Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
As for "doing something better", I would hope to see two (B things happening ... (B (B 1) you begin to use queue_app for your call centre (B requirements and if you need any assistance, you ask about (B it here (B (B 2) having experienced Asterisk's superior queue management (B system, you tell us that you were wrong and consider (B withdrawing your bounty (B (BSorry to say, I can't do both. (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Olle E. Johansson wrote: (B (BAgreed. I wasn't clear enough. Asterisk have users in (Bmany (Bplaces, but no centralized view of a "user". (B (BAgreed. (B (BI haven't said that parallel forking is my recommended (Bway of doing this. (B (BFair enough. (B (BI've stated several times that it doesn't (Breally comply with the architecture of Asterisk, unless (Bdone in an Asterisk-architecture friendly way :-) (BAnd adding it will not solve the user dilemma, as you (Bsay. (B (BThe trouble is that some folks, the ones who only got a (Bhammer, ignored that part of your comments, possibly even (Bon purpose. (B (BI am concerned about the danger of introducing features (Binto Asterisk which water down its philosophy. If features (Bare to be added it should be for the right reasons, not (Bfor the wrong ones. (B (B (B Not if you give them a means to provision it (Bthemselves. (B This can be as easy as an extension that asks for a PIN (B number and then executes a shell script. (B (BRight. Please send samples of this so we can add it to (Bthe Wiki! (B (BThe account provisioning scripts I have done so far are (Bcalled either from some GUI interface or they are executed (Bon the command line. I haven't used them for self (Bprovisioning yet but I know that it would be fairly easy (Bto "misappropriate" them for that purpose. (B (BWhen I get the time, I will add a bit of "end user glue" (Band post an example. (B (BThats where we should go. [peer]s and [user]s being (Bdevices (B(IMEI) and a new user architecture representing the IMSI. (BWe have accountcode now. It's not enough. (B (BIt may well be worth while implementing (parts of) the GSM (BIMSI specification into Asterisk. Combined with support (Bfor SIM card readers this would make it possible to use (Bstandard SIM cards to sign on to an Asterisk driven (Bnetwork. (B (BWithout the SIM card the devices could still log on to (BAsterisk, but they would be placed in a restricted (Bcontext. Then, when a SIM card is inserted and (Bauthenticated, the account will be attached to the device (Band it will then be placed in the context associated with (Bthat user. (B (BOf course there could be alternative methods for user (Bauthentication in addition to SIM cards. Still I think it (Bwould make sense to use the GSM specification *where (Bapplicable* instead of creating yet another format. (B (BThe authentication algorithm would have to be replaced (Bwith RSA or something though, because the GSM A5 algorithm (Bis secret and available only to licensed GSM operators who (Bjoined the GSM Association. Rather silly that those folks (Bstill seem to believe in secret authentication algorithms, (Beh?! (B (BBut apart from that pesky A5 thing, the way GSM (Bauthenticates users is pretty well designed. You have got (Bthe handset which can access the network even without a (BSIM but then only emergency services are authorised. If a (BSIM card is present, a request to authenticate will be (Bsent to the user's network's HLR (Home Location Registry) (Bfrom where the SIM card will then be authenticated. Once (Bthis was successful, a temporary entry is created in the (BVLR (Visited Location Registry) that serves the spot in (Bwhich the user happens to be. This entry then determines (Bwhich services the user is authorised to use, voice, (Bmessaging, data, value added etc. (B (BIf you are interested to look into this a bit further and (Bif only for inspiration, but haven't got access to the (Bdocumentation, I have the entire ETSI GSM standard (Bdocumentation on CD. So, if you want to take a peek, (Bcontact me by email at benjamin (at) sunrise (dash) tel (B(dot) com. (B (Brgds (Bbenjk (B (B__ (BDo You Yahoo!? (Bhttp://bb.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
On Wed, 2004-07-14 at 07:55, Sunrise Ltd wrote: Thats where we should go. [peer]s and [user]s being devices (IMEI) and a new user architecture representing the IMSI. We have accountcode now. It's not enough. It may well be worth while implementing (parts of) the GSM IMSI specification into Asterisk. Combined with support for SIM card readers this would make it possible to use standard SIM cards to sign on to an Asterisk driven network. Without the SIM card the devices could still log on to Asterisk, but they would be placed in a restricted context. Then, when a SIM card is inserted and authenticated, the account will be attached to the device and it will then be placed in the context associated with that user. It should be enough to implement a seperate user layer for accounting and authentification purposes based on password and/or public/private key authentification. On the client side (hardphones / softphone on pc with smartcard-reader, like the ones build in notebooks today) the implementation afterwards would be an easy task. Asterisk doesn't need to know anything about that. All it would need is the authetification keys, that could be placed on the smart/sim-type card. These readers are available in notebooks and should be quite inexpensive for other equipment manufacturers. All it would need is to agree on a standard (doesn't have to involve asterisk) on how the keys would be stored on the smartcards. I quite like the approach, because you could give manufacturers and implementors the possibility to choose whatever solution they like: smartcards, download keys to phone via tftp, put them on a usbkey etc. All you should define is the format for the keys and the passcode format (typically digits, like the 4 digit pin for simcards). But asterisk only needs to handle user authenfication based on public/private keys. Everything else would be a seperate development. Kind regards, Martin LIst-Petersen ___ 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: SIP simultaneous registry possible workaround (was Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry)
Andrew Kohlsmith wrote:
(B
(BI wasn't talking about bandwidth but rather lengthy
(BDial() commands...
(B
(Bexten = s,1,Dial(SIP/someuserSIP/someuserSIP ..
(B
(Bkind of thing... seems awfully unwieldy.
(B
(BThat's why you would stick the members into a global
(Bvariable
(B
(B[globals]
(B
(BDIYCALLGROUP = SIP/111SIP/112SIP113 etc.
(B
(Bthen dial using Dial(${DIYCALLGROUP},...)
(B
(BAlso, you can use the callgroup feature in sip.conf
(B
(B[111]
(B...
(Bcallgroup=1
(Bcallerid="Member 1"12345
(B
(B[112]
(B...
(Bcallgroup=1
(Bcallerid="Member 2"12345
(B
(B[113]
(B...
(Bcallgroup=1
(Bcallerid="Member 3"12345
(B
(Bthen in your dialplan
(B
(Bexten = 12345,1,Dial(SIP/111) ; dialling one member
(Brings them all
(B
(Bthis should call the entire call group. There have been
(Bsome issues with callgroups and SIP some while ago but
(Bthey may have been fixed.
(B
(BIn the event that they haven't been fixed, I suggest once
(Bagain that the bounty would be better spent on fixing
(Bwhatever issues there may still be with callgroups in SIP.
(B
(Brgds
(Bbenjk
(B
(B__
(BDo You Yahoo!?
(Bhttp://bb.yahoo.co.jp/
(B
(B___
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Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Kannaiyan Natesan wrote: (B (BI hope you clearly understand that everyone here (B**KNOWS** (Bto use alternative software such as SER, what is needed (Bhere is (Bthe same facility in asterisk. (B (BYou have not shown us ANY example yet for which this (Bfacility is *NEEDED*. (B (BYou have only shown us examples for which the facility MAY (Bbe used, all of which have been shown to have OTHER, (Bbetter solutions. (B (BFor call centres you use call queues, for taking workload (Boff admins you use self provisioning, for call groups you (Buse callgroup=1 or a dial string with multiple (Bdestinations, for multi line SIP phones you use multiple (Bextensions. (B (BNone of those problems warrant the use of parallel (Bforking. (B (BYour problem seems to be that you want a facility for its (Bown sake, not because you really need it. That, however is (Bnot good enough a reason to add something to Asterisk. (B (Brgds (Bbenjk (B (B__ (BDo You Yahoo!? (Bhttp://bb.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Duane wrote: (B (BWe're running SER and Asterisk on the same system with (BLike2Fone.com and we just stuck Asterisk on a different (Bport then redirect calls as needed, although I doubt it (Bwould (Bbe as difficult as your making out, if you stick SER on (Ban (Balternative port and then just use that to connect your (Bclients to problem solved, in effect the opposite to what (Bwe wanted to achieve... (B (BInteresting. I assume by "redirect calls as needed" you (Bmean passing calls between Asterisk and SER. (B (BIt is unclear to me how you achieve that. (B (BIf Asterisk is directed to speak SIP on port 5061 and SER (Bremains on port 5060, then how do you get Asterisk to talk (Bto SER and vice versa? (B (BWould you care to share this with us? (B (Brgds (Bbenjk (B (B__ (BDo You Yahoo!? (Bhttp://bb.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Kannaiyan Natesan wrote: (B (B I hope you clearly understand that everyone here (B **KNOWS** (B to use alternative software such as SER, what is needed (B here is (B the same facility in asterisk. (B (B You have not shown us ANY example yet for which this (B facility is *NEEDED*. (B (B (BHave you used 5 welcome service in fwd? (BIf not try that. You are invited to join as a volunteer to provide support (Band talk to new people on fwd. (B (BAs I explained to you before we use it for our customer service in call (Bcenter and implemented in many call centres which really makes $. (B (BCan you help me to know how that be achieved with * alone. (B (B-Kannaiyan. (B (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous - Implementation
Based upon the analysis I think we need to modify two things, (B (B1. chan_sip.c (Registrar) (B2. app_dial.c (Dial Command for simultaneous dialling, as of now it (Bsupports simultaneous dialling too) (B (BThe registrar of SIP need to collect the array of registrants and the Dial (Bcommand need to take care of dialling to all possible registrants which I (Bthink should be easier to implement. Anybody thinks will there be any other (Bproblems in handling the same? (B (B-Kannaiyan. (B (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
You have not shown us ANY example yet for which this facility is *NEEDED*. Well, I have users that get an account on my PBX. I really don't care how many phones they want to use, hardware phones on their desktop or soft phones on their laptop while travelling. It's still a user with one account. When the PBX dials them, all their phones should ring. Asterisk doesn't really bother with *users*, it has a device-centric view of life, universe and propably everything. With Asterisk, the user has to call me each time he wants a new device connected and I have to reconfigure his setup. If I had support for multiple registrations on one [peer] account, the [peer] would become a user account instead of a device. And the user could add as many devices as he wanted (up to a defined limit) without bothering the administrator. I guess that's why a lot of people ask for this function. However, since Asterisk doesn't really bother with a user concept, we really have to teach Asterisk about users. And user groups. Life is much more than hardware, little Asterisk :-) I've been discussing this many times, and so has many other people. I think we need an elegant way of defining users to asterisk so we connect peers, users, agents and mailboxes to a *user* with one set of credentials. If you look into your Asterisk configuration, you will find that there are users and credentials for logging in everywhere. It's not easy to maintain at all. After a lot of discussions on the IRC, I'm convinced that we at some point in time have to add ast_auth - a common infrastructure for handling users and authentication. This is a good topic for the Asterisk Developer's Day at Astricon. Let's bring it up on the agenda - A new user and authentication structure for Asterisk. YALMIATASQ - Yet Another Long Mail in answer to a short question. Hint: I have a new idea for a solution on multiple reg's. Raise the bounty and I might give it a try. ;-) /Olle ___ 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: SIP simultaneous registry possible workaround (was Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry)
Also, you can use the callgroup feature in sip.conf [111] ... callgroup=1 callerid=Member 112345 [112] ... callgroup=1 callerid=Member 212345 [113] ... callgroup=1 callerid=Member 312345 then in your dialplan exten = 12345,1,Dial(SIP/111) ; dialling one member rings them all Seems like a s a weird setup. I can't call them individual that way, can't I? ___ 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
Ok I'll kick in $25 (just based on your email alone). Is there a formal system for bounty registration? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Tuesday, 13 July 2004 5:54 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous You have not shown us ANY example yet for which this facility is *NEEDED*. Well, I have users that get an account on my PBX. I really don't care how many phones they want to use, hardware phones on their desktop or soft phones on their laptop while travelling. It's still a user with one account. When the PBX dials them, all their phones should ring. Asterisk doesn't really bother with *users*, it has a device-centric view of life, universe and propably everything. With Asterisk, the user has to call me each time he wants a new device connected and I have to reconfigure his setup. If I had support for multiple registrations on one [peer] account, the [peer] would become a user account instead of a device. And the user could add as many devices as he wanted (up to a defined limit) without bothering the administrator. I guess that's why a lot of people ask for this function. However, since Asterisk doesn't really bother with a user concept, we really have to teach Asterisk about users. And user groups. Life is much more than hardware, little Asterisk :-) I've been discussing this many times, and so has many other people. I think we need an elegant way of defining users to asterisk so we connect peers, users, agents and mailboxes to a *user* with one set of credentials. If you look into your Asterisk configuration, you will find that there are users and credentials for logging in everywhere. It's not easy to maintain at all. After a lot of discussions on the IRC, I'm convinced that we at some point in time have to add ast_auth - a common infrastructure for handling users and authentication. This is a good topic for the Asterisk Developer's Day at Astricon. Let's bring it up on the agenda - A new user and authentication structure for Asterisk. YALMIATASQ - Yet Another Long Mail in answer to a short question. Hint: I have a new idea for a solution on multiple reg's. Raise the bounty and I might give it a try. ;-) /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Hello,
From: Sunrise Ltd [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Date: Tue, 13 Jul 2004 16:31:58 +0900 (JST)
snip
If Asterisk is directed to speak SIP on port 5061 and SER
remains on port 5060, then how do you get Asterisk to talk
to SER and vice versa?
