Re: [Asterisk-Users] How to query a table from the keypad?
Or you could skip the overhead associated with an AGI and use the
dialplan command availabe after installing asterisk-addons MYSQL.
exten => _X.,1,Read(PO-NUMBER,enter-yr-po-num)
exten => _X.,2,MYSQL(Connect connid)
exten => _X.,3,MYSQL(Query resultid ${connid} SELECT balance FROM
account-payables WHERE po_num=${PO-NUMBER})
exten => _X.,4,MYSQL(Fetch fetch ${resultid} AMOUNT-DUE)
exten => _X.,5,MYSQL(Clear ${resultid})
Of course you will want to put in place all the error traps and when
using this function I always have a check in my hangup routine to make
sure I close the open mysql connection. So at the end of the above
dialplan, you should have the value you want in the AMOUNT-DUE variable.
-Chris
Mike Pollitt wrote:
>
> Hi Richard –
>
> What you want is AGI:
> http://www.voip-info.org/tiki-index.php?page=Asterisk+AGI
>
> You could write a perl script to read the PO number from stdin and
> spit back the balance (or whatever). To read the PO number from the
> user, use the Read() dialplan application. To play back the balance,
> you could use SayDigits() (but there’s probably a more elegant
> solution specifically for speaking amounts of money).
>
>
>
> *From:* [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] *On Behalf Of
> *Richard Reina
> *Sent:* Friday, 24 February 2006 9:34 AM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [Asterisk-Users] How to query a table from the keypad?
>
> I am trying to give users the option to query our accts. payable
> database by supplying their PO number. I able to write queries via
> perl->DBI->mysql but have no idea how to get * to do it from the IVR.
> Is this possible? Can anyone point me in the right direction for help
> or examples?
>
> Thanks,
>
> Richard
>
>
>
>
> What are the most popular cars? Find out at Yahoo! Autos
>
>
>
>
>
> ___
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>
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Re: [Asterisk-Users] Recommended rack-mountable server anyone?
I've used quite a few of the rack-mount servers from http://www.siliconmechanics.com/. I've had both digium and sangoma cards in them with zero issues. -Chris ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 5,000 concurrent calls system rollout question
Even if you could, you wouldn't want to use just one system to handle this call load. What happens when you lose a power supply or a hard drive, or any other random failure? I would think you would want a more robust design. While you can go the signate way and use SGI hardware to increase your load per footprint, you can alos go the way of a large cluster of low priced systems as well. I would do something like this: Two SIP router systems (all signalling, no media) that all SIP devices (end user UA's provider trunks etc.) communicate with in a load balanced fashion. These two routers recieve registrations all SIP signalling. They keep track of dynamic UA locations (SER or Asterisk could be used here). They use a SIP 302 redirect where possible and re-invite where redirect isn't supported to route call requests to a cluster of asterisk systems. For 5000 calls with no media, two systems should be good enough for N+1 redundancy (in other words one server is enough, but you have two so you can fail one at any time). Behind this you stick as many asterisk servers as is needed based upon the hardware and it's load ability. Again, N+1 should be your minimum design basis for the number of systems. The two routing systems should have a method of knowing the load on each node so that when redirecting a call, they can do so intelligently. This would also allow you to build in the ability to take nodes offline for maintenance or other requirements. Just throwing together a bunch of asterisk systems and using 'round-robin' routing will quickly become a management nightmare. While this can definately be done using asterisk, like someone else said, if you want to do it right, you are going to be looking at the need for a strong implementation team. -Chris ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] username not stabled?
Ronald Wiplinger wrote: > > > 601, 602, 605, 606, 608, 609, 610, 615 and 616 are in sip.conf > 621 and 626 are in Real-time sip_buddies > > 621 and 626 changes username back from name to number (name) in the > database, and never shows it in "sip show peer" > > 615 changed username "Ronald office" to 615, although no change in > sip.conf > > Did anybody else experienced that? > > *CLI> show version > Asterisk SVN-trunk-r8447M built by root @ vpbx on a x86_64 running > Linux on 2006-01-25 15:33:01 UTC > There is some code in asterisk which I'm not sure why it exists, that will set the username in memory to the user value in the SIP Contact header upon registration. While this isn't normally a big deal, if you are using realtime, when a SIP UA registers, some things are written back to the realtime database, username being one of them. I am not sure if this is a bug or not, as I don't understand the thought process behind allowing a sip ua to modify the username asterisk uses based on a sip header when it registers. I went into the code and removed the username as a field that got written back to the realtime db upon registration and it fixed my problem. -Chris ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Realtime IP peer with static IP won't load
In CVS Head from both 2005-08-28 and 2005-11-03, I've created a SIP peer and user with the same settings. When this peer/user sends a SIP invite to the asterisk system running realtime, it gets a 404 not found back, and the realtime system says it's unable to find a match for :5060. However if I forcefully load the peer by using sip show peer/user load. Then everything works just fine. Also if I set the peer/user as a dynamic IP address, when it registers, the realtime entry is loaded. This seems like bad behaviour for the SIP realtime peer/user with static IP address. Has anyone else experienced the same? Is there a special configuration of the realtime config settings in sip.conf that enables this? -Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Storing extension prefs. in MySQL
Andreas Sikkema wrote:
[EMAIL PROTECTED] wrote:
I would like to store these seetings in a mysql database, so
that they are more easily accessible from a user
configuration page on a webserver. Since these settings need
to be checked in the dialplan for each call to the extension,
it seems a bit to much to have to connect, query and
disconnect from mysql every time. Is there any way to keep a
persistent connection to mysql that can be queried from the
dialplan?
Well, if you do this before answering, nobody is going to
notice. Even querying during an answered call will have
hardly any outside consequences...
Using the MYSQL functions from asterisk-addons
extensions.conf
[macro-open-connection]
exten => s,1,MYSQL(Connect connid .
exten => s,2,SetVar(OPEN-CON=1)
[macro-close-connection]
exten => s,1,GotoIf(${OPEN-CON}?5)
exten => s,5,MYSQL(Close ${connid})
Then call the open macro when you first receive a call, and make sure
in your hangup (exten => h) function you call the close macro. If
you have an include => hangup in all your contexts, the close macro
will be called anytime a hangup is received and it will close any
connection you opened for that call.
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Re: [Asterisk-Users] Realtime IAX
Dana Olson wrote: Chris, Thanks for the reply. I checked those settings, and they were commented out, so I uncommented them. I assumed you meant rtnoupdate=yes, so that's what I put, but that didn't work. I tried rtnoupdate=no, and that didn't work either. I do have a register statement in my iax.conf, and that works - I can get my inbound calls no problem. Dana Actually, the current CVS Head usage is rtupdate=, it was changed from rtnoupdate= not too long ago. If you are using 1.2 I'm not sure which is correct. I went through this battle of getting this to work the beginning of this week, and the four settings I listed in my last post made all the difference. -Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime IAX
> I am having the exact same issues. I even tried to madk my IAX peer > account in both the database, and in the iax.conf file (with different > names, but same info) and the static one works, but not the database > one. I am using 1.2.0-beta1. > > If I specify the user:[EMAIL PROTECTED] on the dialplan, it works, but > this is bypassing the peer in the iaxpeers table in the database. > > I contacted my IAX provider, and he was not seeing the dial request > come across or anything, so where that circuit-busy is coming from, I > don't know... > > Did you ever get a resolution? Is this maybe a bug that should be > opened on the Digium tracker? > > -- > Dana Make sure you have the following setting in your iax.conf file. rtcachefriends=yes rtupdate=yes rtautoclear=no rtignoreexpire=yes Also, you will still need your register => statement if you needed it before you started using realtime -Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] canreinvite=no being ignored?
Am I reading the data below incorrectly, or does it appear that even
though I have the directive canreinvite=no set for the two asterisk
boxes, they are trying to do a reinvite (which fails) anyway?
Is this expected behaviour in this situation? If so, how can I prevent
this?
Lots of output
Using CVS Head from 2005-08-28, I have two asterisk boxen, one (box A)
has a sip ua (2608) attached which is generating a call, the other
machine (box B) has the final destination.
Sip config for the phone on box A (via Realtime):
pbx3*CLI> sip show peer 2608
* Name : 2608
Secret :
MD5Secret:
Context : assigned-device
Language :
AMA flags: Unknown
CallingPres : Presentation Allowed, Not Screened
Callgroup:
Pickupgroup :
Mailbox :
VM Extension : asterisk
LastMsgsSent : -1
Inc. limit : 0
Outg. limit : 0
Dynamic : Yes
Callerid : "" <>
Expire : 386
Expiry : 900
Insecure : no
Nat : RFC3581
ACL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr->IP : 192.168.10.32 Port 5060
Defaddr->IP : 0.0.0.0 Port 5060
Def. Username: 2608
SIP Options : (none)
Codecs : 0x4 (ulaw)
Codec Order : (ulaw)
Status : OK (16 ms)
Useragent: Sipura/SPA841-0.9.1
Reg. Contact : sip:[EMAIL PROTECTED]:5060
Sip config for Box B on box A:
pbx3*CLI> sip show peer pbx1
* Name : boxb
Secret :
MD5Secret:
Context : inter-system-inbound-main
Language :
AMA flags: Unknown
CallingPres : Presentation Allowed, Not Screened
Callgroup:
Pickupgroup :
Mailbox :
VM Extension : asterisk
LastMsgsSent : -1
Inc. limit : 0
Outg. limit : 0
Dynamic : No
Callerid : "" <>
Expire : -1
Expiry : 900
Insecure : no
Nat : RFC3581
ACL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr->IP : Port 5060
Defaddr->IP : 0.0.0.0 Port 0
Def. Username:
SIP Options : (none)
Codecs : 0x4 (ulaw)
Codec Order : (ulaw)
Status : Unmonitored
Useragent:
Reg. Contact :
Sip config for Box A on Box B
pbx1*CLI> sip show peer pbx3
pbx1*CLI>
* Name : boxa
Secret :
MD5Secret:
Context : inter-system-inbound-main
Language :
AMA flags: Unknown
CallingPres : Presentation Allowed, Not Screened
Callgroup:
Pickupgroup :
Mailbox :
VM Extension : asterisk
LastMsgsSent : -1
Inc. limit : 0
Outg. limit : 0
Dynamic : No
Callerid : "" <>
Expire : -1
Expiry : 900
Insecure : no
Nat : No
ACL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr->IP : Port 5060
Defaddr->IP : 0.0.0.0 Port 0
Def. Username:
SIP Options : (none)
Codecs : 0x4 (ulaw)
Codec Order : (ulaw)
Status : Unmonitored
Useragent:
Reg. Contact :
Dial command as appears on boxa
-- Executing Dial("SIP/2608-8049", "SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
Aug 31 02:01:29 NOTICE[10496]: chan_sip.c:9028 handle_response: Failed
to authenticate on INVITE to '"2608"
>;tag=as4124f74a'
-- SIP/boxb-ae96 is circuit-busy
SIP Debug as it appears on boxb from the call above
<-- SIP read from :5060:
INVITE sip:c1#1234@ SIP/2.0
Via: SIP/2.0/UDP :5060;branch=z9hG4bK1d216175;rport
From: "2608" ;tag=as4124f74a
To:
Contact:
Call-ID: 3f1250096c1a12b0259689006888f106@
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 31 Aug 2005 09:01:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 214
v=0
=root 10496 10496 IN IP4
s=session
c=IN IP4
t=0 0
m=audio 19014 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (12 headers 10 lines)---
Using INVITE request as basis request -
3f1250096c1a12b0259689006888f106@
Sending to : 5060 (non-NAT)
Reliably Transmitting (NAT) to :5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
:5060;branch=z9hG4bK1d216175;received=;rport=5060
From: "2608" >;tag=as4124f74a
To: >;tag=as2ac1a098
Call-ID: 3f1250096c1a12b0259689006888f106@
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK
Re: [Asterisk-Users] Graphical Management Interface - Comments requested
In the next week to two weeks I'll be posting some information concerning a system I've been designing. It currently does three layer hosted VoIP pbx services as well as hosted ITSP services (the model is System Owner - you, Affiliates - pbx owner/operators or ITSP operators, and end users). The GUI is Zope/python based. The system supports automated clustering, fail-over, and server specialization (you can have a centralized voicemail server if you like, you can centralize sip registrations if you like, etc.). The Hosted PBX model is in beta testing right now, and we are adding a full billing/provisioning system in September. The bad news, it's not going to be open source. I'll have a web url soon for screen shots and an on-line demo. -Chris Erick Perez wrote: >Hi, >I want to start managing my asterisk boxes with a centralized >graphical based interface so I can (due to customers request) give >control to customers to add/change extensions to their current PBX >intallations such as (not complete list) >Add/del/mod extensions >sound recordings (ivr or voice attendants) >email to fax/ fax to email >voicemail to email >SIP and ZAP, no IAX needed >configure calls routes (server in office A to server in office B, etc) > >What I want to do is lock them out of the command line (linux) and >provide them with some graphical tool (or some manageable mixture of) >that can also help me. > >This list contains gpl and non gpl providers. > >So far I have looked at. >ACTOS >PBX MANAGER from third lane technologies >AMP >PBXWARE >switchvox > >So far I liked AMP (open source) but switchvox (paid) looks nice too. > >Comments on AMP integration? I do not want to start a >discussion/flame, I just want some links to AMP modules and see if i >can build from different sources a "graphical interface that does >end-user pbx functions" > >BTW call accounting and billing will be nice too. > >Thanks, > > > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fedora Core 4 x86_64
And make sure you diable the funky cpu scaling feature in the kernel. Otherwise the first time it tries to slow down a proc (assuming these are dual core procs), your system will freeze hard. -Chris Carlos Chavez wrote: On Fri, 2005-08-26 at 18:21 -0400, Michael Stahl wrote: I have FC4 working well! I have sucessfully deployed Asterisk on several AMD 64 servers using Fedora Core 3 and 4. My office PBX currently uses FC4. Just be sure to install the kernel-devel packages so you can compile Zaptel. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small office setup/using analog lines w/ Ast erisk
Colin Anderson wrote: I'm quite aware that Opterons et al have gained signifigant market share in the past couple years and AMD and their supporting chipsets have dramatically improved in quality. I'm actually an AMD fan. However, when you are spec'ing a system that a business will depend on (and your Asterisk server is arguably the most important piece of kit in the rack) why would you introduce an unknown variable in the equation just to save, say, 30% on the price of the chip and motherboard? The only comment I have in the same line as your comment is this. Anyone who just goes out and buys a piece of hardware to run asterisk on for thier business and does so, because it's a "tier 1" or reputable manufacturer falls in my category of "gambler". If you haven't taken the hardware you plan to build your business phone system on, and installed the software it will be running and beat the ever living snot out of it on a test bench, then you are asking for a suprise. This smells of the "You never get fired for buying IBM (replace with Cisco, etc)" quote. Sure you can take the small gamble that a "tier 1" platform will meet your needs, however, nothing beats a full battery of tests including burn in, capacity and failure mode tests. If you don't know how your system is going to fail and when it will fail, then how are you going to monitor it? All in all, had Jenny's contractor done this, she would be in much better shape now, however given the fact they didn't much care that the card was sharing an IRQ with the network interface. -Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small office setup/using analog lines w/ Ast erisk
Colin Anderson wrote: Oh, great. ;-) For next time, does anyone have recommendations for a particular motherboard or a particular type of motherboard? Yup. Intel chip. I know people will say I'm trolling, but I wouldn't use an AMD for an Asterisk box. Workstation, yes. I run one myself, they work fine. Note I said Chip not Chipset. I've had good luck with older Intel chipsets but from what I understand some other guys have had problems with the new Intel chipsets. I totally trust Asus though. What I would do were I you is Asus with no onboard stuff as much as possible with a P4 chip. Colin, Would you possibly explain why you prefer Intel based systems over the AMD based system for Asterisk? When you speak of Intel here are you talking about Celerons, Pentium 3, Pentium 4, Xeon, Pentium D? When you speak of AMD are you talking XP, MP, Duron, Sempron, 64, 64 FX, X2, Opteron? -Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] unsolicited NOTIFY messages from Asterisk
Olle E. Johansson wrote: Chris A. Icide wrote: Note however, that in the unsolicited NOTIFY that Asterisk sends for MWI, it includes a ;tag= as part of the NOTIFY. This will break some devices as they will not accept the NOTIFY because the tag doesn't match any transaction that is open. The correct way to send an unsolicited NOTIFY would be without the tag altogether. That's interesting. Any suggestions on such a device? I haven't seen that. I guess if they do that, it's ok, since sending unsolicited NOTIFY is theoretically wrong anyway. If the phone software wants to support it, they wouldn't bother with the tag, I guess. /O ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Olle, Yes, a Metaswitch softswitch (and I suspect some other softswitches would as well). It returns a 403 error if you send a NOTIFY for MWI to it with a tag attached. This is a little different implementation. The Metaswitch is providing the end user (over PSTN) service, and when a phone goes unanswered, it sends a the call to asterisk over sip. I extract the phone number that was originally dialed from the SIP Header using the dialplan command for reading sip header messages, and send the call to voicemail. Now I have to send back indication that a message exists. Since the Metaswitch isn't sending a subscribe for every single line, we have a push model. So we push out MWI, but Asterisk tags the push, and the Metaswitch rejects it since it has no previous transaction with a matching tag. I patched the code so it won't send a tag on an MWI Notify for my own use, but it's specific to this implementation and would cause problems where a subscribe has been sent. -Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] unsolicited NOTIFY messages from Asterisk
