[asterisk-users] Presence Management - use of hint
Hi, I am working on Presence management on a SIP client. I have something working based on SUBSCRIBE / NOTIFY mechanism and Asterisk hints. I know that an other solution could be implemented using peer to peer SUBSCRIBE / PUBLISH mechanism. I would like to understand the advantage and drawback of each solution. My main concern is the case of a complex VOIP environment such as Asterisk server to server connection. As example if we have : [client A] -- [Asterisk 1] --- [Asterisk 2] ---[client B] Using hints and SUBSCRIBE / NOTIFY would it be possible for client A to be notified of client B presence modification ? If client A can call client B, it would make sense to have SUBSCRIBE / PUBLISH with peer to peer mechanism working. Am I right ? Thanks for your advice, Eloi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Presence Registration on the D40
So I'm working with our Digium D40's and we're not using DPMA. This video ( http://www.youtube.com/watch?v=zcuocp01pfM#t=35s ) shows presence information being displayed in the Contacts application. Obviously the video is showing DPMA in play. Is it possible to enable this functionality without it? Is this status information only available on higher-end Digium phones? In the contacts XML data, I am supplying the appropriate parameters: contact first_name=John last_name=Doe organization=Acme contact_type =sip account_id=123 subscribe_to=123 but I am not seeing the icons shown in the video at all. On the Asterisk CLI, I can run: etc*CLI core show hint 2003 2003@default : SIP/charrington_desk State:IdleWatchers 0 1 hint matching extension 2003 and Watchers is always 0 for all extensions. Is there a separate way I can test subscribing for presence information? I don't even know, at this point, if it's the phones or Asterisk. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Presence subscription from other pbx systems
Hi members, I have a question regarding presence in asterisk. I have two PBX systems and would like to connect them. After configuring each other as sip providers calls between users of the pbx systems are possible. Now I'm trying to implement presence between the systems. PBX1 sends dialog-event SUBSCRIBE messages to PBX2. Asterisk just answers 404 not found although user 410 exists. I think this is for security reasons. Is there an option to allow presence subscription from configured providers? Sincerely Jan PS: Here are sample sip messages: --- SIP read from UDP:10.99.10.2:5060 --- SUBSCRIBE sip:410@10.99.10.14;sipx-noroute=VoiceMail;sipx-userforward=false SIP/2.0 Record-Route: sip:10.99.10.2:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EWEhLMUd5%21f063cbdfa e9d680ffaa83f6db4234704 From: sip:sipXrls@10.99.10.1:51829;tag=XHK1Gy To: sip:410@10.99.10.14;sipx-noroute=VoiceMail;sipx-userforward=false Call-Id: eZwhSebwLCc187 Cseq: 2 SUBSCRIBE Contact: sip:10.99.10.1:51829;transport=udp;x-sipX-nonat Event: dialog Accept: application/dialog-info+xml Expires: 3153 Date: Mon, 13 Feb 2012 09:45:50 GMT Max-Forwards: 19 User-Agent: sipXecs/4.4.0 sipXecs/rls (Linux) Accept-Language: en Proxy-Authorization: Digest username=~~id~sipXrls, realm=voip.mydomain.local, nonce=3998fbca7da46e21895d383a16356f424f38dbce, uri=sip:410@10.99.10.14;sipx-noroute=VoiceMail;sipx-userforward=false, response=53ae73a9ce6a3a6acbe35deda3f731be, cnonce=a42sMg, qop=auth, nc=0001 Via: SIP/2.0/UDP 10.99.10.2;branch=z9hG4bK-XX-18ddpkccVUQr6IO02D7a9Q5x0A Via: SIP/2.0/UDP 10.99.10.1:51829;branch=z9hG4bK-XX-f75bT2Zlly8RPJBMDcOw5dyOxw Content-Length: 0 - --- (18 headers 0 lines) --- Creating new subscription Sending to 10.99.10.2:5060 (no NAT) list_route: hop: sip:10.99.10.2:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EWEhLMUd5%21f063cbdfa e9d680ffaa83f6db4234704 No matching peer for 'sipXrls' from '10.99.10.2:5060' Looking for 410 in public-direct-dial (domain 10.99.10.14) --- Transmitting (no NAT) to 10.99.10.2:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.99.10.2;branch=z9hG4bK-XX-18ddpkccVUQr6IO02D7a9Q5x0A;received=10.99.10.2 Via: SIP/2.0/UDP 10.99.10.1:51829;branch=z9hG4bK-XX-f75bT2Zlly8RPJBMDcOw5dyOxw From: sip:sipXrls@10.99.10.1:51829;tag=XHK1Gy To: sip:410@10.99.10.14;sipx-noroute=VoiceMail;sipx-userforward=false;tag=as73 d6e628 Call-ID: eZwhSebwLCc187 CSeq: 2 SUBSCRIBE Server: AskoziaPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Presence for channels other than SIP.
Hi How to get the presence status of channels than SIP like Phone,Dahdi ,gsm and etc. I have checked the DEVICE_STATE function in dialplan but it shows only SIP channels status may be IAX too ,for other type channels(Phone,Dahdi,gsm) it is not showing anything.,And I tried hint too then also same result. Is this feature available in asterisk ? Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] presence with polycom DND
You might have to look at writing a forward macro on the server that would be dialed by the DND button - that also changed the device status to busy(via the devstate app?). My guess is that it would be less than 10 lines of dialplan code, but maybe 1.6 only PaulH cfh wrote: hi, I have configured asterisk 1.4.21 to control the presence BLF (hint + watch buddy parameter) of Polycom phones (650,550,330) and it works good. But when I set the phones on Do Not Disturb (DND) on the server there arent sip notifications and the presence doesnt change. On the Polycom configuration I have try to use the server based DND option but i dont know how to use this with asterik. What can i do ? Are there some workaround to use the DND button and the BLF on asterisk? thanks cfh ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] presence with polycom DND
hi, I have configured asterisk 1.4.21 to control the presence BLF (hint + watch buddy parameter) of Polycom phones (650,550,330) and it works good. But when I set the phones on Do Not Disturb (DND) on the server there arent sip notifications and the presence doesnt change. On the Polycom configuration I have try to use the server based DND option but i dont know how to use this with asterik. What can i do ? Are there some workaround to use the DND button and the BLF on asterisk? thanks cfh ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Presence on Polycom 301 partially broke?
Hi all- Equipment: Xlite softphone Polycom 301 with SIP 2.1.1 and BootROM 3.2.3 Polycom 501 with SIP 2.1.1 and BootROM 3.2.3 Asterisk 1.4.2 SIP Trunk to FWD I wanted to post this problem as I haven't found it described in any of the past presence threads on here. I use an identical configs for a Polycom 501 and 301. (I actually unplug one when the other is in use). The only difference between the two is that the different mac.cfg and mac-directory.cfg have different MAC addresses. On the 501, presence works fine under the Buddy Status screen and when a Contact is put as a speed dial on a line key, the icon next to the line key changes correctly depending on the buddy's status. On the 301, presence works fine under the Buddy Status screen, but the icon for the Contact when put as a speed dial on a line key remains the dial pad icon no matter what the current status of the buddy is. *The buddies ARE added to the contact list with the buddy watch enabled. I have had a very experience Polycom/Asterisk person from the #asterisk IRC channel recreate the problem. I am wondering if anyone has seen this and found a fix for it. Or is it a known problem without a work around? I am doing the most basic config possible on the phones. From the default configuration, I change on the following: sip.cfg: voIpProt.server.1.address=192.168.0.34 feature.1.enabled=1 phone1.cfg: reg.1.address=station1 reg.1.label=101 sip.conf ;Polycom Phone [station1] disallow=all allow=ulaw allow=alaw type=peer context=internal host=dynamic dtmfmode=rfc2833 callerid=101 101 nat=no qualify=yes canreinvite=no notifyringing=yes notifyhold=yes call-limit=99 ;X-Lite softphone [station2] disallow=all allow=ulaw allow=alaw type=peer context=internal host=dynamic dtmfmode=rfc2833 callerid=102 102 nat=no qualify=yes canreinvite=no notifyringing=yes notifyhold=yes mailbox=1234 call-limit=99 extensions.conf [internal] exten = 101,hint,SIP/station1 exten = 101,1,Dial(SIP/station1) exten = 102,hint,SIP/station2 exten = 102,1,Dial(SIP/station2) exten = yy,1,Dial(SIP/[EMAIL PROTECTED]) ;(yy is really a known fwd number) Any thoughts? Thanks in advance! DR ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Presence issues with Got SUBSCRIBE for extensions without hint. Please add hint to s
Hello all, I have a number of Polycom phones 601's and 430's and I'm seeing: Got SUBSCRIBE for extensions without hint. Please add hint to s to context local-hints in the CLI over and over. I have: [local-hints] exten = 110,hint,SIP/110 exten = 111,hint,SIP/111 exten = 112,hint,SIP/112 exten = 113,hint,SIP/113 exten = 114,hint,SIP/114 The hints seem to be working, however why is it looking for a hint for s - should I define one? Polycom's are running 1.6.7, Asterisk is 1.2.9.1 Thanks in advance wulf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Presence issues with Got SUBSCRIBE for extensions without hint. Please add hint to s
Are you sure there are no VoIP Phone users with Eyebeam or even polycom requesting SUBSCRIBE for other extensions? It happened to me, that users on my network were adding Subscribe for PSTN numbers that aren't even extensions on my * server. On 12/29/06, Lorentz Hinrichsen [EMAIL PROTECTED] wrote: Hello all, I have a number of Polycom phones 601's and 430's and I'm seeing: Got SUBSCRIBE for extensions without hint. Please add hint to s to context local-hints in the CLI over and over. I have: [local-hints] exten = 110,hint,SIP/110 exten = 111,hint,SIP/111 exten = 112,hint,SIP/112 exten = 113,hint,SIP/113 exten = 114,hint,SIP/114 The hints seem to be working, however why is it looking for a hint for s - should I define one? Polycom's are running 1.6.7, Asterisk is 1.2.9.1 Thanks in advance wulf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Presence issues with Got SUBSCRIBE for extensions without hint. Please add hint to s
