Re: [Asterisk-Users] Channel Bank Woes...
On Saturday 17 January 2004 23:16, Brent Franks wrote: > I have a new Carrier Access Adit 600 and am having a problem dialing > out and receiving incoming calls. > > After the initial ring, the calling party will here nothing. There > is a silence, but the SIP phones that is supposed to ring will ring. > The calling party will not hear the sound of the ring that * sends > out on the line. Additionally, if someone picks up the receiver > there is no sound, but occasionally you can here something trying to > get through that sounds like tin-foil is over the MIC of the phone. What does your incoming context look like in extensions.conf? -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbee question
You can easily have an incoming call ring multiple extensions. You could also send the incoming call to an alternate extension. Paul Mahler mail:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp von Klitzing Sent: Saturday, January 17, 2004 11:34 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Newbee question Hi! > > in a large/ distributed environment users move about either office to > > office or branch to branch can they log in and have their virtual > > extension routed to the one they are on? > > http://bugs.digium.com/bug_view_page.php?bug_id=102 > > .. currently its not somthing that is supported by Asterisk, you may be > able to write you own application to support it if you need it.. I agree with the comments of JT on that bug report: You can rather easily arrange this going using DBGet and DBPut. Still it might be a good idea to create an example and present that on the wiki, even include it in the default extensions.conf. Apart from that you have the option to use a phone (like the SNOM) that allows for multiple users. But unless you take extra provisions you'll still need to register with _your_ server, and not the branch's server. Let's assume we divide our number space into physical phones and personal numbers: - 1000 to 1999 for physical phones - 2000 to 2999 for persons Now all you need to do is write a small set of macros that - sets my caller ID (with DBget) to 2555 when I call 2301 using phone 1022 - offers a tiny menu where I can say "I am 2555 and can be found at 1022" and store that with DBput. You'll have to notify the other server(s) using AGI about this change or - better - use a shared database. Your voicemail, however, will not move with you, so you'll still need to call your "home server" to retrieve that. Also the MWI function on phone 1022 will need extra treatment in sip.conf (or whatever channel you use) to adjust the mailbox= setting - ok, that's a bit ugly to fix, but maybe you can also do without vm while away, or use the web interface...? Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel Bank Woes...
Hello all, I have a new Carrier Access Adit 600 and am having a problem dialing out and receiving incoming calls. After the initial ring, the calling party will here nothing. There is a silence, but the SIP phones that is supposed to ring will ring. The calling party will not hear the sound of the ring that * sends out on the line. Additionally, if someone picks up the receiver there is no sound, but occasionally you can here something trying to get through that sounds like tin-foil is over the MIC of the phone. The Channel bank is grounded properly, and only has one 8 Port FXO card. Also, there is nothing in any of the PCI slots except the 1 T100P card. Sometimes calls will work great (it's very sporadic), and everything sounds really clear. Sometimes there will be no sound on the line, others when trying to place an outgoing call the dialtone will sound like it has a lot of static I have six phone lines from Verizon and the problem will occur on all of the phone lines, not just one. I have trouble-shooted this thing extensively and have hit a wall. I have attached my configurations (zaptel.conf, zapata.conf, and print config (from the CB)). Has anyone seen anything like this before? Any help or other avenues to try to trouble shoot would be greatly appreciated! Thanks, B ---/etc/zaptel.conf--- span=1,1,0,esf,b8zs fxsks=1-6 # BMZ has the option to add more cards / lines to the CB # Setup T1 Span 2 # span=2,1,0,esf,b8zs # fxoks=25-48 loadzone = us defaultzone=us ---/etc/asterisk/zapata.conf [channels] language=en ; Set the context for dialplan calling group=1 context=incoming ; FXO Cards use FXS signalling, lets use Kickstart signalling=fxs_ks ; Cancel that Echo echocancel = yes echocancelwhenbridged = yes busydetect=no callprogress=no channel => 1-6 ---ADIT CB 600 print config > print config - -Adit 600 configuration file -Created on 01/02/2002 at 07:13:02 -This file is valid for the following configuration only: - -CardType - -SLOT A T1x2 SW Version: 6.1.2 -SLOT 1 FXOx8 -SLOT 2 FXOx8 -SLOT 3 FXSx8 -SLOT 4 FXSx8 -SLOT 5 FXSx8 -SLOT 6 FXSx8 -NOTES: -1. It is necessary to issue the commands 'restore defaults' - and 'reset' BEFORE downloading the configuration file to - ensure proper configuration. -2. Lines beginning with '-' will be ignored as comments - by the CLI. Before downloading, review the sections of - the configuration file delimited by these comments and - delete the commands that are not needed (e.g. 'set ip - address' and 'add user' are likely candidates for - deletion). -3. While downloading, a character delay of 5 ms and a line - delay of 300 ms is recommended. - -Turning off verification messages. set verification off -Setting local off. set local off -Disconnecting all connections. disconnect a disconnect 1 disconnect 2 disconnect 3 disconnect 4 disconnect 5 disconnect 6 -Setting IP addresses. set ethernet ip address 10.10.1.100 255.255.0.0 set ip gateway 10.10.1.1 -Setting the SNMP MIB-II System Group objects. set snmp getcom "public" set snmp setcom "public" set snmp trapcom "public" set snmp trapauth enable set snmp trapevent all set snmp trapvers 2 -Setting slot a. set a:1 up set a:1 fdl none set a:1 lbo 1 set a:1 framing esf set a:1 id "CAC DS1# A:1" set a:1 linecode b8zs set a:1 loopdetect on set a:1:1-24 side drop set a:1:1-24 type voice set a:1:1-24 signal ls set a:2 down set a:2 fdl none set a:2 lbo 1 set a:2 framing esf set a:2 id "CAC DS1# A:2" set a:2 linecode b8zs set a:2 loopdetect on set a:2:1-24 side drop set a:2:1-24 type voice set a:2:1-24 signal ls -Setting slot 1. set 1:1-8 signal lscpd set 1:1-8 txgain -3 set 1:1-8 rxgain -3 -Setting slot 2. set 2:1-8 signal ls set 2:1-8 txgain -3 set 2:1-8 rxgain -6 -Setting slot 3. set 3:1-8 signal ls set 3:1-8 txgain -3 set 3:1-8 rxgain -6 set 3:1-8 linelength short -Setting slot 4. set 4:1-8 signal ls set 4:1-8 txgain -3 set 4:1-8 rxgain -6 set 4:1-8 linelength short -Setting slot 5. set 5:1-8 signal ls set 5:1-8 txgain -3 set 5:1-8 rxgain -6 set 5:1-8 linelength short -Setting slot 6. set 6:1-8 signal ls set 6:1-8 txgain -3 set 6:1-8 rxgain -6 set 6:1-8 linelength short -Setting primary and secondary clock sources. set clock1 internal set clock2 internal -Making connections. connect a:1:1 2:1 connect a:1:17-24 3:1-8 connect a:2:1-8 4:1-8 connect a:2:9-16 5:1-8 connect a:2:17-24 6:1-8 -Turning verification on. set verification on ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Integration with Lucent Definity g3si
Matthew Branton wrote: Come monday I will see if I can get the PRI line working if we have an extra 767 circuit pack. I promise that if/when we get this working I will definitely write up a detailed explanation of the steps involved. Right now we have a partial setup but a fully integrated box seems within reach... any more specifics would be great. Matt Matt, Been doing a little digging, so far found some good info on Cisco's web site, here's an example.. http://www.cisco.com/application/pdf/en/us/guest/products/ps259/c1237/ccmigration_09186a00801475be.pdf (The above url probally got split) Also go to Cisco's web site and do a search on "G3 migration" Lots of info on migrationg legacy G3 to voip and also some good war storys. There's also a very good definity forum that's very active with many Definity consultants willing to give a hand on all aspects of G3 system admin. http://www.tek-tips.com/gthreadminder.cfm/lev2/9/lev3/89/pid/690 Sounds like you're a step ahead of me on this, but the above forum is a good stop as soon as I dig through the manuals a little and brush up on my G3 talk. Let us all know how it goes, I'm sure it will work fine and just as soon as I can get to it, I will get it going. Ken ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Indications
Hi, Just wondering if someone could better explain how the indications.conf file actually affects Asterisk? I am using a Cisco 7940 from my Asterisk system, and have set in indications.conf “country=au” thinking that this would make the dialtones/call progress sound like the familiar Australian tones? However when I call another extension on my system, it still sounds like the American ring tone. Does the indications perhaps only effect Analog FXS cards and not SIP phones? Also, when loading the Asterisk configs as shown below, it displays a message about “Removed default indication country ‘au’ and at the end proceeds to set default indication country to ‘au’… the Removed part has me thinking it’s forgotten all about the particular indications for au? ==cut from Asterisk console=== -- Unregistered indication country 'us' Jan 18 14:02:36 NOTICE[262161]: indications.c:390 ast_unregister_indication_coun try: Removed default indication country 'au' -- Unregistered indication country 'au' -- Unregistered indication country 'fr' -- Unregistered indication country 'de' -- Unregistered indication country 'nl' -- Unregistered indication country 'uk' -- Unregistered indication country 'fi' -- Unregistered indication country 'no' == Parsing '/etc/asterisk/indications.conf': == Parsing '/etc/asterisk/indic ations.conf': Found -- Registered indication country 'us' -- Registered indication country 'au' -- Registered indication country 'fr' -- Registered indication country 'de' -- Registered indication country 'nl' -- Registered indication country 'uk' -- Registered indication country 'fi' -- Registered indication country 'no' -- Setting default indication country to 'au' == Thanks, Chris Lee
Re: [Asterisk-Users] New sounds also now in CVS
At 7:25 PM -0800 1/17/04, Ken Alker wrote: --On Saturday, January 17, 2004 8:49 PM -0500 John Todd <[EMAIL PROTECTED]> wrote: Ideas welcome for more text; I may have another timeslot with Allison early next week in which there will be some leftover room for additional words. Short phrases and meaningful sets of words for existing applications are desired; please don't give me words for apps that aren't even thought out yet. I don't know where to look to find out if these phrases already exist, so forgive me if they do. These are both used on my NorTel NAM II voice mail system for call transfer screening. Steve Murphy has written privacy features (not only thought out, but written) that could use these phrases, IMHO. For exact intonation of the below, Allison can dial 805/692-2323 and then x234. You'll hear the first two messages after dialing x234. To hear the third message, one must wait for the voice mail, wait for the BEEP, then *don't say anything* for a few seconds. "please record your name at the tone" "one moment please" "please speak louder, or speak directly into the telephone to ensure a clear recording" The first two phrases are complete, and in the CVS repository, and if installed on your system, in /var/lib/sounds/asterisk : one-pls-rcrd-name-at-tone.gsm moment-please.gsm The third will be done early next week, with any luck. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P Configs for Australia
Hi, Just wondering if anyone else in Australia is using the X100P to connect to the PSTN, and what configs they have for it? I’m finding at present when I make a call I get a fair bit of echo of myself speaking, and also the person on the other end can’t hear me very well (perhaps need to up the digial Tx Gain? I don’t have it configured at present) Asterisk is running on Slackware Linux 9.1 and I built from the latest CVS just last night (Saturday 17th Jan 04). The phone I’m using to call from is a Cisco 7940 running the SIP 6.0 firmware. If I make calls between the two Cisco 7940’s on my Asterisk system the voice quality is fine. The settings I have for now are:- Zapata.conf = [channels] language=en context=inbound-analog signalling=fxs_ks usecallerid=no immediate=no busydetect=no callprogress=no relaxdtmf=yes echocancel=yes echocancelwhenbridged=yes callerid=asreceived channel => 1 = Zaptel.conf = fxsks=1 loadzone=au defaultzone=au = Thanks, Chris Lee
Re: [Asterisk-Users] New sounds also now in CVS
I just found another thread showing where the files are hiding. Thank you for recording the first two below already - greatly appreciated. Please note that the third is a new one, however. For exact intonation of the below, Allison can dial 805/692-2323 and then x234. You'll hear the first two messages after dialing x234. To hear the third message, one must wait for the voice mail, wait for the BEEP, then *don't say anything* for a few seconds. DONE"please record your name at the tone" DONE"one moment please" PENDING "please speak louder, or speak directly into the telephone to ensure a clear recording" /** Ken Alker [EMAIL PROTECTED]ham radio: KA6SDU Impulse Internet Services http://www.impulse.net Santa Barbara, San Luis Obispo, Ventura, Los Angeles, Orange T-3 / T-1 / ADSL / ISDN / 56K / web hosting / wireless / co-lo ***/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Class features in dialplan ?
