Re: [Asterisk-Users] suggested hardware especially sound cards
Armand A. Verstappen wrote: I think we should have these setups listed: - home user with 1-2 telco lines and 2-5 phones - small office with 4-8 telco lines and 8-16 phones - small office with a fractional E1/T1 and 12-24 phones - medium office with full E1/T1 and 24-48 phones - medium office with 2-4 E1/T1s and 48-100 phones - large office with 4-16 E1/T1s and 100-500 phones - multi-location corporate offices with 16-64 E1/T1s distributed and 500-2500 phones - ACD heavy office suggestions - IVR or Conference heavy suggestions You can add a section this to the wiki (http://www.voip-info.org/wiki-Asterisk), and fill out the suggestions you have information for, then invite others to complete the others. All you need to do is register, which is free. http://www.voip-info.org/tiki-index.php?page=Asterisk+hardware+recommendations I've just added the meny above, the rest is open for addition. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Web Voicemail Permissions
Tilghman Lesher wrote: On Monday 06 October 2003 05:13 pm, Carlton J. O'Riley wrote: Are there any plans to incorporate the running of Asterisk as a non-root user into the current CVS? There is nothing in Asterisk that requires root access as far as I know and this would solve the vmail.cgi script permissions problem. Here's a reason why it might need to run as root: bash# ls -l /dev/zap/ctl crw-r--r--1 root root 196, 0 Oct 6 13:15 /dev/zap/ctl We need to open some ports for listening as root, but after that we can change user ID the way other daemons do. Tilghman, can we handle this ctl device as another user after we opened it? I agree that it would be good to have Asterisk running with another user ID. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 802.11 phone review: WiSIP
At 10:42 PM -0500 10/3/03, Steven Critchfield wrote: On Fri, 2003-10-03 at 20:11, Masakazu Nakano wrote: I found it. but that webite is chinese BIG-5. take care. http://www.mpn.com.tw/index-big5-PRODUCT.html The WiFi600 described in the above URL is the device I currently have, and on which I wrote my review. They are the same unit. and that already released by Fujitsu. http://www.net-2com.com/jp/product/hw/wireless_ipphone/ The Fujitsu is also the WiFi600, the device I have, and it's not made by Fujitsu. Note that the pictures of the device in my article have Japanese characters on the keys, probably due to lack of new silkscreening before they shipped the first US models. Actually, this is the one I was referring to http://www.symbol.com/products/wireless/netvisionphoneds.html [snip] I wish the Symbol NetVision supported SIP, but it doesn't as far as I've been able to find. Anyone who has new data is welcome to update me, or to ship me a demo unit for evaluation. :-) To answer a few more questions that were sent to me: - I don't know if transfer works; I have been unable to find any key that enables a SIP transfer action - Same goes for hold - maybe it works, but I don't know how to make the magic, since the manual was so small. Updates to my review: SIP password authentication with Asterisk now works with latest rev of code. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Web Voicemail Permissions
On Tuesday 07 October 2003 01:23, Olle E. Johansson wrote: Tilghman Lesher wrote: On Monday 06 October 2003 05:13 pm, Carlton J. O'Riley wrote: Are there any plans to incorporate the running of Asterisk as a non-root user into the current CVS? There is nothing in Asterisk that requires root access as far as I know and this would solve the vmail.cgi script permissions problem. Here's a reason why it might need to run as root: bash# ls -l /dev/zap/ctl crw-r--r--1 root root 196, 0 Oct 6 13:15 /dev/zap/ctl We need to open some ports for listening as root, but after that we can change user ID the way other daemons do. None of the ports are below 1024, so root access is not needed to bind them. Tilghman, can we handle this ctl device as another user after we opened it? Check with Mark. Also, note that there's no guarantee that some kernel developer might think this is a bad idea (read: security hole) and disallow it in some future version. I agree that it would be good to have Asterisk running with another user ID. If you're that concerned about it, why not use the NSA kernel with ACLs? It would probably be even better served if you worked to secure the entire execution environment (e.g. chroot, ACLs, etc.) instead of just changing the uid. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Let's TALK ABOUT IT!!!
It is that type of mechanism that enum uses and yes it was to solve a similar goal, but in this case you need a 'route server' type system - in particular as this is for IP routing of PSTN end points not on an IP network. A discussion about this came up a while ago. I suggested something along the lines of BGP, where each endpoint announces prefixes of what they can get to. You'll need a central machine that everyone peers up with and then you can use a switch = statement or exten = _.,1,Dial in * to query that machine and get the best route for your call. If you make sure that your destination machines are not behind NAT or a firewall, you can do an IAX handoff to get the connection set peer to peer instead of through the central server. Example: 4 remote * machines, each configured with our BGP software. Machine 1 announces that it can terminate calls to country code 1 with a cost of .02. Machine 2 announces that it can terminate calls to 1 with a cost of .05. Machine 3 announces that it can terminate calls to 1-830 with a cost of 0. Machine 4 announces that it can terminate calls to 1-830-751 with a cost of 0. You place a call to 1-830-751-2000 and the system determines that it can place that call for a cost of 0 to machine 4. You place a call to 1-240-988-4000 and the system determines that it can place that call via either machine 1 or 2, but lowest cost is machine 1. [general summary to all branches of this thread] Yes, that describes TRIP (RFC3219 - http://www.zvon.org/tmRFC/RFC3219/Output/index.html) fairly accurately. While not having quite a central machine with which everyone peers, it may be that each ITAD (Internet Telephony Administrative Domain, like an ASN) would have one main router to which all their local Asterisk servers would be leafs, and then that core router would peer with other core routers at other ITADs or maybe some large IRR-like servers which were clearinghouse-only style route distributors. I offered money here on this list previously to anyone who thinks that they're qualified to develop and integrate a TRIP implementation into Asterisk. Warning: it's not a trivial issue, and I will only consider programmers with a full understanding of the magnitude of the task. This could threaten to be a surprisingly large mesh with possibly hundreds of thousands or millions of routes of an extremely dynamic nature, and such an implementation is not for the beginner. I'm still taking applications. In other notes: I saw in other parts of this thread the discussion about how to do number routing via DNS. This is a good idea, so good, in fact, that it already exists in Asterisk and is a set of RFCs. It's called ENUM, and it routes phone numbers via the DNS. See enum.conf and show application EnumLookup - the good folks at nic.at were kind enough to pay for and work on these improvements to Asterisk. ENUM is great, but it's going slowly as far as the hyper adoption rates of Internet time are any comparison. The main issues seem to be political, since the triple whammy of ownership, authorization, and administration seem to be in the way. If you are in a country that hates VoIP, don't expect to see above-board ENUM any time in the near future. :-( BUT: The nice thing about ENUM, especially in Asterisk (and hopefully soon in SER) is that one can specify cascading trees in which to look up data that are not necessarily e164.arpa. as the root. I will leave it to the reader to figure out why this is a good thing and a bad thing at the same time. ENUM and TRIP provide DIFFERENT functions: ENUM gives out exact answers, and TRIP provides gateway answers. First, you look up the number in ENUM. Is there an answer? If so, send call to that SIP/H323/IAX gateway. If no answer, then look up the number in TRIP and find someone who has a cheap/good/fast/whatever gateway to that particular number range, and send the call to that SIP/H323/IAX/etc. gateway. In fact, I had a really nasty thought the other day: make a DNS resolver hack that allows ENUM lookups to incorporate TRIP replies. Yuck, yuck, yuck... but it would allow TRIP integration into any system that supports ENUM with no additional work on the telephony client side. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ChanIsAvail app setting ${AVAILCHAN} to an unusable value.
On Sunday 05 October 2003 16:41, Robert Hajime Lanning wrote:
I sent this earlier under Editting variable contents but no-one
has responded. So, the subject is now more to the problem, instead
of the solution I was trying to implement.
ChanIsAvail returns the channel ID plus -session.
How can I edit ${AVAILCHAN} to remove this session ID, so I can use
its contents in a subsequent Dial statement?
Oh, it's quite simple. You just write your own application to remove
the suffix. Or you wait for someone else to write it.
Untested code. UAYOR.
-Tilghman
Attachment converted: PrivateSpace:app_cut.c (TEXT/ttxt) (69546196)
I don't recall if -session is a fixe number of digits. If so, you
can use the string manipulation features within Asterisk to cut it
off. I don't have the manual reference right here with me, but note
that you can put negative numbers for ${EXTEN:-1:-3} and the like,
which will chop things up based on fixed positions within the string.
JT
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ChanIsAvail app setting ${AVAILCHAN} to anunusable value.
quote who=John Todd
On Sunday 05 October 2003 16:41, Robert Hajime Lanning wrote:
I sent this earlier under Editting variable contents but no-one
has responded. So, the subject is now more to the problem, instead
of the solution I was trying to implement.
ChanIsAvail returns the channel ID plus -session.
How can I edit ${AVAILCHAN} to remove this session ID, so I can use
its contents in a subsequent Dial statement?
Oh, it's quite simple. You just write your own application to remove
the suffix. Or you wait for someone else to write it.
Untested code. UAYOR.
-Tilghman
I don't recall if -session is a fixe number of digits. If so, you
can use the string manipulation features within Asterisk to cut it
off. I don't have the manual reference right here with me, but note
that you can put negative numbers for ${EXTEN:-1:-3} and the like,
which will chop things up based on fixed positions within the string.
JT
Not fixed length. Well it maybe fixed per technology. (Zap vs. SIP...)
I ended up just writing an AGI script.