Would you care to share this with us?
It is something like this:
Asterisk extensions.conf:
[globals]
SERADDRESS=XXX.XXX.XXX.XXX:5060
[context]
exten = yourexten,1,Dial(SIP/[EMAIL PROTECTED],20,r)
In ser.cfg:
if (method == INVITE) {
if (uri =~ sip:[EMAIL PROTECTED]){
log(1, Forwarding to Asterisk\n);
rewritehostportt(XXX.XXX.XXX.XXX:5061);
t_relay();
break;
}
}
rgds
benjk
Regards, Girish
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under the sun.
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Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
I can see the point of the discussion somewhere, but let's take it the other way around (comments though mail): On Tue, 2004-07-13 at 08:53, Olle E. Johansson wrote: You have not shown us ANY example yet for which this facility is *NEEDED*. Well, I have users that get an account on my PBX. I really don't care how many phones they want to use, hardware phones on their desktop or soft phones on their laptop while travelling. It's still a user with one account. When the PBX dials them, all their phones should ring. Now .. the problem is, that every hardware phone, every softphone etc. actually might need a different configuration, some IAX, some SIP, some one codec, some other codecs (now that we are talk asterisk). It will get quite problematic to get all solutions under one account without breaking one or the other. Asterisk doesn't really bother with *users*, it has a device-centric view of life, universe and propably everything. With Asterisk, the user has to call me each time he wants a new device connected and I have to reconfigure his setup. Or you provide him with a webinterface, where he has one username and one password. He manages the accounts and settings for each friend,peer,user there and you wouldn't have the work. Kind regards, Martin List-Petersen ___ 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 13/07/2004 at 11:48 Martin List-Petersen wrote: I can see the point of the discussion somewhere, but let's take it the other way around (comments though mail): On Tue, 2004-07-13 at 08:53, Olle E. Johansson wrote: You have not shown us ANY example yet for which this facility is *NEEDED*. Well, I have users that get an account on my PBX. I really don't care how many phones they want to use, hardware phones on their desktop or soft phones on their laptop while travelling. It's still a user with one account. When the PBX dials them, all their phones should ring. Now .. the problem is, that every hardware phone, every softphone etc. actually might need a different configuration, some IAX, some SIP, some one codec, some other codecs (now that we are talk asterisk). It will get quite problematic to get all solutions under one account without breaking one or the other. Yes, this is a problem I''d forsee... but ignoring that for one moment :P Imagine that asterisk accepts multiple registrations for a single entry in sip.conf ([myphone]) simply adding each to an internal variable: The first phone registers: WHO_I_DIAL = sip:[EMAIL PROTECTED] then joe comes along and also registers a line on his phone WHO_I_DIAL = sip/[EMAIL PROTECTED]sip/[EMAIL PROTECTED] now when I execute a dial, asterisk internally replaces the occurrence of myphone with the WHO_I_DIAL variable: eg: Dial(SIP/myphone,120) becomes (internally) Dial(WHO_I_DIAL,120) In essence DIAL sees nothing different at all and doesn;t need to be changed because the internal reference SIP/myphone actually = the content of WHO_I_DIAL So what we affectively achieve is: Dial(sip/[EMAIL PROTECTED]sip/[EMAIL PROTECTED],120) Which is what people have been saying everyone should do... but this process becomes automatic, which is a feature that people want. I'm pretty sure you'd do this with an array rather than a string, but I think it explains the theory behind it all. Of course I've ignored the issue with different configs required for different SIP devices (eg DTMFMODE=), but that artistic license ;) I may have explained it badly, so let me know Andy ___ 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: SIP simultaneous registry possible workaround (was Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry)
On Tuesday 13 July 2004 03:07, Sunrise Ltd wrote: (B exten = s,1,Dial(SIP/someuserSIP/someuserSIP .. (B (B That's why you would stick the members into a global (B variable (B (BYou global variable is still unwieldy. All you did was move the problem. (B (B Also, you can use the callgroup feature in sip.conf (B (B [111] (B ... (B callgroup=1 (B callerid="Member 1"12345 (B (B [112] (B ... (B callgroup=1 (B callerid="Member 2"12345 (B (BNow *that* is what I was looking for -- so it is possible to group SIP peers (Blike you can Zap channels :-) (B (BThank you. I don't use SIP unless I have to, but I was hoping Asterisk could (Bhandle SIP grouping to help this particular fellow. :-) (B (B-A. (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SIP simultaneous registry possible workaround (was Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry)
On Tue, 2004-07-13 at 03:54, Holger Schurig wrote: Also, you can use the callgroup feature in sip.conf [111] ... callgroup=1 callerid=Member 112345 [112] ... callgroup=1 callerid=Member 212345 [113] ... callgroup=1 callerid=Member 312345 then in your dialplan exten = 12345,1,Dial(SIP/111) ; dialling one member rings them all Seems like a s a weird setup. I can't call them individual that way, can't I? Sure you can. It just depends on you set up the extensions. For example, you could have a section in extensions.conf for the individual SIP extensions, and another section where all the SIP extensions should be rung (e.g. an IVR menu or something). HTH, Kanwar Systems Aligned Inc. www.systemsaligned.com ___ 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
I have Asterisk running against 5060 and 5061 servers: [general] Port=5060 register = 8775551212:[EMAIL PROTECTED] register = 18005551212:[EMAIL PROTECTED]:5061 And then [sip-vonage] secret=secret username=18005551212 host=sphone.vopr.vonage.net port=5061 type=peer nat=yes canreinvite=no dtmfmode=rfc2833 fromuser=18005551212 context=incoming fromdomain=sphone.vopr.vonage.net [sip-bv1] secret=secret username=8775551212 host=sip.broadvoice.com Port=5060 type=peer nat=yes canreinvite=no dtmfmode=inband fromuser=8775551212 callerid=8775551212 context=incoming fromdomain=sip.broadvoice.com Runs without a problem. It's conceivable that you can run SER on port 5061 and tell Asterisk to register/dial using localhost:5061. -Original Message- From: Sunrise Ltd [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 13, 2004 2:32 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous If Asterisk is directed to speak SIP on port 5061 and SER remains on port 5060, then how do you get Asterisk to talk to SER and vice versa? ___ 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
Kannaiyan Natesan wrote: (B (BHave you used 5 welcome service in fwd? (BIf not try that. You are invited to join as a volunteer (Bto provide support and talk to new people on fwd. (B (BAsterisk can do that much better than SER because it has (Bgot a queue management system built-in. (B (BAs I explained to you before we use it for our customer (Bservice in call (Bcenter and implemented in many call centres which really (Bmakes $. (B (BNo matter how many times you claim that parallel forking (Bis the right solution to implement a call centre, it (Bdoesn't change the fact that you are still wrong. Call (Bcentres use queue management systems. (B (BCan you help me to know how that be achieved with * (Balone. (B (BSure. Read /etc/asterisk/queues.conf and (B/etc/asterisk/agents.conf. Everything you need is there. (B (Brgds (Bbenjk (B (B__ (BDo You Yahoo!? (Bhttp://bb.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Girish Gopinath wrote:
(B
(B[globals]
(BSERADDRESS=XXX.XXX.XXX.XXX:5060
(B
(B[context]
(Bexten =
(Byourexten,1,Dial(SIP/[EMAIL PROTECTED],20,r)
(B
(BIn ser.cfg:
(B
(Bif (method == "INVITE") {
(Bif (uri =~ "sip:[EMAIL PROTECTED]"){
(Blog(1, "Forwarding to Asterisk?n");
(Brewritehostportt("XXX.XXX.XXX.XXX:5061");
(Bt_relay();
(Bbreak;
(B}
(B}
(B
(BOK, that looks kind of promising.
(B
(BIn other words you're saying to do away with
(Bauthentication for calls between Asterisk and SER since
(Bboth run on the same box?
(B
(BI haven't looked at it this way, but I guess it makes
(Bsense.
(B
(B
(BUnfortunately though, this doesn't seem to also be a
(Bsolution for what I would like to do, which is run X-Lite
(Band Asterisk on the same box, my Powerbook G4, so I can
(Buse it as an IAX phone when travelling.
(B
(Bthanks anyway for the interesting insight.
(B
(Brgds
(Bbenjk
(B
(B__
(BDo You Yahoo!?
(Bhttp://bb.yahoo.co.jp/
(B
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Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Olle E. Johansson wrote: (B (BWell, I have users that get an account on my PBX. (B (BI really don't care how many phones they want to use, (Bhardware phones on their desktop or soft phones on their (Blaptop while travelling. It's still a user with one (Baccount. (B (BTwo words: self provisioning. (B (BAsterisk doesn't really bother with *users*, it has a (Bdevice-centric view of life, universe and propably (Beverything. (B (BThat's only partly correct. The queue management system (Bhas a user view, called agents, and agents can (Bauthenticate themselves independently from the device they (Bare using and then attach themselves to call queues (Bmanaged by Asterisk. (B (BHowever, for anything unrelated to queue management, you (Bare correct in that Asterisk doesn't apply this concept (Bthere. (B (BI may even agree with you that it would be worthwhile to (Bapply this user concept to other areas outside of queue (Bmanagement. (B (BStill I disagree that parallel forking is the way to do (Bthis. I even disagree that it would introduce a user view. (BInstead it would water down the device view. So you go (Bfrom an system with a very clean device view but without a (Buniversally applied user view to a system with a messy (Bdevice view and still no user view. (B (BWith Asterisk, the user has to call me each time he wants (Ba new (Bdevice connected and I have to reconfigure his setup. (B (BNot if you give them a means to provision it themselves. (BThis can be as easy as an extension that asks for a PIN (Bnumber and then executes a shell script. (B (BIf I had support for multiple registrations on one [peer] (Baccount, the (B[peer] would become a user account instead of a device (B (BWell, that's an opinion. (B (BI'd rather prefer to have a user layer on top and in (Baddition to a device layer instead of trading one for the (Bother. (B (BThis is how GSM works BTW, you have the IMEI which (Bidentifies the device and the IMSI which identifies the (Bsubscriber. A subscriber may be using the same IMSI on (Bdifferent devices, but the IMEI for each device is unique. (BThe IMEI lives in the device. The IMSI lives on the SIM (Bcard. (B (BThe customer care and billing system is mostly concerned (Babout the IMSI when dealing with a subscriber, but some (Blow level network elements need the IMEI to do their job. (BThe conclusion here is that there is a use for both, (Bdevice and user views. (B (BI think it would be wise to take a lesson from GSM in (Brespect of having both a device and a user view, and not (Bjust trade one for the other. (B (BAnd the user could add as many devices as he wanted (B(up to a defined limit) without bothering the (Badministrator. (B (BEarly mobile phone systems made the same mistake you are (Bproposing here. They too said "device = user" and it (Bopened the door to plenty of problems, from inconvenience (Bwhen changing a device to fraud. (B (BThe introduction of GSM introduced a user layer on top of (Bthe device layer and you got both convenience (ie move the (BSIM card to another phone and secondary SIM for a family (Bmember etc etc) and better security (no device cloning, (Bstolen equipment can be blocked through EIR network (Belements etc etc). (B (B (BI guess that's why a lot of people ask for this function. (B (BNo, people asking for this because "If all you have is a (Bhammer, everything looks like a nail." (B (BHowever, since Asterisk doesn't really bother with a user (Bconcept, (Bwe really have to teach Asterisk about users. And user (Bgroups. (B (BI agree with that in priniciple, but parallel forking (Bdoesn't do that. (B (BI've been discussing this many times, and so has many (Bother (Bpeople. I think we need an elegant way of defining users (Bto (Basterisk so we connect peers, users, agents and mailboxes (Bto a *user* with one set of credentials. If you look into (Byour (BAsterisk configuration, you will find that there are (Busers and (Bcredentials for logging in everywhere. It's not easy to (Bmaintain at all. (B (BAgreed again, but still fail to see how parallel forking (Bwould contribute anything to what you ask for here. (B (BHint: I have a new idea for a solution on multiple reg's. (BRaise the bounty and I might give it a try. ;-) (B (BIf you absolutely have to mess with it, just make sure it (Bcan be disabled by the rest of us who don't want to deal (Bwith any potential problems it may introduce. (B (Brgds (Bbenjk (B (B__ (BDo You Yahoo!? (Bhttp://bb.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
It goes somewhere. (BI will close it from my side. (B (BThanks for your info. (BLet us do something better. (B (B-Kannaiyan. (B (B (B- Original Message - (BFrom: "Sunrise Ltd" [EMAIL PROTECTED] (BTo: [EMAIL PROTECTED] (BSent: Tuesday, July 13, 2004 5:20 PM (BSubject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous (B (B (B Kannaiyan Natesan wrote: (B (B Have you used 5 welcome service in fwd? (B If not try that. You are invited to join as a volunteer (B to provide support and talk to new people on fwd. (B (B Asterisk can do that much better than SER because it has (B got a queue management system built-in. (B (B As I explained to you before we use it for our customer (B service in call (B center and implemented in many call centres which really (B makes $. (B (B No matter how many times you claim that parallel forking (B is the right solution to implement a call centre, it (B doesn't change the fact that you are still wrong. Call (B centres use queue management systems. (B (B Can you help me to know how that be achieved with * (B alone. (B (B Sure. Read /etc/asterisk/queues.conf and (B /etc/asterisk/agents.conf. Everything you need is there. (B (B rgds (B benjk (B (B __ (B Do You Yahoo!? (B http://bb.yahoo.co.jp/ (B (B ___ (B Asterisk-Users mailing list (B [EMAIL PROTECTED] (B http://lists.digium.com/mailman/listinfo/asterisk-users (B To UNSUBSCRIBE or update options visit: (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (B (B (B (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
As I explained to you before we use it for our customer service in call (B center and implemented in many call centres which really makes $. (B (BAll this stuff to do a simple call queue system??? Man, You need to read (Bwiki. Read agents.conf and queue.conf before to begin a war here... (BAll you need to do can be achieved with app_queue. (B (BRegards, (B (BGus (B (B (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SIP simultaneous registry possible workaround (was Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry)
Hi! That sound unwieldy as the number of simultaneous ringers increases... Can SIP peers be grouped like Zap channels? Yes - use a queue. http://www.voip-info.org/wiki-Asterisk+call+queues Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SIP simultaneous registry possible workaround (was Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry)
On Jul 13, 2004, at 5:13 AM, Andrew Kohlsmith wrote: On Tuesday 13 July 2004 03:07, Sunrise Ltd wrote: exten = s,1,Dial(SIP/someuserSIP/someuserSIP .. That's why you would stick the members into a global variable You global variable is still unwieldy. All you did was move the problem. It scales a bit further then DIAL(SIP/...SIP/...). If you want to go much further, you'll probably want to move towards auto-generating your Asterisk configs from a database or simple specification language. That'll scale a lot further, and be more reliable in the face of editing mistakes. Scott ___ 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
Sunrise Ltd wrote: Olle E. Johansson wrote: Well, I have users that get an account on my PBX. I really don't care how many phones they want to use, hardware phones on their desktop or soft phones on their laptop while travelling. It's still a user with one account. Two words: self provisioning. Right. Asterisk doesn't really bother with *users*, it has a device-centric view of life, universe and propably everything. That's only partly correct. The queue management system has a user view, called agents, and agents can authenticate themselves independently from the device they are using and then attach themselves to call queues managed by Asterisk. However, for anything unrelated to queue management, you are correct in that Asterisk doesn't apply this concept there. Agreed. I wasn't clear enough. Asterisk have users in many places, but no centralized view of a user. Still I disagree that parallel forking is the way to do this. I even disagree that it would introduce a user view. Instead it would water down the device view. So you go from an system with a very clean device view but without a universally applied user view to a system with a messy device view and still no user view. I haven't said that parallel forking is my recommended way of doing this. I've stated several times that it doesn't really comply with the architecture of Asterisk, unless done in an Asterisk-architecture friendly way :-) And adding it will not solve the user dilemma, as you say. I hope I did not say it that way. That was why I brought up that we regardless of parallell forking will have to look at the user and authentication architecture of Asterisk. Not if you give them a means to provision it themselves. This can be as easy as an extension that asks for a PIN number and then executes a shell script. Right. Please send samples of this so we can add it to the Wiki! If I had support for multiple registrations on one [peer] account, the [peer] would become a user account instead of a device Well, that's an opinion. I'd rather prefer to have a user layer on top and in addition to a device layer instead of trading one for the other. Right, me too. This is how GSM works BTW, you have the IMEI which identifies the device and the IMSI which identifies the subscriber. A subscriber may be using the same IMSI on different devices, but the IMEI for each device is unique. The IMEI lives in the device. The IMSI lives on the SIM card. Thats where we should go. [peer]s and [user]s being devices (IMEI) and a new user architecture representing the IMSI. We have accountcode now. It's not enough. I guess that's why a lot of people ask for this function. No, people asking for this because If all you have is a hammer, everything looks like a nail. :-) /O ___ 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: SIP simultaneous registry possible workaround (was Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry)
