[Asterisk-Users] Throroughly confused about SetCallerID
Folks, I've been trying to handle the problem where blocked callerids appear as coming from asterisk asterisk on the email notification, and the message envelope simply doesn't say anything (does not actually play the vm-unknown message). So, following the tip provided by several previous posters, I tried putting this in my extensions.conf (the xx's are my DID, obviously: exten = xxx,1,SetCallerID(Unknown ID); exten = xxx,2,Goto(veeport_context,s,1); Okay. This works find for BLOCKED callerids. But the problem is that now all callerids are being treated as blocked, not just the really blocked ones!Help. What I want to do is: For calls that don't have blocked caller id, I want asterisk to do what it does normally: tell me the call is from phonenumber xx For calls that have a blocked callerid, I want asterisk to indicate that the callerid is unknown, both in the email and in the message envelope on the actual voicemail. Is there any way to achieve both these phenomena? I tried editing chan_sip.c to change the default_callerid there, but that causes its own set of problems. Any help would be much appreciated! Thanks, Maya __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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] Throroughly confused about SetCallerID
I forgot to state that this is only for INCOMING calls. I'm not making outgoing calls, so I really don't care what the outgoing caller id is. I'm running Asterisk 1.0.5 stable ... it's a production environment, and the users are getting really confused about the caller id strings on their incoming voicemail notifications and voicemail envelopes! Cheers, Maya --- beonice [EMAIL PROTECTED] wrote: Folks, I've been trying to handle the problem where blocked callerids appear as coming from asterisk asterisk on the email notification, and the message envelope simply doesn't say anything (does not actually play the vm-unknown message). So, following the tip provided by several previous posters, I tried putting this in my extensions.conf (the xx's are my DID, obviously: exten = xxx,1,SetCallerID(Unknown ID); exten = xxx,2,Goto(veeport_context,s,1); Okay. This works find for BLOCKED callerids. But the problem is that now all callerids are being treated as blocked, not just the really blocked ones!Help. What I want to do is: For calls that don't have blocked caller id, I want asterisk to do what it does normally: tell me the call is from phonenumber xx For calls that have a blocked callerid, I want asterisk to indicate that the callerid is unknown, both in the email and in the message envelope on the actual voicemail. Is there any way to achieve both these phenomena? I tried editing chan_sip.c to change the default_callerid there, but that causes its own set of problems. Any help would be much appreciated! Thanks, Maya __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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 __ Yahoo! Music Unlimited Access over 1 million songs. Try it free. http://music.yahoo.com/unlimited/ ___ --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] Throroughly confused about SetCallerID
--- Dan Journo [EMAIL PROTECTED] wrote:
Hi Beonice,
Just told i was dealing with the same problem.
[incoming]
exten = _!,1,GotoIf($[${CALLERID} =
unknown]?2:5)
exten = _!,2,Set(CALLERID(name)=Withheld Number)
exten = _!,3,Set(CALLERID(number)=00)
exten = _!,4,Goto(8)
exten = _!,5,GotoIf($[${CALLERID} =
asterisk]?2)
exten = _!,6,Set(CALLERID(name)=${CALLERID})
exten = _!,7,Set(CALLERID(number)=${CALLERID})
exten = _!,8,Dial(SIP/${EXTEN})
Maybe you can use that to solve your problem.
With regards to voicemail etc, i cant help you with
that.
Take care
Dan
www.sippal.com http://www.sippal.com
Thanks, Dan. I figured out that I needed to use the
GotoIf variety just as soon as I finished posting. It
does work great now. I guess it works the same on all
incoming calls, whether or not it gets sent off to
voicemail after that is irrelevant.
Thanks again!
Maya
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[Asterisk-Users] Unknown or blocked ID now shows up as asterisk
Hello, all. This is rather annoying. I know that this _used_ to work in my production system. I'm running Asterisk 1.0.5 and am using IAX exclusively. Used to be, when someone called in from a caller-id-blocked number, the notification email said: From an unknown caller, and the message envelope when I listened to it also said From an unknown caller. Something seems to have changed. I have NOT upgraded asterisk versions, but I did recompile asterisk recently. Did a full make clean; make; make install thing. Any clues? The stuff I'm seeing on previous discussions talks about editing the conf files to recognise blocked callers, but I don't think I did that before, so what's likely to have changed? I'm not seeing the forest for the trees, obviously. Any help would be appreciated! Thanks, Maya __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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] asterisk showing more than once on ps
--- [EMAIL PROTECTED] wrote: Actually this is on an enterprise system - I have never seen more than one. Just checked now and there were several calls up and only one ps. On Jul 1, 2005, at 8:33 PM, Luki wrote: Do not know why, but have noticed redhat = 1, and debian = many Not quite. RedHat Enterprise also = many at times, depending on number of concurrent calls; usually one when idle. Maybe it has something to do with kernel 2.4 vs 2.6 and how threads show up in ps. --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users What I've noticed (running the auto-start script on White-Box Enterprise Linux ... basically the same as RedHat Enterprise Linux of some vintage) is that when the auto-start executes safe_asterisk, I see multiple instances of asterisk under ps. However, if I use asterisk -r to connect, then stop and restart safe_asterisk manually, ps drops to showing exactly 1 instance of asterisk running. For what it's worth ... Maya Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.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] Unable to connect to remote asterisk
Jason, when I've seen this, it's usually meant that I was logged in as a user who didn't have the right permissions. :) Check your notes to figure out which user you gave the permission to (root, maybe?) and log in as that user, then try to connect using asterisk -r. Hope that helps, Maya --- Jason Greene [EMAIL PROTECTED] wrote: Hello, I'm trying to figure out why the asterisk service starts fine, but when i try to connect by typing asterisk -r I get: Unable to connect to remote asterisk The service is running and lists under ps -ef as: asterisk -vvvg -c any help is appreciated ___ 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 __ Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html ___ 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] Asterisk server with remote monitoring capabilities
--- Tzafrir Cohen [EMAIL PROTECTED] wrote: Hi On Thu, Jun 23, 2005 at 05:52:44PM -0700, beonice wrote: Okay, so what makes more sense: 1) a remote management card that will let me actually log in to the machine to monitor it as well as to reboot it vs. 2) a remote-accessible powerstrip that will allow me to remotely reboot the server? Linux also has a software watchdog module. Maybe it could work here without extra hardware to set up. I'm realising that sometimes the problem may simply be processes out of control, and may be something that doesn't require killing the entire machine, but just some processes. In my current setup (an ordinary PIII 1.someting GHz machine, not a server-class machine), when a process goes haywire, I lose remote access via SSH, so I drive to the colo, log in, sigh in frustration, and reboot because I'm already here, so why not?. Because you destroy any evidence of the problem. What processes are taking much CPU time? Are there any relevant log messages? Is this a case of over-swapping? (not 100% CPU usage, but rather large swap usage, CPU spends too much time at system, though the latter may be probably normal for an Asterisk server). Could you login from the console? Did you manage to move between virtual consoles? Install the package sysstat and run sar to get some stats. Consider adding a cron job to gather more relevant stats every 5 minutes or every minute. BTW: does asterisk run with real-time priority? try removing it, so at least asterisk won't hang the whole system. Though I doubt it if this would help. Some of the problems were caused by my old router ... since I replaced it, the need to drive the 40 miles each way has gone down significantly ... in fact, to pretty much zero. So I have time to contemplate my options here. :) You have an extra router there? I recall that there was a kernel patch to reboot the system upon recieving a specific ICMP packet. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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 Thanks, everyone, for all the responses and suggestions. I already do have remote access via SSH. The problem is that, occasionally, the box simply won't let me SSH in, so I have to drive over and reboot. Several of those times, it turned out to be simply the router acting up, so power-cycling the router would bring things back to life, including SSH. As I mentioned, the router has now been replaced and the new one seems a lot more stable. However, on a _few_ occasions, when I went into the colo, it turned out that the box itself was not responding to input, even on the console. At those times, the ONLY thing I could do was a hard reboot ... and yes, I'm aware of the potential hazards involved in a hard reboot. :) The frustrating thing is that since I upgraded the router, the box hasn't crashed at all ... now I'm thinking back and wondering if I imagined those occasions. :) Based on the suggestions provided by all of you, I think I'm definitely going to try to get better stats on what exactly is going on (thanks for the tip that there are production Asterisk servers with months/years uptime ... that was an eye-opener!) before I invest in any new hardware. Thanks again, everyone! Cheers, Maya Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.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 server with remote monitoring capabilities
Hello, all. I'm tired of having to drive out to the colocation facility each time my dedicated asterisk server craps out, just to press the button to do a hard reboot. (I'm running 1.05 stable at present, no telephony hardware, as this is mainly a system that receives calls, no dial-out ability is needed.) I've been looking at the fancy xeon-based systems listed on ebay for a couple of hundred dollars, in the hope that some of them have remote reboot capabilities, but most of the sellers don't mention this ability, and by the time I send out email, the item is already taken anyway. :) So, to cut the long story short, has anyone used one of these server-class machines with remote reboot capability, and does it really help? Are there any particular configurations to stay away from? The wiki doesn't talk specifically about issues regarding dual-CPU machines, but in following the chat here on asterisk-users, it seems there are definitely issues there ... can anyone elaborate? I don't want to spend money on a fancy system that turns out to be useless for my purposes. Thanks for any insight! Cheers, Maya Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.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] Asterisk server with remote monitoring capabilities
--- Michael Welter [EMAIL PROTECTED] wrote: William Boehlke wrote: Dell sells a remote management card for under $400 that enables remote reboots. I know there are others out there but have no experience with them. William Boehlke Signate -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of beonice I'm tired of having to drive out to the colocation facility each time my dedicated asterisk server craps out, just to press the button to do a hard reboot. (I'm running 1.05 stable at present, no telephony hardware, as this is mainly a system that receives calls, no dial-out ability is needed.) APC makes a power strip with a web server. Each socket has its own IP address. All you have to do to power cycle is access the IP address from your web browser and give the power cycle command. It is sooo cool. Thanks for your responses, folks. Okay, so what makes more sense: 1) a remote management card that will let me actually log in to the machine to monitor it as well as to reboot it vs. 2) a remote-accessible powerstrip that will allow me to remotely reboot the server? I'm realising that sometimes the problem may simply be processes out of control, and may be something that doesn't require killing the entire machine, but just some processes. In my current setup (an ordinary PIII 1.someting GHz machine, not a server-class machine), when a process goes haywire, I lose remote access via SSH, so I drive to the colo, log in, sigh in frustration, and reboot because I'm already here, so why not?. Some of the problems were caused by my old router ... since I replaced it, the need to drive the 40 miles each way has gone down significantly ... in fact, to pretty much zero. So I have time to contemplate my options here. :) Cheers, Maya __ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail ___ 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, SIP and NAT: Help needed!
I've been googling and talking with Libretel about my setup and the fact that incoming calls to my asterisk box through the Libretel number reach my box (I hear the greeting being played) but then don't accept DTMF. Here is a rough diagram of my setup: Asterisk | server| NAT Libretel | router | Note that there are NO SIP devices, just the Asterisk box itself. Apparently my asterisk server is trying to accept audio on port 31347 at 192.168.0.2 (which is, obviously, the internal IP address). I'm already forwarding port 5060, so I presume that's what's allowing the incoming call to hear my recorded greeting. From google, I see that I can: a) Set my externip address to the address of the NAT router b) Set nat=yes c) Somehow mangle headers??? d) change the dtmfmode in [general] Okay, so I've done all of the above: a) I've set the externip to the static IP address of the router b) I've set nat=yes in the [general] context (I'm only interested in incoming calls right now) d) I've tried each of the dtmfmode options in the [general] context As for c) I'm not sure what needs to be done there. I've also, additionally forwarded ports 1 through 4 from the router to the asterisk box (Libretel said the box is listening to audio on port 31347, so I needed at least to ensure that's open, and got a little generous there). I'm really bewildered. Especially as IAX was so straightforward to set up and get running! Thanks for any help! Cheers, Maya Attached is my current sip.conf: -- sip.conf ; ; SIP Configuration for Asterisk ; ; Syntax for specifying a SIP device in extensions.conf is ; SIP/devicename where devicename is defined in a section below. ; ; You may also use ; SIP/[EMAIL PROTECTED] to call any SIP user on the Internet ; (Don't forget to enable DNS SRV records if you want to use this) ; ; If you define a SIP proxy as a peer below, you may call ; SIP/proxyhostname/user or SIP/[EMAIL PROTECTED] ; where the proxyhostname is defined in a section below ; ; Useful CLI commands to check peers/users: ; sip show peers Show all SIP peers (including friends) ; sip show users Show all SIP users (including friends) ; sip show registry Show status of hosts we register with ; ; sip debug Show all SIP messages ; [general] ;context=default; Default context for incoming calls context=from-sip; Default context for incoming calls ; After all, we don't want any random ; incoming calls to have access to outbound ; calling - Maya, May 1, 2005 ;recordhistory=yes ; Record SIP history by default ; (see sip history / sip no history) ;realm=mydomain.tld ; Realm for digest authentication ; defaults to asterisk ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=192.168.0.2; 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 Internet ;pedantic=yes ; Enable slow, pedantic checking for Pingtel ; and multiline formatted headers for strict ; SIP compatibility (defaults to no) ;tos=184; Set IP QoS to either a keyword or numeric val ;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none ;maxexpirey=3600; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration defaultexpirey=600 ; length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all ; First disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=ilbc ; Note: codec order is respected only in
Re: [Asterisk-Users] Voicemail Passwords
--- Jeff Heath [EMAIL PROTECTED] wrote: On Tue, 2005-05-10 at 21:25, BJ Weschke wrote: voicemail.conf edit that file and issue a reload to change them. I tried this, but I still can't get access to voicemail from one of the phones. This is a test system that I setup about a month ago. Got busy and am just now getting back to it. I have 2 SIP phones and the Asterisk server. The default voicemail password is 1234 for both extensions. I changed the password for one of them and (doh!) forgot/lost it. Since this is a test system, I tried an experiment. I went into the phone where I can get access to voicemail, and I manually changed the password from 1234 to 4567. Then I issued a reload (the default passwords in voicemail.conf are 1234). Then I accessed voicemail again, and the password is 4567 not 1234. When you say manually, what do you mean? :) By editing the voicemail.conf file? See below: --- snipped --- voicemail.conf [general] format = wav49|gsm|wav serveremail = asterisk attach = yes maxmessage = 180 maxgreet = 60 skipms = 3000 maxsilence = 10 silencethreshold = 128 maxlogins = 3 [default] 4009 = 1234,Jeff 4035 = 1234,Pam Right there in the [default] context is where you would change 1234 to whatever else you want. :) Then, even if you restart, you should have the new password. You don't ever _need_ to remember the password. :) Hope that helps. Cheers, Maya Discover Yahoo! Stay in touch with email, IM, photo sharing and more. Check it out! http://discover.yahoo.com/stayintouch.html ___ 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] Getting DTMF to work with SIP?
Folks, from googling, I see that the dtmfmode parameter is not valid in the [general] context. My problem is that my overseas DID through Libretel seems to want to come into the [general] context! And, having done that, I get my welcome message, but then the DID does not accept the DTMF when I try to dial an extension! It plays the welcome message, waits, and then times out (and hangs up nicely, yes). I've actually tried setting the dtmfmode parameter in the [general] context, obviously with no success. Also, I note that when the incoming call uses SIP, a physical hangup by the caller is not recognised ... it basically goes through my timeout routine and hangs up at some point. The same context in extensions.conf can be reached through my California DID which uses IAX, and in that case I have no problem with DTMF or hangup. Any ideas on how to solve this? I've got a question open at Libretel, too, but was hoping someone here may have already dealt with this? Thanks, Maya __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ 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] Voice mail Greetings
--- Michael D Schelin [EMAIL PROTECTED] wrote:
Hi all, What would cause the greetings not to play.