extensions.conf:
; Now we dial
exten = 8901,6,AGI(strip-sess,DIALCHANS)
exten = 8901,7,Macro(stdexten,8901,${DIALCHANS})
-
#! /usr/bin/perl
$|=1;
$variable = shift;
while ($line = STDIN,$line =~ /[^ \n\r]/) { }
print STDOUT GET VARIABLE $variable\n;
$response = STDIN;
$response =~ /^\d+ +result=(\d+) +\((.*)\)\s*$/;
$response = $1;
$data = $2;
if ($response == 1) {
$data = join(,map {$_ =~ s/\-\w+$//;$_;} split(//,$data));
print STDOUT SET VARIABLE $variable \$data\\n;
$response = STDIN;
}
exit(0);
-
--
END OF LINE
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: DB virtualization for multiple database support - Was Re: [Asterisk-Users] How to use vmdb.sql in voicemail.conf/extension.conf
If you write code in C/C++, you'd better use sqlrelay : http://sqlrelay.sourceforge.net/ SQL Relay is a persistent database connection pooling, proxying and load balancing system for Unix and Linux supporting ODBC, Oracle, MySQL, mSQL, PostgreSQL, Sybase, MS SQL Server, IBM DB2, Interbase, Lago and SQLite with APIs for C, C++, Perl, Perl-DBI, Python, Python-DB, Zope, PHP, Ruby, Ruby-DBI, TCL and Java, command line clients, a GUI configuration tool and extensive documentation. The APIs support advanced database operations such as bind variables, multi-row fetches, client side result set caching and suspended transactions. It is ideal for speeding up database-driven web-based applications, accessing databases from unsupported platforms, migrating between databases, distributing access to replicated databases and throttling database access. :-) Garry Adkins wrote: Not familiar with it... You have a URL? - Original Message - From: Matteo Brancaleoni [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 05, 2003 4:52 PM Subject: [Asterisk-Users] Re: DB virtualization for multiple database support - Was Re: [Asterisk-Users] How to use vmdb.sql in voicemail.conf/extension.conf Like what PEARS (php libs) do for db backends? Matteo. Garry Adkins wrote: I am trying a scenerio where the * will take the email and mailbox number from the Mysql database for sendming mail to a voicemail user. I have seen vmdb.sql file but is not able to determine its use. You can't anymore MySQL was ripped from Asterisk because the client libs are GPL. I would be more than happy to help write a DB Virtualization function for *. I *love* the way it works in Java, but that's not a real possibility. It wouldn't need to be as complicated as JDBC but it's a nice model. We could however: 1) Abstract out the schema from the database calls 2) Develop a pluggable driver interface to translate to whatever DB you're using. This way... You want MySQL, you develop a translation driver that maps * db calls to MySql. (fairly trivial) Same for Postgres (I'd suggest making this the default, as no GPL issues for mark, etc.) Same for Oracle, etc. ___ 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] Message Waiting on Cisco 7960
Jaun, In your sip.conf try changing the [EMAIL PROTECTED] to [EMAIL PROTECTED] My MWI works rock solid now almost instantaneously coming on and off. Babak Juan J. Sierralta P. wrote: On Fri, 2003-10-03 at 14:53, Babak Pasdar wrote: This issue was resolved by adding the @context in the voicemail.conf file for the extension to the mailbox=XXX command. [EMAIL PROTECTED] Thanks so much for your help. Is there anything special I need to configure on the Cisco phone to get MWI ? Because a I have a 7960 hanging from asterisk and I have followed all the suggestions here and I have not MWI on the phone. Here are my confs: --- extensions.conf - [demo] exten = 8991,1,Dial(SIP/8991,20) exten = 8991,2,Voicemail2([EMAIL PROTECTED]) exten = 8991,102,Voicemail2([EMAIL PROTECTED]) exten = 8991,103,Hangup voicemail.conf - [demo] 8991 = 8991,Juanjo,[EMAIL PROTECTED] --- sip.conf [8991] type=friend username=8991 secret= nat=no ; This phone may be natted host=dynamic canreinvite=no ; Cisco poops on reinvite sometimes qualify=500 ; Qualify peer is no more than 200ms away ;defaultip=192.168.0.15 context=demo [EMAIL PROTECTED] Any sugestions will be appreciated. -- Juanjo sin .sig ___ 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] Snom100 H.323 sample config
Tilghman Lesher schrieb: I'm trying to get a Snom100 configured with H.323. Right now, the ... Hi, I had Snoms (100 and 200) configured with H.323 working with asterisk-0.4.0. Since I upgraded to asterisk-0.5.0 and I had problems with H.323 I switch to sip. (The problem was: when I transfered from one H.323 phone to another, chan_oh323 crashed and I had to restart asterisk. But when a gatekeeper avoids that situation, it works fine - I tried with gnugk.) But I still have the config file. The important things are: fastStart=no h245Tunnelling=no h245inSetup=yes Same settings for the phone! For the rest you can take the example config from chan_oh323. Hope, that helps. Roger. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with questions for initial Asterisk wizard (GUI)
Hi Leif Very good Idea Everything you have wrote is right!!! Many thanks Dimitri On Tuesday 07 October 2003 04:59, Leif Madsen wrote: Hey all, I am in the middle of creating a new user wizard which will generate all the .conf's the new Asterisk user will require to get themselves up and running in Asterisk without having to touch a single configuration file. This is what I have come up with as a rough draft. It is far from complete, so I'm asking people to submit things that should be added, changed, removed etc.. etc... so please help me come up with a good logic for the questions so that I may start the work on the actual wizard! Thanks in advance for all your help! Hardware Are you using Digium Hardware? (yes/no) + TE410P + How many? + TDM400P + How many? + How many modules (each card) + T400P + How many? + T100P + How many? + E100P + How many? + X100P + How many? Other hardware + Do you have a soundcard? (yes/no) + ALSA or OSS? Do you have any SIP devices? (yes/no) + SIP phone + Softphone Extension Ranges + Start and Stop range + Would like you to enable voicemail on any of these extensions? (yes/no) + all of them (yes/no) + which ones? + What should the default voicemail password be? + Default formats for writing voicemail + GSM, wav49, WAV + Email notification? (yes/no) + Who should the email appear to come from? + Should we attach it to the email? + Would you like to specify a maximum message length? (yes/no) + How long? + Would you like to specify a maximum greeting length? (yes/no) + How long? + Which country are you in? (for indication) + United States + Australia + France + Netherlands + United Kingdom + Which language? (for zapata.conf) Would you like to activate any of these extensions now? (yes/no) + List extensions with CONFIG | EDIT | DELETE | ADD links + Who is going to use this extension? (name) + What is the email address of the person as this extension? (for email notification) + For each channel of the hardware the user has + Which signalling for this channel? kewl start loop start ground start + Enable three way calling? + Enable transfer? + Enable call waiting? + Enable busy detection? + Use CallerID? + rxgain + txgain + Immediate? (yes/no) + CallerID String + Name + Number + Enable mailbox indication? + Mailbox number(s) to be associated with this channel + Context Would you like to setup registration for FWD, IAXtel, SIPPhone or iptel? + -- +--+ |Leif Madsen - http://www.hacklocalhost.com| +--+ |@| leif at hacklocalhost dot com | | SMS| sms at hacklocalhost dot com | | FWD| 18924 IAX| 1700-363-0761 | |iptel| 8972-1969sipph| 1-747-386-1618 | +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with questions for initial Asterisk wizard (GUI)
On Mon, 2003-10-06 at 23:59, Leif Madsen wrote: Hey all, I am in the middle of creating a new user wizard which will generate all the .conf's the new Asterisk user will require to get themselves up and running in Asterisk without having to touch a single configuration file. This is what I have come up with as a rough draft. It is far from complete, so I'm asking people to submit things that should be added, changed, removed etc.. etc... so please help me come up with a good logic for the questions so that I may start the work on the actual wizard! Maybe it's me, but only the most basic of setups will be handled by a wizard. The person who does the setup necessarily needs to know a fair amount about routing of calls. Thanks in advance for all your help! Hardware Are you using Digium Hardware? (yes/no) + TE410P + How many? + TDM400P + How many? + How many modules (each card) + T400P + How many? + T100P + How many? + E100P + How many? + X100P + How many? This section should be expanded to know about ISDN and signaling should really be combined with this section since you only use kewlstart/groundstart/loopstart when actually touching the PSTN interfaces. Also you should either interrogate the system to know how it plans to load the drivers so you know which order the channels will be in. Other hardware + Do you have a soundcard? (yes/no) + ALSA or OSS? Do you have any SIP devices? (yes/no) + SIP phone + Softphone Does this make a difference as to whether it is software or not? As a side note, you might want to disable the sip module if it isn't going to be used to help close down unused ports. If you plan to coddle a user, might as well do it right. Extension Ranges + Start and Stop range + Would like you to enable voicemail on any of these extensions? (yes/no) + all of them (yes/no) + which ones? + What should the default voicemail password be? + Default formats for writing voicemail + GSM, wav49, WAV + Email notification? (yes/no) + Who should the email appear to come from? + Should we attach it to the email? + Would you like to specify a maximum message length? (yes/no) + How long? + Would you like to specify a maximum greeting length? (yes/no) + How long? Where do you plan on connecting these with actual channels? Also do you plan on building out menu generation. Don't forget to explain to the user your coddling about significant digits, ie. for x number of extensions you will probably need to extend 1 more digit so they all start with the same digit, or only consume 1 or 2 significant digits since one or more will also be used to signify outbound dialing and any number of other lead digits might need to be used in other functions like meetme apps. Might as well add configuration of outbound prefixes for dialing. Ready to tackle all those international codes too. I'm sure our euro friends will love it if you get those patterns all down for them. + Which country are you in? (for indication) + United States + Australia + France + Netherlands + United Kingdom + Which language? (for zapata.conf) Would you like to activate any of these extensions now? (yes/no) + List extensions with CONFIG | EDIT | DELETE | ADD links + Who is going to use this extension? (name) + What is the email address of the person as this extension? (for email notification) + Shouldn't this be folded into the extension setup? For each channel of the hardware the user has + Which signalling for this channel? kewl start loop start ground start + Enable three way calling? + Enable transfer? + Enable call waiting? + Enable busy detection? + Use CallerID? + rxgain + txgain + Immediate? (yes/no) + CallerID String + Name + Number + Enable mailbox indication? + Mailbox number(s) to be associated with this channel + Context This should be folded into channel setup. Of course this is where it gets tricky as you can't specify a mailbox until you specified extensions and worked out the problems with the extension logic. Would you like to setup registration for FWD, IAXtel, SIPPhone or iptel? + Set up Meetme apps, upload and use voice
[Asterisk-Users] Voicetronics
Title: Message has anyone got a voicetronics openline4 card working ?? If so do you have any notes etc. thanks in advance. Regards Mick
[Asterisk-Users] Vioce Modems
Title: Leterhead Hi I am a newbie and just set up my first Asterisk box. I have got 2 x Grandstream 101s working as extensions and am now looking to get to the outside world. Q.) Can you use a voice/fax modem as an FXO interface? If yes, then how would I configure it. Regards Dave Registered Office: - 23 First Street, Low Moor, Bradford, West Yorkshire, BD12 0JQ. Company Registration Number: - 03807643. VAT Registration Number: - 734-3363-42 Telephone / Fax: - 44 (0) 7092 154039. SIP_Phone: - 1 (747)669 1957 http://www.codepipe.ltd.uk / http://www.codepipe.com / E-Mail: - [EMAIL PROTECTED]
Re: [Asterisk-Users] direct-inward-dialing (DID)
Example: 456-7000 is your main number and you have 7001 to 7099 as DIDs:
exten = 7000,1,Goto(AutoAttendant|s|1)
exten = _7XXX,1,Macro(yourdialmacro|${EXTEN})
How are you dropping the 456 there? I thought extensions picked up what
either the SIP phone had dialled, or what DTMF detection picked up when *
answered the line...?
I'm looking at purchasing a PRI with 30 DIDs (can't get any fewer from Bell
Canada) and routing the calls coming in to multiple remote * boxes based on
the called number. I was going to ask a question similar to John's but
just didn't get around to it yet. :-)
If you could explain in a little more detail how you turn the CNID into an
extension I'd really appreciate it.
Andrew
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicetronics
Title: Message got the voicetronics openline4 card sort of working I keep getting this error -- Event [7=[03] Record fifo overflow] on vpb/1-4 and the auto attendant is not clear. thanks in advance Regards Mick
Re: [Asterisk-Users] direct-inward-dialing (DID)
The number of digits that your telco sends to you is a configurable figure
(at least it is here in Aus). The example assumes that the telco is sending
you the last 4 digits.
Paul
Example: 456-7000 is your main number and you have 7001 to 7099 as DIDs:
exten = 7000,1,Goto(AutoAttendant|s|1)
exten = _7XXX,1,Macro(yourdialmacro|${EXTEN})
How are you dropping the 456 there? I thought extensions picked up what
either the SIP phone had dialled, or what DTMF detection picked up when *
answered the line...?
I'm looking at purchasing a PRI with 30 DIDs (can't get any fewer from
Bell
Canada) and routing the calls coming in to multiple remote * boxes based
on
the called number. I was going to ask a question similar to John's but
just didn't get around to it yet. :-)
If you could explain in a little more detail how you turn the CNID into an
extension I'd really appreciate it.
Andrew
___
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] Help with questions for initial Asterisk wizard (GUI)
On Tue, Oct 07, 2003 at 04:40:36AM -0500, Steven Critchfield wrote: What point do you feel that a user is too advanced to us your wizard, or at what point do you think a user of your wizard will be more pissed at being hindered by the product than helped? I'm not trying to insult you, or necessarily put down what you want to do. I just feel that it is way to simplistic to think a wizard will make anyone happy but a small fraction of users. If you -- Steven Critchfield [EMAIL PROTECTED] Good suggestions for changing his program. I strongly believe this kind of coddling will be helpfull to many more people that you expect. First, a large number of people are coming in with the I saw * and want to try to do a ??? phone or I have a X100P and want to do. Those are realy (according to you old timers) trivial. Most of them are looking for help getting started...here is that start. Many will only need to run through a simple config and can them look at the conf files for more advanced setups. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with questions for initial Asterisk wizard (GUI)
What about the following device: USB FXS device (S100U)? This sounds like a nice thing. What language are you using? I know this would be adding a burden to coding, but can you also make the app be data driven so any future additions or new hardware would just be added to a a text file? (I know, this would be making another .conf file.) Thanks -- Original Message -- From: Leif Madsen [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Tue, 07 Oct 2003 00:59:17 -0400 Hey all, I am in the middle of creating a new user wizard which will generate all the .conf's the new Asterisk user will require to get themselves up and running in Asterisk without having to touch a single configuration file. This is what I have come up with as a rough draft. It is far from complete, so I'm asking people to submit things that should be added, changed, removed etc.. etc... so please help me come up with a good logic for the questions so that I may start the work on the actual wizard! Thanks in advance for all your help! Hardware Are you using Digium Hardware? (yes/no) + TE410P + How many? + TDM400P + How many? + How many modules (each card) + T400P + How many? + T100P + How many? + E100P + How many? + X100P + How many? Other hardware + Do you have a soundcard? (yes/no) + ALSA or OSS? Do you have any SIP devices? (yes/no) + SIP phone + Softphone Extension Ranges + Start and Stop range + Would like you to enable voicemail on any of these extensions? (yes/no) + all of them (yes/no) + which ones? + What should the default voicemail password be? + Default formats for writing voicemail + GSM, wav49, WAV + Email notification? (yes/no) + Who should the email appear to come from? + Should we attach it to the email? + Would you like to specify a maximum message length? (yes/no) + How long? + Would you like to specify a maximum greeting length? (yes/no) + How long? + Which country are you in? (for indication) + United States + Australia + France + Netherlands + United Kingdom + Which language? (for zapata.conf) Would you like to activate any of these extensions now? (yes/no) + List extensions with CONFIG | EDIT | DELETE | ADD links + Who is going to use this extension? (name) + What is the email address of the person as this extension? (for email notification) + For each channel of the hardware the user has + Which signalling for this channel? kewl start loop start ground start + Enable three way calling? + Enable transfer? + Enable call waiting? + Enable busy detection? + Use CallerID? + rxgain + txgain + Immediate? (yes/no) + CallerID String + Name + Number + Enable mailbox indication? + Mailbox number(s) to be associated with this channel + Context Would you like to setup registration for FWD, IAXtel, SIPPhone or iptel? + -- +--+ |Leif Madsen - http://www.hacklocalhost.com| +--+ |@| leif at hacklocalhost dot com | | SMS| sms at hacklocalhost dot com | | FWD| 18924 IAX| 1700-363-0761 | |iptel| 8972-1969sipph| 1-747-386-1618 | +--+ -- +--+ |Leif Madsen - http://www.hacklocalhost.com| +--+ |@| leif at hacklocalhost dot com | | SMS| sms at hacklocalhost dot com | | FWD| 18924 IAX| 1700-363-0761 | |iptel| 8972-1969sipph| 1-747-386-1618 | +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Costas Menico Meezon Software Corp 201-224-8111 [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] direct-inward-dialing (DID)
The number of digits that your telco sends to you is a configurable figure (at least it is here in Aus). The example assumes that the telco is sending you the last 4 digits. Hmmm ok so DIDs are not what is stuffed into the CNID field? Or rather pieces of the DID make it into the CNID? Confused, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digium FXO
Is it possible to send an external hookflash command to the Digium FXO card from the asterisk PBX?