Yes, it appears that the Polycom is trying to subscribe to s - why? I've triple checked the directory xml file and it is only bw'ing 110,111,112,113,114 no other extensions. See the sip log below: -- SIP read from 192.168.1.134:5060: SUBSCRIBE sip:192.168.1.65:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.134;branch=z9hG4bKba9b690c844C2BE1 From: 113 sip:[EMAIL PROTECTED];tag=DAC0ED6D-27373C0E To: sip:192.168.1.65 CSeq: 1 SUBSCRIBE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0094 Max-Forwards: 70 Expires: 3600 Content-Length: 0 --- (13 headers 0 lines)--- Using latest SUBSCRIBE request as basis request Sending to 192.168.1.134 : 5060 (NAT) Transmitting (no NAT) to 192.168.1.134:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.134;branch=z9hG4bKba9b690c844C2BE1;received= 192.168.1.134 From: 113 sip:[EMAIL PROTECTED];tag=DAC0ED6D-27373C0E To: sip:192.168.1.65;tag=as37029a1e Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX A pbx*CLI llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:192.168.1.65:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=3b34afb0 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user '113' pbx*CLI -- SIP read from 192.168.1.134:5060: SUBSCRIBE sip:192.168.1.65:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.134;branch=z9hG4bK1a8ce17b31705644 From: 113 sip:[EMAIL PROTECTED];tag=DAC0ED6D-27373C0E To: sip:192.168.1.65 CSeq: 2 SUBSCRIBE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0094 Authorization: Digest username=113, realm=asterisk, nonce=3b34afb0, uri=sip:192.168.1.65:5060, response=bf28cd2382f065f3ab3502c0a98074f1, algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 --- (14 headers 0 lines)--- Found user '113' Looking for s in bella-out (domain 192.168.1.65) Scheduling destruction of call '[EMAIL PROTECTED]' in 361 ms Dec 29 08:32:32 ERROR[26486]: chan_sip.c:10988 handle_request_subscribe: Got SUBSCRIBE for extensions without hint. Please add hint to s in context bella-presence Transmitting (no NAT) to 192.168.1.134:5060: SIP/2.0 404 Not found Via: SIP/2.0/UDP 192.168.1.134;branch=z9hG4bK1a8ce17b31705644;received= 192.168.1.134 From: 113 sip:[EMAIL PROTECTED];tag=DAC0ED6D-27373C0E To: sip:192.168.1.65;tag=as37029a1e Call-ID: [EMAIL PROTECTED] CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 On 12/29/06, Marco Mouta [EMAIL PROTECTED] wrote: Are you sure there are no VoIP Phone users with Eyebeam or even polycom requesting SUBSCRIBE for other extensions? It happened to me, that users on my network were adding Subscribe for PSTN numbers that aren't even extensions on my * server. On 12/29/06, Lorentz Hinrichsen [EMAIL PROTECTED] wrote: Hello all, I have a number of Polycom phones 601's and 430's and I'm seeing: Got SUBSCRIBE for extensions without hint. Please add hint to s to context local-hints in the CLI over and over. I have: [local-hints] exten = 110,hint,SIP/110 exten = 111,hint,SIP/111 exten = 112,hint,SIP/112 exten = 113,hint,SIP/113 exten = 114,hint,SIP/114 The hints seem to be working, however why is it looking for a hint for s - should I define one? Polycom's are running 1.6.7, Asterisk is 1.2.9.1 Thanks in advance wulf ___ --Bandwidth and Colocation provided by Easynews.comhttp://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Presence-awareness in Asterisk
Hi,How would you monitor screensaver activity ?Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Presence-awareness in Asterisk
Or you can look at PHP-AGI; use the php to query mysql (probably more scalable than dialplan MYSQL) Take a look at http://www.jivesoftware.org/ - perhaps some way you can use that? rajeevOn 11/10/06, Andrea Spadaccini [EMAIL PROTECTED] wrote: Ciao Ondrej, That's why I was more thinking about mysql - it is already running on my * box and remote access is no problem. Question is, if I could do the same trick you did with Asterisk DB with Mysql.Of course you can. In asterisk-addons there's the app MYSQL(), that does exactly what you want.See http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL for moredetails.HTH,--Andrea Spadaccini Multimedia Technologies Institute s.r.l.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Presence-awareness in Asterisk
On Sun, Nov 12, 2006 at 12:51:27AM +0530, Rajeev Natarajan wrote: Or you can look at PHP-AGI; use the php to query mysql (probably more scalable than dialplan MYSQL) Running an external php script which will open a separate mysql connection, query it, close and be done is not exactly scalable. At least not more scalable then using mysql from the dialplan. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Presence-awareness in Asterisk
On 21:44, Sat 11 Nov 06, Tzafrir Cohen wrote: On Sun, Nov 12, 2006 at 12:51:27AM +0530, Rajeev Natarajan wrote: Or you can look at PHP-AGI; use the php to query mysql (probably more scalable than dialplan MYSQL) Running an external php script which will open a separate mysql connection, query it, close and be done is not exactly scalable. At least not more scalable then using mysql from the dialplan. unless you run it as fastagi in some kind of daemon that uses persistant connections. AFAIK asterisk does not support persistant connections. That's why I built a http server with php in persistant mode. Really scales ok (not really tested because I dont have that much of call volume yet) -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Presence-awareness in Asterisk
Hello all, I am just wondering - how can I implement presence awareness in Asterisk? I know there is the hint feature that might be useful (for someone) but it is not exactly what I am looking for. My idea is some fairly simple application running on user desktop and having just 3-4 buttons like - online - do not disturb - forward to my mobile and possibly also monitoring xscreensaver activity. This application could then communicate with the * server (via AGI or SQL database or something) and amend the dialplan accordingly. Does anyone implemented it somewhere? How can I achieve this? I am happy with just any hint pointing me to the right direction. Thanks, Ondrej ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Presence-awareness in Asterisk
Am Freitag, den 10.11.2006, 14:35 +0100 schrieb Ondrej Valousek:
Hello all,
I am just wondering - how can I implement presence awareness in Asterisk?
I know there is the hint feature that might be useful (for someone) but
it is not exactly what I am looking for.
My idea is some fairly simple application running on user desktop and
having just 3-4 buttons like
- online
- do not disturb
- forward to my mobile
and possibly also monitoring xscreensaver activity. This application
could then communicate with the * server (via AGI or SQL database or
something) and amend the dialplan accordingly.
Does anyone implemented it somewhere? How can I achieve this?
I am happy with just any hint pointing me to the right direction.
The implementation on the Asterisk side is quite easy.
Consider the case where you have
exten = 234,1,Dial(SIP/sip234)
Now you want to replace that with some kind of *-magic such that either
of the three options you mentioned can be selected.
exten = 234,1,GotoIf($[${DB(Status/${EXTEN})} = dnd]?10)
exten = 234,2,GotoIf($[${DB(Status/${EXTEN})} = away]?20)
exten = 234,3,Dial(SIP/sip234)
exten = 234,10,VoiceMail(b${EXTEN})
exten = 234,20,Dial( your mobile number)
(this is not beautiful, but you get the idea)
This way, any time someone calls the Asterisk database will be queried
for status information. You can put that information by hand from the
CLI ( database put Status 234 dnd ) or use some other means to set it. I
could imagine an Apache CGI script to do that, or you write a
proprietary (Windows,KDE,...) APP that runs in the user taskbar and is
able to somehow update the status in the Asterisk DB.
BTW you can set something in the asterisk DB from the shell with the
asterisk -rx database set. command.
HTH
Anselm
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Re: [asterisk-users] Presence-awareness in Asterisk
Hi Anselm,
Yes it looks promising.
somehow update the status in the Asterisk DB
and that's the problem - how can I access Asterisk DB remotely (in some
nice and elegant way)?
That's why I was more thinking about mysql - it is already running on my
* box and remote access is no problem.
Question is, if I could do the same trick you did with Asterisk DB with
Mysql.
Thanks!
Ondrej
P.S.
Apache cgi is a possibility, indeed.
Anselm Martin Hoffmeister wrote:
Am Freitag, den 10.11.2006, 14:35 +0100 schrieb Ondrej Valousek:
Hello all,
I am just wondering - how can I implement presence awareness in Asterisk?
I know there is the hint feature that might be useful (for someone) but
it is not exactly what I am looking for.
My idea is some fairly simple application running on user desktop and
having just 3-4 buttons like
- online
- do not disturb
- forward to my mobile
and possibly also monitoring xscreensaver activity. This application
could then communicate with the * server (via AGI or SQL database or
something) and amend the dialplan accordingly.
Does anyone implemented it somewhere? How can I achieve this?
I am happy with just any hint pointing me to the right direction.
The implementation on the Asterisk side is quite easy.
Consider the case where you have
exten = 234,1,Dial(SIP/sip234)
Now you want to replace that with some kind of *-magic such that either
of the three options you mentioned can be selected.
exten = 234,1,GotoIf($[${DB(Status/${EXTEN})} = dnd]?10)
exten = 234,2,GotoIf($[${DB(Status/${EXTEN})} = away]?20)
exten = 234,3,Dial(SIP/sip234)
exten = 234,10,VoiceMail(b${EXTEN})
exten = 234,20,Dial( your mobile number)
(this is not beautiful, but you get the idea)
This way, any time someone calls the Asterisk database will be queried
for status information. You can put that information by hand from the
CLI ( database put Status 234 dnd ) or use some other means to set it. I
could imagine an Apache CGI script to do that, or you write a
proprietary (Windows,KDE,...) APP that runs in the user taskbar and is
able to somehow update the status in the Asterisk DB.