If what u mean by CLASS is Class of Service, ie: the ability to allow/denny access to users to/from resources like public network based on the number they dial, this can by nicely achieved by using a powerful tool that * calls "context". Playing with "contexts" you can define several different class of service levels that can be separately applied to every phone and will work independently of the type of technology of the phone (SIP, H323, IAX, Legacy, etc). Sam - Original Message - From: "Lance Arbuckle" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, January 16, 2004 8:58 PM Subject: [Asterisk-Users] Class features in dialplan ? > > hey guys > I thought I was making progress on my dialplan when I realized that the > class features that are available for zap channels aren't available for > SIP channels. I see references in the archives to adding pattern > matches in the dialplan for CLASS features which has raised a couple > questions. > > 1. Is implementing CLASS like features via the dialplan the currently > recommended way to do this ? > > 2. In general, are there any problems using non numeric characters in a > pattern match with SIP phones (i.e. _*67) ? > > So far I'm planning to do Call Forward Unconditional, Call Forward Busy, > Call Forward No-Answer, and Do not disturb and maybe some speed dials > but I haven't thought that one through yet. > > 3. Anyone willing to share some of their cool features that they've > come up with ??? I'd be most appreciative :) > > Thanks. > > -Lance > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New sounds also now in CVS
--On Saturday, January 17, 2004 8:49 PM -0500 John Todd <[EMAIL PROTECTED]> wrote: Ideas welcome for more text; I may have another timeslot with Allison early next week in which there will be some leftover room for additional words. Short phrases and meaningful sets of words for existing applications are desired; please don't give me words for apps that aren't even thought out yet. I don't know where to look to find out if these phrases already exist, so forgive me if they do. These are both used on my NorTel NAM II voice mail system for call transfer screening. Steve Murphy has written privacy features (not only thought out, but written) that could use these phrases, IMHO. For exact intonation of the below, Allison can dial 805/692-2323 and then x234. You'll hear the first two messages after dialing x234. To hear the third message, one must wait for the voice mail, wait for the BEEP, then *don't say anything* for a few seconds. "please record your name at the tone" "one moment please" "please speak louder, or speak directly into the telephone to ensure a clear recording" JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users /** Ken Alker [EMAIL PROTECTED]ham radio: KA6SDU Impulse Internet Services http://www.impulse.net Santa Barbara, San Luis Obispo, Ventura, Los Angeles, Orange T-3 / T-1 / ADSL / ISDN / 56K / web hosting / wireless / co-lo ***/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Remote reloading Cisco phones...
Here's a simple small expect script ... I call it "phreboot", usage: phreboot IP $ phreboot 10.99.1.1 -- cut here -- #!/usr/bin/expect -f set timeout -1 spawn $env(SHELL) match_max 1 send -- "telnet [lrange $argv 0 0]\r" expect -exact "word :" send -- "cisco\r" expect -exact "Phone> " send -- "reset\r" send -- "" ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Registering multiple FWD accounts
Can multiple FWD accounts be
registered?
I have the following output in my sip.conf
file:
register=74928:[EMAIL PROTECTED]/74928register=75160:[EMAIL PROTECTED]/75160register=74573:[EMAIL PROTECTED]/74573
[fwd-74928]type=friendsecret=xxxusername=74928host=fwd.pulver.com
[fwd-75160]type=friendsecret=xxxusername=75160host=fwd.pulver.com
[fwd-74573]type=friendsecret=xxxusername=74573host=fwd.pulver.com
Two questions :
What exactly does [fwd-74573] mean in this case
anyway? Does this category define a local users which another client can then
logon to (e.g. I don't really need this) - or is this actually necessary to get
FWD working? I notice in online examples that people use [fwd.pulver.com] - but
I gather I can't have three of these as it would conflict. The register=74928:[EMAIL PROTECTED]/74928
clause already does the logging in surely - so
are the [] required?
Secondly, using the above setup - when an incoming
call comes in to 74928 I get the following output:
-- Executing Answer("SIP/74573-aeb6",
"") in new stack -- Executing Wait("SIP/74573-aeb6", "1")
in new stack -- Executing Playback("SIP/74573-aeb6",
"welcomemsg") in new stack -- Playing 'welcomemsg'
(language 'en')
- which is strange, considering it was 74928 that
was dialled!! However, the extension correctly associated with 74928 does ring
(even when called from a true FWD client outside of this asterisk server). The
other extensions work fine too. So it almost seems like FWD is correctly
registered three times... but just the cosmetic log output is
wrong.
Is there actually a problem with this?
My extensions.conf is attached below for
reference.
Thanks,
Terence
-
exten => 74928,1,Answerexten
=> 74928,2,Wait(1)exten => 74928,3,Playback(welcomemsg)exten =>
74928,4,Dial(SIP/TerenceParker&SIP/DavidLiu&SIP/GardenVista&SIP/PeterLiu,90,tm)exten
=> 74928,5,Voicemail,u999exten => 74928,6,Hangupexten =>
74928,102,Voicemail,b999exten => 74928,103,Hangup
exten => 75160,1,Answerexten
=> 75160,2,Dial(SIP/DavidLiu,180,tm)exten =>
75160,3,Voicemail,u1000exten => 75160,4,Hangupexten =>
75160,102,Voicemail,b1000exten => 75160,103,Hangup
exten => 74573,1,Answerexten
=> 74573,2,Dial(SIP/TerenceParker,180,tm)exten =>
74573,3,Voicemail,u1001exten => 74573,4,Hangupexten =>
74573,102,Voicemail,b1001exten =>
74573,103,Hangup
[Asterisk-Users] Unavailable versus private in extensions.conf?
