[asterisk-users] Registration timed out after a sip reload
Hi, I have configured a sip provider account, with register = user :[EMAIL PROTECTED]/user, with Asterisk 1.4.2. Then I start Asterisk, which register successfully to the sip provider: sip show registry show me the provider and status Registered. I do a sip reload in the CLI, and now registration timed out, sip show registry show me Request Send, and after 10 minutes registration is still not complete and retries.. Why ? Greg ___ --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] Asterisk 1.4.2
Hi, (Salut thomas ;) ) I have a working trunk with a SIP provider on asterisk 1.4.1 and 1.4.2, without problems. Send us your sip peers configuration to try to solve your problem. Greg. Hi all, I upgrade my asterisk from 1.2.11 to 1.4.2 changing my entire dial plan but I have the following errors and I'm not able to call anymore. Do you know what can I have to do? My Asterisk is connected to a patton with a SIP trunk. [Mar 22 10:19:03] WARNING[16462]: chan_sip.c:12311 handle_response: Remote host can't match request BYE to call '[EMAIL PROTECTED]'. Giving up. [Mar 22 10:19:04] WARNING[16462]: chan_sip.c:12317 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. Thanks a lot, Thomas ___ --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] Query Failed because: Incorrect information in file: './asterisk/sip.frm'
Hi, I have a working asterisk 1.4.0 with Mysql Realtime configuration, and today I encountered this error. Now, I have no acces to any information in mysql realtime, so nothing work now ! [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Everything is fine. [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions WHERE exten = 'h' AND context = ' interne' AND priority = '1' [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Query: SELECT * FROM extensions WHERE exten = 'h' AND context = 'interne ' AND priority = '1' [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Query Failed because: Incorrect information in file: './asterisk/extensi ons.frm' [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Everything is fine. [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions WHERE exten LIKE '\\_%' AND conte xt = 'interne' AND priority = '1' ORDER BY exten [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Query: SELECT * FROM extensions WHERE exten LIKE '\\_%' AND context = 'i nterne' AND priority = '1' ORDER BY exten [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Query Failed because: Incorrect information in file: './asterisk/extensi ons.frm' [Jan 24 10:32:40] DEBUG[31026] res_config_mysql.c: MySQL RealTime: Everything is fine. [Jan 24 10:32:40] DEBUG[31026] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sip WHERE name = '129.200.1.51' [Jan 24 10:32:40] DEBUG[31026] res_config_mysql.c: MySQL RealTime: Query: SELECT * FROM sip WHERE name = '129.200.1.51' [Jan 24 10:32:40] DEBUG[31026] res_config_mysql.c: MySQL RealTime: Query Failed because: Incorrect information in file: './asterisk/sip.frm 'ast ___ --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] Asterisk and 3PCC
Hi all, I have an Asterisk server running, and some hardware phones, and I want to do 3PCC : third party call control. The third party is a software running on the asterisk box, which can for example ask a hard SIP phone to put a call on hold. To do that, this software has to send a SIP message to this phone. How can I do that ? Greg ___ --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] Asterisk and 3PCC
Seems that this has to be implemented by the phones, or by a B2BUA I think that a B2BUA could be used for 3PCC, but dont know if an open-source B2BUA exists and works with Asterisk Greg _ De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Koen Van Impe Envoyé : mardi 9 janvier 2007 10:33 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Asterisk and 3PCC Gregory, I know there is something called SIP CTI TR87. It's used by Nortel to integrate with Microsoft's Live Communication Server. Don't know if something similar exists for Asterisk. This links could be helpfull: http://www.ecma-international.org/publications/techreports/E-TR-087.htm Regards, Koen On 1/9/07, Gregory Duchatelet [EMAIL PROTECTED] wrote: Hi all, I have an Asterisk server running, and some hardware phones, and I want to do 3PCC : third party call control. The third party is a software running on the asterisk box, which can for example ask a hard SIP phone to put a call on hold. To do that, this software has to send a SIP message to this phone. How can I do that ? Greg ___ --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] Asterisk and 3PCC
True :) Here is an example of what i want to do : - a phone call extension 100 - asterisk enter the context, and execute Dial() to call another phone - ringing... - now I want that asterisk ask the called phone to answer : how to do that ?? Greg uhm... On Tue, 2007-01-09 at 12:28 +0100, Gregory Duchatelet wrote: Seems that this has to be implemented by the phones, or by a B2BUA. I think that a B2BUA could be used for 3PCC, but don't know if an open-source B2BUA exists and works with Asterisk . asterisk IS a B2BUA just my 2cents. Matteo. ___ --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] 2 devices using same sip account
Hi, It seems that they both can make calls, but only one can receive call: the last registered... Greg Hi all, What will happen if 2 devices using the same set of sip account to connect to the same asterisk? Do they both can make call? Can they receive call as normal? ___ --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] sip peer name channel variable?