I wasn't talking about bandwidth but rather lengthy Dial() commands...
Variables can shorten the parameter greatly
GROUP1=SIP/1SIP/2SIP/3SIP/4
GROUP2=SIP/5SIP/6
dial(${GROUP1}${GROUP2})
I have no idea on the theoretical limit for a dial string, but I suspect it
should be quite long
Youness
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Re: SIP simultaneous registry possible workaround (was Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry)
On Tuesday 13 July 2004 17:16, Youness El Andaloussi wrote:
Variables can shorten the parameter greatly
GROUP1=SIP/1SIP/2SIP/3SIP/4
GROUP2=SIP/5SIP/6
dial(${GROUP1}${GROUP2})
You're missing the point. This is STILL unwieldy. I'd have to put together a
list of variables, ensuring each entry is typed correctly not only in
sip.conf but also in extensions.conf, and then group all these strings
together.
We have group=x functionality in zapata.conf, does the same exist in sip.conf?
What about iax and skinny and mgcp? It should, it simplifies this all
incredibly. Add a user, add them to the group and your dialplan doesn't have
to change every time you add a new peer to the list.
I dunno, just my $0.02.
-A.
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Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Jay Milk wrote: (B (B[general] (BPort=5060 (B (Bregister = [EMAIL PROTECTED]:5061 (B (B[sip-vonage] (B... (Bhost=sphone.vopr.vonage.net (Bport=5061 (B (Bvery interesting, thanks. (B (BStill need to get my head around this and see how it may (Bbe used for running X-Lite along Asterisk on the same (Bnotebook in order to dial out from X-Lite through Asterisk (Bgoing from SIP to IAX in the process. (B (Bthanks again (Bbenjk (B (B__ (BDo You Yahoo!? (Bhttp://bb.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
(BKannaiyan Natesan wrote: (B (BSubject: Re: [Asterisk-Users] New Asterisk bounty: SIP (Bsimultaneous (B (BI will close it from my side. (B (BThanks for your info. (BLet us do something better. (B (BDid you check out /etc/asterisk/queues.conf and (B/etc/asterisk/agents.conf ??? (B (BOr do you mean to say you won't be satisfied with anything (Bother than your pet feature, no matter how unsuited it may (Bbe and regardless of better suited solutions Asterisk has (Bto offer? (B (BAs for "doing something better", I would hope to see two (Bthings happening ... (B (B1) you begin to use queue_app for your call centre (Brequirements and if you need any assistance, you ask about (Bit here (B (B2) having experienced Asterisk's superior queue management (Bsystem, you tell us that you were wrong and consider (Bwithdrawing your bounty (B (Bthx (Brgds (Bbenjk (B (B__ (BDo You Yahoo!? (Bhttp://bb.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
So isn't this the problem * has? The first client registers as the address of record, then the second client comes in with the same registration and becomes the address of record? I think you are making this look more complicated than it actually is. We do this with our SER Network all the time. Its called parallel forking. For example, our subscribers can have 2 or more Sipuras with the same number and registration info. They have one Sipura at the office and another at home. When a call is destined for that sub, SER will lookup the location database to see where it should send the INVITE. If it sees 2 or more locations then it sends multiple INVITES, ie.. Parallel Fork. The first INVITE to answer will be the one that establishes the RTP Session and all the others will receive a CANCEL. Its quite simple and works perfectly. -- Andres Network Admin http://www.telesip.net ___ 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
Well Andres is right but there are numerous problems with quite a few SIP clients that do NOT follow the the SIP RFC correctly. There is a problem with dialog creation in a number of SIP products out there. SIP dialog creation is the critical part of the spec that supports parallel forking - so be careful. Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andres Sent: 12 July 2004 08:54 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous So isn't this the problem * has? The first client registers as the address of record, then the second client comes in with the same registration and becomes the address of record? I think you are making this look more complicated than it actually is. We do this with our SER Network all the time. Its called parallel forking. For example, our subscribers can have 2 or more Sipuras with the same number and registration info. They have one Sipura at the office and another at home. When a call is destined for that sub, SER will lookup the location database to see where it should send the INVITE. If it sees 2 or more locations then it sends multiple INVITES, ie.. Parallel Fork. The first INVITE to answer will be the one that establishes the RTP Session and all the others will receive a CANCEL. Its quite simple and works perfectly. -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Paul Mahler wrote: Well, this is certainly getting exciting. Yes, it is. Sorry for coming in late to this debate... Andy, I took your advice and re-read the RFP. It's actually RFC, not RFP. (teasing :-) So, gentlemen, help me out here. The spec says: The Address of record is the SIP address that the registry knows the registrand. . . The Address of record is the public SIP uri you want people to call you at, regardless of the address of the phone you are answering on. It's the SIP phone address you place on your business card. A client uses the REGISTER method to register the address listed in the TO header field with a SIP Server. A client registers a temporary address, the address to a SIP UA, to the SIP registrar that is responsible for the domain in the AOR. This tells the SIP registrar (or location server) where to find you if someone calls your URI. When sending mail, I am not addressing the mail to the IP address you are reading the mail on, I am using your public e-mail address that is mapped to an e-mail server that is responsible for all e-mail to your domain. Later on, you fetch the e-mail from an e-mail client somewhere, with an IP address that propably changes as you travel around signing books ;-) SIP works the same way. You have a public address and a SIP proxy being responsible for keeping track of where you want to answer your calls. You can surely register several phones that you want to answer on. The proxy takes care of hiding this to the callee, so that the caller only get one set of responses. That's what the forking stuff is all about. If one phone is busy and the other one is answering, we should only signal 200 OK in SIP lingo to the caller. I don't see how two different clients can register with a server as the same address of record. Doesn't the second registration from a new client change the address of record for the registered client? You have one address-of-record that maps into several SIP URIs, one for each device. These are not as long-term as your a-o-r SIP URI. From the RFC: Location Service: A location service is used by a SIP redirect or proxy server to obtain information about a callees possible location(s). It contains a list of bindings of address-of-record keys to zero or more contact addresses. The bindings can be created and removed in many ways; this specification defines a REGISTER method that updates the bindings. If the second client is trying the same registration as the first client, and it's the responsibility of the client to provide the complete list of bindings, how does the second client know the list of bindings for the first client that bound the registration? It's *not* the responsibility of the *client* to provide a list, it's the server that responds with a list, telling the client by the way, these devices are also registred for the same a-o-r. So isn't this the problem * has? The first client registers as the address of record, then the second client comes in with the same registration and becomes the address of record? The address of record does not change because of a registration. The stored address (the contact: header) of where we can reach you (location) changes. And yes, if you have multiple devices registering for the same Asterisk sip [peer] account, it will be changing for each registration. This is not the behaviour of most SIP Proxys. 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. Due to Asterisk's multi-protocol architecture we have to make some compromises in the SIP channel to be able to have some kind of generic view of calls and phones in the core. A SIP proxy is never the end point of a call, it should never handle the media stream. The power is in the edge, in the phones. This is why transfers and other PBX functions is a bit messy with SIP and Asterisk, we are trying to find a way to do it centralized as Asterisk but de- centralized as SIP... 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. The architecture of Asterisk is a PBX, and the dial plan and a lot of apps wants to be in control of the device. 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. If you are looking for a SIP PBX, check Pingtel's Open Source software. If you are looking for a SIP proxy, test SIP Express Router from
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
* No, there's no quick fix for a 100 USD bounty How much you estimate on quick fix? -Kannaiyan. ___ 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
Excellent Post! Very Informative. Thanks a lot Sir! Regards, Girish From: Olle E. Johansson [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous Date: Mon, 12 Jul 2004 10:52:33 +0200 Paul Mahler wrote: Well, this is certainly getting exciting. Yes, it is. Sorry for coming in late to this debate... Andy, I took your advice and re-read the RFP. It's actually RFC, not RFP. (teasing :-) So, gentlemen, help me out here. The spec says: The Address of record is the SIP address that the registry knows the registrand. . . The Address of record is the public SIP uri you want people to call you at, regardless of the address of the phone you are answering on. It's the SIP phone address you place on your business card. A client uses the REGISTER method to register the address listed in the TO header field with a SIP Server. A client registers a temporary address, the address to a SIP UA, to the SIP registrar that is responsible for the domain in the AOR. This tells the SIP registrar (or location server) where to find you if someone calls your URI. When sending mail, I am not addressing the mail to the IP address you are reading the mail on, I am using your public e-mail address that is mapped to an e-mail server that is responsible for all e-mail to your domain. Later on, you fetch the e-mail from an e-mail client somewhere, with an IP address that propably changes as you travel around signing books ;-) SIP works the same way. You have a public address and a SIP proxy being responsible for keeping track of where you want to answer your calls. You can surely register several phones that you want to answer on. The proxy takes care of hiding this to the callee, so that the caller only get one set of responses. That's what the forking stuff is all about. If one phone is busy and the other one is answering, we should only signal 200 OK in SIP lingo to the caller. I don't see how two different clients can register with a server as the same address of record. Doesn't the second registration from a new client change the address of record for the registered client? You have one address-of-record that maps into several SIP URIs, one for each device. These are not as long-term as your a-o-r SIP URI. From the RFC: Location Service: A location service is used by a SIP redirect or proxy server to obtain information about a callees possible location(s). It contains a list of bindings of address-of-record keys to zero or more contact addresses. The bindings can be created and removed in many ways; this specification defines a REGISTER method that updates the bindings. If the second client is trying the same registration as the first client, and it's the responsibility of the client to provide the complete list of bindings, how does the second client know the list of bindings for the first client that bound the registration? It's *not* the responsibility of the *client* to provide a list, it's the server that responds with a list, telling the client by the way, these devices are also registred for the same a-o-r. So isn't this the problem * has? The first client registers as the address of record, then the second client comes in with the same registration and becomes the address of record? The address of record does not change because of a registration. The stored address (the contact: header) of where we can reach you (location) changes. And yes, if you have multiple devices registering for the same Asterisk sip [peer] account, it will be changing for each registration. This is not the behaviour of most SIP Proxys. 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. Due to Asterisk's multi-protocol architecture we have to make some compromises in the SIP channel to be able to have some kind of generic view of calls and phones in the core. A SIP proxy is never the end point of a call, it should never handle the media stream. The power is in the edge, in the phones. This is why transfers and other PBX functions is a bit messy with SIP and Asterisk, we are trying to find a way to do it centralized as Asterisk but de- centralized as SIP... 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. The architecture of Asterisk is a PBX, and the dial plan and a lot of apps wants to be in control of the device. 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
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Kannaiyan Natesan wrote: * No, there's no quick fix for a 100 USD bounty How much you estimate on quick fix? I apologize for my Swenglish language... 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. In order to provide a better answer, I need some time and funding to research this a bit more. Every problem has a solution. /O ___ 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 11/07/2004 at 18:11 Paul Mahler wrote: Well, this is certainly getting exciting. Andy, I took your advice and re-read the RFP. Andy--I don't think you are a Sorry, I was sleeping when these new emails came in I've read the other responses which seem to make it pretty clear.. and address all the points and give most of the info you need...(do I need to add to it?) I couldn't for the life of me remember the name (it was late) and Andres reminded us all that it's called Parallel Forking - it's by far the best feature of SIP and nearly, but not quite, negates the NAT problems. The reason i've been so adamant about this, is that I use it every day... my * box and 2 of my phones register with a local sip proxy for the same sip address... I use this just incase my * box dies, since it's my development box too and I'm always mesing with it. good candidate for a beginner's book on *, but if you send my your address, I'll send you a copy on me. :-) Or some Ninja assasin... ;) Perhaps you could also sign it :D (not the Ninja assasin ;) ) Andy, I'm in your hands. I was too late... I took the liberty of getting some sleep... appologies. Andy ___ 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
I don't think we should let these misunderstandings judge the quality of Paul's Asterisk book. Even authors need to learn now and then :-) Can I just point out that the reason I said what I said (see, I can't write) was because Paul steadfastly refused to believe what we were saying, rather than investigating it.ie His response was more like: You're wrong, I'm right. rather than: Oh... maybe there's something I'm not aware of. I shall investigate immediately. I'll admit techies always argue over stuff like this, primarily because they don't want to be seen to not know something.. anyho.. I'd consider the discussion of the existance of forking closed and proven and we can now begin arguing over why it would/wouldn't be a good idea to include this behaviour in * ;) Andy ___ 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
SIP simultaneous registry possible workaround (was Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry)
This may sound like a stupid work around, but how about registering
different extensions and putting both of them in the Dial String (so they
would ring at once) and giving both extensions the same caller id?
