Re: [Asterisk-Users] MP3 streams for MOH: idea
allow this to happen. Do you know of any tools that convert ASF to mp3? mplayer/mencoder understands ASF, mp3 and lots of other formats. -- http://graphics.cs.uni-sb.de/VoIP/ pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] OT: Creating documentation using a web interface
Why not create the docs in a word processor and save it out as HTML? Virtually all the word processor programs will do that now. OpenOffice (available free for Linux, MS Windows, and possibly Mac) is nice for this, or you could use KWord (also free), or software for that other platform (if you don't mind writing the same document 5 times because it crashed in the middle of a save or right after you wrote a whole page more). If you do not need a collaborative ability in the software this is a good way to go. Tim On Saturday 06 September 2003 08:42 am, Leif Madsen wrote: Hello all, I would like to document some things I am doing with asterisk, but would prefer to do this from a web interface. I am unfamiliar with any software that allows you to create online documentation from a web interface. Ideally I will be able to create documentation online from a browser, which then when saved, is immediately ready to be read online. Perhaps I can setup different authors who are also allowed to create documentation, or have a section where users to the site can create their own documentation and submit it for inclusion. A section to submit documentation edits would be nice, as well as maybe a history timeline or something like that? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FYI: Perl module for Cisco 79x0 phones...
hi, not sure is anyone is aware, but I found a perl module that makes interfacing with a cisco 79x0 phone a breeze http://www.cpan.org/modules/by-module/Cisco/ Though it might be of some use... Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialplan question
Title: Message
Fredrik,
Your
dialplan looks correct, however you disallow 112, the emergency
number!
Does
it fail for local or interlocal calls?
I
use:
[dutchdial];;emergency (112) and other 11x
numbers;exten = _1XX,1,Dial(${ISDN}:${EXTEN})exten =
_1XX,2,Congestionexten = _01XX,1,Dial(${ISDN}:${EXTEN:1})exten =
_01XX,2,Congestion;;0900 0800 numbers;exten =
_00[89]00.,1,SetCIDNum(0206408219)exten =
_00[89]00.,2,Dial(${ISDN}:${EXTEN:1})exten =
_00[89]00.,3,Congestion;;International;exten =
_000.,1,Dial(${ISDN}:${EXTEN:1})exten =
_000.,2,Congestion;;Local (7 digits, add area code);exten =
_0XXX,1,SetCIDNum(0206408219)exten =
_0XXX,2,Dial(${ISDN}:020${EXTEN:1})exten =
_0XXX,3,Congestion;;Interlocal, 10 digits;exten =
_0XX,1,SetCIDNum(0206408219)exten =
_0XX,2,Dial(${ISDN}:${EXTEN:1})exten =
_0XX,3,Congestion
And it
works fine
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of fredrik
chabotSent: zaterdag 6 september 2003 18:25To:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Dialplan
questionHi,Dialplan QuestionI'm in
holland and I have:[naarbuiten]ignorepat = 0;
interlocaalexten =
_00[1-9],1,Dial(Modem/g1:${EXTEN})
exten = _00[1-9],2,Congestion; locaalexten =
_0[1-9]XX,1,Dial(Modem/g1:${EXTEN})
exten = _0[1-9]XX,2,CongestionAnd sometimes I can get
out, most of the time however I get a busy signal halfway throu the
number.It works more often if I change Early Dial:
No Yes (use "Yes" only if proxy supports 484
response)to No. In the Budgetone 100
phone.regards,fredrik chabot---*CLI
show dialplan [ Context 'default' created by 'pbx_config' ]
Include =
'demo'
[pbx_config][ Context 'demo' created by 'pbx_config' ] '#'
= 1.
Playback(demo-thanks)
[pbx_config]
2.
Hangup()
[pbx_config] '100'
= 1.
Dial(SIP/100)
[pbx_config] '101'
= 1.
Dial(SIP/101)
[pbx_config] '190'
= 1.
Dial(Modem/g1:006400)
[pbx_config] '8500'
= 1.
VoicemailMain()
[pbx_config]
2.
Goto(s|6)
[pbx_config] 'i'
= 1.
Playback(invalid)
[pbx_config] 's'
= 1.
Wait(1)
[pbx_config]
2.
Answer()
[pbx_config]
3.
DigitTimeout(5)
[pbx_config]
4.
ResponseTimeout(10)
[pbx_config]
5.
BackGround(demo-congrats)
[pbx_config]
6.
BackGround(demo-instruct)
[pbx_config] 't'
= 1.
Goto(#|1)
[pbx_config] Include
=
'naarbuiten'
[pbx_config][ Context 'naarbuiten' created by 'pbx_config' ]
'_00[1-9]' = 1.
Dial(Modem/g1:${EXTEN})
[pbx_config]
2.
Congestion()
[pbx_config] '_0[1-9]XX' = 1.
Dial(Modem/g1:${EXTEN})
[pbx_config]
2.
Congestion()
[pbx_config] Ignore pattern =
'0'
[pbx_config][ Context 'vanbuiten' created by 'pbx_config' ]
's' = 1.
Wait(1)
[pbx_config]
2.
Answer()
[pbx_config]
3.
DigitTimeout(5)
[pbx_config]
4.
ResponseTimeout(10)
[pbx_config]
5.
Playback(tt-weasels)
[pbx_config]
6.
Dial(SIP/100|4)
[pbx_config]
7.
Dial(SIP/100SIP/101|10)
[pbx_config]
8. Dial(SIP/100SIP/101Modem/g1:0064000)
[pbx_config]___
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[Asterisk-Users] Asterisk Application Documentation
I've spent some time on the Wiki adding documentation on all asterisk applications from the cli 'show application ' commands. I've also added some cross references and pointers. http://www.voip-info.org/tiki-index.php?page=Asterisk If you find this useful, please go there and help us build a reference database. A Wiki is a wonderful collaboration and documentation tool - anyone can add or edit information. At this point, you'll find the documentation on all commands you can use in extensions.conf. After reading them all, I would humbly suggest that some of the documentation is not for users who are not used to read the source... Additions and clarifications are welcome, maybe someone later can add new versions back into the source. Anyway, it's a starting point. If this copying of mine is a violation of any copyright, just mail me and I'll ask the web site owner to remove those parts. I haven't copied anything from Digiums Asterisk Handbook, even though some of that text would clarify. Since it's only in PDF on line, it's hard to point to the proper chapter in the PDF. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Creating documentation using a web interface
Do-you knox PhpNuke : http://phpnuke.org/ or SPIP : http://www.spip.net/en I use SPIP for a bunch of web sites and I guess this could be very easy for you to publish doc with that I would like to document some things I am doing with asterisk, but would prefer to do this from a web interface. I am unfamiliar with any software that allows you to create online documentation from a web interface. Ideally I will be able to create documentation online from a browser, which then when saved, is immediately ready to be read online. Perhaps I can setup different authors who are also allowed to create documentation, or have a section where users to the site can create their own documentation and submit it for inclusion. A section to submit documentation edits would be nice, as well as maybe a history timeline or something like that? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN problems (Take 2)
I have asterisk loaded and seemingly working with 4 snom 200 phones. I cannot dial in or out using a DIVA isdn card When I dial out i get not found xxx on the handset. When I dial in i see in /var/log/message isdn_net: Incoming call without OAD, assuming '0' isdn_net: call from 0 - 0 1856 ignored isdn_tty: Incoming call without OAD, assuming '0' isdn_tty: call from 0 - 1856 ignored I have tried isdnctrl eaz MSNNumber but am still seeing these messages. Am I missing something fundamental here? Here is my modem.conf context=remote driver=i4l language=en type=autodetect stripmsd=1 dialtype=tone mode=immediate device = /dev/ttyI0 msn=1856 TIA, Max ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: Creating documentation using a web interface
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Marc Verprat Sent: Sunday, September 07, 2003 9:20 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] OT: Creating documentation using a web interface Do-you knox PhpNuke : http://phpnuke.org/ or SPIP : http://www.spip.net/en I use SPIP for a bunch of web sites and I guess this could be very easy for you to publish doc with that Well... I was kind of trying to find something that was a little more designed to be a document creation system as opposed to the blogging system since I already have a blog style website. I'm looking at TWiki right now.. but it seems to be actually a little overkill I'm thinking. I just want something simple that other people can use to contribute documentation to as well without having to learn a whole interface thing. I may just end up using a blogging style interface for now until I can find something better suited to what I'm thinking. I'd just make my own, but I really just don't have the time. Not to mention, I really just don't enjoy website design :) Since I already have a website setup, maybe I'll just add an asterisk specific section and create little documents with that... although that's not what I'm really after. Leif Madsen. FWD# 18924 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New cvs compile; basic operational question, please.
Can someone offer a hint on what I'm doing wrong with the basic * config?
Just implemented * for the first time using yesterday's cvs. The initial
configs are based on John Todd's article at http://www.onlamp.com/lpt/a/3956,
and using two 7960's for initial testing. When one 7960 calls the other, I
get the following and the call is dropped:
-- Executing Dial(SIP/3001-ec1c, SIP/3000|20) in new stack
-- Called 3000
-- Got SIP response 488 Not Acceptable Here back from 206.222.193.92
== No one is available to answer at this time
-- Executing VoiceMail(SIP/3001-ec1c, u3000) in new stack
== Parsing '/etc/asterisk/voicemail.conf': Found
-- Playing 'vm/3000/unavail'
My sip.conf looks like:
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
allow=all ; Allow all codecs
context = bogon-calls ; Send SIP callers that we don't know about here
[3000]
type=friend ; This device takes and makes calls
username=3000 ; Username on device
secret=npi2003 ; Password for device
host=dynamic; This host is not on the same IP addr every time
context=from-sip; Inbound calls from this host go here
mailbox=100 ; Activate the message waiting light if this
; voicemailbox has messages in it
[3001]
type=friend ; This device takes and makes calls
username=3001 ; Username on device
secret=npi2003 ; Password for device
host=dynamic; This host is not on the same IP addr every time
context=from-sip; Inbound calls from this host go here
mailbox=100 ; Activate the message waiting light if this
---
and my extensions.conf looks like:
[general]
static=yes ; These two lines prevent the command-line interface
writeprotect=yes; from overwriting the config file. Leave them here.