Note however, that in the unsolicited NOTIFY that Asterisk sends for MWI, it includes a ;tag= as part of the NOTIFY. This will break some devices as they will not accept the NOTIFY because the tag doesn't match any transaction that is open. The correct way to send an unsolicited NOTIFY would be without the tag altogether. -Chris dbruce wrote: If the SIP device is sending a SUBSCRIBE, then it is specifically asking for a NOTIFY. The only time that Asterisk wilol send an unsolicited NOTIFY is in the case of a MWI. The reason that the unsolicited NOTIFY is sent as a voicemail notification is that the vast majority of vendors have implimented it in this way. very few devices can explicitly subscribe for a MWI, so Asterisk uses the registration of the device as an implicit request. For us to be able to help you with your issue, you need more information that just a SIP debug. You need to: 1) Tell us the device you are using (is it a phone (specific model), is it a connection with an ITSP). 2) get a debug output from the console. 3) get a debug from the SIP DEBUG command. Remember, the more information you give, the less obscure your problem will be to the reader, and the higher your likelyhood of receiving an answer that will be of help to you. Regards, Derek Bruce - Original Message - From: Mark Edwards To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, July 18, 2005 3:05 PM Subject: Re: [Asterisk-Users] unsolicited NOTIFY messages from Asterisk can you give a little more detail here? I assume * is sending the SIP messages. Where is is sending them. To the ITSP or the PHONE? What does the SIP message contain? Is is MWI? is it 'OPTIONS' keepalive? Try posting a Sip Debug output from * that contains the dialogue in question cheers Mark On 7/18/05, Subashini C V - CTD, Chennai <[EMAIL PROTECTED]> wrote: hi, i am getting unsolicited NOTIFY messages from Asterisk after the subscription. Is this type of NOTIFY messages is supported in any of the RFCs..? Because, the SIP stack with 3265 compliance, does not support any such NOTIFY messages and discarding those. I need the justification for this kind of NOTIFY messages sent by Asterisk. I need your valuable inputs... Thanks in Advance, Subashini ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: passing through MWI info from SBC
John Novack wrote: Mike Myers wrote: Wow, this is a serious problem for me. I don't need to actually check the voicemail itself from Asterisk, just to be able to tell that there is voicemail waiting. Are you saying there is no way in Asterisk to do this? Is that true for using Digium hardware as well as FXO ports on a SIP ATA? Vonage VM doesn't matter to me, since I'll turn it off and use Asterisk for that functionality, but determining SBC's VM status is very important. My whole wife's family (multiple households) uses it. In the past, if one family tried to switch to a non SBC provider, they always returned in less than a week because of lack of VM interoperation. So my wife will put the kibosh on the whole Asterisk project unless I can light the MWI light when SBC VM is waiting. Since the cheapest analog phones can do this, I don't think she's going to understand that these $200 Polycom phones can't... :-( Is there no way around this? Thanks, Mike Here is what I would do. Install a TDM04 card with a couple fxos. Connect the analog phones that your wife will be using to the tdm card. In zapata.conf, set those phones to immediate=yes, and when you get an event on the fxo port, connect it to the fxs port with the stutter tone. This way, when she picks up the phone, it will immediately connect her to the sbc provided dial tone, and she can hear the stutter or lack thereof. When a call comes inbout however, you can still route it as you want. Not a perfect solution, since the phones she will be using are forced to use SBC, but the best solution I can think of. -Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GUI that supports virtual PBX's/users
Deon wrote: A friend of mine runs a small office building, 10-15 tenants. Each have their own company, their own thing, renting space from him. His main PBX is getting dated and his tenants are complaining. I was telling him about Asterisk but his main concern is he doesn't want to have to always be the one to add/remove extensions, or change the IVR hours or whatever. Does anybody know of a free or even a commercial (he's got money) GUI for Asterisk that will support multiple logins that have restricted access to Asterisk? Like Tenant A log's in, he can control his stuff specifically, extensions and such, can't do anything insane like restart the server or shutdown Asterisk. Tenant B log's in and he can control his stuff specifically, etc. I looked at AMP but AMP is like an all or nothing deal, it has complete control of the config files, no way to section off users and contexts and such. It's designed for a dedicated system to one person/company. Any commercial offerings or anything else? Deon, I have a product I'm working on that does exactly what your friend is looking for, I'll send you an email directly with contact information. It's basically a three level Web GUI, providing system management by the owner (who assigns DID's and lower level management rights), Company level management (who adds end users, routes did's / extensions, manages IVR, conference configs, company speed dials, company blacklist, etc.) and a User level including follow-me configuration, personal speed dial and black/white lists. -Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk failover solution
Of course this solution can be broken by phones that cache DNS answers, and the phone may require a reboot to actually force it to go and contact the smart DNS server for a new ip address. -Chris Michael Stahl wrote: If your phones are setup to connect to the asterisk box by name, then a smart DNS server can just point phones to the backup box after failure. However, since asterisk running on the backup box doesn't know about the phones, this is only half the solution From: Mohamed A. Gombolaty [mailto:[EMAIL PROTECTED]] Sent: Thursday, June 30, 2005 8:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk failover solution Dear All, I am using Linux-High Availability between two Asterisk servers, everything is fine but I do have one problem with this, When a server fails and the other assumes the ip address and start asterisk on server 2, the ip phone must re-register themselves again, otherwise the phones are dead. Does anyone have Ideas of how to overcome this. -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [Fwd: Asterisk Balancing solution]
Matthew Boehm wrote: Damon Estep wrote: Dear All, I am using Linux-High Availability between two Asterisk servers, everything is fine but I do have one problem with this, When a server fails and the other assumes the ip address and start asterisk on server 2, the ip phone must re-register themselves again, otherwise the phones are dead. Does anyone have Ideas of how to overcome this. SIP RealTime should solve this problem. If both machines use SIP RealTime then it doesn't matter which server the phone registers to cause if the other server recieves a call for the phone, it will lookup in the db how to contact that phone. -Matthew Interesting, but I'm using realtime right now, and even though the sip phones register, the realtime db for each sip device is not updated with any location information. -Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting Caller ID after Dial
What about setting and using Accountcode for each sip client? It tracks separately than callerid in the cdr. so in your sip.conf, add an accountcode= statement for each sip entry, and in the AccountCode field in the CDR, you'll have the correct entry needed to determine who made the call. -Chris Chee Foong Chiew wrote: Hello, I have the following situation: I have a PRI with 200 DID numbers and I have set up 200 sip extensions that matches the last 4 digit of the corresponding DID numbers so that when any of the 200 DID number is called, asterisk can pass the call to the respective sip extension. Incomming has been fine. But when making out going calls I want the called party to always see the same number (which is one of the number selected from the 200 DID numbers). This I can be achieved in asterisk by calling SetCallerID before Dial command. However in the CDR, the caller id number of the number that i set using SetCallerID is always logged and there is no trace of which sip extension is making the call since the caller is always the same. This has become a serious trouble for billing. I have been searching around and could not seems to get a solution. I have tried DIAL_STATUS variable (only work if call is not answered), using 'g' option in Dial command (does not work if calling party hangup first), etc. Is there a solution or work around for this? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gnudialer
Jesus Mogollon wrote: Hi there! Is there anyone out there using gnudialer? I tried vicidial but couldn't get it to work (does vicidial support SIP trunks anyways?). Gnudialer seems to be simpler, though their web interface needs a little work (version 2.0 seems like a step in the right direction but it isn't out yet). How do you register an agent? The documentation is lacking... Jesus Mogollon I've got it running. It took a little work, but it's running. Use the agent program to log in the agents. Version 2 should be out this week, by the way, if you can wait, I would. -Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Remote Voicemail Notifier / enter Dialplan onSIPRegister
Luki wrote:
OK -- follow up, for the record.
After looking at the patch code, it looks like the documentation is wrong:
[default]
1000:sip-reg-srv-001 => 1000,John Doe,[EMAIL PROTECTED]
should be:
[default]
[EMAIL PROTECTED] => 1000,John Doe,[EMAIL PROTECTED]
... and then extensions.conf can have the usual call to VoiceMail
without specifying the remote server or context:
VoiceMail(u${EXTEN})
That works fine for me :-). Nice.
--Luki
Luki,
Thanks! I'll update the docs, sorry for the pita error in the docs.
-Chris
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Re: [Asterisk-Users] voicemail comprehension
Decarpentrie Guy wrote: Hi all, In order to do loadbalancing between my two *, i wanted to stock all things concerning voicemail on a NFS partition... I see that the voicemail system put his files onto two differents directories : /var/spool/asterisk/voicemail/mycontext etc. and /var/lib/asterisk/voicemail/mycontext etc. I've two questions : Why ? and how can i do to centralize the destination of the messages AND of annonces in a unique directorie ? thx in advance. Take a look at this patch on mantis (http://bugs.digium.com/view.php?id=4371) With a little creative dialplan work, you can centralize your voicemail server, not using NFS, but over IAX. -Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Remote Voicemail Notifier / enter Dialplan onSIPRegister
I've added the patch for CVS HEAD 05-02-2005 to the Mantis entry ( http://bugs.digium.com/view.php?id=4371) last night as well as the patch to CVS HEAD as of 5/24 -Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Remote Voicemail Notifier / enter Dialplan on SIPRegister
Anton Krall wrote: Damn! This is very nice Chris! Actually it's Martin's work, I just funded the development. If you like the features, please go to Mantis and download the patch, test and comment. -Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Remote Voicemail Notifier / enter Dialplan on SIP Register
There is a patch on Mantis (http://bugs.digium.com/view.php?id=4371) Which includes several features. 1. Support for central voicemail server(s) with remote server notification via IAX In other words, this patch allows you to configure an Asterisk server as a central voicemail server and to send out voicemail notification to remote Asterisk servers who can then pass the notification on to local clients. 2. Diaplan command to update voicemail pointers The patch includes a dialplan command (ChangeMailbox) which allows you to change both the central voicemail server as well as the remote clients mailbox pointers. In other words, you can add/move 'mailbox=' fields via the dialplan. 3. Entry into dial plan upon device registration or loss of registration This function allows you to configure a dialplan context in which SIP or IAX devises will activate a dialplan thread allowing you to execute any non-audio stream dependant dialplan command(s). An example might be identifying the location of the registration and using the changemailbox command to direct the correct voicemail indication to the registering device. We need community support if we want to get this added to asterisk. Please take a gander. I also have a patch against CVS head 05-02-2005 if you want to test against CVS HEAD prior to all the function changes. Below are notes on the configuration: First, the remote voicemail notification configuration: In voicemail.conf under the general section we need to identify that this voicemail server is a centralized voicemail server supplying remote notification. This is done by adding an entry that points at an iax entry for this server. Note that you must add an entry for this server in your IAX configuration. [general] voicemail_server= On the same server we will configure the voicemail boxes. There is one new entry required over the normal user configuration and that is the remote asterisk server hosting the client device to which we will be sending voicemail indication. This is included as an IAX entry (must also exist in your IAX config) as shown here [default] : => ,,etc.. or if we have voicemail box 1000 being served for a client device attached to IAX peer 'sip-reg-srv-001' as defined in your iax.conf file, the line would look like this: [default] 1000:sip-reg-srv-001 => 1000,John Doe,[EMAIL PROTECTED] All servers (both central voicemail and remote client) must have iax entries for each other (and the central server must include an entry for itself in it's own iax.conf file). Servers can be dyanmic (host=dynamic) as well as static. In the sip.conf or iax.conf for each device there is a change to the mailbox= argument. Note that the mailbox= still works for locally served mailboxes. For remote mailboxes the format is this: mailbox=:@ so in the example above, it would look like this sip.conf [1000] type=friend host=dynamic mailbox=1000:[EMAIL PROTECTED] where 'vm-srv-001' is the entry in the iax.conf for the central voicemail server. This configuration does not include automatic routing of voicemail commands (voicemail and voicemailmain). You must still route calls to voicemail to the central voicemail server through your dialplan. If you try and call a voicemailbox directly on a remote client you will get a no such voicemailbox error. So when you want to leave voicemail or retrieve it, the call needs to be sent to the central voicemail server where you would then execute the VoiceMail or VoiceMailMain functions. Second, the ChangeMailbox function There is an application that you can change the mailbox= definition in either iax.conf/sip.conf or voicemail.conf (changes are NOT reflected in the configuration files, they are not updated by this command) ChangeMailbox(tech,name=newsetting) On the central voicemail server, you can change both the voicemail entry as well as the sip or iax entry. On the remote client servers you can only change the sip/iax entries. To change the remote server in which a client's voicemail status is sent, you would use the Voicemail keyword for technology in the following structure: 1) Voicemail ChangeMailbox(Voicemail,@) On the system which hosts the UA device, you would use the following structure, with SIP or IAX as the technology: 2) IAX/SIP/etc ChangeMailbox(IAX,=:@) To completely change the location a voicemail indication is being sent, you may need to exectute this command on both the central vm server as well as the remote client server. The entry on the voicemail server tells the vm server what remote asterisk server to send the notification to. The entry on the remote client server tells the local asterisk system which device to notify. So in the case that the device only changes (you want to send or add vmail indication to another device on the same server) only the SIP or IAX entry need be changed. However if the device moves to a different remot
[Asterisk-Users] Remote Voicemail Notifier / enter Diaplplan on SIP Register
There is a patch on Mantis (http://bugs.digium.com/view.php?id=4371) Which includes several features. 1. Support for central voicemail server(s) with remote server notification via IAX In other words, this patch allows you to configure an Asterisk server as a central voicemail server and to send out voicemail notification to remote Asterisk servers who can then pass the notification on to local clients. 2. Diaplan command to update voicemail pointers The patch includes a dialplan command (ChangeMailbox) which allows you to change both the central voicemail server as well as the remote clients mailbox pointers. In other words, you can add/move 'mailbox=' fields via the dialplan. 3. Entry into dial plan upon device registration or loss of registration This function allows you to configure a dialplan context in which SIP or IAX devises will activate a dialplan thread allowing you to execute any non-audio stream dependant dialplan command(s). An example might be identifying the location of the registration and using the changemailbox command to direct the correct voicemail indication to the registering device. We need community support if we want to get this added to asterisk. Please take a gander. I also have a patch against CVS head 05-02-2005 if you want to test against CVS HEAD prior to all the function changes. I will try and post both on Mantis and here on -Users some config file samples and a little better explanation of the configuration sometime later today. -Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rack Mount Server Recommendations
Michael B. Murdock wrote: Is there anywhere (or anyone) who has compiled some recommendations on rack mount servers for Asterisk? We are currently using Dell 2650 and Dell 2850 but are seeing some problems with the raid controllers failing and are now shopping for a suitable alternative. Ideally the server would be 19in rack mount, build with similar quality to the the Dell's, and have a -48VDC power supply option. Oh yeah, and run asterisk like a champ. Any suggestions? -- Mike I use http://www.siliconmechanics.com for all my Asterisk work. I recommend them to all my clients. You might pay a little more than you will for dell, but the build quality is better, and the systems run like champs. Ask for Ken Hostetler and tell him Chris Icide sent you. He'll point you in the direction of the right hardware. -Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Old DBGet/DBPut vs. new Set(var=${DB(...
On 07:57 PM 5/15/2005, Jean-Yves Avenard wrote:
Hello
I upgraded to CVS head yesterday (due to the lack of zaptel drivers
working with 2.6.10)
And noticed that now DBGet and DBPut have been deprecated in favour of the
new Set/DB one.