Hi I'm using an X101P as a Asterisk based answering machine and it works great. Thanks. Is there a way to differentiate a caller ID of "private" versus "unavailable" in the extensions.conf file? I can not find anything in the docs that says it or isn't possible. thanks in advance, Bob Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Integration with Lucent Definity g3si
Come monday I will see if I can get the PRI line working if we have an extra 767 circuit pack. I promise that if/when we get this working I will definitely write up a detailed explanation of the steps involved. Right now we have a partial setup but a fully integrated box seems within reach... any more specifics would be great. Matt On Jan 16, 2004, at 10:47 PM, Ken Godee wrote: PBXtech wrote: We have our G3R setup on a PRI connection. Your trunk group should be set to "tie". If anybody would be willing to share just alittle more info on how to set this up it would be great. I've just started thinking about this also. A brief outline would be great, circuit packs used, ds1 settings, trunk group settings and how are you guys setting up the private network to route calls through to/from * ? There's just not a whole lot of info out there on this and I'd rather cut my left nut off, rather than try to talk to Avaya about this. I noticed a spot on the wiki for just this thing, but no ones contributed. http://www.voip-info.org/wiki-Asterisk+legacy+integration Either it's simpler or more complex than I'm making it * via TE410P ISDN-PRI -> G3 ISDN-PRI DS1 TN767E (dependancy circuit packs in place, via existing ISDN-PRI) define G3 DS1 assign (tie) trunk group to the G3 DS1 A.)How should one route calls through the G3 out to extentions defined in *? B.) How should one route calls from the * server to/through the G3's ext.s and outbound lines? Any info would be great, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playing background message
Thanks for the replies I've decided to simply add 'm' to the dialplan for now, but i'll investigate call queues later - this sounds like the ideal setup for me though. For the meantime though, music on hold works fine! Thanks again. Terence > I agree, I'd rather have the caller hear ringing instead of MOH as > ringing gives the caller some feedback as to what is happening. I'd > save the music until they've talked to someone or heard a message and > are put on hold or get dumped into a queue. > > -Lance ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zone Paging
> I see a lot of chatter in the archives about intercom and paging, but has anyone addressed zone paging? Each of the 50 telephones in a large clinic would be members of one or more paging zones. Someone could then page Dr. X in zone #1. Would this be possible with analog phones? SIP? Summarizing from memory only, I believe this has been discussed more then once on the list and usually comes down to... what device(s) provide an auto-answer extension and includes an audio output jack that can be plugged into a paging amplifier? The two most common suggestions (from memory again) has been: 1. Use of the sound card on the asterisk machine (which sort of implies a limit of one paging zone), or, 2. Use a sip phone (Cisco 7960 with v6 as one example), configure the phone to support auto-answer, and connect the external headset to the paging amp. (Implies one sip phone per paging zone.) I've not tried either, so not sure of success/failure rates or problems. Seems like a fair number of people have problems getting the sound card to play nicely with asterisk, and most of the chatter seems to be oriented around sound card driver issues, etc. Don't know if the ata-186 supports auto-answer in current software, but if it did, jury-rigging a matching transformer as a source of zone audio would not seem like it would be very difficult. (Anyone know whether the 186 can be configured for auto answer?) Rich I have experimented with this in tentative ways in the following manner with Cisco 7960 phones with the new 6.1 SIP image which has the ability to auto-answer: - configure members of your group with 7960 phones such that they have a line that auto-answers. If you want to be clever, you can make it so that any outbound calls made on that key will lead to your paging extension ( use _. ) - create a special "paging" extension, ringable from other lines - when the paging extension is called, it runs a short AGI script that dumps out several .call files that call the members of the paging group, and sends them all to a conference call. The call duration is forced to be 20 seconds (AbsoluteTimeout) and all the called parties are set to mute. The caller makes their <20 second announcement, and hangs up. The other phones go silent, and after the 20 second timer, they are hung up. - the caller then has to pause for a few seconds to wait for RTP to catch up; this is a minor inconvenience, and users hopefully will quickly figure out that they need to pause for a few seconds before they start talking. - you may experience strange out-of-sync jitter if the phones are turned up, since each station will be firing up a separate RTP stream to the * server. This is meant for quiet, at-the-desk paging instead of the kind of blaring-trumpets-and-cannons paging you get at somewhere like a machine shop or an auto body garage JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New sounds also now in CVS
The soundfiles I submitted earlier today have been cleaned up, and added to the Digium CVS server in a more formal manner. Also, some of the really bad formatting in my .txt description file has been rectified. All of the sounds on my website are now on the Digium site, and I will be submitting future changes via patches to Digium for additional sounds. Ideas welcome for more text; I may have another timeslot with Allison early next week in which there will be some leftover room for additional words. Short phrases and meaningful sets of words for existing applications are desired; please don't give me words for apps that aren't even thought out yet. [follow the instructions on http://www.asterisk.org/index.php?menu=download, then add this: ] # mkdir asterisk-sounds # cvs checkout asterisk-sounds # cd asterisk-sounds; make install JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Remote reload Cisco 7960
Their is no need to telnet with perl you can just shove a notify packet down the cisco and let it reboot on its own. Really easy. bkw On Sat, 17 Jan 2004, Steven Critchfield wrote: > On Fri, 2004-01-16 at 15:26, B. J. Bomar wrote: > > Yes, I was wanting to do it via a script, but telneting in will work as a > > stop gap. > > So learn how to telnet with perl. > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Walker Haddock > > Sent: Friday, January 16, 2004 13:46 > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] Remote reload Cisco 7960 > > > > > > On Fri, Jan 16, 2004 at 12:13:19PM -0600, B. J. Bomar wrote: > > > Does anyone have a working way of having a Cisco 7960 reload its config > > > remotely. I have tried some of the scripts that I have found on the web, > > > but to no avail. Thanks for the help. > > > > I just telnet to it and then enter the `reset` command. Are you trying to > > do it automatically from a script? > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_odbc not logging integers eg duration
Check your tables. I logged everything as integer. set verbose 10 and make a call and watch it.. then do reload and watch the output. It will unload and reload and you can check to make sure your accually connetcing to the database. bkw On Sat, 17 Jan 2004, Iain Stevenson wrote: > > I've just noticed that since swapping from the direct mysql cdr driver to > cdr_odbc, the call duration (and anything else that's an integer) isn't > logged - anyone else seen this and know the reason. The cdr_odbc driver > gives no error messages and records any string based fields correctly. > > Iain > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cdr_odbc not logging integers eg duration
I've just noticed that since swapping from the direct mysql cdr driver to cdr_odbc, the call duration (and anything else that's an integer) isn't logged - anyone else seen this and know the reason. The cdr_odbc driver gives no error messages and records any string based fields correctly. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec matching weirdness
On Saturday 17 January 2004 15:44, Olle E. Johansson wrote: > Dustin Goodwin wrote: > > I did find something interesting. If you set reinvite=yes then * > > can setup the RTP stream so that it avoids the media proxy in the * > > box completely. I haven't tested to see if it changes anything. > > Can we please kill "reinvite" - it does not exist in the SIP channel > as an option for anything. Period. Bugnote 873 -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec matching weirdness
> Can we please kill "reinvite" - it does not exist in the SIP channel as an > option for anything. Period. > > There is an option called "canreinvite" that you can set to yes or no. > Setting "reinvite" to anything will not change anything at all. Olle, I thought I was the only that was loosing it with folks interchanging the two options. ;) Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec matching weirdness
Maybe someone will write a patch to print an error to the console if reinvite= is found in the config file.? On Sat, 2004-01-17 at 15:44, Olle E. Johansson wrote: > Dustin Goodwin wrote: > > > I did find something interesting. If you set reinvite=yes then * can > > setup the RTP stream so that it avoids the media proxy in the * box > > completely. I haven't tested to see if it changes anything. > > > Can we please kill "reinvite" - it does not exist in the SIP channel as an > option for anything. Period. > > There is an option called "canreinvite" that you can set to yes or no. > Setting "reinvite" to anything will not change anything at all. > > However, setting "canreinvite" to something will change ASterisk's > behaviour during a SIP call. It may also break your conversation > if your SIP device does not support the SIP re-invite mechanism. > > Please read: > http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+canreinvite > for more information. > > /Olle > > PS. I know that the "reinvite" option is mentioned in many archived > e-mails, which does not help at all. Please do not add any more messages > with this option, as it will only confuse users. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Go to http://www.digium.com/index.php?menu=documentation and look at the "Unofficial Links" section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the "Asterisk Resource Pages". BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re Grandstream 1.0.4.38