The u command is
supposed to play the unavailable greeting. It
doesn't work. with this
setup. Maybe I'm missing something. The voice
prompts play well. What
do you think? Thanks
exten = 9007,1,VoicemailMain
exten = _.,2,Voicemail(u${EXTEN})
I'm guessing if you try changing the priority of the
above line to 1, you may get somewhere. Haven't tried
it, but there's not much else obvious to me. :)
exten = _.,2,AbsoluteTimeout(180)
exten = _.,4,Congestion
^^ change priority to 3
exten = _.,5,Hangup
^^ change priority to 4
I'm not sure what asterisk does when priorities skip
numbers in sequence, but it's my guess that this would
be part of the problem.
Hope that helps ...
Cheers,
Maya
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[Asterisk-Users] Forwarding incoming calls via SIP
I accidentally posted this on the biz list, but it probably belongs here. --- Folks, Has anyone successfully got a Libretel DID forwarded to a Voicepulse Connect number for incoming calls? I have signed up for an overseas DID with Libretel. My main VOIP number is a VoicepulseConnect number, which I've been happily using for incoming calls via IAX. Libretel insists on SIP (or IAX via FWD ... but that's adding another point of failure, potentially, so I don't want to go there). The choices for forwarding seem to be FWD, Generic SIP and Broadvoice. So, I selected Generic SIP, and gave my address as: [EMAIL PROTECTED] (the same userid and srvr I use to register asterisk at voicepulse connect). I see on the asterisk console that it successfully registers asterisk on the Voicepulse SIP server. When I call my Libretel DID, however, all I get is a busy signal. Libretel tells me that when they try to forward a call through to my VPC SIP-address, they get a user not known. VPC tells me they are not set up for IP-forwarding incoming calls, that calls must be dialed with a phone or PBX. So, what are my options? Has anyone successfully got Libretel and Voicepulse interacting? Any ideas on what to do next? Thanks, Maya __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Set up SIP, now I'm getting a busy tone and weird (to me)messages on the console
--- Tim Connolly [EMAIL PROTECTED] wrote: Is NAT=yes on, are you behind a firewall? Give us some connectivity details. Usually when you see maximum retries, its because you have one-way communications with the far end for some reason. Are you setting externip statically? THANKS, Tim! That nat=yes bit did the trick as far as the console messages go. Now I see a registration successful message on the console. (Is it normal for this to happen every few minutes? I haven't timed it but it looks like the registration is recurring every couple of minutes, maybe even once every minute.) Unfortunately, I still get only a busy tone when I dial the Canadian DID. When I dial the Canadian number, absolutely nothing happens on the console! Interestingly, even when I dial the local California number (I'm assuming it's still using IAX at that point), it seems to go into the unwelcome-calls extension (which I thought was sip-specific) and, according to the console, plays the congestion stuff which I never hear. Thanks again, Maya __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Set up SIP, now I'm getting a busy tone and weird (to me) messages on the console
Folks, I'm hoping someone has already run into this ... the only other complaint I've seen is here: http://lists.digium.com/pipermail/asterisk-bsd/2005-March/000640.html and that basically was a problem with the /etc/hosts ... my server is definitely described in my hosts file. I've been using asterisk with IAX and a voicepulse connect number. No problems at all receiving calls. Now, I've just purchased a DID in Canada from another provider, and their proxy only supports SIP. So, following the generic instructions I've found off the web, I set up my SIP.conf to point to voicepulse's server, and set up the other DID to point into this newly defined sip context, i.e., to uid:[EMAIL PROTECTED]/888 The problem? The remote DID, when called, simply gives me a busy signal. Also, on the asterisk console, I'm seeing these messages that don't tell me anything: May 1 18:37:09 WARNING[12065]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 103 (Critical Request) May 1 18:37:23 NOTICE[12065]: chan_sip.c:4036 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]' timed out, trying again May 1 18:37:23 DEBUG[12065]: chan_sip.c:4150 transmit_register: Scheduled a registration timeout # 5 May 1 18:37:29 WARNING[12065]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 104 (Critical Request) May 1 18:37:43 NOTICE[12065]: chan_sip.c:4036 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]' timed out, trying again May 1 18:37:43 DEBUG[12065]: chan_sip.c:4150 transmit_register: Scheduled a registration timeout # 7 It looks like the remote DID is failing to register with the voicepulse server. Any hints on what could be the problem? If it helps, here is the relevant portion of my sip.conf file. [general] ;context=default; Default context for incoming calls context=unwelcome-calls ; Default context for incoming calls ; After all, we don't want any random ; incoming calls to have access to outbound ; calling - Maya Kurup, May 1, 2005 ;recordhistory=yes ; Record SIP history by default ; (see sip history / sip no history) ;realm=mydomain.tld ; Realm for digest authentication ; defaults to asterisk ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name port=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) ... register = uid:[EMAIL PROTECTED] ; We need to allow at least incoming calls to ; accept calls via libretel, etc. ; So, let's add a context for that: [888]; For incoming calls ONLY type=user ; This device takes incoming calls username=uid ; Username on device secret=secret ; Password for device host=srvr.voicepulse.com ; This host will not ; change frequently context=allowed_context ; Inbound calls from ; this host go ; to the normal context - and I have allowed_context described in my extensions.conf, it's the same one I'm using for regular IAX incoming calls, and works fine. The context for unwelcome-calls is as follows: [unwelcome-calls] ; ; Take unknown callers that may have found ; our system, and send them to a re-order tone. ; The string _. matches any dialed sequence, so all ; calls will result in the Congestion tone application ; being called. They'll get bored and hang up eventually. ; exten = _.,1,Congestion --- Any help would be appreciated. Thanks, Maya __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Set up SIP, now I'm getting a busy tone and weird (to me)messages on the console
--- Tim Connolly [EMAIL PROTECTED] wrote: Is NAT=yes on, are you behind a firewall? Give us some connectivity details. Usually when you see maximum retries, its because you have one-way communications with the far end for some reason. Are you setting externip statically? To answer your questions, yes, I am behind a firewall. The asterisk server is the only device connected to a cheapo Netgear 4-port router/firewall. I'm not setting externip myself, so whatever the default is, it's getting used. I'm also NOT making outgoing calls, and there are no actual SIP devices attached ... I'm just trying to receive incoming calls forwarded from a different provider via SIP. Here is a complete sip.conf file ... do I need to provide anything else? sip.conf: ; ; SIP Configuration for Asterisk ; ; Syntax for specifying a SIP device in extensions.conf is ; SIP/devicename where devicename is defined in a section below. ; ; You may also use ; SIP/[EMAIL PROTECTED] to call any SIP user on the Internet ; (Don't forget to enable DNS SRV records if you want to use this) ; ; If you define a SIP proxy as a peer below, you may call ; SIP/proxyhostname/user or SIP/[EMAIL PROTECTED] ; where the proxyhostname is defined in a section below ; ; Useful CLI commands to check peers/users: ; sip show peers Show all SIP peers (including friends) ; sip show users Show all SIP users (including friends) ; sip show registry Show status of hosts we register with ; ; sip debug Show all SIP messages ; [general] ;context=default; Default context for incoming calls context=unwelcome-calls ; Default context for incoming calls ; After all, we don't want any random ; incoming calls to have access to outbound ; calling ;recordhistory=yes ; Record SIP history by default ; (see sip history / sip no history) ;realm=mydomain.tld ; Realm for digest authentication ; defaults to asterisk ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name port=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 Internet ;pedantic=yes ; Enable slow, pedantic checking for Pingtel ; and multiline formatted headers for strict ; SIP compatibility (defaults to no) ;tos=184; Set IP QoS to either a keyword or numeric val ;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none ;maxexpirey=3600; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY ;videosupport=yes ; Turn on support for SIP video ;disallow=all ; First disallow all codecs ;allow=ulaw ; Allow codecs in order of preference ;allow=ilbc ; Note: codec order is respected only in [general] ;musicclass=default ; Sets the default music on hold class for all SIP calls ; This may also be set for individual users/peers ;language=en; Default language setting for all users/peers ; This may also be set for individual users/peers ;relaxdtmf=yes ; Relax dtmf handling ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity ; when we're not on hold ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity ; when we're on hold (must be rtptimeout) ;trustrpid = no ; If Remote-Party-ID should be trusted ;progressinband=no ; If we should generate in-band ringing always ;useragent=Asterisk PBX ; Allows you to change the user agent string ;nat=no ; NAT settings ; yes = Always ignore info and assume NAT ; no = Use NAT mode only according to RFC3581 ; never = Never attempt NAT mode or RFC3581
[Asterisk-Users] VoicpulseConnect problems?
Folks, I'm having trouble with my voicepulse numbers. Over the past week, incoming calls have been very slow to be answered, but they seem fine while the call is in progress. When the caller hangs up, asterisk takes a while (over 2 minutes in some cases). This system does not make outgoing calls. Today, after rebooting my machine and rotating the log files, I have absolutely NO incoming calls being received. My cell phone dials the number, tells me it's connected, and then happily hangs up 10 to 12 seconds later, while asterisk (and the logs) show no indication at all of any incoming calls. Looking at my syslog and asterisk messages, the only thing I'm seeing over the past week that did not use to happen is this message in the asterisk logs: Apr 28 10:06:45 WARNING[4282]: Host 'gwiax-in-01.voicepulse.com' not found at line 72 But that's been happening for about the same time as the slow-down issue, and still calls _were_ being answered, albeit slowly. I'm HoSed. :) Has anyone else run into this? Got any ideas on what's up at VPConnect? Do I need to placate the rain-god or something? Any help would be appreciated! Thanks, Maya __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] VoicpulseConnect problems?
--- Tim Burt [EMAIL PROTECTED] wrote: I was having problems with voicepulse about a week or two ago... Incoming calls would fail, and one incoming call, would block all outgoing calls. Then one day, the DTMF tones stopped working. I could call into Asterisk, but I could not navigate because my tones were being ignored. I ran debug, and Asterisk was not seeing the tones. I emailed the voicepulse folks, and they fixed it the next day. It was like they knew the bug, and reset whatever needed fixing. Their reply was terse, but the fix was good. Aside from this event, voicepulse has been very good for me! Try to email their support I did open a ticket on my account at voicepulse. What you had sounds exactly like what I'm experiencing. If I don't get a fix by tomorrow, I will probably just call them and ask them when they plan to get to it. :) Thanks for the feedback! Cheers, Maya __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Has anyone used Libretel DIDs with Asterisk?
Hello, all. Google searches reveal that a couple of users have used Libretel port-of-call service via FWD using Asterisk. I don't have a FWD account. I don't have a SIP phone (no SIP setup at all, in fact). I just purchased a DID in Brazil from Libretel and discovered that they assume that I either have a SIP phone or a VOIP number through BroadVoice. My provider is actually VoicepulseConnect in California. My server here is an Asterisk box connecting to a Voicepulse DID using IAX. HELP. :( I was assuming everyone supports IAX. I guess I should have made sure first, huh? Any help would be much appreciated. Thanks, Maya __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Help using wav files for IVR
--- trixter http://www.0xdecafbad.com; [EMAIL PROTECTED] wrote: This script does mp3- wav - gsm, if you omit the first part it should work for you. requires mpg123 and sox for wav-gsm conversion and mp3-decoder to start from mp3. will process all .mp3 files in the current directory.. #!/bin/sh for i in *.mp3; do file=`echo $i | sed -e s/\.mp3$//g` mp3-decoder -w $file.wav $i normalize-audio $file.wav sox $file.wav -r 8000 decafbad-$file.gsm rm $file.wav done for ease will do wav-gsm for your pleasure.. #!/bin/sh for i in *.wav; do file=`echo $i | sed -e s/\.wav$//g` normalize-audio $i sox $i -r 8000 $file.gsm done Thanks! This script (modified to just look at the .wav files) did exactly what I hoped it would ... I now have decent-sounding gsm files that I can use. I still don't know why the original wav files were not played back properly, though. Thanks again, Maya __ Yahoo! Messenger Show us what our next emoticon should look like. Join the fun. http://www.advision.webevents.yahoo.com/emoticontest ___ 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] Help using wav files for IVR
Hi, all. We needed to add a few custom menus to the IVR, so got some recordings done professionally over the past week. Now, the files (very good sound quality) have been delivered to us, in .wav format. From googling at the digium site, I'm seeing that people in the past have successfully used wav files for IVRs. I've tried sticking them into the /var/libs/asterisk/sounds directory (after removing corresponding .gsm files where necessary). However, if I leave the extension as .wav (the format it came to me in), then I get the error message: ast_streamfile: Unable to open test-welcome (format ulaw): No such file or directory and if I just change the extension to .WAV, I get the message: ast_streamfile failed on IAX2/[EMAIL PROTECTED]:4569/5 for test-welcome Any ideas what I'm doing wrong? It looks like (from the messages above) the .WAV is the preferred format, but it's obviously not working. Help! Thanks, Maya __ Yahoo! Messenger Show us what our next emoticon should look like. Join the fun. http://www.advision.webevents.yahoo.com/emoticontest ___ 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] SetDigitTimeout question
Folks, I'm trying to slow down the speed at which Asterisk decides I've finished typing in an extension for forwarding voicemail. I've tried using exten = s,2,DigitTimeout(5); exten = s,3,ResponseTimeout(5) ; in my extensions.conf, but it still seems only about 2 seconds (or less!) before the forward functionality decides that I've had enough time to type in the next digit in the extension. Given that I occasionally have brain-freeze when I look at an extension, this isn't making me a happy camper today. Any suggestions? A _quick_ look in the app_voicemail.c didn't give any strong clues about timeouts at all. HELP! Wild bears are ... oh, never mind, it's just an asterisk. Thanks, Maya __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ 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] Asterisk@Home Install Problem
--- Scheda [EMAIL PROTECTED] wrote: Whenever I try to install [EMAIL PROTECTED], I get this error at about 43% There was an error installing rpmdb-redhat-3.4-0.20050105. This can indicate media failure, lack of disk space, and/or hardware problems. This is a fatal error and your install will be aborted. Please verify your media and try your install again. I've gotten this on 3 different cd's I've burned. To make sure it wasn't a hardware problem of some sort, I installed Debain last night. Went without a hitch, but [EMAIL PROTECTED] doesn't like it... Any ideas as to why? Scheda, You _did_ verify that you have sufficient disk space, right? I recently installed Linux on an old 6 GB hard drive and forgot to tell it to take the whole drive, so it only took what was left over after a Win2000 installation (not much was left, obviously), and I got the same error before the OS finished installing. Doesn't hurt to double check using df. Other than that, I don't know ... looks like it's not the media, unless those CDs won't let you install on a different Linux box either. Hardware problems can be a pain to isolate ... good luck! Cheers, Maya __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ 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] SetCallerID({$NEWCALLERID})
--- Steven Frazier [EMAIL PROTECTED] wrote:
I am trying to SetCallerID to a variable I have
defined. This obviously is
wrong. It actually sets the caller ID to
$NEWCALLERID. I have search
through the examples on wiki but wasn't able to find
something similar to
see what I was doing wrong. Could someone tell me
the correct way to
SetCallerID to a defined variable?
exten = 2125551212,5,SetCallerID({$NEWCALLERID})
--- snipped the rest ---
Off-hand, not having actually tested this, I'd guess
that you have the $ in the wrong place. Move it one
character to the left.