Re: [Asterisk-Users] Help with questions for initial Asterisk wizard (GUI)
This is a great idea. I have a good understanding of Asterisk but would use this for initial setup if it were quick and easy, then go in and tweak the settings. This is especially good for a client that would like total ownership and admin over the product but do not have the time or desire to ascend the fairly steep learning curve of Asterisk. Most network admins I know don't like having an outsider come in to admin a machine on their network (myself included). You will experience alot of negativity from people that would like to see themselves as the Best or the Master of anything and feel threatened if something promises to take that away from them. This is their crutch to feel Special in the world. To that I say Ba! Thanks, Steve Totaro - Original Message - From: PJ Welsh [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, October 07, 2003 5:47 AM Subject: Re: [Asterisk-Users] Help with questions for initial Asterisk wizard (GUI) On Tue, Oct 07, 2003 at 04:40:36AM -0500, Steven Critchfield wrote: What point do you feel that a user is too advanced to us your wizard, or at what point do you think a user of your wizard will be more pissed at being hindered by the product than helped? I'm not trying to insult you, or necessarily put down what you want to do. I just feel that it is way to simplistic to think a wizard will make anyone happy but a small fraction of users. If you -- Steven Critchfield [EMAIL PROTECTED] Good suggestions for changing his program. I strongly believe this kind of coddling will be helpfull to many more people that you expect. First, a large number of people are coming in with the I saw * and want to try to do a ??? phone or I have a X100P and want to do. Those are realy (according to you old timers) trivial. Most of them are looking for help getting started...here is that start. Many will only need to run through a simple config and can them look at the conf files for more advanced setups. ___ 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] Anyone else use Audacity for prompts?
Why compress all your prompts to .gsm files? Isn't * going to have to reformat them anyway based on the codec being used for the call? I have all my voice prompts as 8khz/16bit .wav files (* can't seem to play back 8 bit files). I recorded them through soundforge as a 48Khz/16bit mono .wav - did a little tweaking to compress and brighten them up - then resampled to 8khz. Quality is as good as any I've heard from any commercial PBX... It is important to me, if I'm going to use * for my business, that the voice prompts sound as clean and clear as any other system - from a marketing/PR standpoint. Dave Redmore ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can AGI be used in this way?
Hi all, I'm about to build a basic browser based call management module with some basic functions and was wondering if I can use AGI in the following way: browser app -- calls perl script containing AGI stuff -- controls asterisk. The most important task I'm hoping to integrate is call transfer via web browser. If not, are there any other ways to do this? My web app is built with CFML but I can access/interact with any of the following: linux shell - can pass any command/arg combos; java libraries; perl libraries; php libraries TIA, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium FXO
I think that app_flash should do the trick : astro*CLI -= Info about application 'Flash' =- [Synopsis]: Flashes a Zap Trunk [Description]: Flash(): Sends a flash on a zap trunk. This is only a hack for people who want to perform transfers and such via AGI and is generally quite useless otherwise. Returns 0 on success or -1 if this is not a zap trunk btw, I never used it really. Matteo. Il mar, 2003-10-07 alle 15:07, Kevin ha scritto: Is it possible to send an external hookflash command to the Digium FXO card from the asterisk PBX? -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connect with another PBX
Hi, I would like to connect my * with another PBX. Which card should I use ?? X100P, TDM400P, etc... Which signalling ??? Does Anyone knows any tutorial or samples config on web that focus this problem... Thanks In Advanced Andre Lomonaco ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can AGI be used in this way?
On Tue, 2003-10-07 at 08:37, Dave Wilson wrote: Hi all, I'm about to build a basic browser based call management module with some basic functions and was wondering if I can use AGI in the following way: browser app -- calls perl script containing AGI stuff -- controls asterisk. agi is for call control from within the call, not external. You need the manager interface. The most important task I'm hoping to integrate is call transfer via web browser. If not, are there any other ways to do this? My web app is built with CFML but I can access/interact with any of the following: linux shell - can pass any command/arg combos; java libraries; perl libraries; php libraries I think someone here has perl manager libraries available. Use google. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] direct-inward-dialing (DID)
On Tue, 2003-10-07 at 07:53, Andrew Kohlsmith wrote: The number of digits that your telco sends to you is a configurable figure (at least it is here in Aus). The example assumes that the telco is sending you the last 4 digits. Hmmm ok so DIDs are not what is stuffed into the CNID field? Or rather pieces of the DID make it into the CNID? The telco will send x(configurable) digits as CNID on a PRI. This becomes the extension inside of asterisk automagically. I know you can have anywhere from 3 digits up to the entire 10 digits sent to you. While my company was on channelized em t1 we had 4 digits of DID, and now on PRI we get the whole 10 digit number. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can AGI be used in this way?
I think someone here has perl manager libraries available. Use google. -- Thanks Steven. Plenty of samples from google. Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Message Waiting on Cisco 7960
On Mon, 2003-10-06 at 23:05, Brian West wrote: use mailbox=500 instead of [EMAIL PROTECTED] [EMAIL PROTECTED] Thanks guys, changing voicemail by mailbox did the trick ! -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Iconnect Incomming calls
I have an IconnectHere account
with a Inbound number and have setup the sip.conf to register and am recieving
the call but When I answer the call it disconnect. I have tried sending the call
to from * to a Softphone, Pingtel, and FXS port and all result the same. As soon
as I accept the call it disconnects. I believe it may be some type of codec
issue but I am not very familiar with that layer.
Below is the .conf's SIP
debug
Thank for any
help
; SIP Configuration for
Asterisk;[general]port =
5060
; Port to bind tobindaddr =
0.0.0.0
; Address to bind tocontext =
sipinbound ;
Default for incoming callsregister =
1410344:[EMAIL PROTECTED]/1410344
--=-=-=-= extentions.conf-=-=-=-=-=- have also
tried sip phone same results
[sipinbound]Exten =
_.,1,Dial,Zap/5-1
-=-=-=-=-=-=-=-=-=- upgraded to lastest cvs with
same results
pbx1*CLI show versionAsterisk
CVS-10/03/03-13:40:08 built by [EMAIL PROTECTED] on a i686
running Linux
-=-=-=-=-=-=-=-=-=-=
pbx1*CLI Sip read: INVITE
sip:[EMAIL PROTECTED] SIP/2.0Record-Route:
sip:[EMAIL PROTECTED]:5060;maddr=213.137.73.176Via:
SIP/2.0/UDP
213.137.73.176:5060;branch=7fdf8984-233992d8-bccd43ce-d4d3adac-1Via:
SIP/2.0/UDP 213.137.65.234:5060From:
sip:[EMAIL PROTECTED];tag=17B49340-1BD5To:
sip:[EMAIL PROTECTED]Date: Fri, 03 Oct 2003 20:37:52
GMTCall-ID: [EMAIL PROTECTED]Supported:
timer,100relMin-SE: 1800Cisco-Guid:
1267048311-4111995351-2493635217-4243844325User-Agent:
Cisco-SIPGateway/IOS-12.xAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK,
COMET, REFER, SUBSCRIBE, NOTIFY, INFOCSeq: 101 INVITEMax-Forwards:
9Remote-Party-ID:
sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=offTimestamp:
1065213472Contact: sip:[EMAIL PROTECTED]:5060Diversion:
sip:[EMAIL PROTECTED];reason=unconditionalExpires:
180Allow-Events: telephone-eventContent-Type:
application/sdpContent-Length: 332
v=0o=CiscoSystemsSIP-GW-UserAgent 7043 3136 IN
IP4 213.137.65.234s=SIP Callc=IN IP4 213.137.65.234t=0 0m=audio
16826 RTP/AVP 4 18 101 19c=IN IP4 213.137.65.234a=rtpmap:4
G723/8000a=fmtp:4 annexa=noa=rtpmap:18 G729/8000a=fmtp:18
annexb=noa=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=rtpmap:19
CN/8000
23 headers, 14 linesUsing latest request as
basis requestSending to 213.137.73.176 : 5060 (non-NAT)Found audio
format ULAWFound audio format UNKNFound audio format UNKNFound audio
format UNKNFound description format G723Found description format
G729Found description format telephone-eventFound description format
CNCapabilities: us - 524302, them - 257/0, combined - 0Non-codec
capabilities: us - 1, them - 3, combined - 1Sip read: INVITE
sip:[EMAIL PROTECTED] SIP/2.0Record-Route:
sip:[EMAIL PROTECTED]:5060;maddr=213.137.73.176Via:
SIP/2.0/UDP
213.137.73.176:5060;branch=7fdf8984-233992d8-bccd43ce-d4d3adac-1Via:
SIP/2.0/UDP 213.137.65.234:5060From:
sip:[EMAIL PROTECTED];tag=17B49340-1BD5To:
sip:[EMAIL PROTECTED]Date: Fri, 03 Oct 2003 20:37:52
GMTCall-ID: [EMAIL PROTECTED]Supported:
timer,100relMin-SE: 1800Cisco-Guid:
1267048311-4111995351-2493635217-4243844325User-Agent:
Cisco-SIPGateway/IOS-12.xAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK,
COMET, REFER, SUBSCRIBE, NOTIFY, INFOCSeq: 101 INVITEMax-Forwards:
9Remote-Party-ID:
sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=offTimestamp:
1065213472Contact: sip:[EMAIL PROTECTED]:5060Diversion:
sip:[EMAIL PROTECTED];reason=unconditionalExpires:
180Allow-Events: telephone-eventContent-Type:
application/sdpContent-Length: 332
v=0o=CiscoSystemsSIP-GW-UserAgent 7043 3136 IN
IP4 213.137.65.234s=SIP Callc=IN IP4 213.137.65.234t=0 0m=audio
16826 RTP/AVP 4 18 101 19c=IN IP4 213.137.65.234a=rtpmap:4
G723/8000a=fmtp:4 annexa=noa=rtpmap:18 G729/8000a=fmtp:18
annexb=noa=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=rtpmap:19
CN/8000
23 headers, 14 linesIgnoring this
requestLooking for 14103445557 in sipinboundRDNIS is
4103445557list_route: hop:
sip:[EMAIL PROTECTED]:5060;maddr=213.137.73.176list_route:
hop: sip:[EMAIL PROTECTED]:5060Transmitting (no
NAT):SIP/2.0 100 TryingVia: SIP/2.0/UDP
213.137.73.176:5060;branch=7fdf8984-233992d8-bccd43ce-d4d3adac-1Via:
SIP/2.0/UDP 213.137.65.234:5060From:
sip:[EMAIL PROTECTED];tag=17B49340-1BD5To:
sip:[EMAIL PROTECTED];tag=as72f8d457Call-ID: [EMAIL PROTECTED]CSeq:
101 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFERContact: sip:[EMAIL PROTECTED]Content-Length:
0
to 213.137.73.176:5060 -- Executing
Dial("SIP/-080e9768", "Zap/5-1") in new stack -- Called
5-1 -- Zap/5-1 is ringingTransmitting (no
NAT):SIP/2.0 180 RingingVia: SIP/2.0/UDP
213.137.73.176:5060;branch=7fdf8984-233992d8-bccd43ce-d4d3adac-1Via:
SIP/2.0/UDP 213.137.65.234:5060From:
sip:[EMAIL PROTECTED];tag=17B49340-1BD5To:
sip:[EMAIL PROTECTED];tag=as72f8d457Call-ID: [EMAIL PROTECTED]CSeq:
101 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFERContact: sip:[EMAIL PROTECTED]Content-Length:
0
to
[Asterisk-Users] Communication between 2 telephones
Hi! I have installed everything, asterisk, pwlib,openh323, chan_oh323. And now? I want to install ophone to talk, but I don't see what is the asterisk role. I mean, ophone lets us to talk with another phone,... why do we need asterisk? What does ophone do and what dows asterisk do? Thanks for all your help Mireia ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with questions for initial Asterisk wizard (GUI)
Steven Critchfield wrote: On Mon, 2003-10-06 at 23:59, Leif Madsen wrote: Hey all, I am in the middle of creating a new user wizard which will generate all the .conf's the new Asterisk user will require to get themselves up and running in Asterisk without having to touch a single configuration file. This is what I have come up with as a rough draft. It is far from complete, so I'm asking people to submit things that should be added, changed, removed etc.. etc... so please help me come up with a good logic for the questions so that I may start the work on the actual wizard! Maybe it's me, but only the most basic of setups will be handled by a wizard. The person who does the setup necessarily needs to know a fair amount about routing of calls. Exactly. This is not meant for the advanced user to come in and manage their entire system (at least not at this point). This is meant to give the clueless newbie a little place to start in making their configuration instead of having to search all over the web for the handbook, websites from different people, asking questions in the IRC channel that have been asked 1000 times, etc.. etc.. I understand your concern, but it is simply meant to be a stepping stone for those with NO knowledge of how to setup a .conf file in Asterisk. I realize it won't be able to take a non-existant * box to an advanced setup ready for production, and that is not what I am trying to accomplish. I'd simply like to give the user a place to start and get some basic configuration files so that they can kind of visualize how they work. Do you have any SIP devices? (yes/no) + SIP phone + Softphone Does this make a difference as to whether it is software or not? No, but it was late, and I was just writing stuff as it came to me :) As a side note, you might want to disable the sip module if it isn't going to be used to help close down unused ports. If you plan to coddle a user, might as well do it right. Good idea. I'll have to keep stuff like that in mind for disabling modules that aren't being used. Where do you plan on connecting these with actual channels? Also do you plan on building out menu generation. Menu generation would be something I would like to add, perhaps using the festival application? It might not be something I worry about at first though. Don't forget to explain to the user your coddling about significant digits, ie. for x number of extensions you will probably need to extend 1 more digit so they all start with the same digit, or only consume 1 or 2 significant digits since one or more will also be used to signify outbound dialing and any number of other lead digits might need to be used in other functions like meetme apps. Hrm... right. Yet another thing to keep in mind. Might as well add configuration of outbound prefixes for dialing. Ready to tackle all those international codes too. I'm sure our euro friends will love it if you get those patterns all down for them. Might as well eh? :) Perhaps I can just have it ask for the prefixes that the user would like added, then the administrator can worry about adding all their own prefixes. If I get THAT ambitious, then we'll see... Would you like to activate any of these extensions now? (yes/no) + List extensions with CONFIG | EDIT | DELETE | ADD links + Who is going to use this extension? (name) + What is the email address of the person as this extension? (for email notification) Shouldn't this be folded into the extension setup? Probably. This should be folded into channel setup. Of course this is where it gets tricky as you can't specify a mailbox until you specified extensions and worked out the problems with the extension logic. Right... perhaps voicemail configuration should come more towards the end, or at least after all the extensions are configured... ? Set up Meetme apps, upload and use voice prompts, layout appropriate contexts for incoming outgoing and menus, DISA. What point do you feel that a user is too advanced to us your wizard, or at what point do you think a user of your wizard will be more pissed at being hindered by the product than helped? I don't feel that any user should go in thinking this wizard will help them put a machine into production. This wizard is simple meant to give the beginning a user to start and learn how to actually manage the .conf files him/herself. I look at it like this: I'm a new user, and I have never even thought about looking at a configuration file for Asterisk, what kinds of things might I want to start out with. I'd probably want to setup my dev kit I just ordered from Digium. I might want to connect to some sort of SIP proxy so I can receive calls, such as FWD, SIPPhone.. etc.. etc.. I may have a pair of Asterisk boxes with SIP devices that I want to call between either with SIP or IAX (and as has been mentioned, the SIP devices can be either softphones or
[Asterisk-Users] Dialling problems
Hey all, I'm having problems reliably dialling out my FXO card. About 30% of the time I'll get a your call cannot be completed as dialed. I'm thinking it might be the dialling speed, but I can't find any configs that change that setting. Any suggestions for troubleshooting? Thanks, Brad Waite ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conferencing Calls on Cisco 7940
On Mon, 2003-10-06 at 20:47, Brian West wrote: Works fine on my 7960 with 5.3 firmware. bkw Im having a similar problem with my 7960 when I receive two incoming calls I cannot join them. Hello, I am trying to conference two or more calls on a Cisco 7940 phone. When I have one inbound call and one outbound (I initiate the second call by pressing conference) I get the join button at the bottom of the screen and I can conference. When I initiate both calls or I receive both calls I dont get the join button. As a side question what would represent the hook flash on a Cisco 7940 or is this capability not possible. Thanks Babak -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Xten X-lite codec problem ???