[bogon-calls]
exten = _.,1,Congestion
[from-sip]
exten = 3000,1,Dial(SIP/3000,20)
exten = 3000,2,Voicemail(u3000)
exten = 3000,102,Voicemail(b3000)
exten = 3000,103,Hangup
exten = 3001,1,Dial(SIP/3001,20)
exten = 3001,2,Voicemail(u3001)
exten = 3001,102,Voicemail(b3001)
exten = 3001,103,Hangup
exten = 3999,1,VoicemailMain(${CALLERIDNUM})
Apparently I'm doing something wrong, but since this is my first attempt
at making * work, I don't actually have a clue what I'm doing (yet).
Asterisk did complile and install clean the first time (on new RH9 system),
and both 7960's are registered. In some attempts to dial, I do receive
vmail announcements, etc, so whatever I've done wrong I'm guessing it must
be in the above config statements.
If someone would kindly point out my error (and maybe a constructive comment
about the error so I can learn), if would be greatly appreciated.
TIA,
Rich
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RE: [Asterisk-Users] OT: Creating documentation using a web interface
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Marc Verprat Sent: Sunday, September 07, 2003 9:20 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] OT: Creating documentation using a web interface Do-you knox PhpNuke : http://phpnuke.org/ or SPIP : http://www.spip.net/en I use SPIP for a bunch of web sites and I guess this could be very easy for you to publish doc with that Now that I'm reading that SPIP site a little more carefully... it does seem closer to what I might be looking for. I'll let you know after I do some testing with it. Leif Madsen. FWD #18924 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie IVR question
On Sun, 2003-09-07 at 00:43, Tom Forbes wrote: Steven Critchfield wrote: On Tue, 2003-09-02 at 10:00, [EMAIL PROTECTED] wrote: php is not just a web scripting language anymore. it has been used in other ways for quite a while now. it works nicely from the command line, can be used with ncurses and with gtk. there are several well-known respectable large projects out there built upon php. i usually find that php's biggest critics are those who know the least about the language. however that holds true with pretty much any technology. linux suffers from the same type of critics. Just to point out, I am a php developer. I actually am employed to create and maintain a large webapp in php. I like the fact that I can take my php or perl scripts and not have to change much to them to work in the other language. Well if they are simple enough. There is enough well known documented problems with php. Such as? This is just an example that a co worker submitted recently. Now that bugs is back up I can point to it. http://bugs.php.net/bug.php?id=25281 A fair number of problems seem to be from the developers personalities. This is known in other open source software as well. Take the fact that so many people avoid qmail due to DJB. Monty Widenius of mysql causes people to continually search for something better. Although it doesn't support the php argument, here is a link for amusement. http://www.rickbradley.com/tour/ Just saying that because it is used in large projects doesn't change whether it is suited to the task. There are enough people on this planet, that statistically you will find enough people who refuse to admit the are using a square peg for the round hole. If we go back to PERL's roots, we find that it was never intended as a general, all-purpose language, but one for extracting and formatting data. Now it seems as though it's being touted as the cure-all for *anything* that requires scripting. PHP's intent, on the other hand was a bit more sophisticated. Being a web-based scripting languange, it, by necessity, had to interface with other components (and do it efficiently) in order to acquire, manipulate, and pass data between the user and any backend processes. I'm more curious to know what exactly it is about AGI scripting that would make PHP an inappropriate choice. Perl has always been intended to be glue between processes. I don't consider it the cure all for everything. While I have used the gtk extensions for php and perl for curiosity, I wouldn't suggest using them for anything that needs to be done on a production system. When you consider what it is you are doing, perl seems the perfect choice. AGI is a textual interface to your app, which then must respond in text. This is what perl was written to handle. Php is intended to take in user input, chew on it a moment, maybe consult backends, then spew data and die. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MP3 streams for MOH: idea
allow this to happen. Do you know of any tools that convert ASF to mp3? mplayer/mencoder understands ASF, mp3 and lots of other formats. Wont play an ASF stream, though...which is what he's looking for. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New cvs compile; basic operational question, please.
Just stumbled across the problem noted in my original post below. I added:
allow=ulaw
allow=ilbc
to sip.conf instead of the recommended 'allow=all' and now all phones work.
Can someone help me understand this? (It would appear, based on my very much
lack of experience, that * was attempting to set up the conversation using g723,
when all of the phones have 'default=ulaw' definitions. Should I leave the
ulaw definition for future production use, or is this really something that
I did to read/learn more about for a very small office use?)
Rich
Can someone offer a hint on what I'm doing wrong with the basic * config?
Just implemented * for the first time using yesterday's cvs. The initial
configs are based on John Todd's article at http://www.onlamp.com/lpt/a/3956,
and using two 7960's for initial testing. When one 7960 calls the other, I
get the following and the call is dropped:
-- Executing Dial(SIP/3001-ec1c, SIP/3000|20) in new stack
-- Called 3000
-- Got SIP response 488 Not Acceptable Here back from 206.222.193.92
== No one is available to answer at this time
-- Executing VoiceMail(SIP/3001-ec1c, u3000) in new stack
== Parsing '/etc/asterisk/voicemail.conf': Found
-- Playing 'vm/3000/unavail'
My sip.conf looks like:
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
allow=all ; Allow all codecs
context = bogon-calls ; Send SIP callers that we don't know about here
[3000]
type=friend ; This device takes and makes calls
username=3000 ; Username on device
secret=npi2003 ; Password for device
host=dynamic; This host is not on the same IP addr every time
context=from-sip; Inbound calls from this host go here
mailbox=100 ; Activate the message waiting light if this
; voicemailbox has messages in it
[3001]
type=friend ; This device takes and makes calls
username=3001 ; Username on device
secret=npi2003 ; Password for device
host=dynamic; This host is not on the same IP addr every time
context=from-sip; Inbound calls from this host go here
mailbox=100 ; Activate the message waiting light if this
---
and my extensions.conf looks like:
[general]
static=yes ; These two lines prevent the command-line interface
writeprotect=yes; from overwriting the config file. Leave them here.
[bogon-calls]
exten = _.,1,Congestion
[from-sip]
exten = 3000,1,Dial(SIP/3000,20)
exten = 3000,2,Voicemail(u3000)
exten = 3000,102,Voicemail(b3000)
exten = 3000,103,Hangup
exten = 3001,1,Dial(SIP/3001,20)
exten = 3001,2,Voicemail(u3001)
exten = 3001,102,Voicemail(b3001)
exten = 3001,103,Hangup
exten = 3999,1,VoicemailMain(${CALLERIDNUM})
Apparently I'm doing something wrong, but since this is my first attempt
at making * work, I don't actually have a clue what I'm doing (yet).
Asterisk did complile and install clean the first time (on new RH9 system),
and both 7960's are registered. In some attempts to dial, I do receive
vmail announcements, etc, so whatever I've done wrong I'm guessing it must
be in the above config statements.
If someone would kindly point out my error (and maybe a constructive comment
about the error so I can learn), if would be greatly appreciated.
TIA,
Rich
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---End of Original Message-
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Re: [Asterisk-Users] MP3 streams for MOH: idea
On Sun, Sep 07, 2003 at 12:21:18PM -0500, James Sharp wrote: allow this to happen. Do you know of any tools that convert ASF to mp3? mplayer/mencoder understands ASF, mp3 and lots of other formats. Wont play an ASF stream, though...which is what he's looking for. you're sure? e.g. mplayer http://live.atlas.cz/radio1/radio1-32.asx works fine here. (using mplayer version 0.90rc5-2.95.4 here with all the codec packages from the mplayer website) -- http://graphics.cs.uni-sb.de/VoIP/ pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] MP3 streams for MOH: idea
Wont play an ASF stream, though...which is what he's looking for. you're sure? e.g. mplayer http://live.atlas.cz/radio1/radio1-32.asx works fine here. Well, hell. Make a liar out of me. It wouldn't last time I looked. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] app_queue input needed...