Cheers,
Maya
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RE: [Asterisk-Users] Asterisk MySQL Blobs
--- Colin Anderson [EMAIL PROTECTED] wrote: Has anyone had production experience using * w/ MySQL Blobs to store sound files? The application I am working on requires all user data resides in a database. I am currently reading/writing the files to disk via a phpagi scripts but I would love to read the blob into a variable in the dial plan, etc. It seems like a waste of resources to write and delete the file. Too bad your requirement is to have everything in the DB, 'cause you will be asking for trouble in the long run. BLOBs are probably the fastest way to kill your DB once you scale. I did an experiment a few years ago to stream faxes as BLOB's into a SQL server and performance beyond a few thousand records was to put it mildly crap. Fair enough, I was thinking the same thing myself. IMO, use filesystem for files. Use DB for DB. Put a pointer in a field to the file. Your DB will love you for it. The problem I suspect will arise is the number of inodes allowed by the file system. I don't know the exact size of the typical inode-max, but this will also presumably become an issue when the user tries to scale to really large amounts of faxes or voicemail! Would it help to split the db off to a separate server (that should reduce the CPU load on the asterisk server)? Any other alternatives? Anyone verified whether the BLOB storage solution breaks down first or the number of inodes runs out first? :) Cheers, Maya __ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web http://birthday.yahoo.com/netrospective/ ___ 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] Asterisk MySQL Blobs
--- Steven Critchfield [EMAIL PROTECTED] wrote: On Mon, 2005-03-07 at 13:30 -0800, beonice wrote: ... snip ... Would it help to split the db off to a separate server (that should reduce the CPU load on the asterisk server)? Any other alternatives? Anyone verified whether the BLOB storage solution breaks down first or the number of inodes runs out first? :) On a 40gig drive I have, there are 4.7million inodes. On an 80gig drive I have there are 9.4million inodes. Block size is (I think) 32k on these machines. At 32k I could only use 2.6 million inodes pointing to minimum sized files on the 80gig drive. If you are worried about inodes, I believe it is xfs that dynamically creates inodes as needed and won't run out. Thanks, Steven. This is exactly what I was hoping to find out. DB performance issues aside, I was worried about running out of resources on the filesystem itself if I used the present vm storage mechanism. Cheers, Maya __ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web http://birthday.yahoo.com/netrospective/ ___ 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] Question about AGI vs. FastAGI vs. straight C/DB development
Folks, I want to build a custom IVR for my setup. I've got it working (well, the bells and whistles are not there yet, but the basic stuff works) using AGI, but I'm worried about how well this will scale. I've seen references to FastAGI, and presumably this will be more efficient. Question, though: how well do either of these (AGI or FastAGI) scale if my system is handling a large number of simultaneous calls? I'm assuming that even with FastAGI, there's going to be some CPU overhead that will increase as the number of simultaneous calls increases. Would it make more sense to write a custom application in C instead, designing it to work sort-of like the built-in app_voicemail.c and others? I do know C, but is it worth the effort in terms of ROI? :) I'd like to figure this out in advance rather than after all my customers (hopefully there _will_ be customers) start yelling at me about performance issues. :) Cheers, Maya __ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web http://birthday.yahoo.com/netrospective/ ___ 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] Question about AGI vs. FastAGI vs. straight C/DB development
--- Steven Critchfield [EMAIL PROTECTED] wrote: On Mon, 2005-03-07 at 14:21 -0800, beonice wrote: --- snip --- Would it make more sense to write a custom application in C instead, designing it to work sort-of like the built-in app_voicemail.c and others? I do know C, but is it worth the effort in terms of ROI? :) I'd like to figure this out in advance rather than after all my customers (hopefully there _will_ be customers) start yelling at me about performance issues. :) You said a magic word there, customers. You need to become versed in the ins and outs of licensing. If you write it in C and distribute it, you must also offer to those you distribute asterisk your modifications and linked in code. Hmm. I'm not going to be distributing this, it'll be an in-house application, for my eyes only. :) The customers will merely be using the in-house server. So GPL or not would not be an issue. Robustness, scaleability and ROI definitely are the issues I'm interested in. Again, the licensing issues for AGI or FastAGI using any Asterisk libraries should not be an issue either. --- snipped some of Steven's response --- FastAGI offers faster initial response times than AGI as it will most likely already be running and ready to accept a new call. FastAGI should also be capable of being load balanced if that process is doing too much work to be responsive upon scaling. AGI is not capable of being run from a different machine. Okay, so it looks like AGI itself will not be worth pursuing. That's okay, at least I have a handle on what I'm doing now. So, FastAGI or straight C? I just want something that's easy to write. FastAGI will have to be wrapped around Asterisk commands, so I'm anticipating a slight overhead there, but if it's really not that much slower, it may be faster to get the IVR code up and running. Opinions? Would any of you bother writing an IVR-only application (like the app_voicemail application) in C? Why or why not? Thanks, Maya __ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web http://birthday.yahoo.com/netrospective/ ___ 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] Question with email notification
--- J P Edmund [EMAIL PROTECTED] wrote: I have been searching all over for the answer on all sources online and have come to the conclusion that it must be rudimentary or I am asking the wrong question. I cannot figure out how to configure the box to set the from address to a correct domain, as my outgoing isp will not pass mail from [EMAIL PROTECTED], as I expect it wouldn't. Any help is appreciated, even just what should I be looking for to find the correct information. This is more of a straight Linux question ... you probably need to either set hostname for your machine(s) or configure sendmail (assuming you're using the default mailer) properly to set the correct from address. You won't find the answer on the asterisk wiki or mailing-list archives, I suspect. :) Try reading the man pages for hostname and sendmail ... or do a google search. I remember having to do something like that myself a while ago when I was setting up my Linux boxen. Cheers, Maya __ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web http://birthday.yahoo.com/netrospective/ ___ 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] Weird behaviour on incoming DIDs
Folks,
I have a problem here. I have 2 DIDs, one a 415 number
and the other a 650 number. I have my extensions.conf
set up to handle both of them exactly the same way,
passing them to an internal context. When _I_ dial
either DID, I get exactly the same behaviour that I
have specified (the call is answered, and then I play
my own welcome mesage, then handle any extension
dialed).
However, when one of my friends dials in, the 415 DID
consistently works as designed, but the 650 DID
sometimes just tells him goodbye, and then hangs up on
him!
I don't know if it matters, but I am calling from the
650 area code and he's calling sometimes from the 415
area code and sometimes from 408. No, there is no
pattern as to which incoming call gets hung-up!
Here are the relevant sections of my extensions.conf:
[incoming_context]
; This is the incoming call/DID context only.
exten = 415xxx,1,Goto(internal_context,s,1)
exten = 650xxx,1,Goto(internal_context,s,1)
; munged numbers, obviously
exten = i,1,Background(invalid)
exten = #,1,Background(goodbye)
exten = #,2,Wait(2)
exten = #,3,Hangup
exten = t,1,Background(goodbye)
exten = t,2,Wait(2)
exten = t,3,Hangup
exten = h,1,Hangup
exten = 1000,1,Background(goodbye);
exten = 1000,2,Wait(2);
exten = 1000,3,Hangup
#include other_extensions.conf
And, my other_extensions.conf has:
[internal_context]
exten = #,1,Goto(incoming_context,1000,1)
exten = *,1,VoiceMailMain()
exten = *,2,Background(demo-congrats)
exten = h,1,Goto(incoming_context,h,1))
exten = i,1,Background(invalid)
exten = s,1,Answer()
exten = s,2,Background(test-welcome)
exten = t,1,Goto(incoming_context,1000,1)
exten = _[1-9]XX,1,VoiceMail(u${EXTEN})
exten = _[1-9]XX,2,Goto(incoming_context,1000,1)
Also, when this happens, I don't see Asterisk logging
the hung-up call in Master.csv. Other calls seem to be
logged fine.
Can this by any chance be caused by Master.csv getting
too large? If so, how come the same asterisk can still
handle calls coming in on the other DID with no
problems?
Thanks,
Maya
~
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Re: [Asterisk-Users] Weird behaviour on incoming DIDs
--- Michael Loftis [EMAIL PROTECTED] wrote: --On Monday, February 28, 2005 08:46 -0800 beonice [EMAIL PROTECTED] wrote: -- snipped -- When _I_ dial either DID, I get exactly the same behaviour that I have specified (the call is answered, and then I play my own welcome mesage, then handle any extension dialed). However, when one of my friends dials in, the 415 DID consistently works as designed, but the 650 DID sometimes just tells him goodbye, and then hangs up on him! ... snipped How are these DIDs being delivered? SIP, IAX, PRI? Either way I would connect to my asterisk console in verbose mode (set verbose 255) and get someone to call and then see whats different about the failed call versus the successful calls. The DIDs are delivered via IAX. Yes, I've now set verbose to extremely high, but of course, the problem isn't happening right now. :) I'll keep monitoring it ... it looks like there may be something happening when I call that 650 DID that clears up the problem, then repeated calls from outside cause the weirdness again. Of course, then he tells me it's broken, so I try calling it, which clears up whatever the problem was. At least that is what the _symptom_ is. :) Thanks, Maya __ Do you Yahoo!? Yahoo! Mail - 250MB free storage. Do more. Manage less. http://info.mail.yahoo.com/mail_250 ___ 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] Custom Menu Not Working
Try this: In the extensions_custom.conf file set: [custom-myapp] exten = s,1,SayDigits(1234) exten = s,2,Hangup() You're telling it to go to priority s, (exten = 3,1,Goto(custom-myapp,s,1) but in the custom-myapp context, you have priority 3 instead. Hope that helps. Maya --- Chris Blake [EMAIL PROTECTED] wrote: Greetings *`s, I am having what appears to be a small problem, but the frustration is erally getting to me, what am I doing wrong here ? I used AMP to set up a custom menu, so if caller presses 1 it goes to ext200, if caller presses 2 it goes to ext201 etc etc... Now I have created a third option that when the caller presses 3 it must play a sound and hang up. No rocket science yet. When adding the details in AMP for when caller dials 3, I have referenced it using 'custom-myapp,s,1', and if I go to 'extensions_additional.conf' I see the following line under the rest of menu item info that was created : exten = 3,1,Goto(custom-myapp,s,1) ; and in the extensions_custom.conf file I have [custom-myapp] exten = 3,1,SayDigits(1234) exten = 3,2,Hangup() But when you call and press option 3, it hangs up immediately. I have followed examples from the documentation, and this should be working. Any other places I can check where something is perhaps missing ? Regards -- Chris Blake Cell: 082 775 1492 Work: +27 11 782 0840 Fax : +27 11 782 0841 Mail: [EMAIL PROTECTED] When the ax entered the forest, the trees said, The handle is one of us! -- Turkish proverb ___ 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 __ Do you Yahoo!? The all-new My Yahoo! - What will yours do? http://my.yahoo.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] Custom Menu Not Working
I'll let someone else speak to the missing .conf files. If you could post your extensions.conf and extensions_additional.conf, it would be easier to help you debug this. The suggestion from timebandit _should_ have worked given your original post. Personally, for learning my way around Asterisk, I found that AMP actually got in the way. If you browse the wiki, you'll find lots of helpful examples of the config files, which is really what gets you going with Asterisk. Cheers, Maya --- Chris Blake [EMAIL PROTECTED] wrote: On Tue, 2005-02-22 at 10:48, [EMAIL PROTECTED] wrote: When adding the details in AMP for when caller dials 3, I have referenced it using 'custom-myapp,s,1', and if I go to 'extensions_additional.conf' I see the following line under the rest of menu item info that was created : exten = 3,1,Goto(custom-myapp,s,1) ; and in the extensions_custom.conf file I have [custom-myapp] exten = 3,1,SayDigits(1234) exten = 3,2,Hangup() Change to this [custom-myapp] exten = s,1,SayDigits(1234) exten = s,2,Hangup() since you send it to s,1 hth Howdy TimeBandit, thanks for replying... I tried your suggestions and it made no difference, so I went back to AMP and removed entry pointing to custom-myapp, leaving the other two that I had also created earlier, which worked. Now when I dial in the digital receptionist doesn`t do anything, and the call just disconnects. Here is the output of the CLI from the time the call is initiated, to termination : == Starting simple switch on 'Zap/4-1' -- Executing GotoIf(Zap/4-1, 0?from-pstn-reghours|s|1:) in new stack -- Executing GotoIf(Zap/4-1, 0?from-pstn-afthours|s|1:) in new stack -- Executing GotoIfTime(Zap/4-1, ||*|*?from-pstn-reghours|s|1:) in new stack -- Goto (from-pstn-reghours,s,1) -- Executing GotoIf(Zap/4-1, 0?from-pstn-reghours-nofax|s|1:2) in new stack -- Goto (from-pstn-reghours,s,2) -- Executing Answer(Zap/4-1, ) in new stack -- Executing Wait(Zap/4-1, 1) in new stack -- Executing SetVar(Zap/4-1, intype=) in new stack -- Executing Cut(Zap/4-1, intype=intype|-|1) in new stack -- Executing GotoIf(Zap/4-1, 0?7:9) in new stack -- Goto (from-pstn-reghours,s,9) -- Executing GotoIf(Zap/4-1, 0?10:12) in new stack -- Goto (from-pstn-reghours,s,12) -- Executing Goto(Zap/4-1, |s|1) in new stack -- Goto (,s,1) -- Hungup 'Zap/4-1' So I reload, and I notice that there are a few entries which seem a bit suspect : snip == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) snip == Parsing '/etc/asterisk/adsi.conf': Not found (No such file or directory) snip Reloading MGCP == Parsing '/etc/asterisk/mgcp.conf': Not found (No such file or directory) snip Based on the recommendations on the AMP site, I should not manually edit the extensions.conf, extensions_custom.conf and extensions_additional.conf files, so if removing the entries via AMP does not solve the problem, how do I go about fixing this ? Commenting out the lines inserted by AMP into the above files does not make a difference. Does the custom-myapp entry I listed earlier (with corrections of course) have to be in extensions_additional.conf as well? Regards -- Chris Blake Cell: 082 775 1492 Work: +27 11 782 0840 Fax : +27 11 782 0841 Mail: [EMAIL PROTECTED] Enzymes are things invented by biologists that explain things which otherwise require harder thinking. -- Jerome Lettvin ___ 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 __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ 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] voice recognition xml
Dean, I'd be very interested in helping with this effort. I've worked with both SGML and XML in the past (I used to work at SoftQuad in Toronto, one of the original providers of SGML and HTML tools), and have written several DTDs, both for SGML and XML. I think it would be fun to work on an XML interchange design for voice recognition ... please let me know if your contact would be interested. Cheers, Maya Kurup --- dean collins [EMAIL PROTECTED] wrote: Anyone here technical enough to design a voice recognition voice xml interchange for asterisk please email me; I've been speaking with a contact of mine that is in the voice recognition space and he is interested in 'donating' some technical support to the Asterisk community to assist with this project. This can only help benefit the Asterisk Community if this comes off. If this got up and running it would mean that Asterisk users would be able to offer voice recognition capabilities to their clients (or on their own installations) in an on-net ASP capability. Email me and I'll send you the details of the working group. Cheers, Dean ___ 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 __ Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. http://promotions.yahoo.com/new_mail ___ 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] Still asterisk startup crash plz help
It looks like it's breaking at the iax.conf file. Have you set up your iax.conf with the registration info your service provider gave you? It should look something like this: register = iaxid:[EMAIL PROTECTED] So, in my case, I have a line that says register = myid:[EMAIL PROTECTED] where myid, mypwd and the provider string were all supplied to me by voicepulse when I signed up for a DID with them. If you're using a SIP phone, you'll also have to set up sip.conf for your specific device. Also, you'll need to modify extensions.conf and voicemail.conf to suit your specific configuration. I hope that helps. Cheers, Maya --- Edward Banfa [EMAIL PROTECTED] wrote: Hi, First i would like to thank the kind people of the list who have answered my previuos mail, but i am still stuck as asterisk still crashes upon startup, i have read the install article at http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation and i have search the asterisk archives, but i still cant get asterisk to work, i have tried reinstalling asterisk but it still complains and exits with the output below, i have trie editing the config files but to no avail, plz i would appreciate any pointers to anything that can help me here. Once again thank u Edward ASTERISK OUTPUT [EMAIL PROTECTED] asterisk-1.0.5]# /usr/sbin/asterisk -c == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk 1.0.5, Copyright (C) 1999-2004 Digium. Written by Mark Spencer [EMAIL PROTECTED] = == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Started /var/log/asterisk/event_log == Manager registered action Ping == Manager registered action Events == Manager registered action Logoff == Manager registered action Hangup == Manager registered action Status == Manager registered action Setvar == Manager registered action Getvar == Manager registered action Redirect == Manager registered action Originate == Manager registered action MailboxStatus == Manager registered action Command == Manager registered action ExtensionState == Manager registered action AbsoluteTimeout == Manager registered action MailboxCount == Manager registered action ListCommands == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 1 - 2 Asterisk PBX Core Initializing Registering builtin applications: [AbsoluteTimeout] == Registered application 'AbsoluteTimeout' [Answer] == Registered application 'Answer' [BackGround] == Registered application 'BackGround' [Busy] == Registered application 'Busy' [Congestion] == Registered application 'Congestion' [DigitTimeout] == Registered application 'DigitTimeout' [Goto] == Registered application 'Goto' [GotoIf] == Registered application 'GotoIf' [GotoIfTime] == Registered application 'GotoIfTime' [Hangup] == Registered application 'Hangup' [NoOp] == Registered application 'NoOp' [Prefix] == Registered application 'Prefix' [Progress] == Registered application 'Progress' [ResetCDR] == Registered application 'ResetCDR' [ResponseTimeout] == Registered application 'ResponseTimeout' [Ringing] == Registered application 'Ringing' [SayNumber] == Registered application 'SayNumber' [SayDigits] == Registered application 'SayDigits' [SayAlpha] == Registered application 'SayAlpha' [SayPhonetic] == Registered application 'SayPhonetic' [SetAccount] == Registered application 'SetAccount' [SetAMAFlags] == Registered application 'SetAMAFlags' [SetGlobalVar] == Registered application 'SetGlobalVar' [SetLanguage] == Registered application 'SetLanguage' [SetVar] == Registered application 'SetVar' [StripMSD] == Registered application 'StripMSD' [Suffix] == Registered application 'Suffix' [Wait] == Registered application 'Wait' [WaitExten] == Registered application 'WaitExten' Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [chan_modem.so] = (Generic Voice Modem Driver) == Parsing '/etc/asterisk/modem.conf': Found == Loading modem driver chan_modem_aopen.so = (A/Open (Rockwell Chipset) ITU-2 VoiceModem Driver) == Registered channel type 'Modem' (Generic Voice Modem Channel Driver) [res_musiconhold.so] = (Music On Hold Resource) == Parsing '/etc/asterisk/musiconhold.conf': Found == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' [res_adsi.so] = (ADSI Resource) == Parsing '/etc/asterisk/adsi.conf': Found [res_features.so] = (Call Parking Resource) == Parsing '/etc/asterisk/features.conf': Found -- Registered extension context 'parkedcalls' --
Re: [Asterisk-Users] Anyone having trouble with VoicePulse Connect?