I do something with my zaptel and x lite phones... I assign them both the
same number and they both come out as the same caller ID. All lines that
you want will ring, plus outgoing caller ID will be what you want it to be.
This gives you also the possibility to have one line which will never be
busy. You pu
I hope this helps,
Youness
ie:
--- in extensions.conf ---:
phone1=SIP/32SIP/33SIP/34
[incoming]
s,1,Dial(${PHONE1})
-- in sip.conf
[phone1]
callerid=Youness Mobile 21
type=friend
secret=secret
[phone2]
callerid=Youness Mobile 21
type=friend
secret=secret
Youness
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Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
in response to Olle's excellent post, ... (B (Bin particular ... (B (BAsterisk is *not* a SIP proxy. It's a SIP registrar and (Blocation server. (BIt's a very clever SIP UA. It wants to be in the middle (Bof the call (Band wants to be in control of each device. This (Bdevice-slave view doesn't match the SIP architecture. (B (Band ... (B (BI've spent a considerable amount of time investigating (Bsupport for (Bmultiple registrations on one Asterisk sip [peer] account (Band after (Blearning about Asterisk's architecture come to the (Bconclusion that (Bit is not an easy or even a desirable feature to (Bimplement. (B (Band ... (B (BIt may be possible, but will probably lead to a lot of (Bchanges to (BAsterisk, both core and applications, that no other (Bchannel will (Bbenefit from. A quick hack to support it may lead to a (Blot of (Bconfusion on how to handle other apps. And it's a lot (Bmore work (Bthan the bounty will cover. I suggest that you use a (Bforking SIP (Bproxy in conjunction with Asterisk to get this (Bfunctionality. (B (BPrecisely! A fairly simple and elegant solution. (B (BFor those rare occasions where one would really need (Bmultiple concurrent SIP registrations I'd say one should (Bconsider running Asterisk in combination with a SIP proxy. (BSince SER is a free download, this wouldn't seem to be (Bsuch a big deal IF IT WASNT for the fact that one will (Bthen need to run two boxes. (B (BIt would make a lot of sense to provide support for an (Beasy-to-configure set up where Asterisk can live together (Bwith another SIP speaking piece of software on the same (Bbox. (B (BSomething along the lines of ... (B (B(ip1:5060)---[*]---[portswapper]---(ip1:5061)---[SER]---(ip2:5060) (B (BSomething like this should allow you to run Asterisk on (Bone address (ie LAN side) and SER on another (ie WAN (Bside), so you get the best of both Asterisk and a SIP (Bproxy all in one box. (B (BThis would also make it possible to run a SIP softphone (Balongside Asterisk on a notebook, so it would solve two (Bbirds with one stone. (B (BI'd like to emphasise however, that most of the problems (Bdescribed in this thread are NOT good reasons for multiple (Bconcurrent SIP registrations. Those problems have other (Bsolutions. Let's take a look at them. (B (B1) Call centre scenario (B (BProblem: multiple agents should receive calls on the same (Bphone number (B (BSolution: assign a number to a call queue and let the call (Bqueue distribute incoming calls to the agents on different (BSIP phones, each of which should have unique logins for (Breasons of accounting and quality assurance. (B (Bmultiple concurrent registrations on the same SIP account (Bin call centres is a BAD IDEA. (B (B2) Overworked admin scenario (B (BProblem: asterisk admin doesn't want to deal with support (Bcalls for adding additional SIP phones (B (BSolution: a simple self provisioning system, either web (Bbased or even IVR based. (B (B3) Dual line desk phone scenario (B (BProblem: dual line desk phone requires multiple (Bregistrations, one per line (B (BSolution: let the phone register on two different SIP (Baccounts, which is how any conventional PBX handles dual (Bline phones: one extension per line. (B (B4) Call group scenario (B (BProblem: multiple phones to ring on the same extension (B (BSolution: use the call group feature or use the dial (Bcommand with multiple SIP peers (B (B (BFor the avoidance of doubt, I am not saying there is no (Bsituation for which multiple concurrent SIP registrations (Bmay be the right solution, but the problems described so (Bfar are *not*. (B (BBut if anybody has a problem that truly warrants parallel (Bforking, then I propose you look into sponsoring somebody (Bto work on the little port swapping trick to run SER (Bconcurrently on your Asterisk box. (B (Brgds (Bbenjk (B (B (B__ (BDo You Yahoo!? (Bhttp://bb.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SIP simultaneous registry possible workaround (was Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry)
You need to go through all the messages again.
You feel it as stupid, I sometimes feel * is stupid and could not accept
this simply. To resolve this problem I simply use SER. But it is another
software, that makes me worried.
If you still think this is not at all needed. I better suggest you not to
waste of others around here. I think Olle, Andy gave better explanation on
the same. I You can't expect more than that.
-Kannaiyan.
- Original Message -
From: Youness El Andaloussi [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 12, 2004 4:40 PM
Subject: SIP simultaneous registry possible workaround (was Re:
[Asterisk-Users] New Asterisk bounty: SIP simultaneous registry)
This may sound like a stupid work around, but how about registering
different extensions and putting both of them in the Dial String (so they
would ring at once) and giving both extensions the same caller id?
I do something with my zaptel and x lite phones... I assign them both the
same number and they both come out as the same caller ID. All lines that
you want will ring, plus outgoing caller ID will be what you want it to
be.
This gives you also the possibility to have one line which will never be
busy. You pu
I hope this helps,
Youness
ie:
--- in extensions.conf ---:
phone1=SIP/32SIP/33SIP/34
[incoming]
s,1,Dial(${PHONE1})
-- in sip.conf
[phone1]
callerid=Youness Mobile 21
type=friend
secret=secret
[phone2]
callerid=Youness Mobile 21
type=friend
secret=secret
Youness
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Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Jason Penton wrote: Well Andres is right but there are numerous problems with quite a few SIP clients that do NOT follow the the SIP RFC correctly. There is a problem with dialog creation in a number of SIP products out there. SIP dialog creation is the critical part of the spec that supports parallel forking - so be careful. Jason Yes, but this is a problem of the client not *. The clients need to fix their firmware. You know this already though. -- Daniel Jimenez djimenez[at]pobox[dot]com ___ 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
It would appear to me that you could run Asterisk and SER on the same box. Asterisk's sip-binding can be configured to be 5061 instead of 5060, and I'd assume SER can be configured as well. What am I missing? -Original Message- From: Sunrise Ltd [mailto:[EMAIL PROTECTED] Sent: Monday, July 12, 2004 11:20 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous 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) ___ 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
Re: SIP simultaneous registry possible workaround (was Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry)
On Mon, 2004-07-12 at 10:40, Youness El Andaloussi wrote: This may sound like a stupid work around, but how about registering different extensions and putting both of them in the Dial String (so they would ring at once) and giving both extensions the same caller id? That's exactly how I and everyone else that needs this functionality does it. --Eric -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ 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
Sunrise Ltd wrote: But if anybody has a problem that truly warrants parallel forking, then I propose you look into sponsoring somebody to work on the little port swapping trick to run SER concurrently on your Asterisk box. We're running SER and Asterisk on the same system with Like2Fone.com and we just stuck Asterisk on a different port then redirect calls as needed, although I doubt it would be as difficult as your making out, if you stick SER on an alternative port and then just use that to connect your clients to problem solved, in effect the opposite to what we wanted to achieve... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the confrontation between the stream and the rock, the stream always wins; not through strength, but through persistence. ___ 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
I hope you clearly understand that everyone here **KNOWS** to use (Balternative software such as SER, what is needed here is the same facility (Bin asterisk. (B (BWhen I see beauty, I want to see all the things in a same figure rather than (Bsplitted. If you see with splitted face doen't mean there is no beauty but (Bit cause inconvinience. (B (BWhat everyone here want to see the same beauty in asterisk and when this can (Bbe done in another softwares why not in asterisk. (B (BAs Olle says, nothing is impossible. There is a possible solution, but takes (Btime. (B (B-Kannaiyan. (B (B (B- Original Message - (BFrom: "Sunrise Ltd" [EMAIL PROTECTED] (BTo: [EMAIL PROTECTED] (BSent: Monday, July 12, 2004 5:20 PM (BSubject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous (B (B (B in response to Olle's excellent post, ... (B (B in particular ... (B (B Asterisk is *not* a SIP proxy. It's a SIP registrar and (B location server. (B It's a very clever SIP UA. It wants to be in the middle (B of the call (B and wants to be in control of each device. This (B device-slave view doesn't match the SIP architecture. (B (B and ... (B (B I've spent a considerable amount of time investigating (B support for (B multiple registrations on one Asterisk sip [peer] account (B and after (B learning about Asterisk's architecture come to the (B conclusion that (B it is not an easy or even a desirable feature to (B implement. (B (B and ... (B (B It may be possible, but will probably lead to a lot of (B changes to (B Asterisk, both core and applications, that no other (B channel will (B benefit from. A quick hack to support it may lead to a (B lot of (B confusion on how to handle other apps. And it's a lot (B more work (B than the bounty will cover. I suggest that you use a (B forking SIP (B proxy in conjunction with Asterisk to get this (B functionality. (B (B Precisely! A fairly simple and elegant solution. (B (B For those rare occasions where one would really need (B multiple concurrent SIP registrations I'd say one should (B consider running Asterisk in combination with a SIP proxy. (B Since SER is a free download, this wouldn't seem to be (B such a big deal IF IT WASNT for the fact that one will (B then need to run two boxes. (B (B It would make a lot of sense to provide support for an (B easy-to-configure set up where Asterisk can live together (B with another SIP speaking piece of software on the same (B box. (B (B Something along the lines of ... (B (B (ip1:5060)---[*]---[portswapper]---(ip1:5061)---[SER]---(ip2:5060) (B (B Something like this should allow you to run Asterisk on (B one address (ie LAN side) and SER on another (ie WAN (B side), so you get the best of both Asterisk and a SIP (B proxy all in one box. (B (B This would also make it possible to run a SIP softphone (B alongside Asterisk on a notebook, so it would solve two (B birds with one stone. (B (B I'd like to emphasise however, that most of the problems (B described in this thread are NOT good reasons for multiple (B concurrent SIP registrations. Those problems have other (B solutions. Let's take a look at them. (B (B 1) Call centre scenario (B (B Problem: multiple agents should receive calls on the same (B phone number (B (B Solution: assign a number to a call queue and let the call (B queue distribute incoming calls to the agents on different (B SIP phones, each of which should have unique logins for (B reasons of accounting and quality assurance. (B (B multiple concurrent registrations on the same SIP account (B in call centres is a BAD IDEA. (B (B 2) Overworked admin scenario (B (B Problem: asterisk admin doesn't want to deal with support (B calls for adding additional SIP phones (B (B Solution: a simple self provisioning system, either web (B based or even IVR based. (B (B 3) Dual line desk phone scenario (B (B Problem: dual line desk phone requires multiple (B registrations, one per line (B (B Solution: let the phone register on two different SIP (B accounts, which is how any conventional PBX handles dual (B line phones: one extension per line. (B (B 4) Call group scenario (B (B Problem: multiple phones to ring on the same extension (B (B Solution: use the call group feature or use the dial (B command with multiple SIP peers (B (B (B For the avoidance of doubt, I am not saying there is no (B situation for which multiple concurrent SIP registrations (B may be the right solution, but the problems described so (B far are *not*. (B (B But if anybody has a problem that truly warrants parallel (B forking, then I propose you look into sponsoring somebody (B to work on the little port swapping trick to run SER (B concurrently on your Asterisk box. (B (B rgds (B benjk (B (B (B __ (B Do You Yahoo!? (B http://bb.yahoo.co.j
