[Asterisk-Users] CallerID Transfer
When I receveid a call (num1) in the my office (num2), I transfer the call at the num3, but the callerid is num2, in the telephone3. What can I doing for show the callerid num1? Thanks Yahoo! Mail: gratis 1GB per i messaggi, antispam, antivirus, POP3___ --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] CallerID Transfer
Use the o flag to force the original callerid, not the num2 callerid. example: exten = s,1,Dial(SIP/200,30,ortT) SKM From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk183Sent: Monday, December 12, 2005 2:59 AMTo: asteriskSubject: [Asterisk-Users] CallerID Transfer When I receveid a call (num1) in the my office (num2), I transfer the call at the num3, but the callerid is num2, in the telephone3.What can I doing for show the callerid num1?Thanks Yahoo! Mail: gratis 1GB per i messaggi, antispam, antivirus, POP3 ___ --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
[Asterisk-Users] asterisk1.2.1+realtimedb+voicemail+contexts
It doesn't work anymore! When Voicemail([EMAIL PROTECTED]) is executed, the context is ignored in Asterisk 1.2.1 with realtime voicemailboxes. I upgraded from 1.2.0 to 1.2.1 and it stopped working. When the option searchcontext=yes, and looking at the debug messages of the database queries the WHERE statement doesn't contain the search for context! As a result several mailboxes are returned, all with the same box# but different contexts. When putting searchcontexts=no it simply puts context='' in the WHERE statement, only giving mailboxes with an empty context as a result! Reading through online documentation, the searchcontexts parameter should only apply to behaviour when a mailbox is NOT found with the given parameters. It then should look in the default context (when sc=no) or look in all contexts (sc=yes). I am assuming searchcontexts applies to any Voicemail execution now, not only when a mailbox is not found. Commenting out the searchcontexts parameter has the same behaviour as if it is disabled. When initialising the mailboxes statically, they do work. Only the realtime voicemailboxes seem to have the bug. Cheers, Frank Aartman ___ --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
[Asterisk-Users] [helpp] Problem in astersik
Hi I am very new to asterisk
I am facing some problems
I have installed asterisk on my fedora core 3 by tar.gz
by
#cd /usr/local
#tar -xzvf asterisk.tar.gz
#make
#make install
#make samples
i made following changes in the sip.conf and extention.conf
In sip.conf
[500]
context=fromsip
type=friend
username=500
secret=shanee
callerid=shanee 500
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=info
callgroup=3
pickupgroup=3
qualify=1000
[501]
context=fromsip
type=friend
username=501
secret=shanee
callerid=shanee 501
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=info
callgroup=3
pickupgroup=3
qualify=1000
In externsion.conf
[fromsip]
exten = s,1,Answer( )
exten = _5XX,1,Dial(SIP/${EXTEN},100,tr)
exten = h,1,Hangup
exten = t,1,Hangup
exten = i,1,Hangup
Then What i did is
[EMAIL PROTECTED] asterisk]# asterisk -rvvv
Unable to connect to remote asterisk
[EMAIL PROTECTED] asterisk]# asterisk -c
Asterisk 1.0.9, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
[ Booting...Dec 10
07:09:47 WARNING[865]: chan_oss.c:257 sound_thread: Read error on sound
device: Resource temporarily unavailable
...Dec 10 07:09:47 WARNING[865]: chan_mgcp.c:4050 reload_config:
Unable to get our IP address, MGCP disabled
...Dec 10 07:09:47 WARNING[865]: chan_skinny.c:2587 reload_config:
Unable to get our IP address, Skinny disabled
Illegal instruction
I gave these error to forum and i got reply that you should unload the
mgcp and skinny modules in the modules.conf
so i unload the following modules by
noload = chan_mgcp.so
noload = chan_skinny.so
noload = chan_oss.so
[EMAIL PROTECTED] asterisk]# asterisk -c
Asterisk 1.0.9, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
[
Booting..Illegal
instruction
[EMAIL PROTECTED] asterisk]#
Then i try to start it
[EMAIL PROTECTED] asterisk]# asterisk -r
Unable to connect to remote asterisk
[EMAIL PROTECTED] asterisk]#
So can you tell me why i am having this problem and how can i solve it
Regard
Talat
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Re: [Asterisk-Users] Long and variable echo
The problem has been consistent from 1.0 through CVS to 1.2, and across different machines and distributions. Does anyone have any suggestions on how I can deal with this? I have had echo cancellation happening, but half-duplex speech is not acceptable. You're not using zaptel, what are you using to connect to the outside? (and is it PSTN/ISDN ?) Can you advice on what hardware you're running, that would help.. Regards, Kristof ___ --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
[Asterisk-Users] Problems with current chan-capi-cm
Hi, as of at least Dec 9, but also today, the cvs version of the chan-capi on sf.net gives problems dialing out. The call gets out, but no audio in any direction. Going back to a version from Dec 4th gives a working system again. All against svn branch 1.2 as of Dec 9th. Anyone else experiencing problems with the chan-capi Here is an entry into the log file: Dec 12 09:53:39 ERROR[859] chan_capi.c: CAPI error sending DATA_B3_REQ ID=005 #0 x0232 LEN=0030 Controller/PLCI/NCCI= 0x10103 Data32 = 0x8164078 DataLength = 0xa0 DataHandle = 0x14f Flags = 0x0 Data64 = 0x0 (NCCI=0x10103) (error=0x1103) -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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
[Asterisk-Users] Got clone lock for masquerade crash
Several times asterisk has crashed with this message: Dec 12 09:17:09 DEBUG[6792] chan_sip.c: Setting NAT on RTP to 0 Dec 12 09:17:09 DEBUG[6792] chan_sip.c: Checking SIP call limits for device Dec 12 09:17:09 DEBUG[6792] chan_sip.c: build_route: Contact hop: sip:[EMAIL PROTECTED] Dec 12 09:17:09 DEBUG[6781] channel.c: Avoiding initial deadlock for 'SIP/172.31.0.8-b7402a60' Dec 12 09:17:09 VERBOSE[12050] logger.c: -- Executing Set(SIP/172.31.0.8-b7402a60, GROUP()=active_calls) in new stack Dec 12 09:17:09 DEBUG[12050] pbx.c: Function result is '1' Dec 12 09:17:09 DEBUG[12050] pbx.c: Expression result is '0' Dec 12 09:17:09 VERBOSE[12050] logger.c: -- Executing GotoIf(SIP/172.31.0.8-b7402a60, 0?106) in new stack Dec 12 09:17:09 DEBUG[12050] pbx.c: Not taking any branch Dec 12 09:17:09 VERBOSE[12050] logger.c: -- Executing Dial(SIP/172.31.0.8-b7402a60, SIP/703|30|t) in new stack Dec 12 09:17:09 DEBUG[12050] chan_sip.c: Setting NAT on RTP to 0 Dec 12 09:17:09 DEBUG[12050] chan_sip.c: Outgoing Call for 703 Dec 12 09:17:09 VERBOSE[12050] logger.c: -- Called 703 Dec 12 09:17:09 DEBUG[6792] chan_sip.c: Acked pending invite 102 Dec 12 09:17:09 DEBUG[6792] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Dec 12 09:17:09 VERBOSE[6792] logger.c: -- Got SIP response 302 Moved Temporarily back from 10.0.13.73 Dec 12 09:17:09 DEBUG[6792] chan_sip.c: Found 302 Redirect to extension '601' Dec 12 09:17:09 VERBOSE[12050] logger.c: -- Now forwarding SIP/172.31.0.8-b7402a60 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/703-7e11) Dec 12 09:17:09 DEBUG[12050] chan_sip.c: update_call_counter(703) - decrement call limit counter Dec 12 09:17:09 VERBOSE[12052] logger.c: -- Executing Set(Local/[EMAIL PROTECTED],2, GROUP()=active_calls) in new stack Dec 12 09:17:09 DEBUG[12052] pbx.c: Function result is '2' Dec 12 09:17:09 DEBUG[12052] pbx.c: Expression result is '0' Dec 12 09:17:09 VERBOSE[12052] logger.c: -- Executing GotoIf(Local/[EMAIL PROTECTED],2, 0?106) in new stack Dec 12 09:17:09 DEBUG[12052] pbx.c: Not taking any branch Dec 12 09:17:09 VERBOSE[12052] logger.c: -- Executing Set(Local/[EMAIL PROTECTED],2, CALLERID(all)=Foo 12345678) in new stack Dec 12 09:17:09 VERBOSE[12052] logger.c: -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/[EMAIL PROTECTED]) in new stack Dec 12 09:17:09 DEBUG[12052] chan_sip.c: Setting NAT on RTP to 0 Dec 12 09:17:09 DEBUG[12052] chan_sip.c: Outgoing Call for 601 Dec 12 09:17:09 VERBOSE[12052] logger.c: -- Called [EMAIL PROTECTED] Dec 12 09:17:09 DEBUG[6792] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Dec 12 09:17:09 DEBUG[6792] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Dec 12 09:17:09 VERBOSE[12052] logger.c: -- SIP/lpbx01-3fa5 is making progress passing it to Local/[EMAIL PROTECTED],2 Dec 12 09:17:09 VERBOSE[12050] logger.c: -- Local/[EMAIL PROTECTED],1 is making progress passing it to SIP/172.31.0.8-b7402a60 Dec 12 09:17:13 DEBUG[6792] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Dec 12 09:17:13 VERBOSE[12052] logger.c: -- SIP/lpbx01-3fa5 is ringing Dec 12 09:17:13 VERBOSE[12050] logger.c: -- Local/[EMAIL PROTECTED],1 is ringing Dec 12 09:17:17 DEBUG[6792] chan_sip.c: Acked pending invite 102 Dec 12 09:17:17 DEBUG[6792] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Dec 12 09:17:17 DEBUG[6792] chan_sip.c: build_route: Contact hop: sip:[EMAIL PROTECTED] Dec 12 09:17:17 VERBOSE[12052] logger.c: -- SIP/lpbx01-3fa5 answered Local/[EMAIL PROTECTED],2 Dec 12 09:17:17 VERBOSE[12050] logger.c: -- Local/[EMAIL PROTECTED],1 stopped sounds Dec 12 09:17:17 VERBOSE[12050] logger.c: -- Local/[EMAIL PROTECTED],1 answered SIP/172.31.0.8-b7402a60 Dec 12 09:17:17 DEBUG[6781] channel.c: Avoiding initial deadlock for 'SIP/172.31.0.8-b7402a60' Dec 12 09:17:17 DEBUG[6792] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Response 102: Match Found Dec 12 09:17:17 DEBUG[12050] channel.c: Planning to masquerade channel SIP/172.31.0.8-b7402a60 into the structure of Local/[EMAIL PROTECTED],2 Dec 12 09:17:17 DEBUG[12050] channel.c: Done planning to masquerade channel SIP/172.31.0.8-b7402a60 into the structure of Local/[EMAIL PROTECTED],2 Dec 12 09:17:17 DEBUG[12050] chan_local.c: Not posting to queue since already masked on 'Local/[EMAIL PROTECTED],1' Dec 12 09:17:17 DEBUG[12052] channel.c: Got clone lock for masquerade on 'SIP/172.31.0.8-b7402a60' at 0xb7401dc4 The 6011 and the 12345678 numbers are fake, the rest of the log is genuine. After this the log ends because asterisk is dead. Each time asterisk crashed, the Got clone lock for masquerade appeared as the last log entry -- and the log entry never
Re: [Asterisk-Users] [helpp] Problem in astersik
Talat:
asterisk -r means to connect to an asterisk that is already running.
Try asterisk -gc
This will start asterisk and give you a console.
If you just want to run asterisk in the background, just run
asterisk
Then you can connect to that background asterisk with
asterisk -rc
HTH
Roger
Talat Ishtiaq wrote:
Hi I am very new to asterisk
I am facing some problems
I have installed asterisk on my fedora core 3 by tar.gz
by
#cd /usr/local
#tar -xzvf asterisk.tar.gz
#make
#make install
#make samples
i made following changes in the sip.conf and extention.conf
In sip.conf
[500]
context=fromsip
type=friend
username=500
secret=shanee
callerid=shanee 500
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=info
callgroup=3
pickupgroup=3
qualify=1000
[501]
context=fromsip
type=friend
username=501
secret=shanee
callerid=shanee 501
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=info
callgroup=3
pickupgroup=3
qualify=1000
In externsion.conf
[fromsip]
exten = s,1,Answer( )
exten = _5XX,1,Dial(SIP/${EXTEN},100,tr)
exten = h,1,Hangup
exten = t,1,Hangup
exten = i,1,Hangup
Then What i did is
[EMAIL PROTECTED] asterisk]# asterisk -rvvv
Unable to connect to remote asterisk
[EMAIL PROTECTED] asterisk]# asterisk -c
Asterisk 1.0.9, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
[ Booting...Dec 10
07:09:47 WARNING[865]: chan_oss.c:257 sound_thread: Read error on sound
device: Resource temporarily unavailable
...Dec 10 07:09:47 WARNING[865]: chan_mgcp.c:4050 reload_config:
Unable to get our IP address, MGCP disabled
...Dec 10 07:09:47 WARNING[865]: chan_skinny.c:2587 reload_config:
Unable to get our IP address, Skinny disabled
Illegal instruction
I gave these error to forum and i got reply that you should unload the
mgcp and skinny modules in the modules.conf
so i unload the following modules by
noload = chan_mgcp.so
noload = chan_skinny.so
noload = chan_oss.so
[EMAIL PROTECTED] asterisk]# asterisk -c
Asterisk 1.0.9, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
[ Booting..Illegal instruction
[EMAIL PROTECTED] asterisk]#
Then i try to start it
[EMAIL PROTECTED] asterisk]# asterisk -r
Unable to connect to remote asterisk
[EMAIL PROTECTED] asterisk]#
So can you tell me why i am having this problem and how can i solve it
Regard
Talat
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--
Roger Hill 07739 707 180
Perseverance is the hard work you do after you get
tired of doing the hard work you already did.
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[Asterisk-Users] Cisco 7941 difference
Does anybody know what is difference between Cisco phones that end with 1 and 0 (7970 vs 7971; 7960 vs 7961 and 7940 vs 7941)? So far, as I could see and read, only difference is that button which you choose channel, lights on when is activated. Are there any other difference? Models that end with GE have Gigabit Ethernet port. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)393447 e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Asterisk cluster and astdb
Dear dude, Please, if you solve this one and you find enough time, send one e-mail on list with explanation how did you manage to make it all work. Thank you! Tomislav -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dashy dude Sent: 1. prosinac 2005 2:38 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk cluster and astdb Dear All I am trying to build a high availability cluster of asterisk. I am using RedHat cluster management suit on Enterprise edition AS3 Origianally, astdb was located on native hard disk of each server. All my end points are configured for Reinvite=Yes Everrything was working fine and if active server is rebooted, the standby would take over and the ongoing calls will continue without any problem. But this had a problem that the astdb file is not updated with latest end-point information and phones dont get a call untill they re register. To avoid this, I moved the astdb file on the shared storage and created sym links from individual servers. Now, when the active server is rebooted, all the active calls are dropped. Please help me in resolving this. ___ --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
[Asterisk-Users] Variables naming, may be a BUG??