Hello folks, I trying to get working my xten (X-lite V2) working with Asterisk !!! It's working nice with my development server without nat but NO in my production server with NAT=yes !!! Below my client configuration for asterisk: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls allow=all [phone1] type=friend host=dynamic username=phone1 secret=phone1 mailbox=; Mailbox for message waiting indicator context=demo callerid=Windows 2124 My X-lite is configured how I read in some old threads Under System SettingsNetwork 1- Set the IP of you * box in Outbound SIP Proxy Under System SettingsSip Proxy 1- Enable yes 2- Username (the name or number in your SIP.CONF [brackets] 3- Leave Authorized User blank (and remark out in SIP.CONF if you have it in there.) 4- Obviously set the password 5- Domain/Realm: the IP of your * box. 6- Sip Proxy: the IP of your * box. 7- Send Internal IP: ON When I use the dev server is working correctly... but to the production server, however it is seems to work fine in asterisk side (if I check the console mode, or sip trace) but I don't hear anything in the client X-lite... I don't understand why cause the only things I changed is nat=yes... A bit of help will be really welcome, Ares ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conferencing Calls on Cisco 7940
Im having a similar problem with my 7960 when I receive two incoming calls I cannot join them. ya you can't join them. That sucks.. but you can park one call, go back to call number 1. Press conf. Dial the parking orbit.. then press join! bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX and Jitter problem
Hello, I've been playing around with * for quite a while now, and have run into a problem that I just cannot seem to figure out. When using * and any IAX client (I have tested with GnoPhone and both clients from iaxclient.sourceforge.net) I have incredibly bad jitter on the connection. What I'm running is a P3-1Ghz machine with 512mb ram for a server. The other end has been various machines (all connected via 100mb switch) ranging from a AMD K6-2 350 running Win98 to another P3-1Ghz running RH Linux 9.0 and GnoPhone. I've tried changing the jitterbuffer settings in iax.conf (including turning it off as I've seen some recommendations on the archives) and I've even tried rebuilding zaptel with the various jitter control switches. At this point I have extension 8500 setup to take me to voicemailmain. When I connect (IAX only - I do not have any Digium cards in the server at all) I can generaly not tell what is being said at all. I've used sox and a player and know that the .gsm files are okay. Anybody have any suggestions of what to try? So far this has been something I've been playing with before I attempt to put it in a production system, but so far am not having a whole lot of luck. I've not been able to try SIP as of yet, as I've not found a softclient and the application I will be using * for would require this. Thanks, Mike Atkinson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialling problems
On Tue, 7 Oct 2003, Brad Waite wrote:
I'm having problems reliably dialling out my FXO card. About 30% of the time
I'll get a your call cannot be completed as dialed. I'm thinking it might be
the dialling speed, but I can't find any configs that change that setting.
We had the same problem and had to modify our extensions like so:
exten = _9NXX,5,Dial(Zap/g1/w${EXTEN:1}||Tr)
Add a w before the number and it will pause a bit.
--
Dave Weis I believe there are more instances of the abridgment
[EMAIL PROTECTED] of the freedom of the people by gradual and silent
encroachments of those in power than by violent
and sudden usurpations.- James Madison
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk-Users] IVR Questions?
OK, I've got my script all set up and running, but now Asterisk crashes when the digits are entered with the following error: Ouch ... error while writing audio data: : Broken pipe I just retrieved and compiled the latest CVS this morning, as well as the latest AGI perl module. Why won't the AGI-get_data() function work correctly? Joe Richard Lyman [EMAIL PROTECTED] wrote the Oct 6, 2003 6:08 PM: simply add... .. my $AGI = new Asterisk::AGI; my %input = $AGI-ReadParse(); ## this line .. Joe Dennick wrote: That makes a lot of sense, but...it still doesn't work. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Crick Sent: Monday, October 06, 2003 4:18 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] IVR Questions? Try putting an Answer() in your extensions.conf before you call the AGI code - a common gotcha I think? ___ 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] IAX and Jitter problem
On Tue, 2003-10-07 at 11:14, [EMAIL PROTECTED] wrote: At this point I have extension 8500 setup to take me to voicemailmain. When I connect (IAX only - I do not have any Digium cards in the server at all) I ^^ ding ding ding, we have a winner. Please try and install one of the psuedo channels such as ztdummy or the rtc channel driver so as to provide you with timing. Your problems are most likely associated with the lack of something throttling your output. can generaly not tell what is being said at all. I've used sox and a player and know that the .gsm files are okay. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] clocking source for T100P?
is it preferred that the T100P generate the T1 clock or that whatever it is plugged in to (channel bank, PRI, whatever) generate the clock? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] agi exit problem
Hello
Is it possible to make an agi script keep going after a Dial is exectued?
Example:
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
$AGI-verbose(-- Hello);
$AGI-exec('Dial',IAX2/whatever); when this call ends the agi script
ends.
$AGI-verbose(-- Hello again); --- it never gets to here :(
$AGI-hangup();
exit(0);
Any help would be appreciated,
Thanks
Panny
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call parking on specific park number
Hello, Is there any way to park a call on a specific park number? If this is not possible, is there any way to create multiple park orbits? Also, is there any way to invoke call parking of an active call coming through a Zap channel from the manager interface? MATT--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] agi exit problem
Hello
Is it possible to make an agi script keep going after a Dial
is exectued?
Example:
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
$AGI-verbose(-- Hello);
$AGI-exec('Dial',IAX2/whatever); when this call ends
the agi script
ends.
$AGI-verbose(-- Hello again); --- it never gets to here :(
$AGI-hangup();
exit(0);
Not sure if it's possible to keep the script running after Dial but perhaps
you could explain what you're attempting to achieve and there may be a
workaround.
Dave
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX and Jitter problem
Thought I would just mention that I have a Pentium 150 with 64MB of RAM, asterisk installed, 2 Budgetone 102's and an X100P. No problem with jitter here or anything like that. I don't use mp3 music on hold because I doubt the hardware would cope particularly well. Has anybody got Asterisk running on anything lower spec than this? Michael On Tue, 7 Oct 2003 [EMAIL PROTECTED] wrote: Hello, I've been playing around with * for quite a while now, and have run into a problem that I just cannot seem to figure out. When using * and any IAX client (I have tested with GnoPhone and both clients from iaxclient.sourceforge.net) I have incredibly bad jitter on the connection. What I'm running is a P3-1Ghz machine with 512mb ram for a server. The other end has been various machines (all connected via 100mb switch) ranging from a AMD K6-2 350 running Win98 to another P3-1Ghz running RH Linux 9.0 and GnoPhone. I've tried changing the jitterbuffer settings in iax.conf (including turning it off as I've seen some recommendations on the archives) and I've even tried rebuilding zaptel with the various jitter control switches. At this point I have extension 8500 setup to take me to voicemailmain. When I connect (IAX only - I do not have any Digium cards in the server at all) I can generaly not tell what is being said at all. I've used sox and a player and know that the .gsm files are okay. Anybody have any suggestions of what to try? So far this has been something I've been playing with before I attempt to put it in a production system, but so far am not having a whole lot of luck. I've not been able to try SIP as of yet, as I've not found a softclient and the application I will be using * for would require this. Thanks, Mike Atkinson ___ 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] agi exit problem
Not sure if it's possible to keep the script running after Dial but perhaps you could explain what you're attempting to achieve and there may be a workaround. I want to know how long the call lasted :) Panny ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] agi exit problem
On Tue, 7 Oct 2003, Panny Malialis wrote: Not sure if it's possible to keep the script running after Dial but perhaps you could explain what you're attempting to achieve and there may be a workaround. I want to know how long the call lasted :) Your AGI will continue to run, but after the call has hungup you can no longer exectue any AGI commands. Your verbose will fail, but if you print to STDERR you will see that your script is still running. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with questions for initial Asterisk wizard (GUI)
Hi Leif im not good programmer but if need some help mail to me for everything. Thanks Dimitri On Tuesday 07 October 2003 04:59, Leif Madsen wrote: Hey all, I am in the middle of creating a new user wizard which will generate all the .conf's the new Asterisk user will require to get themselves up and running in Asterisk without having to touch a single configuration file. This is what I have come up with as a rough draft. It is far from complete, so I'm asking people to submit things that should be added, changed, removed etc.. etc... so please help me come up with a good logic for the questions so that I may start the work on the actual wizard! Thanks in advance for all your help! Hardware Are you using Digium Hardware? (yes/no) + TE410P + How many? + TDM400P + How many? + How many modules (each card) + T400P + How many? + T100P + How many? + E100P + How many? + X100P + How many? Other hardware + Do you have a soundcard? (yes/no) + ALSA or OSS? Do you have any SIP devices? (yes/no) + SIP phone + Softphone Extension Ranges + Start and Stop range + Would like you to enable voicemail on any of these extensions? (yes/no) + all of them (yes/no) + which ones? + What should the default voicemail password be? + Default formats for writing voicemail + GSM, wav49, WAV + Email notification? (yes/no) + Who should the email appear to come from? + Should we attach it to the email? + Would you like to specify a maximum message length? (yes/no) + How long? + Would you like to specify a maximum greeting length? (yes/no) + How long? + Which country are you in? (for indication) + United States + Australia + France + Netherlands + United Kingdom + Which language? (for zapata.conf) Would you like to activate any of these extensions now? (yes/no) + List extensions with CONFIG | EDIT | DELETE | ADD links + Who is going to use this extension? (name) + What is the email address of the person as this extension? (for email notification) + For each channel of the hardware the user has + Which signalling for this channel? kewl start loop start ground start + Enable three way calling? + Enable transfer? + Enable call waiting? + Enable busy detection? + Use CallerID? + rxgain + txgain + Immediate? (yes/no) + CallerID String + Name + Number + Enable mailbox indication? + Mailbox number(s) to be associated with this channel + Context Would you like to setup registration for FWD, IAXtel, SIPPhone or iptel? + -- +--+ |Leif Madsen - http://www.hacklocalhost.com| +--+ |@| leif at hacklocalhost dot com | | SMS| sms at hacklocalhost dot com | | FWD| 18924 IAX| 1700-363-0761 | |iptel| 8972-1969sipph| 1-747-386-1618 | +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] agi exit problem
Thanks, that makes sense now :) Panny - Original Message - From: James Golovich [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, October 07, 2003 7:18 PM Subject: Re: [Asterisk-Users] agi exit problem On Tue, 7 Oct 2003, Panny Malialis wrote: Not sure if it's possible to keep the script running after Dial but perhaps you could explain what you're attempting to achieve and there may be a workaround. I want to know how long the call lasted :) Your AGI will continue to run, but after the call has hungup you can no longer exectue any AGI commands. Your verbose will fail, but if you print to STDERR you will see that your script is still running. James ___ 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] agi exit problem
The way I worked around this is to log the uniqueid in a database when the
call is placed with the start time and then execute an agi script upon all
hangups:
exten = h,1,AGI(call_log.agi,${EXTEN})
That script queries the database for the uniqueid and if it exists in the
table it figures out the call length and updates the record
It's a little bit of perl overhead but if you have a fast system and a fast
DB it should have very little delay.