Re: SIP simultaneous registry possible workaround (was Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry)
On Monday 12 July 2004 15:18, Eric Wieling wrote: On Mon, 2004-07-12 at 10:40, Youness El Andaloussi wrote: This may sound like a stupid work around, but how about registering different extensions and putting both of them in the Dial String (so they would ring at once) and giving both extensions the same caller id? That's exactly how I and everyone else that needs this functionality does it. That sound unwieldy as the number of simultaneous ringers increases... Can SIP peers be grouped like Zap channels? -A. ___ 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: SIP simultaneous registry possible workaround (was Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry)
Dude relax, take a deep breath and be zen :). From the tone of your email, looks like you expect me to apologize for trying to be helpful :) There is more than one way to skin a cat. You may not like my solution, but it would work and saying I am wasting your time is a bit over reacting in my humble opinion. Automated with a couple shell scripts it could work nicely around and with shell scripts generating the configs, dhcp and TFTP, you could actually make a lot of things automatic and take your 100$ to buy some ice to cool down. Youness At 17:16 12/07/2004, you wrote: You need to go through all the messages again. You feel it as stupid, I sometimes feel * is stupid and could not accept this simply. To resolve this problem I simply use SER. But it is another software, that makes me worried. If you still think this is not at all needed. I better suggest you not to waste of others around here. I think Olle, Andy gave better explanation on the same. I You can't expect more than that. -Kannaiyan. ___ 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: SIP simultaneous registry possible workaround (was Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry)
Not sure how many phones you need ringing at the same time, but it should not be a major problem as only an invite is sent... and it should not require much bandwidth just to invite. At 23:57 12/07/2004, you wrote: That sound unwieldy as the number of simultaneous ringers increases... Can SIP peers be grouped like Zap channels? -A. ___ 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 same SIP account in call centres is a BAD IDEA. 2) Overworked admin scenario Problem: asterisk admin doesn't want to deal with support calls for adding additional SIP phones Solution: a simple self
Re: SIP simultaneous registry possible workaround (was Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry)
On Monday 12 July 2004 22:41, Youness El Andaloussi wrote: That sound unwieldy as the number of simultaneous ringers increases... Can SIP peers be grouped like Zap channels? Not sure how many phones you need ringing at the same time, but it should not be a major problem as only an invite is sent... and it should not require much bandwidth just to invite. I wasn't talking about bandwidth but rather lengthy Dial() commands... exten = s,1,Dial(SIP/someuserSIP/someuserSIP/someuserSIP/someuserSIP/someuserSIP/someuserSIP/someuserSIP/someuserSIP/someuserSIP/someuserSIP/someuserSIP/someuserSIP/someuserSIP/someuserSIP/someuserSIP/someuserSIP/someuserSIP/someuserSIP/someuserSIP/someuserSIP/someuserSIP/someuserSIP/someuserSIP/someuserSIP/someuserSIP/someuser) kind of thing... seems awfully unwieldy. -A. ___ 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: SIP simultaneous registry possible workaround (was Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry)
If you talk about alternatives and loosing my hair, the *BEST* solution for me to go for is SER, which is fast, best and reliable SIP Proxy and we do that right now for the same purpose. Also the discussion is not about choosing alternatives but having it on the same. -Kannaiyan. - Original Message - From: Youness El Andaloussi [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 13, 2004 3:34 AM Subject: Re: SIP simultaneous registry possible workaround (was Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry) Dude relax, take a deep breath and be zen :). From the tone of your email, looks like you expect me to apologize for trying to be helpful :) There is more than one way to skin a cat. You may not like my solution, but it would work and saying I am wasting your time is a bit over reacting in my humble opinion. Automated with a couple shell scripts it could work nicely around and with shell scripts generating the configs, dhcp and TFTP, you could actually make a lot of things automatic and take your 100$ to buy some ice to cool down. Youness At 17:16 12/07/2004, you wrote: You need to go through all the messages again. You feel it as stupid, I sometimes feel * is stupid and could not accept this simply. To resolve this problem I simply use SER. But it is another software, that makes me worried. If you still think this is not at all needed. I better suggest you not to waste of others around here. I think Olle, Andy gave better explanation on the same. I You can't expect more than that. -Kannaiyan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
When I call a SIP user, the phone should ring in more
(Bthan one
(Bextentions. Also more than one phone should be able to
(Bregister with
(Basterisk. Right now it is not the case.
(B
(BThere is no issue here. You seem to be confused, that's
(Ball.
(B
(BA SIP account is a SIP account and an extension is an
(Bextension. You can assign an extension to an account (or
(Bto multiple accounts) and the tool for that is the dial
(Bcommand.
(B
(BHowever, there is no implicit assignment between an
(Bextension and an account and that is good so. This should
(Bnot be changed because it would harm Asterisk's
(Bflexibility and manageability.
(B
(B
(BThis type of situations might be needed in call centres.
(B
(BCalled 12345
(B|---(12345) Ringing
(B|---(12345) Ringing
(B|---(12345) Ringing
(B
(BAs I said, you are confusing extensions with accounts. The
(Bfirst "12345" is an extension, the three "(12345)"s are
(Baccounts. Those are different layers, don't mix them up.
(B
(BYou should always be able to distinguish between devices,
(Beven if they are assigned the same phone number. In fact,
(Bin a call centre you'd be using a call queue. It would be
(Brather nonsensical for a call queue management to have to
(Bdistinguish between multiple identical agents.
(B
(BTherefore, setting up multiple devices with the same
(Baccount credentials is not a good idea, especially not in
(Ba call centre. Each device and each agent should have
(Btheir own unique account credentials and assigning
(Bextensions to them should always be done through the
(Bdialplan and only the dialplan.
(B
(BAsterisk has been designed this way. It is a good design.
(BIt should NOT be changed nor undermined.
(B
(BYou may want to do something like this ...
(B
(B[GLOBALS]
(B
(BA-GROUP = SIP/2001 SIP2002 SIP/2003
(B
(BB-BROUP = SIP/jdoe SIP/dflint SIP/bsmith
(B
(B...
(B
(B
(B[Support]
(B
(Bexten = 12345,1,Dial(${A-GROUP},30,r)
(B...
(B
(Bexten = 54321,1,Dial(${B-GROUP},30,r)
(B...
(B
(B
(BThere is of course an issue when you want to let different
(Bphones start ringing at different times, for example, the
(Bfirst phone is supposed to start ringing immediately and
(Bthe other two are only to join in if the first phone
(Bhasn't been picked up in 10 seconds, like so
(B
(Bexten = 12345,1,Dial(${JDOE},10,r)
(Bexten = 12345,2,Dial(${JDOE}{DFLINT}${BSMITH},20,r)
(B
(BThis works but if JDOE was to pick up right between those
(Btwo dial commands, then it will have been too late for the
(Bfirst and JDOE will be "on the phone" for the second dial
(Bcommand, so there is some room for improvement. A bounty
(Bmight better be spent on solving this little problem.
(B
(BAlso, Asterisk supports call groups and pickup groups.
(BIndeed, there have been some bugs with those features and
(BI am not sure if they have have been fixed, but if they
(Bhaven't, then it would again make more sense to put the
(Bbounty on fixing those rather than creating an ugly
(Bworkaround.
(B
(B
(BI feel this is a great feature
(B
(BI don't and if you spent some more time with Asterisk and
(Bimmerse its philosophy, then you'll very likely change
(Byour mind.
(B
(Bin other SIP proxy server this can be done easily
(B
(BAsterisk is not a SIP proxy. It's a telephone exchange.
(B
(Bi mean its default 1 or more phone could be registered
(Bat 1 number (12345) and resulting same effect
(B
(BA phone does not register at a number. It registers at an
(Baccount to which Asterisk can assign one or more numbers.
(BThis makes perfect sense and it is a far more flexible and
(Bbetter design.
(B
(BSIP proxies' auto assignment of extensions to SIP
(Busernames is a serious limitation, not an advantage.
(B
(B
(BThe only situation where one might want to consider
(Bsupporting multiple concurrent logins on the same account
(Bis for public VoIP service providers where end users might
(Bhave a SIP phone on their desk and use a softphone on
(Btheir notebook when they are traveling.
(B
(BBut here again, it is more likely to be a disadvantage.
(BConsider the following situation ...
(B
(B1) Incoming call to 12345
(B
(B2) both deskphone 12345 and road warrior's notebook 12345
(Bring
(B
(B3) Secretary of Mr. 12345 picks up before he himself is
(Bable to do so
(B
(B4) Caller asks for Mr.12345 but secretary has no way of
(Btrying to transfer the call
(B
(BOTOH, Asterisk handles this situation much better ...
(B
(B1) Incoming call to extension 12345
(B
(B2) Dial command determines to ring both deskphone and road
(Bwarrior's notebook which are on different extensions
(B
(B3) Secretary of Mr. Road Warrior picks up before he
(Bhimself is able to do so
(B
(B4) Caller asks for Mr. Road Warrior, secretary transfers
(Bto internal extension of road warrior notebook's softphone
(B
(B
(BI am sorry but your bounty doesn't seem to make sense. It
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
I accept your views.
(B
(BI have a specific requirements, can you help to attain the same.
(BIn our business we have 25 employees handling customer service.
(B
(BI want to add or remove employees in the customer service so does the
(Bdevices connected to it.
(BI don't want to make any changes in the asterisk, and all I need is to plug
(Bin the VoIP Phone and start handling the customer service. I would like to
(Bdo for as many employees as I want without any problems.
(B
(BCan you think of a better solution?
(B
(B-Kannaiyan.
(B
(B- Original Message -
(BFrom: "Sunrise Ltd" [EMAIL PROTECTED]
(BTo: [EMAIL PROTECTED]
(BSent: Sunday, July 11, 2004 9:15 AM
(BSubject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
(B
(B
(B When I call a SIP user, the phone should ring in more
(B than one
(B extentions. Also more than one phone should be able to
(B register with
(B asterisk. Right now it is not the case.
(B
(B There is no issue here. You seem to be confused, that's
(B all.
(B
(B A SIP account is a SIP account and an extension is an
(B extension. You can assign an extension to an account (or
(B to multiple accounts) and the tool for that is the dial
(B command.
(B
(B However, there is no implicit assignment between an
(B extension and an account and that is good so. This should
(B not be changed because it would harm Asterisk's
(B flexibility and manageability.
(B
(B
(B This type of situations might be needed in call centres.
(B
(B Called 12345
(B |---(12345) Ringing
(B |---(12345) Ringing
(B |---(12345) Ringing
(B
(B As I said, you are confusing extensions with accounts. The
(B first "12345" is an extension, the three "(12345)"s are
(B accounts. Those are different layers, don't mix them up.
(B
(B You should always be able to distinguish between devices,
(B even if they are assigned the same phone number. In fact,
(B in a call centre you'd be using a call queue. It would be
(B rather nonsensical for a call queue management to have to
(B distinguish between multiple identical agents.
(B
(B Therefore, setting up multiple devices with the same
(B account credentials is not a good idea, especially not in
(B a call centre. Each device and each agent should have
(B their own unique account credentials and assigning
(B extensions to them should always be done through the
(B dialplan and only the dialplan.
(B
(B Asterisk has been designed this way. It is a good design.
(B It should NOT be changed nor undermined.
(B
(B You may want to do something like this ...
(B
(B [GLOBALS]
(B
(B A-GROUP = SIP/2001 SIP2002 SIP/2003
(B
(B B-BROUP = SIP/jdoe SIP/dflint SIP/bsmith
(B
(B ...
(B
(B
(B [Support]
(B
(B exten = 12345,1,Dial(${A-GROUP},30,r)
(B ...
