowsky wrote:
How would you use the t extension to accomplish this?
exten = s,1,Wait(1)
exten = s,2,Answer
exten = s,3,SetVar(loopCnt=0)
exten = s,4,Background(welcome)
exten = s,5,Background(parties)
exten = t,1,SetVar(loopCnt=$[${loopCnt} + 1])
exten = t,2,GotoIf($[${loopCnt} 3]?s|4)
exten = t,3,Background(vm-goodbye)
exten = t,4,Hangup
-Tilghman
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Re: [Asterisk-Users] If you know your party's extension # please dial it now ...
Asterisk doesn't accept keystrokes during playback,
use BackGround to play while waiting for keystrokes.
/Stig
At 08:37 2004-03-21 -0500, you wrote:
Hi all,
I've built the usual press one for sales, 2 for support IVR which works
fine but I'm having difficulty in allowing callers to type in whole
extension numbers.
My internal extn ranges are 3xxx and 4xxx. I have pasted the IVR below
(just in case someone wants one). The welcome message states callers
should type in the extension number they want or choose from the options.
It seems though that one can only press one number before the IVR moves to
the next step.
I'm starting to think that if my extn's are 3xxx and 4xxx I can't have any
menu choices beginning with 3 or 4. Would this be correct? If so how does
the received DTMF break out of the IVR and get matched to the relevant
dialplan entry?
[mainmenu]
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,3
exten = s,4,ResponseTimeout,5
;SAI menu - 1 for tech support, 2 for voicemail, 3 for echo test
exten = s,5,Background(welcomemsg)
exten = s,6,Background(choosemsg)
; Sales
exten = 1,1,Dial,SIP/3400|20
exten = 1,2,Voicemail(3400)
exten = 1,3,Goto(mainmenu,s,60
; Tech support
exten = 2,1,Dial,SIP/3401|20
exten = 2,2,Voicemail(3401)
exten = 2,2,Goto(mainmenu,s,1)
; Echo Test
exten = 3,1,Playback(demo-echotest)
exten = 3,2,Echo
exten = 3,3,Playback(demo-echodone)
exten = 3,4,Goto(mainmenu,s,6)
; Parrot Test
exten = 4,1,Goto(205,1)
; Access VoiceMail
exten = 5,1,VoicemailMain
exten = 5,2,Goto(mainmenu,s,6)
; Play the weasels
exten = 6,1,Wait,3
exten = 6,2,Playback(tt-somethingwrong)
exten = 6,3,Playback(tt-weasels)
exten = 6,4,Wait,2
exten = 6,5,Goto(mainmenu,s,6)
; # to hangup
exten = #,1,Playback(vm-goodbye)
exten = #,2,Hangup
exten = t,1,Goto(#,1) ; If they take too long, give up
exten = i,1,Playback(invalid) ; That's not valid, try again
Whilst writing this I've had a thought. What would happen if I had an
entry like this?
; transfer to regular extension #
exten = _3XXX,1,Dial(SIP/{EXTN}|20|T)
exten = _4XXX,1,Dial(SIP/{EXTN}|20|T)
Thanks
--
Mark Phillips, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com/
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RE: [Asterisk-Users] firefly softphone
Yup, also experienced the crashes, however... the version available from virbigage site (1.4) does not seem to support SIP even though it is available as choice when setting it up from the installation. Select IAX during setup. When finished setup, select options and choose codecs (they are unselected as per default ). SIP was released beta a few days ago with a message to this list, i believe the download location was same as from their website, but name was firefly-dev.exe. Search the archive here... Works ok, some small bugs - can't send DTMF after connection, but fix is said to be on the way. /Stig At 07:39 2004-03-20 +1100, Simon Brown wrote: I had exactly the same problem. I tried removing and reinstalling several times but it always crashed. I sent an email to verbiage asking for help and all I got in response was Have you got it working yet? from them. I have been unable to get a reply since. Simon Brown -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: Saturday, 20 March 2004 3:48 To: Asterisk List Subject: Re: [Asterisk-Users] firefly softphone On Fri, 2004-03-19 at 17:31, Nick Knight wrote: Hello all, I have tried the firefly softphone on a couple of computers now - and as soon as it registers with the Asterisk server (in fact tries to register) but then crashes and tries to send crash report to MS. Has any one had experience of this. IIRC it's because no codecs have been selected. -- Dave Cotton [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 - This mail was content checked for malicious code and viruses by GFI MailSecurity. ___ 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] Session numbers?