Or this link :
http://www.voip-info.org/wiki/index.php?page=Asterisk+Local+channels
http://www.voip-info.org/wiki/index.php?page=Asterisk+Local+channels
se the /n parameter of Local/ channels.
Cheers
Greg
_
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de William Piper
Envoyé : lundi 18 décembre 2006 06:03
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] sip peer name channel variable?
Check out this page:
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sipchaninfo
bp
On 12/17/06, Damon Estep [EMAIL PROTECTED] wrote:
Started out looking for what I thought was going to be a simple variable
name, have not found it.
Does anyone know of a variable that would contain only the SIP peer name of
the originating channel?
${CHANNEL} contains it, but it needs to be parsed and our dial plan
sometimes uses local channels, in one case it may be SIP/peer-id and in
another case local/peer-id
The peer is defined as type=friend
v1.2.13
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[asterisk-users] How to know who hangup ?
Hi, Using AMI or dial plan, how can i know which leg (channel ?) of a bridged call, hangup ? AMI send 2 hangup events, which have both cause 16 (normal clearing), and the first hangup event is the called leg hangup event, not the one who hangup. Greg ___ --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] How to know who hangup ?
adding g in your dial application and the call will go on the extension
when the callee hangup
Yes, i could also use h extension, but how to know which one hangup first
? ${HANGUPCAUSE} always say 16 (Normal clearing), and ${CHANNEL} is set to
the current channel in the dial plan...
Greg
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RE: [asterisk-users] Diva Server V-BRI-2 and internal numbers
It looks like 107 is busy ;-) Please increase verbosity, like set verbose 5 capi debug to see what is happening. Hi Armin, Verbose was at 30 :) 107 is not busy since i can call it from 102, which is another internal phone. All internal phones are busy for Asterisk... Here is the log with verbose at 100 and capi debug enabled : -- Executing Dial(SIP/Greg-081f5a10, CAPI/ISDN1/b:107||rtT) in new stack data = ISDN1/b:107 parsed dialstring: 'ISDN1' 'b' '107' '' capi request for interface 'ISDN1' parsed dialstring: 'ISDN1' 'b' '107' '' == ISDN1: Call CAPI/ISDN1/107-1e (pres=0x00, ton=0x00) CONNECT_REQ ID=001 #0x1002 LEN=0047 Controller/PLCI/NCCI= 0x1 CIPValue= 0x1 CalledPartyNumber = 80107 CallingPartyNumber = 00 80b CalledPartySubaddress = default CallingPartySubaddress = default BProtocol B1protocol = 0x1 B2protocol = 0x1 B3protocol = 0x0 B1configuration= default B2configuration= default B3configuration= default GlobalConfiguration= default BC = default LLC = default HLC = default AdditionalInfo BChannelinformation= 00 00 Keypadfacility = default Useruserdata = default Facilitydataarray = default SendingComplete= default -- Called ISDN1/b:107 CAPI devicestate requested for ISDN1/107 CAPI devicestate requested for ISDN1/107 CONNECT_CONF ID=001 #0x1002 LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 -- ISDN1: received CONNECT_CONF PLCI = 0x101 INFO_IND ID=001 #0x11a8 LEN=0016 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x18 InfoElement = 8a INFO_RESP ID=001 #0x11a8 LEN=0012 Controller/PLCI/NCCI= 0x101 -- ISDN1: info element CHANNEL IDENTIFICATION 8a INFO_IND ID=001 #0x11a9 LEN=0015 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x800d InfoElement = default INFO_RESP ID=001 #0x11a9 LEN=0012 Controller/PLCI/NCCI= 0x101 -- ISDN1: info element SETUP ACK INFO_IND ID=001 #0x11ab LEN=0015 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x8002 InfoElement = default INFO_RESP ID=001 #0x11ab LEN=0012 Controller/PLCI/NCCI= 0x101 -- ISDN1: info element CALL PROCEEDING -- CAPI/ISDN1/107-1e is proceeding passing it to SIP/Greg-081f5a10 INFO_IND ID=001 #0x11ad LEN=0037 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x1c InfoElement = 91 a1 13 02 02 8f 02 01 2200a a1 05003 02 01 00 82 01 01 INFO_RESP ID=001 #0x11ad LEN=0012 Controller/PLCI/NCCI= 0x101 -- ISDN1: info element FACILITY INFO_IND ID=001 #0x11ae LEN=0017 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x8 InfoElement = 81 d8 INFO_RESP