Hi, i discovered that in version 1.2.0 stable if i use a variable like: CALLERID_FOO=12345 in the extension.conf, the variable is not evaluated and left empty. In the 1.0.9 it was not this way. May be a bug? Bye, Marcello ___ --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
[Asterisk-Users] click to dial applications
Greetings to all. I am looking for an application that allows dialing of phone numbers from any Windows application, either via right-click, or via function keys. I am successfully using AstTapi, but want to take this one step further and make Asttapi available from applications other than just Outlook. If anyone knows of such an application, please let me know. I have seen this done on several systems, but have not found any software that allows this. Regards to all, Joe ___ --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
[Asterisk-Users] Problem with Speex
Hi I have installed asterisk 1.0.9 on my laptop which is running Redhat el3. As it is when i use ulaw / alaw codecs my calls r easily getting thru with good quality, but when i resort to speex i am getting the error message on console : chan_sip.c:2792 process_sdp: No compatible codecs! my sip.conf looks like [12] type=friend secret=kk host=dynamic canreinvite=no disallow=all allow=SPEEX context=test_direct dtmfmode=rfc2833 outgoinglimit=1 ;incominglimit=1 [21] type=friend secret=amit host=dynamic canreinvite=no disallow=all allow=SPEEX context=test_direct dtmfmode=rfc2833 outgoinglimit=1 ;incominglimit=1 I am also using a linksys PAP2NA so as to connect two Analogue phones, further i downloaded the latest version of speex for el3 and also the libogg libraries. Further the devel package for speex is also installed. Still when i am making calls i am having problems with asterisk console displaying the above mentioned codec related error message. Regards Hrishikesh shrivastaw India ___ --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] [helpp] Problem in astersik
Hi.First off, the illegal instruction doesn't look at all pretty.The best way to start a new installation is to start asterisk thus:/usr/sbin/asterisk -vcwhen you get a clean start with no fatal or significant errors, you can close it out and then start it as a daemon
/usr/sbin/asteriskOnly then can you use -r to connect to the remote console.I would suggest if you are having illegal instruction' messages, then asterisk isn't starting cleanly, and you have a significant compilation issue on your hands.
I would make sure your FC3 box is up2date and that you are not seeing any significant compilation errors first.Once you have got things going this far, we can then look at your dialplan...cheers,
MarkOn 12/12/05, Talat Ishtiaq [EMAIL PROTECTED] wrote:
Hi I am very new to asteriskI am facing some problemsI have installed asterisk on my fedora core 3 by tar.gzby#cd /usr/local#tar -xzvf asterisk.tar.gz#make#make install#make samples
i made following changes in the sip.conf and extention.confIn sip.conf[500]context=fromsiptype=friendusername=500secret=shaneecallerid=shanee 500host=dynamicnat=yes
canreinvite=nodisallow=allallow=ulawdtmfmode=infocallgroup=3pickupgroup=3qualify=1000[501]context=fromsiptype=friendusername=501secret=shaneecallerid=shanee 501
host=dynamicnat=yescanreinvite=nodisallow=allallow=ulawdtmfmode=infocallgroup=3pickupgroup=3qualify=1000In externsion.conf[fromsip]exten = s,1,Answer( )exten = _5XX,1,Dial(SIP/${EXTEN},100,tr)
exten = h,1,Hangupexten = t,1,Hangupexten = i,1,HangupThen What i did is[EMAIL PROTECTED] asterisk]# asterisk -rvvvUnable to connect to remote asterisk[EMAIL PROTECTED] asterisk]# asterisk -c
Asterisk 1.0.9, Copyright (C) 1999-2004 Digium.Written by Mark Spencer [EMAIL PROTECTED]=
[ Booting...Dec 1007:09:47 WARNING[865]: chan_oss.c:257 sound_thread: Read error on sounddevice: Resource temporarily unavailable...Dec 10 07:09:47 WARNING[865]: chan_mgcp.c:4050 reload_config:
Unable to get our IP address, MGCP disabled...Dec 10 07:09:47 WARNING[865]: chan_skinny.c:2587 reload_config:Unable to get our IP address, Skinny disabledIllegal instruction
I gave these error to forum and i got reply that you should unload themgcp and skinny modules in the modules.confso i unload the following modules bynoload = chan_mgcp.sonoload = chan_skinny.so
noload = chan_oss.so[EMAIL PROTECTED] asterisk]# asterisk -cAsterisk 1.0.9, Copyright (C) 1999-2004 Digium.Written by Mark Spencer [EMAIL PROTECTED]
=[ Booting..Illegal instruction[EMAIL PROTECTED] asterisk]#
Then i try to start it[EMAIL PROTECTED] asterisk]# asterisk -rUnable to connect to remote asterisk[EMAIL PROTECTED] asterisk]#So can you tell me why i am having this problem and how can i solve it
RegardTalat___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list
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regards,Mark P. EdwardsFWD: 667917
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[Asterisk-Users] PRI E1 - HDLC Bad FCS / HDLC Abort errors
Hello,
Iam trying to configure asterisk with a PRI E1 line. I got to a point
where incoming calls on PRI is landing on asterisk and asterisk
immediately starts throwing the below errors on the console. The call is
dropped at this point somehow.
Dec 12 15:02:57 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Bad FCS (8) on Primary D-channel of span 1
Dec 12 15:02:58 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Bad FCS (8) on Primary D-channel of span 1
Dec 12 15:02:59 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
Dec 12 15:03:00 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Bad FCS (8) on Primary D-channel of span 1
Dec 12 15:03:02 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Bad FCS (8) on Primary D-channel of span 1
Dec 12 15:03:03 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
Dec 12 15:03:04 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
Dec 12 15:03:04 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
*CLI show version
Asterisk CVS-v1-0-11/08/05-01:22:43 built by [EMAIL PROTECTED] on a i686
running Linux
/etc/zaptel.conf
span=1,1,0,cas,ami
;span=1,0,0,cas,ami ; have tried this too
bchan=1-15
dchan=16
bchan=17-31
/etc/asterisk/zapata.conf
; T110P - PRI Configuration
signalling=pri_cpe
;switchtype=national
;switchtype=5ess
switchtype=euroisdn
callerid=asreceived
group=1
context=di_mainmenu
channel = 1-15,17-31
cat /proc/zaptel/1
Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 AMI/
1 WCT1/0/1 Clear (In use)
2 WCT1/0/2 Clear (In use)
3 WCT1/0/3 Clear (In use)
4 WCT1/0/4 Clear (In use)
5 WCT1/0/5 Clear (In use)
6 WCT1/0/6 Clear (In use)
7 WCT1/0/7 Clear (In use)
8 WCT1/0/8 Clear (In use)
9 WCT1/0/9 Clear (In use)
10 WCT1/0/10 Clear (In use)
11 WCT1/0/11 Clear (In use)
12 WCT1/0/12 Clear (In use)
13 WCT1/0/13 Clear (In use)
14 WCT1/0/14 Clear (In use)
15 WCT1/0/15 Clear (In use)
16 WCT1/0/16 HDLCFCS (In use)
17 WCT1/0/17 Clear (In use)
18 WCT1/0/18 Clear (In use)
19 WCT1/0/19 Clear (In use)
20 WCT1/0/20 Clear (In use)
21 WCT1/0/21 Clear (In use)
22 WCT1/0/22 Clear (In use)
23 WCT1/0/23 Clear (In use)
24 WCT1/0/24 Clear (In use)
25 WCT1/0/25 Clear (In use)
26 WCT1/0/26 Clear (In use)
27 WCT1/0/27 Clear (In use)
28 WCT1/0/28 Clear (In use)
29 WCT1/0/29 Clear (In use)
30 WCT1/0/30 Clear (In use)
31 WCT1/0/31 Clear (In use)
/etc/asterisk/extensions.conf
[di_mainmenu]
exten = s,1,Answer ; Answer the line
exten = s,2,SetVar(mloop=0) ; main menu loop count
exten = s,3,SetVar(mloop=$[${mloop} + 1]) ; increment by 1 , first run
exten = s,4,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten = s,5,ResponseTimeout,8 ; Set Response Timeout to 8 seconds
exten = s,6,(di_welcome)
exten = 1,1,Goto(di_sales,s,1)
exten = 2,1,Goto(di_techsupport,s,1)
exten = 3,1,Goto(di_hostingsupport,s,1)
exten = 4,1,Goto(di_billing,s,1)
exten = 5,1,Directory(default)
exten = 9,1,Goto(operator,0,1)
include = internal
Iam based in India and the PRI line is from TATA indicom. The switch iam
connected to is a Lucent 5ess. I have tried 5ess as the switchtype too
in zapata.conf and same errors.
I also get the below warning if i remove and put the PRI line into the
T110p.
Dec 12 14:26:30 WARNING[5783]: No D-channels available! Using Primary
channel 16 as D-channel anyway!
zttool shows the PRI status as ok and there are no alarms.
Interrupts details are below. X server is not installed and i have
disabled dma on my disk.
[EMAIL PROTECTED] ~]# cat /proc/interrupts
CPU0 CPU1
0: 751204 690283IO-APIC-edge timer
1: 9 0IO-APIC-edge i8042
8: 1 0IO-APIC-edge rtc
9: 0 0 IO-APIC-level acpi
14: 6203 1100IO-APIC-edge ide0
169: 0 0 IO-APIC-level uhci_hcd
177: 682240 672161 IO-APIC-level uhci_hcd, wctdm
185: 0 0 IO-APIC-level uhci_hcd
193: 1021351252274 IO-APIC-level wcte11xp
201:1257747 100020 IO-APIC-level wctdm
217: 94 6793 IO-APIC-level eth0
225: 8722 0 IO-APIC-level eth1
NMI: 0 0
LOC:14414411441440
ERR: 0
MIS: 0
# hdparm -v /dev/hda
/dev/hda:
multcount= 16 (on)
IO_support = 0 (default 16-bit)
unmaskirq= 1 (on)
using_dma= 1 (on)
keepsettings = 0 (off)
readonly = 0 (off)
readahead= 256 (on)
geometry =
RE: [Asterisk-Users] [helpp] Problem in astersik
Unable to get our IP address, Skinny
disabledIllegal instruction
It seem a processor invalid
instruction.
See the Makefile,
what is you processor ? Pentium ? Amd ? a had this problem
on Soekris.
i had change PROC=i486, i586 and
so
De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Mark
EdwardsEnvoyé: lundi 12 décembre 2005 11:16À:
[EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
DiscussionObjet: Re: [Asterisk-Users] [helpp] Problem in
astersik
Hi.First off, the illegal instruction doesn't look at all
pretty.The best way to start a new installation is to start asterisk
thus:/usr/sbin/asterisk -vcwhen you get a clean start with no
fatal or significant errors, you can close it out and then start it as a daemon
/usr/sbin/asteriskOnly then can you use "-r" to connect to the
remote console.I would suggest if you are having "illegal instruction'
messages, then asterisk isn't starting cleanly, and you have a significant
compilation issue on your hands. I would make sure your FC3 box is
up2date and that you are not seeing any significant compilation errors
first.Once you have got things going this far, we can then look at your
dialplan...cheers,Mark
On 12/12/05, Talat
Ishtiaq [EMAIL PROTECTED]
wrote:
Hi
I am very new to asteriskI am facing some problemsI have installed
asterisk on my fedora core 3 by tar.gzby#cd /usr/local#tar -xzvf
asterisk.tar.gz#make#make install#make samples i made
following changes in the sip.conf and extention.confIn
sip.conf[500]context=fromsiptype=friendusername=500secret=shaneecallerid="shanee"
500host=dynamicnat=yes
canreinvite=nodisallow=allallow=ulawdtmfmode=infocallgroup=3pickupgroup=3qualify=1000[501]context=fromsiptype=friendusername=501secret=shaneecallerid="shanee"
501
host=dynamicnat=yescanreinvite=nodisallow=allallow=ulawdtmfmode=infocallgroup=3pickupgroup=3qualify=1000In
externsion.conf[fromsip]exten = s,1,Answer( )exten =
_5XX,1,Dial(SIP/${EXTEN},100,tr) exten = h,1,Hangupexten =
t,1,Hangupexten = i,1,HangupThen What i did is[EMAIL PROTECTED]
asterisk]# asterisk -rvvvUnable to connect to remote
asterisk[EMAIL PROTECTED] asterisk]# asterisk -c Asterisk 1.0.9, Copyright
(C) 1999-2004 Digium.Written by Mark Spencer [EMAIL PROTECTED]=
[ Booting...Dec
1007:09:47 WARNING[865]: chan_oss.c:257 sound_thread: Read error on
sounddevice: Resource temporarily unavailable...Dec 10
07:09:47 WARNING[865]: chan_mgcp.c:4050 reload_config: Unable to get our
IP address, MGCP disabled...Dec 10 07:09:47 WARNING[865]:
chan_skinny.c:2587 reload_config:Unable to get our IP address, Skinny
disabledIllegal instructionI gave
these error to forum and i got reply that you should unload themgcp and
skinny modules in the modules.confso i unload the following modules
bynoload = chan_mgcp.sonoload = chan_skinny.so noload
= chan_oss.so[EMAIL PROTECTED] asterisk]# asterisk -cAsterisk
1.0.9, Copyright (C) 1999-2004 Digium.Written by Mark Spencer [EMAIL PROTECTED]=[
Booting..Illegal
instruction[EMAIL PROTECTED] asterisk]#Then i try to start
it[EMAIL PROTECTED] asterisk]# asterisk -rUnable to connect to remote
asterisk[EMAIL PROTECTED] asterisk]#So can you tell me why i am
having this problem and how can i solve
itRegardTalat___--Bandwidth
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visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- regards,Mark P. EdwardsFWD:
667917
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Re: [Asterisk-Users] [helpp] Problem in astersik
Hi Roger
#asterisk -vvvgc
gave long list to verbosity and at the end it say
Illegal instruction (core dumped)
when i run
[EMAIL PROTECTED] postfix]# asterisk -rc
Unable to connect to remote asterisk
[EMAIL PROTECTED] postfix]#
According to my last mail i told that in modules.conf i unload few
modules b/c they were giving error when i run #astersik -c
Plz tell me how to solve it
Regard
Talat
On Mon, 2005-12-12 at 09:23 +, Roger Hill wrote:
Talat:
asterisk -r means to connect to an asterisk that is already running.
Try asterisk -gc
This will start asterisk and give you a console.