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David C. Troy Sent: Sunday, September 07, 2003 11:05 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] app_queue input needed... Brian, I just finished doing both of these mods myself. The patch is available here: http://asterisk.toad.net/app_queue.c-pos+holdtimepatch You might take a look and see if it helps you at all. I make no claims that mine is a definitive version; I just did it to suit my own needs. A couple of things you might want to watch out for with your own coding: - someone had posted a similar patch in June (from pbx.usedontmiss.com) from which I borrowed some ideas, however, this version made the mistake of announcing queue position from inside a thread-locked code section, which will interrupt music-on-hold for all callers while their positions are announced. My version does the announcements outside of the thread-lock. See end of leave_queue. Been tested with over 90 calls in the queue. We did not run into this situation. - If you are caller #4 and callers #3 and #2 leave in rapid succession, you'll be trying to announce position to caller #4 twice, quickly, possibly interleaving your announcements. To solve this problem, no announcement is made if a position announcement was last made within the last 15 seconds. See the first couple of lines of say_position. I see that you set this to 10 seconds actually, but even at 15, that's a little crazy. IMO, announcements should be made no more often than 60 seconds. Personally, I find it quite annoying when something interupts the MOH 3-4 times a minute to say your call is important to us or we value your business or you are caller number 8274738, please wait. If you look at our code, the default is to announce every 60 seconds. It doesn't matter if 1 person or 30 people in front of you drop out in rapid succession (actually, we did test with 30 callers dropping at one time). The next announcement will not come before 60 seconds have passed. My version operates along these lines: - To turn on position/holdtime announcements, just define 'announcetimeout' in queues.conf; position/holdtime announcements will be made at that interval. Not enough control is being offered here. At a minmum, each of the following should be doable: - Announce without position or time - Announce with time only - Announce with position only - Announce with position and time - The guts of the operation is in say_position and get_avg_holdtime; changes to the rest of the code have been minimized. - No position/holdtime announcement is made if the caller is the only caller in the queue when they enter the queue. - A special 'you're next' announcement is played when position == 1 I like this idea very much. - Holdtime is only announced if it is 4 minutes or greater; this is arbitrary and could be rolled into a configurable queues.conf setting. Yes, this needs to a config option for sure. I would want announcements all the way down to less than 2 minutes. - You need to add the following files to /var/lib/asterisk/sounds: YouAreNext You are now first in line; your call will be next ThereAre There are currently... CallsWaiting Calls Waiting AvgHoldTime The approximate average hold time is currently... MinutesMinutes ThankYou Thank you for your patience It would be cool if we could get Mark to have Allison record these files for distribution with the package, once this family of features is committed to app_queue. Perhaps it would be cool if you got Allison to record these. I was going to have her do the ones for our patch, but John Todd volunteered before I even knew that Brian had posted the patches up (which is why there's also like 3 versions of the patch so far). I like the idea of rounding up to under 5, under 10, under 15 announcements as well, as it would sound less cheesy and be better at setting realistic, but not exact, holdtime expectations. My vote for announcing hold times, is to use a configuration varaible to specify minutes and seconds, or to round up to the nearest minute (or 2,3,4,5 minutes). For that matter, you could even decrease the granularity based on the position and how long the wait is. If I'm in position 3, and the wait time is less than 1 minute, why not announce the seconds? Chances are, the call will be answered right after the caller hears your estimated wait time is 38 seconds. On the other hand, that might be annoying to caller 10 who's got a hefty 12 minute wait ahead of her, so that announcement should be less than 15 minutes. I also want to see the ability to fail out of a queue (perhaps defined in terms of X number of retry/timeout cycles) and will be working on this today some. That patch was done by outtolunc, and I expect that
Re: [Asterisk-Users] New cvs compile; basic operational question, please.
Rich -
Leave out the allow lines entirely, including the allow=all -
this was a problem I discovered post-publishing (that I thought I
corrected in the notes, but I see that it's not there.) The allow=
lines need a disallow= line to balance them. If you leave both
out, the system will choose the right codec, but if you only put
one in, things get twisted up a bit. I've updated the article
(again?)
JT
Just stumbled across the problem noted in my original post below. I added:
allow=ulaw
allow=ilbc
to sip.conf instead of the recommended 'allow=all' and now all phones work.
Can someone help me understand this? (It would appear, based on my very much
lack of experience, that * was attempting to set up the conversation
using g723,
when all of the phones have 'default=ulaw' definitions. Should I leave the
ulaw definition for future production use, or is this really something that
I did to read/learn more about for a very small office use?)
Rich
Can someone offer a hint on what I'm doing wrong with the basic * config?
Just implemented * for the first time using yesterday's cvs. The initial
configs are based on John Todd's article at
http://www.onlamp.com/lpt/a/3956,
and using two 7960's for initial testing. When one 7960 calls the other, I
get the following and the call is dropped:
-- Executing Dial(SIP/3001-ec1c, SIP/3000|20) in new stack
-- Called 3000
-- Got SIP response 488 Not Acceptable Here back from 206.222.193.92
== No one is available to answer at this time
-- Executing VoiceMail(SIP/3001-ec1c, u3000) in new stack
== Parsing '/etc/asterisk/voicemail.conf': Found
-- Playing 'vm/3000/unavail'
My sip.conf looks like:
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
allow=all ; Allow all codecs
context = bogon-calls ; Send SIP callers that we don't know about here
[3000]
type=friend ; This device takes and makes calls
username=3000 ; Username on device
secret=npi2003 ; Password for device
host=dynamic; This host is not on the same IP addr every time
context=from-sip; Inbound calls from this host go here
mailbox=100 ; Activate the message waiting light if this
; voicemailbox has messages in it
[3001]
type=friend ; This device takes and makes calls
username=3001 ; Username on device
secret=npi2003 ; Password for device
host=dynamic; This host is not on the same IP addr every time
context=from-sip; Inbound calls from this host go here
mailbox=100 ; Activate the message waiting light if this
---
and my extensions.conf looks like:
[general]
static=yes ; These two lines prevent the command-line interface
writeprotect=yes; from overwriting the config file. Leave them here.
[bogon-calls]
exten = _.,1,Congestion
[from-sip]
exten = 3000,1,Dial(SIP/3000,20)
exten = 3000,2,Voicemail(u3000)
exten = 3000,102,Voicemail(b3000)
exten = 3000,103,Hangup
exten = 3001,1,Dial(SIP/3001,20)
exten = 3001,2,Voicemail(u3001)
exten = 3001,102,Voicemail(b3001)
exten = 3001,103,Hangup
exten = 3999,1,VoicemailMain(${CALLERIDNUM})
Apparently I'm doing something wrong, but since this is my first attempt
at making * work, I don't actually have a clue what I'm doing (yet).
Asterisk did complile and install clean the first time (on new RH9 system),
and both 7960's are registered. In some attempts to dial, I do receive
vmail announcements, etc, so whatever I've done wrong I'm guessing it must
be in the above config statements.
If someone would kindly point out my error (and maybe a constructive comment
about the error so I can learn), if would be greatly appreciated.
TIA,
Rich
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Re: [Asterisk-Users] New cvs compile; basic operational question, please.
Make sure that both phones are set to accept the same codecs. The Not
Acceptable Here is usually when the SIP negotiation fails for a common
codec.
Use SIP DEBUG at the CLI to see the transcripts for details. You might
want to use in sip.conf allow and disallow statements for codecs as
well.
On Sunday, September 7, 2003, at 07:04 PM, Rich Adamson wrote:
Can someone offer a hint on what I'm doing wrong with the basic *
config?
Just implemented * for the first time using yesterday's cvs. The
initial
configs are based on John Todd's article at
http://www.onlamp.com/lpt/a/3956,
and using two 7960's for initial testing. When one 7960 calls the
other, I
get the following and the call is dropped:
-- Executing Dial(SIP/3001-ec1c, SIP/3000|20) in new stack
-- Called 3000
-- Got SIP response 488 Not Acceptable Here back from
206.222.193.92
== No one is available to answer at this time
-- Executing VoiceMail(SIP/3001-ec1c, u3000) in new stack
== Parsing '/etc/asterisk/voicemail.conf': Found
-- Playing 'vm/3000/unavail'
My sip.conf looks like:
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
allow=all ; Allow all codecs
context = bogon-calls ; Send SIP callers that we don't know about
here
[3000]
type=friend ; This device takes and makes calls
username=3000 ; Username on device
secret=npi2003 ; Password for device
host=dynamic; This host is not on the same IP addr every
time
context=from-sip; Inbound calls from this host go here
mailbox=100 ; Activate the message waiting light if this
; voicemailbox has messages in it
[3001]
type=friend ; This device takes and makes calls
username=3001 ; Username on device
secret=npi2003 ; Password for device
host=dynamic; This host is not on the same IP addr every
time
context=from-sip; Inbound calls from this host go here
mailbox=100 ; Activate the message waiting light if this
---
and my extensions.conf looks like:
[general]
static=yes ; These two lines prevent the command-line
interface
writeprotect=yes; from overwriting the config file. Leave them
here.
[bogon-calls]
exten = _.,1,Congestion
[from-sip]
exten = 3000,1,Dial(SIP/3000,20)
exten = 3000,2,Voicemail(u3000)
exten = 3000,102,Voicemail(b3000)
exten = 3000,103,Hangup
exten = 3001,1,Dial(SIP/3001,20)
exten = 3001,2,Voicemail(u3001)
exten = 3001,102,Voicemail(b3001)
exten = 3001,103,Hangup
exten = 3999,1,VoicemailMain(${CALLERIDNUM})
Apparently I'm doing something wrong, but since this is my first
attempt
at making * work, I don't actually have a clue what I'm doing (yet).
Asterisk did complile and install clean the first time (on new RH9
system),
and both 7960's are registered. In some attempts to dial, I do receive
vmail announcements, etc, so whatever I've done wrong I'm guessing it
must
be in the above config statements.
If someone would kindly point out my error (and maybe a constructive
comment
about the error so I can learn), if would be greatly appreciated.
TIA,
Rich
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RE: [Asterisk-Users] app_queue input needed...