In the UPGRADING.txt in Asterisk it says:
* The applications DBGet and DBPut have been deprecated in favor of
functions. Here is a table of their replacements:
DBGet(foo=family/key)Set(foo=${DB(family/key)})
DBPut(family/key=${foo}) Set(${DB(family/key)}=${foo})
I fail to see how DBGet and DBPut can be replaced by those two commands
If I want to create a new database entry:
DBPut(CFIM/200=300)
I will create the entry if it doesn't exist
With the new Set(${DB(CFIM/200)}=300) I get:
May 16 12:39:39 WARNING[1]: func_db.c:54 function_db_read: DB:
CFIM/200 not found in database.
-- Executing Set("SIP/ipp100-1d45", "=300") in new stack
-- Executing Playback("SIP/ipp100-1d45", "auth-thankyou") in new stack
as abviously DB(CFIM/200) always get replaced by its value which in this
instance doesn't exist yet
the other serious problem is that DBGet used to automatically jump to
prioriy n+101 if the entry didn't exist. Now I will do things like:
Set(temp=${DB(CFIM/200)})
which will set temp to "" instead of jumping to an error.
I wish DBGet and DBPut weren'\Ãwë t removed their replacements are no
good and can't be made to behave the same without serious re-work (like
testing the returned entry is not null etc...)
Any ideas on work-around or did I miss anything?
First, either go post on -dev about the differences between the old way and
the new. Functionality was NOT supposed to change. You can continue to
use the old functions for now, if they have been removed (they should still
be there for the time being) then fall back a little in time. The May 2nd
CVS HEAD should be working the way you are used to.
Definately, post this bug to -dev if it's not already there
-Chris
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Re: [Asterisk-Users] Is there a product to simulate a PRI trunk?
On 01:00 PM 5/13/2005, Robert Goodyear wrote: > > >On May 13, 2005, at 12:39 PM, Chris A. Icide wrote: > >> You should note that while this "works" technically, it won't catch >> any issues that you may experience when connecting to other PRI >> switches. >> >> In other words, two asterisk servers connected back to back with a T1 >> cross-over cable won't tell you that Asterisk's NFAS code doesn't work >> with Lucent 5ESS switches, or that Sangoma's code pre-firmware v1.1 >> and pre-driver beta6 versions won't bring up the D channels when >> connected with certain switches. Everything will work between two >> asterisk boxes perfectly. >> > >Right. Which is why I wanted to figure out if there was a way to >emulate the particular signaling protocol; in my instance, NI2, to >ensure I'm doing all the right things on my end based on what I tell >the simulated telco end to impersonate. > There is no way, with an asterisk box to emulate a particular vendor's switch. You can always change your signalling type and present different signalling protocols (asterisk supports quite a few), but you will never be able to emulate a switch from a different manufacturer, and then of course you have different firmware and model versions to deal with as well. I can say from experience that connecting two asterisk boxes back to back will get you 80% of the way there, when testing for functionality, but you won't know if you are going to run into a problem until you connect it to the actual switch hardware you plan to connect. PGP.sig Description: PGP signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is there a product to simulate a PRI trunk?
You should note that while this "works" technically, it won't catch any issues that you may experience when connecting to other PRI switches. In other words, two asterisk servers connected back to back with a T1 cross-over cable won't tell you that Asterisk's NFAS code doesn't work with Lucent 5ESS switches, or that Sangoma's code pre-firmware v1.1 and pre-driver beta6 versions won't bring up the D channels when connected with certain switches. Everything will work between two asterisk boxes perfectly. -Chris On 11:28 AM 5/13/2005, Charlie Watts wrote: > >That is incorrect, you need a crossover cable to go directly between two >T1 jacks if the Telco isn't in the middle. A T1/E1 crossover cable swaps >pins 1&2 with 4&5. > >For PRI, configure one end as pri_net, and the other end as pri_cpe. > >Works just fine. Lots and lots of folks do this. Channel banks, legacy >phone systems, multiple asterisk systems, modem servers. > PGP.sig Description: PGP signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GR-303 zaptel and zapata configurations
On 12:08 PM 5/12/2005, Michael B. Murdock wrote: > >Did you ever may any progress on this Chris? > >I am also interested in figuring out how to configure the GR-303 as a CPE >device. > >-- Mike > No luck as of yet. The person I was working with on the switch side went on vacation, so my testing and configuration attempts have been put off until the 16th of this month. I've heard nothing at all back from anyone at all otherwise. -Chris PGP.sig Description: PGP signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Connecting 20+ asterisk servers together
I would think you could try something like this to at least reduce the management nightmare. Configure 1 (or more for redundancy sake) route servers (dual-power supply, raid 1, etc.) boxes, and maintain routing information on these. These servers should know exactly how to get to all the end UA's in your system. Each route server would have an IAX entry for all the other asterisk servers in your network. When a call is made, route it to the central route server(s) which will then pass it to the correct far end. Make sure you do not disable IAX transfer, and the central server *should* create the connection (including trunking?) and then step out of the path, allowing the originating server a direct IAX connection to the terminating server without having to maintain a huge full mesh of IAX entries on each server. In this model if you add more servers on the edge, you only need update the central servers. Of course the downside of this is what happens if you lose the central servers. I've not tried this myself, but it should work. -Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zttool: BLU/RED Alarm
-BEGIN PGP SIGNED MESSAGE- Using zttool with a Sangoma A104 card, I am seeing a BLU/RED alarm on a circuit. What is the meaning/significance of the BLU alarm? I've never seen a blue alarm before. Is this specific to the Sangoma card, or do Digium cards generate the BLU alarm as well? - -Chris -BEGIN PGP SIGNATURE- Version: PGP 8.1 iQCVAwUBQneqvu0LTNca2q41AQF8dwQAvmvwxh2j8r06rxqeHTSsRF4aeztt+O4H luxiiQSsPICk4RCHSio/3gz025ClquY0KWSdk9J2KlSReAmHyL8FyMcgPwy8O50f oaY0YJbwVs0FHuFrnOsAijmSfJNmo/D2Q/iP7gIwkzqoPjdWam30A8VptoTk6lJs ghJESOoHZFY= =fHr1 -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 "multi-line" configuration
-BEGIN PGP SIGNED MESSAGE- On 09:50 PM 5/2/2005, Matthew Boehm wrote: > >Hold up. So you have Phone #1. And all 6 lines register with the username of >"phone1" ? > >And you have phone #2; and all 6 lines register with username of "phone2"? > >And the phone only registers once? Interesting..I'm gonna test this. Sounds >like it'd be a solution to 1 of my many problems. Yes, I have a 7960, and lines 1 through 6 are set to the same auth name and auth password. They all point at a single entry in the sip.conf table. The 7960 however only sends one register to the server. It just now has six presentations of that single entry (and actually can support 12 calls to that device if you allow call waiting) - -Chris -BEGIN PGP SIGNATURE- Version: PGP 8.1 iQCVAwUBQncHJ+0LTNca2q41AQHf/QP7BB5ni6GOzc7JxvavF+ryg172gBtlIWku hmw5JkcinUBGKcRQ9paMXcZ+NRMokUFyljF+Yc1xLWPp4Gt1u/PCYmnU2tO/RIYg JAffPN5fVcA5zq5+uw/n0utwpUpo0VTzKPErcHonLJrr+ZF7MYxIiQ3NxHpQjAeR zOZw81xfyvU= =vHoj -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 "multi-line" configuration
-BEGIN PGP SIGNED MESSAGE- On 06:36 PM 5/2/2005, Matthew Boehm wrote: > > >>> You can't use the same extension on multiple line buttons but you can >> Yes you can with the 7940's and 7960's. It works fine for a lot of people >> so calls will "roll" to the next button. I have this set up on several >> customer sites. > >You can use the same "extension" but the usernames must be different. >Right? > No, you can use the same username and secret for all 6 lines A 7940 or 7960 will just "do the right thing" That right thing being, roll over the new call to the second line if the first is busy, etc. - -Chris -BEGIN PGP SIGNATURE- Version: PGP 8.1 iQCVAwUBQnbg1O0LTNca2q41AQEXPwP8DIZZo3W2vEZjVzmFUHa4J6uOHI/WB1JH FD8STuiXMb0g+KmABe89DDtR521q4Yqr4D230GxDKqNm4WzFmd5n39MAiIVTrgam sbrAiCGU6qw+15jer2KnuD5HAWLuJuHgcKG5tdrhogdLXBPQ8wI/Ng12lyTLLv6K BKY7RM3G9o8= =pd7W -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GR-303 zaptel and zapata configurations
-BEGIN PGP SIGNED MESSAGE- Does anyone have any working example GR-303 configurations for zaptel and zapata conf? The information available on the wiki as well as in the sample configurations just doesn't seem to be enough to bridge the gap for me. In Zaptel.conf, Do you set up a GR-303 circuit like a PRI with b and d channels or do you set fxo or fxs, ks signalling? How do you configure the channels for the control and timing channels (12 and 24 on a T1 from a 5ESS switch)? In zapata.conf, Trunk groups require 1 or more signalling channels, so this would indicate that you need to set up the circuits in zaptel as PRI? signalling can be gr303fxoks_cpe/net? Do you assign CRV's instead of channels, or do you assign channels and crv's? I would appreciate any help here. I'm trying to set up an asterisk system as a GR-303 CPE device off of a 5ESS switch using 2 T1 circuits. - -Chris -BEGIN PGP SIGNATURE- Version: PGP 8.1 iQCVAwUBQnJ91u0LTNca2q41AQEeXAQA5irA7dJYvLYR2xSIYl3E/dGklbBxR/k7 n2AxrDfTQIwIAPDaSKZZp4VvCGL7fo6fT5Gvdhdu4tRWe9blqRV5GpdF/IpD/zVq h3XVvBn7xArJi6pK8Ed+pjFnqIbwszK+WpFYJQs0CLVAc+xdDJqrAsKNC0lfXmt7 VPEl5dtxUJ4= =z9ey -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E1/T1 back to back ??
On 07:46 PM 3/13/2005, Brett, Gary wrote: > > >Hi there > >Just a quick question, I will be building some servers in a lab utilizing >Digium E1 cards. I would like if possible to avoid the expense of installing >an e1/ISDN30 in my lab. I have two questions really, first does anybody know >of an effective simulation tool I can use to replicate a real world PRI but >without the telco line being installed. And secondly, can I have a scenario >with 2 asterisk servers with digium e1 cards 'back to back' one configured >as the network side and the other configured as the client side (can I just >use a single cat5 straight through cable between them ?? and cant the Digium >e1 cards operate ok in both modes?) > >Any advice would be greatly appreciated Yes, you can connect two asterisk systems together back to back using t1/e1 interfaces. You will need a T1 crossover cable (do a google on T1 Crossover). Make sure you set signalling to net and cpe (if using pri signalling). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium : no lead time!
If you do a web search for ipVolution TDM120, you should find someone who claims to have a card that does such a thing. -Chris On 02:21 PM 3/9/2005, Brandon Patterson wrote: > >Supposed to be someone in Minn. working on this. I heard the name Dan. >He might be in the hardware biz. > > >> On Wed, 2005-03-09 at 15:18 -0600, Matthew Boehm wrote: >>> Again, may be off topic but are there any cards out there supported by >>> asterisk that have on-board DSPs to do better 729->711 or 729->PRI >>> conversion? >> >> Not yet, and I don't know if anyone is working on the drivers for such a >> card. >> -- >> Steven Critchfield <[EMAIL PROTECTED]> >> >> ___ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >___ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZAP/PRI Error: channel reported in use
I have a system with two 4 port T1 cards, with 5 PRI's configured. Each
PRI is configured as an individual PRI and belongs to it's own group
(groups 1-5)
This system is handling roll-over from another system, where any error in
processing the call on that system results in it being sent here. This
mainly results in all calls resulting in a busy being sent for retry
here. I then try the groups in order, 1-5. However, spans 2,3, and 4 are
erroring out with a strange error. They are configured the same as
circuits 1 and 5, and the carrier claims they are all configured the same.
The error I'm concerned with it this:
-- Forcing restart of channel 0/21 on span 2 since channel reported in use
See error in context below.