> > I got 1.0.4.38 from SIPphone's server at 130.94.123.253 but last time > I > > tried it was offering 1.0.4.35. For me 1.0.4.38 cleared all my > problems > > but from SIPphone's email it had hosed some phones, such that they > were > > talking about replacement units. > > I'm still on fscking 1.0.3.81, and I can't get anything from the > server > above... > can someone tell me how I can get a newer version? that's the latest > on > gs's pages as well, and it's really really buggy > > roy Another nice Asterisk user posted 1.0.4.39 on their website. I really like it. I haven't had much time to test it, but it is good so far. http://www.supercomputo.com/b13p4.39.zip Kevin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec matching weirdness
Dustin Goodwin wrote: I did find something interesting. If you set reinvite=yes then * can setup the RTP stream so that it avoids the media proxy in the * box completely. I haven't tested to see if it changes anything. Can we please kill "reinvite" - it does not exist in the SIP channel as an option for anything. Period. There is an option called "canreinvite" that you can set to yes or no. Setting "reinvite" to anything will not change anything at all. However, setting "canreinvite" to something will change ASterisk's behaviour during a SIP call. It may also break your conversation if your SIP device does not support the SIP re-invite mechanism. Please read: http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+canreinvite for more information. /Olle PS. I know that the "reinvite" option is mentioned in many archived e-mails, which does not help at all. Please do not add any more messages with this option, as it will only confuse users. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec matching weirdness
I did find something interesting. If you set reinvite=yes then * can setup the RTP stream so that it avoids the media proxy in the * box completely. I haven't tested to see if it changes anything. - Dustin - [EMAIL PROTECTED] wrote: I am experiencing a problem that from list archive it appears others are running into. When I dial from Cisco 7960 via the * to Free World Dialup destinations that supports G.729 the call fails. The major error from the debug log is Jan 15 00:11:14 NOTICE[22545]: channel.c:1481 ast_set_read_format: Unable to find a path from G729A to ULAW Jan 15 00:11:14 NOTICE[22545]: channel.c:1451 ast_set_write_format: Unable to find a path from ULAW to G729A Me too? I've been wondering the same thing. I asked before and didn't really get anywhere either. Kevin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New sounds posted
So, per the discussion last week and generous donations, we have some new sound files with which to work. The sounds are located in: http://www.loligo.com/asterisk/sounds/ For those of you who just want to download the _new_ sounds, please fetch: http://www.loligo.com/asterisk/sounds/20040117.newsounds.tar All of the sounds in that tarball are also in the main ../sounds/ directory in individual .gsm format. I've created a directory called ../sounds/AIF which contains the high-quality versions of the new sounds that I'm clipping apart, for those of you who want to process them in a different way. Typical users do not need to download or work with these files. There was a balance left over from the donations for this sound file creation. The good news is that the people at Digium will shortly receive (as noted, "at John's discretion") a wide array of unusual and fun foodstuffs, possibly much of it caffeinated to ensure long hours of * hacking. :-) As soon as it's received by Digium, I'll notify the list of the final contents - wouldn't want to ruin the surprise! The next round of ideas for words is open; send your ideas to me and I'll try to make them part of the next cycle, or contact Allison ([EMAIL PROTECTED]) yourself if you have commercial and/or non-BSD license utterances to be recorded. Phrases included in this iteration are below. JT a%a b%b c%c d%d e%e f%f g%g h%h i%i j%j k%k l%l m%m n%n o%o p%p q%q r%r s%s t%t u%u v%v w%w x%x y%y z%z %National Weather Service%national-weather-service %Persons in the immediate path of the%persons-in-path-of %Please enter the conference call number for the conference you wish to join.%enter-conf-pin-number %Please enter the conference pin number.%enter-conf-pin %Sorry, but the user's mailbox can't accept more messages. %sorry-mailbox-full %That pin is invalid for this conference.%pin-invalid %abandons% abandons %afternoon% afternoon %ampersand% ampersand %approximately% approximately %are advised to seek shelter immediately% advised-to-seek-shelter %at sign% at-sign %average speed of answer%avg-speed-answer %backslash%backslash %bar%bar %barometric%barometric %beaufort%beaufort %calls%calls %celsius%celsius %chance of%chance-of %changing%changing %clear%clear %clearing%clearing %close parenthesis%close-parenthesis %clouds%clouds %cloudy%cloudy %comma%comma %cyclone%cyclone %dash%dash %degrees%degrees %dollar%dollar %doppler radar%doppler-radar %dot%dot %early%early %east%east %easterly%easterly %equals%equals %euro%euro %euros%euros %evening%evening %exclamation point%exclaimation-point %fahrenheit%fahrenheit %falling%falling %fast%fast %fiftieth%fiftieth %fog%fog %foggy%foggy %followed by%followed-by %fortieth%fortieth %freeze%freeze %freezing%freezing %gale%gale %greater than%greater-than %hail%hail %half%half %has issued a%has-issued-a %hecto-pascal%hectopascal %high%high %humidity%humidity %hurricane%hurricane %ice%ice %icy%icy %in the%in-the %indicated%indicated %kilometers per hour%kilometers-per-hour %knots%knots %late%late %later%later %left bracket%left-bracket %less than%less-than %lightning%lightning %local authorities%local-authorities %low%low %maximum%maximum %meter%meter %meters%meters %miles per hour%miles-per-hour %minimum%minimum %mist%mist %misty%misty %morning%morning %mostly%mostly %moving%moving %nautical miles%nautical-miles %north%north %northerly%northerly %one moment, please.%one-moment-please %open parenthesis%open-parenthesis %partially%partially %partly%partly %pascal%pascal %pascal%pascal2 %patchy%patchy %percent%percent %period%period %pipe%pipe %please record your name at the tone.%pls-rcrd-name-at-tone %plus%plus %pounds%pounds %pressure%pressure %quarter%quarter %quickly%quickly %quote%quote %rain%rain %rainy%rainy %right bracket%right-bracket %rising%rising %scattered%scattered %severe%severe %sighted%sighted %slash%slash %sleet%sleet %sleeting%sleeting %slow%slow %slowly%slowly %snow%snow %snowy%snowy %south%south %southerly%southerly %space%space %staffing%staffing %sterling%sterling %storm%storm %sun%sun %sunny%sunny %temperature%temperature %thunderstorm%thunderstorm %tide%tide %tornado%tornado %turning to%turning-to %typhoon%typhoon %warning%warning %watch%watch %weather%weather %west%west %westerly%westerly %wind%wind %windy%windy %with%with %zed%zed %Alpha%alpha %Bravo%bravo %Charlie%charlie %Delta%delta %Echo%echo %Foxtrot%foxtrot %Golf%golf %Hotel%hotel %India%india %Juliet%juliet %Kilo%kilo %Lima%lima %Mike%mike %November%november %Oscar%oscar %Papa%papa %Quebec%quebec %Romeo%romeo %Sierra%sierra %Tango%tango %Uniform%uniform %Victor%victor %Whiskey%whiskey %Xray%xray %Yankee%yankee %Zulu%zulu %Niner%niner %So, _you_ sound cute.%you-sound-cute %So, do you dial here often?%dial-here-often %What are you wearing?%what-are-you-wearing %giggle%giggle1 %Oooo! Is that a telephone in your pocket, or are you just happy to see me?%telephone-i
Re: [Asterisk-Users] Newbee question
Hi! > > in a large/ distributed environment users move about either office to > > office or branch to branch can they log in and have their virtual > > extension routed to the one they are on? > > http://bugs.digium.com/bug_view_page.php?bug_id=102 > > .. currently its not somthing that is supported by Asterisk, you may be > able to write you own application to support it if you need it.. I agree with the comments of JT on that bug report: You can rather easily arrange this going using DBGet and DBPut. Still it might be a good idea to create an example and present that on the wiki, even include it in the default extensions.conf. Apart from that you have the option to use a phone (like the SNOM) that allows for multiple users. But unless you take extra provisions you'll still need to register with _your_ server, and not the branch's server. Let's assume we divide our number space into physical phones and personal numbers: - 1000 to 1999 for physical phones - 2000 to 2999 for persons Now all you need to do is write a small set of macros that - sets my caller ID (with DBget) to 2555 when I call 2301 using phone 1022 - offers a tiny menu where I can say "I am 2555 and can be found at 1022" and store that with DBput. You'll have to notify the other server(s) using AGI about this change or - better - use a shared database. Your voicemail, however, will not move with you, so you'll still need to call your "home server" to retrieve that. Also the MWI function on phone 1022 will need extra treatment in sip.conf (or whatever channel you use) to adjust the mailbox= setting - ok, that's a bit ugly to fix, but maybe you can also do without vm while away, or use the web interface...? Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware for Asterisk
On Fri, 2004-01-16 at 16:55, Robert L Mathews wrote: > At 1/16/04 7:25 AM, Andrew Kohlsmith <[EMAIL PROTECTED]> > wrote: > > >That's pure bullshit -- I use software RAID *specifically* because I value > >my data. I don't want to buy two hardaware RAID controllers to have one > >sit on the shelf just in case the first dies... and if the second dies > >you're SOL because they've lasted long enough that they're no longer > >available. Linux software RAID is available on any Linux system and if the > >system blows up I can put the drives in another system and *not* worry > >about it not being detected. > > Yeah, I couldn't agree more. > > We originally thought hardware RAID was the way to go, and we bought a > couple of fully loaded Dell PowerEdge 2550s with SCSI hardware RAID 5 > arrays at about $4500 a pop. We also bought a PowerEdge 600SC for around > $900 with lots of disk space to use as a network backup machine (backing > up the 2550s) with Linux software RAID 5. I've also had a crappy old > desktop machine running Linux software RAID 1 for a couple of years. > > It turns out that the software RAID is just as reliable (more so, in fact > -- we have had a number of lockups on the 2550s that appear to be due to > the hardware RAID subsystem locking up, and the software RAID machines > have never done that, even though the backup server does more disk I/O > than the others). The software RAID on the 600SC is faster than the > hardware RAID in bonnie tests. I believe there is a recall option on those machines. So far no one has identified what exactly is the problem there. I was reading the aac-raid list for a while, some people point the finger at the firmware on the disks, and some at the drivers. Either way, there are a few machines that Dell acknoledges trouble with. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Codec matching weirdness
> I am experiencing a problem that from list archive it appears others are > > running into. When I dial from Cisco 7960 via the * to Free World > Dialup > destinations that supports G.729 the call fails. The major error from > the debug log is > > Jan 15 00:11:14 NOTICE[22545]: channel.c:1481 ast_set_read_format: > Unable to find a path from G729A to ULAW > Jan 15 00:11:14 NOTICE[22545]: channel.c:1451 ast_set_write_format: > Unable to find a path from ULAW to G729A Me too? I've been wondering the same thing. I asked before and didn't really get anywhere either. Kevin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Remote reload Cisco 7960