Hi, The messages produced by asterisk console, in vvv mode, what are the numbers after the brackets? in this example, /4 and /5 = Releasing [EMAIL PROTECTED]/4 and IAX2[ulf]/5 Are these session numbers or? Are they reused? When the first call comes after asterisk is restarted, they begin at /1 but 8 hours later, a new single call can have /4 I'm investigating why some calls do not go through to a Firefly client (IAX2) after the client has been busy. I'm suspecting som kind of zombie sessions... anyone? Any ideas? /Stig ___ 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] New Firefly Beta - with SIP and G.729
Hi again, Installed your new release today (after the sip bugfix). Now SIP registers OK with asterisk, but calling fails... Firefly says: Couldn't start call. Asterisk in SIP debug mode shows the registration, but shows no response when firefly tries to call. Using NO stun, asterisk and Firefly on the same net, using only code G:711 u/alaw Registration data follows if of interrest... Regards Stig - Sip read: REGISTER sip:asterisk.ymex.com:5060;transport=udp SIP/2.0 To: sip:[EMAIL PROTECTED]:5060;transport=udp From: sip:[EMAIL PROTECTED]:5060;tag=014ee749 Via: SIP/2.0/UDP 217.119.162.35:5060;branch=z9hG4bK-c87542-386924776-1--c87542-;rport Call-ID: c75e00726c471711 CSeq: 1 REGISTER Contact: sip:[EMAIL PROTECTED]:5060 Expires: 3600 Max-Forwards: 70 User-Agent: Firefly Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 217.119.162.35 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.119.162.35:5060;branch=z9hG4bK-c87542-386924776-1--c87542-;rport From: sip:[EMAIL PROTECTED]:5060;tag=014ee749 To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as13cb66e9 Call-ID: c75e00726c471711 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 217.119.162.35:5060 Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 217.119.162.35:5060;branch=z9hG4bK-c87542-386924776-1--c87542-;rport From: sip:[EMAIL PROTECTED]:5060;tag=014ee749 To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as13cb66e9 Call-ID: c75e00726c471711 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=30bc622a Content-Length: 0 to 217.119.162.35:5060 asterisk*CLI Sip read: REGISTER sip:asterisk.ymex.com:5060;transport=udp SIP/2.0 To: sip:[EMAIL PROTECTED]:5060;transport=udp From: sip:[EMAIL PROTECTED]:5060;tag=6c3de14a Via: SIP/2.0/UDP 217.119.162.35:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport Call-ID: c75e00726c471711 CSeq: 1 REGISTER Contact: sip:[EMAIL PROTECTED]:5060 Expires: 3600 Max-Forwards: 70 Proxy-Authorization: Digest username=stig,realm=asterisk,nonce=30bc622a,uri=sip:asterisk.ymex.com:5060;transport=udp,response=d39488505ce4c15723e4b8f3a7a2bb69,algorithm=MD5 User-Agent: Firefly Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 217.119.162.35 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.119.162.35:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport From: sip:[EMAIL PROTECTED]:5060;tag=6c3de14a To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as13cb66e9 Call-ID: c75e00726c471711 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 217.119.162.35:5060 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 217.119.162.35:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport From: sip:[EMAIL PROTECTED]:5060;tag=6c3de14a To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as13cb66e9 Call-ID: c75e00726c471711 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: sip:[EMAIL PROTECTED];expires=3600 Date: Wed, 17 Mar 2004 07:24:46 GMT Content-Length: 0 to 217.119.162.35:5060 At 17:34 2004-03-17 +1100, you wrote: Just a quick update, there's was a problem with SIP - if you were getting SIP registration failed, grab the new version. (http://www.virbiage.com/firefly/download/firefly-dev.exe) thanks for the feedback about this bug, Adam Adam Hart wrote: I've been sitting on this release for a week so I thought I'd better just release it :) Firefly now has SIP but it's still in a beta state. If you manage to crash it, send me the hex address of the crash. If you find it doesn't work with another SIP phone, let me know and I'll happy get it working for you. I'll be interested to hear people's experiences behind NATs. To download the beta version of Firefly: http://www.virbiage.com/firefly/download/firefly-dev.exe (the current stable version of firefly will not have sip or g.729) G729 support via dll - basically as we all know, G.729 ain't free but you can get a free development version from Voiceage (Sipro), so I've added support for using that. Download http://www.virbiage.com/firefly/download/g729.zip and follow the instructions in the Readme. You'll need to agree to their license and download their library. Firefly's Protocol Support now is: Voip Protocols: SIP, IAX Codecs: ulaw, alaw, iLBC, GSM, G.729 (via DLL) Next major feature will be conferencing. feel free to email me, Adam Hart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To