If you just want to run asterisk in the background, just run
asterisk
Then you can connect to that background asterisk with
asterisk -rc
HTH
Roger
Talat Ishtiaq wrote:
Hi I am very new to asterisk
I am facing some problems
I have installed asterisk on my fedora core 3 by tar.gz
by
#cd /usr/local
#tar -xzvf asterisk.tar.gz
#make
#make install
#make samples
i made following changes in the sip.conf and extention.conf
In sip.conf
[500]
context=fromsip
type=friend
username=500
secret=shanee
callerid=shanee 500
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=info
callgroup=3
pickupgroup=3
qualify=1000
[501]
context=fromsip
type=friend
username=501
secret=shanee
callerid=shanee 501
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=info
callgroup=3
pickupgroup=3
qualify=1000
In externsion.conf
[fromsip]
exten = s,1,Answer( )
exten = _5XX,1,Dial(SIP/${EXTEN},100,tr)
exten = h,1,Hangup
exten = t,1,Hangup
exten = i,1,Hangup
Then What i did is
[EMAIL PROTECTED] asterisk]# asterisk -rvvv
Unable to connect to remote asterisk
[EMAIL PROTECTED] asterisk]# asterisk -c
Asterisk 1.0.9, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
[ Booting...Dec 10
07:09:47 WARNING[865]: chan_oss.c:257 sound_thread: Read error on sound
device: Resource temporarily unavailable
...Dec 10 07:09:47 WARNING[865]: chan_mgcp.c:4050 reload_config:
Unable to get our IP address, MGCP disabled
...Dec 10 07:09:47 WARNING[865]: chan_skinny.c:2587 reload_config:
Unable to get our IP address, Skinny disabled
Illegal instruction
I gave these error to forum and i got reply that you should unload the
mgcp and skinny modules in the modules.conf
so i unload the following modules by
noload = chan_mgcp.so
noload = chan_skinny.so
noload = chan_oss.so
[EMAIL PROTECTED] asterisk]# asterisk -c
Asterisk 1.0.9, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
[
Booting..Illegal
instruction
[EMAIL PROTECTED] asterisk]#
Then i try to start it
[EMAIL PROTECTED] asterisk]# asterisk -r
Unable to connect to remote asterisk
[EMAIL PROTECTED] asterisk]#
So can you tell me why i am having this problem and how can i solve it
Regard
Talat
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[Asterisk-Users] Re: Got clone lock for masquerade crash
Now I have unintentionally reproduced the problem again. The good news is that I have a backtrace: #0 0x08063c25 in ast_do_masquerade (original=0x818e500) at channel.c:2841 2841AST_LIST_INSERT_TAIL(original-varshead, AST_LIST_FIRST(clone-varshead), entries); (gdb) bt #0 0x08063c25 in ast_do_masquerade (original=0x818e500) at channel.c:2841 #1 0x08065d1c in ast_read (chan=0x818e500) at channel.c:1792 #2 0x080686e5 in ast_channel_bridge (c0=0x818e500, c1=0x81bf6f0, config=0xb757ee6c, fo=0xb757e048, rc=0xb757e044) at channel.c:3248 #3 0xb7acc64e in ast_bridge_call (chan=0x818e500, peer=0x81bf6f0, config=0xb757ee6c) at res_features.c:1312 #4 0xb7840774 in dial_exec_full (chan=0x818e500, data=Variable data is not available. ) at app_dial.c:1558 #5 0xb7841e0c in dial_exec (chan=0x818e500, data=0xb7582fe8) at app_dial.c:1600 #6 0x0808cdb3 in pbx_extension_helper (c=0x818e500, con=Variable con is not available. ) at pbx.c:544 #7 0x0808e254 in __ast_pbx_run (c=0x818e500) at pbx.c:2220 #8 0x0808ee5c in pbx_thread (data=0x818e500) at pbx.c:2507 #9 0xb7fb4b80 in start_thread () from /lib/libpthread.so.0 #10 0xb7e989ce in clone () from /lib/libc.so.6 /Benny ___ --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] Problems with current chan-capi-cm
On Mon, 12 Dec 2005, Peer Oliver Schmidt wrote: Hi, as of at least Dec 9, but also today, the cvs version of the chan-capi on sf.net gives problems dialing out. The call gets out, but no audio in any direction. Going back to a version from Dec 4th gives a working system again. All against svn branch 1.2 as of Dec 9th. Anyone else experiencing problems with the chan-capi Here is an entry into the log file: Dec 12 09:53:39 ERROR[859] chan_capi.c: CAPI error sending DATA_B3_REQ ID=005 #0 x0232 LEN=0030 Controller/PLCI/NCCI= 0x10103 Data32 = 0x8164078 DataLength = 0xa0 DataHandle = 0x14f Flags = 0x0 Data64 = 0x0 (NCCI=0x10103) (error=0x1103) error 0x1103 is 'queue full', so the capi driver (isdn card) does not accept further voice packets. Did you try latest CVS (11.12.)? Can you please provide a full log and one with the older, working version too? Armin ___ --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] Long and variable echo
We have had terrible echo when calling brisbanewhere are you calling from/to? PaulH Melb - Original Message - From: James Andrewartha [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, December 12, 2005 6:15 PM Subject: [Asterisk-Users] Long and variable echo Hi all, Yep, I have really bad echo. I've measured it (recording in Asterisk and then measuring in Audacity), and it varies from 160ms to 250ms, which is far above what normal echo cancellation deals with. Additionally, I don't have Zaptel hardware, so I can't use ztmonitor as described in http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html to set my gains, not that I know of a milliwat line in Australia anyway. The problem has been consistent from 1.0 through CVS to 1.2, and across different machines and distributions. Does anyone have any suggestions on how I can deal with this? I have had echo cancellation happening, but half-duplex speech is not acceptable. TIA, -- James Andrewartha Systems Administrator Data Analysis Australia Pty Ltd ___ --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] [helpp] Problem in astersik
Hi Mark
The
#/usr/sbin/astersik -vc
---(many verbosity)
Illegal instruction
[EMAIL PROTECTED] postfix]# /usr/sbin/asterisk
[EMAIL PROTECTED] postfix]# /usr/sbin/asterisk -r
Unable to connect to remote asterisk
still give me illegal instruction in the end.
As long as i remember i did not get any error during make and make
install command.
What packages do you want me to update on my fedora core 3 machine
Plz tell me
Regard
Talat
On Mon, 2005-12-12 at 21:16 +1100, Mark Edwards wrote:
Hi.
First off, the illegal instruction doesn't look at all pretty.
The best way to start a new installation is to start asterisk thus:
/usr/sbin/asterisk -vc
when you get a clean start with no fatal or significant errors, you
can close it out and then start it as a daemon
/usr/sbin/asterisk
Only then can you use -r to connect to the remote console.
I would suggest if you are having illegal instruction' messages, then
asterisk isn't starting cleanly, and you have a significant
compilation issue on your hands.
I would make sure your FC3 box is up2date and that you are not seeing
any significant compilation errors first.
Once you have got things going this far, we can then look at your
dialplan...
cheers,
Mark
On 12/12/05, Talat Ishtiaq [EMAIL PROTECTED] wrote:
Hi I am very new to asterisk
I am facing some problems
I have installed asterisk on my fedora core 3 by tar.gz
by
#cd /usr/local
#tar -xzvf asterisk.tar.gz
#make
#make install
#make samples
i made following changes in the sip.conf and extention.conf
In sip.conf
[500]
context=fromsip
type=friend
username=500
secret=shanee
callerid=shanee 500
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=info
callgroup=3
pickupgroup=3
qualify=1000
[501]
context=fromsip
type=friend
username=501
secret=shanee
callerid=shanee 501
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=info
callgroup=3
pickupgroup=3
qualify=1000
In externsion.conf
[fromsip]
exten = s,1,Answer( )
exten = _5XX,1,Dial(SIP/${EXTEN},100,tr)
exten = h,1,Hangup
exten = t,1,Hangup
exten = i,1,Hangup
Then What i did is
[EMAIL PROTECTED] asterisk]# asterisk -rvvv
Unable to connect to remote asterisk
[EMAIL PROTECTED] asterisk]# asterisk -c
Asterisk 1.0.9, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
[ Booting...Dec 10
07:09:47 WARNING[865]: chan_oss.c:257 sound_thread: Read error
on sound
device: Resource temporarily unavailable
...Dec 10 07:09:47 WARNING[865]: chan_mgcp.c:4050
reload_config:
Unable to get our IP address, MGCP disabled
...Dec 10 07:09:47 WARNING[865]: chan_skinny.c:2587
reload_config:
Unable to get our IP address, Skinny disabled
Illegal instruction
I gave these error to forum and i got reply that you should
unload the
mgcp and skinny modules in the modules.conf
so i unload the following modules by
noload = chan_mgcp.so
noload = chan_skinny.so
noload = chan_oss.so
[EMAIL PROTECTED] asterisk]# asterisk -c
Asterisk 1.0.9, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
[
Booting..Illegal
instruction
[EMAIL PROTECTED] asterisk]#
Then i try to start it
[EMAIL PROTECTED] asterisk]# asterisk -r
Unable to connect to remote asterisk
[EMAIL PROTECTED] asterisk]#
So can you tell me why i am having this problem and how can i
solve it
Regard
Talat
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Re: [Asterisk-Users] Problems with current chan-capi-cm
Armin Schindler schrieb: On Mon, 12 Dec 2005, Peer Oliver Schmidt wrote: Hi, as of at least Dec 9, but also today, the cvs version of the chan-capi on sf.net gives problems dialing out. The call gets out, but no audio in any direction. Going back to a version from Dec 4th gives a working system again. [..] error 0x1103 is 'queue full', so the capi driver (isdn card) does not accept further voice packets. Did you try latest CVS (11.12.)? I cvs checkouted today. Can you please provide a full log and one with the older, working version too? set verbose 50 enough? Or another type of log? The file is massive, and I don't want to waste everybodies bandwith. Thanks for your help. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] [helpp] Problem in astersik
Hello,
This looks like the asterisk you are using was compiled for a higher class
CPU then the one running it.
Regards
Kiss Karoly
On Mon, 12 Dec 2005, Talat Ishtiaq wrote:
Date: Mon, 12 Dec 2005 15:48:11 +0500
From: Talat Ishtiaq [EMAIL PROTECTED]
To: Mark Edwards [EMAIL PROTECTED]
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] [helpp] Problem in astersik
Hi Mark
The
#/usr/sbin/astersik -vc
---(many verbosity)
Illegal instruction
[EMAIL PROTECTED] postfix]# /usr/sbin/asterisk
[EMAIL PROTECTED] postfix]# /usr/sbin/asterisk -r
Unable to connect to remote asterisk
still give me illegal instruction in the end.
As long as i remember i did not get any error during make and make
install command.
What packages do you want me to update on my fedora core 3 machine
Plz tell me
Regard
Talat
On Mon, 2005-12-12 at 21:16 +1100, Mark Edwards wrote:
Hi.
First off, the illegal instruction doesn't look at all pretty.
The best way to start a new installation is to start asterisk thus:
/usr/sbin/asterisk -vc
when you get a clean start with no fatal or significant errors, you
can close it out and then start it as a daemon
/usr/sbin/asterisk
Only then can you use -r to connect to the remote console.
I would suggest if you are having illegal instruction' messages, then
asterisk isn't starting cleanly, and you have a significant
compilation issue on your hands.
I would make sure your FC3 box is up2date and that you are not seeing
any significant compilation errors first.
Once you have got things going this far, we can then look at your
dialplan...
cheers,
Mark
On 12/12/05, Talat Ishtiaq [EMAIL PROTECTED] wrote:
Hi I am very new to asterisk
I am facing some problems
I have installed asterisk on my fedora core 3 by tar.gz
by
#cd /usr/local
#tar -xzvf asterisk.tar.gz
#make
#make install
#make samples
i made following changes in the sip.conf and extention.conf
In sip.conf
[500]
context=fromsip
type=friend
username=500
secret=shanee
callerid=shanee 500
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=info
callgroup=3
pickupgroup=3
qualify=1000
[501]
context=fromsip
type=friend
username=501
secret=shanee
callerid=shanee 501
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=info
callgroup=3
pickupgroup=3
qualify=1000
In externsion.conf
[fromsip]
exten = s,1,Answer( )
exten = _5XX,1,Dial(SIP/${EXTEN},100,tr)
exten = h,1,Hangup
exten = t,1,Hangup
exten = i,1,Hangup
Then What i did is
[EMAIL PROTECTED] asterisk]# asterisk -rvvv
Unable to connect to remote asterisk
[EMAIL PROTECTED] asterisk]# asterisk -c
Asterisk 1.0.9, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
[ Booting...Dec 10
07:09:47 WARNING[865]: chan_oss.c:257 sound_thread: Read error
on sound
device: Resource temporarily unavailable
...Dec 10 07:09:47 WARNING[865]: chan_mgcp.c:4050
reload_config:
Unable to get our IP address, MGCP disabled
...Dec 10 07:09:47 WARNING[865]: chan_skinny.c:2587
reload_config:
Unable to get our IP address, Skinny disabled
Illegal instruction
I gave these error to forum and i got reply that you should
unload the
mgcp and skinny modules in the modules.conf
so i unload the following modules by
noload = chan_mgcp.so
noload = chan_skinny.so
noload = chan_oss.so
[EMAIL PROTECTED] asterisk]# asterisk -c
Asterisk 1.0.9, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
[
Booting..Illegal
instruction
[EMAIL PROTECTED] asterisk]#
Then i try to start it
[EMAIL PROTECTED] asterisk]# asterisk -r
Unable to connect to remote asterisk
[EMAIL PROTECTED] asterisk]#
So can you tell me why i am having this problem and how can i
solve it
Regard
Talat
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To
[Asterisk-Users] Re: Problem with Speex
I do not think that speex is installed by default. run show translations in asterisk and see what you get. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- hrishikesh shrivastaw [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi I have installed asterisk 1.0.9 on my laptop which is running Redhat el3. As it is when i use ulaw / alaw codecs my calls r easily getting thru with good quality, but when i resort to speex i am getting the error message on console : chan_sip.c:2792 process_sdp: No compatible codecs! my sip.conf looks like [12] type=friend secret=kk host=dynamic canreinvite=no disallow=all allow=SPEEX context=test_direct dtmfmode=rfc2833 outgoinglimit=1 ;incominglimit=1 [21] type=friend secret=amit host=dynamic canreinvite=no disallow=all allow=SPEEX context=test_direct dtmfmode=rfc2833 outgoinglimit=1 ;incominglimit=1 I am also using a linksys PAP2NA so as to connect two Analogue phones, further i downloaded the latest version of speex for el3 and also the libogg libraries. Further the devel package for speex is also installed. Still when i am making calls i am having problems with asterisk console displaying the above mentioned codec related error message. Regards Hrishikesh shrivastaw India ___ --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] Mechanisms for Implementing a Common Contact Database
Hi Douglas! 2. It'd be cool if the regcontext command actually did something. There's a myth out there that it does something like execute a command upon registration. Even the O'Reilly The Future of Telephony seems to think this. After reading some posts in the developer discussion I can say it doesn't. It would be great though, if upon registration from a phone, Asterisk could perform some action, say for example copying the registration to another Asterisk system. I have some more reading for that might help you in your quest: manager API peer registration: http://www.voip- info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Events http://www.asteriskjava.org/latest/apidocs/net/sf/asterisk/manager/event/R egistryEvent.html hint and metermaid (not exactly what you are looking for, but still): http://bugs.digium.com/view.php?id=5779 Cheers, Philipp ___ --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
[Asterisk-Users] Re: click to dial applications
I assume that it would have to use a key sequence (Ctrl+Shift+A, etc.) that does a copy of whatever is highlighted (in any app that supports text copy) and pastes it into the dialing app. (either Tapi or softphone) -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Joseph Rothstein [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Greetings to all. I am looking for an application that allows dialing of phone numbers from any Windows application, either via right-click, or via function keys. I am successfully using AstTapi, but want to take this one step further and make Asttapi available from applications other than just Outlook. If anyone knows of such an application, please let me know. I have seen this done on several systems, but have not found any software that allows this. Regards to all, Joe ___ --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
[Asterisk-Users] Re: [helpp] Problem in astersik
/var/log/asterisk/full text file may give you a more specific error.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---- --- - - - -- - - -- - - - --- - -- -
- --- - - -- - -- -- - --
Talat Ishtiaq [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Hi I am very new to asterisk
I am facing some problems
I have installed asterisk on my fedora core 3 by tar.gz
by
#cd /usr/local
#tar -xzvf asterisk.tar.gz
#make
#make install
#make samples
i made following changes in the sip.conf and extention.conf
In sip.conf
[500]
context=fromsip
type=friend
username=500
secret=shanee
callerid=shanee 500
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=info
callgroup=3
pickupgroup=3
qualify=1000
[501]
context=fromsip
type=friend
username=501
secret=shanee
callerid=shanee 501
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=info
callgroup=3
pickupgroup=3
qualify=1000
In externsion.conf
[fromsip]
exten = s,1,Answer( )
exten = _5XX,1,Dial(SIP/${EXTEN},100,tr)
exten = h,1,Hangup
exten = t,1,Hangup
exten = i,1,Hangup
Then What i did is
[EMAIL PROTECTED] asterisk]# asterisk -rvvv
Unable to connect to remote asterisk
[EMAIL PROTECTED] asterisk]# asterisk -c
Asterisk 1.0.9, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
[ Booting...Dec 10
07:09:47 WARNING[865]: chan_oss.c:257 sound_thread: Read error on sound
device: Resource temporarily unavailable
...Dec 10 07:09:47 WARNING[865]: chan_mgcp.c:4050 reload_config:
Unable to get our IP address, MGCP disabled
...Dec 10 07:09:47 WARNING[865]: chan_skinny.c:2587 reload_config:
Unable to get our IP address, Skinny disabled
Illegal instruction
I gave these error to forum and i got reply that you should unload the
mgcp and skinny modules in the modules.conf
so i unload the following modules by
noload = chan_mgcp.so
noload = chan_skinny.so
noload = chan_oss.so
[EMAIL PROTECTED] asterisk]# asterisk -c
Asterisk 1.0.9, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
[
Booting..Illegal
instruction
[EMAIL PROTECTED] asterisk]#
Then i try to start it
[EMAIL PROTECTED] asterisk]# asterisk -r
Unable to connect to remote asterisk
[EMAIL PROTECTED] asterisk]#
So can you tell me why i am having this problem and how can i solve it
Regard
Talat
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Re: [Asterisk-Users] [helpp] Problem in astersik
Hi Kiss
I am trying to run it on p3 machine i think it should be enough
Regard
Talat
On Mon, 2005-12-12 at 12:18 +0100, Kiss Karoly wrote:
Hello,
This looks like the asterisk you are using was compiled for a higher class
CPU then the one running it.
