[Asterisk-Users] I killed my install, help me restore :(
Never try upgrades half-asleep and 1/4-knowledgable! Got a link from a friend about the FLITE TTS that was rewritten to work really well with Asterisk. So I downloaded and installed it on my 1.0.9 server - oops. So, I downloaded Asterisk 1.2.7.1 did the proper install process, got all kinds of warnings about incompatible modules. Forget what all I did but I eventually got it to compile and install, but now when I run asterisk -c it dies at chan_oss. What all directories/files do I need to remove (I have backups at least) to completely remove Asterisk so I can start over with 1.2.7.1? 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
Re: [Asterisk-Users] I killed my install, help me restore :(
I did have some extra modules (mysql_cdr, cepstral tts) but I can start-over. Based on your suggestion, I went one step further. I have gone through and deleted (rm -Rf just to make sure :) ) /etc/asterisk /var/lib/asterisk /usr/lib/asterisk /usr/include/asterisk /usr/sbin/asterisk I am just running the install process again. make clean make make install Will post results as soon as my poor machine finishes the compiling. -Original Message- From: Gareth Blades [mailto:[EMAIL PROTECTED] Sent: Thursday, May 11, 2006 9:38 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Bulk] Re: [Asterisk-Users] I killed my install, help me restore :( It could be an old module still left behind from the previous version. I would delete everything in /usr/lib/asterisk/modules and then reinstall (make install) and see if it will start. On Thu, 2006-05-11 at 14:30, Shawn Porter wrote: Never try upgrades half-asleep and 1/4-knowledgable! Got a link from a friend about the FLITE TTS that was rewritten to work really well with Asterisk. So I downloaded and installed it on my 1.0.9 server - oops. So, I downloaded Asterisk 1.2.7.1 did the proper install process, got all kinds of warnings about incompatible modules. Forget what all I did but I eventually got it to compile and install, but now when I run asterisk -c it dies at chan_oss. What all directories/files do I need to remove (I have backups at least) to completely remove Asterisk so I can start over with 1.2.7.1? 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
RE: [Asterisk-Users] How to make Asterisk to generate and terminatecalls
Ravi, Take a look here. http://www.voip-info.org/wiki-Asterisk+auto-dial+out I would think that for what you are doing use a cron job and a shell script. Shawn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ravi Shankar Sent: Friday, December 23, 2005 8:41 AM To: Asterisk Users Subject: [Asterisk-Users] How to make Asterisk to generate and terminatecalls Hi, I would like to connect two linux machines running asterisk and then originate SIP calls from one asterisk and terminate it on the other asterisk. Terminating the call is not a problem because I can give the call handle to say AGI application on the terminating asterisk. How do i originate a call from the asterisk ? Is this possible using AGI ? Any pointers in this regard would be of great help. This type of application can be used two simulate bulk calls and find out what is the maximum limit for the asterisk in terms of CPU utilization, memory, etc. before it can be deployed in production environment. thanks, Ravi ___ --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] Asterisk Broadvoice help??