Thanks, Robert. Yes, I _finally_ figured out why I need multiple extension contexts. I'm now one happy camper. Thanks again, Maya --- Robert Hajime Lanning [EMAIL PROTECTED] wrote: quote who=beonice Robert, thank you very much for that informative write-up. Of course, I now have more questions. The first is really basic. I thought extension meant something the caller dials _after_ reaching asterisk. How come incoming DIDs have to be handled as if they are extensions? Actually the answer is better shown when looking at DID analog trunks. Analog trunks are just POTS lines. With inbound DID analog trunks, the side that provides the voltage and dialtone are reversed. The CO looks like a phone and your PBX looks like the CO. So when a call comes in, the CO picks up the phone to your PBX and dials the DID number. In this case, there is no callerID available. [DID] exten = _X.,1,Goto(PublicExtensions,8001,1) But won't this match every single number possible? Including the extensions I set up? So, for example, if I had an extension '1234', it would also be sent to PublicExtensions extension 8001 with priority 1, as would someone calling extension 8001, as well as someone calling in from any DID. I'm totally confused about the way this would behave. This is why you use a seperate context ([DID]) for DID's. DID trunks go into the DID context. Your station channels go into some other internal context, that contains your real extension numbers. The DID context will match DID's to extensions that exist in other contexts. In my example, I have PublicExtensions which is a list of extensions that are available via my main phone number and it gets included in my PrivateExtensions context which is where my internal channels drop into. Just remember that contexts seperate out extensions. Extension 8001 in context internal has nothing to do with extension 8001 in context inbound. You can definately point on towards the other, but there is no intrinsic relationship between the two. -- END OF LINE -MCP ___ 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 __ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail ___ 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] Anyone having trouble with VoicePulse Connect?
Robert, thank you very much for that informative write-up. Of course, I now have more questions. The first is really basic. I thought extension meant something the caller dials _after_ reaching asterisk. How come incoming DIDs have to be handled as if they are extensions? More questions follow: --- Robert Hajime Lanning [EMAIL PROTECTED] wrote: -- snip -- s is the start extension if there is not one already provided. When DID comes in, the channel is kindof predialed. This is with most digital calls, SIP, IAX, H323, ISDN/PRI... So the DID sort of dials its own number as an extension? I'm not sure I understand, but that seems to be the only way it would make sense to treat an incoming DID as an extension. You purchase a DID. When a call comes in it says, This is the number they were calling, you do your own matching to whatever extension you want. Okay, that sort of helps me get a mental picture of what's going on. -- snip -- Usually you do not use wildcards for DIDs. This is because people normally purchase more than one. So, you need to distinguish between phone numbers. I currently have two numbers from VoicePulse, so my extensions.conf has this: (numbers are changed to avoid crank calls) [DID] exten = _4157611829,1,Goto(PublicExtensions,8001,1) exten = _4157611763,1,Goto(PublicExtensions,8003,1) So, all inbound calls from VoicePulse goto this context. I jump from here to the extension I want the external phone number mapped to. OKAY. That example really helped. So you're basically saying that I can treat each and every incoming DID differently. That makes sense. Especially that I realised that the wild-card approach would mean that if I were to actually allocate a 10-digit extension, it would get treated exactly the same as the incoming DID-id'ed call. I tried it, and that's exactly what happened. Your example is a better solution. If you get multiple numbers (say regional numbers) and you want all of them to goto the same place, you can wildcard like this: (gets past the international numbering differences) [DID] exten = _X.,1,Goto(PublicExtensions,8001,1) But won't this match every single number possible? Including the extensions I set up? So, for example, if I had an extension '1234', it would also be sent to PublicExtensions extension 8001 with priority 1, as would someone calling extension 8001, as well as someone calling in from any DID. I'm totally confused about the way this would behave. Thanks again, Maya __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Anyone having trouble with VoicePulse Connect?
--- Martijn van Oosterhout [EMAIL PROTECTED] wrote: --- snipped my quote of what 's' is -- I guess it implies that calls coming from DIDs have digits associated with them. Correct. On ISDN lines, E1, T1 and related digital protocols, details such as CallerID, Dialled Number, CLI Presentation, etc are passed as part of the call setup, before there is any discussion of ringing. So Asterisk can go straight into the part of the script that matches. Ah. I see. However, on an analog line, you start with ringing and you still know nothing about the call. CallerID comes later and Dialled number is generally never sent at all. So you always start in s. Hope this helps, It did clear up why an incoming DID is treated as an extension. :) Thanks! Cheers, Maya __ Do you Yahoo!? Take Yahoo! Mail with you! Get it on your mobile phone. http://mobile.yahoo.com/maildemo ___ 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] Anyone having trouble with VoicePulse Connect?
Andrew, thanks for the explanation ... see more questions below. :) --- Andrew Thompson [EMAIL PROTECTED] wrote: beonice wrote: --- snipped some --- I guess the fundamental question is why is a call coming in from a DID any different? And, of course, does a call coming in _not_ from a DID (maybe via an SIP device? I don't know what the options are!) get automagically handled by the 's' handler without special mappings? I think you've confused your DID with inbound callerid. Unless you have a international DID, the exten=_NXXNXX pattern should always accept any call bound for you from that context. Even if someone calls you from an international location, voicepulse *should* always present a unique DID to you the same way every time. Okay. I think you unintentionally cleared up a visualization problem for me. I was treating ALL my extensions the same, including the inbound DID number. So, I had: [voicepulse_connect_context] exten=_NXXNXX,... exten = s,... exten = i,... exten = h,... exten = t,... exten = 1000,... exten = 1001,... etc. As you can see, all were lumped in the same context. So (bear with me here, I'm trying to reason my way through this), if I split up the internal extensions into a different context, like so: [voicepulse_connect_context] exten= 9995551212 Goto(my_internal_extensions, s, 1) ; where 9995551212 is the incoming DID exten = s,... exten = i,... exten = h,... exten = t,... [my_internal_extensions] exten = s,1,Answer exten = #,1,Hangup exten = 1000,... exten = 1001,... exten = 9995551212,... ; where 9995551212 is ; an internal extension , etc. then I'd basically be able to treat the DID value differently from what I want to do when the actual extension 9995551212 is dialed?? All calls coming in from that DID would go to the my_internal_extensions handler for 's'. After that, if the caller dials an extension, the my_internal_extensions context takes over and handles the actual extensions. Am I misunderstanding this? I'm going to go try it out. :) Now, if you want to do processing of an inbound call diferently based on it's origination number(it's callerid), you handle that afterwards. -- snipped an interesting example -- Nh. That wasn't what I was trying to achieve. I just wasn't at all understanding what that let's handle an incoming DID as an extension achieved. If my new understanding of contexts is correct, then I can see lots of possibilities. :) Thanks a bunch! Cheers, Maya __ Do you Yahoo!? The all-new My Yahoo! - Get yours free! http://my.yahoo.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] Having trouble with extensions in an include file and retrieve_extensions_from_mysql.pl
Folks, I've been running asterisk successfully using the extensions.conf and voicemail.conf. Now that I've got asterisk happily looking up MySQL tables for the VM configuration, I decided to try out the contributed script /usr/src/asterisk/contrib/scripts/retrieve_extensions_from_mysql.pl I edited the script so that its output goes to a separate extensions_from_mysql.conf file. The resulting extensions_from_mysql.conf file looks something like this: [vp_context] exten = 1000,1,Record(/tmp/rec:gsm); exten = 1000,2,Playback(/tmp/rec) ; exten = 1000,3,Background(goodbye) ; exten = 1000,4,Hangup(); I decided to #include this in my main extensions.conf, like so: [main_vp_context] exten = s,1,Answer #include extensions_from_mysql.conf exten = #,1,Background(goodbye) ; Notify caller exten = #,2,Hangup() ; Hang up exten = t,1,Hangup() ; Hang up if timeout exten = i,1,Playback(invalid) ; Play invalid ; extension if caller ; misdials an extension Basically, I expect asterisk to load the two as separate contexts, and I could swear that it used to. In fact, when I set the verbosity higher, asterisk is definitely still loading them as separate contexts. As of yesterday, though, when I have this format, asterisk won't accept incoming calls. It barfs with the message: Feb 16 21:53:14 NOTICE[4330]: chan_iax2.c:5757 socket_read: Rejected connect attempt from 66.234.228.170, request '[EMAIL PROTECTED]' does not exist The only way to get asterisk to receive calls again is to edit the included file to ensure it does not have a context line in it. So I commented out the line where the retrieve_extensions_from_mysql.pl sticks the context information into the created file. Now, it all works fine. But it's no good. What about when I want to have a sip.conf and have a list of extensions that do different things in the sip context? I really like the contributed script for its ability to add multiple context sections. Anyone see a possible reason for the problem? Do you have any ideas how to use an include file which contains multiple contexts? Or will I have to generate multiple include files, one per included context, without the context lines in these files? Thanks for any help! Cheers, Maya __ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail ___ 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] Having trouble with extensions in an include file and retrieve_extensions_from_mysql.pl
--- Andrew Thompson [EMAIL PROTECTED] wrote: --- snip --- The only thing that seems out of place to me is your #include in [main_vp_context]. It looks to me like you intend for the s, #, t, and i extensions to be in [main_vp_context]. The way you layed out this example, that's not what is happenning. I think you wanted this: Your extensions_from_mysql.conf should still look like: [vp_context] exten = 1000,1,Record(/tmp/rec:gsm); exten = 1000,2,Playback(/tmp/rec) ; exten = 1000,3,Background(goodbye) ; exten = 1000,4,Hangup(); Then, in extensions.conf: #include extensions_from_mysql.conf [main_vp_context] exten = s,1,Answer exten = #,1,Background(goodbye) ; Notify caller exten = #,2,Hangup() ; Hang up exten = t,1,Hangup() ; Hang up if timeout exten = i,1,Playback(invalid) ; Play invalid ; extension if caller ; misdials an extension include = vp_context This way, you define both contexts, and include the extensions that were defined in [vp_context] into [main_vp_context]. I don't know if this will resolve your other problem, but I believe this is the dialplan you were trying to build. Hi, Andrew. Yes, I see what you are saying. This sounds backwards, but it's actually doing what I _want_ it to do. :) From what I see in the dialplan, what asterisk does is, it loads the handlers for '#', 't' and 'i' as part of vp_context, not as part of main_vp_context. That actually happens to be as I wanted it. main_vp_context is simply a place-holder for when I am testing without the include file, and in those cases, I simply comment out my include file and voila, those handlers now handle the main_vp_context incoming cases. I know, I'm weird. :) I'm seriously concerned that my problem may be caused by some interaction between asterisk and voicepulse: at the time of writing this, even with a simple extensions.conf that has no included files at all, I cannot dial in to the asterisk box ... all calls are being rejected. Now I've spent a few minutes on (non-toll-free) hold with Voicepulse, sent them copies of my extensions.conf and iax.conf and am waiting for a response. Life really is exciting on the bleeding edge. Cheers, Maya __ Do you Yahoo!? Yahoo! Mail - now with 250MB free storage. Learn more. http://info.mail.yahoo.com/mail_250 ___ 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] Anyone having trouble with VoicePulse Connect?