Troy, These is all good feedback; I did my patch primarily based on my own needs, so YMMV. The business about the thread-locked code was related to the usedontmiss.com patch, not yours. Yours seems to avoid that problem, as does mine. I spent the last couple of hours digging deeper into this, and I incorporated a 'queuetimeout' (similar to what outtolunc did, but rather than transferring to 's,1,...' within the queue context, it just exits and lets the originating extension list figure out what to do with it. I agree that more control needs to be added to make a general solution; that will mean the addition of a few, preferably a minimal number, queues.conf parameters. To me, that's up to the community and to Mark. I'll post the new patch in a few minutes, after a bit more testing, but it seems to work very smoothly now; I figured out some stuff I didn't understand about the various loops in the code, and that's well commented now. Patch will be located here: http://asterisk.toad.net/app_queue.c-pos+holdtime+queuetimeout.patch Take a look at this patch and let's see what we want to do about the holdtime logic; at a minimum this patch cleans up and packages all that logic very neatly, so it then becomes a simple question of how to invoke it rather than the harder question of how to do it generally. Dave = David C. Troy [EMAIL PROTECTED] 410-384-2500 Sales ToadNet - Want to go fast?410-544-1329 FAX 570 Ritchie Highway, Severna Park, MD 21146-2925 www.toad.net On Sun, 7 Sep 2003, Troy Settle wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David C. Troy Sent: Sunday, September 07, 2003 11:05 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] app_queue input needed... Brian, I just finished doing both of these mods myself. The patch is available here: http://asterisk.toad.net/app_queue.c-pos+holdtimepatch You might take a look and see if it helps you at all. I make no claims that mine is a definitive version; I just did it to suit my own needs. A couple of things you might want to watch out for with your own coding: - someone had posted a similar patch in June (from pbx.usedontmiss.com) from which I borrowed some ideas, however, this version made the mistake of announcing queue position from inside a thread-locked code section, which will interrupt music-on-hold for all callers while their positions are announced. My version does the announcements outside of the thread-lock. See end of leave_queue. Been tested with over 90 calls in the queue. We did not run into this situation. - If you are caller #4 and callers #3 and #2 leave in rapid succession, you'll be trying to announce position to caller #4 twice, quickly, possibly interleaving your announcements. To solve this problem, no announcement is made if a position announcement was last made within the last 15 seconds. See the first couple of lines of say_position. I see that you set this to 10 seconds actually, but even at 15, that's a little crazy. IMO, announcements should be made no more often than 60 seconds. Personally, I find it quite annoying when something interupts the MOH 3-4 times a minute to say your call is important to us or we value your business or you are caller number 8274738, please wait. If you look at our code, the default is to announce every 60 seconds. It doesn't matter if 1 person or 30 people in front of you drop out in rapid succession (actually, we did test with 30 callers dropping at one time). The next announcement will not come before 60 seconds have passed. My version operates along these lines: - To turn on position/holdtime announcements, just define 'announcetimeout' in queues.conf; position/holdtime announcements will be made at that interval. Not enough control is being offered here. At a minmum, each of the following should be doable: - Announce without position or time - Announce with time only - Announce with position only - Announce with position and time - The guts of the operation is in say_position and get_avg_holdtime; changes to the rest of the code have been minimized. - No position/holdtime announcement is made if the caller is the only caller in the queue when they enter the queue. - A special 'you're next' announcement is played when position == 1 I like this idea very much. - Holdtime is only announced if it is 4 minutes or greater; this is arbitrary and could be rolled into a configurable queues.conf setting. Yes, this needs to a config option for sure. I would want announcements all the way down to less than 2 minutes. - You need to add the following files to
RE: [Asterisk-Users] app_queue input needed...
I think the hold time needs to be announced only once when the caller is injected into the call queue. Otherwise callers will hear hold times shoot up if you have a few long calls. bkw On Sun, 7 Sep 2003, David C. Troy wrote: Troy, These is all good feedback; I did my patch primarily based on my own needs, so YMMV. The business about the thread-locked code was related to the usedontmiss.com patch, not yours. Yours seems to avoid that problem, as does mine. I spent the last couple of hours digging deeper into this, and I incorporated a 'queuetimeout' (similar to what outtolunc did, but rather than transferring to 's,1,...' within the queue context, it just exits and lets the originating extension list figure out what to do with it. I agree that more control needs to be added to make a general solution; that will mean the addition of a few, preferably a minimal number, queues.conf parameters. To me, that's up to the community and to Mark. I'll post the new patch in a few minutes, after a bit more testing, but it seems to work very smoothly now; I figured out some stuff I didn't understand about the various loops in the code, and that's well commented now. Patch will be located here: http://asterisk.toad.net/app_queue.c-pos+holdtime+queuetimeout.patch Take a look at this patch and let's see what we want to do about the holdtime logic; at a minimum this patch cleans up and packages all that logic very neatly, so it then becomes a simple question of how to invoke it rather than the harder question of how to do it generally. Dave = David C. Troy [EMAIL PROTECTED] 410-384-2500 Sales ToadNet - Want to go fast?410-544-1329 FAX 570 Ritchie Highway, Severna Park, MD 21146-2925 www.toad.net On Sun, 7 Sep 2003, Troy Settle wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David C. Troy Sent: Sunday, September 07, 2003 11:05 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] app_queue input needed... Brian, I just finished doing both of these mods myself. The patch is available here: http://asterisk.toad.net/app_queue.c-pos+holdtimepatch You might take a look and see if it helps you at all. I make no claims that mine is a definitive version; I just did it to suit my own needs. A couple of things you might want to watch out for with your own coding: - someone had posted a similar patch in June (from pbx.usedontmiss.com) from which I borrowed some ideas, however, this version made the mistake of announcing queue position from inside a thread-locked code section, which will interrupt music-on-hold for all callers while their positions are announced. My version does the announcements outside of the thread-lock. See end of leave_queue. Been tested with over 90 calls in the queue. We did not run into this situation. - If you are caller #4 and callers #3 and #2 leave in rapid succession, you'll be trying to announce position to caller #4 twice, quickly, possibly interleaving your announcements. To solve this problem, no announcement is made if a position announcement was last made within the last 15 seconds. See the first couple of lines of say_position. I see that you set this to 10 seconds actually, but even at 15, that's a little crazy. IMO, announcements should be made no more often than 60 seconds. Personally, I find it quite annoying when something interupts the MOH 3-4 times a minute to say your call is important to us or we value your business or you are caller number 8274738, please wait. If you look at our code, the default is to announce every 60 seconds. It doesn't matter if 1 person or 30 people in front of you drop out in rapid succession (actually, we did test with 30 callers dropping at one time). The next announcement will not come before 60 seconds have passed. My version operates along these lines: - To turn on position/holdtime announcements, just define 'announcetimeout' in queues.conf; position/holdtime announcements will be made at that interval. Not enough control is being offered here. At a minmum, each of the following should be doable: - Announce without position or time - Announce with time only - Announce with position only - Announce with position and time - The guts of the operation is in say_position and get_avg_holdtime; changes to the rest of the code have been minimized. - No position/holdtime announcement is made if the caller is the only caller in the queue when they enter the queue. - A special 'you're next' announcement is played when position == 1 I like this idea very much. - Holdtime is only announced if it is 4
Re: [Asterisk-Users] digium dev kit - X100P TDM400P
On Sunday 07 September 2003 06:10 am, Timothy Soos wrote: I too am using Linux Mandrake 9.1 with the DevKit TDM. You may like to know that once you get asterisk properly installed, it works very well with these 2 boards. Tell me about the sound. I use ALSA as it's the preferred, or so the hardware diagnostics say's. asterisk is looking for OSS and and shows a failure during startup, although I can hear the demo messages when I dial in. Any ideas about this ? Regards...Martin -- When in doubt, do what the President does -- guess. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New cvs compile; basic operational question, please.
John,
Excellent, I removed them and works fine.
I started playing with MOH, but haven't seen an example of how to
specify this on a per system or per extension (but I haven't googled
for it yet either). I do have one line uncommented in the moh config
file, but I assume I need to do something in sip.conf or extensions.conf
to make it work.
Suggestions?
Rich
Rich -
Leave out the allow lines entirely, including the allow=all -
this was a problem I discovered post-publishing (that I thought I
corrected in the notes, but I see that it's not there.) The allow=
lines need a disallow= line to balance them. If you leave both
out, the system will choose the right codec, but if you only put
one in, things get twisted up a bit. I've updated the article
(again?)
JT
Just stumbled across the problem noted in my original post below. I added:
allow=ulaw
allow=ilbc
to sip.conf instead of the recommended 'allow=all' and now all phones work.
Can someone help me understand this? (It would appear, based on my very much
lack of experience, that * was attempting to set up the conversation
using g723,
when all of the phones have 'default=ulaw' definitions. Should I leave the
ulaw definition for future production use, or is this really something that
I did to read/learn more about for a very small office use?)
Rich
Can someone offer a hint on what I'm doing wrong with the basic * config?
Just implemented * for the first time using yesterday's cvs. The initial
configs are based on John Todd's article at
http://www.onlamp.com/lpt/a/3956,
and using two 7960's for initial testing. When one 7960 calls the other, I
get the following and the call is dropped:
-- Executing Dial(SIP/3001-ec1c, SIP/3000|20) in new stack
-- Called 3000
-- Got SIP response 488 Not Acceptable Here back from 206.222.193.92
== No one is available to answer at this time
-- Executing VoiceMail(SIP/3001-ec1c, u3000) in new stack
== Parsing '/etc/asterisk/voicemail.conf': Found
-- Playing 'vm/3000/unavail'
My sip.conf looks like:
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
allow=all ; Allow all codecs
context = bogon-calls ; Send SIP callers that we don't know about here
[3000]
type=friend ; This device takes and makes calls
username=3000 ; Username on device
secret=npi2003 ; Password for device
host=dynamic; This host is not on the same IP addr every time
context=from-sip; Inbound calls from this host go here
mailbox=100 ; Activate the message waiting light if this
; voicemailbox has messages in it
[3001]
type=friend ; This device takes and makes calls
username=3001 ; Username on device
secret=npi2003 ; Password for device
host=dynamic; This host is not on the same IP addr every time
context=from-sip; Inbound calls from this host go here
mailbox=100 ; Activate the message waiting light if this
---
and my extensions.conf looks like:
[general]
static=yes ; These two lines prevent the command-line interface
writeprotect=yes; from overwriting the config file. Leave them here.
[bogon-calls]
exten = _.,1,Congestion
[from-sip]
exten = 3000,1,Dial(SIP/3000,20)
exten = 3000,2,Voicemail(u3000)
exten = 3000,102,Voicemail(b3000)
exten = 3000,103,Hangup
exten = 3001,1,Dial(SIP/3001,20)
exten = 3001,2,Voicemail(u3001)
exten = 3001,102,Voicemail(b3001)
exten = 3001,103,Hangup
exten = 3999,1,VoicemailMain(${CALLERIDNUM})
Apparently I'm doing something wrong, but since this is my first attempt
at making * work, I don't actually have a clue what I'm doing (yet).
Asterisk did complile and install clean the first time (on new RH9 system),
and both 7960's are registered. In some attempts to dial, I do receive
vmail announcements, etc, so whatever I've done wrong I'm guessing it must
be in the above config statements.