On Fri, 2004-01-16 at 15:26, B. J. Bomar wrote: > Yes, I was wanting to do it via a script, but telneting in will work as a > stop gap. So learn how to telnet with perl. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Walker Haddock > Sent: Friday, January 16, 2004 13:46 > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Remote reload Cisco 7960 > > > On Fri, Jan 16, 2004 at 12:13:19PM -0600, B. J. Bomar wrote: > > Does anyone have a working way of having a Cisco 7960 reload its config > > remotely. I have tried some of the scripts that I have found on the web, > > but to no avail. Thanks for the help. > > I just telnet to it and then enter the `reset` command. Are you trying to > do it automatically from a script? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk/X100 - Sipura Configuration
Title: Asterisk/X100 - Sipura Configuration I have a Sipura behind my Asterisk Box. I have a X100 card in the box. Calls are coming in ok. When I try to configure my extensions for dialing out, I can dial 9X and then get a fast busy. I followed an example from moxilla on the Spirua and Asterisk before but I am questioning the dialing plan in the Sipura causing my problems. Has anyone got the above configuration working? I would like to be able to (for now) I will worry about voicepulse, fwd, iax, later. Just dial: 9XXX 9XX 91XX And have all traffic go out over my X100 from either line of my Sipura. Thanks in advance. Steve
Re: [Asterisk-Users] Newbee question
Chris Lee wrote: I am new to asterisk and am wanting to know if it can do some things: in a large/ distributed environment users move about either office to office or branch to branch can they log in and have their virtual extension routed to the one they are on? naturaly this implies the question: if branch servers are used can they ceep track of where a virtual extension is currently attached? This is somthing that I said was needed a while ago.. you can view the thread here.. http://bugs.digium.com/bug_view_page.php?bug_id=102 .. currently its not somthing that is supported by Asterisk, you may be able to write you own application to support it if you need it.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbee question
I am new to asterisk and am wanting to know if it can do some things: in a large/ distributed environment users move about either office to office or branch to branch can they log in and have their virtual extension routed to the one they are on? naturaly this implies the question: if branch servers are used can they ceep track of where a virtual extension is currently attached? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 Phone disconnects when dialing using speaker
Quoting Brian West <[EMAIL PROTECTED]>: > Works fine here.. got two of em. > > bkw Hmpf! I donno whats wrong then, both phones do the same thing. So you can keep the headset in the cradle, hit the 'speaker' button, dial a call and it doesn't disconect ? I wonder, are you using an xml dial plan or anything on you phones ? Thanks -bh > > On Fri, 16 Jan 2004, Bill Hamel wrote: > > > Hi, > > > > Just got some CISCO 7960 phones. They seem to work great except if I make > any > > SIP call using the speaker phone (leaving the hand set in the cradle)the > call > > will disconnect in about 5 or so seconds. If I pick up the hand set and > make a > > call, it's fine. > > > > Has anyone else run into this ? Any solution ? > > > > The phone is on SIP v6.1 - it did the same thing on 4.4 5.0 and 6.0. > > > > Thank you in advance, > > -bh > > > > -- > > > > > > > > This message was sent using IMP, the Internet Messaging Program. > > This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Again: 7920 Cisco IP Phone Skinny & SIP
Hi Jan, >in the sccp_registration i would then handle the registration for the >7920 how the callmanager is behaving. I've just gotten one step further with my 7920: Got it successfully registered to asterisk. Still doesn't actually work, but definitely a step in the right direction. The problem is that the 7920 expects SelectSoftKeys Messages to finish its setup - probably it sends the offhook/onhook sequence to trigger these. Asterisk however doesn't send these until it's finished registering the device on reception of a message of type 2d - which the 7920 never sends. Hack/workaround, against current cvs: diff -urN chan_sccp/sccp_actions.c chan_sccp.mbe/sccp_actions.c --- chan_sccp/sccp_actions.c2004-01-17 11:18:36.0 +0100 +++ chan_sccp.mbe/sccp_actions.c2004-01-17 17:14:15.0 +0100 @@ -94,7 +94,7 @@ } } - sccp_dev_set_registered(d, RsProgress); + sccp_dev_set_registered(d, RsOK); d->currentLine = d->lines; REQ(r1, RegisterAckMessage); This probably breaks all kind of other things, but it does let the 7920 register with asterisk. Next Problem: Softkeys need special attention, currently the onhook/offhook keys aren't mapped correctly, so you can't actually accept a call or hang up :-) Dialing works bzw. Bye, Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Combining 2 AUDIO Frames
Hello Does anybody has a good ideea for combining 2 audio frames (from 2 different channels) in order to send them to an 3rd channel ? I can't send them both cause it seems that i'm overflowing the 3rd channels and it drops the 2nd frame (kind of logic, though).. I have two channels , "bridged" by a loop , and I have access to frames from each channels.. Thanks Alex
RE: [Asterisk-Users] Zone Paging
There are a number of paging interfaces available which connect to a regular phone line on one side and to a paging amplifier on the other side. Use one or more of these - connect them to FXS cards either Digiums 4xFXS or a channel bank and make the extensions your paging zones in the dialplan. Regards Alfred R. Nurnberger --- F L O S Y S Making Communications Flow Tel: +1 (503) 972-9300 Fax: +1 (503) 972-9309 US Toll Free: 1-877-4FLOSYS h323: 208.187.136.227 http://www.flosys.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Welter Sent: Saturday, January 17, 2004 6:06 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Zone Paging I see a lot of chatter in the archives about intercom and paging, but has anyone addressed zone paging? Each of the 50 telephones in a large clinic would be members of one or more paging zones. Someone could then page Dr. X in zone #1. Would this be possible with analog phones? SIP? Thanks, Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Long time to detect that an IAX2 user is not logged on
I notice that when my dial plan directs a call to an IAX user on my own asterisk server that is not logged in, it waits the full dial time before skipping to voice mail. Why is that and is there any way to get it to skip straight away if the person is not logged in? - Kim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ultra-cheap asterisk box
You can get a DELL 400SC server for $400 or so. www.dell.com Check out the refurbished units, although lately the new ones have been a better deal. Paul Paul Mahler -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aram Ter-Martirosyan Sent: Friday, January 16, 2004 6:22 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] ultra-cheap asterisk box I don't need to use the lowest end server for asterisk, but something reasonable. I need to put one TE410P and 2 TDM400P boards in it. We have about 20 heavy telephone users in the company. Can someone suggest reasonable priced and reliable box? Thanks, Aram Ter-Martirosyan Senior Account Manager Hi-Tech Gateway, Inc. http://www.hi-teck.com 1225 Grand Central Ave. Glendale, CA 91201 [EMAIL PROTECTED] tel 818.546.4601 fax 818.546.4617 Turning Technology Into Business Solutions -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James H. Thompson Sent: Friday, January 16, 2004 5:44 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ultra-cheap asterisk box FAQ for Dell 400SC: http://www.aaltonen.us/forums/viewtopic.php?t=8 Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: "calvis" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, January 16, 2004 10:33 AM Subject: RE: [Asterisk-Users] ultra-cheap asterisk box I got in on the same Dell deal I think. You must hang out on the bargain boards just like I do? I hang out mainly at fatwallet.com. This is the thread that I got in on the Dell machines that I just recently purchased. http://www.fatwallet.com/forums/messageview.php?start=920&catid=24&threadid= 264777 I found out by another 400SC user and you can not control assign interrupts on the PCI slots on this machine. Does that point bother you if you are going to run this unit with *? I want to put 3 X100P cards and 1 TDM400P in my up coming 400SC, but not sure if I will have conflict if I use up all the PCI slots in the machine. Charles Alvis Internet Technology Group Redmond, WA -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler Sent: Thursday, January 15, 2004 4:54 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] ultra-cheap asterisk box I have a Dell 400sc sever on order. It will be shipped on the 27th. It is a 2.4GHz P4 with a 533 MHz front side bus, a 40GB disk, 128MB of memory, sound card, ethernet, and year of on-site next day maintenance. It is $318 delivered after rebates. Yes, $318. This is a real server, by the way, not a desktop machine. It also makes NO noise. I can't hear a thing with my ear right next to it. Why would you even THINK about getting anything else? Paul Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Albertson Sent: Thursday, January 15, 2004 9:32 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ultra-cheap asterisk box I'm looking to do about the same thing, build very low cost systems. (I'm looking at putting Asterisk at some non-profit organizations.) but one thing you can't make a compromise on is reliabilty. It has to work and keep working for years to come. I was able to keep the price of a new PC to about $300 ad still use an ASUS mainboard and an AMD XP2600+ The trick is to add absolutly nothing not needed. No floppy, no CDROM so you can run off a 200W P/S. Next I'll experiment with a notebook sized IDE disk drives and to see if _underclocking_ the CPU reduces it's power comsumption enough that we can save one fan. Ideally Asterisk will be ported one day to Linux/ARM or some other very low cost platform. for VOIP you do not need the PCI slots. In theory Asterisk could run on a Lynksys router box with re-flashed EEPROM. After all Lynksys' latest wireless router runs Linux inside Low cost to me means "low total cost of ownership" To get this I don't think buying the lowest priced parts is the way to go. I want quality mainboard, and a quality power supply and, this is importernt: A low internal case temperature. for this reason I'll spend the extra $50 to go with Antec cases and ASUS mainboards over the generic ones. What I'm finding is that the PCs are so cheap that the cost of electric power to run them is now a large part of the cost. (assume 0.20/kwh times 200W times 365 days = $350. So you pay for the PC again every year in electric power to run it. Worse. In an office with airconditioning _all_ of that PC's 200W goes to heat and your A/C unit will use about 220W of power to remove that 200W of heat.) and at a small office they will not have a server room so noise from the fan is an issue. --- Roy Sigurd Karlsbakk <[EMAIL PROTECTED]> wrote: > hi all > > what about this... > I just put together a box on a web shop (komplett.no) that will cost > me > NO