Would someone be so kind as to point out what stupid little mistake I have made. I thought I did everything according to the setup page but I fail to register. HOSTS file contains 147.135.8.128 sip.broadvoice.com SIP.CONF [general]context=iaxclients; Default context for incoming callsport=5060; UDP Port to bind to (SIP standard port is 5060)bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all)srvlookup=yes; Enable DNS SRV lookups on outbound calls; Note: Asterisk only uses the first host ; in SRV records; Disabling DNS SRV lookups disables the ; ability to place SIP calls based on domain ; names to some other SIP users on the Internetpedantic=no; Enable slow, pedantic checking for Pingtel; and multiline formatted headers for strict; SIP compatibility (defaults to "no")disallow=all; First disallow all codecsallow=ulaw,alaw,g723,speex.ilbc; Allow codecs in order of preferencedtmfmode=inbandregister = [EMAIL PROTECTED]:XX:[EMAIL PROTECTED]/1001 [1001];shawntype=friendhost=dynamic;dtmfmode=inbandsecret=context=iaxclientscallerid="Oghma Consulting" 647-283- [666]type=friendhost=10.0.0.101canreinvite=nodefaultip=10.0.0.101context=iaxclientsinsecure=very [sip.broadvoice.com]type=peeruser=phonehost=sip.broadvoice.comfromdomain=sip.broadvoice.comfromuser=7723821447secret=xxxusername=7723821447insecure=verycontext=iaxclientsauthname=7723821447dtmfmode=inbanddtmf=inband;Disable canreinvite if you are behind a NATcanreinvite=no SIP DEBUG Asterisk Ready.*CLI sip debugSIP Debugging Enabled*CLI Dec 20 10:51:51 NOTICE[14126]: chan_sip.c:4017 sip_reregister: -- Re-registration for [EMAIL PROTECTED]@sip.broadvoice.com11 headers, 0 linesReliably Transmitting:REGISTER sip:sip.broadvoice.com SIP/2.0Via: SIP/2.0/UDP 10.0.0.164:5060;branch=z9hG4bK4168ff8cFrom: sip:[EMAIL PROTECTED];tag=as565f9ec4To: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 102 REGISTERUser-Agent: Asterisk PBXExpires: 120Contact: sip:[EMAIL PROTECTED]Event: registrationContent-Length: 0 (no NAT) to 147.135.8.128:5060 Sip read:SIP/2.0 401 UnauthorizedCall-ID: [EMAIL PROTECTED]CSeq: 102 REGISTERFrom: sip:[EMAIL PROTECTED];tag=as565f9ec4To: sip:[EMAIL PROTECTED]Via: SIP/2.0/UDP sip.broadvoice.com:5060;branch=z9hG4bK4168ff8cWWW-Authenticate: DIGEST realm="BroadWorks",algorithm=MD5,nonce="1135093911710"Content-Length: 0 8 headers, 0 linesResponding to challenge, registration to domain/host name sip.broadvoice.com12 headers, 0 linesReliably Transmitting:REGISTER sip:sip.broadvoice.com SIP/2.0Via: SIP/2.0/UDP 10.0.0.164:5060;branch=z9hG4bK220b3020From: sip:[EMAIL PROTECTED];tag=as565f9ec4To: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 103 REGISTERUser-Agent: Asterisk PBXAuthorization: Digest username="7723821447", realm="BroadWorks", algorithm=MD5, uri="sip:sip.broadvoice.com", nonce="1135093911710", response="2c73b280cd7857c8f6d2b56acd6e71eb", opaque=""Expires: 120Contact: sip:[EMAIL PROTECTED]Event: registrationContent-Length: 0 (no NAT) to 147.135.8.128:5060 Sip read:SIP/2.0 401 UnauthorizedCall-ID: [EMAIL PROTECTED]CSeq: 103 REGISTERFrom: sip:[EMAIL PROTECTED];tag=as565f9ec4To: sip:[EMAIL PROTECTED]Via: SIP/2.0/UDP sip.broadvoice.com:5060;branch=z9hG4bK220b3020WWW-Authenticate: DIGEST realm="BroadWorks",algorithm=MD5,nonce="1135093911970"Content-Length: 0 8 headers, 0 linesResponding to challenge, registration to domain/host name sip.broadvoice.com12 headers, 0 linesReliably Transmitting:REGISTER sip:sip.broadvoice.com SIP/2.0Via: SIP/2.0/UDP 10.0.0.164:5060;branch=z9hG4bK890cFrom: sip:[EMAIL PROTECTED];tag=as565f9ec4To: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 104 REGISTERUser-Agent: Asterisk PBXAuthorization: Digest username="7723821447", realm="BroadWorks", algorithm=MD5, uri="sip:sip.broadvoice.com", nonce="1135093911970", response="fb0d8ac4bc042e67a716976d4f10004f", opaque=""Expires: 120Contact: sip:[EMAIL PROTECTED]Event: registrationContent-Length: 0 (no NAT) to 147.135.8.128:5060 Sip read:SIP/2.0 401 UnauthorizedCall-ID: [EMAIL PROTECTED]CSeq: 104 REGISTERFrom: sip:[EMAIL PROTECTED];tag=as565f9ec4To: sip:[EMAIL PROTECTED]Via: SIP/2.0/UDP sip.broadvoice.com:5060;branch=z9hG4bK890cWWW-Authenticate: DIGEST realm="BroadWorks",algorithm=MD5,nonce="1135093912150"Content-Length: 0 8 headers, 0 linesDec 20 10:51:52 NOTICE[14126]: chan_sip.c:6854 handle_response: Failed to authenticate on REGISTER to 'sip:[EMAIL PROTECTED];tag=as565f9ec4'Destroying call '[EMAIL PROTECTED]'Dec 20 10:52:11 NOTICE[14126]: chan_sip.c:4045 sip_reg_timeout: -- Registration for '[EMAIL PROTECTED]@sip.broadvoice.com' timed out, trying again12 headers, 0 linesReliably Transmitting:REGISTER sip:sip.broadvoice.com SIP/2.0Via: SIP/2.0/UDP 10.0.0.164:5060;branch=z9hG4bK5242a6a3From: sip:[EMAIL PROTECTED];tag=as1c7995a0To: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 105 REGISTERUser-Agent: Asterisk PBXAuthorization:
RE: [Asterisk-Users] Asterisk Broadvoice help??