ID=001 #0x11ae LEN=0012 Controller/PLCI/NCCI= 0x101 -- ISDN1: info element CAUSE 81 d8 INFO_IND ID=001 #0x11af LEN=0015 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x8045 InfoElement = default INFO_RESP ID=001 #0x11af LEN=0012 Controller/PLCI/NCCI= 0x101 -- ISDN1: info element DISCONNECT -- ISDN1: Disconnect case 1 -- CAPI/ISDN1/107-1e is busy == ISDN1: CAPI Hangingup -- ISDN1: activehangingup (cause=88) DISCONNECT_REQ ID=001 #0x1003 LEN=0018 Controller/PLCI/NCCI= 0x101 AdditionalInfo BChannelinformation= default Keypadfacility = default Useruserdata = default Facilitydataarray = default SendingComplete= default == Everyone is busy/congested at this time (1:1/0/0) -- Executing Hangup(SIP/Greg-081f5a10, ) in new stack == Spawn extension (macro-appel_sortant, s, 2) exited non-zero on 'SIP/Greg-081f5a10' in macro 'appel_sortant' == Spawn extension (macro-appel_sortant, s, 2) exited non-zero on 'SIP/Greg-081f5a10' CAPI devicestate requested for ISDN1/107 CAPI devicestate requested for ISDN1/107 DISCONNECT_CONF ID=001 #0x1003 LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 INFO_IND ID=001 #0x11b1 LEN=0015 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x805a InfoElement = default INFO_RESP ID=001 #0x11b1 LEN=0012 Controller/PLCI/NCCI= 0x101 -- ISDN1: info element RELEASE COMPLETE DISCONNECT_IND ID=001 #0x11b3
[asterisk-users] Diva Server V-BRI-2 and internal numbers
Hi, I have an asterisk with a DIVA Server V-BRI-2 card, connected to a Siemens PABX. From a SIP phone, I can call other internal SIP phones, external numbers (to PSTN), but I can't call internal phones connected to the internal phone network. When I call 107, which is an internal phone, heres the logs from asterisk: -- Executing Dial(SIP/Greg-081f5a10, CAPI/ISDN1/b:107||rtT) in new stack -- Called ISDN1/b:107 -- CAPI/ISDN1/107-1a is proceeding passing it to SIP/Greg-081f5a10 -- CAPI/ISDN1/107-1a is busy == ISDN1: CAPI Hangingup == Everyone is busy/congested at this time (1:1/0/0) -- Executing Hangup(SIP/Greg-081f5a10, ) in new stack == Spawn extension (macro-appel_sortant, s, 2) exited non-zero on 'SIP/Greg-081f5a10' in macro 'appel_sortant' == Spawn extension (macro-appel_sortant, s, 2) exited non-zero on 'SIP/Greg-081f5a10' BUT! If I call an internal isdn number like 122 which is a fax, the call is answered. How can I call 107 ? Greg ___ --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] CTI: put on hold a call
Hi list, I need no control a call via AMI or AGI or whatever. I don't know how to put a call on hold. Example: an external call ring, in the dial plan I call Dial application to an internal SIP phone. But my SIP phone does not have the on hold feature, so how to put the callee on hold ? Thanks Greg ___ --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] CTI: put on hold a call
Another way would be to control the channel from asterisk. It is a SIP feature, not an asterisk feature. I have a SIP phone (not a softphone) and want to control it from the computer. Greg One suggestion is to transfer the call to an on-hold extension that plays music, then go pick up the call later. or get a new SIP phone. : ) ~Joel _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Duchatelet Sent: Friday, December 08, 2006 9:51 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] CTI: put on hold a call Hi list, I need no control a call via AMI or AGI or whatever. I don't know how to put a call on hold. Example: an external call ring, in the dial plan I call Dial application to an internal SIP phone. But my SIP phone does not have the on hold feature, so how to put the callee on hold ? Thanks Greg ___ --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] Detecting no answers and/or disconnected numbers
Hi, I am interested about that, too. If someone have some more informations... Greg Hi, Using call files, is there a way to identify no answered calls from disconnected numbers (no longer in service). Both return the same value and so far I can not find a way to know one from the other. Thank you, Andre Courchesne ___ --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] iax/sip registering and real-time