Robert, Thanks for the suggestion. I also came to the same conclusion, but Asterisk wasn't showing any errors. Then I decided to go back to basics, so I backed up my extensions.conf and used the sample one provided by voicepulse voila, it worked. The culprit? Me. I'd commented out the line: exten = _NXXNXX,1,Background(welcome) ; which is apparently a critical one. I was under the impression that exten = s,1,Answer was all that was needed to make sure Asterisk answers the phone ... obviously not. :) The importance of this line probably belongs in a FAQ somewhere. I'll see if I can figure out how to update the information on the wiki somehow. By the way, what _is_ the known bug in #include files that I need to look out for (for future debugging hints ... I'm sure I'll run into in somewhere down the line!)? Thanks again, Maya --- Robert Goodyear [EMAIL PROTECTED] wrote: Look at an EXTENSIONS RELOAD and make sure the include is being parsed -- and not throwing file not found errors. I broke my include functionality last week by reMAKEing and not paying attention to a known bug in the #INCLUDE function that existed in non-HEAD versions. /rg __ Do you Yahoo!? Yahoo! Mail - now with 250MB free storage. Learn more. http://info.mail.yahoo.com/mail_250 ___ 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] Having trouble with extensions in an include file and retrieve_extensions_from_mysql.pl
I've finally figured the problem. It turns out that it had nothing to do with the include file. When I was commenting out unnecessary extensions in the main extensions.conf, I commented out the line: exten = _NXXNXX,1,Background(welcome) ; I did not realise (it's not mentioned anywhere) that this particular extension NEEDS to be handled, but as soon as I added this line back in, it fixed the problem. Now I have asterisk reading multiple contexts happily from the included file, answering the phone when it rings, and letting me leave myself voicemail. Very cool. Could someone explain why this line is so important? I'm assuming that incoming calls in other countries will have different patterns for the incoming dialed number. How does asterisk handle that? Can we add the explanation to that question into the FAQ somewhere? Thanks again, everyone! Cheers, Maya --- beonice [EMAIL PROTECTED] wrote: --- Andrew Thompson [EMAIL PROTECTED] wrote: --- snip --- The only thing that seems out of place to me is your #include in [main_vp_context]. It looks to me like you intend for the s, #, t, and i extensions to be in [main_vp_context]. The way you layed out this example, that's not what is happenning. I think you wanted this: Your extensions_from_mysql.conf should still look like: [vp_context] exten = 1000,1,Record(/tmp/rec:gsm); exten = 1000,2,Playback(/tmp/rec) ; exten = 1000,3,Background(goodbye) ; exten = 1000,4,Hangup(); Then, in extensions.conf: #include extensions_from_mysql.conf [main_vp_context] exten = s,1,Answer exten = #,1,Background(goodbye) ; Notify caller exten = #,2,Hangup() ; Hang up exten = t,1,Hangup() ; Hang up if timeout exten = i,1,Playback(invalid) ; Play invalid ; extension if caller ; misdials an extension include = vp_context This way, you define both contexts, and include the extensions that were defined in [vp_context] into [main_vp_context]. I don't know if this will resolve your other problem, but I believe this is the dialplan you were trying to build. Hi, Andrew. Yes, I see what you are saying. This sounds backwards, but it's actually doing what I _want_ it to do. :) From what I see in the dialplan, what asterisk does is, it loads the handlers for '#', 't' and 'i' as part of vp_context, not as part of main_vp_context. That actually happens to be as I wanted it. main_vp_context is simply a place-holder for when I am testing without the include file, and in those cases, I simply comment out my include file and voila, those handlers now handle the main_vp_context incoming cases. I know, I'm weird. :) I'm seriously concerned that my problem may be caused by some interaction between asterisk and voicepulse: at the time of writing this, even with a simple extensions.conf that has no included files at all, I cannot dial in to the asterisk box ... all calls are being rejected. Now I've spent a few minutes on (non-toll-free) hold with Voicepulse, sent them copies of my extensions.conf and iax.conf and am waiting for a response. Life really is exciting on the bleeding edge. Cheers, Maya __ Do you Yahoo!? Yahoo! Mail - now with 250MB free storage. Learn more. http://info.mail.yahoo.com/mail_250 ___ 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 __ Do you Yahoo!? Yahoo! Mail - now with 250MB free storage. Learn more. http://info.mail.yahoo.com/mail_250 ___ 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] Having trouble with extensions in an include file and retrieve_extensions_from_mysql.pl
--- Andrew Thompson [EMAIL PROTECTED] wrote: ... snip ... Back to your original post... As of yesterday, though, when I have this format, asterisk won't accept incoming calls. It barfs with the message: Feb 16 21:53:14 NOTICE[4330]: chan_iax2.c:5757 socket_read: Rejected connect attempt from 66.234.228.170, request '[EMAIL PROTECTED]' does not exist So, where is this voicepulse_connect_context context? Ah. When I posted, I shortened it to vp_context. This is what used to be in the main extensions.conf, then that got changed to main_vp_context while voicepulse_connect_context moved into extensions_from_mysql.conf. I should be more careful when I post. :) Cheers, Maya __ Do you Yahoo!? The all-new My Yahoo! - Get yours free! http://my.yahoo.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] Anyone having trouble with VoicePulse Connect?
--- Andrew Thompson [EMAIL PROTECTED] wrote: beonice wrote: The culprit? Me. I'd commented out the line: exten = _NXXNXX,1,Background(welcome) ; which is apparently a critical one. I was under the impression that exten = s,1,Answer Will s be traveled if a call arrives at it with a DID? The pattern you have above matches any US did that arrives into that context. If I understood the little documentation I found on 's', it's supposed to be a catchall for ALL incoming calls. That's why I assumed it would catch a DID as well. If that's not the case, it really should be updated in some meta-doc somewhere. :) So what happens if the DID is _not_ a US DID? I've seen users here from Europe and Asia as well ... does each country need its own mapping to catch the appropriate incoming DID? Now, what about the folks who are trying to call other countries, and potentially be called by other DIDs themselves? I'm assuming this sort of thing is very likely. How are others solving this issue? Really wondering what's going on ... Maya __ Do you Yahoo!? The all-new My Yahoo! - What will yours do? http://my.yahoo.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] Anyone having trouble with VoicePulse Connect?
--- Robert Webb [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Goodyear Sent: Thursday, February 17, 2005 8:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Anyone having trouble with VoicePulse Connect? On Feb 17, 2005, at 4:32 PM, Andrew Thompson wrote: beonice wrote: The culprit? Me. I'd commented out the line: exten = _NXXNXX,1,Background(welcome) ; which is apparently a critical one. I was under the impression that exten = s,1,Answer Will s be traveled if a call arrives at it with a DID? The pattern you have above matches any US did that arrives into that context. I can't get the default handler (s) to work either; I'm using VP Connect and I RTFMed a lot. I had to do a little trickery to get mine to work. Here is what I have: iax.conf [general] register = dqo75qMa71:[EMAIL PROTECTED] [voicepulse-in-01] type=user context=fromVoicePulse auth=rsa inkeys=voicepulse01 extensions.conf [fromVoicePulse] exten = _NXXNXX,1,Goto(fromPSTN,s,1) [fromPSTN] exten = s,1,LookupBlacklist exten = s,2,DigitTimeout(3) exten = s,3,ResponseTimeout(5) exten = s,4,Wait(1) exten = s,5,Background(custom/pls-wait) exten = s,102,Goto(blacklisted,s,1) Etc... For me, I have all incoming lines going to the fromPSTN context. Since I am using it at home I treat most all incoming calls the same. But hopefully this will give you an idea of the way I had to do it. Not sure why... Robert Hmm. I guess that would be one way to do it. But we still have the issue of what happens when calls come in from DIDs in other countries. How are our colleagues in Europe and Asia handling this? Are you all creating handlers that special-case your incoming DID pattern and then map it to the handler for 's' as Robert demonstrated above? I guess the fundamental question is why is a call coming in from a DID any different? And, of course, does a call coming in _not_ from a DID (maybe via an SIP device? I don't know what the options are!) get automagically handled by the 's' handler without special mappings? Any deep insight into this issue would be welcome. I hate not knowing WHY I have to do magic incantations. Cheers, Maya __ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail ___ 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] Anyone having trouble with VoicePulse Connect?
--- beonice [EMAIL PROTECTED] wrote: ... snipped ... this thread is getting long ... Hmm. I guess that would be one way to do it. But we still have the issue of what happens when calls come in from DIDs in other countries. How are our colleagues in Europe and Asia handling this? Are you all creating handlers that special-case your incoming DID pattern and then map it to the handler for 's' as Robert demonstrated above? I guess the fundamental question is why is a call coming in from a DID any different? And, of course, does a call coming in _not_ from a DID (maybe via an SIP device? I don't know what the options are!) get automagically handled by the 's' handler without special mappings? Any deep insight into this issue would be welcome. I hate not knowing WHY I have to do magic incantations. To answer my own question, at least partially, here is a quote from the Asterisk Configuration chapter in Paul Mahler's book VoIP Telephony With Asterisk: Table 1. Reserved Extension Names -- Character NameUsage - - -- s Start A call that does not have digits associated with it, for example a loopstart analog line, begins at the s extension Interesting. I don't understand it fully, but I'm sure I will if I stare at it long enough. :) I guess it implies that calls coming from DIDs have digits associated with them. Cheers, Maya __ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail ___ 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] Anyone having trouble with VoicePulse Connect?
I've been using my voicepulse connect number for over a month now, but today it simply won't connect. My partner and I each have a number, both are mapped in my iax.conf and extensions.conf files. This has been working fine. Today, either number gives this message: Feb 16 21:53:14 NOTICE[4330]: chan_iax2.c:5757 socket_read: Rejected connect attempt from 66.234.228.170, request '[EMAIL PROTECTED]' does not exist where ## is the number I dialed. Obviously chan_iax2 knows which of the numbers I dialed, but for some reason is unable to connect my machine to voicepulse? Any troubleshooting tips? This is frustrating! I _have_ asked a question via my VoicePulse Connect web account, but I don't know when they will respond. Thanks, BeOnIce. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Anyone having trouble with VoicePulse Connect?
I was doing some testing and it seems to be related to
my extensions.conf.
I have a #include extensions_from_mysql.conf that
was working fine yesterday:
[voicepulse_connect_context2]
exten = s,1,Answer
exten = s,2,NoOp,${CALLERID}
#include extensions_from_mysql.conf
and extensions_from_mysql.conf is:
[voicepulse_connect_context]
exten = 1234567890,1,VoiceMail(u1234567890);
exten = 1234567890,2,Background(goodbye)() ;
exten = 1234567890,3,Hangup() ;
exten = 1234567890,102,VoiceMail(b1234567890) ;
exten = 1234567890,103,Background(goodbye)() ;
exten = 1234567890,104,Hangup();
Today this _will_NOT_ work. Today, the only way I can
get it to work is to set the context in
extensions.conf back to
[voicepulse_connect_context]
exten = s,1,Answer
exten = s,2,NoOp,${CALLERID}
; stick my extensions in here
;
;#include extensions_from_mysql.conf
Note that I've had to comment out the #include!
Has anyone run into this weird behaviour? I am not
convinced as to where the problem is, given that the
message still seems to indicate somehow that the
interaction with voicepulse is the problem.
Thanks,
BeOnIce.
--- beonice [EMAIL PROTECTED] wrote:
I've been using my voicepulse connect number for
over
a month now, but today it simply won't connect. My
partner and I each have a number, both are mapped in
my iax.conf and extensions.conf files. This has been
working fine.
Today, either number gives this message:
Feb 16 21:53:14 NOTICE[4330]: chan_iax2.c:5757
socket_read: Rejected connect attempt from
66.234.228.170, request
'[EMAIL PROTECTED]' does not
exist
where ## is the number I dialed. Obviously
chan_iax2 knows which of the numbers I dialed, but
for
some reason is unable to connect my machine to
voicepulse?
Any troubleshooting tips? This is frustrating! I
_have_ asked a question via my VoicePulse Connect
web
account, but I don't know when they will respond.
Thanks,
BeOnIce.
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Re: [Asterisk-Users] ast_data does not patch
Lonnie, If you look at: http://www.voip-info.org/wiki-Asterisk+RealTime it says that MySQL _is_ supported. I don't know whether RealTime PostgreSQL, but I can't upgrade to RealTime anyway ... I need a stable version of asterisk, and the current stable version does not include RealTime. :( I am hoping to be able to use MySQL as well ... I've got all the config files set up correctly, I believe, but I get nowhere because: a) ast_data seems to be currently broken b) the shipped version of asterisk app_voicemail.c seems to have a bug in its select statement ... MySQL's log shows that it's connecting correctly to the database, but then it issues a command SEL instead of the command SELECT * from users WHERE ... Very frustrating. I don't really want to debug this code, but it looks like I'm going to give it a shot anyway. (No, not the ast_data code, the basic app_voicemail.c code.) Lonnie, if you have any luck getting ast_data to work, or find a contact address for rgagnon (who seems to have done the original development and presumably is still maintaining it), please forward that to me. Cheers, Maya --- [EMAIL PROTECTED] wrote: Thanks I'll look into it, but from the little that I read on RealTime, I was under the impression that it did not use MySQL or PostgreSQL which is a database feature that I was hoping to use. --Lonnie Why not just use the built-in database features to do what you want? Its called RealTime. Lots of info on it on the wiki. -Matthew - Original Message - From: [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Saturday, February 12, 2005 4:56 PM Subject: [Asterisk-Users] ast_data does not patch Hello all, I have just been trying to install the latest ast_data from: http://svn.asteriskdocs.org/res_data/ast_data/ into my cvs version of Asterisk and have found that the install patching fails. - patching file contrib/scripts/sip-friends.sql patching file contrib/scripts/iax-friends.sql patching file apps/app_voicemail.c patching file apps/app_directory.c patching file channels/chan_sip.c Hunk #2 succeeded at 621 (offset 9 lines). Hunk #3 FAILED at 1480. Hunk #4 succeeded at 1549 (offset 11 lines). Hunk #5 succeeded at 1617 (offset 18 lines). Hunk #6 succeeded at 1972 (offset 11 lines). 1 out of 6 hunks FAILED -- saving rejects to file channels/chan_sip.c.rej patching file channels/chan_iax2.c Hunk #2 succeeded at 593 with fuzz 2 (offset 13 lines). Hunk #3 FAILED at 944. Hunk #4 succeeded at 4441 (offset 57 lines). Hunk #5 FAILED at 5234. 2 out of 5 hunks FAILED -- saving rejects to file channels/chan_iax2.c.rej patching file Makefile patching file pbx.c Hunk #6 succeeded at 1390 (offset 18 lines). Hunk #8 succeeded at 1439 (offset 18 lines). Hunk #10 succeeded at 1508 (offset 18 lines). patching file asterisk.c Hunk #2 succeeded at 1922 (offset 76 lines). -- Does anyone know how to get in touch with the developer or have another viable and working option that will allow me to dynamically place my users information in a MySQL database? Thanks, Lonnie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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 __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ 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] ast_data does not patch
Heh. Good point, Kevin. I didn't realise that ast_data was also a third party add-on. :) So I submitted a bug report to digium with my gdb trace (http://bugs.digium.com/bug_view_page.php?bug_id=0003580), and markster there suggested that I should update to the latest stable asterisk from CVS. I did. And now the core asterisk can see my voicemail configuration in MySQL just fine. I must have originally retrieved a buggy version of the stable asterisk. :) Thanks, everyone, for all your help! Cheers, Maya --- Kevin P. Fleming [EMAIL PROTECTED] wrote: beonice wrote: I don't know whether RealTime PostgreSQL, but I can't upgrade to RealTime anyway ... I need a stable version of asterisk, and the current stable version does not include RealTime. :( You need a stable version of Asterisk, but you're willing to patch with an unsupported change like ast_data? Seems a little contradictory to me :-) ___ 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 __ Do you Yahoo!? All your favorites on one personal page Try My Yahoo! http://my.yahoo.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] Still stuck trying to make Asterisk read MySQL (SOLVED)
Thanks to everyone who responded. I submitted a bug report to digium (http://bugs.digium.com/bug_view_page.php?bug_id=0003580), and markster responded, suggesting that I get an updated version of stable asterisk from CVS. I did, and now it's all working fine. I must have initially downloaded a not-quite-stable stable asterisk. :) The only glitch I seem to notice is that it demands that the VM context be 'default'. :) I set it to something else, and the generated SQL code is still looking for context=default. Oh well, at least that's easy to work around. Thanks again, Maya --- Joe Dennick [EMAIL PROTECTED] wrote: I've been working with RealTime configuration from MySQL Server, and have had good results. You might check it out. You can do a search for 'realtime' on the Wiki and get some good documentation on how to set it up. I think in the extconfig.conf file, not only do you need to identify the engine (ODBC in your case), but you also need to identify the actual table you used for your Voicemail configuration. If I recall correctly, the default is a table named 'voicemail' and since you are using a different name, you need to specify the name in the extconfig.conf file so it can find it. beonice ([EMAIL PROTECTED]) wrote: I've been continuing to experiment with MySQL. I'm having absolutely no luck getting asterisk to read voicemail configuration data and mailbox configuration data from mysql tables instead of from voicemail.conf. The default Asterisk setup that reads from voicemail.conf and extensions.conf works fine. I'm using Asterisk CVS-v1-0-12/12/04-15:58:29 on a Whitebox Enterprise Linux box. I'm not using any telephony hardware or SIP phones. I've just got a voicepulse DID talking to asterisk via IAX. I've got mysql downloaded and installed and have successfully got the contributed script reading from my asterisk_vm database to set up the extensions.conf, as per the instructions at: http://www.voip-info.org/wiki-Asterisk+extensions+from+mysql Now I'm trying to get Asterisk to look up voicemail configs from the asterisk_vm database. In order to do this, I've been following the instructions at: http://www.voip-info.org/wiki-Asterisk+voicemail+database So, I've: 1) Updated the /usr/src/asterisk/apps/Makefile to have USE_MYSQL_VM_INTERFACE=1 and recompiled asterisk, with make clean; make; make install 2) Updated voicemail.conf to have the appropriate entries: dbuser=username ;; Yes I changed this to my username dbpass=password ;; Yes I changed this to my password dbhost=localhost dbname=asterisk_vm 3) Created the users table in the asterisk_vm database. +-++--+--+---+---+++ | context | mailbox| password | fullname | email | pager | options| stamp | +-++--+--+---+---+++ | default | | 1234 | Moron Tester | [EMAIL PROTECTED] | | attach=yes | 20050211131641 | +-++--+--+---+---+++ 4) Updated extensions.conf to have the following line: exten = ,1,VoiceMail(u) I tried restarting asterisk at this point, called in and tried to leave voicemail for extension (and mailbox) . Here's the message I get: *CLI Feb 11 13:21:36 WARNING[18393]: app_voicemail.c:1539 leave_voicemail: No entry in voicemail config file for '' So I dug around some more and found http://www.voip-info.org/wiki-Asterisk+res_config Decided to try these instructions as well. So: 5) I created the ast_config table as directed: Here is the data: ++++---++--+--+-+ | id | cat_metric | var_metric | commented | filename | category | var_name | var_val | ++++---++--+--+-+ | 1 | 0 | 0 | 0 | voicemail.conf | default | | | ++++---++--+--+-+ 6) I edited /etc/asterisk/configs/res_odbc.conf to contain: [mysql1] dsn = MySQL-asterisk username = myuser password = mypass pre-connect = yes [mysql1] dsn = asterisk_vm username = myuser ;; changed to my userid on mysql password = mypass ;; changed to my password on mysql pre-connect = yes [mysql2] dsn = MySQL2-asterisk username = myuser2 password = mypass2 enabled = no [ENV] VAR=VALUE 7) Inserted glue to tell asterisk where to look: ; /etc/asterisk/res_config_odbc.conf [settings] table = ast_config connection = mysql1 8) Rerouted Asterisk's config engine: ; /etc
[Asterisk-Users] Asterisk-MySQL: Not loading voicemail config from MySQL
Folks,
I'm trying to get Asterisk to load my voicemail
configuration from MySQL. I've followed the
instructions at:
http://www.voip-info.org/wiki-Asterisk+voicemail+database
I restarted Asterisk, but no luck: the voicemail.conf
does not get updated. I started with a sample
voicemail.conf that I found on the Wiki. Or was it
from Voicepulse? I can't remember. For initial
testing, I added extensions and 100 in the
[voicepulse_connect_context] with appropriate settings
in extensions.conf to direct incoming calls to those
mailboxes, and that works. I was expecting that after
I added in the db details, reloading or restarting
Asterisk would add the new extension from MySQL's
'users' table into the voicemail.conf. It doesn't.