If someone would kindly point out my error (and maybe a constructive comment
about the error so I can learn), if
[Asterisk-Users] Sound error during launch
Hello. Although I can hear the demo etc. now, I notice during asterisk launch I get :- [chan_oss.so] = (OSS Console Channel Driver) == Console is full duplex == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found WARNING[98311]: File chan_oss.c, Line 232 (sound_thread): Read error on sound device: Resource temporarily unavailable I'm running mandrake 9.1 with the ALSA sound driver. In the control center | Hardware | HardDrake | soundcard | help it suggest this is preferred. Is that warning important ? How do I correct it ? Regards...Martin -- It is better to be bow-legged than no-legged. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is 3 minutes a magic number of some kind??
All -- Having the a problem with calls cutting off after 3 minutes -- not 2:30 or 3:12, but *exactly* 180 seconds. To illustrate, I have the following extension: Exten = 8005551212,1,MusicOnHold If this extension is dialed from a Cisco 7960 (SIP), the call plays the music, and then disconnects after 3 minutes/180 seconds exactly. If this extension is dialed from a PSTN Zap interface, the call plays the music, and disconnects after 3 minutes/180 seconds exactly, giving the caller the congestion tone. If this extension is dialed from a Pingtel Xpressa or ATA186, the calls stay connected after 3 min and indefinitely just fine. Wondering what sort of value could be affecting both Zap and SIP call progress/timeouts, etc. Any input appreciated. Regards, Dave = David C. Troy [EMAIL PROTECTED] 410-384-2500 Sales ToadNet - Want to go fast?410-544-1329 FAX 570 Ritchie Highway, Severna Park, MD 21146-2925 www.toad.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] freebsd and asterisk ?? anyone yet
so has anyone gotten * ported to freeBSD yet ?? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_local environments: unexpected results
I'm having some difficulty with chan_local dial requests. It seems
that when a chan_local call is picked up, that the native bridge
pops the environment back to the settings of the original call.
This is unexpected and leads to very frustrating results. My
example below is a very distilled sample of a much more complex
dialplan problem I'm having with chan_local, but it illustrates the
point.
In my example below, the variable MYTEST is set in the original
call. Then, the call is handed to chan_local at a different dialed
number. Then, as expected, a new environment is set up, and I cannot
see the value I set for MYTEST in this new environment - so far,so
goo. I create another variable called OTHERTEST in this new
environment.
However, as soon as the Answer application is called, I see the
Local channel hangs up. Suddenly, the environment of the call is
handed back to the original call's settings! This seems
counter-intuitive, since my dial plan is following a path and
expecting variable settings that may not be available in the original
call's environment. I would think that Local calls would be
sandboxed such that they cannot see the environments from other
calls, since that is how all the other channel types work...
Is this the expected behavior?
JT
Calls are handed from my SIP phone directly to [local]. The user at
SIP extension 2209 is not registered, so the Busy (priority 105 in
this case) routines will be called.
[local]
exten = 2213,1,SetVar(MYTEST=ishouldnotseethis)
exten = 2213,2,Dial(Local/[EMAIL PROTECTED])
exten = 2209,1,SetVar(OTHERTEST=goodness)
exten = 2209,2,NoOp(${MYTEST})
exten = 2209,3,NoOp(${OTHERTEST})
exten = 2209,4,Dial(SIP/2209)
exten = 2209,105,Answer
exten = 2209,106,Playback(invalid)
exten = 2209,107,NoOp(${MYTEST})
exten = 2209,108,NoOp(${OTHERTEST})
exten = 2209,109,Hangup
ms1*CLI
-- Executing SetVar(SIP/2203-2496, MYTEST=ishouldnotseethis)
in new stack
-- Executing Dial(SIP/2203-2496, Local/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
-- Executing SetVar(Local/[EMAIL PROTECTED],2,
OTHERTEST=goodness) in new stack
-- Executing NoOp(Local/[EMAIL PROTECTED],2, ) in new stack
-- Executing NoOp(Local/[EMAIL PROTECTED],2, goodness) in new stack
-- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/2209) in new stack
== Everyone is busy at this time
-- Executing Answer(Local/[EMAIL PROTECTED],2, ) in new stack
-- Local/[EMAIL PROTECTED],1 answered SIP/2203-2496
-- Executing Playback(Local/[EMAIL PROTECTED],2, invalid) in new stack
-- Playing 'invalid'
== Spawn extension (local, 2213, 2) exited non-zero on
'Local/[EMAIL PROTECTED],2ZOMBIE'
ms1*CLI [I hear the recording on my SIP phone at this point]
ms1*CLI
-- Executing NoOp(SIP/2203-2496, ishouldnotseethis) in new stack
-- Executing NoOp(SIP/2203-2496, ) in new stack
-- Executing Hangup(SIP/2203-2496, ) in new stack
== Spawn extension (local, 2209, 109) exited non-zero on 'SIP/2203-2496'
ms1*CLI
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Re: [Asterisk-Users] Is 3 minutes a magic number of some kind??
On Sun, 2003-09-07 at 16:07, David C. Troy wrote: All -- Having the a problem with calls cutting off after 3 minutes -- not 2:30 or 3:12, but *exactly* 180 seconds. To illustrate, I have the following extension: Exten = 8005551212,1,MusicOnHold If this extension is dialed from a Cisco 7960 (SIP), the call plays the music, and then disconnects after 3 minutes/180 seconds exactly. If this extension is dialed from a PSTN Zap interface, the call plays the music, and disconnects after 3 minutes/180 seconds exactly, giving the caller the congestion tone. If this extension is dialed from a Pingtel Xpressa or ATA186, the calls stay connected after 3 min and indefinitely just fine. Wondering what sort of value could be affecting both Zap and SIP call progress/timeouts, etc. Any input appreciated. The case of the SIP to SIP calls get routed outside of asterisk after initial setup and therefore do not get limited. Your Zap to SIP, Sip to Zap, and Sip or Zap to asterisk would be limited by what asterisk does. Look for an Absolutetimeout in your dialplan. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New cvs compile; basic operational question, please.
Rich -
To get MOH working, I'd suggest looking at the archives for this
list a bit - lots of clues there. Ensure that you actually have
mpg123, and not mpg321. Restart asterisk. It should just work with
7960's.
JT
John,
Excellent, I removed them and works fine.
I started playing with MOH, but haven't seen an example of how to
specify this on a per system or per extension (but I haven't googled
for it yet either). I do have one line uncommented in the moh config
file, but I assume I need to do something in sip.conf or extensions.conf
to make it work.
Suggestions?
Rich
Rich -
Leave out the allow lines entirely, including the allow=all -
this was a problem I discovered post-publishing (that I thought I
corrected in the notes, but I see that it's not there.) The allow=
lines need a disallow= line to balance them. If you leave both
out, the system will choose the right codec, but if you only put
one in, things get twisted up a bit. I've updated the article
(again?)
JT
Just stumbled across the problem noted in my original post below. I added:
allow=ulaw
allow=ilbc
to sip.conf instead of the recommended 'allow=all' and now all phones work.
Can someone help me understand this? (It would appear, based on
my very much
lack of experience, that * was attempting to set up the conversation
using g723,
when all of the phones have 'default=ulaw' definitions. Should I leave the
ulaw definition for future production use, or is this really something that
I did to read/learn more about for a very small office use?)
Rich
Can someone offer a hint on what I'm doing wrong with the
basic * config?
Just implemented * for the first time using yesterday's cvs. The initial
configs are based on John Todd's article at
http://www.onlamp.com/lpt/a/3956,
and using two 7960's for initial testing. When one 7960 calls
the other, I
get the following and the call is dropped:
-- Executing Dial(SIP/3001-ec1c, SIP/3000|20) in new stack
-- Called 3000
-- Got SIP response 488 Not Acceptable Here back from
206.222.193.92
== No one is available to answer at this time
-- Executing VoiceMail(SIP/3001-ec1c, u3000) in new stack
== Parsing '/etc/asterisk/voicemail.conf': Found
-- Playing 'vm/3000/unavail'
My sip.conf looks like:
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
allow=all ; Allow all codecs
context = bogon-calls ; Send SIP callers that we don't know about here
[3000]
type=friend ; This device takes and makes calls
username=3000 ; Username on device
secret=npi2003 ; Password for device
host=dynamic; This host is not on the same IP addr
every time
context=from-sip; Inbound calls from this host go here
mailbox=100 ; Activate the message waiting light if this
; voicemailbox has messages in it
[3001]
type=friend ; This device takes and makes calls
username=3001 ; Username on device
secret=npi2003 ; Password for device
host=dynamic; This host is not on the same IP
addr every time
context=from-sip; Inbound calls from this host go here
mailbox=100 ; Activate the message waiting light if this
---
and my extensions.conf looks like:
[general]
static=yes ; These two lines prevent the
command-line interface
writeprotect=yes; from overwriting the config file.
Leave them here.
[bogon-calls]
exten = _.,1,Congestion
[from-sip]
exten = 3000,1,Dial(SIP/3000,20)
exten = 3000,2,Voicemail(u3000)
exten = 3000,102,Voicemail(b3000)
exten = 3000,103,Hangup
exten = 3001,1,Dial(SIP/3001,20)
exten = 3001,2,Voicemail(u3001)
exten = 3001,102,Voicemail(b3001)
exten = 3001,103,Hangup
exten = 3999,1,VoicemailMain(${CALLERIDNUM})
Apparently I'm doing something wrong, but since this is my first attempt
at making * work, I don't actually have a clue what I'm doing (yet).
Asterisk did complile and install clean the first time (on new
RH9 system),
and both 7960's are registered. In some attempts to dial, I do receive
vmail announcements, etc, so whatever I've done wrong I'm
guessing it must
be in the above config statements.