Hi all, I configure an Asterisk with real-time sip and iax users/peers. I want to do an action when users register/unregister to asterisk, how to do that ? Thanks Greg ___ --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] Diva Server, chan_capi and tone detection
This would require a change in chan-capi. To get the extended tone detection indications, additional request/parameter via CAPI must be issued. First, thanks for your reply. Do you have the CxDtmf.pdf document, from Eicon ? If I understand good, you have to enable DTMF facilities 248, 249 and 250, and then you receive DTMF code for tone detection : 0x81 for unidentified ton detected 0x80 for end of signal detected 0xC9 for human speech detected Etc... Another thing is, how do you want to get these indications for use in your dialplan? So, with DTMF code, you could handle it like for fax : redirect to extension vad or something ... Greg ___ --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] Diva Server, chan_capi and tone detection
Hi all, I have a Diva Server V-BRI-2 card, which support, as written in reference guide: Extended tone processing (human talker detection, generation and detection of country-specific tones) I would like to detect human speech and fax tone with asterisk. I think that the diva card transmit a DTMF code when detecting voice, but chan_capi doesn't receive this DTMF code. I verbose it more, displaying all DTMF received, and only DTMF code CNG is received. Did you know how I can enable this detection (see DivaReportTones in Diva Server SDK) or how can I receive this DTMF in chan_capi ? Thanks Greg ___ --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] Send event from dialplan
Hi all, Another question for today, hope an answer for this one. I have a program talking with asterisk via the AMI. I receive events, and I would like to insert some events in the dialplan, which could be catch by my program. Any idea how to do this ? Greg ___ --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] Send event from dialplan
Sorry, asking too quickly, thats what im looking for : http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Eve nts Greg _ De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Gregory Duchatelet Envoyé : mercredi 22 novembre 2006 15:33 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Send event from dialplan Hi all, Another question for today, hope an answer for this one I have a program talking with asterisk via the AMI. I receive events, and I would like to insert some events in the dialplan, which could be catch by my program. Any idea how to do this ? Greg ___ --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] Asterisk 1.4 and Queues RealTime
Hi all, I would like to use the Agent Login feature with real-time queues it is not possible with asterisk 1.2, as described here : http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue The mantis bug describing the implementation of realtime queue is bug 4037. This bug includes some discussion on how to extend dynamic queues to also work with the member login feature. So, did you know if this is possible natively in asterisk 1.4 (or will be) ? Thanks Greg ___ --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] best gui
Hi, Can you tel us more information about this GUI ? I'm interested about it. And what about you ? Will DeStar become a commercial licensed product in a few years ? ;-) When will the 0.3 be released ? Thanks you for your reply. Greg Destar is based on 'configlets' so you can add or remove functionality on an easy way. You can get the destar code or more information at: http://destar.berlios.de/ or you can join the IRC channel #destar on irc.freenode.net. Bye. ___ --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] best gui
Check it out for yourself. Installation is quite simple. Just make sure you backup your existing Asterisk configuration files first. I'm currently trying to install it... and have some problems with cdr_sqlite3_custom and Asterisk 1.2.13 Some other questions : - what about support for asterisk real-time ? - what about AJAX support in fronted ? Thx Greg ___ --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] asterisk and HMP