-- Executing Dial("IAX2/[EMAIL PROTECTED]/1", "Zap/G1/5551212") in
new stack
-- Called G1/5551212
-- Channel 0/23, span 1 got hangup
-- Zap/23-1 is busy
-- Hungup 'Zap/23-1'
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing Dial("IAX2/[EMAIL PROTECTED]/1", "Zap/G2/5551212") in
new stack
-- Called G2/5551212
-- Channel 0/21, span 2 got hangup
-- Forcing restart of channel 0/21 on span 2 since channel reported in use
-- Hungup 'Zap/45-1'
== No one is available to answer at this time (1:0/0/0)
-- Executing Goto("IAX2/[EMAIL PROTECTED]/1", "203") in new stack
-- Goto (unipoint,5551212,203)
-- Executing Dial("IAX2/[EMAIL PROTECTED]/1", "Zap/G3/5551212") in
new stack
-- Called G3/5551212
-- Channel 0/21, span 3 got hangup
-- Forcing restart of channel 0/21 on span 3 since channel reported in use
-- Hungup 'Zap/69-1'
== No one is available to answer at this time (1:0/0/0)
-- Executing Goto("IAX2/[EMAIL PROTECTED]/1", "304") in new stack
-- Goto (unipoint,5551212,304)
-- Executing Dial("IAX2/[EMAIL PROTECTED]/1", "Zap/G4/5551212") in
new stack
-- Called G4/5551212
-- Channel 0/21, span 4 got hangup
-- Forcing restart of channel 0/21 on span 4 since channel reported in use
-- Hungup 'Zap/93-1'
== No one is available to answer at this time (1:0/0/0)
-- Executing Goto("IAX2/[EMAIL PROTECTED]/1", "405") in new stack
-- Goto (unipoint,5551212,405)
-- Executing Dial("IAX2/[EMAIL PROTECTED]/1", "Zap/G5/5551212") in
new stack
-- Called G5/5551212
-- Channel 0/22, span 5 got hangup
-- Zap/118-1 is busy
-- Hungup 'Zap/118-1'
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing Hangup("IAX2/[EMAIL PROTECTED]/1", "") in new stack
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[Asterisk-Users] Help with configuring CFAS groups
I've got a system with 5 pri circuits configured into a CFAS group with a primary and secondary d channel. There are three TE410P cards in the system. The 5 circuit span are located as follows: circuit 1 on span 5 circuit 2 on span 1 circuit 3 on span 6 circuit 4 on span 2 circuit 5 on span 9 primary d chan is on chan 24 of span 5 (chan 120) secondary d chan is on chan 24 of span 1 (chan 24) T1 circuits are configured correctly in zaptel.conf and show up and good zapata looks like this: trunkgroup => 1,120,24 spanmap => 1,1,2 spanmap => 2,1,4 spanmap => 5,1,1 spanmap => 6,1,3 spanmap => 9,1,5 [channels] signalling=pri_cpe switchtype=national pridialplan=unknown ;group 1 group=1 context=cfas-in channel=1-23 channel=25-48 channel=97-119 channel=121-144 channel=193-216 -- pri show span 5 (seems it sets the span number by the physical span with the primary d channel) shows: localhost*CLI> pri show span 5 Primary D-channel: 120 Status: Provisioned, Up, Active Switchtype: National ISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 Secondary D-channel: 24 Status: Provisioned, Up, Standby Switchtype: National ISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 --- When the system is started, or I issue a ztcfg command, I see the following: Dec 10 13:31:40 WARNING[5649]: chan_zap.c:7569 pri_dchannel: Restart requested on odd/unavailable channel number 0/1 I get this for 0/1 through 0/23 When an inbound call to this CFAS group is placed, I receive the following: Dec 10 13:43:06 WARNING[5649]: chan_zap.c:7640 pri_dchannel: Ring requested on unconfigured channel 0/1 on span 5. What am I doing wrong here to cause this unconfigured error? The provider is seeing both of my d channels, so I have the spans mapped right. -Chris ___ 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] Headsets for Cisco 7940/7960
I've tried 5 different headsets, some amplified, some using the headset jack. I've settled on one. The Toughset 10, is very comfortable, and has excellent sound quality. http://www.vxicorp.com/storefront/detail.asp?PRODUCT_ID=200585&l2=callcenter -Chris On 03:01 PM 11/21/2004, Brian Pavane wrote: >What headsets have people found work well with the Cisco 7940 and 7960 >phones? To date, I have tried a couple of the headsets within the >Plantronics H series (H41-N), and noticed that the volume of my speaking >is lower over the headset than on the regular handset. I am currently >looking for headsets that are known to work well. I do know that Cisco >lists the H-91 and H-101 as certified to work, however these are both >over-the-head type models. I was looking for an over-the-ear model, as >I would like to be able to provide a variety of headsets depending on >the individuals taste. I am not looking for a headset that requires an >external amplifier, but rather a headset that can make use of the >headset jack on the phone itself. >___ >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] Using CallingPres to set up CallerID blocking
On 12:08 AM 11/21/2004, Chris A. Icide wrote: Okay, ignore the previous question, I figured it out. for anyone else who may also not completely grasp the wiki explanation: number is a octet, and the only bits you need worry about are bits 1,2,6 and 7 bits 1 and 2 define the screening indicator, and bits 6 and 7 define the Presentation indicator. The placement of the Bits and Meaning Header messed me up a little bit, as did the '(octet 3a)' as I was wondering what importance the 3a had. So some general settings: Presentation Allowed, Network Provided: 3 (0011) Presentation Restricted, User-provided, verified, and passed: 33 (0011) Presentation Restricted, Network Provided: 35 (00100011) -Chris ___ 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] Using CallingPres to set up CallerID blocking
From the Wiki: Presentation indicator (octet 3a) Bits 7 6 Meaning 0 0 Presentation allowed 0 1 Presentation restricted 1 0 Number not available due to interworking 1 1 Reserved Screening indicator (octet 3a) Bits 2 1 Meaning 0 0 User-provided, not screened 0 1 User-provided, verified and passed 1 0 User-provided, verified and failed 1 1 Network provided How do these bits fit into "number" where CallingPres(number)? Is number is binary field or a decimal field? It is 8 bits long (the word octet sure seems to indicate that)? What would number be if you wanted to block Presentation on the far end (when placing a call through a Q.931 PRI) if you wanted to present Anonymous or Unavailable? What would number be if you wanted to present a full callerid as User-provided, verified, and passed? -Chris ___ 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] Multiple asterisk process
On 10:42 AM 11/20/2004, Jose Hernandez wrote: > >>Did you bother using google? > >I searched google but could not find an answer. Any other suggestions? > http://lists.digium.com/pipermail/asterisk-users/2004-April/043852.html ___ 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] MYSQL Dialplan Question
On 10:19 AM 11/15/2004, Michael Shuler wrote: >If you want good mysql/postgres/odbc/etc. support use >http://svn.asteriskdocs.org/res_data/ > I may be incorrect, but I believe that res_data only lets you move configuration information into a database, however, what if you want to access databases and tables that have nothing at all to do with configuration data? The MYSQL application allows you to access any MySQL host.database.table. -Chris ___ 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] high-capacity systems / trouble with Tyan
one other question: What kind of power supply do you have in the AMD system? On my 2466, I had alot of problems until I upgraded my power supply to a high quality 500W unit. I seem to remember a while back reading that AMD systems were much more sensitive to power issues that comparable Pentium units. On 02:23 PM 11/10/2004, mattf wrote: >Hello, > >I've had a Tyan dual Athlon MP(2800) machine for a year now and have had >several lockups for strange reasons on stock redhat kernel and on custom >compiled kernel off of Slackware. I've tried every combination of BIOS >settings and changed out all assiciated hardware and found the problem: It's >the Tyan. I've also had issues with a couple of SCSI RAID cards when I tried >using them with the Tyan card. > >This all would have really upset me if the Athlon MP platform performed >better than the Intel platform, but it doesn't. This Dual Athlon MP system >actually handles LESS total Asterisk load than a single P4 3.2 GHz, and the >P4 has a lot more Motherboard options and cost much less. > >This is just my experience, I'm sure I am using Asterisk a little >differently than you, I don't have 3 Quad T1 cards in any of my machines, >but if that's what you're looking for, I'd suggest the PowerPC(Mac) >platform. Asterisk installs just fine right on top of Yellow Dog Linux and >the bus speed of a Mac mops the floor with most x86 motherboards, meaning >more bandwidth for those bus-hungry Digium boards. > >MATT--- ___ 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: FW: [Asterisk-Users] Need a creative solution - Caller IDanda stupidupstream
On 11:39 AM 11/8/2004, Paul Rodan wrote:
>Help anyone? I hate caller ID.
I would do something like this:
Set accountcode to the callerid number for each sip ua. In other words if
my callerid for a sip UA was "John F. Doe" <2025551212>, then I would set
the accountcode to 2025551212
Then I would create an context/extension that people would dial for setting
a forward number and include it in the contexts available to the SIP UA:
[features]
exten => 1234,1,Answer
exten => 1234,2,Playback(enter-fwd-number-at-tone)
exten => 1234,3,Read(number,,11) ;set length as you see fit, 11 allows +1
US dialing
exten => 1234,4,Wait(1)
exten => 1234,5,Playback(You-entered)
exten => 1234,6,SayDigits(${number})
exten => 1234,7,Background(press-1-if-correct-2-if-incorrect)
exten => 1234,8,Goto(7)
exten => 4321,1,Answer
exten => 4321,2,DBPut(${ACCOUNTCODE}/FEATURE/FORWARD=0)
exten => 4321,3,Playback(forwarding-disabled)
exten => 4321,4,Hangup
exten => 1,1,DBPut(${ACCOUNTCODE}/FEATURE/FORWARD=1)
exten => 1,2,DBPut(${ACCOUNTCODE}/FEATURE/FWDNUMBER=${number})
exten => 1,3,Playback(thankyou)
exten => 1,4,Hangup
exten => 2,1,Goto(1234,2)
Then for inbound calls I which go to the SIP UA, I would check forward
status:
[macro-ring-sip-ua]
; ARG1 is sip extension, ARG2 is timeout, ATG3 is options, ARG4 is callers
callerid
exten => s,1,DBGet(FWDSTATUS=${ARG1}/FEATURE/FORWARD)
exten => s,2,GotoIf($[${FWDSTATUS} = 1]?s,20:s,10)
exten => s,102,NoOp(No DB entry FORWARD for ${ARG1})
exten => s,103,Goto(s,10)
exten => s,10,Dial(SIP/${ARG1},${ARG2},${ARG3})
exten => s,11,Voicemail(u${ARG1})
exten => s,12,Hangup
exten => s,111,Voicemail(b${ARG1})
exten => s,112,Hangup
exten => s,20,DBGet(FWDNUM=${ARG1}/FEATURE/FWDNUMBER)
exten => s,21,SetCallerID("${ARG4}" ${ACCOUNTCODE})
exten => s,22,Dial(local/[EMAIL PROTECTED])
exten => s,121,NoOp(No DB entry for FOWARDNUMBER for ${ARG1})
exten => s,122,Goto(s,10)
The idea here is that you are sending out the original caller's ID as the
TEXT field and your callerid as the number field.
Please forgive any typos above, I did this in a few minutes. It should at
least point you in a good direction if this solution is of interest to you.
-Chris
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RE: [Asterisk-Users] Queue announce behavior for callback agents?
On 02:25 PM 11/7/2004, Damon Estep wrote: >I guess I need to understand the relationship between queue, agent, and >ackcall. Seems like if ackcall can tell the queue the call has been >answered it must be somewhat aware of what queue called it. Is it that >there is a generic token passed? > ackcall doesn't interact with queue at all. Basically, here is the way things appear to work. The key to understanding how agents work is to remember it is a channel, not an application. Think of it like you would think of a SIP channel or ZAP channel. Call is top of the queue Queue then through whatever strategy it's using pools an agent. The message it sends to chan_agent could be considered the same as a SIP invite for all purposes. At this time agent rings the end user through whatever technology is set in agent.conf The agent answers and (in this case) presses #. Now, agent sends back a message to queue that the channel has answered the invite. Queue now plays the announcement to the agent if it is configured, and then connects the two channels together. As you can see, there is no logic in the two modules to send the announcement message prior to considering the channel answered. app_queue would need to be modified to accept an intermediate signal from chan_agent indicating that app_queue should play an announcement, but not consider the channel answered. chan_agent would need to be modified to provide the new message. -Chris ___ 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] Queue announce behavior for callback agents?
Damon, You could do something like that, but ackcall wouldn't know what queue called it, so the message would end up needing to be fairly generic. After you hit #, then queue would play back a message that can be unique to the queue that called the agent. To get a message unique to the queue that polled the agent, you still will need to modify both app_queue and app_agent. Unfortunately, my C skills are so poor, doing this would probably take me a week alone. I think your best option would be to either post a bounty, post a feature request on bugs.digium.com and add a note that you will pay for the work, or as a last resort (you might get somewhat flamed for this) go to the asterisk-dev list (or IRC channel might be better) and ask if someone would be willing to do the coding for you (for a fee of course). If you just post the request on bugs.digium.com and don't offer a bounty, you might get the code done for free, but I wouldn't count on it being soon. -Chris On 01:51 PM 11/7/2004, Damon Estep wrote: >Chris, > >What if ackcall could be given parameters, like the name of a sound file >to play between the answer event and the end of the timeout event? This >would then only be a modification to ackcall, right? > >Here is the sequence I envision; > >Agent call out event; Answer event; Ackcall sequence, modified to loop a >sound file for the remaining timeout duration, like "I have a call for >the support queue"; # to ack; Notify queue if there is an ack; Queue >plays announcement, like "connecting..."; (optional in the current >code). Seems like this would not require modifying the queue code based >on your earlier analysis. > >Now, if I only had the skills to envision the c required to implement >it! > >Is this a smaller project? Since we have no in-house coders we would >need to have this written for us (yes, I realize there would be a cost), >with hopes that it would then be useful enough to others to be added to >the project so we do not have to maintain workarounds. > ___ 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] "night" mode ideas
[features]
exten => 001,1,DBPut(FEATURE/STATUS=1)
exten => 001,2,Hangup
exten => 002,1,DBPut(FEATURE/STATUS=2)
exten => 002,2,Hangup
exten => 003,1,DBPut(FEATURE/STATUS=3)
exten => 003,2,Hangup
[inbound]
exten => s,1,DBGet(STATUS=FEATURE/STATUS)
exten => s,2,GotoIf($[${STATUS} = 1]?ringall,s,1)
exten => s,3,GotoIf($[${STATUS} = 2]?ringone,s,1)
exten => s,4,GotoIf($[${STATUS} = 3]?ringvoicemail,s,1)
exten => s,5,NoOp(STATUS is UNSET, please set STATUS immediately by
dialing 001,002, or 003)
exten => s,6,Hangup
where ringall is the context that
handles ringing X, Y, and Z
ringone is the context hangling ringing Z only
ringvoicemail is the context sending the caller to Voicemail
immediately
You can have a cron command like such
* 21 * * 1,2,3,4,5 asterisk -rx "database put FEATURE STATUS
3"
* 8 * * 1,2,3,4,5 asterisk -rx "database put FEATURE STATUS
1"
This should do what you described below
-Chris
On 12:30 PM 11/7/2004, Tom Lahti wrote:
>Hi all. I have a system I'm going to build that I have a
pretty good idea
>how I'm going to accomplish this feature, but I was wondering if
anyone had
>a better idea/method since mine seems somewhat "hackish"
for some reason I
>can't explain :)
>
>The system wants to have different incoming call handling
more-or-less
>based on time of day, but we don't want it to be at some precise UNIX
time,
>but rather under human control, so if people stay late they can keep
it in
>a more appropriate mode.
>
>What we're looking for is a way for someone on the system to pick up
an
>extension and dial some code that changes the incoming call context,
i.e.
>manually switching incoming call handling. We want incoming
calls to ring
>extension X->Y->Z in one mode, ring only Z in a second mode,
and only take
>voicemail in a 3rd mode. Th idea I came up with to solve this
was to code
>a special extension for each mode that used Authenticate() and then
>System() to copy a different extensions.conf into place and restart
>Asterisk for each mode. Restarting seems a bit harsh since it
will drop
>any calls in progress.
>
>There would also be a cron job that would swap modes if they hadn't
been
>swapped by some time of day.
>
>Does this sound like a reasonable solution, or do any of the experts
(ahem,
>not me) have a more elegant solution idea?
>
>--
>Tom
>
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Re: [Asterisk-Users] Queue announce behavior for callback agents?
From a quick glance, it appears that this is not a trivial change. the ackcall is handled by the agent channel, and the agent channel doesn't tell queue that the channel has been aswered until the # key is pressed. Once queue understands that the channel is answered it plays the announcement. To make what you want work, both applications would need additional coding to have app_agent be able to notify app_queue that the end device has answered but wants an announcement message before accepting the caller, and also supply information to app_queue that the end user accepts or rejects the call after hearing the annoucement. You may want to submit a feature request at http://bugs.digium.com/ asking for this feature. -Chris > >CVS-HEAD-10/30/04 > >When using the queue announce= in conjunction with a callback agent it >appears the # key must be pressed before the optional announcement is >heard by the agent (as far as I can tell), this seems to make sense for >always logged in agents, but is in the reverse order of what one might >expect for callback agents. > >I realize the # is an acknowledgement that not only was the call >answered, but there is a real live body on the answering end. Perhaps >the announcement could loop until timeout or ackcall? > >Is there a method available to play the announcement before the ackcall ># key is pressed? What about to loop it until timeout? > ___ 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] Need a dial plan as follows
Take a look at queue.conf and agent.conf. These are some of the features that should be most interesting: the annouceoveride and timeout options of the Queue command and ackcall in agents.conf. You should be able to achieve what you are looking for with these hints. -Chris On 05:04 PM 11/6/2004, Damon Estep wrote: >I need to put together a dial plan for an extension that has this >behavior; > >Answers the incoming call to the extension. Puts the caller on hold with >music. Calls several other extensions and external numbers >(sequentially). Presents each user that answers one of the extensions or >external numbers with an option to accept the call that is on hold or >not (important!). Bridges the calls if one of the called parties >accepts. Sends the caller to VM if no one can be reached or no one >accepts the call. > ___ 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] res_config problems
Hello, I'm having some issues with res_config. I've used BKW's perl program to take a working sip.config and place it into the database. Asterisk doesn't load in the sip table from the database. What am I doing wrong? Here are CLI lines: == Parsing '/etc/asterisk/res_odbc.conf': Found Nov 5 11:53:56 NOTICE[12572]: res_odbc.c:133 load_odbc_config: registered database handle 'mysql1' dsn->[MYSQL-asterisk] Nov 5 11:53:56 NOTICE[12572]: res_odbc.c:345 odbc_obj_connect: Calling 0x8110d68/0x8110d78 Nov 5 11:53:56 NOTICE[12572]: res_odbc.c:359 odbc_obj_connect: res_odbc: Connected to mysql1 [MYSQL-asterisk] Nov 5 11:53:56 NOTICE[12572]: res_odbc.c:385 load_module: res_odbc loaded. [res_config_odbc.so] => (ODBC Configuration) Nov 5 11:53:56 NOTICE[12572]: config.c:556 ast_config_register: Registered Config Engine odbc == Parsing '/etc/asterisk/extconfig.conf': Found == Binding sip.conf to odbc/asterisk/ast_config res_config_odbc loaded. [chan_sip.so] => (Session Initiation Protocol (SIP)) Nov 5 11:53:56 NOTICE[12572]: config.c:492 __ast_load: Loading Config sip.conf via odbc engine == Parsing '/etc/asterisk/sip.conf': Not found (No such file or directory) Nov 5 11:53:56 NOTICE[12572]: chan_sip.c:8477 reload_config: Unable to load config sip.conf, SIP disabled odbc.conf: [global] dsn=MySQL-asterisk username= password= where username and password are replaced with functional username and password with access to the database/table res_config_odbc.conf: [settings] table => ast_config connection => mysql1 res_odbc.conf: ;;; odbc setup file [mysql1] dsn => MYSQL-asterisk username => password => pre-connect => yes databse table: mysql> select * from ast_config; ++++---++--+--+--+ | id | cat_metric | var_metric | commented | filename | category | var_name | var_val | ++++---++--+--+--+ | 1 | 0 | 0 | 0 | sip.conf | general | context | default | | 2 | 0 | 1 | 0 | sip.conf | general | port | 5060 | | 3 | 0 | 2 | 0 | sip.conf | general | bindaddr | 0.0.0.0 | | 4 | 0 | 3 | 0 | sip.conf | general | srvlookup| no | | 5 | 1 | 0 | 0 | sip.conf | 2000 | type | friend | | 6 | 1 | 1 | 0 | sip.conf | 2000 | secret | 2000 | | 7 | 1 | 2 | 0 | sip.conf | 2000 | auth | md5 | | 8 | 1 | 3 | 0 | sip.conf | 2000 | disallow | all | | 9 | 1 | 4 | 0 | sip.conf | 2000 | allow| ilbc | | 10 | 1 | 5 | 0 | sip.conf | 2000 | allow| ulaw | | 11 | 1 | 6 | 0 | sip.conf | 2000 | canreinvite | no | | 12 | 1 | 7 | 0 | sip.conf | 2000 | nat | yes | | 13 | 1 | 8 | 0 | sip.conf | 2000 | mailbox | [EMAIL PROTECTED]| | 14 | 1 | 9 | 0 | sip.conf | 2000 | callerid | "Test Config" <2000> | | 15 | 1 | 10 | 0 | sip.conf | 2000 | host | dynamic | | 16 | 1 | 11 | 0 | sip.conf | 2000 | qualify | 200 |
Re: [Asterisk-Users] Looking for a SQL or ODBC Application
Thanks, I just found the add-ons one while searching mantis. This looks like exactly what I was looking for. -Chris On 01:37 PM 11/4/2004, William Suffill wrote: >there should be 1 addons for mysql and anthm wrote res_sqlite which >would add the same functionality but use sqlite to backend it >___ >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] Looking for a SQL or ODBC Application
Is anyone aware of an Asterisk application that allows you to read and write values to multiple tables in a database. Im not looking for an app that will let me externalize asterisk's database. I'm looking for one that will let me do normal SQL like queries and inserts. An example: exten => _XX.,1,SQL(PEER=Select peer from egress where route_id=1) ... exten => _XX.,1,SQL(update egress set technology='Zap' where route_id=1) etc. Chris ___ 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] Cannot start asterisk - CAPI issues - Hurray!