BTW you can set something in the asterisk DB from the shell with the
asterisk -rx database set. command.
HTH
Anselm
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Re: [asterisk-users] Presence-awareness in Asterisk
Am Freitag, den 10.11.2006, 16:33 +0100 schrieb Ondrej Valousek: Hi Anselm, Yes it looks promising. somehow update the status in the Asterisk DB and that's the problem - how can I access Asterisk DB remotely (in some nice and elegant way)? That's why I was more thinking about mysql - it is already running on my * box and remote access is no problem. Question is, if I could do the same trick you did with Asterisk DB with Mysql. There I cannot help you. But - there is an Apache Manager API that can be used over the network: http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk+Manager+(AJAM) It seems to have support for a DBPut command, which is what you need here. HTH Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Presence-awareness in Asterisk
Ciao Ondrej, That's why I was more thinking about mysql - it is already running on my * box and remote access is no problem. Question is, if I could do the same trick you did with Asterisk DB with Mysql. Of course you can. In asterisk-addons there's the app MYSQL(), that does exactly what you want. See http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL for more details. HTH, -- Andrea Spadaccini Multimedia Technologies Institute s.r.l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Presence SUBSCRIBE/NOTIFY behaviour
I'd appreciate some feedback on the behaviour of some tests relating to presence SUBSCRIBE/NOTIFY. In the tests no NAT or proxies are involved. We have a client using a SIP stack accepting requests on one port (eg 5060) but handling responses on a 'temporary' port. In other words it sends a request on a port '', quotes port '' in the 'Via' header, and then handles the response to that request on port ''. What we're seeing is that * sends the notification requests to the '' port associated with the SUBSCRIBE request/response rather than 5060. Naturally this port is no longer open, so they don't get through. I'm interested in how * figures out the addressing for the notifications. Is it from the Via header on the SUBSCRIBE (although I thought this was just used for responses?) or does it assume that the source of the subscription is the target for the notifications, or is it through some other means? Is it expected behaviour? OPTIONS requests generated by * are targetted at 5060 so they get through without any problem. Any light you could shed on this would be helpful - at this stage I'm just trying to establish what the correct behaviour is. Thanks, Shaun Bailey ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Presence SUBSCRIBE/NOTIFY behaviour
18 aug 2006 kl. 16.04 skrev Shaun Bailey: I'd appreciate some feedback on the behaviour of some tests relating to presence SUBSCRIBE/NOTIFY. In the tests no NAT or proxies are involved. We have a client using a SIP stack accepting requests on one port (eg 5060) but handling responses on a 'temporary' port. In other words it sends a request on a port '', quotes port '' in the 'Via' header, and then handles the response to that request on port ''. What we're seeing is that * sends the notification requests to the '' port associated with the SUBSCRIBE request/response rather than 5060. Naturally this port is no longer open, so they don't get through. I'm interested in how * figures out the addressing for the notifications. Is it from the Via header on the SUBSCRIBE (although I thought this was just used for responses?) or does it assume that the source of the subscription is the target for the notifications, or is it through some other means? Is it expected behaviour? OPTIONS requests generated by * are targetted at 5060 so they get through without any problem. Any light you could shed on this would be helpful - at this stage I'm just trying to establish what the correct behaviour is. That is a very good question that I can't answer. We are sending the NOTIFY as transactions within the existing dialog, the SUBSCRIBE, which means that the via header int he SUBSCRIBE applies. Whether this is the right or wrong way, is something we have to find out from the RFCs - a task that is not always easy. /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Presence support on GrandStream GXP-2000
[EMAIL PROTECTED] wrote: It does with the latest BETA firmware. But it dosn't seem to work to well. It stops working and the phones have to be rebooted. works good, as long as asterisk doesn't get restarted. then you need to reboot the phone. it's a bug. http://www.voip-info.org/wiki/view/GXP-2000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Presence support on GrandStream GXP-2000
trixter aka Bret McDanel wrote: I havent looked, I am sure that its there somewhere on grandstreams site but where is the latest beta located? all info can be found on http://www.voip-info.org/wiki/view/GXP-2000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Presence support on GrandStream GXP-2000
Same here. I believe there's some funkiness (spelling?) with lights staying on when calls are transferred to another extension. Rebooting the phone and/or asterisk is required for me sometimes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Bowyer Sent: Tuesday, January 10, 2006 1:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Presence support on GrandStream GXP-2000 On 10/01/06, Richard Smith [EMAIL PROTECTED] wrote: Hi folks, Just a quick question. Does the GrandStream GXP-2000 phone support presence (hints)? Yes - with the latest beta firmware (1.0.1.13). Working well for me in a SOHO environment. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Presence support on GrandStream GXP-2000
Hi folks, Just a quick question. Does the GrandStream GXP-2000 phone support presence (hints)? Cheers, Richard. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Presence support on GrandStream GXP-2000
It does with the latest BETA firmware. But it dosn't seem to work to well. It stops working and the phones have to be rebooted. --- Richard Smith [EMAIL PROTECTED] wrote: Hi folks, Just a quick question. Does the GrandStream GXP-2000 phone support presence (hints)? Cheers, Richard. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Yahoo! DSL Something to write home about. Just $16.99/mo. or less. dsl.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Presence support on GrandStream GXP-2000
On 10/01/06, Richard Smith [EMAIL PROTECTED] wrote: Hi folks, Just a quick question. Does the GrandStream GXP-2000 phone support presence (hints)? Yes - with the latest beta firmware (1.0.1.13). Working well for me in a SOHO environment. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Presence support on GrandStream GXP-2000
On Tue, 2006-01-10 at 07:21 +, Peter Bowyer wrote: On 10/01/06, Richard Smith [EMAIL PROTECTED] wrote: Hi folks, Just a quick question. Does the GrandStream GXP-2000 phone support presence (hints)? Yes - with the latest beta firmware (1.0.1.13). Working well for me in a SOHO environment. I havent looked, I am sure that its there somewhere on grandstreams site but where is the latest beta located? -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Presence + Eyebeam + Asterisk 1.2
Hi, anyone managed to get a Presence Agent configuration with Asterisk 1.2 and X-Ten Eyebeam working. I believe this should be paritally supported now in 1.2 ? regards Mark___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Presence + Eyebeam + Asterisk 1.2
Don't waste your time asterisk does not support presence --- Mark van Kerkwyk [EMAIL PROTECTED] a écrit : Hi, anyone managed to get a Presence Agent configuration with Asterisk 1.2 and X-Ten Eyebeam working. I believe this should be paritally supported now in 1.2 ? regards Mark ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] presence and Asterisk crash
Hi all. Ive got Asterisk CVS Head running on Fedora Core 3. It has been running for 4 months with no particular problem. Recently I tried to enable presence. On dialplan I added hint extensions for all my SIP users and on my Eyebeam clients (v. 1.1 3008q) I set Peer-to-Peer presence mode. Presence works right, but when an incoming or outogoing call is answered, Asterisk crashes with the following message: Ouch ... error while writing audio data: : Broken pipe Segmentation fault I tried to restart Asterisk many times but it always stop with this message. As I disable presence (on Eyebeam clients, not even in Asterisk dial plan) Asterisk stays on. Is this a bug or do I miss something with presence? Thank you, _fangi_ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] presence and Asterisk crash
Francesco Angi wrote: Hi all. I’ve got Asterisk CVS Head running on Fedora Core 3. It has been running for 4 months with no particular problem. Recently I tried to enable presence. On dialplan I added hint extensions for all my SIP users and on my Eyebeam clients (v. 1.1 3008q) I set Peer-to-Peer presence mode. Presence works right, but when an incoming or outogoing call is answered, Asterisk crashes with the following message: Ouch ... error while writing audio data: : Broken pipe Segmentation fault I tried to restart Asterisk many times but it always stop with this message. As I disable presence (on Eyebeam clients, not even in Asterisk dial plan) Asterisk stays on. Is this a bug or do I miss something with presence? There is a bug report open on this in the bug tracker. Collect some data, add a backtrace and SIP debug output up to the point where it crashes and you will help us track that bug down and kill it. Thank you for your assistance. /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] presence settings and Eyebeam
Olle E. Johansson wrote:
Vahan Yerkanian wrote:
What is the proper way of adding hints to multiple extensions?
In my case extensions are the same as the sip usernames, while as per
http://www.voip-info.org/tiki-index.php?page=Asterisk%20presence
exten = 1234,hint,SIP/1234 works,
exten = _1,hint,SIP/${EXTEN} doesn't. Not sure if I can even use
${EXTEN} here...
Any hints?
File a bug report if it does not work. I think it would be a good idea
if it works, even though I usually don't recommend using the extension
as the peer name. ;-)
/O
Can you elaborate on why you don't recommend using the extension as the
peer name?
Cheers,
Kevin
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[Asterisk-Users] Presence Fully Supported?
I've seen lots about presence and Polycom phones recently. I've got one here for evaluation but noticed other phones only seem to appear busy when they initiate a call. If they receive a call, they still show as available. Is this a config problem on my part, or is that as far as presence is working right now? Thanks! Trev ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Presence Fully Supported?
The latest CVS versions support Presence a lot better. PaulH - Original Message - From: Trevor Peirce [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, September 12, 2005 8:57 AM Subject: [Asterisk-Users] Presence Fully Supported? I've seen lots about presence and Polycom phones recently. I've got one here for evaluation but noticed other phones only seem to appear busy when they initiate a call. If they receive a call, they still show as available. Is this a config problem on my part, or is that as far as presence is working right now? Thanks! Trev ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] presence settings and Eyebeam
What is the proper way of adding hints to multiple extensions?
In my case extensions are the same as the sip usernames, while as per
http://www.voip-info.org/tiki-index.php?page=Asterisk%20presence
exten = 1234,hint,SIP/1234 works,
exten = _1,hint,SIP/${EXTEN} doesn't. Not sure if I can even use
${EXTEN} here...
Any hints?