Regards
Kiss Karoly
On Mon, 12 Dec 2005, Talat Ishtiaq wrote:
Date: Mon, 12 Dec 2005 15:48:11 +0500
From: Talat Ishtiaq [EMAIL PROTECTED]
To: Mark Edwards [EMAIL PROTECTED]
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] [helpp] Problem in astersik
Hi Mark
The
#/usr/sbin/astersik -vc
---(many verbosity)
Illegal instruction
[EMAIL PROTECTED] postfix]# /usr/sbin/asterisk
[EMAIL PROTECTED] postfix]# /usr/sbin/asterisk -r
Unable to connect to remote asterisk
still give me illegal instruction in the end.
As long as i remember i did not get any error during make and make
install command.
What packages do you want me to update on my fedora core 3 machine
Plz tell me
Regard
Talat
On Mon, 2005-12-12 at 21:16 +1100, Mark Edwards wrote:
Hi.
First off, the illegal instruction doesn't look at all pretty.
The best way to start a new installation is to start asterisk thus:
/usr/sbin/asterisk -vc
when you get a clean start with no fatal or significant errors, you
can close it out and then start it as a daemon
/usr/sbin/asterisk
Only then can you use -r to connect to the remote console.
I would suggest if you are having illegal instruction' messages, then
asterisk isn't starting cleanly, and you have a significant
compilation issue on your hands.
I would make sure your FC3 box is up2date and that you are not seeing
any significant compilation errors first.
Once you have got things going this far, we can then look at your
dialplan...
cheers,
Mark
On 12/12/05, Talat Ishtiaq [EMAIL PROTECTED] wrote:
Hi I am very new to asterisk
I am facing some problems
I have installed asterisk on my fedora core 3 by tar.gz
by
#cd /usr/local
#tar -xzvf asterisk.tar.gz
#make
#make install
#make samples
i made following changes in the sip.conf and extention.conf
In sip.conf
[500]
context=fromsip
type=friend
username=500
secret=shanee
callerid=shanee 500
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=info
callgroup=3
pickupgroup=3
qualify=1000
[501]
context=fromsip
type=friend
username=501
secret=shanee
callerid=shanee 501
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=info
callgroup=3
pickupgroup=3
qualify=1000
In externsion.conf
[fromsip]
exten = s,1,Answer( )
exten = _5XX,1,Dial(SIP/${EXTEN},100,tr)
exten = h,1,Hangup
exten = t,1,Hangup
exten = i,1,Hangup
Then What i did is
[EMAIL PROTECTED] asterisk]# asterisk -rvvv
Unable to connect to remote asterisk
[EMAIL PROTECTED] asterisk]# asterisk -c
Asterisk 1.0.9, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
[ Booting...Dec 10
07:09:47 WARNING[865]: chan_oss.c:257 sound_thread: Read error
on sound
device: Resource temporarily unavailable
...Dec 10 07:09:47 WARNING[865]: chan_mgcp.c:4050
reload_config:
Unable to get our IP address, MGCP disabled
...Dec 10 07:09:47 WARNING[865]: chan_skinny.c:2587
reload_config:
Unable to get our IP address, Skinny disabled
Illegal instruction
I gave these error to forum and i got reply that you should
unload the
mgcp and skinny modules in the modules.conf
so i unload the following modules by
noload = chan_mgcp.so
noload = chan_skinny.so
noload = chan_oss.so
[EMAIL PROTECTED] asterisk]# asterisk -c
Asterisk 1.0.9, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
[
Booting..Illegal
instruction
[EMAIL PROTECTED] asterisk]#
Then i try
Re: [Asterisk-Users] Asterisk Dial Failover
On Fri, 2005-12-09 at 06:51 -0600, [EMAIL PROTECTED] wrote: Your other option is to setup the OpenSER boxes in a truly redundant configuration using Linux HA (www.linux-ha.org). That way you setup all your PSTN calls to forward to one shared virtual IP between the boxes. One of the boxes is the Master, the other is the Slave. There is a heartbeat between the boxes that goes at a configurable rate. If the Master fails then the Slave will take over and it can even be configured for sub-second failover. I think there is a article on voip-info.org about this, but don't have time to look it up. Not attempting to hijack the thread but does OpenSER/Linux-HA support stateful failover? If not wouldn't you be better off without the virtual IP address and phones that support a secondary proxy so the phone switches over the moment it detects that the primary is down? Regards, Patrick ___ --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] Asterisk on PPC chan_capi issue
On Fri, 2005-12-09 at 15:53 +, Jason Williams wrote: chan_capi registers fine: ** [chan_capi.so] = (Common ISDN API for Asterisk) == This box has 1 capi controller(s). == Reading config for BRI1 -- ast_capi_pvt BRI1-pseudo-D (MSN1,MSN2,capi-in,0,2) (1,4,128) -- ast_capi_pvt BRI1 (MSN1,MSN2,capi-in,0,2) (1,4,128) -- ast_capi_pvt BRI1 (MSN1,MSN2,capi-in,0,2) (1,4,128) -- listening on contr1 CIPmask = 0x1fff03ff == Registered channel type 'CAPI' (Common ISDN API Driver ($Revision: 1.115 $) ) == Registered application 'capiCommand' == Registered custom function VANITYNUMBER Call from my GSM to a SIP phone (exten 1003) via ISDN/CAPI (MSN2): ** == BRI1: Incoming call 'my GSM' - 'MSN2' -- Executing Macro(CAPI/BRI1/MSN2-0, stdexten| 1003|SIP/1003) in new stack -- Executing Dial(CAPI/BRI1/MSN2-0, SIP/1003|10| TtwW) in new stack Dec 6 02:30:47 WARNING[28889]: channel.c:2494 ast_request: No translator path exists for channel type SIP (native 65535) to 0 Dec 6 02:30:47 NOTICE[28889]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) Looks like a codec problem when making calls to the SIP phone, ensure your sip phone has Alaw enabled in sip.conf, and supports the g711alaw codec. In its config The phone has alaw enabled and this exact same setup works fine on a i686 setup. Regards, Patrick ___ --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
[Asterisk-Users] Production Upgrades
OK, I did some testing with asterisk 1.0.7. I went into production with asterisk 1.0.9. I am using a 2 PCI slot 1U Dell 1750 server. I am using 2 TE110P cards. 1 PRI to Telco. 1 EM to Panasonic DBS PBX with 150 people on it. (all Panasonic people reach Telco via asterisk) I technically put the asterisk server into full production a little earlier than I wanted. I used to have Telco into Panasonic and EM to asterisk from there, but my PRI card in the Panasonic died and since we were intending to replace it, I didn't want to spend the money on the card, so I swapped the cables and put the asterisk first. Needs: I need to add a 4port FXO/FXS card. (911 failover and fax) I need to upgrade to a newer server. I would like to go to the 1.2 branch of Asterisk. Questions: How are other upgrading asterisk on production systems? Are you buying duplicate Digium cards to test configs and reduce downtime? (do you have spares on the shelf anyways?) If I do buy duplicate cards, would I be better served to get a 2 or 4 port T1 card instead of 2 1port cards? (actually considering hardware echo can anyway) I intend to recompile on the new server, copy over configs, VMs and retweak my faxtoemail, etc. But I would still need to test my configs to make sure that all previous options are still supported with the 1.2 branch, right? I know that I could have some redundancy in SIP if I use a SER server or two, but this is the server with my T1/PRI links. Please post how you folks are upgrading your production systems. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- ___ --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] Asterisk Dynamic DNS
Branko Samardzic wrote: Hi everyone, I am running two Asterisk servers on two machines that have dynamic DNS due to ISP changing IP address daily. Both servers are registered on DynDns.org and IP update scripts work fine on both machines. However, if one machine changes IP address, other one (that has trunk pointing to machine that changed address) starts displaying that trunk host is not reachable. O.k. I thought, it is DNS propagation problem, but it is NOT! Even one hour after IP change, machine A still points to old IP address and says that it is not reachable. Register both servers to our FreeVoip service. It's free and the registration will update every minute. Then you can use the freevoip numbers to contact each other. This will cost you nothing and is the easiest way in my (not so neutral) opinion. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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
[Asterisk-Users] Digium PCI-X timeline
Does anyone know if there are plans by Digium to have a PCI-X T1 card? If so, any timing information? Although throughput is higher on PCI-X, is interrupt processing any better/worse than standard PCI? -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- ___ --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] Re: Asterisk Users Newsgroup
Thank you! Thank you! Thank you! This is what I was looking for! If you ever come in Croatia, give me a cool. I'll buy you a drink! Tomislav -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Sent: 2. prosinac 2005 16:24 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Asterisk Users Newsgroup I am using http://www.gmane.com/ with my newsreader. You still have to be a list member to post. You can then turn on the vacation option in the list manager to stop receiving emails. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ___ --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] Problems with current chan-capi-cm
On Mon, 12 Dec 2005, Peer Oliver Schmidt wrote: Armin Schindler schrieb: On Mon, 12 Dec 2005, Peer Oliver Schmidt wrote: Hi, as of at least Dec 9, but also today, the cvs version of the chan-capi on sf.net gives problems dialing out. The call gets out, but no audio in any direction. Going back to a version from Dec 4th gives a working system again. [..] error 0x1103 is 'queue full', so the capi driver (isdn card) does not accept further voice packets. Did you try latest CVS (11.12.)? I cvs checkouted today. Can you please provide a full log and one with the older, working version too? set verbose 50 enough? Or another type of log? The file is massive, and I don't want to waste everybodies bandwith. Use 'set verbose 5' and 'capi debug'. You can send the logs to me directly. Armin ___ --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] mpg123 on x86_64 (Opteron MP)
On Sun, 2005-12-11 at 09:56 +1100, Brad wrote: [snip] Anyone able to point me in the right direction to compile this app? It is running ubuntu.. Not really but iirc there is a Mandrake/Mandriva mpg123 SRPM floating around that had some x86_64 patches in it. Maybe you could try to track the SRPM down and use their patches to make it compile. Regards, Patrick ___ --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] Problems with current chan-capi-cm
Hello again, as of at least Dec 9, but also today, the cvs version of the chan-capi on sf.net gives problems dialing out. The call gets out, but no audio in any direction. Going back to a version from Dec 4th gives a working system again. [..] error 0x1103 is 'queue full', so the capi driver (isdn card) does not accept further voice packets. Did you try latest CVS (11.12.)? I cvs checkouted today. Can you please provide a full log and one with the older, working version too? set verbose 50 enough? Or another type of log? The file is massive, and I don't want to waste everybodies bandwith. After reading your notes regarding capi debug I did some more investigation. The solution was simple. One of the ISDN ports of my AVM C4 did not contain PTP in the capi.conf. It did not seem to matter before, but did now. Changed it, and everything is fine and dandy. Thanks for your help, and sorry for the bothering. Have a good week. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Small / embedded system recommendations
Chris Bagnall wrote: We've recently ordered a pair of these: http://www.soekris.com/net4801.htm Which have a standard PCI slot into which I'm hoping a TDM card will work. Their Belgian distributor (kd85.com) appears to have a nice range of expanded cases that might (hopefully) take a TDM card. I'll find out when they arrive I guess. I'm not sure whether a 266Mhz processor would stand a hope in hell of running 60 calls though - I'll leave that one for someone else to answer. Fortunately our requirement is only for 4-6 concurrent calls. Regards, Chris Chris, I had a look at these and came to the conclusion that they're underpowered for our needs. We really need at least 20 calls at once, preferably 60. Alistair Cunningham, Integrics Ltd, +44 (0)7870 699 479 http://integrics.com/ ___ --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] Small / embedded system recommendations
Kristian Kielhofner wrote: P.S. - You should run AstLinux on your net4801: http://www.astlinux.org P.P.S. - I created AstLinux, and it rocks ;)! Kristian, We haven't decided yet, but AstLinux is definitely on our short list. I'm a Debian person myself (I used to be a Debian developer), so am leaning towards making a small Debian install set of packages, but we're still looking at options for now. Persuade me that AstLinux is better! Alistair Cunningham, Integrics Ltd, +44 (0)7870 699 479 http://integrics.com/ ___ --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] Small / embedded system recommendations
On 12/12/05, Alistair Cunningham [EMAIL PROTECTED] wrote: Chris Bagnall wrote: We've recently ordered a pair of these: http://www.soekris.com/net4801.htm Which have a standard PCI slot into which I'm hoping a TDM card will work. Their Belgian distributor (kd85.com) appears to have a nice range of expanded cases that might (hopefully) take a TDM card. I'll find out when they arrive I guess. I'm not sure whether a 266Mhz processor would stand a hope in hell of running 60 calls though - I'll leave that one for someone else to answer. Fortunately our requirement is only for 4-6 concurrent calls. Regards, Chris Chris, I had a look at these and came to the conclusion that they're underpowered for our needs. We really need at least 20 calls at once, preferably 60. Alistair, You're likely to be better off with a Shuttle type PC. That will probably suit your needs nicely. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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] Long and variable echo
Kristof Hardy wrote: The problem has been consistent from 1.0 through CVS to 1.2, and across different machines and distributions. Does anyone have any suggestions on how I can deal with this? I have had echo cancellation happening, but half-duplex speech is not acceptable. You're not using zaptel, what are you using to connect to the outside? (and is it PSTN/ISDN ?) Can you advice on what hardware you're running, that would help.. Echo needs to be canceled at the point where the PSTN is converted to VoIP. ___ --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
[Asterisk-Users] persistentagents, persistentmembers
Is there a way to persist agent statuses after a restart? Support I have to restart Asterisk for some reason, is it possible that all logged in (AgentCallBackLogin) would remain logged in? Thank you Dov ___ --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] Attack dialing
there is an asterisk agi (may not be an agi but is a program, I think it was an agi) to do this for radio stations, perhaps a google for that, I dont remember the exact name used but do remember that someone was speaking about mass dialing to a radio contest line and bridging to their phone once it was connected. I am sure that it can be modified to do just one chanenl if that is desired. If it doesnt exist a timeout could most likely be easily added so that it doesnt continue to dial after some period has elapsed. For radio contests you most likely dont want it to dial all day as the call in parts are short lived. Oh now that's interesting! I don't see it anywhere though... where did you originally see this?! ___ --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] Synthesized Voice for Asterisk
I'll third this Cepstral is superb! And it's at the right price! On 12/10/05, Steve Totaro [EMAIL PROTECTED] wrote: Cepstral sounds great. You can test it for free but I will append some message about being free until you pay for the license. I will be purchasing a license shortly but the way I read it (could be wrong), the licensing is similar to g729 in that a license is only good for a simultaneous use. Thanks, Steve -Original Message- From: John Cianfarani [mailto:[EMAIL PROTECTED] Sent: Friday, December 09, 2005 10:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Synthesized Voice for Asterisk Cepstral has some pretty decent quality voices at like $29 they don't break the bank. https://www.cepstral.com It also can integrate directly into asterisk I believe. Hope that helps John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dakota Sent: Friday, December 09, 2005 7:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Synthesized Voice for Asterisk Are there any cool free software I can use to create automated voice message greetings for my PBX? I want to customized some of my messages, however prefer to use a standard voice. ___ --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 ___ --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
[Asterisk-Users] ChefSec function
Somebody implemented the Chef-Secretary function in asterisk? ___ --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
[Asterisk-Users] Zap Transfer
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I am attempting to connect asterisk to a pbx to setup a 4 port voicemail with auto attendant. What I would like to do is if a call comes in on Zap/1 I would like to play the auto attendant and when they select an extension use a transfer to send them back out Zap/1. I have not found an easy way to do this and I am looking for some guidance. Thank you, Sean -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFDnYQky9wPyZpnL2URArOqAJsGcNqmjGufRS6fN5KW0k/7LtXu6gCgg+R+ jvnUF9Ef/EfK9vA8Pyck/Fc= =KZJW -END PGP SIGNATURE- ___ --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] Zap Transfer
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Disregard, I just found what I was looking for exten = s,1,Answer exten = s,2,Wait(1) exten = s,3,Background(pls-wait-connect-call) exten = s,4,Flash() exten = s,5,SendDTMF($ARG1) exten = s,6,Hangup() Sean Cook wrote: I am attempting to connect asterisk to a pbx to setup a 4 port voicemail with auto attendant. What I would like to do is if a call comes in on Zap/1 I would like to play the auto attendant and when they select an extension use a transfer to send them back out Zap/1. I have not found an easy way to do this and I am looking for some guidance. Thank you, Sean ___ - --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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFDnYTqy9wPyZpnL2URArPrAJ9c7kMiO1FAHV4xclQ/svL8zxYtcACfdc1i emupFjIaNQE3OWt/CafXlm4= =WUNo -END PGP SIGNATURE- ___ --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
[Asterisk-Users] ASTCC/ASTCC anything wrong with that?
