[asterisk-users] 回覆︰ WebRTC softphone for Asterisk - any suggestion?
you can use sipml5 web page ,,, It work for webrtc... 寄件人︰ Lenz Emilitri lenz.lo...@gmail.com 收件人︰ Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com 傳送日期︰ 2013年06月3日 (週一) 6:34 PM 主題︰ Re: [asterisk-users] WebRTC softphone for Asterisk - any suggestion? Looks yummy! http://phono.com/webrtc 2013/5/31 Adnan 112linuxstockh...@gmail.com Voxeo/Phono webrtc. /Adnan On Fri, May 31, 2013 at 1:53 PM, Lenz Emilitri lenz.lo...@gmail.com wrote: Hi All, I wonder if any of you has some suggestions on which WebRTC client/softphone to use for a click-to-dial, webpage hosted solution. Any suggestions? Thanksl. -- Loway - home of QueueMetrics - http://queuemetrics.com/ Test-drive WombatDialer beta @ http://wombatdialer.com/ -- _ -- 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 -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com/ Test-drive WombatDialer beta @ http://wombatdialer.com/ -- _ -- 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 -- _ -- 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] 回覆︰ WebRTC softphone for Asterisk - any suggestion?
Use sipml5 as webrtc web client . Webrtc2sip GW and asterisk put in to internet . It is work . I try before Regard/chui king man 寄件人︰ Bob Kyeyune bkyey...@gmail.com 收件人︰ Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com 傳送日期︰ 2013年06月1日 (週六) 1:25 PM 主題︰ Re: [asterisk-users] WebRTC softphone for Asterisk - any suggestion? hello; hopefully u can help me i have asterisk vanilla installation 11 and i have also managed to install webrtc2sip how do i make asterisk to communicate with webrtc2sip c'se right now both run independently Regards. Kyeyune Bob Network IT Engineer +256 774 702 258 bob.kyey...@onesolutions.ug Integrated IT services from Plot 57B Luthuli Avenue Bugolobi, Kampala On Fri, May 31, 2013 at 11:36 PM, Adnan 112linuxstockh...@gmail.com wrote: Voxeo/Phono webrtc. /Adnan On Fri, May 31, 2013 at 1:53 PM, Lenz Emilitri lenz.lo...@gmail.com wrote: Hi All, I wonder if any of you has some suggestions on which WebRTC client/softphone to use for a click-to-dial, webpage hosted solution. Any suggestions? Thanksl. -- Loway - home of QueueMetrics - http://queuemetrics.com/ Test-drive WombatDialer beta @ http://wombatdialer.com/ -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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] 回覆︰ Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger
Hi, So , how to connect asterisk to whatapps ??Please advice .. Thank Regard/chui king man 寄件人︰ Lenz Emilitri lenz.lo...@gmail.com 收件人︰ isr...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com 傳送日期︰ 2013年04月19日 (週五) 4:34 PM 主題︰ Re: [asterisk-users] Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger Depends on what you are trying to do. Not in general (AFAIK) but you may find a number of scripts around. 2013/4/18 isr...@gmail.com I think facebook uses xmpp so you could use asterisk jabber or so Don't know about the rest -Original Message- From: bilal ghayyad bilmar...@yahoo.com Sender: asterisk-users-boun...@lists.digium.com Date: Wed, 17 Apr 2013 14:41:53 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger Hello; Is there any modules or channels or integration between asterisk and any of the following: whatsapp, facebook, viber, yahoo and hotmail messanger? Regards Bilal -- _ -- 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 -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com/ Test-drive WombatDialer beta @ http://wombatdialer.com/ -- _ -- 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 -- _ -- 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] 回覆︰ app_rtsp.c ported to Asterisk 11.x
Dear Sir, I want the source code . Please email to me ... Thank Regard/chui king man 寄件人︰ Robert Krakora rob.krak...@messagenetsystems.com 收件人︰ asterisk-users@lists.digium.com; asterisk-...@lists.digium.com 傳送日期︰ 2013年03月16日 (週六) 4:10 AM 主題︰ [asterisk-users] app_rtsp.c ported to Asterisk 11.x Hi, If anyone is interested, I have ported app_rtsp.c to Asterisk 11.x. I have tested it with GStreamer RTSP server and a C920 webcam streaming H264 SVC video from one machine to another machine running Linphone. Contact me at this e-mail address robkrak...@messagenetsystems.com for source code. Best Regards, -- Rob Krakora MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032 (317)566-1677Ext 212 (317)663-0808 Fax -- _ -- 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 -- _ -- 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] 回覆︰ 回覆︰ Directmedia question