Re: [Asterisk-Users] Playing background message
Rich Adamson wrote: > > Unless I'm missing something here, from the CLI do a 'show application dial' > and checkout the "r" option, as in: > exten => 3015,1,Dial(SIP/3015,15,tr) I agree, I'd rather have the caller hear ringing instead of MOH as ringing gives the caller some feedback as to what is happening. I'd save the music until they've talked to someone or heard a message and are put on hold or get dumped into a queue. -Lance ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER & Asterisk
On Sat, 2004-01-17 at 01:33, [EMAIL PROTECTED] wrote: > Thanks guys, thought SER had to 'register' to be able to use > any Asterisk contexts. > But just defining a new entry in the sip.conf with just context & ip worked! > > But now i'm stumbling on another problem.. Asterisk seems to want > to send the SIP udp packets directly to the SIP clients. > In the case of a SIP user/client behind a NAT, this obviously doesn't > work. My guess'd be that this is a problem of your ser configuration (such as a missing record_route()) rather than an issue with *. One thing I would take a look at, would be the incoming INVITE request using "sip debug" and check whether or not you you find a header field Record-Route: pointing to you SER proxy. Thilo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: PID
> "T" == T Chan <[EMAIL PROTECTED]> writes: T> Thanks alot for your explanation. Can you tell me if there is a way T> to confirm if I have the nptl in the boxes ? grep for nptl in the installed pthread libs: grep -i nptl /lib/libpthread.so.0 /usr/lib/libpthread.a does it on my box. -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec problems (SIP)
Terence Parker wrote: Hi again, I've finally got Voicetronix OpenLine4 working so am a happy man ... thanks again to all those who helped! Just a few outstanding questions of curiosity : 1. I have finally got my setup to work by allowing ONLY g711alaw and nothing else. Why should enabling a few extra codecs cause problems? Surely if two phones are able to work at g711alaw, and either side had a compatibility problem with anything else (i.e. g729a at one end but not at the other) - they would automatically negotiate to use g711alaw anyway? Is the system/phones not smart enough to do this and I have to explicitly specify what everything should use? My little experience on * tell me that some phones and/or channels do not negotiate very well the codec selection. Secondly, also regarding codecs - I don't understand this as, surely, I have already enabled g729a and ulaw ... how can it complain that it can't transmit in that format, or that it can't find a path? How do you got the g729 codec? * does not include it. You must to pay for that. ... okay, fine. But where can I buy it? And is there something specific I have to buy, or does any old thing work with asterisk? Or...? * is OS, thus I can not include any think that needs license. See Digium if you want to buy g729 codecs. Jorge Thanks again! Terence ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Notice Messages??? What does it mean
Is there a way to tell which device it is coming from? We have Grandstream phones and it is intereconnected with a Nextone MSW using SIP. Todd -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brancaleoni Matteo Sent: Saturday, January 17, 2004 8:41 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Notice Messages??? What does it mean there isn't already in the wiki... this is really a FAQ! btw, that means that your device is using silence suppression. since * doesn't support that, it issue the NOTICE below. that's not harmful, but if you're annoyed by those msgs, just turn off silence suppression in your device. > Jan 16 15:56:11 NOTICE[240654]: File rtp.c, Line 263 > (process_rfc3389): RFC3389 matteo. -- Brancaleoni Matteo <[EMAIL PROTECTED]> Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ultra-cheap asterisk box
That's about the scale we're planning per server. We just purchased a Dell PowerEdge 1600SC for the job (able to go dual Xeon, but we're starting with one 2.8GHz, will add a second and recompile if we need 2 procs). The nice thing about this box is that it seems to have 2 of each kind of PCI slot: 2 x 64-bit/100MHz PCI-X (Supports 3.3v or Universal cards), 2 x 64-bit/66MHz PCI (Supports 3.3v or Universal cards), and 2 x 32-bit/33MHz PCI (Supports legacy 5v or Universal cards) Price after rebate: about $800 US. I'll let you know how it goes when it comes in. It should be early next week. - Original Message - From: "Aram Ter-Martirosyan" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, January 16, 2004 8:21 PM Subject: RE: [Asterisk-Users] ultra-cheap asterisk box > I don't need to use the lowest end server for asterisk, but something > reasonable. I need to put one TE410P and 2 TDM400P boards in it. We have > about 20 heavy telephone users in the company. Can someone suggest > reasonable priced and reliable box? > > Thanks, > > Aram Ter-Martirosyan > Senior Account Manager > Hi-Tech Gateway, Inc. > http://www.hi-teck.com > 1225 Grand Central Ave. > Glendale, CA 91201 > [EMAIL PROTECTED] > tel 818.546.4601 > fax 818.546.4617 > Turning Technology Into Business Solutions > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of James H. > Thompson > Sent: Friday, January 16, 2004 5:44 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] ultra-cheap asterisk box > > > FAQ for Dell 400SC: > > http://www.aaltonen.us/forums/viewtopic.php?t=8 > > > Jim > > James H. Thompson > [EMAIL PROTECTED] > > - Original Message - > From: "calvis" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Friday, January 16, 2004 10:33 AM > Subject: RE: [Asterisk-Users] ultra-cheap asterisk box > > > I got in on the same Dell deal I think. > > You must hang out on the bargain boards just like I do? I hang out mainly > at fatwallet.com. This is the thread that I got in on the Dell machines > that I just recently purchased. > > http://www.fatwallet.com/forums/messageview.php?start=920&catid=24&threadid= > 264777 > > I found out by another 400SC user and you can not control assign interrupts > on the PCI slots on this machine. Does that point bother you if you are > going to run this unit with *? I want to put 3 X100P cards and 1 TDM400P > in my up coming 400SC, but not sure if I will have conflict if I use up all > the PCI slots in the machine. > > > Charles Alvis > Internet Technology Group > Redmond, WA > > > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler > Sent: Thursday, January 15, 2004 4:54 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] ultra-cheap asterisk box > > I have a Dell 400sc sever on order. It will be shipped on the 27th. It is a > 2.4GHz P4 with a 533 MHz front side bus, a 40GB disk, 128MB of memory, sound > card, ethernet, and year of on-site next day maintenance. > > It is $318 delivered after rebates. Yes, $318. > > This is a real server, by the way, not a desktop machine. It also makes NO > noise. I can't hear a thing with my ear right next to it. > > Why would you even THINK about getting anything else? > > Paul > > Paul Mahler > mail:[EMAIL PROTECTED] > phone: 650.207.9855 > fax: 877.408.0105 > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Chris Albertson > Sent: Thursday, January 15, 2004 9:32 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] ultra-cheap asterisk box > > > > I'm looking to do about the same thing, build very low cost > systems. (I'm looking at putting Asterisk at some > non-profit organizations.) but one thing you can't make > a compromise on is reliabilty. It has to work and keep working > for years to come. I was able to keep the price of a new PC > to about $300 ad still use an ASUS mainboard and an AMD XP2600+ > The trick is to add absolutly nothing not needed. No floppy, > no CDROM so you can run off a 200W P/S. Next I'll experiment > with a notebook sized IDE disk drives and to see if _underclocking_ > the CPU reduces it's power comsumption enough that we can save > one fan. > > Ideally Asterisk will be ported one day to Linux/ARM or some > other very low cost platform. for VOIP you do not need the > PCI slots. In theory Asterisk could run on a Lynksys router > box with re-flashed EEPROM. After all Lynksys' latest wireless > router runs Linux inside > > Low cost to me means "low total cost of ownership" To get this > I don't think buying the lowest priced parts is the way to go. > I want quality mainboard, and a quality power supply and, this > is importernt: A low internal case temperature. for this reason > I'll spend the extra $50 to go with Antec cases and ASUS mainboards > over the generic ones. > > What I'm finding is that the PCs are so cheap tha
Re: [Asterisk-Users] Class features in dialplan ?
> Here's how Mark wants it done:
>
> In the channel_pvt structure, we have a pointer to a new structure:
>
> struct ast_common_features {
> char fwd[AST_MAX_EXTENSION];
> char fwd_off[AST_MAX_EXTENSION];
> .
> .
> .
> };
>
> This allows the codes to be redefined per installation (for those who
> don't like the NANPA assignments) or even per channel. In addition,
> those who want to handle the codes in the dialplan can turn off various
> featuresets by setting their codes to "".
>
> Then, when we're processing extensions in the channel driver, we do
> the following:
>
> if (ast_feature_match(&p->common_features, exten, ...)) {
> /* Make channel if it doesn't already exist */
> ast_feature_handle(chan, &p->common_features, exten, ...);
> }
>
> and in the config, the features would be configured with another common
> routine:
>
> else if (!ast_feature_config(&p->common_features, v->var, v->value)) {
> ast_log(LOG_WARNING, "Unknown keyword '%s'\n", v->var);
> }
>
> All these common features could then be handled either in a device
> independent way (such as forward, where the end number is entered into
> a database for later lookup) or in a device-specific way, as
> appropriate.
>
> I've looked at implementing this, but it is quite complex and extensive,
> and I haven't had the time to complete it. Add to that all the
> negativity of people who don't like this plan and have vociferously
> expressed their distaste (intending to implement these in the dialplan
> instead), and you see why this code is even more difficult to write.
>
> -Tilghman
So, it sounds like everyone is stuck reinventing the wheel for a while
unless these featurs are going to be available "real soon now" :) I
guess I'll go throught the nanpa list and pick out the ones that are
must haves for now.