Vahan
begin:vcard
fn:Vahan Yerkanian
n:Yerkanian;Vahan
org:ARMINCO Global Telecommunications;Head, Research Development dept.
adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia
email;internet:[EMAIL PROTECTED]
x-mozilla-html:FALSE
url:http://www.arminco.com/
version:2.1
end:vcard
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Re: [Asterisk-Users] presence settings and Eyebeam
Vahan Yerkanian wrote:
What is the proper way of adding hints to multiple extensions?
In my case extensions are the same as the sip usernames, while as per
http://www.voip-info.org/tiki-index.php?page=Asterisk%20presence
exten = 1234,hint,SIP/1234 works,
exten = _1,hint,SIP/${EXTEN} doesn't. Not sure if I can even use
${EXTEN} here...
Any hints?
File a bug report if it does not work. I think it would be a good idea
if it works, even though I usually don't recommend using the extension
as the peer name. ;-)
/O
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Re: [Asterisk-Users] presence settings and Eyebeam
Done. Not sure if picked categories under SIP Mantis correct but here it is: http://bugs.digium.com/view.php?id=5149 VY Olle E. Johansson wrote: File a bug report if it does not work. I think it would be a good idea if it works, even though I usually don't recommend using the extension as the peer name. ;-) /O begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] presence in cvs head - how does one map extension to sip user?
Hello, I found, that in CVS Head, in chan_sip.c, there's some support of asterisk. I have one question -- how does it map extensions to sip user names? When my client subscribes to other extensions' presence, they see all users online, but it may be because of voicemail fallback. Is there a way to map extension to some sip user's presence? Any ideas are welcome. Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] presence in cvs head - how does one map extension to sip user?
Juraj Bednar wrote: Hello, I found, that in CVS Head, in chan_sip.c, there's some support of asterisk. I have one question -- how does it map extensions to sip user names? When my client subscribes to other extensions' presence, they see all users online, but it may be because of voicemail fallback. Is there a way to map extension to some sip user's presence? Yes, there are. Check the hint priority in your extensions.conf.sample in the source directory. Basically you connect an extension to one or several devices by entering a hint: exten = 500,hint,SIP/juraj /Olle --- Astricon 2005 - http://www.astricon.net/2005/ October 12-14, Anaheim, California, USA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] presence in cvs head - how does one map extension to sip user?
Hello, I found, that in CVS Head, in chan_sip.c, there's some support of asterisk. I have one question -- how does it map extensions to sip user names? When my client subscribes to other extensions' presence, they see all users online, but it may be because of voicemail fallback. Is there a way to map extension to some sip user's presence? Yes, there are. Check the hint priority in your extensions.conf.sample in the source directory. Basically you connect an extension to one or several devices by entering a hint: exten = 500,hint,SIP/juraj /Olle again, thank you very much for explaining this. I added this piece of information to the voip-info.org wiki, as many people have been asking this on -users list before and there was a lack of information. If anything on the page is not correct, feel free to edit: http://www.voip-info.org/tiki-index.php?page=Asterisk+presence Best wishes, Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] presence and IM again, want to develop a workinghack
Hi, -Original Message- I personally don't think it's a good idea to implement it in chan_sip. One reason for this is that user1 wants msn, user2 wants jabber, user3 wants icq, user4 wants gadugadu etc etc. Are you gonna implement all this ? That is, if you mean Instant Messaging in SIP ;) Forgive me if I'm wrong... You actually need a little of both. For PC interfaces, a universal messaging would be nice. There are SIP devices which support SIP messaging though, and they will not be able to deal with much else. Some form of integration in chan_sip will make sense... Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] presence and IM again, want to develop a working hack
Hello, I was again asked to try to add support for presence (SUBSCRIBE/NOTIFY) and IM using SIMPLE. I have few questions: a.) are there any, at least partial projects, patches, anything, that at least partly implements presence and/or IM to chan_sip? I don't care about presence on other channels, I have one SIP client per user. I've read this list's archive several times and found lots of wonderful proposals, which try to convince asking users, what needs to be done to support this well (multichannel, multiple phones per user, ...), mainly saying, that without very difficult reworking of internals, it would not be supported. What I really need is to hack it into chan_sip.c. I need the support of other channels and applications (f.e. MeetMe), but where I really care about presence and IM is SIP. So, any project, hack, patch, anything, that would allow me to go further with this would be greatly appreciated. I found this page in Russian: http://www.asterisk-support.ru/forums/development/53843189454 that somehow deals with the problem. I tried babelfish translation, (http://babelfish.altavista.com/babelfish/trurl_pagecontent?lp=ru_entrurl=http%3a%2f%2fwww.asterisk-support.ru%2fforums%2fdevelopment%2f53843189454) but I was not able to find out, if it really at least partially solves this problem, but as far as I understand it, Windows Messanger makes use of Subscribe/Notify, so this should be it. b.) Anyone has a partial solution using SER (which supports presence and IM) as a frontend, but routing all calls through Asterisk? Can this be done? I need the calls to go via Asterisk (I don't mean only sip notifications, but also the data, so I have canreinvite=no). So basically, SER would be a registrar proxy to Asterisk, which would do the authentication. The only thing, that SER would do would be to handle presence and IM and pass everything else on to Asterisk (as far as I know, SER can't pass traffic through it. I need the data to pass through the SIP server, since machines in my network topology don't see each other, it's a star with Asterisk in centre -- quite poetic indeed:). Any ideas, pointers to similiar configurations, ... are welcome. c.) If there is no solution to start with, is it possible to implement it only to chan_sip? I'm not familiar with Asterisk source code at all. Where are the places to look (in chan_sip.c) which are best to hook this code. Again, any code, hints, etc. about the structure of the source code are really welcome. Doing this in a clean way (although it's a hack) so it can be reused by community as much as possible is my intent. If anyone wants to help with the project by donating coder's time, mail me off the list. I hope I'll be able to support presence for hardphones and Xten's eyeBeam softphone in a few days with your help. Best wishes and thanks for any replies, Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] presence and IM again, want to develop a working hack
Juraj Bednar wrote: Hello, I was again asked to try to add support for presence (SUBSCRIBE/NOTIFY) and IM using SIMPLE. I have few questions: a.) are there any, at least partial projects, patches, anything, that at least partly implements presence and/or IM to chan_sip? I don't care about presence on other channels, I have one SIP client per user. I've read this list's archive several times and found lots of wonderful proposals, which try to convince asking users, what needs to be done to support this well (multichannel, multiple phones per user, ...), mainly saying, that without very difficult reworking of internals, it would not be supported. What I really need is to hack it into chan_sip.c. I need the support of other channels and applications (f.e. MeetMe), but where I really care about presence and IM is SIP. So, any project, hack, patch, anything, that would allow me to go further with this would be greatly appreciated. I found this page in Russian: http://www.asterisk-support.ru/forums/development/53843189454 that somehow deals with the problem. I tried babelfish translation, (http://babelfish.altavista.com/babelfish/trurl_pagecontent?lp=ru_entrurl=http%3a%2f%2fwww.asterisk-support.ru%2fforums%2fdevelopment%2f53843189454) but I was not able to find out, if it really at least partially solves this problem, but as far as I understand it, Windows Messanger makes use of Subscribe/Notify, so this should be it. b.) Anyone has a partial solution using SER (which supports presence and IM) as a frontend, but routing all calls through Asterisk? Can this be done? I need the calls to go via Asterisk (I don't mean only sip notifications, but also the data, so I have canreinvite=no). So basically, SER would be a registrar proxy to Asterisk, which would do the authentication. The only thing, that SER would do would be to handle presence and IM and pass everything else on to Asterisk (as far as I know, SER can't pass traffic through it. I need the data to pass through the SIP server, since machines in my network topology don't see each other, it's a star with Asterisk in centre -- quite poetic indeed:). Any ideas, pointers to similiar configurations, ... are welcome. c.) If there is no solution to start with, is it possible to implement it only to chan_sip? I'm not familiar with Asterisk source code at all. Where are the places to look (in chan_sip.c) which are best to hook this code. Again, any code, hints, etc. about the structure of the source code are really welcome. Doing this in a clean way (although it's a hack) so it can be reused by community as much as possible is my intent. If anyone wants to help with the project by donating coder's time, mail me off the list. I hope I'll be able to support presence for hardphones and Xten's eyeBeam softphone in a few days with your help. Best wishes and thanks for any replies, Juraj. Hi, Why not use the manager interface to poll if the sip device is logged in? You can make a script that puts your jabber client on/offline based on the manager output. I personally don't think it's a good idea to implement it in chan_sip. One reason for this is that user1 wants msn, user2 wants jabber, user3 wants icq, user4 wants gadugadu etc etc. Are you gonna implement all this ? That is, if you mean Instant Messaging in SIP ;) Forgive me if I'm wrong... Michiel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: SV: [Asterisk-Users] Presence and IM?
Hello all! First of all, thank you for all suggestions. As suggested, FOP does show who's online, but it's not really what I'm looking for. As said before, there's possibilities within the SIP protocol to have presence indication (using SIMPLE?) and that's what I would like to use. Not there yet, but imagine a small department with five staff members, all equipped with laptops. Some of them are constantly on travel. With the ability to use presence, any staff member will be able to tell right away who's online and who's not, without going through an operator or opening up FOP through their web browser. I'd consider this an advantage. Regards, Bjorn -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Juraj Bednar Sendt: 19. juni 2005 19:19 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: SV: [Asterisk-Users] Presence and IM? Hello, We have been running Asterisk for about a month now and one of the things I miss the most is the ability to se who's online and available and who's not. Surely, there's the manager interface, but unless you'd want to install extra software on each client computer, this is not a good option. Then there's the presence / IM function in SIP. Since we're only using SIP clients, this could easily solve some of our problems. However, I cannot get this to work with Asterisk using Eyebeam. Is this because the function is simply not supported within Asterisk? If lack of support is the case, anyone knows if this feature is to be implemented in the near future? I have the same problem and am seeking for few weeks for a suitable solution... If you'll figure out something, please let me know. We use Polycom IP500s which when used with a 'hint' in extensions.conf, can show presence via the 'buddy list.' could you post a snippet? Does this hint work as a presence agent and sending notifies? Does IM work with asterisk? I would really like to support presence in Asterisk with Eyebeam as a client. SIP Express Router has this ability, but it's not a good choice either. Maybe it would be possible to port this feature from SER? Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: SV: [Asterisk-Users] Presence and IM?