As soon as I type (the beginning of the mailbox
and also the extension number), I get the message:
*CLI Feb 11 08:38:38 WARNING[5224]:
app_voicemail.c:1539 leave_voicemail: No entry in
voicemail config file for ''
If I add a line for into my voicemail.conf, all
works well.
Please help me understand what is going on here!
Thanks,
Maya.
--- My configuration ---
My 'users' table has 1 row only, for testing purposes:
+---++--+--+---+---+++
| context | mailbox| password |
fullname | email | pager | options
| stamp |
+---++--+--+---+---+++
| voicemail_connect_context | | 1234 |
Moron Tester | [EMAIL PROTECTED] | | attach=yes
| 00 |
+---++--+--+---+---+++
--
The appropriate settings from extensions.conf:
[voicepulse_connect_context] ; -- Should match the
context you have
; under [voicepulse-in-01] in
iax.conf
exten = 100,1,Playback(tt-monkeys)
exten = 100,2,Record(/tmp/asterisk-recording:gsm)
;exten = 100,3,Wait(2)
exten = 100,3,Playback(/tmp/asterisk-recording)
;exten = 100,5,Wait(2)
exten = ,1,Playback(transfer,skip)
exten = ,2,VoiceMail,u
exten = ,102,VoiceMail,b
exten = ,1,VoiceMail,u
--
My complete voicemail.conf looks like this:
;
; Voicemail Configuration
;
[general]
; Default formats for writing Voicemail
;format=g723sf|wav49|wav
format=wav49|gsm|wav
; Who the e-mail notification should appear to come
from
[EMAIL PROTECTED]
; Should the email contain the voicemail as an
attachment
attach=yes
; Maximum length of a voicemail message in seconds
;maxmessage=180
; Minimum length of a voicemail message in seconds
;minmessage=3
; Maximum length of greetings in seconds
;maxgreet=60
; How many miliseconds to skip forward/back when
rew/ff in message playback
skipms=3000
; How many seconds of silence before we end the
recording
maxsilence=10
; Silence threshold (what we consider silence, the
lower, the more sensitive)
silencethreshold=128
; Max number of failed login attempts
maxlogins=3
; If you need to have an external program, i.e.
/usr/bin/myapp
; called when a voicemail is left, delivered, or your
voicemailbox
; is checked, uncomment this:
;externnotify=/usr/bin/myapp
; If you need to have an external program, i.e.
/usr/bin/myapp
; called when a voicemail password is changed,
; uncomment this:
;externpass=/usr/bin/myapp
; For the directory, you can override the intro file
if you want
;directoryintro=dir-intro
; The character set for voicemail messages can be
specified here
;charset=ISO-8859-1
; The ADSI feature descriptor number to download to
;adsifdn=000F
; The ADSI security lock code
;adsisec=9BDBF7AC
; The ADSI voicemail application version number.
;adsiver=1
; Skip the [PBX]: string from the message title
pbxskip=yes
; Change the From: string
fromstring=The Asterisk PBX
;
;Change the From: string for pager messages
;pagerfromstring=The Asterisk PBX
;
; Change the email body and/or subject, variables:
; VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX,
VM_CALLERID, VM_DATE
;
; Note: The emailbody config row can be up to 512
characters due to a limitation in
; asterisk config files.
;emailsubject=New VM (${VM_MSGNUM}) - ${VM_DUR} long
in mailbox ${VM_MAILBOX} from ${VM_CALLERID}
emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you
know you were just left a ${VM_DUR} long message
(number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from
${VM_CALLERID}, on ${VM_DATE} so you might\nwant to
check it when you get a chance. Thanks!
;
; You can override the default program to send e-mail
if you wish, too
;
;mailcmd=/usr/sbin/sendmail -t
;
; Users may be located in different timezones, or may
have different
; message announcements for their introductory message
when they enter
; the voicemail system. Set the message and the
timezone each user
; hears here. Set the user into one of these zones
with the tz= attribute
; in
Re: [Asterisk-Users] Asterisk-MySQL: Not loading voicemail config fromMySQL
--- Matthew Boehm [EMAIL PROTECTED] wrote: What version of asterisk? -Matthew Asterisk CVS-v1-0-12/12/04-15:58:29 built by [EMAIL PROTECTED] on a i686 running WhiteBox Enterprise Linux By the way, I _have_ created an ast_config db and the content of my ast_config table is: ++++---++--+--+-+ | id | cat_metric | var_metric | commented | filename | category | var_name | var_val | ++++---++--+--+-+ | 1 | 0 | 0 | 0 | voicemail.conf | default | | | ++++---++--+--+-+ I've also created etc/asterisk/configs/res_odbc.conf as described in: http://voip-info.org/wiki-Asterisk+res_config My extconfig.conf says: [settings] ;uncomment to load queues.conf via the db engine. ;queues.conf = odbc voicemail.conf = odbc Unfortunately, I'm not sure what values to put in for [mysql1] dsn = MySQL-asterisk username = myuser password = mypass pre-connect = yes and for [ENV] VAR=VALUE I suspect this MAY be the problem. :) I'm unable to guess what to substitute as an appropriate dsn value and what to put into the [ENV] section. Thanks, Maya __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ 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] Asterisk-MySQL: Not loading voicemail config fromMySQL
By the way, I did fix the typo in my users table so now the context is 'voicepulse_connect_context', just like in the extensions.conf. That didn't fix the problem. Cheers, Maya --- Matthew Boehm [EMAIL PROTECTED] wrote: What version of asterisk? -Matthew - Original Message - From: beonice [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, February 11, 2005 10:36 AM Subject: [Asterisk-Users] Asterisk-MySQL: Not loading voicemail config fromMySQL Folks, I'm trying to get Asterisk to load my voicemail configuration from MySQL. I've followed the instructions at: http://www.voip-info.org/wiki-Asterisk+voicemail+database I restarted Asterisk, but no luck: the voicemail.conf does not get updated. I started with a sample voicemail.conf that I found on the Wiki. Or was it from Voicepulse? I can't remember. For initial testing, I added extensions and 100 in the [voicepulse_connect_context] with appropriate settings in extensions.conf to direct incoming calls to those mailboxes, and that works. I was expecting that after I added in the db details, reloading or restarting Asterisk would add the new extension from MySQL's 'users' table into the voicemail.conf. It doesn't. As soon as I type (the beginning of the mailbox and also the extension number), I get the message: *CLI Feb 11 08:38:38 WARNING[5224]: app_voicemail.c:1539 leave_voicemail: No entry in voicemail config file for '' If I add a line for into my voicemail.conf, all works well. Please help me understand what is going on here! Thanks, Maya. --- My configuration --- My 'users' table has 1 row only, for testing purposes: +---++--+--+ ---+---+++ | context | mailbox| password | fullname | email | pager | options | stamp | +---++--+--+ ---+---+++ | voicemail_connect_context | | 1234 | Moron Tester | [EMAIL PROTECTED] | | attach=yes | 00 | +---++--+--+ ---+---+++ -- The appropriate settings from extensions.conf: [voicepulse_connect_context] ; -- Should match the context you have ; under [voicepulse-in-01] in iax.conf exten = 100,1,Playback(tt-monkeys) exten = 100,2,Record(/tmp/asterisk-recording:gsm) ;exten = 100,3,Wait(2) exten = 100,3,Playback(/tmp/asterisk-recording) ;exten = 100,5,Wait(2) exten = ,1,Playback(transfer,skip) exten = ,2,VoiceMail,u exten = ,102,VoiceMail,b exten = ,1,VoiceMail,u -- My complete voicemail.conf looks like this: ; ; Voicemail Configuration ; [general] ; Default formats for writing Voicemail ;format=g723sf|wav49|wav format=wav49|gsm|wav ; Who the e-mail notification should appear to come from [EMAIL PROTECTED] ; Should the email contain the voicemail as an attachment attach=yes ; Maximum length of a voicemail message in seconds ;maxmessage=180 ; Minimum length of a voicemail message in seconds ;minmessage=3 ; Maximum length of greetings in seconds ;maxgreet=60 ; How many miliseconds to skip forward/back when rew/ff in message playback skipms=3000 ; How many seconds of silence before we end the recording maxsilence=10 ; Silence threshold (what we consider silence, the lower, the more sensitive) silencethreshold=128 ; Max number of failed login attempts maxlogins=3 ; If you need to have an external program, i.e. /usr/bin/myapp ; called when a voicemail is left, delivered, or your voicemailbox ; is checked, uncomment this: ;externnotify=/usr/bin/myapp ; If you need to have an external program, i.e. /usr/bin/myapp ; called when a voicemail password is changed, ; uncomment this: ;externpass=/usr/bin/myapp ; For the directory, you can override the intro file if you want ;directoryintro=dir-intro ; The character set for voicemail messages can be specified here ;charset=ISO-8859-1 ; The ADSI feature descriptor number to download to ;adsifdn=000F ; The ADSI security lock code ;adsisec=9BDBF7AC ; The ADSI voicemail application version number. ;adsiver=1 ; Skip the [PBX]: string from the message title pbxskip=yes ; Change the From: string fromstring=The Asterisk PBX ; ;Change the From: string for pager messages ;pagerfromstring=The Asterisk PBX ; ; Change the email body and/or subject, variables: ; VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX, VM_CALLERID, VM_DATE ; ; Note: The emailbody config
[Asterisk-Users] Still stuck trying to make Asterisk read MySQL
I've been continuing to experiment with MySQL. I'm having absolutely no luck getting asterisk to read voicemail configuration data and mailbox configuration data from mysql tables instead of from voicemail.conf. The default Asterisk setup that reads from voicemail.conf and extensions.conf works fine. I'm using Asterisk CVS-v1-0-12/12/04-15:58:29 on a Whitebox Enterprise Linux box. I'm not using any telephony hardware or SIP phones. I've just got a voicepulse DID talking to asterisk via IAX. I've got mysql downloaded and installed and have successfully got the contributed script reading from my asterisk_vm database to set up the extensions.conf, as per the instructions at: http://www.voip-info.org/wiki-Asterisk+extensions+from+mysql Now I'm trying to get Asterisk to look up voicemail configs from the asterisk_vm database. In order to do this, I've been following the instructions at: http://www.voip-info.org/wiki-Asterisk+voicemail+database So, I've: 1) Updated the /usr/src/asterisk/apps/Makefile to have USE_MYSQL_VM_INTERFACE=1 and recompiled asterisk, with make clean; make; make install 2) Updated voicemail.conf to have the appropriate entries: dbuser=username ;; Yes I changed this to my username dbpass=password ;; Yes I changed this to my password dbhost=localhost dbname=asterisk_vm 3) Created the users table in the asterisk_vm database. +-++--+--+---+---+++ | context | mailbox| password | fullname | email | pager | options| stamp | +-++--+--+---+---+++ | default | | 1234 | Moron Tester | [EMAIL PROTECTED] | | attach=yes | 20050211131641 | +-++--+--+---+---+++ 4) Updated extensions.conf to have the following line: exten = ,1,VoiceMail(u) I tried restarting asterisk at this point, called in and tried to leave voicemail for extension (and mailbox) . Here's the message I get: *CLI Feb 11 13:21:36 WARNING[18393]: app_voicemail.c:1539 leave_voicemail: No entry in voicemail config file for '' So I dug around some more and found http://www.voip-info.org/wiki-Asterisk+res_config Decided to try these instructions as well. So: 5) I created the ast_config table as directed: Here is the data: ++++---++--+--+-+ | id | cat_metric | var_metric | commented | filename | category | var_name | var_val | ++++---++--+--+-+ | 1 | 0 | 0 | 0 | voicemail.conf | default | | | ++++---++--+--+-+ 6) I edited /etc/asterisk/configs/res_odbc.conf to contain: [mysql1] dsn = MySQL-asterisk username = myuser password = mypass pre-connect = yes [mysql1] dsn = asterisk_vm username = myuser ;; changed to my userid on mysql password = mypass ;; changed to my password on mysql pre-connect = yes [mysql2] dsn = MySQL2-asterisk username = myuser2 password = mypass2 enabled = no [ENV] VAR=VALUE 7) Inserted glue to tell asterisk where to look: ; /etc/asterisk/res_config_odbc.conf [settings] table = ast_config connection = mysql1 8) Rerouted Asterisk's config engine: ; /etc/asterisk/extconfig.conf [settings] ;queues.conf = odbc voicemail.conf = odbc 9) I modified the sample script load_res_config.pl and ran it, it successfully updated my ast_config table, stuffing in all the settings that I'm used to seeing in voicemail.conf. 10) I restarted asterisk _again_. I get the exact same message. Feb 11 14:18:40 WARNING[18528]: app_voicemail.c:1539 leave_voicemail: No entry in voicemail config file for '' I'm totally out of ideas now. Anyone else got a clue to lend me? Thanks, Maya __ Do you Yahoo!? Take Yahoo! Mail with you! Get it on your mobile phone. http://mobile.yahoo.com/maildemo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is no one using MySQL on stable asterisk?