Thanks
-Lance
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Re: [Asterisk-Users] Playing background message
> When incoming calls come to us, our PSTN line is picked up almost > immediately - and then asterisk will proceed to dial the SIP extensions. > During this time the caller hears dead slience - obviously not very good as > some would think the line just went dead and hang up. I have toyed with the > idea of playing a 'welcome... your call will be answered shortly' etc... > message, but can't get it to work how want it. > > The caller will hear a recorded message, followed by music. What I want is > the caller to hear this WHILE the SIP phones are ringing - but using the > 'Background' option in extensions.conf seems to make it so that my SIP > phones won't be dialled until AFTER the music clip is finished - i.e. > pointless. > > How do I truly set a background audio to play while the internal phones are > ringing? Is this possible? Music on hold perhaps? Unless I'm missing something here, from the CLI do a 'show application dial' and checkout the "r" option, as in: exten => 3015,1,Dial(SIP/3015,15,tr) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playing background message
hi > > The caller will hear a recorded message, followed by music. What I want is > the caller to hear this WHILE the SIP phones are ringing - but using the > 'Background' option in extensions.conf seems to make it so that my SIP > phones won't be dialled until AFTER the music clip is finished - i.e. > pointless. of course... background is an action in the dialplan, so during its execution, you can't do anything more... > > How do I truly set a background audio to play while the internal phones are > ringing? Is this possible? Music on hold perhaps? yes, moh is the way to go see musiconhold.conf -- Brancaleoni Matteo <[EMAIL PROTECTED]> Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Playing background message
Use music on hold, and tell the dial app to use it.. Luciano -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Terence Parker Enviado el: Sábado 17 de Enero del 2004 11:38 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] Playing background message Sorry for the fragmented messages from me - one last thing I forgot to ask in my last post. When incoming calls come to us, our PSTN line is picked up almost immediately - and then asterisk will proceed to dial the SIP extensions. During this time the caller hears dead slience - obviously not very good as some would think the line just went dead and hang up. I have toyed with the idea of playing a 'welcome... your call will be answered shortly' etc... message, but can't get it to work how want it. The caller will hear a recorded message, followed by music. What I want is the caller to hear this WHILE the SIP phones are ringing - but using the 'Background' option in extensions.conf seems to make it so that my SIP phones won't be dialled until AFTER the music clip is finished - i.e. pointless. How do I truly set a background audio to play while the internal phones are ringing? Is this possible? Music on hold perhaps? Thanks, Terence ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zone Paging
> I see a lot of chatter in the archives about intercom and paging, but > has anyone addressed zone paging? Each of the 50 telephones in a large > clinic would be members of one or more paging zones. Someone could then > page Dr. X in zone #1. Would this be possible with analog phones? SIP? Summarizing from memory only, I believe this has been discussed more then once on the list and usually comes down to... what device(s) provide an auto-answer extension and includes an audio output jack that can be plugged into a paging amplifier? The two most common suggestions (from memory again) has been: 1. Use of the sound card on the asterisk machine (which sort of implies a limit of one paging zone), or, 2. Use a sip phone (Cisco 7960 with v6 as one example), configure the phone to support auto-answer, and connect the external headset to the paging amp. (Implies one sip phone per paging zone.) I've not tried either, so not sure of success/failure rates or problems. Seems like a fair number of people have problems getting the sound card to play nicely with asterisk, and most of the chatter seems to be oriented around sound card driver issues, etc. Don't know if the ata-186 supports auto-answer in current software, but if it did, jury-rigging a matching transformer as a source of zone audio would not seem like it would be very difficult. (Anyone know whether the 186 can be configured for auto answer?) Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Notice Messages??? What does it mean
there isn't already in the wiki... this is really a FAQ! btw, that means that your device is using silence suppression. since * doesn't support that, it issue the NOTICE below. that's not harmful, but if you're annoyed by those msgs, just turn off silence suppression in your device. > Jan 16 15:56:11 NOTICE[240654]: File rtp.c, Line 263 > (process_rfc3389): RFC3389 matteo. -- Brancaleoni Matteo <[EMAIL PROTECTED]> Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Playing background message
Sorry for the fragmented messages from me - one last thing I forgot to ask in my last post. When incoming calls come to us, our PSTN line is picked up almost immediately - and then asterisk will proceed to dial the SIP extensions. During this time the caller hears dead slience - obviously not very good as some would think the line just went dead and hang up. I have toyed with the idea of playing a 'welcome... your call will be answered shortly' etc... message, but can't get it to work how want it. The caller will hear a recorded message, followed by music. What I want is the caller to hear this WHILE the SIP phones are ringing - but using the 'Background' option in extensions.conf seems to make it so that my SIP phones won't be dialled until AFTER the music clip is finished - i.e. pointless. How do I truly set a background audio to play while the internal phones are ringing? Is this possible? Music on hold perhaps? Thanks, Terence ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re:[Asterisk-Users] SS7 over Asterisk ?
Hmm.. I've searched OpenSS7 , but the asterisk projects it's stalled.(http://www.openss7.org/asterix.html) All I'm trying right now is to get raw data from the E1 (from each timeslot) , transmit it to another asterisk server and push it to the other E1.. I don't care what's on that E1.. It's SS7 , it's PPP data , whatever... All I want is to "bridge" the timeslots.. Thanks Alex - Original Message - > From: "Tom Scott" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Saturday, January 17, 2004 3:57 PM > Subject: Re: [Asterisk-Users] SS7 over Asterisk ? > > > > Alexandru Coseru wrote: > > > I have a customer who wants to connect 2 PBX's over IP.. > > > > > > The setup should look like this: > > > > > > > > > [PBX] <-- SS7 --> [Asterisk] <-- IAX--> [Asterisk] <-- SS7 > > > --> [PBX] > > > > If you succeed in doing this, i.e., turning asterisk into a softswitch, > you'll > > be quite famous, at least to me and a few others. For background on the > IPCC > > softswitch and related technologies, you can browse this url: > > > > http://vedatel.com/Isdn/eet290_spring_2004.html > > > > Connecting a VoP (VoIP, VoATM, VoFR, VoMPLS) cloud to the PSTN via SS7 is > one > > of the great challenges, especially if you want to use open-source > software from > > asterisk and other projects. There are proprietary solutions, of course, > but > > I assume you're looking for something more cost effective since you're > using > > asterisk. > > > > Good luck. and plz post your results if you succeed. > > > > -- TT > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Kernel for 1586 vs i686 for Asterisk?
Hi, As I have said before I am not that hot on building kernels and what effects of various options would have on Asterisk.. I am thinking of using a distro called Trustix to build my new Asterisk server.. Its super small and is built for security so everything is disabled by default which is always a good thing since I would probably miss some when I was locking down any other distro.. The thing that is playing on my mind is that all packages are built for i586 not i686.. So the question, for the kernal guru's, is what effect will this have on Asterisk performance if any?? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Notice Messages??? What does it mean
I just started getting the following notice message and was wondering what it meant…. Jan 16 15:56:11 NOTICE[240654]: File rtp.c, Line 263 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible Todd Wallace
Re: [Asterisk-Users] Codec problems (SIP)
Hi again, I've finally got Voicetronix OpenLine4 working so am a happy man ... thanks again to all those who helped! Just a few outstanding questions of curiosity : 1. I have finally got my setup to work by allowing ONLY g711alaw and nothing else. Why should enabling a few extra codecs cause problems? Surely if two phones are able to work at g711alaw, and either side had a compatibility problem with anything else (i.e. g729a at one end but not at the other) - they would automatically negotiate to use g711alaw anyway? Is the system/phones not smart enough to do this and I have to explicitly specify what everything should use? Secondly, also regarding codecs > > - I don't understand this as, surely, I have already enabled g729a and > > ulaw ... how can it complain that it can't transmit in that format, or > > that it can't find a path? > > > How do you got the g729 codec? * does not include it. You must to pay > for that. ... okay, fine. But where can I buy it? And is there something specific I have to buy, or does any old thing work with asterisk? Or...? Thanks again! Terence ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zone Paging
I see a lot of chatter in the archives about intercom and paging, but has anyone addressed zone paging? Each of the 50 telephones in a large clinic would be members of one or more paging zones. Someone could then page Dr. X in zone #1. Would this be possible with analog phones? SIP? Thanks, Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 over Asterisk ?
Alexandru Coseru wrote: I have a customer who wants to connect 2 PBX's over IP.. The setup should look like this: [PBX] <-- SS7 --> [Asterisk] <-- IAX--> [Asterisk] <-- SS7 --> [PBX] If you succeed in doing this, i.e., turning asterisk into a softswitch, you'll be quite famous, at least to me and a few others. For background on the IPCC softswitch and related technologies, you can browse this url: http://vedatel.com/Isdn/eet290_spring_2004.html Connecting a VoP (VoIP, VoATM, VoFR, VoMPLS) cloud to the PSTN via SS7 is one of the great challenges, especially if you want to use open-source software from asterisk and other projects. There are proprietary solutions, of course, but I assume you're looking for something more cost effective since you're using asterisk. Good luck. and plz post your results if you succeed. -- TT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Early B3 on PRI channel (Listen free mex without charge)
Dear sorry for my next post on the same object but i need to know if is possible to hack libPRI to have the function that KAPEJOD made in CAPI for the "early B3" function. I'm the only in the list have this type of problem with Free Message? :-) Thanks in advance Dimitri ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Class features in dialplan ?
Hi! > I thought I was making progress on my dialplan when I realized that the > class features that are available for zap channels aren't available for > SIP channels. http://www.voip-info.org/wiki-CLASS > 3. Anyone willing to share some of their cool features that they've > come up with ??? I'd be most appreciative :) http://www.voip-info.org/wiki-Asterisk+PBX+functions Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER & Asterisk
> But now i'm stumbling on another problem.. Asterisk seems to want > to send the SIP udp packets directly to the SIP clients. > In the case of a SIP user/client behind a NAT, this obviously doesn't > work. Have you tried reinvite=no in your [ser] section of sip.conf? P ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SS7 over Asterisk ?