If someone would kindly
[Asterisk-Users] Queue Viewer Perl CGI App
All -- I hacked together a quick perl script to view queue status via a web page; it uses the manager interface to connect to asterisk on a local/remote box and grabs the queue info, formats it, and that's about it. I'm submitting in the hopes that it might be useful to current users and could serve as a basis for further development. http://asterisk.toad.net/qview.pl It's easy to setup, but requires CGI.pm on your web server; you might want to create a css file for it, too, and store it at /pbxinfo.css on your web server. You can copy mine from: http://asterisk.toad.net/pbxinfo.css If you like it, enhance it. :) Dave = David C. Troy [EMAIL PROTECTED] 410-384-2500 Sales ToadNet - Want to go fast?410-544-1329 FAX 570 Ritchie Highway, Severna Park, MD 21146-2925 www.toad.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CallerID through the GnuGK - does this work?
Hi - can anyone confirm or deny that CallerID works through (passes through) the GnuGK? ie., X100P - Asterisk - GnuGK - Gateway Thanks. Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue Viewer Perl CGI App
You might want to check out the Asterisk Perl modules James has created http://asterisk.gnuinter.net Jeremy McNamara David C. Troy wrote: All -- I hacked together a quick perl script to view queue status via a web page; it uses the manager interface to connect to asterisk on a local/remote box and grabs the queue info, formats it, and that's about it. I'm submitting in the hopes that it might be useful to current users and could serve as a basis for further development. http://asterisk.toad.net/qview.pl It's easy to setup, but requires CGI.pm on your web server; you might want to create a css file for it, too, and store it at /pbxinfo.css on your web server. You can copy mine from: http://asterisk.toad.net/pbxinfo.css If you like it, enhance it. :) Dave = David C. Troy [EMAIL PROTECTED] 410-384-2500 Sales ToadNet - Want to go fast?410-544-1329 FAX 570 Ritchie Highway, Severna Park, MD 21146-2925 www.toad.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk leaving many UDP ports available.
Hi there, Should asterisk have many, many UDP ports open? [EMAIL PROTECTED] asterisk]# netstat -nap|grep asterisk|less udp0 0 0.0.0.0:17920 0.0.0.0:* 19773/asterisk udp0 0 0.0.0.0:12544 0.0.0.0:* 19773/asterisk udp0 0 0.0.0.0:10240 0.0.0.0:* 19773/asterisk udp0 0 0.0.0.0:12800 0.0.0.0:* 19773/asterisk udp0 0 0.0.0.0:17921 0.0.0.0:* 19773/asterisk udp0 0 0.0.0.0:12545 0.0.0.0:* 19773/asterisk udp0 0 0.0.0.0:10241 0.0.0.0:* 19773/asterisk ... [EMAIL PROTECTED] asterisk]# netstat -nap|grep asterisk|wc -l 447 I know that the other day I had to recycle asterisk because I was getting too many open files errors in the log file (and asterisk was dead). ...deon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] digium dev kit - X100P TDM400P
On Sunday 07 September 2003 01:41 pm, marrandy wrote: On Sunday 07 September 2003 06:10 am, Timothy Soos wrote: I too am using Linux Mandrake 9.1 with the DevKit TDM. You may like to know that once you get asterisk properly installed, it works very well with these 2 boards. Tell me about the sound. The sound is quite good: at least as good as any connection I make with a decent residential analog phone to a local phone number. There is no noticeable latency. I use ALSA as it's the preferred, or so the hardware diagnostics say's. I use ARTS. ALSA was always a pain-in-the-butt for me. asterisk is looking for OSS and and shows a failure during startup, although I can hear the demo messages when I dial in. You are probably getting the message: WARNING[196621]: File chan_oss.c, Line 232 (sound_thread): Read error on sound device: Resource temporarily unavailable. I get that too. Still, I am not very concerned about it because the systems works well. I am under a bit of a time crunch, and this does not seem to be a major problem, so I am not overly concerned about it at this moment. Yet, if you do find the solution, I would like to know what the fix is. Any ideas about this ? If your sound is not working or poorly working, switch to ARTS. Other than that suggestion, I do not yet know how to help you on this. -- Thanks, Timothy Soos XQL, LLC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Conference Leader
Hello, Is meetme able to do the following scenario: Say a group of caller calling in to the same conference room where one of them is a conference leader.The leadercan press a key which gives that person options like dial out to a particular person and tranfer him/her to the conference. The leader's password is different from the other. I think the tricky part in this scenario is how todetectthe key press by leader. any idea.
Re: [Asterisk-Users] Asterisk Application Documentation
On Sunday 07 September 2003 05:17 am, Olle E. Johansson wrote: I've spent some time on the Wiki adding documentation on all asterisk applications from the cli 'show application ' commands. I've also added some cross references and pointers. http://www.voip-info.org/tiki-index.php?page=Asterisk Thank you very much from providing this resource. There is a lot of very good information there! :) -- Thanks, Timothy Soos XQL, LLC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GrandStream Phones... White,Black or Green?
If anyone on here knows how to actually purchase Grandstream equipment, please contact me. I will let you decide if it should be on-list or off. Specifically, I want to buy one of the HandyTone's, and maybe later a BudgeTone. I contacted them directly, but had no reply to a request for direct sale. (Resellers contacting me offlist would be fine, if you're not gouging!) -- Andrew Thompson Quoting Brian West [EMAIL PROTECTED]: Just in case you guys haven't been paying attention Grandstream sliped in some diffrent colors on the IP phones and looks like they released the ATA-286 (Cisco is gonna have kittens I suspect) bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GrandStream Phones... White,Black or Green?
http://store.yahoo.com/grandstream-networks-inc/products.html They finally removed the password from their shopping cart! -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andrew Thompson Sent: Sunday, September 07, 2003 10:36 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] GrandStream Phones... White,Black or Green? If anyone on here knows how to actually purchase Grandstream equipment, please contact me. I will let you decide if it should be on-list or off. Specifically, I want to buy one of the HandyTone's, and maybe later a BudgeTone. I contacted them directly, but had no reply to a request for direct sale. (Resellers contacting me offlist would be fine, if you're not gouging!) -- Andrew Thompson Quoting Brian West [EMAIL PROTECTED]: Just in case you guys haven't been paying attention Grandstream sliped in some diffrent colors on the IP phones and looks like they released the ATA-286 (Cisco is gonna have kittens I suspect) bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE410 - 3.3v?
I just noticed that the new TE410 only supports 3.3v PCI. How do you know if it will work in your machine? It's not something commonly listed in your motherboard specs I was thinking about getting one for an Athlon 750 that is a spare box I'm doing testing on... (Original SlotA) -G ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem Installing Open H.323 Channel Driver
Hi,
I am having trouble installing the Open H.323 Channel driver in Asterisk. I
have installed PWLib and OpenH323 V.1.11.7
This is the output l get before it bails out.
asterisk:/usr/src/asterisk/channels/h323# make
g++ -g -c -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DNDEBUG
-DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC -Wmissing-prototypes
-Wmissing-declarations -DP_LINUX -D_REENTRANT -D_GNU_SOURCE
-DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES -DPTRACING
-DP_USE_PRAGMA -I../../include -I/root/pwlib/include/ptlib/unix
-I/root/pwlib/include -I/root/openh323/include -Wno-missing-prototypes
-Wno-missing-declarations ast_h323.cpp
In file included from ast_h323.cpp:28:
ast_h323.h:31: h323.h: No such file or directory
ast_h323.h:32: h323pdu.h: No such file or directory
ast_h323.h:33: mediafmt.h: No such file or directory
ast_h323.h:34: lid.h: No such file or directory
ast_h323.cpp:29: h323t38.h: No such file or directory
In file included from ast_h323.cpp:28:
ast_h323.h:71: parse error before `{'
ast_h323.h:72: virtual outside class declaration
ast_h323.h:72: non-member function `GetClass(unsigned int)' cannot have
`const' method qualifier
ast_h323.h: In function `const char * GetClass(unsigned int = 0)':
ast_h323.h:72: `H323AudioCapability' undeclared (first use this function)
ast_h323.h:72: (Each undeclared identifier is reported only once
ast_h323.h:72: for each function it appears in.)