I'm still (doggedly) trying to get asterisk to read my voicemail configuration from MySQL. I'm using the stable release of Asterisk, from back in December, before realtime was included. If anyone has got it to work, please contact me ... I've posted details, but everyone who's responded so far has been working with the newer version that uses realtime. Unfortunately, this is going to be a production server, so I need a stable release, hence the reluctance to upgrade. Any help or pointers would be deeply appreciated. My original post is at: http://lists.digium.com/pipermail/asterisk-users/2005-February/088949.html Thanks, Maya (I _need_ to get this working. I hate being defeated by a bunch of bits and bytes!) __ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail ___ 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] Wireless LANs and Asterisk
Mike, I'm using Asterisk over a wireless LAN (Netgear something or the other). I use my desktop as the server and a laptop for remote administration and testing. So far, I haven't had any major problems that I would attribute to the wireless connectivity. Cheers, BeOnIce --- Mike Meyer [EMAIL PROTECTED] wrote: Has anyone had any experience with wireless LANs and Asterisk? We have and here are my impressions. We configured an Asterisk in the office as a precaution to see how it would work for our own retail customers. Our office is open space, about 800 sq ft. (20x40 area). We use Snom200 and Grandstream SIP phones. Using the latest Linksys wireless access point (WAP54g) and 3 wireless bridges (WET54g), I have found that it works most of the time with WPA encryption on, but will occasionally drop voice (loosing packets). With no encryption on the WLAN it seems to work without a hitch! Using a less CPU intense encryption such as 64bit WEP, things also work fine. There must be too much delay with higher rate encryption. Also we had one bridge that seemed to be a week puppy in the litter. It could only muster 60-70% signal strength. It seemed to have problems under all configurations. Finally we positioned it such that it too works well running WEP 64b. I wonder if having 3 wireless bridges in close proximity would have anything to do with the signal strength? I would doubt it though. Anyone else with other experiences to share regarding wireless LANs and encryption? I'd me interested to hear them. Thanks, Mike Meyer GenDesign Corporation ___ 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 __ Do you Yahoo!? Yahoo! Mail - 250MB free storage. Do more. Manage less. http://info.mail.yahoo.com/mail_250 ___ 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] Why does Asterisk Hangup cause server to freeze?
Hello all. I'm still investigating the cause of freezes on my asterisk server. It's a minimal installation: the only things I remember running are httpd, sshd, sendmail and asterisk itself. I have a DID from Voicepulse. No telephony cards or SIP phones ... I'm just trying to figure out the voicemail issues at this point. So a call comes in, and the caller can type a voicemail number and leave a message. So far so good. I'm using Asterisk CVS-v1-0-12/12/04-15:58:29 on a i686 running WhiteBox Enterprise Linux. Unfortunately, Asterisk has been crashing multiple times a day. When it crashes, it takes the whole machine with it ... I can't make the server accept any keystrokes or mouse events, no virtual consoles, nothing. Only a hard reboot satisfies this beast. When I reboot, there's nothing particularly interesting in /var/log/asterisk/messages either. After a LOT of reboots, I finally realised that the occasions when the machine would freeze were exactly the occasions when my dialplan indicated that the next priority should cause a Hangup. For example, I had: exten = ,1,Playback(transfer,skip) exten = ,2,VoiceMail,u exten = ,3,Hangup exten = ,102,VoiceMail,b exten = t,1,Hangup exten = i,1,Hangup And this would reliably freeze the machine after I left a voicemail in mailbox , or pressed an invalid key, or allowed Asterisk to timeout. Basically, all I wanted was for Asterisk to hangup on the caller. :) Now I've changed all instances of Hangup to SoftHangup, and the problem does not occur any more. This is unsatisfactory, though, because I DO want asterisk to hang up the connection when it sees fit. Does anyone have any ideas why Hangup isn't working for me? It seems a simple enough command, and others are obviously using it! Has anyone else encountered problems with Hangup? Some folks had suggested that the freeze could be indicative of a hardware problem, but it doesn't seem likely given that changing Hangup to SoftHangup seems to remove the problem. Any help would be much appreciated. Thanks, BeOnIce. __ Do you Yahoo!? Meet the all-new My Yahoo! - Try it today! http://my.yahoo.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 causing server to hang ... any hints?
I am trying to set up a simple Asterisk server. All
it's going to do for now is to act as my voicemail
box. I've got a DID from Voicepulse, and am using IAX
(I'll get to SIP someday when I want to circumvent the
phone company for long-distance, but for now I'd be
happy to get a trial version of Asterisk running).
So far, I've managed to set up voicemail.conf,
extensions.conf and iax.conf so that incoming calls
are answered and a voicemail can be left. Initially I
did not have a handler for the timeout case, and
Asterisk would complain about the missing handler and
occasionally would crash the server after several
messages had been left (I'm the only one testing it,
and the server has nothing else running, just
Asterisk), requiring a hard reboot.
To avoid this, I put in a handler for the timeout.
Now, Asterisk crashes the server as soon as I leave
myself a message! This is irritating. Hard reboots
everytime someone leaves me a voicemail is not going
to be something I can trust if I go on vacation
(hopefully someday soon).
Any ideas what I am doing wrong?
Here's the change I made to my extension file (this
used to crash occasionally):
[old-context]
exten = ,2,VoiceMail,u
exten = ,3,Hangup
exten = ,102,VoiceMail,b
exten = ,3,Hangup
exten = ,103,Hangup
[new-context] ;; crashes all the time!
exten = ,2,VoiceMail,u
exten = ,3,Hangup
exten = ,102,VoiceMail,b
exten = ,3,Hangup
exten = ,103,Hangup
exten = ,1,Playback(transfer,skip)
exten = ,2,Ringing
exten = ,3,Wait(2)
exten = ,4,VoiceMail,u
exten = ,104,VoiceMail,b
exten = 1001,1,Ringing
exten = 1001,2,Wait(2)
exten = 1001,3,VoicemailMain
exten = _NXXNXX,1,Background(beep)
;exten = _NXXNXX,2,SayDigits(${EXTEN})
;exten = _NXXNXX,3,Goto(testdtmf|s|1)
exten = t,1,Ringing
exten = t,2,Hangup
exten = i,1,Ringing
exten = i,2,Hangup
exten = a,1,VoicemailMain,EXTEN
exten = a,2,Hangup
I thought that using the
exten = t,2,Hangup and the
exten = i,2,Hangup
would cause Asterisk to hang up on timeout, but
obviously I misunderstood. By the way, what _does_ the
'a' handler do, anyway? I believe 'i' is for invalid
key and 't' is for timeout. I'm not sure what 'a'
does.
The last time I tried to leave myself voicemail,
here's the output I got. And yes, it crashed. I've got
Asterisk running at a level of 4 in terms of
verbosity, i.e., I start it with /usr/sbin/asterisk
-cp
-- Playing 'beep' (language 'en')
Feb 8 14:07:29 DEBUG[4195]: chan_iax2.c:5310
socket_read: Ooh, voice format changed to 4
== CDR updated on
IAX2/[EMAIL PROTECTED]:4569/1
-- Executing
Playback(IAX2/[EMAIL PROTECTED]:4569/1,
transfer|skip) in new stack
-- Playing 'transfer' (language 'en')
-- Executing
VoiceMail(IAX2/[EMAIL PROTECTED]:4569/1,
u) in new stack
Feb 8 14:07:42 DEBUG[4195]: app_voicemail.c:1381
leave_voicemail:
voicemail/voicepulse_connect_context//unavail
doesn't exist, doing what we can
-- Playing 'vm-theperson' (language 'en')
-- Playing 'digits/7' (language 'en')
-- Playing 'digits/7' (language 'en')
-- Playing 'digits/7' (language 'en')
-- Playing 'digits/7' (language 'en')
-- Playing 'vm-isunavail' (language 'en')
-- Playing 'vm-intro' (language 'en')
-- Playing 'beep' (language 'en')
-- Recording the message
Feb 8 14:07:53 DEBUG[4195]: app.c:549
ast_play_and_record: play_and_record: None,
/var/spool/asterisk/voicemail/voicepulse_connect_context//INBOX/msg0001,
'wav49|gsm|wav'
Feb 8 14:07:53 DEBUG[4195]: app.c:566
ast_play_and_record: Recording Formats: sfmts=wav49
-- x=0, open writing:
/var/spool/asterisk/voicemail/voicepulse_connect_context//INBOX/msg0001
format: wav49, 0x814ded0
-- x=1, open writing:
/var/spool/asterisk/voicemail/voicepulse_connect_context//INBOX/msg0001
format: gsm, 0x814dff0
-- x=2, open writing:
/var/spool/asterisk/voicemail/voicepulse_connect_context//INBOX/msg0001
format: wav, 0x814e100
-- User ended message by pressing #
-- Playing 'auth-thankyou' (language 'en')
localhost*CLI
At this point, the server itself seems to hang. I can
do nothing but a physical reboot of the machine.
HELP.
Thanks,
BeOnIce.
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RE: [Asterisk-Users] Asterisk causing server to hang ... any hints?
By the way, in case it matters, I don't have any
digium cards or SIP phones attached to the Asterisk
setup. Right now, it's all IAX, just receiving
incoming calls.
Cheers,
BeOnIce.
--- Chamberland-Larose, Guillaume [EMAIL PROTECTED]
wrote:
If the asterisk process is hung up you should be
able to debug it. If
the whole machine is hung up this is a totally
different issue isn't it.
If you're running linux and the machine locks up
that often, you must
have a hardware problem. Or maybe you just think the
machine is locked
up while it actually isn't.
You might want to look on google and the wiki about
deadlocks, I've seen
a lot of information around on how to debug them and
report them.
Guills
-Original Message-
From: beonice [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 08, 2005 3:41 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk causing server
to hang ...
any hints?
I am trying to set up a simple Asterisk server.
All it's
going to do for now is to act as my voicemail box.
I've got a
DID from Voicepulse, and am using IAX (I'll get to
SIP
someday when I want to circumvent the phone
company for
long-distance, but for now I'd be happy to get a
trial
version of Asterisk running).
So far, I've managed to set up voicemail.conf,
extensions.conf and iax.conf so that incoming
calls are
answered and a voicemail can be left. Initially I
did not
have a handler for the timeout case, and Asterisk
would
complain about the missing handler and
occasionally would
crash the server after several messages had been
left (I'm
the only one testing it, and the server has
nothing else
running, just Asterisk), requiring a hard reboot.
To avoid this, I put in a handler for the timeout.
Now, Asterisk crashes the server as soon as I
leave myself a
message! This is irritating. Hard reboots
everytime someone
leaves me a voicemail is not going to be something
I can
trust if I go on vacation (hopefully someday
soon).
Any ideas what I am doing wrong?
Here's the change I made to my extension file
(this used to
crash occasionally):
[old-context]
exten = ,2,VoiceMail,u
exten = ,3,Hangup
exten = ,102,VoiceMail,b
exten = ,3,Hangup
exten = ,103,Hangup
[new-context] ;; crashes all the time!
exten = ,2,VoiceMail,u
exten = ,3,Hangup
exten = ,102,VoiceMail,b
exten = ,3,Hangup
exten = ,103,Hangup
exten = ,1,Playback(transfer,skip)
exten = ,2,Ringing
exten = ,3,Wait(2)
exten = ,4,VoiceMail,u
exten = ,104,VoiceMail,b
exten = 1001,1,Ringing
exten = 1001,2,Wait(2)
exten = 1001,3,VoicemailMain
exten = _NXXNXX,1,Background(beep)
;exten = _NXXNXX,2,SayDigits(${EXTEN}) ;exten
=
_NXXNXX,3,Goto(testdtmf|s|1) exten =
t,1,Ringing exten
= t,2,Hangup exten = i,1,Ringing exten =
i,2,Hangup exten
= a,1,VoicemailMain,EXTEN exten = a,2,Hangup
I thought that using the
exten = t,2,Hangup and the
exten = i,2,Hangup
would cause Asterisk to hang up on timeout, but
obviously I
misunderstood. By the way, what _does_ the 'a'
handler do,
anyway? I believe 'i' is for invalid key and 't'
is for
timeout. I'm not sure what 'a'
does.