Thanks Steven Works great. They should put a little more detail in the setup page as to where you get that password!! very difficult to figure to that out in the wee hours of the morning. Shawn -Original Message- From: Steven Job [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 20, 2005 11:27 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Broadvoice help?? What password are you using? This is the special one they created for you correct? This should not the one that you created on your own (that you use to log in). [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=7723821447 secret=xxx username=7723821447 insecure=very context=iaxclients authname=7723821447 dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no I don't have any context in my configuration. Try removing that. register = [EMAIL PROTECTED]:XX:[EMAIL PROTECTED]/1001 Change that to: register = [EMAIL PROTECTED]:XX:[EMAIL PROTECTED] I never got that extension thing to work. But without it, it did. ___ --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] Linking existing channels through Managerinterface. Is it possible?
I have been wondering the same thing. I would like to be able to link 2 channels inside an AGI script. Also, a way to send variables back-and-forth. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Senyo Sent: Tuesday, December 20, 2005 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Linking existing channels through Managerinterface. Is it possible? Hello, I've been looking for a way to merge two existing asterisk channels manually through the manager interface, but have been unable to find any support for this. Does anyone know if it exist or if there is something out there that might accomplish this? Thanks, ~Senyo ___ --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] Distinctive Ring and zapata.conf
Robert,
This configuration is working fine for me (In ontario with Bell Canada)
dring1 is the 2nd ring pattern on our line, it is a double-ring
dring3 is the regular ring, which I wanted to ignore but since you cant do
that I just send it to a wait loop
ZAPATA.CONF
[channels]
usercallerid=yes
signalling=fxs_ks
usedistinctiveringdetection=1
faxdetect=both
dring1=323,0,0
dring1context=work
dring2=90,0,0
dring2context=home
dring3=0,0,0
dring3context=home
echotraining=800
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
musiconhold=default
channel = 1-2
EXTENSIONS.CONF (partial)
[home]
;ignore 853-1073 calls unless it is a fax
exten = s,1,GotoIf($[${CALLERIDNUM} = 8531073]?5:2)
exten = s,2,Wait,30
exten = s,3,system(/var/lib/asterisk/agi-bin/phone_call.sh ${CALLERID})
exten = s,4,goto(s,7)
exten = s,5,answer
exten = s,6,Goto(fax,1)
exten = s,7,Hangup
exten = fax,1,Macro(home_faxreceive)
[work]
;
; We start with what to do when a call first comes in.