On 05:20 AM 11/1/2004, Jean-Michel Hiver wrote: >Funny I had to disable rate_engine.so though... here is the message I got: > > [rate_engine.so]Nov 1 16:18:48 WARNING[1076992544]: loader.c:299 >ast_load_resource: /usr/lib/asterisk/modules/rate_engine.so: undefined >symbol: ast_pthread_create >Nov 1 16:18:48 WARNING[1076992544]: loader.c:480 load_modules: Loading >module rate_engine.so failed! > I had the same issue recently. I was upgrading an asterisk install for some of the recent database features, and it broke rate_engine. So I fell back to the last asterisk version that I knew worked with rate_engine. CVS 9-29-04 seems to work fine with rate_engine. -Chris ___ 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] high-capacity systems / trouble with Tyan
On 06:57 AM 10/30/2004, Andrew Kohlsmith wrote: >On October 29, 2004 11:49 pm, Chris A. Icide wrote: >> Only in the X100P format, and only 2 of them > >I have to ask -- why are you running such high-end equipment for a craptastic >FXO device? Don't you find other issues that going to a TDM4xxP or even a >T1+channel bank would fix? I mean I ran an X100P and TDM410P (1FXO+1FXS) on >a weenie P90 for cryin' out loud... What's this super big system getting you >(except maybe future proofing?) > The box is one of my development boxes for developing Asterisk based solutions for my clients. I've always been of the mind that you can't have enough power... -Chris ___ 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] high-capacity systems / trouble with Tyan
On 06:05 PM 10/29/2004, Jim Gottlieb wrote: >> I have had X100P, TDM4XX, and TE4 cards in it with no issue. > >Have you had multiple cards in it at the same time? Only in the X100P format, and only 2 of them > > >> I never even tried the 2.4 kernels in the >> system, I built the 2.6 kernel before installing asterisk. > >We've been sticking with the 2.4 kernel in Fedora Core 1. I installed >FC2 with its 2.6 kernel and couldn't even get asterisk to compile. The culprit is the RedHat kernel. I don't know what redhat does with their kernel or sources. But If you build your own kernel from non-redhat source, asterisk will compile perfectly. I recently had that problem with a client's system. They wanted to run FC, we tried compiling asterisk with the FC kernel to no avail. I went to kernel.org, grabbed 2.6.6 source compiled it, and everything worked perfectly. > >I realize that Redhat isn't the only Linux. I've only been using it >because it's what we've always used and it's what I'm most familiar >with (though my girlfriend prefers SUSE). The only thing wrong with RedHat as far as asterisk is concerned is that they do something goofy with their kernels and all you need do is recompile a kernel from source. IMHO, you should always compile a kernel for your specific hardware. -Chris ___ 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] high-capacity systems / trouble with Tyan
On 03:47 PM 10/29/2004, Jim Gottlieb wrote: >We wanted to try the new AMD MP 2800 chips on the newer Tyan S2466 >motherboard, but the systems hang or panic (with DMA errors) after >starting the zaptel drivers. We tried putting the older slower CPUs on >the new motherboard and had the same trouble. I currently have a development system I use when developing configurations for my clients. It's a Tyan 2466 motherboard with the latest bios revision, running with two AMD 3000 MP processors. RedHat 9 is the system, but I have a non-redhat 2.6.6 kernel. I have had many versions of asterisk running on it with no issues whatsoever. I have had X100P, TDM4XX, and TE4 cards in it with no issue. I never even tried the 2.4 kernels in the system, I built the 2.6 kernel before installing asterisk. You may want to migrate away from the redhat kernel, and build your own kernel whether you build a 2.4 or a 2.6 is up to you, but from what I understand the 2.6 kernel is more efficient in SMP form. Chris ___ 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 quad T1 install
On 07:24 PM 10/22/2004, Mark Phillips wrote:
>Firstly, I have a PRI which will be connected to the card. How does *
>know that a call for 732 111 3714 should get routed to extn 3714?
Simple enough, your PRI provider will send some portion of the number to
you, many times it's the last 4 digits, sometimes I've seen 10
digits. Either way, it will be enough to make all the numbers mapped to
your pri unique. Contact your PRI provider if you don't know how many
digits they are sending. Then you just match those by extension...
[inbound-pri]
exten => 3714,1,
>
>How do I get 3714's CLI to be 732 111 3714 on outbound calls?
[outboud-us-ld]
exten => _1NXXNXX,1,SetCallerID("My Company Name" <7321113714>)
exten => _1NXXNXX,2,Dial(Zap/g1/${EXTEN})
>
>How does * handle the calls going out? I'm guessing this is something to
>do with Dial(zap/something)?
Read the your sample zapata.conf file. Use a dialing group (for example:
group=1), then dial like above.
>
>Any tips etc would be greatly received.
>
>Mark
>
>
>--
>
>Mark Phillips, G7LTT/KC2ENI
>Randolph, NJ
-Chris
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[Asterisk-Users] Cannot send # to far end, asterisk intercepts.
Okay, I know there have been previous posts about this, and there is the patch 110 on mantis that was never added to CVS. If you have a mixed environment of SIP based phones and ATA adapters, how can you still allow the ATA style phones to use the transfer function, yet allow all dtmf to be passed on to the far end? I thought the double # answer was a good one, but Mark brought up the fact that it's non-intuitive and if people can't figure out how to flash a phone, then they surely can't figure out the timing in pressing two #'s. Has anyone figured out a solution to this, that isn't 'replace the ATA's with SIP phones'? -Chris ___ 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] Least Cost Routing
On 05:44 PM 10/13/2004, Duane wrote: >Matthew Boehm wrote: > >> If any others are successfully using another Least Cost Routing method, >> please pass it along. > >For simple things you can slightly hard code it into the dialing plan in >asterisk, for example in mine I have asterisk setup to check e164.org >then goes to a cheap voip provider and failing that goes to pstn. > >Larger scale I'd probably go with an AGI script load all >routes/providers into a database and then have the AGI script pull all >providers for a route out of the database in order of cost and loop >through them till a call succeeded... > >If you need it to scale better you'd probably have to resort to writing >an app in c, then sending the call into that. > But the trollphone system does exactly this in c code as an asterisk app. Why write a AGI? the folks at trollphone included an sql schema file as part of the package. It includes all table structures. I think it's schema.sql in their distribution. -Chris ___ 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] rate_engine substitue db field?
Is anyone familiar with the Trollphone's LCR package? There is a field in the egress table labeled substitue. Placing a $1 there results in the correct dial extension being passed. However how is this field used to substitute replacement dial extensions... in other words as an example, lets say my EXTEN as passed to and back from the rate_engine is 01144123456789 and I want to have a route in the table which just leaves that be (so I just use $1), but I also have a route that I want to use, but I need to strip off the 011. In an asterisk dial plan I would use $1:3. What is the regexp format to use in the substitute field. I can't seem to find any documentation from Trollphone folks of how to format that field. -Chris ___ 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: Asterisk Conferencing using g729
On 12:53 AM 9/7/2004, Tony Mountifield wrote: > >> Another related question: Is there a way to just use g729 for the conference and for nothing >> else. The problem I have is that I have Broadvoice ( BV rocks, by the way) which requires >> ULAW and sends DTMF inband. If I allow g729 in the sip.conf, Asterisk complains that inband >> dtmf is only supported under ULAW and incoming dtmf does not work through Asterisk, >> something I must have. > You may very well have hit on a bug (well, really a feature request). Asterisk *SHOULD* do the conversion, so if your Sip UA originates the call, and between it and Asterisk, they choose g729, and then asterisk originates a call to Broadvoice, with the selected codes as Ulaw, then Asterisk should take incoming out of band dtmf from the UA and generate in-band dtmf for broadvoice. If this isn't happening, it should. You might be able to reset the codec by using the channel variable for the codec. I forget off the top of my head what it is, but surely the wiki or the archives of this list will provide that information. I'm not sure though, once the invite from the UA is sent and accepted, that changing that variable will force a re-negotiation of the codec? If it does, then you could force ulaw to the UA when you end up in the broadvoice context, or force g729 when going to conference, and default to ulaw -Chris ___ 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] offtopic - channel banks
On 11:24 PM 9/6/2004, Ilia Mirkin wrote: >OK, so we're getting close. E&M is something that rides in the T1 >datastream. Now - I have E&M cards in the channel bank. So I can set up >* to talk through the T1 to the E&M card. (please correct me if I've >misunderstood...) You are correct > >The next problem is what goes on with the pairs coming out the back. >Does * provide the dialtone that the channel bank passes on to the pair >in the back? Is the assumption made that all of those pairs will be >connected to standard analog phones? Yes, Asterisk provides the dialtone (thus you are able to receive stutter tone as well if the configuration is correctly done when you have voicemail) And yes, the assumption is that the lines are connected to regular old buy 'em at k-mart analog phones. > >Getting really close to understanding this > ___ 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] offtopic - channel banks
Ilia, Okay, we are stretching the limits of my official technical knowledge and getting into the realm of empirical knowledge (i.e. I messed around and got it to work). The T1 framing and encoding (esf,b8zs for example), will get you green lights on the T1 side of the channel bank, but if you hook up a phone to the correct pairs matching a channel output on the channel bank, you won't get dial tone. You'll need to match the signalling (E&M, E&M Wink, fxs_ls, fxs_ks, etc.) between the channel bank and the asterisk t1 configuration. This will then allow you top provide correct signalling to the analog handsets giving you dial tone as well as off-hook, on-hook detection, etc. So to answer your question, E&M is not a framing protocol, it rides on top of the framing and encoding protocols and is the protocol that interacts with the channel bank to provide correct signalling between the analog handset and the T1 interface of Asterisk. -Chris On 10:59 PM 9/6/2004, Ilia Mirkin wrote: >I must not be explaining my question clearly. Is E&M a framing protocol >that the signal that travels "through" the T1's channels conforms to, or >is it a wire protocol that will drive a pair on the centronics >connector. If it is the former case, then what drives the pair on the >centronics connector, and will i be able to plug a standard analog phone >into it (and expect it to work)? If it is the latter, then I assume that >there is no easy way to connect an analog phone. > >Thanks to everyone for responding to my questions. > >--- >Ilia Mirkin >[EMAIL PROTECTED] > >On Mon, 2004-09-06 at 11:33, Chris A. Icide wrote: >> Ilia, >> >> Think of a channel bank as a concentrator. In a single T1 channel bank, >> you concentrate 24 analog two wire phone connections into a single 4 wire >> digital interface. >> >> So in your case, the RJ45 connector is for the T1 interface to Asterisk, >> the local CLEC, or whatever you intend to connect it to. On the T1 side, >> there has to be several layers of signalling and encoding. Alot of this >> information is superfluous, but may help you when it comes to understanding >> your configs. >> >> Before we even talk about E&M signalling, you have the T1 framing and >> encoding. This is used to allow both ends of the T1 circuit to understand >> how the 24 channels are being configured on the 4 wire circuit. It's >> generally going to be either sf, d4 or esf, b8zs. This is known as the >> framing and encoding. >> >> Once those are agreed upon, then we need to set up the way the T1 is going >> to signal across the channels. Normal phone lines (analog) use voltages, >> resistances, and dtmf to signal what it is doing. Since a T1 is a digital >> circuit we can't do that, so we need to set up another way to signal, so >> that the channel bank knows what to do when we send some kind of digital >> signal. In this case, this is the E&M signalling you asked about. >> >> Finally, you probably are looking for some way to plug your phone's RJ11 >> connecter into the channel bank. Unfortunately it's not that easy. That >> big Centronics style connecter is where you actually have to plug up the >> phone. there are 24 pairs of contacts in that connecter that are >> associated with each channel on the T1 circuit. Historically, you would >> connect up a cable of 50 conductors connected to the centronics connector >> on one end, and then to one side a punch down block on the other (just a >> quick-connect style access device for copper wire). On the other side of >> the punch down block, you would connect the wires that would then run to >> the remote wall jacks, etc. where your phones plug in. >> >> The problem you have is wither by google mastery or just plain brute force >> testing, you need to figure out the pinout of that centronics port before >> you can connect up any phones successfully. >> >> -Chris >> >> On 10:56 PM 9/5/2004, Ilia Mirkin wrote: >> >While I understand everything that you have said, I'm still a little >> >confused. Yes - I have what looks like a centronics connector on the >> >back. So, I can do "t100p with e&m signalling" <-> "act-1241 e&m card" >> ><-> what? Namely, if the E&M card deals with the T1 end of the channel, >> >how do I get that to a real phone? Will it "just work" if I plug an >> >analog phone onto the correct pair coming out of the connector in the >> >back? If not, what is the output of the E&M card? (and, more >> >importantly, what would I n
Re: [Asterisk-Users] offtopic - channel banks
Ilia, Think of a channel bank as a concentrator. In a single T1 channel bank, you concentrate 24 analog two wire phone connections into a single 4 wire digital interface. So in your case, the RJ45 connector is for the T1 interface to Asterisk, the local CLEC, or whatever you intend to connect it to. On the T1 side, there has to be several layers of signalling and encoding. Alot of this information is superfluous, but may help you when it comes to understanding your configs. Before we even talk about E&M signalling, you have the T1 framing and encoding. This is used to allow both ends of the T1 circuit to understand how the 24 channels are being configured on the 4 wire circuit. It's generally going to be either sf, d4 or esf, b8zs. This is known as the framing and encoding. Once those are agreed upon, then we need to set up the way the T1 is going to signal across the channels. Normal phone lines (analog) use voltages, resistances, and dtmf to signal what it is doing. Since a T1 is a digital circuit we can't do that, so we need to set up another way to signal, so that the channel bank knows what to do when we send some kind of digital signal. In this case, this is the E&M signalling you asked about. Finally, you probably are looking for some way to plug your phone's RJ11 connecter into the channel bank. Unfortunately it's not that easy. That big Centronics style connecter is where you actually have to plug up the phone. there are 24 pairs of contacts in that connecter that are associated with each channel on the T1 circuit. Historically, you would connect up a cable of 50 conductors connected to the centronics connector on one end, and then to one side a punch down block on the other (just a quick-connect style access device for copper wire). On the other side of the punch down block, you would connect the wires that would then run to the remote wall jacks, etc. where your phones plug in. The problem you have is wither by google mastery or just plain brute force testing, you need to figure out the pinout of that centronics port before you can connect up any phones successfully. -Chris On 10:56 PM 9/5/2004, Ilia Mirkin wrote: >While I understand everything that you have said, I'm still a little >confused. Yes - I have what looks like a centronics connector on the >back. So, I can do "t100p with e&m signalling" <-> "act-1241 e&m card" ><-> what? Namely, if the E&M card deals with the T1 end of the channel, >how do I get that to a real phone? Will it "just work" if I plug an >analog phone onto the correct pair coming out of the connector in the >back? If not, what is the output of the E&M card? (and, more >importantly, what would I need to do to hook it up to an analog phone?) > >Thanks for clearing things up. > >--- >Ilia Mirkin >[EMAIL PROTECTED] > >On Sun, 2004-09-05 at 04:31, Steven Critchfield wrote: >> On Sun, 2004-09-05 at 03:10, Ilia Mirkin wrote: >> > hi, >> > >> > i have some newbie questions about channel banks. i have an adtran >> > act-1241 sitting around. it accepts D4 modules, and it contains a number >> > of e&m cards. >> > >> > first of all, how does this thing work? a t1 contains 24 channels, and i >> > noticed that the channel bank has space for 24 cards. what do these >> > cards do? what are their outputs? the ones that are in there have some >> > outputs on the front marked "test", but nothing else. there are a number >> > of wires coming out the back (48, if i had to guess), and it has a few >> > ports on the front which seem to be able to take in a T1. am i correct >> > in understanding that it is the card in the bank that determines the >> > signalling style, and not the t1? as such, is there no way that i could >> > use it in its current configuration to have it talk with analog phones >> > (i.e. something like t100p -> act-1241 with e&m cards -> phone)? i'm a >> > bit unclear on the different signalling types, and their >> > intercompatibilities. >> > >> > if anyone could shed any light into this, i would very much appreciate >> > it. >> >> Think of the T1 as 24 digital digital pathways. The coding of each >> pathway must be compatible on each end. With E&M cards, you signal with >> E&M and the line will work. The cards plug into a backplane where the >> controller routes the digital signal to the card and then optionally >> hook up the output from the card to a connector that consolidates many >> lines. Look for something that looks like an older 50 pin scsi D >> connector. >> >> If there is 2 RJ45 jacks on the front, and 2 50 pin D connectors on the >> back, then it is likely that each card controlls 2 lines each. If there >> is only 1 50 pin connector, then there is only 24 channels. >> >> Hope that helps. > >___ >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/
Re: [Asterisk-Users] Dynamic dialplan
On 06:25 AM 9/1/2004, Juan Jose Comellas wrote:
>We intend to use Asterisk with a very large dialplan (with a lot of
>functionality for 3000+ users). Each user will be able to change several of
>his parameters in the dialplan, so we will be forced to reload the diaplan
>constantly. Has anybody else any previous experience with a similar
>installation? There are some things that we'd like to know, if anybody can
>help us. These are:
>
>- Is this something that can be done safely with Asterisk?