The last time I tried to leave myself voicemail,
here's the
output I got. And yes, it crashed. I've got
Asterisk running
at a level of 4 in terms of verbosity, i.e., I
start it with
/usr/sbin/asterisk -cp
-- Playing 'beep' (language 'en')
Feb 8 14:07:29 DEBUG[4195]: chan_iax2.c:5310
socket_read: Ooh, voice format changed to 4
== CDR updated on
IAX2/[EMAIL PROTECTED]:4569/1
-- Executing
Playback(IAX2/[EMAIL PROTECTED]:4569/1,
transfer|skip) in new stack
-- Playing 'transfer' (language 'en')
-- Executing
VoiceMail(IAX2/[EMAIL PROTECTED]:4569/1,
u) in new stack
Feb 8 14:07:42 DEBUG[4195]: app_voicemail.c:1381
leave_voicemail:
voicemail/voicepulse_connect_context//unavail
doesn't exist, doing what we can
-- Playing 'vm-theperson' (language 'en')
-- Playing 'digits/7' (language 'en')
-- Playing 'digits/7' (language 'en')
-- Playing 'digits/7' (language 'en')
-- Playing 'digits/7' (language 'en')
-- Playing 'vm-isunavail' (language 'en')
-- Playing 'vm-intro' (language 'en')
-- Playing 'beep' (language 'en')
-- Recording the message
Feb 8 14:07:53 DEBUG[4195]: app.c:549
ast_play_and_record: play_and_record: None,
/var/spool/asterisk/voicemail/voicepulse_connect_context//
INBOX/msg0001,
'wav49|gsm|wav'
Feb 8 14:07:53 DEBUG[4195]: app.c:566
ast_play_and_record: Recording Formats:
sfmts=wav49
-- x=0, open writing:
/var/spool/asterisk/voicemail/voicepulse_connect_context//
INBOX/msg0001
format: wav49, 0x814ded0
-- x=1, open writing
[Asterisk-Users] Newbie question: Can't start up asterisk
Folks, I've just successfully set up Asterisk (as part of the Asterisk Management Portal installation). When I say successfully, I mean that I have gone through all the steps detailed for the installation of AMP and not hit any snags there. I can connect to my asterisk server via ssh and can also connect via Http to the portal to change settings in AMP. Now I'm trying to start up Asterisk itself. When I type in /usr/sbin/asterisk -cp, it does a bunch of stuff, then exits with the message Ouch ... error while writing audio data: : Broken pipe From a Google search, this doesn't seem to be anything obvious. Can anyone walk me through this? I'd really like to get asterisk running. My environment: White Box Enterprise Linux, a VoicePulse Connect! DID, no telephony cards. I've tried to edit the system to use the ztdummy, but am not sure it successfully registered that change. Any and all help would be much appreciated! Thanks! BeOnIce __ Do you Yahoo!? Yahoo! Mail - Helps protect you from nasty viruses. http://promotions.yahoo.com/new_mail ___ 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] Newbie question: Can't start up asterisk
Here is the entire output until it dies: [EMAIL PROTECTED] asterisk]# /usr/sbin/asterisk -cp Set to realtime thread == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk CVS-v1-0-12/12/04-15:58:29, Copyright (C) 1999-2004 Digium. Written by Mark Spencer [EMAIL PROTECTED] = == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Started /var/log/asterisk/event_log == Manager registered action Ping == Manager registered action Events == Manager registered action Logoff == Manager registered action Hangup == Manager registered action Status == Manager registered action Setvar == Manager registered action Getvar == Manager registered action Redirect == Manager registered action Originate == Manager registered action MailboxStatus == Manager registered action Command == Manager registered action ExtensionState == Manager registered action AbsoluteTimeout == Manager registered action MailboxCount == Manager registered action ListCommands == Parsing '/etc/asterisk/manager.conf': Found Asterisk Management interface listening on port 5038 == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 1 - 2 Asterisk PBX Core Initializing Registering builtin applications: [AbsoluteTimeout] == Registered application 'AbsoluteTimeout' [Answer] == Registered application 'Answer' [BackGround] == Registered application 'BackGround' [Busy] == Registered application 'Busy' [Congestion] == Registered application 'Congestion' [DigitTimeout] == Registered application 'DigitTimeout' [Goto] == Registered application 'Goto' [GotoIf] == Registered application 'GotoIf' [GotoIfTime] == Registered application 'GotoIfTime' [Hangup] == Registered application 'Hangup' [NoOp] == Registered application 'NoOp' [Prefix] == Registered application 'Prefix' [Progress] == Registered application 'Progress' [ResetCDR] == Registered application 'ResetCDR' [ResponseTimeout] == Registered application 'ResponseTimeout' [Ringing] == Registered application 'Ringing' [SayNumber] == Registered application 'SayNumber' [SayDigits] == Registered application 'SayDigits' [SayAlpha] == Registered application 'SayAlpha' [SayPhonetic] == Registered application 'SayPhonetic' [SetAccount] == Registered application 'SetAccount' [SetAMAFlags] == Registered application 'SetAMAFlags' [SetGlobalVar] == Registered application 'SetGlobalVar' [SetLanguage] == Registered application 'SetLanguage' [SetVar] == Registered application 'SetVar' [StripMSD] == Registered application 'StripMSD' [Suffix] == Registered application 'Suffix' [Wait] == Registered application 'Wait' [WaitExten] == Registered application 'WaitExten' Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [chan_modem.so] = (Generic Voice Modem Driver) == Parsing '/etc/asterisk/modem.conf': Found == Loading modem driver chan_modem_aopen.so = (A/Open (Rockwell Chipset) ITU-2 VoiceModem Driver) == Registered channel type 'Modem' (Generic Voice Modem Channel Driver) [res_musiconhold.so] = (Music On Hold Resource) == Parsing '/etc/asterisk/musiconhold.conf': Found Junk at the beginning 49443303 Warning, flexibel rate not heavily tested! == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' [res_adsi.so] = (ADSI Resource) == Parsing '/etc/asterisk/adsi.conf': Not found (No such file or directory) [res_features.so] = (Call Parking Resource) == Parsing '/etc/asterisk/features.conf': Not found (No such file or directory) == Parsing '/etc/asterisk/parking.conf': Found -- Registered extension context 'default' -- Added extension '700' priority 1 to default == Registered application 'ParkedCall' == Registered application 'Park' == Manager registered action ParkedCalls [res_crypto.so] = (Cryptographic Digital Signatures) -- Loaded PUBLIC key 'iaxtel' -- Loaded PUBLIC key 'freeworlddialup' -- Loaded PUBLIC key 'voicepulse01' [res_indications.so] = (Indications Configuration) == Parsing '/etc/asterisk/indications.conf': Found -- Registered indication country 'us' -- Registered indication country 'au' -- Registered indication country 'fr' -- Registered indication country 'de' -- Registered indication country 'nl' -- Registered indication country 'uk' -- Registered indication country 'fi' -- Registered indication country 'no' -- Registered indication country 'br' -- Registered indication country 'za' -- Registered indication country 'it' -- Setting default indication country to 'us' == Registered application 'Playtones' == Registered application 'StopPlaytones' [res_monitor.so] =
Re: [Asterisk-Users] Newbie question: Can't start up asterisk
Denis wrote: --- Denis Galvão - iSolve [EMAIL PROTECTED] wrote: Em Ter 18 Jan 2005 20:43, Matt Riddell escreveu: beonice wrote: Ouch ... error while writing audio data: : Broken pipe What are the messages before this? Matt I think that is something related to mpg123... -- D e n i s G a l v ã o iSolve - Solve Is Our Business Av. Candido de Abreu, 526 1206B CEP: 80530-000 - Curitiba - PR +55 41 252-2977 http://www.isolve.com.br That _seems_ to be a possibility. But I'm not really sure. I made sure that there is a symbolic link in /usr/bin to mpg123 ... the actual version is in /usr/local/bin. Thanks. By the way, I accidentally created a new post with the details of the output instead of responding to Matt's question right here ... but here is the output again: [EMAIL PROTECTED] asterisk]# /usr/sbin/asterisk -cp Set to realtime thread == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) Asterisk CVS-v1-0-12/12/04-15:58:29, Copyright (C) 1999-2004 Digium. Written by Mark Spencer [EMAIL PROTECTED] = == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Started /var/log/asterisk/event_log == Manager registered action Ping == Manager registered action Events == Manager registered action Logoff == Manager registered action Hangup == Manager registered action Status == Manager registered action Setvar == Manager registered action Getvar == Manager registered action Redirect == Manager registered action Originate == Manager registered action MailboxStatus == Manager registered action Command == Manager registered action ExtensionState == Manager registered action AbsoluteTimeout == Manager registered action MailboxCount == Manager registered action ListCommands == Parsing '/etc/asterisk/manager.conf': Found Asterisk Management interface listening on port 5038 == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 1 - 2 Asterisk PBX Core Initializing Registering builtin applications: [AbsoluteTimeout] == Registered application 'AbsoluteTimeout' [Answer] == Registered application 'Answer' [BackGround] == Registered application 'BackGround' [Busy] == Registered application 'Busy' [Congestion] == Registered application 'Congestion' [DigitTimeout] == Registered application 'DigitTimeout' [Goto] == Registered application 'Goto' [GotoIf] == Registered application 'GotoIf' [GotoIfTime] == Registered application 'GotoIfTime' [Hangup] == Registered application 'Hangup' [NoOp] == Registered application 'NoOp' [Prefix] == Registered application 'Prefix' [Progress] == Registered application 'Progress' [ResetCDR] == Registered application 'ResetCDR' [ResponseTimeout] == Registered application 'ResponseTimeout' [Ringing] == Registered application 'Ringing' [SayNumber] == Registered application 'SayNumber' [SayDigits] == Registered application 'SayDigits' [SayAlpha] == Registered application 'SayAlpha' [SayPhonetic] == Registered application 'SayPhonetic' [SetAccount] == Registered application 'SetAccount' [SetAMAFlags] == Registered application 'SetAMAFlags' [SetGlobalVar] == Registered application 'SetGlobalVar' [SetLanguage] == Registered application 'SetLanguage' [SetVar] == Registered application 'SetVar' [StripMSD] == Registered application 'StripMSD' [Suffix] == Registered application 'Suffix' [Wait] == Registered application 'Wait' [WaitExten] == Registered application 'WaitExten' Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [chan_modem.so] = (Generic Voice Modem Driver) == Parsing '/etc/asterisk/modem.conf': Found == Loading modem driver chan_modem_aopen.so = (A/Open (Rockwell Chipset) ITU-2 VoiceModem Driver) == Registered channel type 'Modem' (Generic Voice Modem Channel Driver) [res_musiconhold.so] = (Music On Hold Resource) == Parsing '/etc/asterisk/musiconhold.conf': Found Junk at the beginning 49443303 Warning, flexibel rate not heavily tested! == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' [res_adsi.so] = (ADSI Resource) == Parsing '/etc/asterisk/adsi.conf': Not found (No such file or directory) [res_features.so] = (Call Parking Resource) == Parsing '/etc/asterisk/features.conf': Not found (No such file or directory) == Parsing '/etc/asterisk/parking.conf': Found -- Registered extension context 'default' -- Added extension '700' priority 1 to default == Registered application 'ParkedCall' == Registered application 'Park' == Manager registered action ParkedCalls [res_crypto.so] = (Cryptographic Digital Signatures) -- Loaded PUBLIC key 'iaxtel' -- Loaded PUBLIC key 'freeworlddialup' -- Loaded
RE: [Asterisk-Users] Newbie question: Can't start up asterisk
--- Colin Anderson [EMAIL PROTECTED] wrote: Ouch ... error while writing audio data: : Broken pipe Did you run make samples from /usr/src/asterisk? Hmm. I thought I had, but I've attempted this Asterisk installation so many times that I seem to have skipped it on this incarnation. I just did the make samples, and now I get this output (note that it still dies with the same message, but now there is more context!): Set to realtime thread == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk CVS-v1-0-12/12/04-15:58:29, Copyright (C) 1999-2004 Digium. Written by Mark Spencer [EMAIL PROTECTED] = == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Started /var/log/asterisk/event_log == Manager registered action Ping == Manager registered action Events == Manager registered action Logoff == Manager registered action Hangup == Manager registered action Status == Manager registered action Setvar == Manager registered action Getvar == Manager registered action Redirect == Manager registered action Originate == Manager registered action MailboxStatus == Manager registered action Command == Manager registered action ExtensionState == Manager registered action AbsoluteTimeout == Manager registered action MailboxCount == Manager registered action ListCommands == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 1 - 2 Asterisk PBX Core Initializing Registering builtin applications: [AbsoluteTimeout] == Registered application 'AbsoluteTimeout ' [Answer] == Registered application 'Answer' [BackGround] == Registered application 'BackGround' [Busy] == Registered application 'Busy' [Congestion] == Registered application 'Congestion' [DigitTimeout] == Registered application 'DigitTimeout' [Goto] == Registered application 'Goto' [GotoIf] == Registered application 'GotoIf' [GotoIfTime] == Registered application 'GotoIfTime' [Hangup] == Registered application 'Hangup' [NoOp] == Registered application 'NoOp' [Prefix] == Registered application 'Prefix' [Progress] == Registered application 'Progress' [ResetCDR] == Registered application 'ResetCDR' [ResponseTimeout] == Registered application 'ResponseTimeout ' [Ringing] == Registered application 'Ringing' [SayNumber] == Registered application 'SayNumber' [SayDigits] == Registered application 'SayDigits' [SayAlpha]%) == Registered application 'SayAlpha' [SayPhonetic] == Registered application 'SayPhonetic' [SetAccount] == Registered application 'SetAccount' [SetAMAFlags] == Registered application 'SetAMAFlags' [SetGlobalVar] == Registered application 'SetGlobalVar' [SetLanguage] == Registered application 'SetLanguage' [SetVar] == Registered application 'SetVar' [StripMSD] == Registered application 'StripMSD' [Suffix] == Registered application 'Suffix' [Wait] == Registered application 'Wait' [WaitExten] == Registered application 'WaitExten' Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [chan_modem.so] = (Generic Voice Mo dem Driver) == Parsing '/etc/asterisk/modem.conf': Found == Loading modem driver chan_modem_aopen.so = (A/Ope n (Rockwell Chipset) ITU-2 VoiceModem Driver) == Registered channel type 'Modem' (Generic Voice Modem Chan nel Driver) [res_musiconhold.so] = (Music On Ho ld Resource) == Parsing '/etc/asterisk/musiconhold.conf': Found == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold ' == Registered application 'SetMusicOnHold' [res_adsi.so] = (ADSI Resource ) == Parsing '/etc/asterisk/adsi.conf': Found [res_features.so] = (Call Parking R esource) == Parsing '/etc/asterisk/features.conf': Found -- Registered extension context 'parkedcalls' -- Added extension '700' priority 1 to parkedcalls == Registered application 'ParkedCall' == Registered application 'Park' == Manager registered action ParkedCalls [res_crypto.so] = (Cryptographic Di gital Signatures) -- Loaded PUBLIC key 'iaxtel' -- Loaded PUBLIC key 'freeworlddialup' -- Loaded PUBLIC key 'voicepulse01' [res_indications.so] = (Indications Configuration) == Parsing '/etc/asterisk/indications.conf': Found -- Registered indication country 'cl' -- Registered indication country 'tw' -- Registered indication country 'us' -- Registered indication country 'au' -- Registered indication country 'fr' -- Registered indication country 'de' -- Registered indication country 'nl' -- Registered indication country 'uk' -- Registered indication country 'fi' -- Registered indication country 'no' -- Registered indication country 'br' -- Registered indication country 'za' -- Registered indication country 'it'
Re: [Asterisk-Users] Newbie question: Can't start up asterisk
--- Matt Riddell [EMAIL PROTECTED] wrote: Colin Anderson wrote: Ouch ... error while writing audio data: : Broken pipe Did you run make samples from /usr/src/asterisk? Yeah he has config files. What is the response you get running mpg123? I.E. what are the first four lines? I.E. what version? :) -- Cheers, Matt Riddell Well, I just tried running mpg123. Here is the version information: High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3. Version 0.59r (1999/Jun/15). Written and copyrights by Michael Hipp. Uses code from various people. See 'README' for more! THIS SOFTWARE COMES WITH ABSOLUTELY NO WARRANTY! USE AT YOUR OWN RISK! __ Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. http://promotions.yahoo.com/new_mail ___ 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] Newbie question: Can't start up asterisk
--- Denis Galvão - iSolve [EMAIL PROTECTED] wrote: Did you install mpg123 from source!? Or you're using a distro native version!? You have to get the mpg123 from its website and then get it compiled to your suystem. I believe I have a freshly compiled version. I've got version 0.59r, is what running mpg123 tells me. Cheers, BeOnIce __ Do you Yahoo!? Yahoo! Mail - now with 250MB free storage. Learn more. http://info.mail.yahoo.com/mail_250 ___ 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] Newbie question: Can't start up asterisk
--- Brian Dingman [EMAIL PROTECTED] wrote: Put /usr/local/lib in /etc/ld.so.conf then run ldconfig. Hmm. I don't understand what that did, but THANKS, Brian. It seems to have at least got Asterisk to successfully run. Now I can have fun with configuring it! Thanks a bunch! BeOnIce. __ Do you Yahoo!? Yahoo! Mail - Easier than ever with enhanced search. Learn more. http://info.mail.yahoo.com/mail_250 ___ 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] Newbie question: Can't start up asterisk
Thanks, Brian. I'm now a happy camper! Cheers, BeOnIce --- Brian Dingman [EMAIL PROTECTED] wrote: It has to do with spandsp and receiving incoming faxes. This should probably be updated in the documentation. On Tue, 18 Jan 2005 17:09:04 -0800 (PST), beonice [EMAIL PROTECTED] wrote: --- Brian Dingman [EMAIL PROTECTED] wrote: Put /usr/local/lib in /etc/ld.so.conf then run ldconfig. Hmm. I don't understand what that did, but THANKS, Brian. It seems to have at least got Asterisk to successfully run. Now I can have fun with configuring it! Thanks a bunch! BeOnIce. ___ 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 __ Do you Yahoo!? The all-new My Yahoo! - Get yours free! http://my.yahoo.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