(B
(B exten = 54321,1,Dial(${B-GROUP},30,r)
(B ...
(B
(B
(B There is of course an issue when you want to let different
(B phones start ringing at different times, for example, the
(B first phone is supposed to start ringing immediately and
(B the other two are only to join in if the first phone
(B hasn't been picked up in 10 seconds, like so
(B
(B exten = 12345,1,Dial(${JDOE},10,r)
(B exten = 12345,2,Dial(${JDOE}{DFLINT}${BSMITH},20,r)
(B
(B This works but if JDOE was to pick up right between those
(B two dial commands, then it will have been too late for the
(B first and JDOE will be "on the phone" for the second dial
(B command, so there is some room for improvement. A bounty
(B might better be spent on solving this little problem.
(B
(B Also, Asterisk supports call groups and pickup groups.
(B Indeed, there have been some bugs with those features and
(B I am not sure if they have have been fixed, but if they
(B haven't, then it would again make more sense to put the
(B bounty on fixing those rather than creating an ugly
(B workaround.
(B
(B
(B I feel this is a great feature
(B
(B I don't and if you spent some more time with Asterisk and
(B immerse its philosophy, then you'll very likely change
(B your mind.
(B
(B in other SIP proxy server this can be done easily
(B
(B Asterisk is not a SIP proxy. It's a telephone exchange.
(B
(B i mean its default 1 or more phone could be registered
(B at 1 number (12345) and resulting same effect
(B
(B A phone does not register at a number. It registers at an
(B account to which Asterisk can assign one or more numbers.
(B This makes perfect sense and it is a far more flexible and
(B better design.
(B
(B SIP proxies' auto assignment of extensions to SIP
(B usernames is a serious limitation, not an advantage.
(B
(B
(B The only situation where one might want to consider
(B supporting multiple concurrent logins on the same account
(B is for public VoIP service providers where end users might
(B have a SIP phone on their desk and use a softphone on
(B their notebook when they are traveling.
RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
I was going to keep out of this (was interesting to read, as I have dealt
(Bwith simmillar situation) however I would like to add just this one commnet.
(B
(BTry to better understand asterisk than to throw about your money. What you
(Bwant to do is perfectly possible with asterisk there is no need to add a new
(Bconfusing feature.
(B
(BAs for your bounty, donate it to the wiki ! :-)
(B
(BUmar.
(B
(B-Original Message-
(BFrom: [EMAIL PROTECTED]
(B[mailto:[EMAIL PROTECTED] Behalf Of Kannaiyan Natesan
(BSent: 11 July 2004 09:51
(BTo: [EMAIL PROTECTED]
(BSubject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
(B
(B
(BI accept your views.
(B
(BI have a specific requirements, can you help to attain the same.
(BIn our business we have 25 employees handling customer service.
(B
(BI want to add or remove employees in the customer service so does the
(Bdevices connected to it.
(BI don't want to make any changes in the asterisk, and all I need is to plug
(Bin the VoIP Phone and start handling the customer service. I would like to
(Bdo for as many employees as I want without any problems.
(B
(BCan you think of a better solution?
(B
(B-Kannaiyan.
(B
(B- Original Message -
(BFrom: "Sunrise Ltd" [EMAIL PROTECTED]
(BTo: [EMAIL PROTECTED]
(BSent: Sunday, July 11, 2004 9:15 AM
(BSubject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
(B
(B
(B When I call a SIP user, the phone should ring in more
(B than one
(B extentions. Also more than one phone should be able to
(B register with
(B asterisk. Right now it is not the case.
(B
(B There is no issue here. You seem to be confused, that's
(B all.
(B
(B A SIP account is a SIP account and an extension is an
(B extension. You can assign an extension to an account (or
(B to multiple accounts) and the tool for that is the dial
(B command.
(B
(B However, there is no implicit assignment between an
(B extension and an account and that is good so. This should
(B not be changed because it would harm Asterisk's
(B flexibility and manageability.
(B
(B
(B This type of situations might be needed in call centres.
(B
(B Called 12345
(B |---(12345) Ringing
(B |---(12345) Ringing
(B |---(12345) Ringing
(B
(B As I said, you are confusing extensions with accounts. The
(B first "12345" is an extension, the three "(12345)"s are
(B accounts. Those are different layers, don't mix them up.
(B
(B You should always be able to distinguish between devices,
(B even if they are assigned the same phone number. In fact,
(B in a call centre you'd be using a call queue. It would be
(B rather nonsensical for a call queue management to have to
(B distinguish between multiple identical agents.
(B
(B Therefore, setting up multiple devices with the same
(B account credentials is not a good idea, especially not in
(B a call centre. Each device and each agent should have
(B their own unique account credentials and assigning
(B extensions to them should always be done through the
(B dialplan and only the dialplan.
(B
(B Asterisk has been designed this way. It is a good design.
(B It should NOT be changed nor undermined.
(B
(B You may want to do something like this ...
(B
(B [GLOBALS]
(B
(B A-GROUP = SIP/2001 SIP2002 SIP/2003
(B
(B B-BROUP = SIP/jdoe SIP/dflint SIP/bsmith
(B
(B ...
(B
(B
(B [Support]
(B
(B exten = 12345,1,Dial(${A-GROUP},30,r)
(B ...
(B
(B exten = 54321,1,Dial(${B-GROUP},30,r)
(B ...
(B
(B
(B There is of course an issue when you want to let different
(B phones start ringing at different times, for example, the
(B first phone is supposed to start ringing immediately and
(B the other two are only to join in if the first phone
(B hasn't been picked up in 10 seconds, like so
(B
(B exten = 12345,1,Dial(${JDOE},10,r)
(B exten = 12345,2,Dial(${JDOE}{DFLINT}${BSMITH},20,r)
(B
(B This works but if JDOE was to pick up right between those
(B two dial commands, then it will have been too late for the
(B first and JDOE will be "on the phone" for the second dial
(B command, so there is some room for improvement. A bounty
(B might better be spent on solving this little problem.
(B
(B Also, Asterisk supports call groups and pickup groups.
(B Indeed, there have been some bugs with those features and
(B I am not sure if they have have been fixed, but if they
(B haven't, then it would again make more sense to put the
(B bounty on fixing those rather than creating an ugly
(B workaround.
(B
(B
(B I feel this is a great feature
(B
(B I don't and if you spent some more time with Asterisk and
(B immerse its philosophy, then you'll very likely change
(B your mind.
(B
(B in other SIP proxy server this can be done easily
(B
(B Asterisk is not a SIP proxy. It's a telephone exchange.
(B
(B i mean its default 1 or more phone could be registered
(B at 1
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
I explained him a sample need.
(BI don't think asterisk does whatever i want in sip. It is an good PBX.
(B
(BPlease help me to understand. Anywhere am I wrong ? Or as you say is that
(BSIP feature is written?
(B
(B-Kannaiyan.
(B
(B
(B- Original Message -
(BFrom: "usedcanon" [EMAIL PROTECTED]
(BTo: [EMAIL PROTECTED]
(BSent: Sunday, July 11, 2004 10:02 AM
(BSubject: RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
(B
(B
(B I was going to keep out of this (was interesting to read, as I have dealt
(B with simmillar situation) however I would like to add just this one
(Bcommnet.
(B
(B Try to better understand asterisk than to throw about your money. What you
(B want to do is perfectly possible with asterisk there is no need to add a
(Bnew
(B confusing feature.
(B
(B As for your bounty, donate it to the wiki ! :-)
(B
(B Umar.
(B
(B -Original Message-
(B From: [EMAIL PROTECTED]
(B [mailto:[EMAIL PROTECTED] Behalf Of Kannaiyan
(BNatesan
(B Sent: 11 July 2004 09:51
(B To: [EMAIL PROTECTED]
(B Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
(B
(B
(B I accept your views.
(B
(B I have a specific requirements, can you help to attain the same.
(B In our business we have 25 employees handling customer service.
(B
(B I want to add or remove employees in the customer service so does the
(B devices connected to it.
(B I don't want to make any changes in the asterisk, and all I need is to
(Bplug
(B in the VoIP Phone and start handling the customer service. I would like to
(B do for as many employees as I want without any problems.
(B
(B Can you think of a better solution?
(B
(B -Kannaiyan.
(B
(B - Original Message -
(B From: "Sunrise Ltd" [EMAIL PROTECTED]
(B To: [EMAIL PROTECTED]
(B Sent: Sunday, July 11, 2004 9:15 AM
(B Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
(B
(B
(B When I call a SIP user, the phone should ring in more
(B than one
(B extentions. Also more than one phone should be able to
(B register with
(B asterisk. Right now it is not the case.
(B
(B There is no issue here. You seem to be confused, that's
(B all.
(B
(B A SIP account is a SIP account and an extension is an
(B extension. You can assign an extension to an account (or
(B to multiple accounts) and the tool for that is the dial
(B command.
(B
(B However, there is no implicit assignment between an
(B extension and an account and that is good so. This should
(B not be changed because it would harm Asterisk's
(B flexibility and manageability.
(B
(B
(B This type of situations might be needed in call centres.
(B
(B Called 12345
(B |---(12345) Ringing
(B |---(12345) Ringing
(B |---(12345) Ringing
(B
(B As I said, you are confusing extensions with accounts. The
(B first "12345" is an extension, the three "(12345)"s are
(B accounts. Those are different layers, don't mix them up.
(B
(B You should always be able to distinguish between devices,
(B even if they are assigned the same phone number. In fact,
(B in a call centre you'd be using a call queue. It would be
(B rather nonsensical for a call queue management to have to
(B distinguish between multiple identical agents.
(B
(B Therefore, setting up multiple devices with the same
(B account credentials is not a good idea, especially not in
(B a call centre. Each device and each agent should have
(B their own unique account credentials and assigning
(B extensions to them should always be done through the
(B dialplan and only the dialplan.
(B
(B Asterisk has been designed this way. It is a good design.
(B It should NOT be changed nor undermined.
(B
(B You may want to do something like this ...
(B
(B [GLOBALS]
(B
(B A-GROUP = SIP/2001 SIP2002 SIP/2003
(B
(B B-BROUP = SIP/jdoe SIP/dflint SIP/bsmith
(B
(B ...
(B
(B
(B [Support]
(B
(B exten = 12345,1,Dial(${A-GROUP},30,r)
(B ...
(B
(B exten = 54321,1,Dial(${B-GROUP},30,r)
(B ...
(B
(B
(B There is of course an issue when you want to let different
(B phones start ringing at different times, for example, the
(B first phone is supposed to start ringing immediately and
(B the other two are only to join in if the first phone
(B hasn't been picked up in 10 seconds, like so
(B
(B exten = 12345,1,Dial(${JDOE},10,r)
(B exten = 12345,2,Dial(${JDOE}{DFLINT}${BSMITH},20,r)
(B
(B This works but if JDOE was to pick up right between those
(B two dial commands, then it will have been too late for the
(B first and JDOE will be "on the phone" for the second dial
(B command, so there is some room for improvement. A bounty
(B might better be spent on solving this little problem.
(B
(B Also, Asterisk supports call groups and pickup groups.