List ... Darren,
In order to use a provider with unusual prefix 00
i.e. 001NXXNXX and providing failover to other providers with
the usual 1NXXNXX, decided to:
1. Change dialstr like that: IAX2/$res-{path}$phone|30|HL (excerpt from
below)
if ( $res-{tech} eq IAX2 ) {
$dialstr =
IAX2/$res-{path}/$phone|30|HL(
. ( $maxtime * 60 * 1000 )
. :6:3);
2. EVery trunk is closed lake that:
iaxprovider/
otherprovider/00
yetanother/
Q: see anything very wrong with that?
Thanks,
benchev
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[Asterisk-Users] RE: CallerID Transfer
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Use the o flag to force the original callerid, not the num2 callerid. example: exten = s,1,Dial(SIP/200,30,ortT) And where to put this one? On first or on second call? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime Subscribecontext
Douglas Garstang wrote: The issues of NAT, call limit handling and registration expiration don't sound quite so bad. I think we can live with those, if we can in fact just get a central location database. Do you have any suggestions or ideas about how this can be implemented with Asterisk? Because, honestly, right now this current limitation is proving to be a real thorn in our side. There is no known answer at this time; there are many discussions occurring about this topic and various ways of addressing it, but they are all theoretical at this point and nobody has come up with a solid design. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC/ASTCC anything wrong with that?
Try it out. It looks to me like it would work but I've been wrong
often. :-)
Darren Wiebe
[EMAIL PROTECTED] wrote:
List ... Darren,
In order to use a provider with unusual prefix 00
i.e. 001NXXNXX and providing failover to other providers with
the usual 1NXXNXX, decided to:
1. Change dialstr like that: IAX2/$res-{path}$phone|30|HL (excerpt from
below)
if ( $res-{tech} eq IAX2 ) {
$dialstr =
IAX2/$res-{path}/$phone|30|HL(
. ( $maxtime * 60 * 1000 )
. :6:3);
2. EVery trunk is closed lake that:
iaxprovider/
otherprovider/00
yetanother/
Q: see anything very wrong with that?
Thanks,
benchev
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--
Darren Wiebe
[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing Calling Cards
www.aleph-com.net/astpp
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Re: [Asterisk-Users] Mechanisms for Implementing a Common ContactDatabase
Douglas Garstang wrote: Someone tell me how this sounds please. We will know the IP addresses of all our phones, and the users/extensions on those phones because we will be the ones provisioning them. We therefore write a script that reads from some source (file/database etc) and somehow (means yet to be determined, probably write to astdb) PRIME Asterisk on startup. Ie when asterisk starts up, it's astdb file will contain the location info for every single phone. This sort of info won't change a lot and if it does, it's easy to edit the entries in astdb. Any opinions? If all of that is true, what do you need Realtime for? Just write out configuration files with the information and do a 'reload chan_sip.so'. ___ --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: RE : [Asterisk-Users] zapata directory not found in svn .
[EMAIL PROTECTED] wrote: Hi Kevin and the list, Yes, please, you must. Why? The CVS server is not going away any time soon, and there are no changes in that project nor any commits happening. ___ --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
[Asterisk-Users] How do I remove the temp greeting?!?!
Hi, In Asterisk voicemail... if I record a temporary greeting, how the heck do I delete it and go back to using the normal greetings again?! ___ --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] Digium PCI-X timeline
Steven wrote: Does anyone know if there are plans by Digium to have a PCI-X T1 card? If so, any timing information? We cannot say anything about our future product plans, sorry :-( Although throughput is higher on PCI-X, is interrupt processing any better/worse than standard PCI? No, it's identical. The major differences are in address/data transfer. ___ --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] Attack dialing
On Mon, 2005-12-12 at 08:49 -0500, Matt wrote: there is an asterisk agi (may not be an agi but is a program, I think it was an agi) to do this for radio stations, perhaps a google for that, I dont remember the exact name used but do remember that someone was speaking about mass dialing to a radio contest line and bridging to their phone once it was connected. I am sure that it can be modified to do just one chanenl if that is desired. If it doesnt exist a timeout could most likely be easily added so that it doesnt continue to dial after some period has elapsed. For radio contests you most likely dont want it to dial all day as the call in parts are short lived. Oh now that's interesting! I don't see it anywhere though... where did you originally see this?! I dont remember but if I had to guess here.. I did a quick google and didnt get anything, but I am fairly sure that I saw someone talking about it and didnt dream that idea up myself, I do remember thinking 'gee that would be nice if I could actually get a radio station where I live'. They were talking about using VoIP (sip I think) to generate a large amount of channels all at the same time to guarantee they could get at least one line in. With the 'enter every 30 days' policies that most radio stations have odds are you will win every month. They said that when it connected it just rang their sip phone. This would be trivial to do (although a bit klugy) with the outgoing queue. You just set it to dial via whatever channel is appropriate and have it dial when connected (I would think about tossing it in a queue so you can deal with the calls in an orderly fashion as some contests answer 'you are caller 1' even though you didnt win). I *think* the original poster said they had a special extension that they called that spawned all this off, which indicates an AGI - although I could be wrong, I am fairly sure they did say something about that. Hopefully this gives people either enough ideas to write something which wouldnt be that hard as described above, or even to find the original author (unless it was all in my mind!) and use his package. I almost think it would be faster to write it yourself than to hunt down the original guy though. What I said above is what 15 minutes tops to write and test? -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --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] How do I remove the temp greeting?!?!
0-4-2 Options, Temporary Greeting, Delete Temporary Greeting Matt wrote: Hi, In Asterisk voicemail... if I record a temporary greeting, how the heck do I delete it and go back to using the normal greetings again?! ___ --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] How do I remove the temp greeting?!?!
Go back into voicemail , mailbox options, press 4 to record temp greeting . Once in this level of the menu you should have options 1 2. Option # 2 will be delete temp greeting. Matt wrote: Hi, In Asterisk voicemail... if I record a temporary greeting, how the heck do I delete it and go back to using the normal greetings again?! ___ --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 -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:[EMAIL PROTECTED] ___ --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] Digium PCI-X timeline
Kevin P. Fleming wrote: Steven wrote: Does anyone know if there are plans by Digium to have a PCI-X T1 card? If so, any timing information? We cannot say anything about our future product plans, sorry :-( Although throughput is higher on PCI-X, is interrupt processing any better/worse than standard PCI? No, it's identical. The major differences are in address/data transfer. Wouldn't anything new and high performance now be PCI-E, and not PCI-X? I know hardly anything but video cards, and the occassional high end RAID card, uses PCI-E, but it seems like that would be the direction for a new card. Steve ___ --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] Realtime Subscribecontext
Could DUNDI help him? Or maybe a OpenSER plus Asterisk environment... Denis. On 12 de dez de 2005, at 12:41, Kevin P. Fleming wrote: Douglas Garstang wrote: The issues of NAT, call limit handling and registration expiration don't sound quite so bad. I think we can live with those, if we can in fact just get a central location database. Do you have any suggestions or ideas about how this can be implemented with Asterisk? Because, honestly, right now this current limitation is proving to be a real thorn in our side. There is no known answer at this time; there are many discussions occurring about this topic and various ways of addressing it, but they are all theoretical at this point and nobody has come up with a solid design. ___ --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] RE:IConnecthere dial out problems
I finally got my issue resolved. It actually had nothing to do with my SIP.conf file. The problem was how I was trying to set the callerid in my extensions.conf file. Anyway, do you have other voip providers that are working? Do incoming calls work at all prior to timing out? Are you NAT'ed, or are you behind a broadband router? If you are NAT'ed and you haven't already configured * for it you may have issues like this. (from http://www.voip-info.org/wiki/view/tips) When sip is behind a NAT do not forget to specify: in sip.conf [general] nat=yes externip = X.X.X.X fromdomain = yourdomain.com localnet = 192.168.X.0/255.255.255.0 I choose to use externhost = yourdomain.com instead of externip since most broadband providers use DHCP and you address can change. I registered a domain with no-ip.com (it's free) and use that in place of yourdomain.com. They also have a client that you can load on a windows box that keeps track of your external ip and updates your domain if your ip ever changes. That way I don't have to worry about it. You may need to add srvlookup to your sip.conf to allow name resolution if you use externhost instead of externip: srvlookup=yes You will also need to setup port forwarding on your broadband router/firewall: (from http://www.voip-info.org/wiki/view/NAT+and+VOIP) SIP signaling: Ports 5060 to 5070 RTP audio: Ports 8766 to 35000 I only forward the following listening ports (read comments in the wiki for this) SIP signaling: Port 5060 RTP audio: Ports 1000 - 2000 (you can restrict this in RTP.conf) Try these wiki pages for more info: http://www.voip-info.org/wiki-Asterisk+config+rtp.conf http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions http://www.voip-info.org/wiki/view/NAT+and+VOIP - Original Message - From: Dennis Gilmore To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] RE:IConnecthere dial out problems Date: Wed, 7 Dec 2005 21:49:01 -0600 Once upon a time Wednesday 07 December 2005 8:42 pm, John Voss wrote: Your SIP.conf file looks much different than mine. I'll give it a try. Hope mine helped [iconnect] type=friend secret= username= host=213.137.73.140 ;sipauth.deltathree.com permit=213.137.73.140/255.255.255.0 permit=208.170.168.0/255.255.255.0 disallow=all context=incoming allow=gsm allow=ulaw allow=alaw allow=G726 insecure=very nat=Yes canreinvite=no I don't know what your register line looks like in your SIP.conf. This is mine. register = ::@213.137.73.140:5060 I was unable to receive calls until I added the insecure=very line. mine is register = ::@natrelay.deltathree.com i can receive incomming calls for a little while after a reload but after some timeouts incomming calls stop -- Dennis Gilmore, RHCE http://www.ausil.us 2.dat -- ___ Play 100s of games for FREE! http://games.mail.com/ ___ --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] How do I remove the temp greeting?!?!
Thank you that will do it. Wow that was slightly not intuitive :) On 12/12/05, Steve Blair [EMAIL PROTECTED] wrote: Go back into voicemail , mailbox options, press 4 to record temp greeting . Once in this level of the menu you should have options 1 2. Option # 2 will be delete temp greeting. Matt wrote: Hi, In Asterisk voicemail... if I record a temporary greeting, how the heck do I delete it and go back to using the normal greetings again?! ___ --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 -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:[EMAIL PROTECTED] ___ --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] Attack dialing
I think I've got something whipped up with the queue... however, it would be nice to not have to pick the phone up when it rings, but rather have asterisk just queue as you speak of.. and connect the call to your handset, already off-hook.. is there a way to do that? On 12/12/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: On Mon, 2005-12-12 at 08:49 -0500, Matt wrote: there is an asterisk agi (may not be an agi but is a program, I think it was an agi) to do this for radio stations, perhaps a google for that, I dont remember the exact name used but do remember that someone was speaking about mass dialing to a radio contest line and bridging to their phone once it was connected. I am sure that it can be modified to do just one chanenl if that is desired. If it doesnt exist a timeout could most likely be easily added so that it doesnt continue to dial after some period has elapsed. For radio contests you most likely dont want it to dial all day as the call in parts are short lived. Oh now that's interesting! I don't see it anywhere though... where did you originally see this?! I dont remember but if I had to guess here.. I did a quick google and didnt get anything, but I am fairly sure that I saw someone talking about it and didnt dream that idea up myself, I do remember thinking 'gee that would be nice if I could actually get a radio station where I live'. They were talking about using VoIP (sip I think) to generate a large amount of channels all at the same time to guarantee they could get at least one line in. With the 'enter every 30 days' policies that most radio stations have odds are you will win every month. They said that when it connected it just rang their sip phone. This would be trivial to do (although a bit klugy) with the outgoing queue. You just set it to dial via whatever channel is appropriate and have it dial when connected (I would think about tossing it in a queue so you can deal with the calls in an orderly fashion as some contests answer 'you are caller 1' even though you didnt win). I *think* the original poster said they had a special extension that they called that spawned all this off, which indicates an AGI - although I could be wrong, I am fairly sure they did say something about that. Hopefully this gives people either enough ideas to write something which wouldnt be that hard as described above, or even to find the original author (unless it was all in my mind!) and use his package. I almost think it would be faster to write it yourself than to hunt down the original guy though. What I said above is what 15 minutes tops to write and test? -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (GNU/Linux) iD8DBQBDnYql+1olxlzQw5cRAgxNAKCuFxQfK2h/GI4NJFrxdWIeVRTjBACeOlDx Zysu9DTS/JppidUmG+m+uuU= =co3+ -END PGP SIGNATURE- ___ --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] Synthesized Voice for Asterisk
I posted on the wiki how to use cepstral with weather.agi. I am no programmer so it is a hack but hey, it works great. I like William the best but I have only tried William, Diane, and David. If someone finds one that is better let me know. Thanks Steve I'll third this Cepstral is superb! And it's at the right price! On 12/10/05, Steve Totaro [EMAIL PROTECTED] wrote: Cepstral sounds great. You can test it for free but I will append some message about being free until you pay for the license. I will be purchasing a license shortly but the way I read it (could be wrong), the licensing is similar to g729 in that a license is only good for a simultaneous use. Thanks, Steve -Original Message- From: John Cianfarani [mailto:[EMAIL PROTECTED] Sent: Friday, December 09, 2005 10:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Synthesized Voice for Asterisk Cepstral has some pretty decent quality voices at like $29 they don't break the bank. https://www.cepstral.com It also can integrate directly into asterisk I believe. Hope that helps John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dakota Sent: Friday, December 09, 2005 7:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Synthesized Voice for Asterisk Are there any cool free software I can use to create automated voice message greetings for my PBX? I want to customized some of my messages, however prefer to use a standard voice. ___ --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 ___ --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 ___ --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] mpg123 on x86_64 (Opteron MP)
I wasted a lot of time on this and never figured it out. Finally went with Madplayer. If you find a solution, please let us know. On Sun, 2005-12-11 at 09:56 +1100, Brad wrote: [snip] Anyone able to point me in the right direction to compile this app? It is running ubuntu.. Not really but iirc there is a Mandrake/Mandriva mpg123 SRPM floating around that had some x86_64 patches in it. Maybe you could try to track the SRPM down and use their patches to make it compile. Regards, Patrick ___ --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] mpg123 on x86_64 (Opteron MP)
Patrick wrote: On Sun, 2005-12-11 at 09:56 +1100, Brad wrote: [snip] Anyone able to point me in the right direction to compile this app? It is running ubuntu.. Not really but iirc there is a Mandrake/Mandriva mpg123 SRPM floating around that had some x86_64 patches in it. Maybe you could try to track the SRPM down and use their patches to make it compile. We use MAD (http://www.underbit.com/products/mad/) on x86_64 systems. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --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] PRI E1 - HDLC Bad FCS / HDLC Abort errors
Try PRI debug span 1 and see if that sheds any light on the problem.