I mean set DTMF =sip info ... not inband .. it sis work .. it do not relay on what codec you use .. it work I test before ... 寄件人︰ Mark Henry markhenry...@gmail.com 收件人︰ kingman chui chuiking...@yahoo.com.hk; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com 傳送日期︰ 2013年03月9日 (週六) 5:21 PM 主題︰ Re: [asterisk-users] 回覆︰ Directmedia question But that is not supported in g729 Inband DTMF is not supported on codec g729. Use RFC2833 Still media is through Asterisk On Sat, Mar 9, 2013 at 3:34 AM, kingman chui chuiking...@yahoo.com.hk wrote: If you want to use direcmedia = yes , in order take to effect.You must not set dtmf = rfc2833 .You should set it dtmf = info. It should work then. Regard/chui king man 寄件人︰ Mark Henry markhenry...@gmail.com 收件人︰ asterisk-users@lists.digium.com 傳送日期︰ 2013年03月9日 (週六) 7:23 AM 主題︰ [asterisk-users] Directmedia question Hello List, I have some doubt about direct media settings. I have an asterisk 1.8.14 instance running on 172.20.255.50, a soft phone on IP 10.100.210.51 and a gateway at 10.100.210.254 I have set both gateway and peer to directmedia=yes but still on gateway I see RTP from asterisk's IP, have tried setting nat=yes/no and also specifying localnet values but not sure where I am doing wrong. Also directrtpsetup is set to yes A sip debug and sip show peer output is here http://pastebin.com/5PwqJ1KW Please assist Thanks -- _ -- 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 -- _ -- 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 -- _ -- 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] 回覆︰ Directmedia question
If you want to use direcmedia = yes , in order take to effect.You must not set dtmf = rfc2833 .You should set it dtmf = info. It should work then. Regard/chui king man 寄件人︰ Mark Henry markhenry...@gmail.com 收件人︰ asterisk-users@lists.digium.com 傳送日期︰ 2013年03月9日 (週六) 7:23 AM 主題︰ [asterisk-users] Directmedia question Hello List, I have some doubt about direct media settings. I have an asterisk 1.8.14 instance running on 172.20.255.50, a soft phone on IP 10.100.210.51 and a gateway at 10.100.210.254 I have set both gateway and peer to directmedia=yes but still on gateway I see RTP from asterisk's IP, have tried setting nat=yes/no and also specifying localnet values but not sure where I am doing wrong. Also directrtpsetup is set to yes A sip debug and sip show peer output is here http://pastebin.com/5PwqJ1KW Please assist Thanks -- _ -- 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 -- _ -- 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] 回覆︰ Question about directmedia or canreinvite in sip.conf
Hi, If you use rfc2833 and set directmedia=yes, diect media will not work. You must set other value like SIP info in order to make diectmedia work ... Regard/chui kingh man 寄件人︰ Shitian Long longst...@gmail.com 收件人︰ asterisk-users@lists.digium.com 傳送日期︰ 2013年01月17日 (週四) 7:27 PM 主題︰ [asterisk-users] Question about directmedia or canreinvite in sip.conf Hello, I have a question about directmedia or canreinvite, I have experience that whatever I set directmedia=yes or no. After I run sip show settings. all settings looks the same. My question is how I could make sure from sip show settings that my directmedia configuration is applied. Thanks -- _ -- 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 -- _ -- 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] 回覆︰ tcptls ssl connection error