;
exten = s,1,Answer ; Answer the line
exten = s,2,Wait,2 ; Wait a second, just for fun
exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
exten = s,5,AGI(openclose.agi)
exten = s,6,GotoIf($[${STATUS} = closed]?10:7)
exten = s,7,GotoIf($[${STATUS} = holiday]?12:8)
exten = s,8,GotoIf($[${STATUS} = afternoon]?14:16)
exten = s,9,Goto(s,16)
exten = s,10,BackGround(goodevening)
exten = s,11,Goto(s,17)
exten = s,12,BackGround(holiday)
exten = s,13,Goto(s,17)
exten = s,14,BackGround(goodafternoon)
exten = s,15,Goto(s,17)
exten = s,16,BackGround(goodmorning)
exten = s,17,BackGround(greeting)
exten = s,18,BackGround(instruct)
exten = o,1,VoiceMailMain
exten = t,1,playback(goodbye)
exten = t,2,Hangup
exten = i,1,Playback,invalid-exten
exten = i,2,Goto,s|17
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Robert La
Ferla
Sent: Tuesday, December 20, 2005 1:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Distinctive Ring and zapata.conf
Does anyone have distinctive ring working with Asterisk? Could you
share your zapata.conf and relevent extensions.conf?
Thanks.
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RE: [Asterisk-Users] IVR Capacity
Serge, How are you going to be building this server? I am not going to claim to be any sort of expert on sizing, but I do have some experience as an IVR designer/developer. In one of your previous posts you mention E1 cards. In order to get 300 msgs at once you would need to be running 10 E1s, is this reasonable? How many questions are there in your survey, how long is an average call going to take? say you have 5 questions, and a complete call will take approx 2 minutes allow for call-setup and call-breakdown - 15 sec should allow for 26calls per hour per channel @ 30 channels = 780 calls hour using just 1 E1 In this case you would only have, at maximum, 30 prompts playing at any one time. From what I have seen in this mailing list there are servers out there doing a lot more than that (but I do not know what the hardware is). I have built systems running at that level of traffic using the Contarra IVR platform and running on a P4-1.6GHz with 512MB on a dialogic card. Surely an * box with a decent CPU Memory can handle it no problem. Now that I have gone stated all this for the world to criticize, please let me know if I have made any critical mistakes/assumptions. Shawn P.S Contarra Envox I know, Asterisk I am learning. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Serge SchumacherSent: Tuesday, December 20, 2005 4:23 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] IVR Capacity Hi, Do you think * could play around 300 voicemenu messages simoultanously? Regs, Serge ___ --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] IVR Capacity
my own criticism. I just talked with a friend about erlang tables. completely blows away all the stuff I just wrote below... -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Shawn PorterSent: Tuesday, December 20, 2005 4:47 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] IVR Capacity Serge, How are you going to be building this server? I am not going to claim to be any sort of expert on sizing, but I do have some experience as an IVR designer/developer. In one of your previous posts you mention E1 cards. In order to get 300 msgs at once you would need to be running 10 E1s, is this reasonable? How many questions are there in your survey, how long is an average call going to take? say you have 5 questions, and a complete call will take approx 2 minutes allow for call-setup and call-breakdown - 15 sec should allow for 26calls per hour per channel @ 30 channels = 780 calls hour using just 1 E1 In this case you would only have, at maximum, 30 prompts playing at any one time. From what I have seen in this mailing list there are servers out there doing a lot more than that (but I do not know what the hardware is). I have built systems running at that level of traffic using the Contarra IVR platform and running on a P4-1.6GHz with 512MB on a dialogic card. Surely an * box with a decent CPU Memory can handle it no problem. Now that I have gone stated all this for the world to criticize, please let me know if I have made any critical mistakes/assumptions. Shawn P.S Contarra Envox I know, Asterisk I am learning. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Serge SchumacherSent: Tuesday, December 20, 2005 4:23 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] IVR Capacity Hi, Do you think * could play around 300 voicemenu messages simoultanously? Regs, Serge ___ --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 many oh323