Thanks,
Steve
Hello,
Iam trying to configure asterisk with a PRI E1 line. I got to a point
where incoming calls on PRI is landing on asterisk and asterisk
immediately starts throwing the below errors on the console. The call
is
dropped at this point somehow.
Dec 12 15:02:57 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Bad FCS (8) on Primary D-channel of span 1
Dec 12 15:02:58 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Bad FCS (8) on Primary D-channel of span 1
Dec 12 15:02:59 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
Dec 12 15:03:00 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Bad FCS (8) on Primary D-channel of span 1
Dec 12 15:03:02 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Bad FCS (8) on Primary D-channel of span 1
Dec 12 15:03:03 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
Dec 12 15:03:04 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
Dec 12 15:03:04 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
*CLI show version
Asterisk CVS-v1-0-11/08/05-01:22:43 built by [EMAIL PROTECTED] on a i686
running Linux
/etc/zaptel.conf
span=1,1,0,cas,ami
;span=1,0,0,cas,ami ; have tried this too
bchan=1-15
dchan=16
bchan=17-31
/etc/asterisk/zapata.conf
; T110P - PRI Configuration
signalling=pri_cpe
;switchtype=national
;switchtype=5ess
switchtype=euroisdn
callerid=asreceived
group=1
context=di_mainmenu
channel = 1-15,17-31
cat /proc/zaptel/1
Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 AMI/
1 WCT1/0/1 Clear (In use)
2 WCT1/0/2 Clear (In use)
3 WCT1/0/3 Clear (In use)
4 WCT1/0/4 Clear (In use)
5 WCT1/0/5 Clear (In use)
6 WCT1/0/6 Clear (In use)
7 WCT1/0/7 Clear (In use)
8 WCT1/0/8 Clear (In use)
9 WCT1/0/9 Clear (In use)
10 WCT1/0/10 Clear (In use)
11 WCT1/0/11 Clear (In use)
12 WCT1/0/12 Clear (In use)
13 WCT1/0/13 Clear (In use)
14 WCT1/0/14 Clear (In use)
15 WCT1/0/15 Clear (In use)
16 WCT1/0/16 HDLCFCS (In use)
17 WCT1/0/17 Clear (In use)
18 WCT1/0/18 Clear (In use)
19 WCT1/0/19 Clear (In use)
20 WCT1/0/20 Clear (In use)
21 WCT1/0/21 Clear (In use)
22 WCT1/0/22 Clear (In use)
23 WCT1/0/23 Clear (In use)
24 WCT1/0/24 Clear (In use)
25 WCT1/0/25 Clear (In use)
26 WCT1/0/26 Clear (In use)
27 WCT1/0/27 Clear (In use)
28 WCT1/0/28 Clear (In use)
29 WCT1/0/29 Clear (In use)
30 WCT1/0/30 Clear (In use)
31 WCT1/0/31 Clear (In use)
/etc/asterisk/extensions.conf
[di_mainmenu]
exten = s,1,Answer ; Answer the line
exten = s,2,SetVar(mloop=0) ; main menu loop count
exten = s,3,SetVar(mloop=$[${mloop} + 1]) ; increment by 1 , first
run
exten = s,4,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten = s,5,ResponseTimeout,8 ; Set Response Timeout to 8 seconds
exten = s,6,(di_welcome)
exten = 1,1,Goto(di_sales,s,1)
exten = 2,1,Goto(di_techsupport,s,1)
exten = 3,1,Goto(di_hostingsupport,s,1)
exten = 4,1,Goto(di_billing,s,1)
exten = 5,1,Directory(default)
exten = 9,1,Goto(operator,0,1)
include = internal
Iam based in India and the PRI line is from TATA indicom. The switch
iam
connected to is a Lucent 5ess. I have tried 5ess as the switchtype too
in zapata.conf and same errors.
I also get the below warning if i remove and put the PRI line into the
T110p.
Dec 12 14:26:30 WARNING[5783]: No D-channels available! Using Primary
channel 16 as D-channel anyway!
zttool shows the PRI status as ok and there are no alarms.
Interrupts details are below. X server is not installed and i have
disabled dma on my disk.
[EMAIL PROTECTED] ~]# cat /proc/interrupts
CPU0 CPU1
0: 751204 690283IO-APIC-edge timer
1: 9 0IO-APIC-edge i8042
8: 1 0IO-APIC-edge rtc
9: 0 0 IO-APIC-level acpi
14: 6203 1100IO-APIC-edge ide0
169: 0 0 IO-APIC-level uhci_hcd
177: 682240 672161 IO-APIC-level uhci_hcd, wctdm
185: 0 0 IO-APIC-level uhci_hcd
193: 1021351252274 IO-APIC-level wcte11xp
201:1257747 100020 IO-APIC-level wctdm
217: 94 6793 IO-APIC-level eth0
225: 8722 0 IO-APIC-level eth1
NMI: 0 0
LOC:14414411441440
ERR: 0
MIS: 0
# hdparm -v
RE: [Asterisk-Users] New Product ID.
This is called Zone Paging and can be implemented on * Thanks, Steve I am asking all the VOIP Gurus and any developers out there if a product exists and if not if anyone would want to help me develop such product. With the onslaught of new homes that are wired with networking capabilities. I was wondering if there is a product out there developed that can be used by Asterisk for intercom systems in homes, business or multi-dwelling buildings. I want to know if there is a system that you can install that will use SIP as the communication mechanism but install in every room and dial the extension of the rooms or do an extension that does a broadcast for all the intercoms. If this product exists can someone tell me who makes it and point me out to the websites. If not if someone is interested in developing such a product and cobranding it let me know. This unit would be an all in one system wall mounted in rooms that can be used inside or outside of entrance doors without a special intercom system. I believe that such a device would allow better marketing for Asterisk and VOIP systems to make their entrance in the residential field. This would allow builders to further push VOIP in their new dwellings. Thanks. ___ --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
[Asterisk-Users] Unable to prevent SIP to SIP calls from removing Asterisk from Media path
Due to problems with SIP transfers and agents, we are using blind transfers in asterisk(# key) for all calls. With 1.2.1, Asterisk is doing a native bridge regardless. Dial(SIP/phone,,to) Using the above dial string and I see on the console that Asterisk is attempting a native bridge. This breaks the blind transfers :( Also tried putting, the below in sip.conf for the phones without success: canreinvite=no Any advice? --johann ___ --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] Synthesized Voice for Asterisk
Also, you can apparently use Cepstral voices with Festival through a wrapper. http://www.cepstral.com/cgi-bin/downloads?page=other Festival Wrapper - This Perl script creates a directory called cepstral_swift in the current working directory for each voice directory named as an argument. When placed inside Festival's voices/ directory allows access to Cepstral Swift voices from within festival. Thanks, Steve I posted on the wiki how to use cepstral with weather.agi. I am no programmer so it is a hack but hey, it works great. I like William the best but I have only tried William, Diane, and David. If someone finds one that is better let me know. Thanks Steve I'll third this Cepstral is superb! And it's at the right price! On 12/10/05, Steve Totaro [EMAIL PROTECTED] wrote: Cepstral sounds great. You can test it for free but I will append some message about being free until you pay for the license. I will be purchasing a license shortly but the way I read it (could be wrong), the licensing is similar to g729 in that a license is only good for a simultaneous use. Thanks, Steve -Original Message- From: John Cianfarani [mailto:[EMAIL PROTECTED] Sent: Friday, December 09, 2005 10:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Synthesized Voice for Asterisk Cepstral has some pretty decent quality voices at like $29 they don't break the bank. https://www.cepstral.com It also can integrate directly into asterisk I believe. Hope that helps John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dakota Sent: Friday, December 09, 2005 7:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Synthesized Voice for Asterisk Are there any cool free software I can use to create automated voice message greetings for my PBX? I want to customized some of my messages, however prefer to use a standard voice. ___ --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 ___ --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 ___ --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] PRI E1 - HDLC Bad FCS / HDLC Abort errors
Look on mantis for some patch to do hdlc in hardware, it might help.
Zoa
Steve Totaro wrote:
Try PRI debug span 1 and see if that sheds any light on the problem.
Thanks,
Steve
Hello,
Iam trying to configure asterisk with a PRI E1 line. I got to a point
where incoming calls on PRI is landing on asterisk and asterisk
immediately starts throwing the below errors on the console. The call
is
dropped at this point somehow.
Dec 12 15:02:57 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Bad FCS (8) on Primary D-channel of span 1
Dec 12 15:02:58 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Bad FCS (8) on Primary D-channel of span 1
Dec 12 15:02:59 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
Dec 12 15:03:00 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Bad FCS (8) on Primary D-channel of span 1
Dec 12 15:03:02 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Bad FCS (8) on Primary D-channel of span 1
Dec 12 15:03:03 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
Dec 12 15:03:04 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
Dec 12 15:03:04 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
*CLI show version
Asterisk CVS-v1-0-11/08/05-01:22:43 built by [EMAIL PROTECTED] on a i686
running Linux
/etc/zaptel.conf
span=1,1,0,cas,ami
;span=1,0,0,cas,ami ; have tried this too
bchan=1-15
dchan=16
bchan=17-31
/etc/asterisk/zapata.conf
; T110P - PRI Configuration
signalling=pri_cpe
;switchtype=national
;switchtype=5ess
switchtype=euroisdn
callerid=asreceived
group=1
context=di_mainmenu
channel = 1-15,17-31
cat /proc/zaptel/1
Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 AMI/
1 WCT1/0/1 Clear (In use)
2 WCT1/0/2 Clear (In use)
3 WCT1/0/3 Clear (In use)
4 WCT1/0/4 Clear (In use)
5 WCT1/0/5 Clear (In use)
6 WCT1/0/6 Clear (In use)
7 WCT1/0/7 Clear (In use)
8 WCT1/0/8 Clear (In use)
9 WCT1/0/9 Clear (In use)
10 WCT1/0/10 Clear (In use)
11 WCT1/0/11 Clear (In use)
12 WCT1/0/12 Clear (In use)
13 WCT1/0/13 Clear (In use)
14 WCT1/0/14 Clear (In use)
15 WCT1/0/15 Clear (In use)
16 WCT1/0/16 HDLCFCS (In use)
17 WCT1/0/17 Clear (In use)
18 WCT1/0/18 Clear (In use)
19 WCT1/0/19 Clear (In use)
20 WCT1/0/20 Clear (In use)
21 WCT1/0/21 Clear (In use)
22 WCT1/0/22 Clear (In use)
23 WCT1/0/23 Clear (In use)
24 WCT1/0/24 Clear (In use)
25 WCT1/0/25 Clear (In use)
26 WCT1/0/26 Clear (In use)
27 WCT1/0/27 Clear (In use)
28 WCT1/0/28 Clear (In use)
29 WCT1/0/29 Clear (In use)
30 WCT1/0/30 Clear (In use)
31 WCT1/0/31 Clear (In use)
/etc/asterisk/extensions.conf
[di_mainmenu]
exten = s,1,Answer ; Answer the line
exten = s,2,SetVar(mloop=0) ; main menu loop count
exten = s,3,SetVar(mloop=$[${mloop} + 1]) ; increment by 1 , first
run
exten = s,4,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten = s,5,ResponseTimeout,8 ; Set Response Timeout to 8 seconds
exten = s,6,(di_welcome)
exten = 1,1,Goto(di_sales,s,1)
exten = 2,1,Goto(di_techsupport,s,1)
exten = 3,1,Goto(di_hostingsupport,s,1)
exten = 4,1,Goto(di_billing,s,1)
exten = 5,1,Directory(default)
exten = 9,1,Goto(operator,0,1)
include = internal
Iam based in India and the PRI line is from TATA indicom. The switch
iam
connected to is a Lucent 5ess. I have tried 5ess as the switchtype too
in zapata.conf and same errors.
I also get the below warning if i remove and put the PRI line into the
T110p.
Dec 12 14:26:30 WARNING[5783]: No D-channels available! Using Primary
channel 16 as D-channel anyway!
zttool shows the PRI status as ok and there are no alarms.
Interrupts details are below. X server is not installed and i have
disabled dma on my disk.
[EMAIL PROTECTED] ~]# cat /proc/interrupts
CPU0 CPU1
0: 751204 690283IO-APIC-edge timer
1: 9 0IO-APIC-edge i8042
8: 1 0IO-APIC-edge rtc
9: 0 0 IO-APIC-level acpi
14: 6203 1100IO-APIC-edge ide0
169: 0 0 IO-APIC-level uhci_hcd
177: 682240 672161 IO-APIC-level uhci_hcd, wctdm
185: 0 0 IO-APIC-level uhci_hcd
193: 1021351252274 IO-APIC-level wcte11xp
201:1257747 100020 IO-APIC-level wctdm
217: 94 6793 IO-APIC-level eth0
225: 8722 0 IO-APIC-level eth1
NMI: 0 0
LOC:14414411441440
ERR: 0
MIS: 0
#
Re: [Asterisk-Users] ASTCC/ASTCC anything wrong with that?
Thanks.
It works fine. I was just curious about
any collateral damages.
Thanks again,
benchev
On Monday 12 December 2005 16:42, Darren Wiebe wrote:
Try it out. It looks to me like it would work but I've been wrong
often. :-)
Darren Wiebe
[EMAIL PROTECTED] wrote:
List ... Darren,
In order to use a provider with unusual prefix 00
i.e. 001NXXNXX and providing failover to other providers with
the usual 1NXXNXX, decided to:
1. Change dialstr like that: IAX2/$res-{path}$phone|30|HL (excerpt from
below)
if ( $res-{tech} eq IAX2 ) {
$dialstr =
IAX2/$res-{path}/$phone|30|HL(
. ( $maxtime * 60 * 1000 )
. :6:3);
2. EVery trunk is closed lake that:
iaxprovider/
otherprovider/00
yetanother/
Q: see anything very wrong with that?