Tried this, but unfortunately no luck. Bjorn -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av [EMAIL PROTECTED] Sendt: 18. juni 2005 03:05 Til: asterisk-users@lists.digium.com Emne: Re: SV: [Asterisk-Users] Presence and IM? We have been running Asterisk for about a month now and one of the things I miss the most is the ability to se whos online and available and whos not. Surely, theres the manager interface, but unless youd want to install extra software on each client computer, this is not a good option. Then theres the presence / IM function in SIP. Since were only using SIP clients, this could easily solve some of our problems. However, I cannot get this to work with Asterisk using Eyebeam. Is this because the function is simply not supported within Asterisk? If lack of support is the case, anyone knows if this feature is to be implemented in the near future? We use Polycom IP500s which when used with a 'hint' in extensions.conf, can show presence via the 'buddy list.' -Ron ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: SV: [Asterisk-Users] Presence and IM?
Asterisk does support the presence support in SIP, at least in CVS head. It takes some fiddling to make it work. Below you'll find a link that will hopefully help you. As for SIMPLE it's actually SIP's messaging protocol, which Asterisk does not ... quite ... support. http://www.voip-info.org/tiki-index.php?page=Asterisk%20standard%20extension s - Details the hint priority, what it is - what it does and gives a link to a scenario where a SNOM phone was used. Please note that the source code mentioned on the See Also link is already present in Asterisk. As well, the context where you put your hints needs to be accessible to the SIP phone. - Joshua Colp. On 6/21/05 9:58 AM, Bjørn Ove Kristiansen [EMAIL PROTECTED] wrote: Hello all! First of all, thank you for all suggestions. As suggested, FOP does show who's online, but it's not really what I'm looking for. As said before, there's possibilities within the SIP protocol to have presence indication (using SIMPLE?) and that's what I would like to use. Not there yet, but imagine a small department with five staff members, all equipped with laptops. Some of them are constantly on travel. With the ability to use presence, any staff member will be able to tell right away who's online and who's not, without going through an operator or opening up FOP through their web browser. I'd consider this an advantage. Regards, Bjorn -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Juraj Bednar Sendt: 19. juni 2005 19:19 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: SV: [Asterisk-Users] Presence and IM? Hello, We have been running Asterisk for about a month now and one of the things I miss the most is the ability to se who's online and available and who's not. Surely, there's the manager interface, but unless you'd want to install extra software on each client computer, this is not a good option. Then there's the presence / IM function in SIP. Since we're only using SIP clients, this could easily solve some of our problems. However, I cannot get this to work with Asterisk using Eyebeam. Is this because the function is simply not supported within Asterisk? If lack of support is the case, anyone knows if this feature is to be implemented in the near future? I have the same problem and am seeking for few weeks for a suitable solution... If you'll figure out something, please let me know. We use Polycom IP500s which when used with a 'hint' in extensions.conf, can show presence via the 'buddy list.' could you post a snippet? Does this hint work as a presence agent and sending notifies? Does IM work with asterisk? I would really like to support presence in Asterisk with Eyebeam as a client. SIP Express Router has this ability, but it's not a good choice either. Maybe it would be possible to port this feature from SER? Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: SV: SV: [Asterisk-Users] Presence and IM?
Hello again! As said below, this was already tried. However, it doesn't work. I should add that I've gotten the hint function to work through the management interface, so the syntax should be right. But for presence it's not fully compatible with SIP devices and software such as EyeBeam. Bjorn -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Joshua Colp Sendt: 21. juni 2005 15:16 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: SV: SV: [Asterisk-Users] Presence and IM? Asterisk does support the presence support in SIP, at least in CVS head. It takes some fiddling to make it work. Below you'll find a link that will hopefully help you. As for SIMPLE it's actually SIP's messaging protocol, which Asterisk does not ... quite ... support. http://www.voip-info.org/tiki-index.php?page=Asterisk%20standard%20extension s - Details the hint priority, what it is - what it does and gives a link to a scenario where a SNOM phone was used. Please note that the source code mentioned on the See Also link is already present in Asterisk. As well, the context where you put your hints needs to be accessible to the SIP phone. - Joshua Colp. On 6/21/05 9:58 AM, Bjørn Ove Kristiansen [EMAIL PROTECTED] wrote: Hello all! First of all, thank you for all suggestions. As suggested, FOP does show who's online, but it's not really what I'm looking for. As said before, there's possibilities within the SIP protocol to have presence indication (using SIMPLE?) and that's what I would like to use. Not there yet, but imagine a small department with five staff members, all equipped with laptops. Some of them are constantly on travel. With the ability to use presence, any staff member will be able to tell right away who's online and who's not, without going through an operator or opening up FOP through their web browser. I'd consider this an advantage. Regards, Bjorn -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Juraj Bednar Sendt: 19. juni 2005 19:19 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: SV: [Asterisk-Users] Presence and IM? Hello, We have been running Asterisk for about a month now and one of the things I miss the most is the ability to se who's online and available and who's not. Surely, there's the manager interface, but unless you'd want to install extra software on each client computer, this is not a good option. Then there's the presence / IM function in SIP. Since we're only using SIP clients, this could easily solve some of our problems. However, I cannot get this to work with Asterisk using Eyebeam. Is this because the function is simply not supported within Asterisk? If lack of support is the case, anyone knows if this feature is to be implemented in the near future? I have the same problem and am seeking for few weeks for a suitable solution... If you'll figure out something, please let me know. We use Polycom IP500s which when used with a 'hint' in extensions.conf, can show presence via the 'buddy list.' could you post a snippet? Does this hint work as a presence agent and sending notifies? Does IM work with asterisk? I would really like to support presence in Asterisk with Eyebeam as a client. SIP Express Router has this ability, but it's not a good choice either. Maybe it would be possible to port this feature from SER? Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: SV: [Asterisk-Users] Presence and IM?
Hello all! First of all, thank you for all suggestions. As suggested, FOP does show who's online, but it's not really what I'm looking for. As said before, there's possibilities within the SIP protocol to have presence indication (using SIMPLE?) and that's what I would like to use. Not there yet, but imagine a small department with five staff members, all equipped with laptops. Some of them are constantly on travel. With the ability to use presence, any staff member will be able to tell right away who's online and who's not, without going through an operator or opening up FOP through their web browser. I'd consider this an advantage. Regards, Bjorn -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Juraj Bednar Sendt: 19. juni 2005 19:19 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: SV: [Asterisk-Users] Presence and IM? Hello, We have been running Asterisk for about a month now and one of the things I miss the most is the ability to se who's online and available and who's not. Surely, there's the manager interface, but unless you'd want to install extra software on each client computer, this is not a good option. Then there's the presence / IM function in SIP. Since we're only using SIP clients, this could easily solve some of our problems. However, I cannot get this to work with Asterisk using Eyebeam. Is this because the function is simply not supported within Asterisk? If lack of support is the case, anyone knows if this feature is to be implemented in the near future? I have the same problem and am seeking for few weeks for a suitable solution... If you'll figure out something, please let me know. We use Polycom IP500s which when used with a 'hint' in extensions.conf, can show presence via the 'buddy list.' could you post a snippet? Does this hint work as a presence agent and sending notifies? Does IM work with asterisk? I would really like to support presence in Asterisk with Eyebeam as a client. SIP Express Router has this ability, but it's not a good choice either. Maybe it would be possible to port this feature from SER? Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: SV: [Asterisk-Users] Presence and IM?
My client (Entourage) did a word wrap... Couldn't fit it all on one line. http://www.voip-info.org/tiki-index.php?page=Asterisk%20standard%20extension Try that ^^^ - Joshua Colp. On 6/21/05 11:04 AM, Anton Krall [EMAIL PROTECTED] wrote: Page cannot be found |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Joshua Colp |Sent: Martes, 21 de Junio de 2005 08:16 a.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: SV: SV: [Asterisk-Users] Presence and IM? | |Asterisk does support the presence support in SIP, at least in |CVS head. It takes some fiddling to make it work. Below you'll |find a link that will hopefully help you. As for SIMPLE it's |actually SIP's messaging protocol, which Asterisk does not ... |quite ... support. | |http://www.voip-info.org/tiki-index.php?page=Asterisk%20standar d%20extension |s - Details the hint priority, what it is - what it does and |gives a link to a scenario where a SNOM phone was used. | |Please note that the source code mentioned on the See Also |link is already present in Asterisk. As well, the context |where you put your hints needs to be accessible to the SIP phone. | |- Joshua Colp. | | |On 6/21/05 9:58 AM, Bjørn Ove Kristiansen [EMAIL PROTECTED] wrote: | | Hello all! | | First of all, thank you for all suggestions. As suggested, FOP does | show who's online, but it's not really what I'm looking for. As said | before, there's possibilities within the SIP protocol to |have presence | indication (using SIMPLE?) and that's what I would like to use. | | Not there yet, but imagine a small department with five |staff members, | all equipped with laptops. Some of them are constantly on |travel. With | the ability to use presence, any staff member will be able to tell | right away who's online and who's not, without going through an | operator or opening up FOP through their web browser. I'd |consider this an advantage. | | Regards, | Bjorn | | -Opprinnelig melding- | Fra: [EMAIL PROTECTED] | [mailto:[EMAIL PROTECTED] På vegne av Juraj | Bednar | Sendt: 19. juni 2005 19:19 | Til: Asterisk Users Mailing List - Non-Commercial Discussion | Emne: Re: SV: [Asterisk-Users] Presence and IM? | | Hello, | | We have been running Asterisk for about a month now and one of the | things I miss the most is the ability to se who's online and | available and who's not. Surely, there's the manager |interface, but | unless you'd want to install extra software on each client |computer, | this is not a good option. | | Then there's the presence / IM function in SIP. Since we're only | using SIP clients, this could easily solve some of our problems. | However, I cannot get this to work with Asterisk using Eyebeam. Is | this because the function is simply not supported within Asterisk? | | If lack of support is the case, anyone knows if this feature is to | be implemented in the near future? | | I have the same problem and am seeking for few weeks for a suitable | solution... If you'll figure out something, please let me know. | | We use Polycom IP500s which when used with a 'hint' in | extensions.conf, can show presence via the 'buddy list.' | | could you post a snippet? | | Does this hint work as a presence agent and sending notifies? Does | IM work with asterisk? | | I would really like to support presence in Asterisk with |Eyebeam as a | client. SIP Express Router has this ability, but it's not a good | choice either. Maybe it would be possible to port this feature from | SER? | | | Juraj. | ___ | Asterisk-Users mailing list | Asterisk-Users@lists.digium.com | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | | | ___ | Asterisk-Users mailing list | Asterisk-Users@lists.digium.com | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit
RE: SV: SV: [Asterisk-Users] Presence and IM?