MATT---
-Original Message-
From: Panny Malialis [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 07, 2003 2:11 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] agi exit problem
Not sure if it's possible to keep the script running after Dial but
perhaps
you could explain what you're attempting to achieve and there may be a
workaround.
I want to know how long the call lasted :)
Panny
___
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] agi exit problem
Citeren Dave Wilson [EMAIL PROTECTED]:
Is it possible to make an agi script keep going after a Dial
is exectued?
Example:
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
$AGI-verbose(-- Hello);
$AGI-exec('Dial',IAX2/whatever); when this call ends
the agi script
ends.
$AGI-verbose(-- Hello again); --- it never gets to here :(
$AGI-hangup();
exit(0);
Not sure if it's possible to keep the script running after Dial but perhaps
you could explain what you're attempting to achieve and there may be a
workaround.
Wouldn't it be cool if you could take the call back, into the IVR script after
the remote party hangs up ?
Might be usefull for callcenters where agents want to allow customers to re-
enter the IVR wherever they left off. Ofcourse one might hack it by
transferring the user back into the IVR one way or another, but it is a hack.
Taking the call back after the Dial statement seems more elegant to me ?
--
Met vriendelijke groet,
Florian Overkamp
ObSimRef BV
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] agi exit problem
exten = h,1, will not work if you park a call then pick it back up. You
are flipping the call direction from what Mark told me. Whats wrong with
CDR data? is that not good enough to tell call lenght?
bkw
On Tue, 7 Oct 2003, mattf wrote:
The way I worked around this is to log the uniqueid in a database when the
call is placed with the start time and then execute an agi script upon all
hangups:
exten = h,1,AGI(call_log.agi,${EXTEN})
That script queries the database for the uniqueid and if it exists in the
table it figures out the call length and updates the record
It's a little bit of perl overhead but if you have a fast system and a fast
DB it should have very little delay.
MATT---
-Original Message-
From: Panny Malialis [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 07, 2003 2:11 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] agi exit problem
Not sure if it's possible to keep the script running after Dial but
perhaps
you could explain what you're attempting to achieve and there may be a
workaround.
I want to know how long the call lasted :)
Panny
___
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] FXO on ATT broadband POTS line?
Does anybody out there run * on an ATT broadband phone line? I'm not seeing any callerid and I can't tell if its ATT doing something funky or if its my setup. I do see CID on my normal phones Thanks, Chris -- The face of a child can say it all, especially the mouth part of the face. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Park on SIP phones
Hi, It is posible to put a call in the parking lot with a SIP phone as a Cisco 7960 ? Anyway, how can I put a call park on a FXS line ? Is there any magic digits ? -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conferencing Calls on Cisco 7940
On Tue, 2003-10-07 at 12:09, Brian West wrote: Im having a similar problem with my 7960 when I receive two incoming calls I cannot join them. ya you can't join them. That sucks.. but you can park one call, go back to call number 1. Press conf. Dial the parking orbit.. then press join! How ? I dont know how to park a call with the 7960. BTW, heres an URL which may be related with the 3-way bug on 7960s. http://paf.se/inoc-dba/17.html -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Compile problem SuSE 8.2
I am trying to compile * on SuSE 8.2. When doing the make install in /usr/src/zaptel I get the following error. ** /usr/src/linux/include/asm/system.h:189: warning: dereferencing type-punned pointer will break strict-aliasing rules freeIn file included from /usr/src/linux/include/linux/highmem.h:5, from /usr/src/linux/include/linux/vmalloc.h:8, from /usr/src/linux/include/asm/io.h:47, from /usr/src/linux/include/asm/pci.h:40, from /usr/src/linux/include/linux/pci.h:654, from zaptel.c:38: /usr/src/linux/include/asm/pgalloc.h: In function `flush_tlb_page': /usr/src/linux/include/asm/pgalloc.h:201: internal compiler error: Segmentation fault Please submit a full bug report, with preprocessed source if appropriate. See URL:http://www.gnu.org/software/gcc/bugs.html for instructions. make: *** [zaptel.o] Error 1 *** Any ideas about where to look for the problem would be appreciated. Robert Friedrichshafen, Germany ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Large-scale Asterisk deployments: VON panel
I'd like to float the idea of a VON panel discussion for large-scale open-source deployments, specifically using Asterisk as an application service and as a gateway service. In order to do that, I'd need to get a list of panelists together who might be interested in speaking. If you manage an installed base of 1000 users, and you're using Asterisk, I'd be interested in hearing from you. I know of at least five of you that I've spoken with directly, and I'm sure there are a dozen or so more out there that are keeping quiet. If you are willing to talk about problems, solutions, and (importantly) your value comparisons and real costs, then I'd like to hear from you. VON is in Santa Clara, California (Silicon Valley) March 28-31 2004. If you are a speaker or panelist, your admission to the conference is free. I'm just fishing to see if the panel might be well-attended; if so, I'll ask the folks at pulver.com if they're interested in such a panel. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call parking on specific park number
Hello, Is there any way to park a call on a specific park number? Not to my knowledge. If this is not possible, is there any way to create multiple park orbits? Not to my knowledge. This seems to be lagging behind some of the other features within Asterisk which allow compartmentalization of phone calls in a way that would suggest that those other features were designed for multiple entities using the same system. The use of contexts in voicemail, as an example, show excellent attention to segregation. However, parking is still a single pool, which would allow inadvertent pickups of calls between organizations if someone fat-fingered the extension. Everyone would be very happy if you re-wrote the parking applications to support multiple contexts and submitted the patch. :-) Also, is there any way to invoke call parking of an active call coming through a Zap channel from the manager interface? MATT--- Does redirect do what you want? JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compile problem SuSE 8.2
On Tue, 2003-10-07 at 14:10, rnc Info Lists wrote: I am trying to compile * on SuSE 8.2. When doing the make install in /usr/src/zaptel I get the following error. ** /usr/src/linux/include/asm/system.h:189: warning: dereferencing type-punned pointer will break strict-aliasing rules freeIn file included from /usr/src/linux/include/linux/highmem.h:5, from /usr/src/linux/include/linux/vmalloc.h:8, from /usr/src/linux/include/asm/io.h:47, from /usr/src/linux/include/asm/pci.h:40, from /usr/src/linux/include/linux/pci.h:654, from zaptel.c:38: /usr/src/linux/include/asm/pgalloc.h: In function `flush_tlb_page': /usr/src/linux/include/asm/pgalloc.h:201: internal compiler error: Segmentation fault Please submit a full bug report, with preprocessed source if appropriate. See URL:http://www.gnu.org/software/gcc/bugs.html for instructions. make: *** [zaptel.o] Error 1 *** Any ideas about where to look for the problem would be appreciated. A segfault during compile is either a hardware problem, or a gcc problem. Specifically this error said it was an internal compiler error and therefore falls squarely on your hardware or gcc version. If you don't have any reason to suspect your hardware, check for gcc upgrades. If gcc wasn't the problem, check your memory, cpu temperature, cpu period, and possibly your hard drive and controller in pretty much this order. Any of these could flip a bit or 2 and screw everything up. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] echo for 15 seconds 002401c38308$2e05e0a0$0102010a@JUPITER m2brszwm6k.fsf@tnuctip.rychter.com 1065158738.26944.4.camel@penguin.isyourdaddy.net
On Sat, 2003-10-04 at 15:09, Jan Rychter wrote: Any chance you could describe the hardware? Was it a Via-based board? I have a setup where I use two *'s, both on Via boards. One is a Mini-ITX and the other is a full-form motherboard. Would interrupt-sharing between the X100P and another card cause this problem? (there is simply no way to avoid it on some hardware!) I can't remember exactly what mobo it was. It was made by a company called Syntax. It was mini-ATX, or whatever the step down from ATX with only 2 PCI slots is called. I believe that it was interrupt sharing that caused the problem. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Re: [Asterisk-Users] IAX and Jitter problem
I'm coming at this thing from an Operational standpoint rather than a development standpoint. Viewing your problem from that angle, I wonder how well your network is performing. Could you have a cable problem that the Asterisk server hasn't reported (Layer 1); or perhaps your * Server is connecting at 100 megabit/half duplex, but the switch is configured (or auto-detected) 100 megabit/full duplex (Layer 2); or perhaps you have a bad port on your switch (Layer 1 or 2); or perhaps there is a mis-typed subnet mask or default gateway somewhere between the systems that hasn't been caught yet (Layer 3). Is your Asterisk server busy doing anything else that's tying up resources? You also menitoned that you haven't yet found a soft-phone. X-Lite (from www.eten.com) works really well on Windows workstations. Michael T Farnworth [EMAIL PROTECTED] wrote the Oct 7, 2003 12:52 PM: Thought I would just mention that I have a Pentium 150 with 64MB of RAM, asterisk installed, 2 Budgetone 102's and an X100P. No problem with jitter here or anything like that. I don't use mp3 music on hold because I doubt the hardware would cope particularly well. Has anybody got Asterisk running on anything lower spec than this? Michael On Tue, 7 Oct 2003 [EMAIL PROTECTED] wrote: Hello, I've been playing around with * for quite a while now, and have run into a problem that I just cannot seem to figure out. When using * and any IAX client (I have tested with GnoPhone and both clients from iaxclient.sourceforge.net) I have incredibly bad jitter on the connection. What I'm running is a P3-1Ghz machine with 512mb ram for a server. The other end has been various machines (all connected via 100mb switch) ranging from a AMD K6-2 350 running Win98 to another P3-1Ghz running RH Linux 9.0 and GnoPhone. I've tried changing the jitterbuffer settings in iax.conf (including turning it off as I've seen some recommendations on the archives) and I've even tried rebuilding zaptel with the various jitter control switches. At this point I have extension 8500 setup to take me to voicemailmain. When I connect (IAX only - I do not have any Digium cards in the server at all) I can generaly not tell what is being said at all. I've used sox and a player and know that the .gsm files are okay. Anybody have any suggestions of what to try? So far this has been something I've been playing with before I attempt to put it in a production system, but so far am not having a whole lot of luck. I've not been able to try SIP as of yet, as I've not found a softclient and the application I will be using * for would require this. Thanks, Mike Atkinson ___ 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] Conferencing Calls on Cisco 7940
I dont see it as a bug.. I see why it don't work.. and why people think it should. Enable # transfers.. and setup call parking to get around this. Also if you conf very much look at app_meetme. bkw On Tue, 7 Oct 2003, Juan J. Sierralta P. wrote: On Tue, 2003-10-07 at 12:09, Brian West wrote: Im having a similar problem with my 7960 when I receive two incoming calls I cannot join them. ya you can't join them. That sucks.. but you can park one call, go back to call number 1. Press conf. Dial the parking orbit.. then press join! How ? I dont know how to park a call with the 7960. BTW, heres an URL which may be related with the 3-way bug on 7960s. http://paf.se/inoc-dba/17.html -- Juanjo sin .sig ___ 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] Call Park on SIP phones
Not yet.. but I sure wish we could... :) On Tue, 7 Oct 2003, Juan J. Sierralta P. wrote: Hi, It is posible to put a call in the parking lot with a SIP phone as a Cisco 7960 ? Anyway, how can I put a call park on a FXS line ? Is there any magic digits ? -- Juanjo sin .sig ___ 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] Help with questions for initial Asterisk wizard (GUI)
Dimitri Bellini wrote: Hi Leif im not good programmer but if need some help mail to me for everything. Yah... me niether :) At this stage it is simply going to be figuring out the logic so that I know I have asked all the questions that need to be asked. If you can think of things I've missed, feel free to chime in! Thanks, -- +--+ |Leif Madsen - http://www.hacklocalhost.com| +--+ |@| leif at hacklocalhost dot com | | SMS| sms at hacklocalhost dot com | | FWD| 18924 IAX| 1700-363-0761 | |iptel| 8972-1969sipph| 1-747-386-1618 | +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] clocking source for T100P?