Yes, see the wiki on scaling issues and hardware requirements
>
>- Can we have a diaplan configuration update every 5 or 10 minutes without
>service interruption?
>
A better idea would be to implement a static dialplan that access a database...
for example...
[inbound-pri]
; this context handles all inbound calls arriving on our PRI
; dnid is 4 digits from the telco
exten => 3748,1,DBGet(user-chan=inbound/${EXTEN})
exten => 3748,2,Dial(${user-chan})
just a very simple example of the concept
>- What happens to new calls while the dialplan configuration is being
>reloaded?
Guess this depends upon where in the process of loading a huge extensions
configuration is when the call comes in...
>
>- What happens to active calls after the dialplan configuration is updated?
Active channels will remain active. In other words a call between to
channels will remain in place during a reload.
>
>- Can we do partial updates of the dialplan (e.g. update a specific context
>instead of the whole dialplan configuration)?
I believe the only way to force a dialplan reload is through the CLI
command: extensions reload, which forces extensions.conf to be loaded in full
>
>- Can Asterisk have its dialplan in a database instead of having it
always in
>memory?
Yep, see above comment for just one of the methods to do this, there are
quite a few.
>
>
>Thanks.
>
>--
>Juan Jose Comellas
>([EMAIL PROTECTED])
>___
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Re: [Asterisk-Users] limit the length of extensions
On 07:05 AM 8/31/2004, Deon Rodden wrote: >How do I limit the length of an extension? In my test IVR/Automated >Attendant (whatever it's called), at the beginning it plays "if you know >your parties 3 digit extension, you may enter it now) and then it gives >a list of options. If the caller puts the 3 digit extension, it goes >through fine, if they press 1, or 2 it goes to the selected menu option, >but if they dial 91235551212 it dials that phone number. Which of >course, is a big security risk. > >Is there a way to limit the length of an extension for an incoming call? >My only solution right now is to duplicate ever single extension (about >50 of them) in a seperate context, one that does not have the _9. >extension in it, and then make the call in menu have access to that >context. However, if I put a limit in the entire context of 3 digits, >then my coworkers who's phones are in that context can only dial each >other, not 9 and an outside number. So it has to be an incoming limit or >something. It sounds like you need to break your dialplan into more focused contexts. Create a context for access to an outside line (you can even break this down into access for toll-free, long distance, local, local-toll, international, '976' numbers, etc. Create contexts for each set of company extensions, create contexts for all the ivr systems separately, so only the options you want are available. Then use include statements to only include the access you want for each context. When I set systems up like this I also include 'anti' contexts. For example, if I include a local context that allows 7, 10, or 11 digit dialing, I also include a context !local that plays back an unauthorized message if someone dials a matching number. This way you can not only catch access to unauthorized features by implicitly denying access, but you provide the user with a reason why they are not able to place the call they just dialed. -Chris ___ 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] Hunt Groups
On 11:15 AM 8/17/2004, Chris Modesitt wrote: I have a question about how Asterisk Parses the Dial Plan. To create a hunt-group which would be the appropriate dial plan: [CompanyABC] exten => 722,1,Dial(SIP/801722,60,r) exten => 722,102,Dial(SIP/8014361234,60,r) exten => 722,203,Dial(SIP/8014362345,60,r) exten => 722,304,Dial(SIP/8014363456,60,r) exten => 722,405,Dial(SIP/8014364567,60,r) exten => 722,506,Dial(SIP/8014365678,60,r) exten => 722,607,Dial(SIP/8014366789,60,r) exten => 722,708,Dial(SIP/8014369876,60,r) exten => 722,809,Dial(SIP/8014368765,60,r) exten => 722,910,Congestion exten => 722,1011,Hangup If extensions extension is busy or fails do you always increment by +100 or just the first time? If the line you are trying is busy (in use and has an incomming-limit of 1, or the soft/hard phone reports back busy), then you increment by 101, however if the phone times out (60 seconds in your example) then the plan only increments by 1. so if you placed a call to 722 in your example which I kept above, the first sip phone would ring if it was available (dynamic and unregistered shows up as not available). If no one answers the call, then the dial plan would try to move to priority 2. However if the phone is busy (according to asterisk), then you would jump to priority 102. So, while it's not as technically 'clean' as using queues, the idea you have above will work as long as you add logic to handle a phone not being answered for 60 seconds. It will make for alot of entries under this exten as well. Use of Goto will allow you to limit the number of lines to about twice as many as you have above... exten => 722,1,Dial(SIP/801722,60,r) exten => 722,2,Goto(102) exten => 722,102,Dial(SIP/8014361234,60,r) exten => 722,103,Goto(203) ... etc
[Asterisk-Users] OH323 and G729
Are there any known issues with using the Digium licensed G729 codec with the OH323 channel? -Chris ___ 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] Formatting in sip.conf...can you have 2 @ signs for register?
On 07:12 PM 8/16/2004, John Todd wrote: > stupid top-posting confuses the hell out of these threads, >but I'll continue the insanity. > >I _swear_ I already brought this problem up and it got resolved, like >1.5 years ago. I explicitly remember talking with kram about a >right-to-left parsing trick he implemented to solve this exact >problem. I don't feel like digging through my 3 gigs of mail to find >it, but I _know_ we hit this a while ago, or something very similar. >Why it's broken now, I haven't the slightest idea... > >JT > Actually, someone (I would if I had any coding skill whatsoever) needs to take the incentive to change the register format from register => blah:[EMAIL PROTECTED] to something like this [sip-provider-a] type=peer host=12.23.34.45 disallow=all allow=g729 allow=ulaw nat=yes [EMAIL PROTECTED] secret=password register=yes This would seem to fix all the issues that keep popping up with the single line register format as it exists now. I haven't checked Olle's latest chan_sip rebuild, but by god, I hope something like this is in it. -Chris ___ 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] Performance testing of asterisk
The problem with this kind of testing is that you are testing asterisk, using asterisk as the test generator. When you start getting performance based artifacts, are you going to be able to tell if it's from the sections of asterisk handling the call transactions and media streams or from the sections that are generating the calls. How much of the system resources are going to be used in the generation of the calls, how does this affect asterisk. Is the program flow of internally generated calls exactly the same as the program flow for the type of calls you are using as the scenario for the test? I think you will find that using the same asterisk box as testee and tester will result in not being able to identify what is causing the artifacts... In other words, your testing tool will significantly affect your test platform. -Chris On 03:07 PM 8/16/2004, Tom Masterson wrote: >We are trying to set up some scripts to test asterisk under various loads. >What we are doing is trying to load a bunch of calls in to various queues >atuomatically from various numbers etc so we can see how it behaves. I >think we can do this by loading files in to the var/spool/asterisk/qcall >directory. However the format of this file has a field named identifier >which appears to be a file of some sort in some location. > >1. Are we going about this in the correct direction? >2. Can someone explain what the fields in the file in qcall are and what >identifier should be? > >Thanks >Tom > >___ >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] one extention, multiple phones
On 05:11 PM 7/31/2004, Sean McKay wrote: > Also I'd like to say I believe this is possible with the CCM otherwise >how could an agent (operator) be able to monitor which extensions are in >use with the 7960 expansion device? CCM doesn't use SIP to do this, it uses SCCP. A proprietary cisco protocol. The SIP protocol has been extended to include a subscribe and notify function, allowing a SIP UA to subscribe to a 'service' and be notified of changes in that service. You could use this to at least be able to turn led's on and off, however, you would have to build some functionality into Asterisk to support a sip barging feature based upon picking up the handset attached to the notified line. I'm not sure what the different sip phones that support subscribe/notify actually do when you hit that button when it's lit as well. Obviously you would need some kind of signal from the phone to asterisk to request being connected to that media stream. Would it be an INVITE? How would it be formatted? I'm sure all this is answered in the RFC that describes the subscribe/notify function, it's been way to long since I skimmed through it. -Chris ___ 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] OH323 and codec selection
I'm having a small issue with the oh323 implementation when it comes to codec selection. Version info: CVS Head 6/30/2004 OH323 0.6.3 OpenPhone for windows version 1.8.1 Asterisk is configured as a h323 endpoint which either terminates to the PSTN locally through a PRI or terminates the h323 call to an IAX provider remotely. Asterisk also has G729 licences installed. in oh323.conf we set codecs allowed in the following order: G729 GSM ULAW ALAW When dialing in with OpenPhone with all codecs besides g729 disabled in the audio codec configuration panel, oh323 in Asterisk still picks and uses GSM as the selected codec. Only if I disable all but G729 in oh323.conf will Asterisk use G729 for an incomming h323 call. Am I doing something wrong? Is the order of the codecs in the oh323.conf significant, or is some other method of codec selection being used? -Chris ___ 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] Strange RTP audio errors on console
I have a system running CVS HEAD 6/30/2004. We've only been using it for PSTN to channel bank handsets, but have decided to add sip phones into the mix. Now I have quite a few systems running sip phones just fine as well as some running both sip and analog via channel banks or tdm cards. When we tried to set up some sip extensions (they are behind nats, we are using xten light, and have canreinvite=no as well as nat=yes set in the sip.conf), we only get one way audio. You can hear the other end (be it the asterisk voice prompts or another non-sip user), but the other user cannot hear the sip phone user talking. It gets even more complex. If using the sip phone to call voicemail, or any other asterisk based services the sip user can get dtmf through (yes rfc2833). The asterisk box is on a public IP address, no firewall or nat. The sip clients are 'generally' behind a nat (we've tested from several locations including my home, which I have multiple sip UA's behind a nat, and the very same xten lite is able to work just fine with any of my local asterisk systems (or any others I've tested - all outside of my home nat). The calls are being made with ulaw as the only codec allowed. The sip debug indicates that the call setup has worked and agreed upon ulaw as the codec. CLI provides multiple repeats of the following two errors: rtp.c:1215 ast_rtp_write: Not sure about sending format SLINR packets rtp.c:1058 ast_rtp_raw_write: Not sure about timestamp format for codec Any thoughts, comments, suggestions? Google has been less than helpful given any keywords I came up with to avoice all the NAT/Firewall one way audio posts. -Chris ___ 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] IRC Etiquette
On 02:51 PM 7/26/2004, Jeremy McNamara wrote: > >In my book, respect is earned. They can earn respect by asking informed >questions, but if the documentation is incorrect, what's the point? > I've always wondered where this particular phrase came from. I generally hear it used mostly in arenas where there is a subset of people who feel they are superior in some way to people not in their subset. I think the fact that almost anyone can identify themselves with a subset of people in which they are superior to people outside of that subset. However, does this give them the right to disrespect the other people? I've always said 'disrespect is earned'. I never was a follower of the earned philosophy. Especially in the commercial environment, respect and treating your customers, vendors, suppliers, etc. with respect is nearly as important (and more so in the service field) as your product. Digium sells product which relies upon software which is given away free of charge. While Digium offers support to people who buy their hardware products, they definitely don't have the resources to provide the level of support needed for new users to begin using Asterisk. Thus they rely upon the community to do so. So actions taken by people who represent the Asterisk product (whether it's via IRC, this mail list, or any other public venue sponsored by Digium) reflects upon Digium, and has a real effect upon tier commercial success. So, I'm not at all surprised by Mark's request to actually respect people on the IRC (and on this list as well, I'm sure Mark would agree) regardless of the amount of time they have spent with Asterisk up to the point. Sure, if they turn out to be obnoxious and disruptive, well, disrespect is earned I always say. -Chris ___ 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] DID VoIP trunk provider for metro Chicago, LA and/or Orlando.
On 01:58 PM 7/20/2004, Carmi Weinzweig wrote: >Chris - >In the real telephony world, one can buy a DID trunk without buying a >PRI. If one wants more than about 10 trunks (depending on provider), it >may be cheaper to buy a PRI instead of individual trunks. > >Having said that, most of these VoIP providers have their pricing model >exactly backwards (they seem to only want to compete with Centrex, not >with regular PBX services), in that they charge a lot for resources >that are freely available and cost them little (phone numbers), but >very little for scarce resources (call terminations) that cost them >much more. > >As an example, they purchase a PRI from either an ILEC or a CLEC for >between $100 and $1000 (depending on distance and market) giving them >23 voice channels and as many numbers as they want (again, numbers cost >them at most between $0.01 and $0.10). > >They assign me a phone number (a value of $0.01 and $0.10) and let me >receive as many simultaneous calls as my bandwidth allows (using these >numbers every call absorbs a channel that costs between $4.35 and >$43.48). > >What I would like is to be limited as to how much of a scarce resource >(channels) I can use, but not be limited as to how much of a plentiful >resource (numbers) I can use. > I understand the telephony side of the business, what I was pointing out is that the VoIP folks are using a different business model, with a focus towards low volume users. In that case selling them a 'number or two' for a few bucks a month is probably more financially rewarding, at least as it applies to the main set of customers they have. The market is very immature. Seems most of these companies get their revenue from people putting an Analog to Sip adapter on the end of their analog home phone and buying a single user account. One would have to think that there should be an aggressive salesperson out there looking for a more efficient customer, you just need to find that salesperson. Nufone has michigan and illinois numbers if I remember right, might want to check with Jeremy on trunks for that area. -Chris ___ 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] DID VoIP trunk provider for metro Chicago, LA and/or Orlando.