Can this hint system be used for gxp2000 phones or just for snoms? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Joshua Colp |Sent: Martes, 21 de Junio de 2005 10:03 a.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: SV: SV: [Asterisk-Users] Presence and IM? | |My client (Entourage) did a word wrap... Couldn't fit it all |on one line. |http://www.voip-info.org/tiki-index.php?page=Asterisk%20standar d%20extension |Try that ^^^ | |- Joshua Colp. | | |On 6/21/05 11:04 AM, Anton Krall |[EMAIL PROTECTED] wrote: | | Page cannot be found | | |-Original Message- | |From: [EMAIL PROTECTED] | |[mailto:[EMAIL PROTECTED] On Behalf Of | |Joshua Colp | |Sent: Martes, 21 de Junio de 2005 08:16 a.m. | |To: Asterisk Users Mailing List - Non-Commercial Discussion | |Subject: Re: SV: SV: [Asterisk-Users] Presence and IM? | | | |Asterisk does support the presence support in SIP, at least in | |CVS head. It takes some fiddling to make it work. Below you'll | |find a link that will hopefully help you. As for SIMPLE it's | |actually SIP's messaging protocol, which Asterisk does not ... | |quite ... support. | | | |http://www.voip-info.org/tiki-index.php?page=Asterisk%20standar | d%20extension | |s - Details the hint priority, what it is - what it does and | |gives a link to a scenario where a SNOM phone was used. | | | |Please note that the source code mentioned on the See Also | |link is already present in Asterisk. As well, the context | |where you put your hints needs to be accessible to the SIP phone. | | | |- Joshua Colp. | | | | | |On 6/21/05 9:58 AM, Bjørn Ove Kristiansen [EMAIL PROTECTED] wrote: | | | | Hello all! | | | | First of all, thank you for all suggestions. As |suggested, FOP does | | show who's online, but it's not really what I'm looking |for. As said | | before, there's possibilities within the SIP protocol to | |have presence | | indication (using SIMPLE?) and that's what I would like to use. | | | | Not there yet, but imagine a small department with five | |staff members, | | all equipped with laptops. Some of them are constantly on | |travel. With | | the ability to use presence, any staff member will be able to tell | | right away who's online and who's not, without going through an | | operator or opening up FOP through their web browser. I'd | |consider this an advantage. | | | | Regards, | | Bjorn | | | | -Opprinnelig melding- | | Fra: [EMAIL PROTECTED] | | [mailto:[EMAIL PROTECTED] På vegne av Juraj | | Bednar | | Sendt: 19. juni 2005 19:19 | | Til: Asterisk Users Mailing List - Non-Commercial Discussion | | Emne: Re: SV: [Asterisk-Users] Presence and IM? | | | | Hello, | | | | We have been running Asterisk for about a month now and |one of the | | things I miss the most is the ability to se who's online and | | available and who's not. Surely, there's the manager | |interface, but | | unless you'd want to install extra software on each client | |computer, | | this is not a good option. | | | | Then there's the presence / IM function in SIP. Since we're only | | using SIP clients, this could easily solve some of our problems. | | However, I cannot get this to work with Asterisk using |Eyebeam. Is | | this because the function is simply not supported |within Asterisk? | | | | If lack of support is the case, anyone knows if this |feature is to | | be implemented in the near future? | | | | I have the same problem and am seeking for few weeks for |a suitable | | solution... If you'll figure out something, please let me know. | | | | We use Polycom IP500s which when used with a 'hint' in | | extensions.conf, can show presence via the 'buddy list.' | | | | could you post a snippet? | | | | Does this hint work as a presence agent and sending |notifies? Does | | IM work with asterisk? | | | | I would really like to support presence in Asterisk with | |Eyebeam as a | | client. SIP Express Router has this ability, but it's not a good | | choice either. Maybe it would be possible to port this |feature from | | SER? | | | | | | Juraj. | | ___ | | Asterisk-Users mailing list | | Asterisk-Users@lists.digium.com | | http://lists.digium.com/mailman/listinfo/asterisk-users | | To UNSUBSCRIBE or update options visit: | |http://lists.digium.com/mailman/listinfo/asterisk-users | | | | | | ___ | | Asterisk-Users mailing list | | Asterisk-Users@lists.digium.com | | http://lists.digium.com/mailman/listinfo/asterisk-users | | To UNSUBSCRIBE or update options visit: | |http://lists.digium.com/mailman/listinfo/asterisk-users | | | | | |___ | |Asterisk-Users mailing list | |Asterisk-Users@lists.digium.com | |http://lists.digium.com/mailman/listinfo/asterisk-users | |To UNSUBSCRIBE or update options visit: | | http://lists.digium.com/mailman/listinfo
Re: SV: SV: [Asterisk-Users] Presence and IM?
On 12:00, Tue 21 Jun 05, Anton Krall wrote: Can this hint system be used for gxp2000 phones or just for snoms? Right now the gxp2000 doesn't support it. I heard rumours on this list that Grandstream is planning this feature for some future firmware. I'm waiting for it as well. Till that time I'll stick to snoms :) -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: SV: SV: [Asterisk-Users] Presence and IM?
Page cannot be found |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Joshua Colp |Sent: Martes, 21 de Junio de 2005 08:16 a.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: SV: SV: [Asterisk-Users] Presence and IM? | |Asterisk does support the presence support in SIP, at least in |CVS head. It takes some fiddling to make it work. Below you'll |find a link that will hopefully help you. As for SIMPLE it's |actually SIP's messaging protocol, which Asterisk does not ... |quite ... support. | |http://www.voip-info.org/tiki-index.php?page=Asterisk%20standar d%20extension |s - Details the hint priority, what it is - what it does and |gives a link to a scenario where a SNOM phone was used. | |Please note that the source code mentioned on the See Also |link is already present in Asterisk. As well, the context |where you put your hints needs to be accessible to the SIP phone. | |- Joshua Colp. | | |On 6/21/05 9:58 AM, Bjørn Ove Kristiansen [EMAIL PROTECTED] wrote: | | Hello all! | | First of all, thank you for all suggestions. As suggested, FOP does | show who's online, but it's not really what I'm looking for. As said | before, there's possibilities within the SIP protocol to |have presence | indication (using SIMPLE?) and that's what I would like to use. | | Not there yet, but imagine a small department with five |staff members, | all equipped with laptops. Some of them are constantly on |travel. With | the ability to use presence, any staff member will be able to tell | right away who's online and who's not, without going through an | operator or opening up FOP through their web browser. I'd |consider this an advantage. | | Regards, | Bjorn | | -Opprinnelig melding- | Fra: [EMAIL PROTECTED] | [mailto:[EMAIL PROTECTED] På vegne av Juraj | Bednar | Sendt: 19. juni 2005 19:19 | Til: Asterisk Users Mailing List - Non-Commercial Discussion | Emne: Re: SV: [Asterisk-Users] Presence and IM? | | Hello, | | We have been running Asterisk for about a month now and one of the | things I miss the most is the ability to se who's online and | available and who's not. Surely, there's the manager |interface, but | unless you'd want to install extra software on each client |computer, | this is not a good option. | | Then there's the presence / IM function in SIP. Since we're only | using SIP clients, this could easily solve some of our problems. | However, I cannot get this to work with Asterisk using Eyebeam. Is | this because the function is simply not supported within Asterisk? | | If lack of support is the case, anyone knows if this feature is to | be implemented in the near future? | | I have the same problem and am seeking for few weeks for a suitable | solution... If you'll figure out something, please let me know. | | We use Polycom IP500s which when used with a 'hint' in | extensions.conf, can show presence via the 'buddy list.' | | could you post a snippet? | | Does this hint work as a presence agent and sending notifies? Does | IM work with asterisk? | | I would really like to support presence in Asterisk with |Eyebeam as a | client. SIP Express Router has this ability, but it's not a good | choice either. Maybe it would be possible to port this feature from | SER? | | | Juraj. | ___ | Asterisk-Users mailing list | Asterisk-Users@lists.digium.com | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | | | ___ | Asterisk-Users mailing list | Asterisk-Users@lists.digium.com | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] Presence and IM?