(B Indeed, there have been so
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry
Daniel Jimenez wrote: http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultaneous+registry From the WIKI: Contributions Manager: Daniel Jimenez (cuban) Bounty: $50 USD Date opened: July 10, 2004 Contributors: cuban ($50) Detail Yes, Yes I know you could do all sorts of fun with the dialplan to produce a similar effect, but I still would like to be able to do this. Plus it's easy money :). I have some users with a 7960 who are administrative assistants who monitor calls for 3 or 4 other people. It'd be nice to have two line instances for them, and one for the person(s) whom they assist. Contact me: djimenez at pobox.com if you're interested in making this happen. Eh... Sort of like shadow lines ??? Remember that Dial(SIP/1 H323/1 ZAP/1,[timeout],[options],[URL]) will dial all 3 extensions simultaneously (regardless of channel choice) and with a little tinkering in your dialplan you can even activate/deactivate this from the Manager Station Why make it harder than it really is ??? I believe this is exactly what they do when programming your regular (old world) PBX systems... -- Soren ___ 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
You are confused about what a SIP session is and what a SIP session does. SIP, session initiation protocol, controls an RTP, real time protocol, session between two IP endpionts. The end points have to have unique IP addresses for the session to run. The unique SIP registration is how * finds a UNIQUE endpoint. You don't want SIP to solve your problem, you want * to solve your problem. You are asking for this SIP feature because you are confused as to how SIP and * work, and how they work together. You can easily fix your business problem with *, but not with mechanism you are asking for. You should spend your money on getting a copy of each of the two books that are now available and learn *. Then it will be clear to you that you don't really want what you are asking for. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kannaiyan Natesan Sent: Sunday, July 11, 2004 1:15 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous I explained him a sample need. I don't think asterisk does whatever i want in sip. It is an good PBX. Please help me to understand. Anywhere am I wrong ? Or as you say is that SIP feature is written? -Kannaiyan. - Original Message - From: usedcanon [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 11, 2004 10:02 AM Subject: RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous I was going to keep out of this (was interesting to read, as I have dealt with simmillar situation) however I would like to add just this one commnet. Try to better understand asterisk than to throw about your money. What you want to do is perfectly possible with asterisk there is no need to add a new confusing feature. As for your bounty, donate it to the wiki ! :-) Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Kannaiyan Natesan Sent: 11 July 2004 09:51 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous I accept your views. I have a specific requirements, can you help to attain the same. In our business we have 25 employees handling customer service. I want to add or remove employees in the customer service so does the devices connected to it. I don't want to make any changes in the asterisk, and all I need is to plug in the VoIP Phone and start handling the customer service. I would like to do for as many employees as I want without any problems. Can you think of a better solution? -Kannaiyan. - Original Message - From: Sunrise Ltd [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 11, 2004 9:15 AM Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous When I call a SIP user, the phone should ring in more than one extentions. Also more than one phone should be able to register with asterisk. Right now it is not the case. There is no issue here. You seem to be confused, that's all. A SIP account is a SIP account and an extension is an extension. You can assign an extension to an account (or to multiple accounts) and the tool for that is the dial command. However, there is no implicit assignment between an extension and an account and that is good so. This should not be changed because it would harm Asterisk's flexibility and manageability. This type of situations might be needed in call centres. Called 12345 |---(12345) Ringing |---(12345) Ringing |---(12345) Ringing As I said, you are confusing extensions with accounts. The first 12345 is an extension, the three (12345)s are accounts. Those are different layers, don't mix them up. You should always be able to distinguish between devices, even if they are assigned the same phone number. In fact, in a call centre you'd be using a call queue. It would be rather nonsensical for a call queue management to have to distinguish between multiple identical agents. Therefore, setting up multiple devices with the same account credentials is not a good idea, especially not in a call centre. Each device and each agent should have their own unique account credentials and assigning extensions to them should always be done through the dialplan and only the dialplan. Asterisk has been designed this way. It is a good design. It should NOT be changed nor undermined. You may want to do something like this ... [GLOBALS] A-GROUP = SIP/2001 SIP2002 SIP/2003 B-BROUP = SIP/jdoe SIP/dflint SIP/bsmith ... [Support
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry
Paul Mahler wrote: If you want to be able to more easily recognize what extension the traffic if for, you can add additional extensions to the 7960. For example, if you have two staff the admin monitors, add two additional extensions to the 7960. The admin can tell who is being called by the extension that rings. Paul Uh no. If I have a 7940 with 112 registered to it on two lines, and then an AA with a 7960 and 113 registered to two lines, I can NOT register 112 to the other 4 lines on the 7960. I could if I could register more than one endpoint. Which is the goal of this bounty. -- Daniel Jimenez djimenez[at]pobox[dot]com ___ 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 registry
Soren Rathje wrote: Eh... Sort of like shadow lines ??? Remember that Dial(SIP/1 H323/1 ZAP/1,[timeout],[options],[URL]) will dial all 3 extensions simultaneously (regardless of channel choice) and with a little tinkering in your dialplan you can even activate/deactivate this from the Manager Station Why make it harder than it really is ??? I believe this is exactly what they do when programming your regular (old world) PBX systems... -- Soren Explain to me how the AA user knows who's line is ringing? Hmm? To them, it's no different then a call directly to them. The bounty stands. -- Daniel Jimenez djimenez[at]pobox[dot]com ___ 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 registry
Kannaiyan Natesan wrote: I hope this helps. Since I feel this is a great feature, I will topup up to $100/- -.Kannaiyan http://www.goods2world.com -- Your Only VoIP Thank you, I updated the wiki with your $25 addition. -- Daniel Jimenez djimenez[at]pobox[dot]com ___ 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 registry
Paul Mahler wrote: If I have the requirement right, you could accomplish this by ringing the staff extension and the admin extension at the same time. The Dial command allows you to ring multiple extensions simultaneously. Paul Did you even read the bounty? Yes, Yes I know you could do all sorts of fun with the dialplan to produce a similar effect, but I still would like to be able to do this. That is an excerpt from my EMAIL and from the WIKI. -- Daniel Jimenez djimenez[at]pobox[dot]com ___ 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 11/07/2004 at 08:42 Paul Mahler wrote: You are confused about what a SIP session is and what a SIP session does. SIP, session initiation protocol, controls an RTP, real time protocol, session between two IP endpionts. The end points have to have unique IP addresses for the session to run. The unique SIP registration is how * finds a UNIQUE endpoint. Sorry, but this is irrelavant... SIP allows multiple endpoints to register with the same account details and will all ring when called. The fact that the rtp stream goes to the first endpoint to pick up (and respond) is what's important ie, if multiple devices are registered with the same account they will *all* be 'spoken' to... Asterisk currently does not support this behaviour. You don't want SIP to solve your problem, you want * to solve your problem. You are asking for this SIP feature because you are confused as to how SIP and * work, and how they work together. No, the idea is to get asterisk to act like a real sip proxy. The dialplan solution is a poor hack. You can easily fix your business problem with *, but not with mechanism you are asking for. You should spend your money on getting a copy of each of the two books that are now available and learn *. Then it will be clear to you that you don't really want what you are asking for. again, irrelavant - the whole beauty of the way SIP works is that I can add to the list of phones that get called by simply registering more phones with the same details. I don't need my users to mess with or make a support call to add to the dial plan. They can add and remove themselves. I'd also suggest adding something like registrationlimit=1 for those that do not want to support multiple client registrations, I'd also like to see the implementation of the q parameter... I'm all for this modification to SIP, although I'd probably want to see DTMF callerid implemented first :D Andy ___ 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
As Daniel Says, Bounty stands. I cannot explain to you anymore. I'm sorry. Please read more what SIP can do with SER. -Kannaiyan. - Original Message - From: Paul Mahler [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 11, 2004 4:42 PM Subject: RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous You are confused about what a SIP session is and what a SIP session does. SIP, session initiation protocol, controls an RTP, real time protocol, session between two IP endpionts. The end points have to have unique IP addresses for the session to run. The unique SIP registration is how * finds a UNIQUE endpoint. You don't want SIP to solve your problem, you want * to solve your problem. You are asking for this SIP feature because you are confused as to how SIP and * work, and how they work together. You can easily fix your business problem with *, but not with mechanism you are asking for. You should spend your money on getting a copy of each of the two books that are now available and learn *. Then it will be clear to you that you don't really want what you are asking for. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kannaiyan Natesan Sent: Sunday, July 11, 2004 1:15 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous I explained him a sample need. I don't think asterisk does whatever i want in sip. It is an good PBX. Please help me to understand. Anywhere am I wrong ? Or as you say is that SIP feature is written? -Kannaiyan. - Original Message - From: usedcanon [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 11, 2004 10:02 AM Subject: RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous I was going to keep out of this (was interesting to read, as I have dealt with simmillar situation) however I would like to add just this one commnet. Try to better understand asterisk than to throw about your money. What you want to do is perfectly possible with asterisk there is no need to add a new confusing feature. As for your bounty, donate it to the wiki ! :-) Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Kannaiyan Natesan Sent: 11 July 2004 09:51 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous I accept your views. I have a specific requirements, can you help to attain the same. In our business we have 25 employees handling customer service. I want to add or remove employees in the customer service so does the devices connected to it. I don't want to make any changes in the asterisk, and all I need is to plug in the VoIP Phone and start handling the customer service. I would like to do for as many employees as I want without any problems. Can you think of a better solution? -Kannaiyan. - Original Message - From: Sunrise Ltd [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 11, 2004 9:15 AM Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous When I call a SIP user, the phone should ring in more than one extentions. Also more than one phone should be able to register with asterisk. Right now it is not the case. There is no issue here. You seem to be confused, that's all. A SIP account is a SIP account and an extension is an extension. You can assign an extension to an account (or to multiple accounts) and the tool for that is the dial command. However, there is no implicit assignment between an extension and an account and that is good so. This should not be changed because it would harm Asterisk's flexibility and manageability. This type of situations might be needed in call centres. Called 12345 |---(12345) Ringing |---(12345) Ringing |---(12345) Ringing As I said, you are confusing extensions with accounts. The first 12345 is an extension, the three (12345)s are accounts. Those are different layers, don't mix them up. You should always be able to distinguish between devices, even if they are assigned the same phone number. In fact, in a call centre you'd be using a call queue. It would be rather nonsensical for a call queue management to have to distinguish between multiple identical agents. Therefore, setting up multiple devices with the same account credentials is not a good idea, especially not in a call centre. Each device and each agent should have their own unique account
RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
The whole point of a SIP registration is to identify a UNIQUE device. You CAN'T HAVE multiple devices registered as the same SIP device. That's WHY the last device that registers gets the traffic. This doesn't have ANYTHING TO DO WITH ASTERISK. This is a SIP issue, not an Asterisk issue. You should just be happy that Asterisk will do what you want, even if SIP won't. If you really, really want to do this, up the bounty to about $50,000 and get the SIP specification changed. Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Powell Sent: Sunday, July 11, 2004 9:57 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous On 11/07/2004 at 08:42 Paul Mahler wrote: You are confused about what a SIP session is and what a SIP session does. SIP, session initiation protocol, controls an RTP, real time protocol, session between two IP endpionts. The end points have to have unique IP addresses for the session to run. The unique SIP registration is how * finds a UNIQUE endpoint. Sorry, but this is irrelavant... SIP allows multiple endpoints to register with the same account details and will all ring when called. The fact that the rtp stream goes to the first endpoint to pick up (and respond) is what's important ie, if multiple devices are registered with the same account they will *all* be 'spoken' to... Asterisk currently does not support this behaviour. You don't want SIP to solve your problem, you want * to solve your problem. You are asking for this SIP feature because you are confused as to how SIP and * work, and how they work together. No, the idea is to get asterisk to act like a real sip proxy. The dialplan solution is a poor hack. You can easily fix your business problem with *, but not with mechanism you are asking for. You should spend your money on getting a copy of each of the two books that are now available and learn *. Then it will be clear to you that you don't really want what you are asking for. again, irrelavant - the whole beauty of the way SIP works is that I can add to the list of phones that get called by simply registering more phones with the same details. I don't need my users to mess with or make a support call to add to the dial plan. They can add and remove themselves. I'd also suggest adding something like registrationlimit=1 for those that do not want to support multiple client registrations, I'd also like to see the implementation of the q parameter... I'm all for this modification to SIP, although I'd probably want to see DTMF callerid implemented first :D Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
It's not what SIP does with SER, it's what SER does with SIP. Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kannaiyan Natesan Sent: Sunday, July 11, 2004 9:58 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous As Daniel Says, Bounty stands. I cannot explain to you anymore. I'm sorry. Please read more what SIP can do with SER. -Kannaiyan. - Original Message - From: Paul Mahler [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 11, 2004 4:42 PM Subject: RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous You are confused about what a SIP session is and what a SIP session does. SIP, session initiation protocol, controls an RTP, real time protocol, session between two IP endpionts. The end points have to have unique IP addresses for the session to run. The unique SIP registration is how * finds a UNIQUE endpoint. You don't want SIP to solve your problem, you want * to solve your problem. You are asking for this SIP feature because you are confused as to how SIP and * work, and how they work together. You can easily fix your business problem with *, but not with mechanism you are asking for. You should spend your money on getting a copy of each of the two books that are now available and learn *. Then it will be clear to you that you don't really want what you are asking for. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kannaiyan Natesan Sent: Sunday, July 11, 2004 1:15 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous I explained him a sample need. I don't think asterisk does whatever i want in sip. It is an good PBX. Please help me to understand. Anywhere am I wrong ? Or as you say is that SIP feature is written? -Kannaiyan. - Original Message - From: usedcanon [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 11, 2004 10:02 AM Subject: RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous I was going to keep out of this (was interesting to read, as I have dealt with simmillar situation) however I would like to add just this one commnet. Try to better understand asterisk than to throw about your money. What you want to do is perfectly possible with asterisk there is no need to add a new confusing feature. As for your bounty, donate it to the wiki ! :-) Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Kannaiyan Natesan Sent: 11 July 2004 09:51 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous I accept your views. I have a specific requirements, can you help to attain the same. In our business we have 25 employees handling customer service. I want to add or remove employees in the customer service so does the devices connected to it. I don't want to make any changes in the asterisk, and all I need is to plug in the VoIP Phone and start handling the customer service. I would like to do for as many employees as I want without any problems. Can you think of a better solution? -Kannaiyan. - Original Message - From: Sunrise Ltd [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 11, 2004 9:15 AM Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous When I call a SIP user, the phone should ring in more than one extentions. Also more than one phone should be able to register with asterisk. Right now it is not the case. There is no issue here. You seem to be confused, that's all. A SIP account is a SIP account and an extension is an extension. You can assign an extension to an account (or to multiple accounts) and the tool for that is the dial command. However, there is no implicit assignment between an extension and an account and that is good so. This should not be changed because it would harm Asterisk's flexibility and manageability. This type of situations might be needed in call centres. Called 12345 |---(12345) Ringing |---(12345) Ringing |---(12345) Ringing As I said, you