On 10:41 AM 7/20/2004, Carmi Weinzweig wrote: >I want many phone numbers so that each phone in my facility has its own >phone number, but I really do not need that many simultaneous calls and >it would be cost prohibitive to pay several dollars for each phone >number. It's a different business plan. By going to a VoIP provider, you alleviate the requirement for hardwware you lease or own to terminate PRI's at multiple locations and distribute the calls to your end users. So, you aren't paying for the physical T1 and associated hardware. The VoIP providers are now incurring that cost and must recuperate it (unless they are operatiing under the '90s dot com business plans in which recuperating costs is not required - but you better be ready to turn up a new provider on a moments notice if you are using one of these). So in the past if I am understanding you, you would buy a PRI and pay some fee for the T1 itself, as well as $0.01 to $0.10 per number assigned. In this case, you want to not pay the T1 fee but still pay low per number rates. Maybe if you talked to the providers they might come to a different pricing plan for you that emulates the old way and gives you a better bang for the number? -Chris ___ 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] Polycom IP 500 Voicemail
On 04:28 PM 7/19/2004, Wiley E. Siler wrote: >Mine does the same. Once in Message center I can choose selection >1.Message Center and then soft key Select.Then I select the >registered line that I want to check voice mail on. That is no less than >4 key strokes just to get into your voice mail. Not many to me but tons >to an unskilled user. However, in the documentation regarding the >bypassInstantMessage value, supposedly, setting bypassInstantMessage to >1 is supposed to allow you to go right into voice mail without >navigating the Message Center. That is the big question on my mind at >this point. I have yet to get this to work and I also don't think I am >receiving any SIMPLE messages ti show me that I have messages waiting. > >Do you get a message waiting indicator? > I do get MWI, there are a few things you need to set, and I forget what off the top of my head, soon as I can look and post it here. I haven't tried the bypassInstantMessage value, but I'll take a look and see if I can get it to work. -Chris ___ 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] Polycom IP 500 Voicemail
On 12:40 PM 7/19/2004, Wiley E. Siler wrote: >My Polycom is on loan as a demo and I assume it is one of the first >revision models. In fact it shows as Rev A on the back of the phone. > >I have all the same buttons you listed save for the Messages button. >The 3rd from the bottom on the right column of buttons sayd Voice Mail >on my version. That corresponds to the location of your button that >says Messages. I assume this was changed by Polycom since their phone >has other messaging capability (isntant message for instance) and it was >easier to use Messages and unify the meaning instead of Voice Mail and >lock it into one type of messaging. > >Does your Messages button dump you right into voice mail or do you have >to navigate a menu first? > >Thanks, >Wiley My messages button dumps me right to message center, which I then have to use soft buttons. My IP500 is Rev. C -Chris ___ 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] Polycom IP 500 Voicemail
Strange, I have an IP500 right out of the new-plastic-gadget-smell box, and it doesn't have a button labelled Voicemail. On the left side are the blue speaker, red mute, and blue headset buttons, then next to them top to bottom are the three Line buttons (clear covers for putting your own labels), Directories, Services, Call Lists, Conference, Transfer, and Redial. On the right of the system, top side are the 4 way selection pad with select and delete, then below that are Menu, Messages, and Do Not Disturb, and finally Hold. In the middle are the 12 keypad keys, 4 soft keys, and volume + and - buttons. No where am I able to find a hard voicemail button. -Chris On 10:42 AM 7/19/2004, Wiley E. Siler wrote: >Thank you! > >Can you tell me more about the dial plan feature? How do you setup the >correct digitmap? > >W > >-Original Message- >From: Russ Beaupre, P.E. [mailto:[EMAIL PROTECTED] >Sent: Monday, July 19, 2004 4:56 AM >To: [EMAIL PROTECTED] >Subject: Re: [Asterisk-Users] Polycom IP 500 Voicemail > >Wiley E. Siler wrote: > >> I have a solution that allows me to assign a soft key with no >problems. >> However, it seems like a waste the the hard button labeled Voice Mail >> is not dialing right into voice mail. Is there a known way yo do >> this? I have tried everything in the manual but it doesn't seem to >> work. I have IP 500s and I want to be able to use all three display >> lines for just lines on the phone. >> >I think that feature is inly available on the 1.2.0 sip firmware. It >works on ours but when you press it, you still have to pick a line, then >connect. The line button goes right to the voicemail. > >> Also, do you know if it is possible to program the buttons along the >> bottom of the screen like normal soft buttons? >> >Probably, but I haven't looked into it enough > >> And finally... >> Is there a way to make the system dial without having to hit the Send >> key after dialing a number? >> >look at the digitmap in sip.cfg > >-rb > >___ >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 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 banks, voicemail, and immediate=no
When using a channel bank for analog handsets, you have a couple options in the way you handle transactions involving the analog handsets and origination. With immediate set to no, it appears to me that soon as a digit is pressed after going off-hook, the single digit is taken and processed against the context that the channel is associated with from the configuration in zapata.conf. With immediate set to yes, the extension s in the channel's context is processed. As far as I know, the method of handling channel bank based analog handsets is to use immediate=yes and then have extension s put the phone directly into a DISA command with no-password and a context for processing the entered calls. I have also tried in the past setting immediate=no, parsing off the first digit and sending the call into separate contexts (see example below) example with immediate=yes exten => s,1,DISA,no-password|internal example with immediate=no exten => 9,1,DISA,no-password|pstn-gateway In the first case, the problem I have is this: If I place the handset directly into DISA, how can I get stuttertone MWI indication? If I use the second method, in many cases, there is NO dialtone provided to the phone until after a dtmf entry is recieved. This I suspect is a channel bank issue because it seems to work on some banks, and not on others. Given the use of channel banks as a method to allow large number of analog phones to access an asterisk system, is there any way (or perhaps any interest in developing a method) to actually treat analog handsets on a channel bank like any other UA? In other words, why not have a method besides the two above so that I can stick the phones into a context (which understands it's for handling analog phones on a channel bank) that actually provides dial tone, and accepts dtmf until a match to the context extensions is found? In other words, with immediate=no, I'd like to see asterisk not jump on the first dtmf and try to match (going to i, if no match exists), but actually wait for as many dtmf's as required to match an extension in the context (e.g. exten => _1NXXNXX waits for 10 digits if dtmf 1 is the first digit). On a different track, am I doing something wrong above? For people who have configured channel banks for use with asterisk, have you found a 'perfect' configuration that you prefer to use? -Chris ___ 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: Re: MYSQL_FRIENDS and IAX problem
On 03:33 PM 7/18/2004, usedcanon wrote: >Bug report might be a good idea, I just dropped the issue as I could do without using IAX. I am sure others may not have that flexibility. > >Umar. > >-Original Message- >Subject: [Asterisk-Users] Re: Re: MYSQL_FRIENDS and IAX problem > > >hmm - this is the bad thing about open source etc. > >Should we make a bugreport ? or are we just doing something wrong ? > >> It seems that way, I asked the same question about a month ago, and no one cared to answer. >> >> Umar. >> >> -Original Message- >> Subject: [Asterisk-Users] Re: MYSQL_FRIENDS and IAX problem >> >> >> Hi, >> >> Are there realy no-one who can help here >> >> "Hans-Henrik Andresen" <[EMAIL PROTECTED]> wrote in message >> news:[EMAIL PROTECTED] >> > Hi, >> > >> > I had compiled support for MYSQL_FRIENDS and it works for SIP, but when use tiwh IAX2 I have some problem, I can register with a client, but when I try to make a call I got this error: >> > >> > Jul 17 12:52:03 NOTICE[229387]: chan_iax2.c:5183 socket_read: Rejected connect attempt from >> > >> > When I google'ed this problem I can see other users also found this error (bug ?) But no-one seems to have solved the problem. >> > >> > Any clue ? >> > I believe that 'ast_data' is the solution to this problem, and will probably obsolete mysql friends. However, I could be incorrect in that manner. There are folks on this list who would be much better informed to say whether or not it will obsolete mysql friends. -Chris ___ 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] oh323 dial structure and oh323 debug?
According to the wiki at voip-info.org, the dial structure for using
oh323 without a gatekeeper is:
OH323/@:
or
OH323/
The second option is valid only in the case where a gatekeeper is
used.
NOTE: OpenH323 library v1.12.0 has a bug in the parsing of the
destination
host. When this version is used then the above syntax should be:
OH323/h323:@:
Now, I've got a '*' box that has oh323 running, and it accepts
inbound h323 calls and processes them perfectly (well, not perfectly, but
thats because they are coming in with g729 and going out as gsm, and we
don't have our g729 licenses from digium yet), and now that this is
working as expected, I've been asked to pass any calls with prefix 572
back out to another h323 gateway. Simple enough, in the dial plan I
just matched _572. and tried sending it out to the other h323 gateway
(not an asterisk platform). This is where the problem is.
I can't seem to get the system to send the extension along no matter what
form I try. And to worsen this the oh323 debug toggle CLI command
does nothing (I haven't checked if I need to go back and compile a debug
flag into oh323 yet). I've tried the following in the way of dial
commands:
First, Dial(OH323/) works in that I actually contact
the remote gw and it gives me a bad user message '-- H.323 call
'ip$localhost/6740' cleared, reason 24 (Call ended with Q.931
cause)
', so I know I'm getting to the right gw.
Dial(OH323/${EXTEN:3}@) doesn't work, we don't seem to
parse the @ and just try to reach the whole
argument as an address, error is '-- H.323 call 'ip$localhost/6738'
cleared, reason 11 (Gatekeeper could not find user)'
Dial(OH323/h323:${EXTEN:3}@) (yes, pulling
at straws here because I'm running 1.13.5 lib), also doesn't work... ' --
H.323 call 'ip$localhost/6737' cleared, reason 11 (Gatekeeper could not
find user)'
I've also tried mixing and matching the exten and ip-addy all around to
no avail. Can someone point out the right format of the Dial
command for oh323 when routing a call with a dialed extension to gateway
with a known ip-address. No gatekeepers involved at all.
-Chris
RE: [Asterisk-Users] Bandwidth requirement with G729A
On 08:56 AM 7/14/2004, Kevin Walsh wrote: >> >The 8k is each way, so that makes 16k for voice. > Huh? I'm not sure the relevance of this as it applies to the original question. If the person asking the original question had said that they were looking at adding both inbound and outbound traffic for a total, then yes. However, I think they were only looking at one way traffic since when most people talk about traffic they speak of one direction (and in many cases in the network world, since your bandwidth is generally identical in capacity for both directions, the largest usage is generally the discussed value). In other words, I don't say my T1 is 3.088 Mbps because it's 1.544 in and 1.544 out. However in the case of ADSL I most often refer to it in a manner denoting both the inbound and outbound (256/2048 for example). And in the case of measurement, most measurement tools display bandwidth usage for each direction separately, otherwise troubleshooting would be severely crippled. So the issue the original poser is having is that they are unable to account for measured traffic as opposed to signal traffic. signal is 8kbps but then you have the overhead associated with the UDP and IP wrappers as well as the overhead associated with the ethernet wrapper. There was a post a while back where someone had gone through and listed the overhead specifically per 'wrapper' at least down to the ethernet frame level. You should be able to google it up for specifics. -Chris ___ 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 Asterisk bounty: SIP simultaneous
There is also something known as feature bloat. feature bloat occurs when you take a system that is fully functional and operation and start to add features to it that don't inherently fit the model the system was designed under, so what you end up with si a bloated system with extra features that have been forced to work in an inefficient manner (thus adding the bloat to the feature). The thing I noticed about what Olle said was something to the effect that Asterisk doesn't inherently have the structure to support this feature, and a good bit of asterisk would need to recoded to support it. Considering you can reproduce this capability by using the dial plan or by putting ser in front of it, do we really need to ask the asterisk dev team to stop what they are working on and move down the path of rewriting a good portion of asterisk to allow it to be a proxy as well as a pbx? Can we toss in making it a h323 gatekeeper as well, I kinda get tired of having to run gnugk when I want a nat'd or registered h323 endpoints to work with asterisk. I guess my point is, mark has said in the past that asterisk is a PBX, NOT a proxy, and apparently that is something that is deep down inside the code and not a "simple fix" as Olle has said. I agree wholeheartedly with you, it would be wonderful if asterisk were a proxy as well as a pbx (and a gatekeeper too!), but at this time, it's not an easy fix. -Chris On 03:30 PM 7/12/2004, Kannaiyan Natesan wrote: >I hope you clearly understand that everyone here **KNOWS** to use >alternative software such as SER, what is needed here is the same facility >in asterisk. > >When I see beauty, I want to see all the things in a same figure rather than >splitted. If you see with splitted face doen't mean there is no beauty but >it cause inconvinience. > >What everyone here want to see the same beauty in asterisk and when this can >be done in another softwares why not in asterisk. > >As Olle says, nothing is impossible. There is a possible solution, but takes >time. > >-Kannaiyan. > > >- Original Message - >From: "Sunrise Ltd" <[EMAIL PROTECTED]> >To: <[EMAIL PROTECTED]> >Sent: Monday, July 12, 2004 5:20 PM >Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous > > >> in response to Olle's excellent post, ... >> >> in particular ... >> >> >Asterisk is *not* a SIP proxy. It's a SIP registrar and >> location server. >> >It's a very clever SIP UA. It wants to be in the middle >> of the call >> >and wants to be in control of each device. This >> device-slave view >doesn't match the SIP architecture. >> >> and ... >> >> >I've spent a considerable amount of time investigating >> support for >> >multiple registrations on one Asterisk sip [peer] account >> and after >> >learning about Asterisk's architecture come to the >> conclusion that >> >it is not an easy or even a desirable feature to >> implement. >> >> and ... >> >> >It may be possible, but will probably lead to a lot of >> changes to >> >Asterisk, both core and applications, that no other >> channel will >> >benefit from. A quick hack to support it may lead to a >> lot of >> >confusion on how to handle other apps. And it's a lot >> more work >> >than the bounty will cover. I suggest that you use a >> forking SIP >> >proxy in conjunction with Asterisk to get this >> functionality. >> >> Precisely! A fairly simple and elegant solution. >> >> For those rare occasions where one would really need >> multiple concurrent SIP registrations I'd say one should >> consider running Asterisk in combination with a SIP proxy. >> Since SER is a free download, this wouldn't seem to be >> such a big deal IF IT WASNT for the fact that one will >> then need to run two boxes. >> >> It would make a lot of sense to provide support for an >> easy-to-configure set up where Asterisk can live together >> with another SIP speaking piece of software on the same >> box. >> >> Something along the lines of ... >> >> (ip1:5060)---[*]---[portswapper]---(ip1:5061)---[SER]---(ip2:5060) >> >> Something like this should allow you to run Asterisk on >> one address (ie LAN side) and SER on another (ie WAN >> side), so you get the best of both Asterisk and a SIP >> proxy all in one box. >> >> This would also make it possible to run a SIP softphone >> alongside Asterisk on a notebook, so it would solve two >> birds with one stone. >> >> I'd like to emphasise however, that most of the problems >> described in this thread are NOT good reasons for multiple >> concurrent SIP registrations. Those problems have other >> solutions. Let's take a look at them. >> >> 1) Call centre scenario >> >> Problem: multiple agents should receive calls on the same >> phone number >> >> Solution: assign a number to a call queue and let the call >> queue distribute incoming calls to the agents on different >> SIP phones, each of which should have unique logins for >> reasons of accounting and quality assurance. >> >> multiple concurrent registrations on the
Re: [Asterisk-Users] E100P and T1 channel banks
On 05:52 AM 7/12/2004, Anton Tinchev wrote: >Andrew Kohlsmith wrote: >> On Monday 12 July 2004 07:36, luan au wrote: >> >>>Could you kind Asterians (should we pick Asteroids then?) confirm if I >>>can use an E100P card with a T1 channel bank via * please? I live in the >>>UK hence the question. >> >> >> Yes. You''l only get 24 channels but it shoudl work fine. >> >> And I prefer the term "Astericians" (think electrician), myself. >> >> -A. >Any signaling and framing issues? I can't imagine that this would work at all. TDM multiplexes a set of signals over a certain time window, thus being able to transmit multiple channels over a single transmission line. In other words, there is a set of windows in which we transmit data. In a T1 we have a set of 24 channels and a 1.536 MHz (where in the world did that clock speed come from?) clock is used to sequence the channles. Each channel is 64 kHz and 24 are stacked together to make a signal, so in one cycle you transmit 24 "frames" one frame for each channel (1,2,3,4,24) and then go back and do it again... the real signal for a channel is achieved by taking each frame for that channel (which is every 24 frames) and putting those together to get a channel. The problem you have is that E1 uses a 2.048 MHz signal and T1 a 1.536 MHz, so right off the bat your timing is going to mismatch, so there is no way a T1 port will natively work with an E1 port. I'm not sure, but I think the 4 port cards will run T1 or E1 mode (never had to find out before), but the T100P and E100P are limited to T1 or E1 (again, this is from memory of reading mail list), so I may be wrong. However the answer to your question is you can't plug T1 and E1 hardware together unless one or the other has an autodetect and autoselect function allowing it to switch clocks, framing and coding. T1's use d4 or esf with ami or b8zs as framing and coding E1's use something completely different ccs, cas? sorry, I'm not up to speed on E1, but a quick glance at a zaptel.conf sample should let you know. -Chris ___ 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 Asterisk bounty: SIP simultaneous