Hello, We have been running Asterisk for about a month now and one of the things I miss the most is the ability to se who's online and available and who's not. Surely, there's the manager interface, but unless you'd want to install extra software on each client computer, this is not a good option. Then there's the presence / IM function in SIP. Since we're only using SIP clients, this could easily solve some of our problems. However, I cannot get this to work with Asterisk using Eyebeam. Is this because the function is simply not supported within Asterisk? If lack of support is the case, anyone knows if this feature is to be implemented in the near future? I have the same problem and am seeking for few weeks for a suitable solution... If you'll figure out something, please let me know. We use Polycom IP500s which when used with a 'hint' in extensions.conf, can show presence via the 'buddy list.' could you post a snippet? Does this hint work as a presence agent and sending notifies? Does IM work with asterisk? I would really like to support presence in Asterisk with Eyebeam as a client. SIP Express Router has this ability, but it's not a good choice either. Maybe it would be possible to port this feature from SER? Juraj. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Presence and IM?
We have been running Asterisk for about a month now and one of the things I miss the most is the ability to se whos online and available and whos not. Surely, theres the manager interface, but unless youd want to install extra software on each client computer, this is not a good option. Then theres the presence / IM function in SIP. Since were only using SIP clients, this could easily solve some of our problems. However, I cannot get this to work with Asterisk using Eyebeam. Is this because the function is simply not supported within Asterisk? If lack of support is the case, anyone knows if this feature is to be implemented in the near future? Regards, Bjorn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Presence and IM?
Hi Bjorn, Maybe it could be done as some form of check against call forward to voicemail etc. Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bjorn Sent: Friday, 17 June 2005 11:51 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Presence and IM? We have been running Asterisk for about a month now and one of the things I miss the most is the ability to se whos online and available and whos not. Surely, theres the manager interface, but unless youd want to install extra software on each client computer, this is not a good option. Then theres the presence / IM function in SIP. Since were only using SIP clients, this could easily solve some of our problems. However, I cannot get this to work with Asterisk using Eyebeam. Is this because the function is simply not supported within Asterisk? If lack of support is the case, anyone knows if this feature is to be implemented in the near future? Regards, Bjorn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Presence and IM?
Maybe, but that would not have been a reliable way of handling it, as not all users would necessarily use voicemail. Besides, I would think that this feature is supported by several SIP devices (it has to do with messaging), so it would be better If Asterisk supported this feature by default, no hacking needed. Regards, Bjorn Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] P vegne av Dean Collins Sendt: 17. juni 2005 18:11 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: RE: [Asterisk-Users] Presence and IM? Hi Bjorn, Maybe it could be done as some form of check against call forward to voicemail etc. Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bjorn Sent: Friday, 17 June 2005 11:51 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Presence and IM? We have been running Asterisk for about a month now and one of the things I miss the most is the ability to se whos online and available and whos not. Surely, theres the manager interface, but unless youd want to install extra software on each client computer, this is not a good option. Then theres the presence / IM function in SIP. Since were only using SIP clients, this could easily solve some of our problems. However, I cannot get this to work with Asterisk using Eyebeam. Is this because the function is simply not supported within Asterisk? If lack of support is the case, anyone knows if this feature is to be implemented in the near future? Regards, Bjorn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] Presence and IM?
Bjorn, Take a look at the Asterisk Management Portal at http://sourceforge.net/projects/amportal It has a flash-based panel that will give you what you are looking for. Bryan M. Johns One Ring Networks 300 West Wieuca Road, NE Building One Suite 205 Atlanta, GA 30342 404.303.9900 x: 104 http://www.oneringnetworks.com On Fri, 2005-06-17 at 20:33 +0200, Bjorn wrote: Maybe, but that would not have been a reliable way of handling it, as not all users would necessarily use voicemail. Besides, I would think that this feature is supported by several SIP devices (it has to do with messaging), so it would be better If Asterisk supported this feature by default, no hacking needed. Regards, Bjorn Fra:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] P vegne av Dean Collins Sendt: 17. juni 2005 18:11 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: RE: [Asterisk-Users] Presence and IM? Hi Bjorn, Maybe it could be done as some form of check against call forward to voicemail etc. Dean From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bjorn Sent: Friday, 17 June 2005 11:51 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Presence and IM? We have been running Asterisk for about a month now and one of the things I miss the most is the ability to se whos online and available and whos not. Surely, theres the manager interface, but unless youd want to install extra software on each client computer, this is not a good option. Then theres the presence / IM function in SIP. Since were only using SIP clients, this could easily solve some of our problems. However, I cannot get this to work with Asterisk using Eyebeam. Is this because the function is simply not supported within Asterisk? If lack of support is the case, anyone knows if this feature is to be implemented in the near future? Regards, Bjorn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] Presence and IM?
Bjorn, Take a look at the Asterisk Management Portal at http://sourceforge.net/projects/amportal It has a flash-based panel that will give you what you are looking for. Bryan M. Johns One Ring Networks 300 West Wieuca Road, NE Building One Suite 205 Atlanta, GA 30342 404.303.9900 x: 104 http://www.oneringnetworks.com On Fri, 2005-06-17 at 20:33 +0200, Bjorn wrote: Maybe, but that would not have been a reliable way of handling it, as not all users would necessarily use voicemail. Besides, I would think that this feature is supported by several SIP devices (it has to do with messaging), so it would be better If Asterisk supported this feature by default, no hacking needed. Regards, Bjorn Fra:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] P vegne av Dean Collins Sendt: 17. juni 2005 18:11 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: RE: [Asterisk-Users] Presence and IM? Hi Bjorn, Maybe it could be done as some form of check against call forward to voicemail etc. Dean From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bjorn Sent: Friday, 17 June 2005 11:51 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Presence and IM? We have been running Asterisk for about a month now and one of the things I miss the most is the ability to se whos online and available and whos not. Surely, theres the manager interface, but unless youd want to install extra software on each client computer, this is not a good option. Then theres the presence / IM function in SIP. Since were only using SIP clients, this could easily solve some of our problems. However, I cannot get this to work with Asterisk using Eyebeam. Is this because the function is simply not supported within Asterisk? If lack of support is the case, anyone knows if this feature is to be implemented in the near future? Regards, Bjorn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] Presence and IM?
Take a look at the Asterisk Management Portal at http://sourceforge.net/projects/amportal It has a flash-based panel that will give you what you are looking for. No need to install AMP to get this, just install FOP : http://www.asternic.org/ hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] Presence and IM?
We have been running Asterisk for about a month now and one of the things I miss the most is the ability to se whos online and available and whos not. Surely, theres the manager interface, but unless youd want to install extra software on each client computer, this is not a good option. Then theres the presence / IM function in SIP. Since were only using SIP clients, this could easily solve some of our problems. However, I cannot get this to work with Asterisk using Eyebeam. Is this because the function is simply not supported within Asterisk? If lack of support is the case, anyone knows if this feature is to be implemented in the near future? We use Polycom IP500s which when used with a 'hint' in extensions.conf, can show presence via the 'buddy list.' -Ron ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] presence and video conference
Hello, I would like to ask, if there's presence support in Asterisk and how to make it work with Xten's Eyebeam client. I tried searching all the possible documentation, google, but I found only a note, that there's a module in SER, that supports the feature. Is there also support in asterisk? Any pointer to documentation describing this is welcome. One more question -- is there a video conferencing support (like meetme, but for video)? I also found some development pages, but without code... Thanks, Juraj Bednar. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] presence and video conference
Hi Juraj, I have been trying for some time to fund video conferencing support and have offered a personal bounty of several thousands of dollars in order to get it developed. So far 5 people have contacted me but apart from one point to point solution I'm still waiting. In the interim I have purchased www.smiletiger.com software for my video conferencing requirements. Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Juraj Bednar Sent: Monday, 13 June 2005 10:21 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] presence and video conference Hello, I would like to ask, if there's presence support in Asterisk and how to make it work with Xten's Eyebeam client. I tried searching all the possible documentation, google, but I found only a note, that there's a module in SER, that supports the feature. Is there also support in asterisk? Any pointer to documentation describing this is welcome. One more question -- is there a video conferencing support (like meetme, but for video)? I also found some development pages, but without code... Thanks, Juraj Bednar. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Presence
This is a wonderful idea. I like the app_im concept a lot. I'd make a few additions though. Like the ability to have festival read the Away message as the Voicemail message. I'd definitely change my voicemail more often if I could do it by changing my Jabber away message. I would suggest that Jabber would be a more effective first target though, as with it comes the ability to hit AIM/ICQ/MSN/Yahoo/etc users via a simple proxy. Having just the one implementation would simplify things. Chris Tooley On Wed, 2004-04-07 at 21:29 -0400, John Todd wrote: At 8:29 PM -0400 on 4/7/04, Shad Mortazavi wrote: I have to agree. A large number of people are looking for this feature. I have written a web script that can show Agent logged into the system. I think integration/gateway between Asterisk and Jabber would be a amazingly wonderful product. There is always MSN. Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Netural Bay Sydney My idea for AIM/Jabber/Yahoo integration is below. Comments and/or programmers are welcome to have at it, and to expand on my ideas. I have mentioned this to several programmers who expressed an interest, but I'm sure that lack of time and funding has kept them from starting on the project, if it indeed is worthwhile. This is a kludge to some degree, but it uses _already existing_ presence tools to extend Asterisk's functionality, without needing to modify any client software or hardware. This is really a one-way presence idea at the moment. There are the glimmerings of two-way presence (see the activewhen keyword) but this is mostly for CTI outbound notices from an * server to humans upon some events defined by the administrator. I would see this most typically used either as a screenpop on an inbound or outbound call, or perhaps as a voicemail notification tool if the administrator is clever enough to embed a URL into the string for the instant message text. Phase 1: Create a set of programs for Asterisk which allows status checking of a particular username on a particular instant messaging system (availability, idle time) and also allows for transmission of instant messages from Asterisk to other users on those instant messaging systems (one-way.) The first systems that come to mind would be AOL's AIM and Yahoo. Phase 2: Add additional instant message systems: maybe Jabber, MSN. Allow examination of user's header line (in AOL, at least) and pass that through the app_imstatus return codes. This would allow me to specify mobile: as the first digits of my status, thus a GotoIf would be able to know that it should send calls to my cell phone. Or when I get to work, and shift between my home account (home: hello, I'm home) to work (work: at my desk) then the system will automatically forward calls appropriately. This might be easy enough to do in Phase 1, but I'm uncertain. Future paths: A true presence application for telephony in a large scale method is lacking today. It may be the case that this could be done by creating a custom telephony presence presentation application that is based on an existing (or multiple existing) chat protocols. As an example, it is possible that I might be able to make my status message on AIM change from avail/sip:[EMAIL PROTECTED] to busy/sip:[EMAIL PROTECTED] every time I pick up the phone; that could be done programmatically by Asterisk. Then, my friends who have the custom telephony presence application would see the little icon beside pinkycaruthers go from green to flashing orange. As soon as I went back to non-busy, they could just click on my icon, and two things would happen: a password-protected message would get fired off to THEIR phone system and extension from the presence application on their desktop, which in turn would be received by an asterisk-aware application on their Asterisk server, which in turn would create a spool call to MY phone system from the SIP URI that I included in my Status message. Presto! We have minimalist call routing, presence, and click-to-dial - we're just missing the little app to do it on Windows, MacOS, Linux, Java, whatever. The core message transport protocols all exist; it's just a matter of layers on top of them. Using standard telephony URI's, we could not just do this with SIP, but with tel, h323, iax2, anything - it's not limited to VoIP. ; im.conf ; ; Use of this file implies that you have an active account with one or more ; instant messaging services, and that you probably use an account that is ; dedicated to your Asterisk server so it knows what's going on. You may ; need to ensure that any other user id's that you expect to receive messages ; are filtered in such a way that the messages from your Asterisk-specific ; account are permitted
[Asterisk-Users] Presence (was FW: pda skype)
Dean Collins just sent out a message a second ago (responding to an earlier posting regarding the new Skype PDA client). He said: Presence based information is the biggest 'seller' in the IP PBX market at the moment, being able to tell what/where a person is certainly driving a lot of sales through my door. I would like to take a moment to second his message. Every presentation at the VON show concurs with his opinion. PRESENCE IS LIKELY THE MOST IMPORTANT SINGLE ADDITION TO ASTERISK THAT COULD BE MADE AT THIS TIME. I know there are people out there working on or waiting for all kinds of features. And I understand that all of them are important in one way or another. But THE feature that turns up time and again on RFPs for VoIP phone systems is PRESENCE. Like it or not, managers like to know where their people are. Friends like knowing where their friends are. Everybody likes being able to communicate how they want, when they want. This is the next step beyond the follow-me/find me applications. Beyond the basics of VoIP. And this step is driving people to deploy VoIP systems (including Asterisk). Perhaps the gory details of implementation are best reserved for the Developer list, but I think everybody out there can comment on the ideas: HOW DO YOU SEE PRESENCE INTEGRATING WITH ASTERISK? Thanks, Steve Steven Sokol Owner/Manager Sokol Associates, LLC Phone: 816.822.1807 IaxTel: 700.613.9004 Web:http://www.sokol-associates.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] Presence
Title: [Asterisk-Users] Presence I have to agree. A large number of people are looking for this feature. I have written a web script that can show Agent logged into the system. I think integration/gateway between Asterisk and Jabber would be a amazingly wonderful product. There is always MSN. Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Netural Bay Sydney
Re: [Asterisk-Users] Presence
Shad Mortazavi wrote: I think integration/gateway between Asterisk and Jabber would be a amazingly wonderful product. firefly, while not 100% bug free I think it has this feature, although I haven't played with it enough to work out how to show someone as being online... -- 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 ___ 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] Presence
They modified iax to include the presence packet but only works on their customized firefly network. I was thinking along the lines of a software app for those of us who use hardware phones but still want to keep TXT chat and presence and perhaps integrated into 1 of the iax soft phones as well to provide a full solution. On Wed, 2004-04-07 at 20:40, Duane wrote: Shad Mortazavi wrote: I think integration/gateway between Asterisk and Jabber would be a amazingly wonderful product. firefly, while not 100% bug free I think it has this feature, although I haven't played with it enough to work out how to show someone as being online... ___ 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] Presence
William Suffill wrote: They modified iax to include the presence packet but only works on their customized firefly network. I was thinking along the lines of a software app for those of us who use hardware phones but still want to keep TXT chat and presence and perhaps integrated into 1 of the iax soft phones as well to provide a full solution. Question is then, how well does their system work? Already have an IAX2 compatible soft phone with that stuff in it, why not make use of the fact and just work out what needs to be sent to their client... -- 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 ___ 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] Presence
At 8:29 PM -0400 on 4/7/04, Shad Mortazavi wrote: I have to agree. A large number of people are looking for this feature. I have written a web script that can show Agent logged into the system. I think integration/gateway between Asterisk and Jabber would be a amazingly wonderful product. There is always MSN. Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Netural Bay Sydney My idea for AIM/Jabber/Yahoo integration is below. Comments and/or programmers are welcome to have at it, and to expand on my ideas. I have mentioned this to several programmers who expressed an interest, but I'm sure that lack of time and funding has kept them from starting on the project, if it indeed is worthwhile. This is a kludge to some degree, but it uses _already existing_ presence tools to extend Asterisk's functionality, without needing to modify any client software or hardware. This is really a one-way presence idea at the moment. There are the glimmerings of two-way presence (see the activewhen keyword) but this is mostly for CTI outbound notices from an * server to humans upon some events defined by the administrator. I would see this most typically used either as a screenpop on an inbound or outbound call, or perhaps as a voicemail notification tool if the administrator is clever enough to embed a URL into the string for the instant message text. Phase 1: Create a set of programs for Asterisk which allows status checking of a particular username on a particular instant messaging system (availability, idle time) and also allows for transmission of instant messages from Asterisk to other users on those instant messaging systems (one-way.) The first systems that come to mind would be AOL's AIM and Yahoo. Phase 2: Add additional instant message systems: maybe Jabber, MSN. Allow examination of user's header line (in AOL, at least) and pass that through the app_imstatus return codes. This would allow me to specify mobile: as the first digits of my status, thus a GotoIf would be able to know that it should send calls to my cell phone. Or when I get to work, and shift between my home account (home: hello, I'm home) to work (work: at my desk) then the system will automatically forward calls appropriately. This might be easy enough to do in Phase 1, but I'm uncertain. Future paths: A true presence application for telephony in a large scale method is lacking today. It may be the case that this could be done by creating a custom telephony presence presentation application that is based on an existing (or multiple existing) chat protocols. As an example, it is possible that I might be able to make my status message on AIM change from avail/sip:[EMAIL PROTECTED] to busy/sip:[EMAIL PROTECTED] every time I pick up the phone; that could be done programmatically by Asterisk. Then, my friends who have the custom telephony presence application would see the little icon beside pinkycaruthers go from green to flashing orange. As soon as I went back to non-busy, they could just click on my icon, and two things would happen: a password-protected message would get fired off to THEIR phone system and extension from the presence application on their desktop, which in turn would be received by an asterisk-aware application on their Asterisk server, which in turn would create a spool call to MY phone system from the SIP URI that I included in my Status message. Presto! We have minimalist call routing, presence, and click-to-dial - we're just missing the little app to do it on Windows, MacOS, Linux, Java, whatever. The core message transport protocols all exist; it's just a matter of layers on top of them. Using standard telephony URI's, we could not just do this with SIP, but with tel, h323, iax2, anything - it's not limited to VoIP. ; im.conf ; ; Use of this file implies that you have an active account with one or more ; instant messaging services, and that you probably use an account that is ; dedicated to your Asterisk server so it knows what's going on. You may ; need to ensure that any other user id's that you expect to receive messages ; are filtered in such a way that the messages from your Asterisk-specific ; account are permitted through. ; ; username= username of the user on this particular messaging system. ; secret=password for the username ; type= type of connection this is. Each messaging system uses it's own protocols, ; so we need to specify which one of the protocols we're using for this particular ; channel. Current choices are: ; aim - the AOL OSCAR protocol ; yahoo - the Yahoo protocol ; statusmessage= Sets the status message for the user on the chat server. Visible to other users. ; activewhen= perform a login only when this channel type is valid or logged in. This is ; reduce
Re: [Asterisk-Users] Presence
Duane wrote: William Suffill wrote: They modified iax to include the presence packet but only works on their customized firefly network. I was thinking along the lines of a software app for those of us who use hardware phones but still want to keep TXT chat and presence and perhaps integrated into 1 of the iax soft phones as well to provide a full solution. Question is then, how well does their system work? Already have an IAX2 compatible soft phone with that stuff in it, why not make use of the fact and just work out what needs to be sent to their client... The protocol is quite simple, it's all text messages. S for subscribe to a user's events, T for send a text message I was half way through discussing this with Mark and more specifically adding it to IAX (along with some other cool stuff). Unforunately, I was told to do another project asap but that'll be released next week (stay tuned). My main concern with IAX were you don't know when someone goes offline until their reg expires - no acceptable in presence. Our solution was to keep the registration session open. Keeping the registration session open actually helps everything else fall in place, you can just send messages over that session without requiring setting up a channel, auth and tear down for each message. -Adam ___ 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] Presence
I'm not familiar with the protocol used in Firefly. If that was known then it would be possible to add the functionality to * so anyone can have the simple presences by dialing extensions in their dial plan or crafted packets at a software level. Jabber is already deployed in my organization so I would lean toward integration to that standard as well. On Wed, 2004-04-07 at 21:05, Duane wrote: William Suffill wrote: They modified iax to include the presence packet but only works on their customized firefly network. I was thinking along the lines of a software app for those of us who use hardware phones but still want to keep TXT chat and presence and perhaps integrated into 1 of the iax soft phones as well to provide a full solution. Question is then, how well does their system work? Already have an IAX2 compatible soft phone with that stuff in it, why not make use of the fact and just work out what needs to be sent to their client... ___ 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