is it preferred that the T100P generate the T1 clock or that whatever it is plugged in to (channel bank, PRI, whatever) generate the clock? That depends on the environment This is what i have read b4 about t1 timing srcs 1. If the T1 is point to point where both ends terminate on a different * boxs. Then one of the T100p supplies clock for the other . The network, unless other wise designed, will not supply clock to the T1 2. If the T1 originates from a PRI or channel bank, switch, etc, then the network supplies clock UNLESS the t100p can supply a T1 clock source traceable to a Stratum 1 Primary Reference Source (PRS) , then the T100 can supply clock back to the T1.. see here for a defintion of stratum src http://www.raltron.com/products/pdfspecs/sync_an02-StratumLevelDefined.pdf I am sure the clock on the digium h/w is not Stratum 1 :) Its is probably a Stratum 4 TTL oscillator... I think there are only a few Stratum 1 src in the NorthAmerica big . If your just clocking a channel bank it maybe a toss up who has the the best timing src, but some might have a Stratum 3 certifed src ?? From what I have read most telcos have Stratum 2 or 3 devices as backup but get an outside feed from a bonfide Stratum 1 src ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Line going to Zombie
I have a problem that sometimes lines will go into what I call never never land. The Asterisk system will put a line with Zombi on it when you type show channels it will make the analog phone line dead. And on the CLI it says: astsvr*CLIZap/1-2ZOMBIE(macro-twoline-exten,s,1)Up Dial Zap/1-2|20|r I have tried to release it with soft hangup Zap/1 also soft hangup Zap/1-2. If I use the last one is say trying to hang up but it never does. I have to shut the system down then back up! I am not able to run the GASTMAN due to I have no XP machine running. I only have Windows 2000 pro. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Line going to Zombie
On Tue, 2003-10-07 at 14:51, Ariel Batista wrote: I have a problem that sometimes lines will go into what I call never never land. The Asterisk system will put a line with Zombi on it when you type show channels it will make the analog phone line dead. And on the CLI it says: astsvr*CLIZap/1-2ZOMBIE(macro-twoline-exten,s,1)Up Dial Zap/1-2|20|r I have tried to release it with soft hangup Zap/1 also soft hangup Zap/1-2. If I use the last one is say trying to hang up but it never does. I have to shut the system down then back up! I am not able to run the GASTMAN due to I have no XP machine running. I only have Windows 2000 pro. I have seen this also. GASTMAN wouldn't have helped you since it would issue the same commands as you did. You can always use astman for similar functions without the gtk widgitry. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conferencing Calls on Cisco 7940
Brian, Would you be kind enough to give me a brief overview of why it doesnt work. I also appreciate the work aorund. This is something I will have to educate my soon to be users on. We do a lot of conferencing of calls as a matter of facilitating clients' immediate needs. For now I will try parking one or more of the calls and conferencing via calling the park extension. I already have a meetme room setup, but it's not quite as convenient as asking someone to hangon while you get the other parties on the line to work out an issue. Especially since it is our policy to authenticate all meetmes. Thanks for everyone's response to this issue. Babak Brian West wrote: I dont see it as a bug.. I see why it don't work.. and why people think it should. Enable # transfers.. and setup call parking to get around this. Also if you conf very much look at app_meetme. bkw On Tue, 7 Oct 2003, Juan J. Sierralta P. wrote: On Tue, 2003-10-07 at 12:09, Brian West wrote: I´m having a similar problem with my 7960 when I receive two incoming calls I cannot join them. ya you can't join them. That sucks.. but you can park one call, go back to call number 1. Press conf. Dial the parking orbit.. then press join! How ? I don´t know how to park a call with the 7960. BTW, here´s an URL which may be related with the 3-way bug on 7960s. http://paf.se/inoc-dba/17.html -- Juanjo sin .sig ___ 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 -- Babak Pasdar Founder/CTO IGX Global 389 Main St. Hackensack, NJ 07601 www.igxglobal.com (201) 498-0555 ext. 2205 The electronic message that you have received and any attachments are solely intended for the use of the addressee(s) and may contain information that is confidential. If you receive this email in error, please advise us by responding to [EMAIL PROTECTED] You are required to delete the contents and destroy any copies immediately. IGX Global is not liable for the views expressed in this electronic message or for the consequences of any computer viruses that may be unknowingly transmitted within this message. This electronic message is also subject to standard copyright/ownership laws. It is not intended to be reproduced, or re-transmitted without the consent of the originator. www.igxglobal.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compile problem SuSE 8.2
I had a similar problem with redhat 9 stock kernel sources. I had to enter the kernel sources dir, do a make mrproper then a make menuconfig save the conf do make dep. after that I was able to build zaptel without issues ;) matteo. Il mar, 2003-10-07 alle 21:10, rnc Info Lists ha scritto: I am trying to compile * on SuSE 8.2. When doing the make install in /usr/src/zaptel I get the following error. ** /usr/src/linux/include/asm/system.h:189: warning: dereferencing type-punned pointer will break strict-aliasing rules freeIn file included from /usr/src/linux/include/linux/highmem.h:5, from /usr/src/linux/include/linux/vmalloc.h:8, from /usr/src/linux/include/asm/io.h:47, from /usr/src/linux/include/asm/pci.h:40, from /usr/src/linux/include/linux/pci.h:654, from zaptel.c:38: /usr/src/linux/include/asm/pgalloc.h: In function `flush_tlb_page': /usr/src/linux/include/asm/pgalloc.h:201: internal compiler error: Segmentation fault Please submit a full bug report, with preprocessed source if appropriate. See URL:http://www.gnu.org/software/gcc/bugs.html for instructions. make: *** [zaptel.o] Error 1 *** Any ideas about where to look for the problem would be appreciated. Robert Friedrichshafen, Germany ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfer from IAX call
On Fri, 3 Oct 2003, Richard Lyman wrote: you'll find that the context is being overwritten. look in chan_iax.c line 1628 and chan_iax2.c line 1645 (or within 3 lines of each) there is a sprintf that is stuff the context, if you comment those out, it should work again. Disclaimer: i have NO CLUE what else this BREAKS!!! THat does look like a problem. Kram - what's the verdict. Is this a bug or a feature? What will changing this line affect? dave Dave Weis wrote: I am using IAX to send a call to my cell phone. I want to be able to hit # and transfer it back into the office. I have added tTr to the dial command and hitting # prompts me for the transfer, but after I start dialing 103, it stops at 1 and tries to transfer it within nufone instead of my dialplan. This is the debug output: -- Called [EMAIL PROTECTED]/1515480 -- Call accepted by 65.127.126.42 (format GSM) -- Format for call is GSM -- IAX2[NuFone]/3 is ringing -- IAX2[NuFone]/3 stopped sounds -- IAX2[NuFone]/3 answered Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 -- Playing 'pbx-transfer' -- Unable to find extension '1' in context 'NANPA' -- Playing 'pbx-invalid' -- Stopped music on hold on Zap/1-1 How do I make this work? dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ 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 -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with questions for initial Asterisk wizard (GUI)
Leif Madsen wrote: think of things I've missed, feel free to chime in! Leif, why don't you put the script up on the wiki so that we all can edit it and add on line with versioning? As a newborn Asterisk user, I had severe problems configuring an ISDN card. I believe a lot of new users start with a linux system and an ISDN bri card for tests. When I mailed the list, I got no answers that solved the problem, but a lot of mail from other users saying I've got the same problem. If you solve it, please inform me how! Oh, and kapejod that suggested I buy a real ISDN card to replace the junk I tried to use. And so I've done. I'm eagerly waiting for my CAPI ISDN card :-) Maybe the CAPI part of the Wiki will be a bit more detailed soon... /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call parking on specific park number
I would love to have separate callparking contexts available, it's omission
the reason I have not been using it up to this point.
As for redirect I haven't tried it yet, I just want to use the manager
interface to send a call on a zap channel to a parkedcall extension(ext.
700){or a specific park extension if possible} and then have manager receive
what park number that call is put on so I can display that on a user
interface application.
Thanks,
MATT---
Everyone would be very happy if you re-wrote the parking applications
to support multiple contexts and submitted the patch. :-)
Also, is there any way to invoke call parking of an active call coming
through a Zap channel from the manager interface?
MATT---
Does redirect do what you want?
JT
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] auto 'modprobe wct1xxp' on startup?
Hi guys, Thanks for your answers on my two questions yesterday. That's exactly what I was looking for, sorry for not noticing it myself, but I'm still getting acclimated to Asterisk and even Linux--from what I see so far, I love it. I've got another one now. Since my Asterisk install and configuration is fairly stable at this point, I'm interested it ensuring that during the event of a power failure, when the power returns (or if the machine is manually restarted) that Asterisk will successfully load on the other side (automatically). I've used the provided asterisk startup script (which I moved to /etc/rc.d/init.d) and RedHat's 'chkconfig' to make sure that Asterisk is started on bootup, but the problem I'm having has to do w/ the wct1xxp module, I believe. When I want to start Asterisk manually, I just type 'modprobe wct1xxp' and my two T1 cards are correctly started and then I can start asterisk w/ the normal commands and everything works. But, when I come back from a restart, it appears that the Asterisk startup failed, and I think it's b/c the wct1xxp module is not loaded. What is the recommended way to ensure this happens? I've been reading and found that modprobe (on startup, it appears) uses /etc/modules.conf, and here's what mine looks like: alias eth0 e1000 alias scsi_hostadapter megaraid alias usb-controller ehci-hcd alias usb-controller1 usb-uhci options torisa base=0xd alias char-major-196 torisa #post-install wcfxs /sbin/ztcfg #post-install wcfxsusb /sbin/ztcfg #post-install torisa /sbin/ztcfg #post-install tor2 /sbin/ztcfg #post-install wcfxo /sbin/ztcfg post-install wct1xxp /sbin/ztcfg #post-install wct4xxp /sbin/ztcfg (I commented out all of the modules I think I don't need, but it didn't work when they weren't commented out anyway). Does this have something to do w/ it? Do I need to add something to indicate that wct1xxp should be loaded on startup elsewhere? I appreciate your willingness to share your knowledge and expertise. jl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] agi exit problem
Why does having a call go through call parking make the h not work?
I currently don't use call parking for other reasons so I've never run into
that.
What is the event to run an agi script after a parked call is hung up?
I don't use CDR data because I have a custom perl/TK interface that grabs
live info from a database and I need a lot more flexibility than CDR can
provide. And I'm spoiled by being able to change the AGI scripts on the fly
without restarting Asterisk.
MATT---
-Original Message-
From: Brian West [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 07, 2003 2:55 PM
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] agi exit problem
exten = h,1, will not work if you park a call then pick it back up. You
are flipping the call direction from what Mark told me. Whats wrong with
CDR data? is that not good enough to tell call lenght?
bkw
On Tue, 7 Oct 2003, mattf wrote:
The way I worked around this is to log the uniqueid in a database when
the
call is placed with the start time and then execute an agi script upon all
hangups:
exten = h,1,AGI(call_log.agi,${EXTEN})
That script queries the database for the uniqueid and if it exists in
the
table it figures out the call length and updates the record
It's a little bit of perl overhead but if you have a fast system and a
fast
DB it should have very little delay.
MATT---
-Original Message-
From: Panny Malialis [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 07, 2003 2:11 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] agi exit problem
Not sure if it's possible to keep the script running after Dial but
perhaps
you could explain what you're attempting to achieve and there may be a
workaround.
I want to know how long the call lasted :)
Panny
___
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] Conferencing Calls on Cisco 7940
On Tue, 2003-10-07 at 15:38, Brian West wrote: I dont see it as a bug.. I see why it don't work.. and why people think it should. Enable # transfers.. and setup call parking to get around this. Also if you conf very much look at app_meetme. I did it, problem that I have now is the dialplan on the Cisco phone, as soon as I push # it dial without any number :( Im trying to get some info on dialplan.xml if somebody has an example to avoid the effect of the # I will appreciate it. Thanks! -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compile problem SuSE 8.2
On Tue, 2003-10-07 at 15:13, Brancaleoni Matteo wrote: I had a similar problem with redhat 9 stock kernel sources. I had to enter the kernel sources dir, do a make mrproper then a make menuconfig This was probably not a good idea as you have configured your kernel source differently than your running kernel. In either the /boot or / directory there should be a Config file that matches your running kernel. You should then ba able to copy it from the /boot or / directory as .config in your kernel source root directory. At that point you should be able to make oldconfig and it will configure your kernel source exactly as the running kernel. I don't know if it would cause problems or not, but it doesn't hurt to have consistency. save the conf do make dep. after that I was able to build zaptel without issues ;) matteo. Il mar, 2003-10-07 alle 21:10, rnc Info Lists ha scritto: I am trying to compile * on SuSE 8.2. When doing the make install in /usr/src/zaptel I get the following error. ** /usr/src/linux/include/asm/system.h:189: warning: dereferencing type-punned pointer will break strict-aliasing rules freeIn file included from /usr/src/linux/include/linux/highmem.h:5, from /usr/src/linux/include/linux/vmalloc.h:8, from /usr/src/linux/include/asm/io.h:47, from /usr/src/linux/include/asm/pci.h:40, from /usr/src/linux/include/linux/pci.h:654, from zaptel.c:38: /usr/src/linux/include/asm/pgalloc.h: In function `flush_tlb_page': /usr/src/linux/include/asm/pgalloc.h:201: internal compiler error: Segmentation fault Please submit a full bug report, with preprocessed source if appropriate. See URL:http://www.gnu.org/software/gcc/bugs.html for instructions. make: *** [zaptel.o] Error 1 *** Any ideas about where to look for the problem would be appreciated. Robert Friedrichshafen, Germany ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Line going to Zombie
-- Original Message -- From: Steven Critchfield [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Tue, 07 Oct 2003 15:06:12 -0500 On Tue, 2003-10-07 at 14:51, Ariel Batista wrote: I have a problem that sometimes lines will go into what I call never never land. The Asterisk system will put a line with Zombi on it when you type show channels it will make the analog phone line dead. And on the CLI it says: astsvr*CLIZap/1-2ZOMBIE(macro-twoline-exten,s,1)Up Dial Zap/1-2|20|r I have tried to release it with soft hangup Zap/1 also soft hangup Zap/1-2. If I use the last one is say trying to hang up but it never does. I have to shut the system down then back up! I am not able to run the GASTMAN due to I have no XP machine running. I only have Windows 2000 pro. I have seen this also. GASTMAN wouldn't have helped you since it would issue the same commands as you did. You can always use astman for similar functions without the gtk widgitry. I still don't understand! Sorry I am new to Asterisk. So there is no way to stop this channel unless we restart the system? I can't find any other documentation on this. I google the list and came up with not much information. I do not have astman loaded on the * box. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] auto 'modprobe wct1xxp' on startup?