On 02:44 AM 7/12/2004, Olle E. Johansson wrote: > >I don't believe there's a quick fix at all. >If you want a quote for a fix, contact me off-list. But remember, that I >believe >that fixing this is chan_sip *will* cause confusion and errors to happen in >other >parts of Asterisk. There is a sort of quick fix - put SER between the SIP UA's and Asterisk. It will take a little work on your part (not Olle, but whomever is implementing this), or you can take the USD 100 and maybe that will buy you enough time from a consultant who understands Asterisk + SER to do a quick config for you. -Chris ___ 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] Transfers (sip or asterisk "#' base) broken in certain scenario
I've got an interesting scenario where transfers while getting an invite seem to break. Here is the scenario: You have a receptionist who has a 6 line phone (in this case, a polycom ip600 - also tested with a Cisco 7960) the receptionist has all six lines available for use (in the case of the cisco I tried registering all lines as one number as well as registering multiple lines and having the dialplan do roll-over). The receptionist receives a call and begins to transfer the call and in the middle of transferring the call, another call is received. This is what happens: If the receptionist is using the cisco/polycom soft button labelled transfer, the transfer goes through, however, the receptionist never knows another call was coming in. It went straight to the 'busy' priority (+101 in the case of a single registered extension or +101, +101, +101, +101... all the way through a roll-over dialplan straight to busy handling even though 5 of the 6 lines were available with no active calls) In the case of using # to effect a transfer, the receptionist hits pound and begins entering the phone number to transfer to and a call comes in. Immediately the receptionist is send to the 'i' extension while doing the transfer, and the new call is presented (rings and LCD screen shows information). In some cases depending on the timing of the new call (if it's received after pressing # but before entering an extension to transfer to) you can get the call back, place it on hold and take the new call, however if you are in the process of entering the number to transfer when the new call comes in, then the original call is immediately acted on. In other words, if I was typing 2004 and had entered 20 when the new call came in, asterisk grabs the 20 and tries to transfer the call to it. No matter what happens, the call is lost to the receptionist, unable to get it back, even if there is a valid 'i' handler. Is there anyone out there who has a busy enough system to have seen this as well? If so, how have you dealt with it? The only solution I can think of is to place all inbound calls into a queue, then pass them to the receptionist as the only agent (permanent agent) of the queue. Then set limits on the number of calls the receptionist is allowed to have incoming 1, outgoing 1 so the queue won't ring the receptionists phone unless there is no active sessions. Relevant info below Asterisk CVS-D2004.07.03.19.00.00-07/05/04-14:41:51 Polycom IP500 sip.ld version 1.2.0.0318 Cisco 7960 SIP image 6.0 sip.conf entries are simple and are not a factor in the problem - items that may be important from sip.conf are: canreinvite=no ; want asterisk in the media stream for features type=friend ; haven't tried this by creating a user and peer for each handset (yuck) extensions.conf entries are either one of the following (tested against both) exten => _,1,Dial(SIP/1000,30,t) exten => _,2,Voicemail(u1000) exten => _,3,Hangup exten => _,102,Voicemail(b1000) exten => _,103,Hangup exten => _,1,Dial(SIP/1000,30,t) exten => _,2,Voicemail(u1000) exten => _,3,Hangup exten => _,102,Dial(SIP/1001,30,t) exten => _,103,Voicemail(u1000) exten => _,104,Hangup exten => _,203,Dial(SIP/1002,30,t) ... exten => _,506,Dial(SIP/1005,30,t) exten => _,507,Voicemail(u1000) exten => _,508,Hangup exten => _,607,Voicemail(b1000) exten => _,608,Hangup -Chris ___ 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] pbx.c:1836 ast_pbx_run: Channel 'Zap/1-1' WARNING
On 12:39 AM 7/8/2004, robert brown wrote: CLI> -- Starting simple switch on 'Zap/1-1' Jul 7 18:42:24 WARNING[1192437440]: pbx.c:1836 ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' In english, ---Starting simple switch on 'Zap/1-1' means, 'Hey! The phone line you have plugged into the first X100P card is ringing! I'm going to answer it!' Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler means, 'I answered the line and in zapata.conf you told me to go to context 'default' to start handling a call on this line. So I went to context 'default' in extensions.conf and since I have no signalling to tell me what number was dialed, I'm going to look for extension 's' (which I look for when the number dialed is a null string), but I couldn't find extension 's' in context default, so I started looking for extension 'i' because thats what I do when I can't find the context/enten I'm looking for (i stands for invalid, and I will alwasy look for extension i when I can't find a matching extension for the call), but alas, extension I wasn't there either. At this point I would normally be creative and try very hard to figure out what you really wanted me to do versus what you told me to do, but alas, unfortunately 99.95% of the programming world thinks such behavior is a bug and not a feature, so I am prevented from reading your mind and must follow my rules, so I am allowed to spit out an error message, which of course I did. -- Hope that helps you understand the error messages you were seeing. What you need to do is create an context/extension in your extensions.conf file like such [default] exten => s,1,Dial(SIP/,20,t) ; see the wiki for what the 20 and t means (or do 'show application dial' from the cli) exten => i,1,Playback(invalid) exten => i,2,Hangup this should at least get you past the errors, the above entries aren't something you want to leave that way, because they don't handle all the possibilities you might run into (like not answering your phone if it rings for more than 20 seconds) -Chris
[Asterisk-Users] GR-303 configuration options?
Can anyone describe the asterisk implementation of this any better than the sample config files do? from zapata.conf ; Trunk groups are used for NFAS or GR-303 connections. ; ; Group: Defines a trunk group. ;group => ,[,...] ; ;trunkgroup is the numerical trunk group to create ;dchannelis the zap channel which will have the ;d-channel for the trunk. ;backup1 is an optional list of backup d-channels. ; ;trunkgroup => 1,24,48 ; ; Spanmap: Associates a span with a trunk group ;spanmap => ,[,] ; ;zapspan is the zap span number to associate ;trunkgroup is the trunkgroup (specified above) for the mapping ;logicalspan is the logical span number within the trunk group to use. ;if unspecified, no logical span number is used. ; ;spanmap => 1,1,1 ;spanmap => 2,1,2 ;spanmap => 3,1,3 ;spanmap => 4,1,4 the trunkgroup seems self explanatory. If I'm reading this correctly, trunkgroup=x,y,z creates a logical trunk group identified by x, with y as the signalling channel (which can contain up to 4 actual signalling channels?), and z being one or more signalling backup channels (also carrying as many as 4 signalling channels in one DS0 channel) spanmap => ,, so this seems to be an entry that will tie the logical trunkgroup to the physical T1 span, but what exactly is the purpose of the 'logicalspan' entry? next lower in the config we have the crv entries, which replace the well known channel entries used in other signalling methods. ; For GR-303, CRV's are created like channels except they must start ; with the trunk group followed by a colon, e.g.: ; ; crv => 1:1 ; crv => 2:1-2,5-8 so again, the first entry in a crv references the logical trunk group, correct? and the second and following entries define GR-303 CRV channels? so crv => 1:1 would reference GR-303 channel 1 on logical trunk 1? since GR-303 allows oversubscription of channels, does that mean that asterisk handles this correctly? If we had a single T1 interface going to a channel bank using GR-303, would asterisk understand: crv => 1:45 in this case if I dialed Zap span 1 channel 45, would asterisk understand this correctly in it's GR-303 implementation? Also on a final note, does anyone know what the user of an analog phone connected to an Adit 600, which uses GR-303 trunking hears when they go to use thier phone and all the trunk lines are active? -Chris ___ 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: Re: iax or sip
On 06:51 AM 7/7/2004, Randy Bush wrote: > >why enum? forcing humans to deal with telephone numbers >is analogous to asking them to use ip addresses instead >of domain names (which are bad enough, but that's another >story). do you want to send email to [EMAIL PROTECTED] >so why not 'dial' [EMAIL PROTECTED] or whatever? Because phones are Ubiquitous. They are everywhere, the proportion of people who have computers versus those who have phones worldwide, is probably still under 10%. By limiting the ability to map the current world wide legacy telephone address space into some method allowing you to reach systems via IP transport as well as TDM transport, you limit the reach of your voice communications network. Consider it backwards compatibility, sure, use [EMAIL PROTECTED] where you can, but I surely know if I told my parents to call me at [EMAIL PROTECTED], they'd suffer the human 'blue screen' event just trying to comprehend what I just said. For the majority of the people out there, saying 'call me' automatically causes a reaction 'what is your telephone NUMBER?' The telephone is such an integrated tool in most people's lives that you can't just switch it out for a keyboard and expect everyone to get it. Over time, perhaps, but then you have to consider compactness and usability. I have some small devices that allow me to use alphanumerics (palm's handwriting detection, or a blueberry's tiny keyboard), but they are bulky compared to my cell phone. And 'compact' keyboards leave a lot to be desired, IMHO. Of course there was the day when you could pick up a phone spin a little handle and then say Hi, Jenny, can you hook me up with Joan over in Jacksonville? Thanks! So maybe not people can make the change back to picking up thier 'phone' and just saying some unique identifier (Joan over in Jacksonville might not cut it now - but [EMAIL PROTECTED] just might) of who they want to call. This has gotten WAY off topic, so I'll just drop it here. -Chris ___ 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: is srv lookup being done when REGISTERing?
On 12:33 PM 7/6/2004, Karl Brose wrote: > >Registering with a provider with a register=yes in the [peer] section is >convenient for small installations when only one user is registering >with that domain since all the info usually is there already (username, >secret, host,...), but what if you have 20, 50, 100 users or accounts to >register. In that case, the old style register is convenient since it is >nice to have things presented in some kind of table (row, column) form, >and all the registrations can be maintained in an #include file or a >database. >Overloading the register syntax with more params naturally is a >drawback, but can actually be simplified by only using the [peer] name >instead of the proxyhost/domain and port and eliminate duplication that way. >Lastly, the existing register line syntax (without hostname portion) >could be moved into the [peer] section to associate it directly with the >peer and still allow multiple account registrations by duplicating the line. Note, he said: I would prefer to see a "register = yes" directive in the "type = peer" sections of both sip.conf and iax.conf, rather than the current method of using a separate "register => whatever" directive. The current method could be maintained for backward compatibility, of course. with the emphasis on 'backward compatibility' However, I would think that ast_data would resolve the issue of handling large numbers of sip entries. And in that case, putting the register=yes as an entry under a peer or friend category would seem to fit the design quite well. -Chris ___ 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] H323 channel
On 03:23 AM 7/6/2004, administrator tootai wrote: >Hello everybody, > >my * box is connected to gnugk with H323 channel. If I call from an H323 >EP to SIP EP (GS HandyTone or Xlite), when callee is picking up, audio >start but noisy (scratch) , then became ok for callee (SIP EP) but still >scratching on H323 EP. Now I stop/start asterisk, call from SIP to H323 >EP and it's ok. And from now, it's also ok when H323 EP call SIP one's! > >No need to say that H323<->H323 is working, as well as SIP<->SIP. >Running CVS version from yesterday. Used codecs are G711U & A, G723.1 >and G729. If I just use G711 it's the same. SIP EP has to call first >when * is started to make it work. Any hint? > >Also, H323 is still broken and working without FastStart. Is there a >workaround existing? Just to help troubleshooting, which h323 implementation for asterisk did you use? -Chris ___ 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] is srv lookup being done when REGISTERing?
On 03:16 AM 7/6/2004, Jasminko Mulahusic wrote: >it looks (to me) like asterisk is not doing an SRV lookup when >REGISTERing with another sip proxy. is that correct? > >what i am trying to achieve is to register [EMAIL PROTECTED] with a >proxy using > >register => jasko:secret:[EMAIL PROTECTED] > >my problem is that asterisk is doing a simple A RR lookup for the >domain telia.net which is pointing to a host that is NOT the proxy for >that domain (resulting in the REGISTER message ending up with the >wrong host). > have you set srvlookup = yes ;Enable DNS SRV lookups on calls in your sip.conf file? see: http://www.voip-info.org/tiki-index.php?page=Asterisk+config+sip.conf -Chris
Re: [Asterisk-Users] Calling an outside phone number as part of a hunt
On 07:51 PM 7/5/2004, Hall, Eric M. wrote: >I'm trying to see if this is even possible. > >When you dial ext 2000 I want it to ring my sip phone for 20 sec then >call my cell and let it ring for 10 sec if I do not pick up the call on >my cell I would like it to go back to * and leave a voice message for >me. Here is what I have so far in my extensions.conf > >Everything works except the call will not go back to * after the 10 sec >rule has expired. Asterisk assumes an analog interface (x100p, TDM4XXX) to have answered a dial command as soon as the dial command completed successfully. You may have some success with using callprogress=yes, however, this may very well cause other issues as can be seen if you do a google on callprogress. So in other words, your dialplan isn't working as you suspect because as soon as asterisk picks up the line on one of your x100p cards and issues the dtmf to the telco, it assumes the line has been answered (and technically it has, the telephone company's switch has answered your call request, and will now just pass audio back to you, whether it's a ring, busy tone, or someone's voice, and without call progress you won't know the difference (at least asterisk won't). The 10 second timeout never makes it into the decision, the line has been answered before 1 second passes. > >My hardware is 2 X100P card > > > >exten => 2000,1,Dial(SIP/2000,20) >exten => 2000,2,Dial(Zap/1/5551212,10) >exten => 2000,3,Voicemail(u2000) >exten => 2000,102,Voicemail(b2000) >exten => 2000,103,Hangup > >Any ideas? > >___ >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] iax or sip
Randy, On 01:11 PM 7/5/2004, Randy Bush wrote: >iax uses udp and traverses nats. neither of these seems useful to >me. i loathe nats, and udp is not well-behaved in the sense of >congestion avoidance. SIP and H323 use in NAT'd environments is problematic. IAX provides another solution to the issue for Asterisk administrators. If you don't use a NAT, then you can ignore this feature of IAX. > >trunking will save some bytes in flight iff one has four or more >streams moving between two pbxes. but who would want to have the >pbxes in the data stream anyway? reinvite rules, especially in a >geographically distributed use scenario. There are many reasons to have an Asterisk box in a stream: 1. Control a call, (maybe you want to do some ACL type filtering, maybe you want to keep track of usage, maybe you just to be in control...) 2. Provide features (access to PSTN, conference capability, music on hold, call parking, agents and queues. the list goes on and on) 3. Endpoints (User Agents) MAY not be able to send data streams to each other directly (firewalls or nats in the middle) And depending upon your view of things (your view might be different than the view of the IT/communications administrator of a large company), using IAX in a geographically distributed use scenario might very well be exactly what you want (use over an encrypted vpn link, etc.) > >now, i could see a network of iaxen if there was some way to >negotiate call routing with costs etc. but trip looks a bit ugly >and kinda far away. and it certainly is not part of current play. > >what am i missing here? Nothing, really. If you don't need IAX, and don't particularly like any of the features of it, then don't use it. At this time, if you don't have NAT's to deal with (or you've already convinced your signalling and media protocols to deal with them), and you don't have a need to force the streams and signalling through your asterisk systems, then 'noload => chan_iax.so' to your hearts content! -Chris ___ 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] Zaptel dacs / dacs
On 06:41 PM 7/2/2004, Andrew Kohlsmith wrote: >On Friday 02 July 2004 21:25, Chris A. Icide wrote: >> I still didn't get an answer to the original question of will DACS do this? > >To my knowlege, DACS is simply a cross-connect. > >dacs=1-24:48 > >When you pick up line 1, * automagically bridges it to line 48. Line 2 to 49, >etc. > >The RBS variant would handle the signalling bits as well. > >That's all just a guess. :-) > >-A. thanks! So if I read this correctly. the first argument in a dacs= line is the channel list (does it have to be a range, or can it be like other channel arguments), and the second argument is the initial channel to begin cross connecting to. So. in the case of a clear channel T1 (em_w or em), could you do something like this? dacs=1-12,25-36:48 (take the first 12 channels of span 1 and the first 12 channels of span 2 and cross connect them all to span 3?) I'm getting way off topic from my original question, but this is interesting as well for some goofy reason. -Chris ___ 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