Altus, Just looking over the voip-info wiki http://www.voip-info.org/wiki/Asterisk+dimensioning it seems you have hit the limit of h323. about 1/3 way down won't be able to run more than 20-25 decent quality calls Shawn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Altus Snyman Sent: Friday, October 21, 2005 1:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] how many oh323 Good day. I configured asterisk and oh323.Im using it as a sip-h323 convertor A call will come in to the asterisk box via IAX and be send to a quintum h323 gateway. in oh323 you can set the max in,out and simultaneous calls, Ive set them all to 100. Calls coming in via iax is alaw and then goes out h323 g729. It is a P4 3.3 and 1Gig of ram.Yet at 20+ calls, calls start failing. Is there someone else with a setup like this.Is the problem on the asterisk side or the quintum Please help Thanks Altus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] how many oh323
oops, typo! http://www.voip-info.org/wiki/view/Asterisk+dimensioning -Original Message- From: Shawn Porter [mailto:[EMAIL PROTECTED] Sent: Friday, October 21, 2005 10:08 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] how many oh323 Altus, Just looking over the voip-info wiki http://www.voip-info.org/wiki/Asterisk+dimensioning it seems you have hit the limit of h323. about 1/3 way down won't be able to run more than 20-25 decent quality calls Shawn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Altus Snyman Sent: Friday, October 21, 2005 1:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] how many oh323 Good day. I configured asterisk and oh323.Im using it as a sip-h323 convertor A call will come in to the asterisk box via IAX and be send to a quintum h323 gateway. in oh323 you can set the max in,out and simultaneous calls, Ive set them all to 100. Calls coming in via iax is alaw and then goes out h323 g729. It is a P4 3.3 and 1Gig of ram.Yet at 20+ calls, calls start failing. Is there someone else with a setup like this.Is the problem on the asterisk side or the quintum Please help Thanks Altus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Diaplan or iax.conf problem
Sorry to bug
everyone with such a silly thing, but I am not having the best of mental days
today.
For some
reason I am unable to make calls from my Diax to my * box (same LAN)
as you can
see by the CLI output below I am registering and authenticating but unable to
call in. Yet, I can make a call from my
sjphone to the Diax client no problem. (SJPhone is on computer A, * is computer
B, Diax on computer C)
any help
would be greatly appreciated.
IAX.CONF
[1002]
type=friend
host=dynamic
username=1002
secret=tumtum
context=iaxclients
notransfer=yes
permit=0.0.0.0/0.0.0.0
EXTENSIONS.CONF
[iaxclients]
exten = _NXXNXX,1,Dial(Zap/1/${EXTEN})
exten = _1NXXNXX,1,Dial(Zap/1/${EXTEN})
exten = _7XXX,1,dial(iax2/RECSERVER:[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])
exten = *9,1,voicemailmain(${CALLERIDNUM})
exten = *1,1,Goto(work,s,1)
exten = _1,1,Dial(IAX2/${EXTEN:1},20,tmf)
exten = _1,2,voicemail(${EXTEN:1})
exten = t,1,Hangup
CLI output
-- Registered '1002' (AUTHENTICATED) at 10.0.0.101:4569
Oct 20 16:48:08 NOTICE[11810]: chan_iax2.c:5448 socket_read: Rejected connect
attempt from 10.0.0.101
-- Registered '1002' (AUTHENTICATED) at 10.0.0.101:4569
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[Asterisk-Users] Asterisk and Dialogic
Hi all, I have a colleague who is very stuck on dialogic boards. I now the asterisk web site says it supports some dialogic boards but has anyone actually installed one and gotten it to work. I tried once to install Dialogic SR 5.1.1 with a D/41JCT-LS but gave up and ended up reformatting and going to a wildcard. I appreciate any feedback, as it will end up being my job to install and configure the server and I am not looking forward to it. Shawn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spandsp and page orientation
I have just installed spandsp-0.0.2 onto my Asterisk 1.0.9 I am using an old Intel 536EP (actually found drivers that work) BUT...all my faxes are coming in landscape mode Has anyone come across this? any fixes? Shawn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] spandsp and page orientation
samples are at http://tumtum.no-ip.com/faxes/1128432831.3.tif http://tumtum.no-ip.com/faxes/853107320051004-150908.tif Both of these were faxed from a Brother intellifax 750 through a ring-it single-line simulator into my asterisk box (through an X100P clone) both were normal 8.5X11 pages in portrait style (the map image should be 8.5 wide and 11 long) I can't take the old fax machine offline until I get this resolved. If anyone has any ideas I am open to suggestion. Shawn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Shawn Porter Sent: Tuesday, October 04, 2005 10:32 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] spandsp and page orientation I have just installed spandsp-0.0.2 onto my Asterisk 1.0.9 I am using an old Intel 536EP (actually found drivers that work) BUT...all my faxes are coming in landscape mode Has anyone come across this? any fixes? Shawn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best way to create IVR/voicemail system