I think this is now the Eicon HMP platform. It looks like Eicon bought this when the fools paid good money for Dialogic. Its amazing how many companies have got on the HMP bandwagon since we started the Zapata work in 1999. If you do a Google search you can find something like 10 companies promoting HMP type products. Few look like coherent products, though. I ask this question but i twas not the good question. Here a schema : Public telephon network - PABX - computer with a soft call center and dialogic cards. I want to connect this computer to an Asterisk, via a SIP trunk, so I start with a HMP driver on the dialogic cards... I'm not sure I'm clear... Greg ___ --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] asterisk and HMP
Hi all, Does Asterisk now support Intel's HMP platforms ? Does it support in 1.4 version ? There's a special driver for Intel-based HMP hardware+software for ABE. On the other hand, Asterisk has always been doing HMP :). In fact, I would say Asterisk's success in HMP is one of the push factors for companies like Intel, NMS to move to HMP from their traditional DSP-based designs. Yes, i known that. Thanks for your reply, you confirm me that it is available only in ABE. Greg ___ --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] asterisk and HMP
Hi all, Does Asterisk now support Intels HMP platforms? Does it support in 1.4 version ? Thanks. Greg ___ --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] using asterisk to do remote control functions
Hi Im very green to asterisk, and I have been asked if asterisk can be used to do remote control, like opening gates etc, say when the user dials a predefined number ... And what hardware is required ... Many Thanks Hi, yes it is possible using AGI scripts ! Greg ___ --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] say Asterisk to answer
Hi list, I have 2 softphones, 1 Idefisk (IAX), 1 Xlite (SIP) registered to Asterisk. One call the other-one, is it possible to order Asterisk to force answering the call ? i.e. Xlite call Idefisk, Idefisk is ringing, I send a command to Asterisk which force answer, so Idefisk answer the call without clicking on Accept button. Greg ___ --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] acami
Hi list, Im searching for a web configuration front-end for Asterisk, and found ACaMI: http://sourceforge.net/projects/acami/ Anyone here try it? any feedback will be great. Greg ___ --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] GPL Softphones
IAXcomm should. So should wengophone and mozphone. And Kiax and Ekiga -- Daniel Ekiga not for Windows platforms... Greg ___ --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] GPL Softphones
Apparently (from what I gathered from #openwengo at irc.freenode.net)Wengo's own network runs on a combination of Asterisk and OPENSer. To get Wengophone working with your asterisk you will need to do some code hackingz...so download the source code and change it. You will need to change the authentication procedure in Wengo phone so that your server ip and port numbers are used. Check the gmane mailing lists, I've posted the format of the XML messages used. So basically all the work you will have to do is hardcode your website's url in place of wengo.fr's and make sure your website sends back the right type of XML and bazzam! Oh just a note - its not a simple compile, so you will be messing around with it for a while, trying to get it compile. But its awesome... and IMHO one of the best softphones in the world. Maybe even the galaxy Also, to make things a bit better, some devs are in fact developing the server agnostic version of Wengophone. Hope that helped a bit ok bye :) It will help me, thanks :) I think too that Wengo seems to be the best GPLed softphone. Tzafrir Cohen I don't try to compile it, before I'm searching for the one :) iaxComm seems good too for a start. Anyone here make some tries with WengoPhone NG ? Greg ___ --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] GPL Softphones
Hi, Im searching for GPLed softphones. I found WengoPhone but actually not available for Asterisk PBX, only for Wengo network. I found Kiax but only for IAX protocol. Did you know a good GPLed softphones which works on Windows ? Thanks Greg ___ --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