cd /usr/src/asterisk; make config; cd /usr/src/zaptel; make config regards Martin On Tue, 7 Oct 2003, john lawler wrote: Hi guys, Thanks for your answers on my two questions yesterday. That's exactly what I was looking for, sorry for not noticing it myself, but I'm still getting acclimated to Asterisk and even Linux--from what I see so far, I love it. I've got another one now. Since my Asterisk install and configuration is fairly stable at this point, I'm interested it ensuring that during the event of a power failure, when the power returns (or if the machine is manually restarted) that Asterisk will successfully load on the other side (automatically). I've used the provided asterisk startup script (which I moved to /etc/rc.d/init.d) and RedHat's 'chkconfig' to make sure that Asterisk is started on bootup, but the problem I'm having has to do w/ the wct1xxp module, I believe. When I want to start Asterisk manually, I just type 'modprobe wct1xxp' and my two T1 cards are correctly started and then I can start asterisk w/ the normal commands and everything works. But, when I come back from a restart, it appears that the Asterisk startup failed, and I think it's b/c the wct1xxp module is not loaded. What is the recommended way to ensure this happens? I've been reading and found that modprobe (on startup, it appears) uses /etc/modules.conf, and here's what mine looks like: alias eth0 e1000 alias scsi_hostadapter megaraid alias usb-controller ehci-hcd alias usb-controller1 usb-uhci options torisa base=0xd alias char-major-196 torisa #post-install wcfxs /sbin/ztcfg #post-install wcfxsusb /sbin/ztcfg #post-install torisa /sbin/ztcfg #post-install tor2 /sbin/ztcfg #post-install wcfxo /sbin/ztcfg post-install wct1xxp /sbin/ztcfg #post-install wct4xxp /sbin/ztcfg (I commented out all of the modules I think I don't need, but it didn't work when they weren't commented out anyway). Does this have something to do w/ it? Do I need to add something to indicate that wct1xxp should be loaded on startup elsewhere? I appreciate your willingness to share your knowledge and expertise. jl ___ 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] auto 'modprobe wct1xxp' on startup?
On Tue, 7 Oct 2003, john lawler wrote: But, when I come back from a restart, it appears that the Asterisk startup failed, and I think it's b/c the wct1xxp module is not loaded. What is the recommended way to ensure this happens? I've been reading and found that modprobe (on startup, it appears) uses /etc/modules.conf, and here's what mine looks like: I've seen similar issues that seem to only happen when /usr is on a seperate filesystem than the root filesystem. I saw this happen on debian. Because /usr/lib/libtonezone.so.1 (or whatever its called) isn't available when ztcfg is run the command does not work. My solution was to have my asterisk startup script execute modprobe to load the module. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] auto 'modprobe wct1xxp' on startup?
On Tue, 2003-10-07 at 15:30, john lawler wrote: Hi guys, Thanks for your answers on my two questions yesterday. That's exactly what I was looking for, sorry for not noticing it myself, but I'm still getting acclimated to Asterisk and even Linux--from what I see so far, I love it. I've got another one now. Since my Asterisk install and configuration is fairly stable at this point, I'm interested it ensuring that during the event of a power failure, when the power returns (or if the machine is manually restarted) that Asterisk will successfully load on the other side (automatically). I've used the provided asterisk startup script (which I moved to /etc/rc.d/init.d) and RedHat's 'chkconfig' to make sure that Asterisk is started on bootup, but the problem I'm having has to do w/ the wct1xxp module, I believe. When I want to start Asterisk manually, I just type 'modprobe wct1xxp' and my two T1 cards are correctly started and then I can start asterisk w/ the normal commands and everything works. But, when I come back from a restart, it appears that the Asterisk startup failed, and I think it's b/c the wct1xxp module is not loaded. What is the recommended way to ensure this happens? I've been reading and found that modprobe (on startup, it appears) uses /etc/modules.conf, and here's what mine looks like: alias eth0 e1000 alias scsi_hostadapter megaraid alias usb-controller ehci-hcd alias usb-controller1 usb-uhci options torisa base=0xd alias char-major-196 torisa #post-install wcfxs /sbin/ztcfg #post-install wcfxsusb /sbin/ztcfg #post-install torisa /sbin/ztcfg #post-install tor2 /sbin/ztcfg #post-install wcfxo /sbin/ztcfg post-install wct1xxp /sbin/ztcfg #post-install wct4xxp /sbin/ztcfg (I commented out all of the modules I think I don't need, but it didn't work when they weren't commented out anyway). Does this have something to do w/ it? Do I need to add something to indicate that wct1xxp should be loaded on startup elsewhere? All that file does is explain what to do when loading the module. You can one of a couple of things. First you can edit the startup script to try and load the modules. This way is easy, but I'm not sure it is a good suggestion since it would try to reinsert the module when asterisk is restarted. You could make a new init.d file and link it in appropriately so that the module loading happens before asterisk startup. You could edit the /etc/modules file and list the appropriate drivers. This may be a debianism, but I don't think so. I hope you are getting the picture that there is a lot of ways to get this type of functionality accomplished under real operating systems. Have fun. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with questions for initial Asterisk wizard (GUI)
Olle E. Johansson wrote: Leif Madsen wrote: think of things I've missed, feel free to chime in! Leif, why don't you put the script up on the wiki so that we all can edit it and add on line with versioning? It's an interesting idea.. but I'm not sure if a wiki is the best place for code... For now, I haven't done any coding, I'm simply trying to come up with the logic for now so that when I start coding, I know where I'm going to be going with it. Thanks, -- +--+ |Leif Madsen - http://www.hacklocalhost.com| +--+ |@| leif at hacklocalhost dot com | | SMS| sms at hacklocalhost dot com | | FWD| 18924 IAX| 1700-363-0761 | |iptel| 8972-1969sipph| 1-747-386-1618 | +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] auto 'modprobe wct1xxp' on startup?
john lawler wrote: Hi guys, Thanks for your answers on my two questions yesterday. That's exactly what I was looking for, sorry for not noticing it myself, but I'm still getting acclimated to Asterisk and even Linux--from what I see so far, I love it. I've got another one now. Since my Asterisk install and configuration is fairly stable at this point, I'm interested it ensuring that during the event of a power failure, when the power returns (or if the machine is manually restarted) that Asterisk will successfully load on the other side (automatically). I've used the provided asterisk startup script (which I moved to /etc/rc.d/init.d) and RedHat's 'chkconfig' to make sure that Asterisk is started on bootup, but the problem I'm having has to do w/ the wct1xxp module, I believe. When I want to start Asterisk manually, I just type 'modprobe wct1xxp' and my two T1 cards are correctly started and then I can start asterisk w/ the normal commands and everything works. But, when I come back from a restart, it appears that the Asterisk startup failed, and I think it's b/c the wct1xxp module is not loaded. What is the recommended way to ensure this happens? I've been reading and found that modprobe (on startup, it appears) uses /etc/modules.conf, and here's what mine looks like: alias eth0 e1000 alias scsi_hostadapter megaraid alias usb-controller ehci-hcd alias usb-controller1 usb-uhci options torisa base=0xd alias char-major-196 torisa #post-install wcfxs /sbin/ztcfg #post-install wcfxsusb /sbin/ztcfg #post-install torisa /sbin/ztcfg #post-install tor2 /sbin/ztcfg #post-install wcfxo /sbin/ztcfg post-install wct1xxp /sbin/ztcfg #post-install wct4xxp /sbin/ztcfg (I commented out all of the modules I think I don't need, but it didn't work when they weren't commented out anyway). Does this have something to do w/ it? Do I need to add something to indicate that wct1xxp should be loaded on startup elsewhere? I appreciate your willingness to share your knowledge and expertise. jl This is the same problem I just had. Don't know if it's the best way, but it works. I created an executable file called rc.modules in my /etc/rc.d/ rc.modules- #!/bin/sh /sbin/modprobe wct4xxp --- and since the module needed to load before the init script called asterisk, I call the rc.modules file from within the rc.sysinit file (at the end of the file) /etc/rc.d/rc.modules boots with no problems now, otherwise asterisk would not just simply start by calling it from an init script. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Line going to Zombie
On Tue, 2003-10-07 at 15:30, Ariel Batista wrote: I have tried to release it with soft hangup Zap/1 also soft hangup Zap/1-2. If I use the last one is say trying to hang up but it never does. I have to shut the system down then back up! I am not able to run the GASTMAN due to I have no XP machine running. I only have Windows 2000 pro. I have seen this also. GASTMAN wouldn't have helped you since it would issue the same commands as you did. You can always use astman for similar functions without the gtk widgitry. I still don't understand! Sorry I am new to Asterisk. So there is no way to stop this channel unless we restart the system? I can't find any other documentation on this. I google the list and came up with not much information. I do not have astman loaded on the * box. astman comes with asterisk. It is a console based manager interface. If you had libnewt-dev on your system, it should have been built and installed for you. As for the problem of zombied channels. You should only have to restart asterisk at worst, not the whole system. I'm not sure what causes the zombied channels and am basically waiting for someone more qualified to finish answering your question. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Line going to Zombie
On Tue, 7 Oct 2003 16:30:52 -0400, Ariel Batista [EMAIL PROTECTED] wrote: astsvr*CLIZap/1-2ZOMBIE(macro-twoline-exten,s,1)Up Dial Zap/1-2|20|r I have tried to release it with soft hangup Zap/1 also soft hangup Zap/1-2. If I use the last one is say trying to hang up but it never does. [...] I still don't understand! Sorry I am new to Asterisk. So there is no way to stop this channel unless we restart the system? Hmm... what happens if you try soft hangup Zap/1-2ZOMBIE ? I occasionally run into Zombies, and that usually clears it up. (Hitting your tab key after the unique parts of the channel name will expand it out, BTW -- worth a shot) -rt -- Ryan Tucker Network Engineer NetAccess, Inc. 1159 Pittsford-Victor Road Bldg. 5, Suite 140 Pittsford, New York 14534 585-419-8200 www.netacc.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with SIP Client!