Angus, This might get you started. As an IVR developer, these examples seem pretty complex for a very simple action. I am also fairly new to *, so maybe I am wrong and will figure it out as I learn more. http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Record http://www.voip-info.org/wiki-Asterisk+tips+ivr+menu Shawn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Angus Comber Sent: Friday, September 30, 2005 1:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Best way to create IVR/voicemail system Hello I want to setup a system where people can dial a number and then a system will ask them questions for which they will leave answers. Eg something like this: Answer Playback(whatisyournamemsg) Record(yourname:gsm) Playback(whatisyourheight) Record(yourheight:gsm) Playback(thankyou) Hangup Is this the best way to do this sort of thing? Do users then just access the responses by eg *98number - or does this work a little differently to voicemail? How do we retrieve the responses? Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk useable DID in Newmarket, Ontario
Does anyone know of a provider that a) allows/works using Asterisk b) provides local DIDs to the Newmarket/Aurora area? thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Diax - * - Diax delay
I found an old article in the mailing list (Jan 2005) but it seemed to just end, no resolution. Also found a bug on the digium reporting site, no real help there. I am running Asterisk 1.0.9 on Fedora Core 4 using Diax v0.9.15a one instance of Diax is on the same network as * the other is remote. we are experiencing about a 6-8second delay. I read about it possibly being timestamps on the packets, I did an ethereal dump but really have little idea what I am looking at. Any thoughts/ideas/suggestions would be appreciated. Shawn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New User - couple of dumb questions
Just installed * 1.0.9 on a FC4 (full install) I am using 2 X100P clones I do not remember what all steps I took to get everything installed. Every time I reboot, I have to modprobe zaptel modprobe wcfxo before asterisk will work. Did I miss a step somewhere? also, I have Xlite on another machine in my network, I have not been able to find (or dont know enough to recognize) Can I setup in such a way that I can call from my Xlite to an extension in my * box? I setup the extensions.conf to enable outbound calls (noise/echo are atrocious) but cannot seem to get the entry right to call an internal extension. Im sure both of these are quite simple, I have probably missed some little thing in my frustrated state. Thanks for any help. Shawn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fedora Core 4 not recognizing X100P cards
I am getting quite frustrated today, so please bear with me. I just installed Fedora Core 4 (was running RedHat 9 with a working Asterisk) now my Fedora does not appear to be recognizing my X100P (clone) at all. Hardware browser just shows them as unknown device. driver: hisax So, of course, my zaptel drivers do not work and therefore my asterisk does not work. any help would be greatly appreciated.. Shawn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to IGNORE distinctive ring
Brad, I posted a similar question on voipuser, no response yet, but I ended up making a separate extension Its not perfect, but it does technically ignore the call. [Home] exten = s,1,Wait(30) exten = s, 2, Hangup Shawn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brad Jacobs Sent: Tuesday, September 13, 2005 5:53 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How to IGNORE distinctive ring PSI System Admin-Message-ID: [EMAIL PROTECTED] Hi list members, I'm sure this question has been posted before but I am still unable to find the answer. I have a TDM 400P line card and I would like to set it up to IGNORE the distinctive ring pattern that I have for a fax machine. Many thanks Brad ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ignore incomingcall?
Is there a way to tell asterisk to ignore an incoming call? I am using distinctinveringdetection and I am only interested in answering calls on the 2nd number. Any call to the main line should just be ignored. right now I have a context set for dring2 cadence 0,0,0 exten = s, 1, wait(30 exten = s, 2, Hangup I thought that would sit until the max 4 rings (we have call-answer) then disconnect itself, but if someone picks up the phone asterisk keeps trying to respond to the call. any thoughts? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] best Fax board?
Prospective user question What is the simplest/inexpensive board to use in order to be able to receive faxes in Asterisk. I have a couple of cards I bought off E-bay think they were TX-1000 (or supposed to be anyways) but I assume I need some form of fax card though. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