Thanks,
benchev
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Re: [Asterisk-Users] mpg123 on x86_64 (Opteron MP)
On Mon, 2005-12-12 at 08:41 -0700, Jason Becker wrote: Patrick wrote: On Sun, 2005-12-11 at 09:56 +1100, Brad wrote: [snip] Anyone able to point me in the right direction to compile this app? It is running ubuntu.. Not really but iirc there is a Mandrake/Mandriva mpg123 SRPM floating around that had some x86_64 patches in it. Maybe you could try to track the SRPM down and use their patches to make it compile. We use MAD (http://www.underbit.com/products/mad/) on x86_64 systems. I am currently using the [files] option and asterisk-addons for music on hold on x86_64. -- Carlos Chavez Director de Tecnologa Telecomunicaciones Abiertas de Mxico S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --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] Re: Problem with Speex
You need to recompile asterisk after you install speex. On 12/12/05, Steven [EMAIL PROTECTED] wrote: I do not think that speex is installed by default.run show translations in asterisk and see what you get.StevenMay you have the peace and freedom that come from abandoning all hope of having a better past. - - -- - - --- - - --- - -- - - --- - - -- --- -- - --hrishikesh shrivastaw [EMAIL PROTECTED] wrote in messagenews:[EMAIL PROTECTED] ...HiI have installed asterisk 1.0.9 on my laptop which is running Redhat el3.As it is when i use ulaw / alaw codecs my calls r easily getting thruwith good quality, but when i resort to speex i am getting the error message on console : chan_sip.c:2792 process_sdp: No compatiblecodecs!my sip.conf looks like[12]type=friendsecret=kkhost=dynamiccanreinvite=nodisallow=allallow=SPEEX context=test_directdtmfmode=rfc2833outgoinglimit=1;incominglimit=1[21]type=friendsecret=amithost=dynamiccanreinvite=nodisallow=allallow=SPEEXcontext=test_directdtmfmode=rfc2833 outgoinglimit=1;incominglimit=1I am also using a linksys PAP2NA so as to connect two Analogue phones, further i downloaded the latest version of speex for el3 and also thelibogg libraries. Further the devel package for speex is also installed.Still when i am making calls i am having problems with asteriskconsole displaying the above mentioned codec related error message.RegardsHrishikesh shrivastawIndia___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo 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] Digium PCI-X timeline
Steve Underwood wrote: Wouldn't anything new and high performance now be PCI-E, and not PCI-X? I know hardly anything but video cards, and the occassional high end RAID card, uses PCI-E, but it seems like that would be the direction for a new card. Yes, I assumed he meant PCI Express, even though he used the wrong acronym :-) ___ --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
[Asterisk-Users] Dial Cmd Outbound CLID Failure (* 1.2.1)
I've been doing AGI now for 2 years, and this problem is making me feel
like I just started. :) I don't have this problem on pre 1.2
installations, so I'm assuming either this is something new, or I've
missed something in the change logs or on wiki.
Scenario:
Customer disables caller id on their IAD. Customer calls in to * where a
perl AGI script reads in RPID info for the customer, and if
privacy=full, set's the callerid variables with RPID info.
Once callerid is set, the AGI script then dials out to a 3rd party,
however, the caller id info is set to 'Unknown'.
If the customer's IAD re-enables callerid, in the same scenario, the
callerid info is passed perfectly through to the 3rd party via *.
It's pretty obvious that * is honoring the privacy=full and/or
recognizing the 'Anonymous' tag in the 'From' field in the sip packet.
Is there a way to disable this behavior so that the callerid can be
forced when the call egresses the * server, regardless of what the
customer's IAD callerid is set to?
I've verified that my RPID parsing subroutine is completely functional
(by verbosing the variables the subroutine sets), and I've verified that
if I just enable the callerid on the IAD, without changing anything
else, that everything works just fine. I completely bypassed this
subroutine in desperation and just set the CLID stuff manually trying to
get it to work.
Any help would be appreciated; thanks in advance,
- Darren
Detailed info below . . .
AGI Excerpts:
Caller ID methods tried:
$AGI-set_variable('CALLERID(name)',\testing\);
$AGI-set_variable('CALLERID(num)',100);
$AGI-set_callerid(\testing\ 100);
$AGI-set_callerid(100);
$AGI-exec('SET',CALLERID 100);
Dial Command: (btw, I've tried using the pipe 'o' as well)
$AGI-exec('Dial',SIP/[EMAIL PROTECTED]|30);
SIP Excerpts (fields modified for protection :) ):
From the IAD to *:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
192.168.1.100:5065;rport=5065;received=xxx.xxx.xxx.xxx;branch=z9hG4bK-5f1f01ef
From: Anonymous sip:[EMAIL PROTECTED];tag=df69fc0c312eb8bo0
To: sip:[EMAIL PROTECTED]
Remote-Party-ID: TEST
sip:[EMAIL PROTECTED];screen=yes;privacy=full;party=calling
From * out to terminate:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK140bc190;rport
From: Unknown sip:[EMAIL PROTECTED];tag=as335c51b2
To: sip:[EMAIL PROTECTED]
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RE: [Asterisk-Users] Mechanisms for Implementing a Common ContactDatabase
Douglas Garstang wrote: Someone tell me how this sounds please. We will know the IP addresses of all our phones, and the users/extensions on those phones because we will be the ones provisioning them. We therefore write a script that reads from some source (file/database etc) and somehow (means yet to be determined, probably write to astdb) PRIME Asterisk on startup. Ie when asterisk starts up, it's astdb file will contain the location info for every single phone. This sort of info won't change a lot and if it does, it's easy to edit the entries in astdb. Any opinions? If all of that is true, what do you need Realtime for? Just write out configuration files with the information and do a 'reload chan_sip.so'. Just for you I put my reply at the bottom. Can't help you with '' marks though. Some of us are using Exchange which doesn't put them against previous text. A database solution would be far cleaner, that's why. After all, that's the WHOLE POINT of a database. I still can't even fathom why this doesn't work. Asterisk is just doing a SELECT statement to find the location of the other user. Why on god's green earth doesn't that work? ___ --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] Production Upgrades
Steven wrote: Questions: How are other upgrading asterisk on production systems? Are you buying duplicate Digium cards to test configs and reduce downtime? We have purchased a duplicate server (with a spare TE410P Digium card). Our primary interface to the telco is a T1 and we use channel banks for our phones. We are using the low tech, unplug the T1 cables from one server and plug them in the other server to cut to our backup system. Depending on what we are doing we may cut over to our backup phone server during the changes. If we don't and something goes wrong, it would take just a few minutes to cut over. We do not have voicemail on our backup server, but since our upgrades happen after hours it has not been a problem. We have a channel bank dedicated for testing any changes. It seems to work for us. Don Pobanz ___ --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
[Asterisk-Users] Outgoing data call
How could I make a modem-based banking software dialing out through Asterisk? I have tried to use an ATA with a lossless compression but the remote modem did not connected. Is it possible to use our Junghanns QuadBRI card as a modem on a dedicated channel and sharing it as a COM port via Samba? Or maybe I am on a wrong track solving this problem? Any advice? Peter Adam ___ --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] PRI E1 - HDLC Bad FCS / HDLC Abort errors
Hey,
Thanks for the suggestion. I did but couldnt understand any of it. Here
is the link if anyone wants to try.
http://pastebin.ca/33394
Iam trying the suggested hardware hdlc patch. Will keep you guys posted.
dushyanth
Try PRI debug span 1 and see if that sheds any light on the problem.
Thanks,
Steve
Hello,
Iam trying to configure asterisk with a PRI E1 line. I got to a point
where incoming calls on PRI is landing on asterisk and asterisk
immediately starts throwing the below errors on the console. The call
is
dropped at this point somehow.
Dec 12 15:02:57 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Bad FCS (8) on Primary D-channel of span 1
Dec 12 15:02:58 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Bad FCS (8) on Primary D-channel of span 1
Dec 12 15:02:59 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
Dec 12 15:03:00 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Bad FCS (8) on Primary D-channel of span 1
Dec 12 15:03:02 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Bad FCS (8) on Primary D-channel of span 1
Dec 12 15:03:03 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
Dec 12 15:03:04 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
Dec 12 15:03:04 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
*CLI show version
Asterisk CVS-v1-0-11/08/05-01:22:43 built by [EMAIL PROTECTED] on a i686
running Linux
/etc/zaptel.conf
span=1,1,0,cas,ami
;span=1,0,0,cas,ami ; have tried this too
bchan=1-15
dchan=16
bchan=17-31
/etc/asterisk/zapata.conf
; T110P - PRI Configuration
signalling=pri_cpe
;switchtype=national
;switchtype=5ess
switchtype=euroisdn
callerid=asreceived
group=1
context=di_mainmenu
channel = 1-15,17-31
cat /proc/zaptel/1
Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 AMI/
1 WCT1/0/1 Clear (In use)
2 WCT1/0/2 Clear (In use)
3 WCT1/0/3 Clear (In use)
4 WCT1/0/4 Clear (In use)
5 WCT1/0/5 Clear (In use)
6 WCT1/0/6 Clear (In use)
7 WCT1/0/7 Clear (In use)
8 WCT1/0/8 Clear (In use)
9 WCT1/0/9 Clear (In use)
10 WCT1/0/10 Clear (In use)
11 WCT1/0/11 Clear (In use)
12 WCT1/0/12 Clear (In use)
13 WCT1/0/13 Clear (In use)
14 WCT1/0/14 Clear (In use)
15 WCT1/0/15 Clear (In use)
16 WCT1/0/16 HDLCFCS (In use)
17 WCT1/0/17 Clear (In use)
18 WCT1/0/18 Clear (In use)
19 WCT1/0/19 Clear (In use)
20 WCT1/0/20 Clear (In use)
21 WCT1/0/21 Clear (In use)
22 WCT1/0/22 Clear (In use)
23 WCT1/0/23 Clear (In use)
24 WCT1/0/24 Clear (In use)
25 WCT1/0/25 Clear (In use)
26 WCT1/0/26 Clear (In use)
27 WCT1/0/27 Clear (In use)
28 WCT1/0/28 Clear (In use)
29 WCT1/0/29 Clear (In use)
30 WCT1/0/30 Clear (In use)
31 WCT1/0/31 Clear (In use)
/etc/asterisk/extensions.conf
[di_mainmenu]
exten = s,1,Answer ; Answer the line
exten = s,2,SetVar(mloop=0) ; main menu loop count
exten = s,3,SetVar(mloop=$[${mloop} + 1]) ; increment by 1 , first
run
exten = s,4,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten = s,5,ResponseTimeout,8 ; Set Response Timeout to 8 seconds
exten = s,6,(di_welcome)
exten = 1,1,Goto(di_sales,s,1)
exten = 2,1,Goto(di_techsupport,s,1)
exten = 3,1,Goto(di_hostingsupport,s,1)
exten = 4,1,Goto(di_billing,s,1)
exten = 5,1,Directory(default)
exten = 9,1,Goto(operator,0,1)
include = internal
Iam based in India and the PRI line is from TATA indicom. The switch
iam
connected to is a Lucent 5ess. I have tried 5ess as the switchtype too
in zapata.conf and same errors.
I also get the below warning if i remove and put the PRI line into the
T110p.
Dec 12 14:26:30 WARNING[5783]: No D-channels available! Using Primary
channel 16 as D-channel anyway!
zttool shows the PRI status as ok and there are no alarms.
Interrupts details are below. X server is not installed and i have
disabled dma on my disk.
[EMAIL PROTECTED] ~]# cat /proc/interrupts
CPU0 CPU1
0: 751204 690283IO-APIC-edge timer
1: 9 0IO-APIC-edge i8042
8: 1 0IO-APIC-edge rtc
9: 0 0 IO-APIC-level acpi
14: 6203 1100IO-APIC-edge ide0
169: 0 0 IO-APIC-level uhci_hcd
177: 682240 672161 IO-APIC-level uhci_hcd, wctdm
185: 0 0 IO-APIC-level uhci_hcd
193: 1021351252274 IO-APIC-level wcte11xp
201:1257747 100020 IO-APIC-level wctdm
217: 94 6793 IO-APIC-level eth0
225: 8722 0
Re: [Asterisk-Users] Problems with current chan-capi-cm
Armin Schindler wrote: On Mon, 12 Dec 2005, Peer Oliver Schmidt wrote: Armin Schindler schrieb: On Mon, 12 Dec 2005, Peer Oliver Schmidt wrote: Hi, as of at least Dec 9, but also today, the cvs version of the chan-capi on sf.net gives problems dialing out. The call gets out, but no audio in any direction. Going back to a version from Dec 4th gives a working system again. [..] error 0x1103 is 'queue full', so the capi driver (isdn card) does not accept further voice packets. Did you try latest CVS (11.12.)? I cvs checkouted today. Can you please provide a full log and one with the older, working version too? set verbose 50 enough? Or another type of log? The file is massive, and I don't want to waste everybodies bandwith. Use 'set verbose 5' and 'capi debug'. You can send the logs to me directly. Armin I know this won't help anybody debug or solve the issue, but I thought it might help to know that others are having the same problem. Mind you I'm using quiet an old Asterisk 1.2 svn version (three weeks ago), with the chan_capi-cm of about three weeks ago too. I didn't even realise I had a problem until a few days ago. For me it works fine after Asterisk is restarted, but at some point later it just stops - it dials, but no audio. I will check out the latest Asterisk and chan_capi-cm and try again over the next week or so. (This is with an AVM Fritz card (BT Speedway) under RH9 btw) Faris. ___ --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] New Product ID.
Interesting, I just came along a product that can do all of this and more, it's called Asterisk. You can find more info here: http://www.asterisk.org/ http://www.digium.com/ http://www.voip-info.org/wiki-asterisk http://www.google.com/ On 12/12/05, Goran Donev [EMAIL PROTECTED] wrote: I am asking all the VOIP Gurus and any developers out there if a product exists and if not if anyone would want to help me develop such product. With the onslaught of new homes that are wired with networking capabilities. I was wondering if there is a product out there developed that can be used by Asterisk for intercom systems in homes, business or multi-dwelling buildings. I want to know if there is a system that you can install that will use SIP as the communication mechanism but install in every room and dial the extension of the rooms or do an extension that does a broadcast for all the intercoms. If this product exists can someone tell me who makes it and point me out to the websites. If not if someone is interested in developing such a product and cobranding it let me know. This unit would be an all in one system wall mounted in rooms that can be used inside or outside of entrance doors without a special intercom system. I believe that such a device would allow better marketing for Asterisk and VOIP systems to make their entrance in the residential field. This would allow builders to further push VOIP in their new dwellings. Thanks. ___ --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
[Asterisk-Users] uniqueid with multiple asterisk hosts
Hello! Soon i will add a second asterisk to my setup and of course i want it to use the same postgresql-db as the first one. Basically it's about the cdr-uniqueid. Since it could be possible that a record with the same uniqueid is written to the cdr-table by both machines i'm lookin for a patch that helps asterisk to produce real unique uniqueid's(don't know why this is not a standard feature anyways). in bugs.digium.com i found several approaches of which the one over here looks the most convincing: http://bugs.digium.com/view.php?id=5825nbn=11 think i'll stick with it but wanted to hear what other people use for this issue? will there be a real uniqueid in the official asterisk code at some point? thanks for your answers. regards christian ___ --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] [helpp] Problem in astersik
On 12/12/05, Talat Ishtiaq [EMAIL PROTECTED] wrote:
Hi Kiss
I am trying to run it on p3 machine i think it should be enough
Sure, but if you got asterisk precompiled with optimizations for PIV
for instance, it won't work.