ast_h323.h:72: parse error before `::'
ast_h323.h:72: no method `H323_G7231Capability::Class'
ast_h323.h:72: warning: control reaches end of non-void function
`GetClass(unsigned int)'
ast_h323.h: At top level:
ast_h323.h:72: virtual outside class declaration
ast_h323.h:72: non-member function `IsClass(const char *)' cannot have
`const' method qualifier
ast_h323.h: In function `BOOL IsClass(const char *)':
ast_h323.h:72: no method `H323_G7231Capability::Class'
ast_h323.h:72: warning: control reaches end of non-void function
`IsClass(const char *)'
ast_h323.h: At top level:
ast_h323.h:72: virtual outside class declaration
ast_h323.h:72: non-member function `IsDescendant(const char *)' cannot have
`const' method qualifier
ast_h323.h: In function `BOOL IsDescendant(const char *)':
ast_h323.h:72: no method `H323_G7231Capability::Class'
ast_h323.h:72: parse error before `::'
ast_h323.h:72: warning: control reaches end of non-void function
`IsDescendant(const char *)'
ast_h323.h: At top level:
ast_h323.h:72: syntax error before `('
ast_h323.h:77: syntax error before `('
ast_h323.h:79: non-member function `Clone()' cannot have `const' method
qualifier
ast_h323.h:81: syntax error before `*'
ast_h323.h:85: non-member function `GetSubType()' cannot have `const' method
qualifier
ast_h323.h:86: non-member function `GetFormatName()' cannot have `const'
method qualifier
ast_h323.h:88: `H245_AudioCapability' was not declared in this scope
ast_h323.h:88: `pdu' was not declared in this scope
ast_h323.h:90: parse error before `)'
ast_h323.h:90: non-member function `OnSendingPDU(...)' cannot have `const'
method qualifier
ast_h323.h:93: parse error before `'
ast_h323.h:101: parse error before `{'
ast_h323.h:103: virtual outside class declaration
ast_h323.h:103: non-member function `GetClass(unsigned int)' cannot have
`const' method qualifier
ast_h323.h: In function `const char * GetClass(unsigned int = 0)':
ast_h323.h:103: redefinition of `const char * GetClass(unsigned int = 0)'
ast_h323.h:72: `const char * GetClass(unsigned int = 0)' previously defined
here
ast_h323.h: In function `const char * GetClass(unsigned int = 0)':
ast_h323.h:103: `H323EndPoint' undeclared (first use this function)
ast_h323.h:103: parse error before `::'
ast_h323.h:103: no method `MyH323EndPoint::Class'
ast_h323.h:103: warning: control reaches end of non-void function
`GetClass(unsigned int)'
ast_h323.h: At top level:
ast_h323.h:103: virtual outside class declaration
ast_h323.h:103: non-member function `IsClass(const char *)' cannot have
`const' method qualifier
ast_h323.h: In function `BOOL IsClass(const char *)':
ast_h323.h:103: redefinition of `BOOL IsClass(const char *)'
ast_h323.h:72: `BOOL IsClass(const char *)' previously defined here
ast_h323.h: In function `BOOL IsClass(const char *)':
ast_h323.h:103: no method `MyH323EndPoint::Class'
ast_h323.h:103: warning: control reaches end of non-void function
`IsClass(const char *)'
ast_h323.h: At top level:
ast_h323.h:103: virtual outside class declaration
ast_h323.h:103: non-member function `IsDescendant(const char *)' cannot have
`const' method qualifier
ast_h323.h: In function `BOOL IsDescendant(const char *)':
ast_h323.h:103: redefinition of `BOOL IsDescendant(const char *)'
ast_h323.h:72: `BOOL IsDescendant(const char *)' previously defined here
ast_h323.h: In function `BOOL IsDescendant(const char *)':
ast_h323.h:103: no method `MyH323EndPoint::Class'
ast_h323.h:103: parse error before `::'
ast_h323.h:103: warning: control reaches end of non-void function
`IsDescendant(const char *)'
[Asterisk-Users] how to connect 2 TE410P
hi guys, do you have any suggestions on how to connect 2 TE410P via E1? (for simulation and testing purposes) asterisk1 -- TE410P ?-? TE410P --asterisk2
[Asterisk-Users] SayNumber patch for spanish
Hello. Does someone have a patch for SayNumber function (say.c) for numbers in spanish ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem Installing Open H.323 Channel Driver
RTFM. Try reading the build instructions for Open H.323:
http://www.openh323.org/build.html
Jeremy McNamara
Phillip Britt wrote:
Hi,
I am having trouble installing the Open H.323 Channel driver in Asterisk. I
have installed PWLib and OpenH323 V.1.11.7
This is the output l get before it bails out.
asterisk:/usr/src/asterisk/channels/h323# make
g++ -g -c -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DNDEBUG
-DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC -Wmissing-prototypes
-Wmissing-declarations -DP_LINUX -D_REENTRANT -D_GNU_SOURCE
-DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES -DPTRACING
-DP_USE_PRAGMA -I../../include -I/root/pwlib/include/ptlib/unix
-I/root/pwlib/include -I/root/openh323/include -Wno-missing-prototypes
-Wno-missing-declarations ast_h323.cpp
In file included from ast_h323.cpp:28:
ast_h323.h:31: h323.h: No such file or directory
ast_h323.h:32: h323pdu.h: No such file or directory
ast_h323.h:33: mediafmt.h: No such file or directory
ast_h323.h:34: lid.h: No such file or directory
ast_h323.cpp:29: h323t38.h: No such file or directory
In file included from ast_h323.cpp:28:
ast_h323.h:71: parse error before `{'
ast_h323.h:72: virtual outside class declaration
ast_h323.h:72: non-member function `GetClass(unsigned int)' cannot have
`const' method qualifier
ast_h323.h: In function `const char * GetClass(unsigned int = 0)':
ast_h323.h:72: `H323AudioCapability' undeclared (first use this function)
ast_h323.h:72: (Each undeclared identifier is reported only once
ast_h323.h:72: for each function it appears in.)
ast_h323.h:72: parse error before `::'
ast_h323.h:72: no method `H323_G7231Capability::Class'
ast_h323.h:72: warning: control reaches end of non-void function
`GetClass(unsigned int)'
ast_h323.h: At top level:
ast_h323.h:72: virtual outside class declaration
ast_h323.h:72: non-member function `IsClass(const char *)' cannot have
`const' method qualifier
ast_h323.h: In function `BOOL IsClass(const char *)':
ast_h323.h:72: no method `H323_G7231Capability::Class'
ast_h323.h:72: warning: control reaches end of non-void function
`IsClass(const char *)'
ast_h323.h: At top level:
ast_h323.h:72: virtual outside class declaration
ast_h323.h:72: non-member function `IsDescendant(const char *)' cannot have
`const' method qualifier
ast_h323.h: In function `BOOL IsDescendant(const char *)':
ast_h323.h:72: no method `H323_G7231Capability::Class'
ast_h323.h:72: parse error before `::'
ast_h323.h:72: warning: control reaches end of non-void function
`IsDescendant(const char *)'
ast_h323.h: At top level:
ast_h323.h:72: syntax error before `('
ast_h323.h:77: syntax error before `('
ast_h323.h:79: non-member function `Clone()' cannot have `const' method
qualifier
ast_h323.h:81: syntax error before `*'
ast_h323.h:85: non-member function `GetSubType()' cannot have `const' method
qualifier
ast_h323.h:86: non-member function `GetFormatName()' cannot have `const'
method qualifier
ast_h323.h:88: `H245_AudioCapability' was not declared in this scope
ast_h323.h:88: `pdu' was not declared in this scope
ast_h323.h:90: parse error before `)'
ast_h323.h:90: non-member function `OnSendingPDU(...)' cannot have `const'
method qualifier
ast_h323.h:93: parse error before `'
ast_h323.h:101: parse error before `{'
ast_h323.h:103: virtual outside class declaration
ast_h323.h:103: non-member function `GetClass(unsigned int)' cannot have
`const' method qualifier
ast_h323.h: In function `const char * GetClass(unsigned int = 0)':
ast_h323.h:103: redefinition of `const char * GetClass(unsigned int = 0)'
ast_h323.h:72: `const char * GetClass(unsigned int = 0)' previously defined
here
ast_h323.h: In function `const char * GetClass(unsigned int = 0)':
ast_h323.h:103: `H323EndPoint' undeclared (first use this function)
ast_h323.h:103: parse error before `::'
ast_h323.h:103: no method `MyH323EndPoint::Class'
ast_h323.h:103: warning: control reaches end of non-void function
`GetClass(unsigned int)'
ast_h323.h: At top level:
ast_h323.h:103: virtual outside class declaration
ast_h323.h:103: non-member function `IsClass(const char *)' cannot have
`const' method qualifier
ast_h323.h: In function `BOOL IsClass(const char *)':
ast_h323.h:103: redefinition of `BOOL IsClass(const char *)'
ast_h323.h:72: `BOOL IsClass(const char *)' previously defined here
ast_h323.h: In function `BOOL IsClass(const char *)':
ast_h323.h:103: no method `MyH323EndPoint::Class'
ast_h323.h:103: warning: control reaches end of non-void function
`IsClass(const char *)'
ast_h323.h: At top level:
ast_h323.h:103: virtual outside class declaration
ast_h323.h:103: non-member function `IsDescendant(const char *)' cannot have
`const' method qualifier
ast_h323.h: In function `BOOL IsDescendant(const char *)':
ast_h323.h:103: redefinition of `BOOL IsDescendant(const char *)'
ast_h323.h:72: `BOOL IsDescendant(const char *)' previously defined here
ast_h323.h: In function `BOOL IsDescendant(const char *)':
ast_h323.h:103: no method `MyH323EndPoint::Class'
Re: [Asterisk-Users] how to connect 2 TE410P
On Sun, 2003-09-07 at 23:24, Kelvin Chua wrote: hi guys, do you have any suggestions on how to connect 2 TE410P via E1? (for simulation and testing purposes) asterisk1 -- TE410P ? - ? TE410P --asterisk2 T1/E1 cross over cable. Or another possible route without needing the second TE410P card is to use a cross over cable to attach port x to port y of the same card. Let asterisk control port x and then use something like my perl wrapper to make calls via port Y. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Time out Problem-Very Urgent!
hi, I have a problem in call time out, An ISDN PRI E1 from PSTN and another ISDN PRI E1 from a Nortel PBX is conneted to my server. But when i do a Dialout(from both E1s)the calls do not timeout. For ex. Dial(Zap/g4/123456|20|t) suppose other side is ringing and is not answering. even after 20 seconds, call doesn't get timeout pls gv me a solutions.. its really urgent.. Surajee --This mail sent through OmniBIS.com-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem Installing Open H.323 Channel Driver
The issue is not when l am trying to build OpenH323, it is in building the
channel driver in:
/usr/src/asterisk/channels/h323
The README file in there gives some basic clues, but not relating to the
problem l have.
Phil
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy McNamara
Sent: Monday, 8 September 2003 2:34 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Problem Installing Open H.323 Channel Driver
RTFM. Try reading the build instructions for Open H.323:
http://www.openh323.org/build.html
Jeremy McNamara
Phillip Britt wrote:
Hi,
I am having trouble installing the Open H.323 Channel driver in Asterisk.