Hello.. I have a customer who wants to connect 2 PBX's over IP.. The setup should look like this: [PBX] <-- SS7 --> [Asterisk] <-- IAX --> [Asterisk] <-- SS7 --> [PBX] Since there are no SS7 cards , I was thinking at a way of carrying the E1 data as bulk...Can I do that ? How ? Is possible a scenario like this ? I'm thinking of IAX because I don't want to use H323 or SIP... Thanks a lot Alex
Re: [Asterisk-Users] VoiceMail - no user pre-registration
Hi! Thanks for your suggestions. How easy is it to use the A and B number of the incoming calls in certain scripts (e.g. are functions present to read them and use them for further operation) I have not worked with AGI scripts before and probably need to have a look at this. Is the AGI capable of doing the issues as you described them below? Cheers, Jeroen PS: Andrew, handling password is not required, an A number analysis is sufficient (environment is 'secured') Scenario: All incoming calls are voicemail calls however the dialled number (called party) does not necessarily have a voicemailbox configured in the Asterisk system. You could: - configure a set of temporary mailboxes in voicemail.conf - direct a caller without mailbox to such a temporary box and record his name somehow (either in the Asterisk Db, or using an AGI script) - after (or at the beginning of the next) call run a clean-up script that 1. renames the voicemail directory and 2. re-creates the temporary mailbox using plain simple shell commands through System(), and 3. modifies voicemail.conf and 4. reloads asterisk (for voicemail.conf I *think* you don't need a restart). I am not sure why you wouldn't want to use a database for voicemail.conf, but I think it can be done rather easily without. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] G.723.1 codec
Saturday, January 17, 2004, 12:49:26 AM, Eric Wieling wrote: EW> You can purchase the G.723.1 reference code from the ITU, then you'll EW> need to make it work with Asterisk I made codec_g723 with this code, but for compression of PCM file 12 sec long requires 37 sec :) (2x600MHz server) So my opinion is: if server has't any hardware DSP you should not do any codec conversation more complex then(gsm -> pcm) EW> On Fri, 2004-01-16 at 13:30, Dan Tusa wrote: >> Hi, >> >> Want to do some experiments with the G.723 codecs - where can I download the >> 723 source code for Asterisk? >> >> I know there are some ongoing discussion regarding patents and license fees >> for the g.723 but I have some hardware on which I only have the 723 and need >> to test it privately. >> >> Thanks! >> Dan >> >> _ >> Use MSN Messenger to send music and pics to your friends >> http://www.msn.co.uk/messenger >> >> ___ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users -- Andrei Koulik. System administrator, Sandy Info Ltd. (ISP), Nizhny Novgorod, Russia ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.723.1 codec
I solve it for h323 in follow way: 1. Exclude all codecs except g723.1 from h323.conf: disallow=ULAW allow=g723.1 2. Add format_g723 module (http://www.agk.nnov.ru/format_g723.c.gz) into project 3. convert all wav and gsm sound into g723 format (use lbccodec from g723_1 demo package, don't ask me where you can download it) 4. set maxsilence=0 in voicemail.conf to suppress conversion into pcm format for silence detection. And it works fine for me. But where are some bugs in h323 module: * not supported g7231 without sound detection (simple to fix). * sometime data transfer (rtp traffic) begins before negotiation complete and first packet is going in g711 codec and channel going down (not yet reviewed). if will any question regards format_g723 module send mail to: f723 >< agk.nnov.ru Friday, January 16, 2004, 10:30:41 PM, Dan Tusa wrote: DT> Hi, DT> Want to do some experiments with the G.723 codecs - where can I download the DT> 723 source code for Asterisk? DT> I know there are some ongoing discussion regarding patents and license fees DT> for the g.723 but I have some hardware on which I only have the 723 and need DT> to test it privately. DT> Thanks! DT> Dan DT> _ DT> Use MSN Messenger to send music and pics to your friends DT> http://www.msn.co.uk/messenger DT> ___ DT> Asterisk-Users mailing list DT> [EMAIL PROTECTED] DT> http://lists.digium.com/mailman/listinfo/asterisk-users DT> To UNSUBSCRIBE or update options visit: DT>http://lists.digium.com/mailman/listinfo/asterisk-users -- Andrei Koulik. System administrator, Sandy Info Ltd. (ISP), Nizhny Novgorod, Russia ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicetronix OpenLine4: disable answering on a particular channel & delay before dial
Hi there, After a lot of valuable insights from the list, incoming and outgoing calls finally work through OpenLine4! Thanks for all the input! Now I have 2 minor issues: Sometimes Voicetronix dials too quickly before an actual dial tone is obtained from the phone company. E.g. Voicetronix picks up a line and then dials immediately, whereas actually it took the phone company may be half a second to actually make the line available to gave a dialtone. As a result? 90% of the time, the first digit dialed was not received by the phone company. Is it possible to tell voicetronix to wait a second or two before dialing? Secondly, I have a phone line plugged into channel 2 that I don't want Asterisk to answer. I only want ASterisk to use it to dialout. So I need to configure Asterisk somehow to ignore incoming calls on channel 2. Is this possible? Thanks! Terence ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR problem with macros
Philipp von Klitzing wrote:
Hi there,
whenever I use a macro to dial out I see only "s" recorded in the dst
field of the CDR. Is there anyway to get around that problem except for
not using a macro?
Example:
)
Try to match every extension before dialing out instead, using "s" is a bad thing for CDRs.
> [default]
> exten => 1234,1,macro(dial-out)
> [macro-dial-out]
> exten => s,1,Dial(SIP/test,30,r)
[default]
exten => 1234,1,macro(dial-out,${EXTEN})
[macro-dial-out]
exten => s,1,goto(dial-out2,${ARGV1})
[dial-out2]
exten => _X.,1,Dial(SIP/test,30,r)
Of course, you could to a goto instead of macro in the first place, but there
might be
another reason that you want to use a macro...
Now, the last extension used is "1234" instead of "s".
/O
___
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Re: [Asterisk-Users] Cisco 79xx Ringtones
Paul Crick wrote: If you want some classic office phone ringers: http://www.leonine.com/~lion/phones.php These are Merlin rings. For some, myself included, they're a bit nostalgic. But cool with it! Now I can make my phone sound like those ones on 24 ;-) Not being too familiar with the Merlin system (hailing from the UK not North America), what's the deal with the different ring tones? Assignable to different phones only, or different line appearances (ie one phone, different lines ring different ways?). You could select one of several available hard-coded ringtones for your Merlin desk phone, primarily to differentiate your phone from your office co-workers. The only thing I forgot to sample was the "poik" sound of the option key presses, but that's not really useful as a ringtone. Lion Templin -- = lion is Lion J Templin [EMAIL PROTECTED] = = 612-605-3613 x3001 FWD 94117 = ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware for Asterisk
On Friday, 16 January, 2004 12:27, Steven Critchfield wrote: > On Fri, 2004-01-16 at 06:47, Andrew Kohlsmith wrote: > > > If you value your data, don't use software raid. If you value > > > performance don't use software raid. If you value uptime/stability > > > don't use any raid on IDE. > > > > That's pure bullshit -- I use software RAID *specifically* because I > > value my data. I don't want to buy two hardaware RAID controllers to > > have one sit on the shelf just in case the first dies... and if the > > second dies you're SOL because they've lasted long enough that they're no > > longer available. Linux software RAID is available on any Linux system > > and if the system blows up I can put the drives in another system and > > *not* worry about it not being detected. > > > > As far as performance goes, I have some bonnie++ tests that I've run that > > show that at least on the few systems I've tested, software RAID 1 beat > > out hardware RAID 1 (these systems were IDE, SCSI-2 and Ultra320, with > > DPT RAID controllers for SCSI on P4 and I think regular Promise IDE RAID > > controllers on P3) -- not a huge difference in speed but one that at > > least tosses your "if you value performance don't use software raid" > > argument. > > > > Perhaps on a _heavily_ loaded server you might be right, but then again I > > feel that you're stupid for letting a server get so loaded up that it > > can't handle the simple mirroring algorithms in addition to normal file > > servering functions without degrading performance to a noticable degree. > > > > I used to believe that HW RAID was the only way to go. With RAID5 I > > still feel that is true to an extent. However if you're just mirroring > > there is _no_ significant advantage to choosing hardware RAID over > > software RAID. Not on IDE, and not on SCSI. In fact, there are > > advantages to choosing software RAID over hardware RAID, as I've > > mentioned above. > > Have you experienced a hardware failure yet that you had to come back > from? If you loose a drive, it is a high probability that you will loose > the controller. So unless you have a add on card, or some motherboard > with 4 IDE ports, you will corrupt the second drive of a mirror. If the > second drive is corrupted, then you are only a hair above not having > anything. If you don't trust that, check out the GOOD IDE raid > controllers. You are only allowed to place 1 drive per port, and they > only use 1 port on a IDE controller. Now here we are seeing that you must have had a really abnormal, bad experience, or you are not talking from experience at all. I have, in fact, used many software and hardware RAID configurations, and I have had a great many drive failures. For mirroring, I use software RAID because is greatly superior due precisely for the reliance on the controller of any given hardware RAID array. Although I think it is very far-fetched to set such a high relational coefficient of drive failure to controller failure, (since I have had _far_ more drives fail than controllers) the facts that hardware controllers are both expensive (compared to free software) and rare (compared to any machine's normal IDE ports) culminates in my use of software RAID. I can stick the good drive of any software-mirrored RAID array into _any_ other system (Linux OR Windows), boot up off my trusty rescue CD with software RAID and networking, and immediately recover data or functionality. Further, this presumes that the machine which housed the failed drive is otherwise in a non-functional state. If this is a false presumption, because I have RAIDed my boot partition the system boots just fine with only one working drive. Even better, when I get the new drive, I can simply install and rebuild the array while I am on-line... a feature not all hardware RAID controllers have. _My_ horror stories are those of single "brick outhouse" servers which all sorts of special hardware failing out in the field with an SCA drive and no SCA backplane/controller within 100 miles. > > Even the large NAS devices that use IDE have the IDE controller built > into the sled that holds the drive and use PCI hotswap technology. > > I don't buy it that any truly redundant raid system is as fast in > software as in hardware on a machine doing anything significant. In raid > 1, you are double or more writing all data to the drives. in a read > environment, it might be able to share the load out to more than 1 drive > and help, but I don't expect it would be much better than a dedicated > controller handling the load. Any load of a software raid solution takes > processor time away from the processes it is trying to complete. So take > our VoIP application, if I am spending time getting the voice recording > to 2 or more drives and the software to get it there, you have > significantly reduced the amount of time available to the CPU to handle > the VoIP packets in a timely manner. This only gets worse
Re: [Asterisk-Users] Hardware for Asterisk
On Saturday 17 January 2004 00:31, Chris Albertson wrote: > Software RAID vs. Hardward RAID??? Welcome to the 80s. > There IS no "Hardward RAID" it's all "software" the difference is > only where the software lives, in ROM on the controler card in > the RAID box or in a Linux driver. Actually, hardware RAID typically runs something called firmware (which is technically software, though it tends to be a little more difficult to alter) and offloads the task of balancing the data across multiple disks off the CPU. This is the primary difference between software RAID (which, since it uses the CPU, reduces the available CPU for other tasks) and hardware RAID. > For Asterisk all you would need is a simple disk mirror at most. That's a gross oversimplification without consideration of a particular setup. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