If you compiled asterisk yourself, be sure you're not using any flags
for a higher processor . Also check youre not using any precompiled
asterisk modules that might have been compiled for a higher class
processor (g723 for instance).
Regard
Talat
On Mon, 2005-12-12 at 12:18 +0100, Kiss Karoly wrote:
Hello,
This looks like the asterisk you are using was compiled for a higher class
CPU then the one running it.
Regards
Kiss Karoly
On Mon, 12 Dec 2005, Talat Ishtiaq wrote:
Date: Mon, 12 Dec 2005 15:48:11 +0500
From: Talat Ishtiaq [EMAIL PROTECTED]
To: Mark Edwards [EMAIL PROTECTED]
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] [helpp] Problem in astersik
Hi Mark
The
#/usr/sbin/astersik -vc
---(many verbosity)
Illegal instruction
[EMAIL PROTECTED] postfix]# /usr/sbin/asterisk
[EMAIL PROTECTED] postfix]# /usr/sbin/asterisk -r
Unable to connect to remote asterisk
still give me illegal instruction in the end.
As long as i remember i did not get any error during make and make
install command.
What packages do you want me to update on my fedora core 3 machine
Plz tell me
Regard
Talat
On Mon, 2005-12-12 at 21:16 +1100, Mark Edwards wrote:
Hi.
First off, the illegal instruction doesn't look at all pretty.
The best way to start a new installation is to start asterisk thus:
/usr/sbin/asterisk -vc
when you get a clean start with no fatal or significant errors, you
can close it out and then start it as a daemon
/usr/sbin/asterisk
Only then can you use -r to connect to the remote console.
I would suggest if you are having illegal instruction' messages, then
asterisk isn't starting cleanly, and you have a significant
compilation issue on your hands.
I would make sure your FC3 box is up2date and that you are not seeing
any significant compilation errors first.
Once you have got things going this far, we can then look at your
dialplan...
cheers,
Mark
On 12/12/05, Talat Ishtiaq [EMAIL PROTECTED] wrote:
Hi I am very new to asterisk
I am facing some problems
I have installed asterisk on my fedora core 3 by tar.gz
by
#cd /usr/local
#tar -xzvf asterisk.tar.gz
#make
#make install
#make samples
i made following changes in the sip.conf and extention.conf
In sip.conf
[500]
context=fromsip
type=friend
username=500
secret=shanee
callerid=shanee 500
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=info
callgroup=3
pickupgroup=3
qualify=1000
[501]
context=fromsip
type=friend
username=501
secret=shanee
callerid=shanee 501
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=info
callgroup=3
pickupgroup=3
qualify=1000
In externsion.conf
[fromsip]
exten = s,1,Answer( )
exten = _5XX,1,Dial(SIP/${EXTEN},100,tr)
exten = h,1,Hangup
exten = t,1,Hangup
exten = i,1,Hangup
Then What i did is
[EMAIL PROTECTED] asterisk]# asterisk -rvvv
Unable to connect to remote asterisk
[EMAIL PROTECTED] asterisk]# asterisk -c
Asterisk 1.0.9, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
[ Booting...Dec
10
07:09:47 WARNING[865]: chan_oss.c:257 sound_thread: Read error
on sound
device: Resource temporarily unavailable
...Dec 10 07:09:47 WARNING[865]: chan_mgcp.c:4050
reload_config:
Unable to get our IP address, MGCP disabled
...Dec 10 07:09:47 WARNING[865]: chan_skinny.c:2587
reload_config:
Unable to get our IP address, Skinny disabled
Illegal instruction
I gave these error to forum and i got reply that you should
unload the
mgcp and skinny modules in the modules.conf
so i unload the following
[Asterisk-Users] Cisco 7940 Reboot
We've currently got 4 servers, and anytime we make any major modifications to the servers, the phones have to be rebooted. We've got about 55 cisco 7940's (which is going to steadily increase over the next few months), does anyone know of a way to reboot the phones without using the telnet function? The powers that be here don't like the telnet cause it's insecure, and I can't really find any other way to do the reboot. Any help would be appreciated. Aaron Daniel Sam Houston State University [EMAIL PROTECTED] ___ --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
[Asterisk-Users] Re: Digium PCI-X timeline
Yes, I meant Express. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Kevin P. Fleming [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Steve Underwood wrote: Wouldn't anything new and high performance now be PCI-E, and not PCI-X? I know hardly anything but video cards, and the occassional high end RAID card, uses PCI-E, but it seems like that would be the direction for a new card. Yes, I assumed he meant PCI Express, even though he used the wrong acronym :-) ___ --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
[Asterisk-Users] executing a reload under stress in Asterisk
Fellow list members, I was wondering if anyone else out there has had issues with Asterisk after doing a reload with a number of users on. I fixed a minor bug in my dial plan and did a reload, and I seemed to have a corrupt config file afterwards. I am also considering there may have been some kind of an issue with copying the files. I keep a seperate copy on a test server, andI copied over to Asterisk before doing the reload. I was getting errors that were basically indicating parts of my dial plan were missing (error indicated I was sending control to priorities that did not exist but they should have). I stopped Asterisk and rebooted the machine and the errors still occured. I copied the files over from my test machine a second time, restarted Asterisk, and everything worked beautifully (as it had on the test machine during the whole process). I have some concern the reload may have been related to this problem, which is serious because it pretty much took down our ability to take calls. It seems like it could have been an error occuring in the process of copying hte files also. For the record I ftp the files between machines to a seperate directory and then "cp" them into the asterisk config directory. Any feedback especially on doing reloads with high call volume and using queues etc, would be appreciated. Thanks in advance, Frank Webb Inter Media Marketing Solutions ___ --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] Attack dialing
Ok.. here is another pet peeve i have.. that maybe someone can answer... When I do a call file. How can I make the call be transfered to the second party BEFORE the first party picks up? In other words.. right now if it's busy it will keep trying.. however if it's ringing... it waits until PARTY 1 picks up and THEN transfers the call to party 2... how can I transfer to party 2 as soon as I start getting ringing to party 1? ___ --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] executing a reload under stress in Asterisk
My guess would be the issue most likely happened in copying the config file over. Why don't you use something like SCP (Secure Copy).. it's like SSH... but for files. On 12/12/05, Franklin Webb [EMAIL PROTECTED] wrote: Fellow list members, I was wondering if anyone else out there has had issues with Asterisk after doing a reload with a number of users on. I fixed a minor bug in my dial plan and did a reload, and I seemed to have a corrupt config file afterwards. I am also considering there may have been some kind of an issue with copying the files. I keep a seperate copy on a test server, and I copied over to Asterisk before doing the reload. I was getting errors that were basically indicating parts of my dial plan were missing (error indicated I was sending control to priorities that did not exist but they should have). I stopped Asterisk and rebooted the machine and the errors still occured. I copied the files over from my test machine a second time, restarted Asterisk, and everything worked beautifully (as it had on the test machine during the whole process). I have some concern the reload may have been related to this problem, which is serious because it pretty much took down our ability to take calls. It seems like it could have been an error occuring in the process of copying hte files also. For the record I ftp the files between machines to a seperate directory and then cp them into the asterisk config directory. Any feedback especially on doing reloads with high call volume and using queues etc, would be appreciated. Thanks in advance, Frank Webb Inter Media Marketing Solutions ___ --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
[Asterisk-Users] Need advice on BRI
Hello all, I need to install a production server with BRI support. I know that exists bristuff, misdn, chan_capi ... I have hcfpci based cards. For a very stable environment, what driver should I use?? Thanks in advance Pedro Nunes ___ --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
[Asterisk-Users] Zultys ZIP2 + asterisk + DTMF on other end? (i.e. ivrs, autoattendants, etc)
Hey all, I have a bunch of Zultys ZIP2 phones, and they work fine with asterisk except on one point, when I make an outgoing call via PSTN/Zap, the call connects all is fine, but if I try to enter in any DTMF tones to navigate a menu at the receiving end, the tones are never recognized by the receiving attendant, any ideas on what needs to be done for the system to send the tones after the call is established? this is only happening on the ZIP2 phones, I have several others that work normally. Thanks in advance ___ --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] Need advice on BRI
Pedro Nunes wrote: I need to install a production server with BRI support. I know that exists bristuff, misdn, chan_capi ... I have hcfpci based cards. For a very stable environment, what driver should I use?? My personal experience is only on using zaptel, it's also the most 'mature' environment. That's why I'm using bristuff, it works with hfc-based cards and quad/octo BRI cards. The advantage is, you can use the more advanced echo cancellers of zaptel. cheers.. ___ --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
[Asterisk-Users] capi incoming call timeout
Hello, Using * 1.2.1 with chan_capi CVS on a Diva server I am mostly happy. However when a phone redirects a call (user forward) and all ISDN channels are busy, the call goes out through an IAX connection and it takes a few seconds to get a ring state from the remote * server. This makes the incoming call (on the Diva) timeout and the caller gets a telco congestion tone. This can be solved by adding a fake ring (r) on the IAX connection Dial() string, as the incoming call now gets a ringing state signaled to it. Is there a way to increase the signaling timeout on the incoming call, so that no fake ringing is required during the IAX call forward? -- I had no wish to arrive, but I had to do my utmost, in order to arrive. -- Samuel Beckett, The Unnamable ___ --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
[Asterisk-Users] Make list of incoming and outgoing calls
Hi, I would like to make a list of all incoming and outgoing calls. From, to, date, duration and for incoming, whether the calls were taken or not and if yes, by which extension. How do I do this? Put a line behind my dial command in the dialplan to save the variables to a file? Any better idea? Cheers, Arik ___ --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] Make list of incoming and outgoing calls
AMP does this by default, and then you can view the call log detail with something like PHPMyAdmin. Otherwise: http://www.google.ca/search?hl=enq=asterisk+call+logmeta= Lots of stuff there. hth -Original Message- From: Arik Funke [mailto:[EMAIL PROTECTED] Sent: Monday, December 12, 2005 11:42 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Make list of incoming and outgoing calls Hi, I would like to make a list of all incoming and outgoing calls. From, to, date, duration and for incoming, whether the calls were taken or not and if yes, by which extension. How do I do this? Put a line behind my dial command in the dialplan to save the variables to a file? Any better idea? Cheers, Arik ___ --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] Make list of incoming and outgoing calls
I would like to make a list of all incoming and outgoing calls. From, to, date, duration and for incoming, whether the calls were taken or not and if yes, by which extension. best is to use the built-in cdr options. (search voip-info.org with asterisk cdr) You can try the mysql cdr, it uses mysql to log all what you need. cheers ___ --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] PRI E1 - HDLC Bad FCS / HDLC Abort errors
Hello,
Iam not sure whether this patch will work for TE110P as the module i
load is wcte11xp and the patches specified at
(http://bugs.digium.com/view.php?id=5313) seems to patch the file
wct4xxp.c in zaptel.
Can anyone please confirm ? Or Wat the heck shuld i just go ahead and try ?
Dushyanth
Look on mantis for some patch to do hdlc in hardware, it might help.
Zoa
Steve Totaro wrote:
Try PRI debug span 1 and see if that sheds any light on the problem.
Thanks,
Steve
Hello,
Iam trying to configure asterisk with a PRI E1 line. I got to a point
where incoming calls on PRI is landing on asterisk and asterisk
immediately starts throwing the below errors on the console. The call
is
dropped at this point somehow.
Dec 12 15:02:57 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Bad FCS (8) on Primary D-channel of span 1
Dec 12 15:02:58 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Bad FCS (8) on Primary D-channel of span 1
Dec 12 15:02:59 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
Dec 12 15:03:00 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Bad FCS (8) on Primary D-channel of span 1
Dec 12 15:03:02 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Bad FCS (8) on Primary D-channel of span 1
Dec 12 15:03:03 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
Dec 12 15:03:04 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
Dec 12 15:03:04 NOTICE[5783]: chan_zap.c:7437 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
*CLI show version
Asterisk CVS-v1-0-11/08/05-01:22:43 built by [EMAIL PROTECTED] on a i686
running Linux
/etc/zaptel.conf
span=1,1,0,cas,ami
;span=1,0,0,cas,ami ; have tried this too
bchan=1-15
dchan=16
bchan=17-31
/etc/asterisk/zapata.conf
; T110P - PRI Configuration
signalling=pri_cpe
;switchtype=national
;switchtype=5ess
switchtype=euroisdn
callerid=asreceived
group=1
context=di_mainmenu
channel = 1-15,17-31
cat /proc/zaptel/1
Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 AMI/
1 WCT1/0/1 Clear (In use)
2 WCT1/0/2 Clear (In use)
3 WCT1/0/3 Clear (In use)
4 WCT1/0/4 Clear (In use)
5 WCT1/0/5 Clear (In use)
6 WCT1/0/6 Clear (In use)
7 WCT1/0/7 Clear (In use)
8 WCT1/0/8 Clear (In use)
9 WCT1/0/9 Clear (In use)
10 WCT1/0/10 Clear (In use)
11 WCT1/0/11 Clear (In use)
12 WCT1/0/12 Clear (In use)
13 WCT1/0/13 Clear (In use)
14 WCT1/0/14 Clear (In use)
15 WCT1/0/15 Clear (In use)
16 WCT1/0/16 HDLCFCS (In use)
17 WCT1/0/17 Clear (In use)
18 WCT1/0/18 Clear (In use)
19 WCT1/0/19 Clear (In use)
20 WCT1/0/20 Clear (In use)
21 WCT1/0/21 Clear (In use)
22 WCT1/0/22 Clear (In use)
23 WCT1/0/23 Clear (In use)
24 WCT1/0/24 Clear (In use)
25 WCT1/0/25 Clear (In use)
26 WCT1/0/26 Clear (In use)
27 WCT1/0/27 Clear (In use)
28 WCT1/0/28 Clear (In use)
29 WCT1/0/29 Clear (In use)
30 WCT1/0/30 Clear (In use)
31 WCT1/0/31 Clear (In use)
/etc/asterisk/extensions.conf
[di_mainmenu]
exten = s,1,Answer ; Answer the line
exten = s,2,SetVar(mloop=0) ; main menu loop count
exten = s,3,SetVar(mloop=$[${mloop} + 1]) ; increment by 1 , first
run
exten = s,4,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten = s,5,ResponseTimeout,8 ; Set Response Timeout to 8 seconds
exten = s,6,(di_welcome)
exten = 1,1,Goto(di_sales,s,1)
exten = 2,1,Goto(di_techsupport,s,1)
exten = 3,1,Goto(di_hostingsupport,s,1)
exten = 4,1,Goto(di_billing,s,1)
exten = 5,1,Directory(default)
exten = 9,1,Goto(operator,0,1)
include = internal
Iam based in India and the PRI line is from TATA indicom. The switch
iam
connected to is a Lucent 5ess. I have tried 5ess as the switchtype too
in zapata.conf and same errors.
I also get the below warning if i remove and put the PRI line into the
T110p.
Dec 12 14:26:30 WARNING[5783]: No D-channels available! Using Primary
channel 16 as D-channel anyway!
zttool shows the PRI status as ok and there are no alarms.
Interrupts details are below. X server is not installed and i have
disabled dma on my disk.
[EMAIL PROTECTED] ~]# cat /proc/interrupts
CPU0 CPU1
0: 751204 690283IO-APIC-edge timer
1: 9 0IO-APIC-edge i8042
8: 1 0IO-APIC-edge rtc
9: 0 0 IO-APIC-level acpi
14: 6203 1100IO-APIC-edge ide0
169: 0 0 IO-APIC-level
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