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry
Daniel Jimenez wrote: Soren Rathje wrote: Eh... Sort of like shadow lines ??? Remember that Dial(SIP/1 H323/1 ZAP/1,[timeout],[options],[URL]) will dial all 3 extensions simultaneously (regardless of channel choice) and with a little tinkering in your dialplan you can even activate/deactivate this from the Manager Station Why make it harder than it really is ??? I believe this is exactly what they do when programming your regular (old world) PBX systems... -- Soren Explain to me how the AA user knows who's line is ringing? Hmm? To them, it's no different then a call directly to them. The bounty stands. Hmm, Googling around a bit I think I understand what you are looking for... Asterisk is missing functionality similar to Cisco CallManagers m option on ephone configuration. The m option allows you to monitor a specific DN but you do not own it... Right ?? If so, Asterisk will have to handle two types of registers. One To: per DN and any number of Cc:'s... Question is if SIP supports this and as I read the RFC it does not. I wonder if this is built into the Asterisk SCCP stuff.. -- Soren ___ 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
Did you even read the RFC? Section 10.2.1 clearly talks about adding multiple bindings to the same address-of record. On Sun, 2004-07-11 at 12:31, Paul Mahler wrote: The whole point of a SIP registration is to identify a UNIQUE device. You CAN'T HAVE multiple devices registered as the same SIP device. That's WHY the last device that registers gets the traffic. This doesn't have ANYTHING TO DO WITH ASTERISK. This is a SIP issue, not an Asterisk issue. You should just be happy that Asterisk will do what you want, even if SIP won't. If you really, really want to do this, up the bounty to about $50,000 and get the SIP specification changed. Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training ___ 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 11/07/2004 at 12:31 Paul Mahler wrote: The whole point of a SIP registration is to identify a UNIQUE device. You CAN'T HAVE multiple devices registered as the same SIP device. That's WHY the last device that registers gets the traffic. WRONG! This doesn't have ANYTHING TO DO WITH ASTERISK. This is a SIP issue, not an Asterisk issue. You should just be happy that Asterisk will do what you want, even if SIP won't. WRONG! If you really, really want to do this, up the bounty to about $50,000 and get the SIP specification changed. This is TOTAL rubbish .. you clearly have no idea how SIP works I think I'll skip your book Andy ___ 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
Mike Machado wrote: On Sun, 2004-07-11 at 12:31, Paul Mahler wrote: The whole point of a SIP registration is to identify a UNIQUE device. You CAN'T HAVE multiple devices registered as the same SIP device. That's WHY the last device that registers gets the traffic. This doesn't have ANYTHING TO DO WITH ASTERISK. This is a SIP issue, not an Asterisk issue. You should just be happy that Asterisk will do what you want, even if SIP won't. If you really, really want to do this, up the bounty to about $50,000 and get the SIP specification changed. Did you even read the RFC? Section 10.2.1 clearly talks about adding multiple bindings to the same address-of record. Just to quote and save everybody the searching: Once a client has established bindings at a registrar, it MAY send subsequent registrations containing new bindings or modifications to existing bindings as necessary. The 2xx response to the REGISTER request will contain, in a Contact header field, a complete list of bindings that have been registered for this address-of-record at this registrar. Nick ___ 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
Paul Mahler wrote: You should spend your money on getting a copy of each of the two books that are now available and learn *. Then it will be clear to you that you don't really want what you are asking for. Shameless plug? It's offical you are trolling. -- Daniel Jimenez djimenez[at]pobox[dot]com ___ 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
Well, this is certainly getting exciting. Andy, I took your advice and re-read the RFP. Andy--I don't think you are a good candidate for a beginner's book on *, but if you send my your address, I'll send you a copy on me. :-) So, gentlemen, help me out here. The spec says: The Address of record is the SIP address that the registry knows the registrand. . . A Sip message is either a request from a client to a server or a response from a server to a client. A client uses the REGISTER method to register the address listed in the TO header field with a SIP Server. And as Nick so cogently pointed out Once a client has established bindings at a registrar, it MAY send subsequent registrations containing new bindings or modifications to existing bindings as necessary. The 2xx response to the REGISTER request will contain, in a Contact header field, a complete list of bindings that have been registered for this address-of-record at this registrar. I don't see how two different clients can register with a server as the same address of record. Doesn't the second registration from a new client change the address of record for the registered client? If the second client is trying the same registration as the first client, and it's the responsibility of the client to provide the complete list of bindings, how does the second client know the list of bindings for the first client that bound the registration? So isn't this the problem * has? The first client registers as the address of record, then the second client comes in with the same registration and becomes the address of record? Andy, I'm in your hands. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nicholas Bachmann Sent: Sunday, July 11, 2004 12:41 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous Mike Machado wrote: On Sun, 2004-07-11 at 12:31, Paul Mahler wrote: The whole point of a SIP registration is to identify a UNIQUE device. You CAN'T HAVE multiple devices registered as the same SIP device. That's WHY the last device that registers gets the traffic. This doesn't have ANYTHING TO DO WITH ASTERISK. This is a SIP issue, not an Asterisk issue. You should just be happy that Asterisk will do what you want, even if SIP won't. If you really, really want to do this, up the bounty to about $50,000 and get the SIP specification changed. Did you even read the RFC? Section 10.2.1 clearly talks about adding multiple bindings to the same address-of record. Just to quote and save everybody the searching: Once a client has established bindings at a registrar, it MAY send subsequent registrations containing new bindings or modifications to existing bindings as necessary. The 2xx response to the REGISTER request will contain, in a Contact header field, a complete list of bindings that have been registered for this address-of-record at this registrar. Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Asterisk bounty: SIP simultaneous registry
http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultaneous+registry From the WIKI: Contributions Manager: Daniel Jimenez (cuban) Bounty: $50 USD Date opened: July 10, 2004 Contributors: cuban ($50) Detail Yes, Yes I know you could do all sorts of fun with the dialplan to produce a similar effect, but I still would like to be able to do this. Plus it's easy money :). I have some users with a 7960 who are administrative assistants who monitor calls for 3 or 4 other people. It'd be nice to have two line instances for them, and one for the person(s) whom they assist. Contact me: djimenez at pobox.com if you're interested in making this happen. -- Daniel Jimenez djimenez[at]pobox[dot]com ___ 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 registry
http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultaneous+registry Updated, Allow a SIP device to register more than once so a single extension may exist in multiple locations. Upped total to $75. Daniel... Daniel Jimenez wrote: http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultaneous+registry From the WIKI: Contributions Manager: Daniel Jimenez (cuban) Bounty: $50 USD Date opened: July 10, 2004 Contributors: cuban ($50) Detail Yes, Yes I know you could do all sorts of fun with the dialplan to produce a similar effect, but I still would like to be able to do this. Plus it's easy money :). I have some users with a 7960 who are administrative assistants who monitor calls for 3 or 4 other people. It'd be nice to have two line instances for them, and one for the person(s) whom they assist. Contact me: djimenez at pobox.com if you're interested in making this happen. -- Daniel Jimenez djimenez[at]pobox[dot]com ___ 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 registry
I'm not sure I understand what you are trying to do. You have an administrative assistant and several other staff. You want the administrator to be able to take calls directed to the staff extensions? If I have the requirement right, you could accomplish this by ringing the staff extension and the admin extension at the same time. The Dial command allows you to ring multiple extensions simultaneously. If you want to be able to more easily recognize what extension the traffic if for, you can add additional extensions to the 7960. For example, if you have two staff the admin monitors, add two additional extensions to the 7960. The admin can tell who is being called by the extension that rings. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Jimenez Sent: Saturday, July 10, 2004 3:05 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+s imultaneous+registry Updated, Allow a SIP device to register more than once so a single extension may exist in multiple locations. Upped total to $75. Daniel... Daniel Jimenez wrote: http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultane ous+registry From the WIKI: Contributions Manager: Daniel Jimenez (cuban) Bounty: $50 USD Date opened: July 10, 2004 Contributors: cuban ($50) Detail Yes, Yes I know you could do all sorts of fun with the dialplan to produce a similar effect, but I still would like to be able to do this. Plus it's easy money :). I have some users with a 7960 who are administrative assistants who monitor calls for 3 or 4 other people. It'd be nice to have two line instances for them, and one for the person(s) whom they assist. Contact me: djimenez at pobox.com if you're interested in making this happen. -- Daniel Jimenez djimenez[at]pobox[dot]com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry
Paul, The question is very simple. When I call a SIP user, the phone should ring in more than one extentions. Also more than one phone should be able to register with asterisk. Right now it is not the case. The last phone which register will be receiving the calls. This type of situations might be needed in call centres. Called 12345 |---(12345) Ringing |---(12345) Ringing |---(12345) Ringing So you don't need to disturb asterisk when you add more devices to it to receive calls. Such facility is not available in asterisk at this moment. I hope this helps. Since I feel this is a great feature, I will topup up to $100/- -.Kannaiyan http://www.goods2world.com -- Your Only VoIP - Original Message - From: Paul Mahler [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 11, 2004 5:44 AM Subject: RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry I'm not sure I understand what you are trying to do. You have an administrative assistant and several other staff. You want the administrator to be able to take calls directed to the staff extensions? If I have the requirement right, you could accomplish this by ringing the staff extension and the admin extension at the same time. The Dial command allows you to ring multiple extensions simultaneously. If you want to be able to more easily recognize what extension the traffic if for, you can add additional extensions to the 7960. For example, if you have two staff the admin monitors, add two additional extensions to the 7960. The admin can tell who is being called by the extension that rings. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Jimenez Sent: Saturday, July 10, 2004 3:05 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+s imultaneous+registry Updated, Allow a SIP device to register more than once so a single extension may exist in multiple locations. Upped total to $75. Daniel... Daniel Jimenez wrote: http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultane ous+registry From the WIKI: Contributions Manager: Daniel Jimenez (cuban) Bounty: $50 USD Date opened: July 10, 2004 Contributors: cuban ($50) Detail Yes, Yes I know you could do all sorts of fun with the dialplan to produce a similar effect, but I still would like to be able to do this. Plus it's easy money :). I have some users with a 7960 who are administrative assistants who monitor calls for 3 or 4 other people. It'd be nice to have two line instances for them, and one for the person(s) whom they assist. Contact me: djimenez at pobox.com if you're interested in making this happen. -- Daniel Jimenez djimenez[at]pobox[dot]com ___ 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
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry
On Sun, 11 Jul 2004, Kannaiyan Natesan wrote: When I call a SIP user, the phone should ring in more than one extentions. Also more than one phone should be able to register with asterisk. Right now it is not the case. The last phone which register will be receiving the calls. This type of situations might be needed in call centres. I think I understand now what you're looking for. But under an arrangement like this, how will asterisk know when a phone which had registered from some IP has re-registered itself sometime later on a different IP? Such a situation could happen in a dhcp environment. Automatic time-outs may be able to avoid or minimize the impact of something like this. What other difficulties might come up? Although the idea does have appeal, it seems like the increased potential for problems outweighs any inconvenience incurred by modifying a line in extensions.conf. Greg ___ 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 registry
in other SIP proxy server, this can be done easily, i mean its default 1 or more phone could be registered at 1 number (12345) and resulting same effect as u ask SER (SIP Express Router, http://iptel.org/ser) can deal with this SER is a friend to asterisk, i think :), you can accept calls with SER and pass it to asterisk to process complex dialplan but if this feature implemented in asterisk alone, it would be nice *** REPLY SEPARATOR *** On 11/07/2004 at 6:00 Kannaiyan Natesan wrote: Paul, The question is very simple. When I call a SIP user, the phone should ring in more than one extentions. Also more than one phone should be able to register with asterisk. Right now it is not the case. The last phone which register will be receiving the calls. This type of situations might be needed in call centres. Called 12345 |---(12345) Ringing |---(12345) Ringing |---(12345) Ringing So you don't need to disturb asterisk when you add more devices to it to receive calls. Such facility is not available in asterisk at this moment. I hope this helps. Since I feel this is a great feature, I will topup up to $100/- -.Kannaiyan http://www.goods2world.com -- Your Only VoIP - Original Message - From: Paul Mahler [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 11, 2004 5:44 AM Subject: RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry I'm not sure I understand what you are trying to do. You have an administrative assistant and several other staff. You want the administrator to be able to take calls directed to the staff extensions? If I have the requirement right, you could accomplish this by ringing the staff extension and the admin extension at the same time. The Dial command allows you to ring multiple extensions simultaneously. If you want to be able to more easily recognize what extension the traffic if for, you can add additional extensions to the 7960. For example, if you have two staff the admin monitors, add two additional extensions to the 7960. The admin can tell who is being called by the extension that rings. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Jimenez Sent: Saturday, July 10, 2004 3:05 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+s imultaneous+registry Updated, Allow a SIP device to register more than once so a single extension may exist in multiple locations. Upped total to $75. Daniel... Daniel Jimenez wrote: http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultane ous+registry From the WIKI: Contributions Manager: Daniel Jimenez (cuban) Bounty: $50 USD Date opened: July 10, 2004 Contributors: cuban ($50) Detail Yes, Yes I know you could do all sorts of fun with the dialplan to produce a similar effect, but I still would like to be able to do this. Plus it's easy money :). I have some users with a 7960 who are administrative assistants who monitor calls for 3 or 4 other people. It'd be nice to have two line instances for them, and one for the person(s) whom they assist. Contact me: djimenez at pobox.com if you're interested in making this happen. -- Daniel Jimenez djimenez[at]pobox[dot]com ___ 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 http://sleepless.ngoprek.org VoIP Rakyat: (0921) 20006 ___ 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