Ok I have the following on the Asterisk every minutes. Got SIP response 481 Call Leg/Transaction Does Not Exist back from 2XX.2XX.133.1XX. The user of this Sip phone is able to make calls and get calls. A Windows 2000 pro using MS Messenger! I loaded it on my PC as well and it does the same for my IP address! Is there some thing I need change on it! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] direct-inward-dialing (DID)
There's been a few replies but thought I'd elaborate on my initial reply.. How are you dropping the 456 there? I thought extensions picked up what either the SIP phone had dialled, or what DTMF detection picked up when * answered the line...? No.. if you have a PRI, the signalling is digital, no DTMFs there.. so Asterisk received the caller ID and dialed number as part of the call setup message. I should have explained in my example that I was assuming your telco was sending you 4 digit DNIS. The stuff I used to work on previously, we'd always ask for full 10 digit DNIS. Easier that way, you know exactly what's going on (and no possibility of clashes if you have DIDs from different exchanges). I'm looking at purchasing a PRI with 30 DIDs (can't get any fewer from Bell Canada) Out of curiosity, where are you located and what's the PRI cost? (I'm in Vancouver and looking to get a T1 in the very near future) and routing the calls coming in to multiple remote * boxes based on the called number. So a sort of central hub/switch, taking calls in then farming them out to remote * boxes over IP? Cheers Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Second Send: Using PCI backplane
I am wondering if it's possible to use a bunch of cards in a PCI backplane instead of going out to the extensions with T1 and then and adapter. How are people connecting to large amounts of extensions? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] agi exit problem
Wouldn't it be cool if you could take the call back, into the IVR script after the remote party hangs up ? Yeah, for the exact reasons you suggested! We had an IVR product that would allow you to zero out to the call centre, get some help, top up your account, whatever, then when the agent hung up the phone we'd plop you back in to the system exactly where you left off.. scored big points when we implemented it cos previously the user would have had to hung up and redialed.. Maybe we need a new DialAndReturn() function? :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Second Send: Using PCI backplane
You are wanting to use a PCI backplane and put a bunch of TDM400P FXS cards instead of a T1 and a channel bank? If that's what you're asking... A T1 card and a channel bank yield 24 extensions. If you figure the TDM400P is $305 for 4 extensions, it would cost $1830 to get enough FXS ports (not to mention the IRQ and other problems you may run into with that many cards). The T1 card is $495, leaving you $1335 to find a 24FXS channel bank (which is more than enough on eBay). A large number of extensions would be handled by way of T1s and channel banks - up to 96 channels on one pci card with the T400P (1 pci card and 4 channel banks). Out of curiosity, why are you reluctant/opposed to a T1 and channel bank? Steve On Tue, 7 Oct 2003, Dennis Gearon wrote: I am wondering if it's possible to use a bunch of cards in a PCI backplane instead of going out to the extensions with T1 and then and adapter. How are people connecting to large amounts of extensions? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Second Send: Using PCI backplane
Each card needs it's own IRQ, not shared with any other device (not even shared with other Digium cards). Adding a PCI backplane gives you more slots, but not more IRQs. On Tue, 2003-10-07 at 16:23, Dennis Gearon wrote: I am wondering if it's possible to use a bunch of cards in a PCI backplane instead of going out to the extensions with T1 and then and adapter. How are people connecting to large amounts of extensions? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call park on SIP phones
On Tue, 2003-10-07 at 16:09, Babak Pasdar wrote: Brian, Would you be kind enough to give me a brief overview of why it doesnt work. I also appreciate the work aorund. This is something I will have to educate my soon to be users on. We do a lot of conferencing of calls as a matter of facilitating clients' immediate needs. I still cannot park calls on my 7960, I have: - extensions.conf --- [demo] ; Juanjo exten = 8991,1,Dial(SIP/8991,20)|t exten = 8991,2,Voicemail2([EMAIL PROTECTED]) exten = 8991,102,Voicemail2([EMAIL PROTECTED]) exten = 8991,103,Hangup [local] ; ; Master context for local, toll-free, and iaxtel calls only ; ignorepat = 9 include = default include = parkedcalls include = trunklocal include = cell include = iaxtel700 include = trunktollfree include = iaxprovider -- parking.conf --- [general] parkext = 700 ; What ext. to dial to park parkpos = 701-720 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in - sip.conf [8991] type=friend username=8991 secret=secret nat=no ; This phone may be natted host=dynamic canreinvite=no ; Cisco poops on reinvite sometimes qualify=500 ; Qualify peer is no more than 200ms context=local [EMAIL PROTECTED] If I dial 700 I got busy tone (440 Not Found) the same happens if I dial #700 which I had to configure in dialplan.xml of the phone (rewriting 700 as #700). Any suggestions ? -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dynamic registration to flakey for production system
Three days after launching our * system with 20 GS phones, I have finally had to give up on dynamic registration. The phones keep dissappearing from the sip peers list, even if just sitting idle. Either I spend half my time re-booting phones to get them registered, or the extension appears busy to outside callers and people get really irritated. Even setting the registration interval to 5 minutes was not enough to guarantee that lines didn't go south. The workaround was to hard code the phone IP addresses in sip.conf and turn off SIP register in the phones. It may cost be a bit of securiity, but since the phones are all on 192.168, I don't think it is too much of a risk. However, if I was on a public network, or, if the phones were using DHCP for addressing, this could be a major headache. Has anyone else had this problem and if so, what was the solution? Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dynamic registration to flakey for production system
Three days after launching our * system with 20 GS phones, I have finally had to give up on dynamic registration. The phones keep dissappearing from the sip peers list, even if just sitting idle. Either I spend half my time re-booting phones to get them registered, or the extension appears busy to outside callers and people get really irritated. Even setting the registration interval to 5 minutes was not enough to guarantee that lines didn't go south. The workaround was to hard code the phone IP addresses in sip.conf and turn off SIP register in the phones. It may cost be a bit of securiity, but since the phones are all on 192.168, I don't think it is too much of a risk. However, if I was on a public network, or, if the phones were using DHCP for addressing, this could be a major headache. Has anyone else had this problem and if so, what was the solution? Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [PATCH] allow announcements in app_dial
Hi.
Since a customer requested us that feature, I wrote this
little patch for app_dial to allow to play an
announcement to the called party, as soon he answers.
you can define the file to play in the dial() option,
using A(filename).
for example:
exten = blah,1,Dial(Zap/blah,30,rA(/my/own/announce)Tt)
that doesn't break anything ...
feel free to blame me for anything bad this patch could do ;)
if for the list is ok, I'll submit to the bug tracker, under
a feature-request.
Matteo
--
Brancaleoni Matteo [EMAIL PROTECTED]
Espia - Emmegi Srl
--- asterisk/apps/app_dial.c2003-10-08 00:05:43.0 +0200
+++ dial-asterisk/apps/app_dial.c 2003-10-08 00:04:20.0 +0200
@@ -337,6 +337,7 @@
struct localuser *u;
char info[256], *peers, *timeout, *tech, *number, *rest, *cur;
char privdb[256] = , *s;
+ char announcemsg[256] = , *ann;
struct localuser *outgoing=NULL, *tmp;
struct ast_channel *peer;
int to;
@@ -344,8 +345,10 @@
int allowredir_out=0;
int allowdisconnect=0;
int privacy=0;
+ int announce=0;
int resetcdr=0;
int clearchannel=0;
+ int cnt=0;
char numsubst[AST_MAX_EXTENSION];
char restofit[AST_MAX_EXTENSION];
char *transfer = NULL;
@@ -419,6 +422,16 @@
} else if (strchr(transfer, 'C')) {
resetcdr = 1;
}
+ /* XXX ANNOUNCE SUPPORT */
+ else if ((ann = strstr(transfer, A())) {
+ announce = 1;
+ strncpy(announcemsg, ann + 2, sizeof(announcemsg) - 1);
+ cnt=0;
+ while(announcemsg[cnt] != ')') {
+ cnt++;
+ }
+ announcemsg[cnt]='\0';
+ }
}
if (resetcdr chan-cdr)
ast_cdr_reset(chan-cdr, 0);
@@ -670,6 +683,11 @@
ast_channel_setoption(chan,AST_OPTION_AUDIO_MODE,x,sizeof(char),0);
ast_channel_setoption(peer,AST_OPTION_AUDIO_MODE,x,sizeof(char),0);
}
+ if (announce announcemsg)
+ {
+ res = ast_streamfile(peer,announcemsg,peer-language);
+ res = ast_waitstream(peer,);
+ }
res = ast_bridge_call(chan, peer, allowredir_in, allowredir_out,
allowdisconnect | clearchannel);
if (clearchannel)
{
[Asterisk-Users] Is there always data at /dev/zap/1?
Hey all..in trying to futher troubleshoot my caller id problem I'm looking at some past troubleshooting tips and this struck me as strange: If I cat /dev/zap/1 I *always* see data...no matter if the line is in use or not...is that typical? Just curious... Thanks, Chris -- People are not homeless if they're sleeping in the streets of their own hometowns.- Dan Quayle http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call park on SIP phones
How are you transfering to 700? You dial # while in a call and then it says transfer and you then dial 700, or are you using a different method? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Juan J. Sierralta P. Sent: Tuesday, October 07, 2003 6:02 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Call park on SIP phones On Tue, 2003-10-07 at 16:09, Babak Pasdar wrote: Brian, Would you be kind enough to give me a brief overview of why it doesnt work. I also appreciate the work aorund. This is something I will have to educate my soon to be users on. We do a lot of conferencing of calls as a matter of facilitating clients' immediate needs. I still cannot park calls on my 7960, I have: - extensions.conf --- [demo] ; Juanjo exten = 8991,1,Dial(SIP/8991,20)|t exten = 8991,2,Voicemail2([EMAIL PROTECTED]) exten = 8991,102,Voicemail2([EMAIL PROTECTED]) exten = 8991,103,Hangup [local] ; ; Master context for local, toll-free, and iaxtel calls only ; ignorepat = 9 include = default include = parkedcalls include = trunklocal include = cell include = iaxtel700 include = trunktollfree include = iaxprovider -- parking.conf --- [general] parkext = 700 ; What ext. to dial to park parkpos = 701-720 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in - sip.conf [8991] type=friend username=8991 secret=secret nat=no ; This phone may be natted host=dynamic canreinvite=no ; Cisco poops on reinvite sometimes qualify=500 ; Qualify peer is no more than 200ms context=local [EMAIL PROTECTED] If I dial 700 I got busy tone (440 Not Found) the same happens if I dial #700 which I had to configure in dialplan.xml of the phone (rewriting 700 as #700). Any suggestions ? -- Juanjo sin .sig ___ 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] [PATCH] allow announcements in app_dial
He he ... too early
Thanks to a quick info from Mark on irc,
I've added the autoservice stuff on the other
channel, that's doing nothing meanwhile.
So here's the correct patch. discard the previous one.
Matteo
Il mer, 2003-10-08 alle 00:11, Brancaleoni Matteo ha scritto:
Hi.
Since a customer requested us that feature, I wrote this
little patch for app_dial to allow to play an
announcement to the called party, as soon he answers.
you can define the file to play in the dial() option,
using A(filename).
for example:
exten = blah,1,Dial(Zap/blah,30,rA(/my/own/announce)Tt)
that doesn't break anything ...
feel free to blame me for anything bad this patch could do ;)
if for the list is ok, I'll submit to the bug tracker, under
a feature-request.
Matteo
--
Brancaleoni Matteo [EMAIL PROTECTED]
Espia - Emmegi Srl
--- asterisk/apps/app_dial.c2003-10-08 00:05:43.0 +0200
+++ dial-asterisk/apps/app_dial.c 2003-10-08 00:25:19.0 +0200
@@ -337,6 +337,7 @@
struct localuser *u;
char info[256], *peers, *timeout, *tech, *number, *rest, *cur;
char privdb[256] = , *s;
+ char announcemsg[256] = , *ann;
struct localuser *outgoing=NULL, *tmp;
struct ast_channel *peer;
int to;
@@ -344,8 +345,10 @@
int allowredir_out=0;
int allowdisconnect=0;
int privacy=0;
+ int announce=0;
int resetcdr=0;
int clearchannel=0;
+ int cnt=0;
char numsubst[AST_MAX_EXTENSION];
char restofit[AST_MAX_EXTENSION];
char *transfer = NULL;
@@ -419,6 +422,16 @@
} else if (strchr(transfer, 'C')) {
resetcdr = 1;
}
+ /* XXX ANNOUNCE SUPPORT */
+ else if ((ann = strstr(transfer, A())) {
+ announce = 1;
+ strncpy(announcemsg, ann + 2, sizeof(announcemsg) - 1);
+ cnt=0;
+ while(announcemsg[cnt] != ')') {
+ cnt++;
+ }
+ announcemsg[cnt]='\0';
+ }
}
if (resetcdr chan-cdr)
ast_cdr_reset(chan-cdr, 0);
@@ -670,6 +683,19 @@
ast_channel_setoption(chan,AST_OPTION_AUDIO_MODE,x,sizeof(char),0);
ast_channel_setoption(peer,AST_OPTION_AUDIO_MODE,x,sizeof(char),0);
}
+ if (announce announcemsg)
+ {
+ int res2;
+ // Start autoservice on the other chan
+ res2 = ast_autoservice_start(chan);
+ // Now Stream the File
+ if (!res2)
+ res2 = ast_streamfile(peer,announcemsg,peer-language);
+ if (!res2)
+ res2 = ast_waitstream(peer,);
+ // Ok, done. stop autoservice
+ res2 = ast_autoservice_stop(chan);
+ }
res = ast_bridge_call(chan, peer, allowredir_in, allowredir_out,
allowdisconnect | clearchannel);
if (clearchannel)
{
Re: [Asterisk-Users] Dynamic registration to flakey for production system
I'm debugging SIP registration too. My next step is to install an Ethernet sniffer to log everything that goes ovr the wire using ports 5060 and 8000~8020. I'll soon know what's up. You would be able to see if the phones are forgetting to register, or if they are and Astrisk is dropping the data on the floor. Switched Ethernet means I have to run the sniffer on the server however or go and find a non-switching hub. --- Stephen R. Besch [EMAIL PROTECTED] wrote: Three days after launching our * system with 20 GS phones, I have finally had to give up on dynamic registration. The phones keep dissappearing from the sip peers list, even if just sitting idle. Either I spend half my time re-booting phones to get them registered, or the extension appears busy to outside callers and people get really irritated. Even setting the registration interval to 5 minutes was not enough to guarantee that lines didn't go south. The workaround was to hard code the phone IP addresses in sip.conf and turn off SIP register in the phones. It may cost be a bit of securiity, but since the phones are all on 192.168, I don't think it is too much of a risk. However, if I was on a public network, or, if the phones were using DHCP for addressing, this could be a major headache. Has anyone else had this problem and if so, what was the solution? Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Line going to Zombie
-- Original Message -- From: Ryan Tucker [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Tue, 07 Oct 2003 17:08:22 -0400 On Tue, 7 Oct 2003 16:30:52 -0400, Ariel Batista [EMAIL PROTECTED] wrote: astsvr*CLIZap/1-2ZOMBIE(macro-twoline-exten,s,1)Up Dial Zap/1-2|20|r I have tried to release it with soft hangup Zap/1 also soft hangup Zap/1-2. If I use the last one is say trying to hang up but it never does. [...] I still don't understand! Sorry I am new to Asterisk. So there is no way to stop this channel unless we restart the system? Hmm... what happens if you try soft hangup Zap/1-2ZOMBIE ? I occasionally run into Zombies, and that usually clears it up. (Hitting your tab key after the unique parts of the channel name will expand it out, BTW -- worth a shot) -rt I have tried the soft hangup. And sometimes it works. But there are times that the only way to get it off and the line working is to shutdown asterisk and then restart it! But Since I have over 50 people using it I have to wait till the end of the day to get this done! -- Ryan Tucker Network Engineer NetAccess, Inc. 1159 Pittsford-Victor Road Bldg. 5, Suite 140 Pittsford, New York 14534 585-419-8200 www.netacc.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