I
have installed PWLib and OpenH323 V.1.11.7
This is the output l get before it bails out.
asterisk:/usr/src/asterisk/channels/h323# make
g++ -g -c -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DNDEBUG
-DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC -Wmissing-prototypes
-Wmissing-declarations -DP_LINUX -D_REENTRANT -D_GNU_SOURCE
-DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES -DPTRACING
-DP_USE_PRAGMA -I../../include -I/root/pwlib/include/ptlib/unix
-I/root/pwlib/include -I/root/openh323/include -Wno-missing-prototypes
-Wno-missing-declarations ast_h323.cpp
In file included from ast_h323.cpp:28:
ast_h323.h:31: h323.h: No such file or directory
ast_h323.h:32: h323pdu.h: No such file or directory
ast_h323.h:33: mediafmt.h: No such file or directory
ast_h323.h:34: lid.h: No such file or directory
ast_h323.cpp:29: h323t38.h: No such file or directory
In file included from ast_h323.cpp:28:
ast_h323.h:71: parse error before `{'
ast_h323.h:72: virtual outside class declaration
ast_h323.h:72: non-member function `GetClass(unsigned int)' cannot have
`const' method qualifier
ast_h323.h: In function `const char * GetClass(unsigned int = 0)':
ast_h323.h:72: `H323AudioCapability' undeclared (first use this function)
ast_h323.h:72: (Each undeclared identifier is reported only once
ast_h323.h:72: for each function it appears in.)
ast_h323.h:72: parse error before `::'
ast_h323.h:72: no method `H323_G7231Capability::Class'
ast_h323.h:72: warning: control reaches end of non-void function
`GetClass(unsigned int)'
ast_h323.h: At top level:
ast_h323.h:72: virtual outside class declaration
ast_h323.h:72: non-member function `IsClass(const char *)' cannot have
`const' method qualifier
ast_h323.h: In function `BOOL IsClass(const char *)':
ast_h323.h:72: no method `H323_G7231Capability::Class'
ast_h323.h:72: warning: control reaches end of non-void function
`IsClass(const char *)'
ast_h323.h: At top level:
ast_h323.h:72: virtual outside class declaration
ast_h323.h:72: non-member function `IsDescendant(const char *)' cannot have
`const' method qualifier
ast_h323.h: In function `BOOL IsDescendant(const char *)':
ast_h323.h:72: no method `H323_G7231Capability::Class'
ast_h323.h:72: parse error before `::'
ast_h323.h:72: warning: control reaches end of non-void function
`IsDescendant(const char *)'
ast_h323.h: At top level:
ast_h323.h:72: syntax error before `('
ast_h323.h:77: syntax error before `('
ast_h323.h:79: non-member function `Clone()' cannot have `const' method
qualifier
ast_h323.h:81: syntax error before `*'
ast_h323.h:85: non-member function `GetSubType()' cannot have `const'
method
qualifier
ast_h323.h:86: non-member function `GetFormatName()' cannot have `const'
method qualifier
ast_h323.h:88: `H245_AudioCapability' was not declared in this scope
ast_h323.h:88: `pdu' was not declared in this scope
ast_h323.h:90: parse error before `)'
ast_h323.h:90: non-member function `OnSendingPDU(...)' cannot have `const'
method qualifier
ast_h323.h:93: parse error before `'
ast_h323.h:101: parse error before `{'
ast_h323.h:103: virtual outside class declaration
ast_h323.h:103: non-member function `GetClass(unsigned int)' cannot have
`const' method qualifier
ast_h323.h: In function `const char * GetClass(unsigned int = 0)':
ast_h323.h:103: redefinition of `const char * GetClass(unsigned int = 0)'
ast_h323.h:72: `const char * GetClass(unsigned int = 0)' previously defined
here
ast_h323.h: In function `const char * GetClass(unsigned int = 0)':
ast_h323.h:103: `H323EndPoint' undeclared (first use this function)
ast_h323.h:103: parse error before `::'
ast_h323.h:103: no method `MyH323EndPoint::Class'
ast_h323.h:103: warning: control reaches end of non-void function
`GetClass(unsigned int)'
ast_h323.h: At top level:
ast_h323.h:103: virtual outside class declaration
ast_h323.h:103: non-member function `IsClass(const char *)' cannot have
`const' method qualifier
ast_h323.h: In function `BOOL IsClass(const char *)':
ast_h323.h:103: redefinition of `BOOL IsClass(const char *)'
ast_h323.h:72: `BOOL IsClass(const char *)' previously defined here
ast_h323.h: In function `BOOL IsClass(const char *)':
ast_h323.h:103: no method `MyH323EndPoint::Class'
ast_h323.h:103: warning: control reaches end of non-void function
`IsClass(const char *)'
ast_h323.h: At
Re: [Asterisk-Users] MOH other than mp3 ??
For fast hacking - Mplayer plays ASFs nice. Try to make some wrapper(in perl maybe), that makes mplayer looks just like mpg123 for asterisk :) John Brown wrote: On Sat, Sep 06, 2003 at 05:40:28PM -0500, Brian West wrote: Just tell em its ASF.. like the would know the diffrence. The system they use to interface with the web is a pre-made system for the station and we can't touch it. Output is ASF and we can't make it something else. bkw On Sat, 6 Sep 2003, John Brown wrote: On Sat, Sep 06, 2003 at 04:56:22PM -0500, Brian West wrote: You didn't just say MS-ASF Yup I did, ducking under the table... Customer requirement MP3's good.. They rock, but the customer is doing something different and wants to insert ASF formated tunes. bkw On Sat, 6 Sep 2003, John Brown wrote: is there a clean way to have MOH (Music on Hold) source its audio from say a MS-ASF streem ??? got a radio station that wants to have their MOH come from their ASF based netbroadcast ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phone - Asterisk - SIP LD Provider question
Peter Pauly wrote: If Asterisk registers with a SIP long distance provider and I make a call from an IP phone through Asterisk to that LD provider, does the RTP (audio) traffic flow between the two end points directly (normally the IP phone and the LD provider) or does it flow through Asterisk? I'm asking because I have Asterisk running behind a NAT firewall along with an IP Phone (software) and I'm trying to get it working with Iconnecthere (ICH). I am able to register, connect , but no audio. I have ports opened up on the firewall, but they point to the Asterisk machine and not the IP phone machine. In this scenario, any audio traffic would have to go through the asterisk box to reach the IP phone. Is that how it works? SIP control connection usualy goes thru firewall. RTP - no. Just put the Asterisk on the machine with the firewall and it will work. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410 - 3.3v?
If it fits, it'll work. Generally motherboards with 64-bit slots and most DELL's including the 600SC work fine. mark On Mon, 8 Sep 2003, Garry Adkins wrote: I just noticed that the new TE410 only supports 3.3v PCI. How do you know if it will work in your machine? It's not something commonly listed in your motherboard specs I was thinking about getting one for an Athlon 750 that is a spare box I'm doing testing on... (Original SlotA) -G ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MOH other than mp3 ??
They have 5 different stations under one company. The idea is to have 5 different MOH sources (URL based) and depending on who/what/when/where placed on hold they would get the right stations current broadcast. So I don't think the cost/value is there for multiple Zap-Channel devices, or multiple sound cards You are not going to get an ASF feed unless you do it yourself. Trust me. This is primarily because ASF could mean anything. Much like .wav files and .avi files, ASF files are only a wrapper that may contain various encodings. If I had to guess, I'd say you're likely going to need to decode the ASF and then decode WMA. The AVIfile library has some support for this. Given the trouble above, I would REALLY just recommend finding some card that will handle like four audio inputs, get the SOX moh code mentioned earlier, and hack together some process for pulling the data from the sound card via sox. IIRC, something like this has been done before by someone on IRC. I have nothing to back that up. Does anyone have any specs on building a converter to take the sound input from, say, a stereo plug and putting it onto a phone line? Is there a box I can buy at radio shack? At that point, it would be a matter of patching MOH to bridge to a zaptel channel (probably the best way to do this). Jayson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ser vs Asterisk?
Rich Adamson wrote: Could someone give me a 10,000 foot view of what the differences are between Ser and Asterisk? I'd like to implement one or the other handle a small number of local ip phones, tie a couple of asterisk (or ser) machines together across the Internet, implement a couple of FX gateways (to handle incoming pstn calls, and for outgoing pstn calls), and use features mostly common to pbx's. No immediate need for CDR. Voice mail, callerid, etc, are wanted. Would like to accept incoming sip calls from anyone on the Internet that might choose to call. Would Ser or Asterisk be the most appropriate choice? Rich I using both in heavy production enviroment. SER is the BEST SIP proxy that i found. But it is just sip proxy. And can serve a _LOT_ connections (10,000 users, 20 cals per second). Asterisk is more like telephone switch with lot of features, but far slower. In your scenario - Asterisk. SIP cannot act as a PSTN Gateway (PC with some telephone board). Mixed scenario - Voice mail, PSTN GWs, Conference ... - Asterisk. Call routing - SIP. You can implement SER only scenario, if you use Hardware Gateways - Cisco AS5350. But i don't recommend you to use HW Gateways - main problem is that this gateways still don't support the speex codec, so if you make long distance calls between, let say, AS5350 and Asterisk you can't use low bandwidth codec. . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to connect 2 TE410P
neat! actually we are just in the process of planning for an asterisk based simulation lab for the university. do you have a cable pin-out descriptions for that purpose? thanks! ~kelvin - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, September 08, 2003 12:52 PM Subject: Re: [Asterisk-Users] how to connect 2 TE410P On Sun, 2003-09-07 at 23:24, Kelvin Chua wrote: hi guys, do you have any suggestions on how to connect 2 TE410P via E1? (for simulation and testing purposes) asterisk1 -- TE410P ? - ? TE410P --asterisk2 T1/E1 cross over cable. Or another possible route without needing the second TE410P card is to use a cross over cable to attach port x to port y of the same card. Let asterisk control port x and then use something like my perl wrapper to make calls via port Y. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Time out Problem-Very Urgent!
hi, I have a problem in call time out,An ISDN PRI E1 from PSTN and another ISDN PRI E1 from a Nortel PBX is conneted to my server.But when i do a Dialout(from both E1s)the calls do not timeout.For ex. Dial(Zap/g4/123456|20|t) suppose other side is ringing and is not answering.even after 20 seconds, call doesn't get timeout pls gv me a solutions..its really urgent.. Surajee