Hi, I set up tls and srtp with lyn and asterisk 1.8 before. I think your ssl connection is not setup . so, it may due to key and certificate problem. If the key and cert is ok with CA, the ssl connection will up auto.. I work this before and I can connect to lync server with TLS and srtp I my case, lync server is CA auth , asterisk is the client only Hope this can help you ... 寄件人︰ Chandrakant Solanki solanki.chandrak...@gmail.com 收件人︰ Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com 傳送日期︰ 2012年11月20日 (週二) 2:39 PM 主題︰ Re: [asterisk-users] tcptls ssl connection error Hello All, Anyone have idea regarding below error. After applying all patch, still faced the same issue. -- Regards, Chandrakant Solanki On Fri, Nov 9, 2012 at 11:39 AM, Chandrakant Solanki solanki.chandrak...@gmail.com wrote: Hello All, I am using asterisk 1.8.13.0 and which is running on TLS port and my request forwarded from opensips which is also run tls port. On both end my certificate is same. During search about this error, I found below blog and apply patch, then also found below error. https://issues.asterisk.org/jira/browse/ASTERISK-18345 https://issues.asterisk.org/jira/browse/ASTERISK-20559 Also applied r375023 [Nov 8 21:57:34] ERROR[16357]: tcptls.c:89 ssl_close: SSL_shutdown() failed: 5 [Nov 8 21:57:36] ERROR[16001]: tcptls.c:89 ssl_close: SSL_shutdown() failed: 5 [Nov 8 21:57:37] == Problem setting up ssl connection: error::lib(0):func(0):reason(0) [Nov 8 21:57:37] WARNING[19274]: tcptls.c:251 handle_tcptls_connection: FILE * open failed! [Nov 8 21:57:39] == Problem setting up ssl connection: error::lib(0):func(0):reason(0) [Nov 8 21:57:39] WARNING[19356]: tcptls.c:251 handle_tcptls_connection: FILE * open failed! [Nov 8 21:57:49] == Problem setting up ssl connection: error::lib(0):func(0):reason(0) [Nov 8 21:57:49] WARNING[19357]: tcptls.c:251 handle_tcptls_connection: FILE * open failed! -- Regards, Chandrakant Solanki -- _ -- 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 -- _ -- 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] 回覆︰ Simple failover configuration
I think you can use virtual IP with a group of sip server that run in HA . so, only one of sipserver is handle the call and other is standby ... Is this what you want ...? Regard/chui king man 寄件人︰ Danny Nicholas da...@debsinc.com 收件人︰ 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com 傳送日期︰ 2012年11月15日 (週四) 11:31 PM 主題︰ Re: [asterisk-users] Simple failover configuration You can actually configure at least some Polycom phones to 3 or more SIP servers. Your problem is going to be that when one of your servers is down for whatever reason, the line key attached to that server will be off. In a Dual Server environment, I would lean toward putting something like Kamailio (sp) in line so it can determine which server is the active one. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Thursday, November 15, 2012 9:27 AM To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Simple failover configuration Polycom phones after firmware 2.x register to BOTH the primary and backup servers. On Thu, Nov 15, 2012 at 8:59 AM, Chris Nighswonger cnighswon...@foundations.edu wrote: Would the simplest approach to failover be to just configure my primary asterisk server as the first SIP server and my backup as the second? -- _ -- 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 -- _ -- 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 -- _ -- 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] 回覆︰ 回覆︰ Simple failover configuration
It is needed to purchase . Any other option that is open source and free ??? 寄件人︰ Michelle Dupuis mdup...@ocg.ca 收件人︰ kingman chui chuiking...@yahoo.com.hk; Asterisk Users List asterisk-users@lists.digium.com 傳送日期︰ 2012年11月16日 (週五) 8:08 AM 主題︰ RE: [asterisk-users] 回覆︰ Simple failover configuration Or...you could use HAAST (http://www.generationd.com/) - it detects failure, switches IP to an Asterisk peer, updates routes, updates ARP tables, synchronizes settings between peers, etc. -M- P.S. I work for generationd, so I think the product is amazing :) From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of kingman chui [chuiking...@yahoo.com.hk] Sent: Thursday, November 15, 2012 6:34 PM To: Asterisk Users List Subject: [asterisk-users] 回覆︰ Simple failover configuration I think you can use virtual IP with a group of sip server that run in HA . so, only one of sipserver is handle the call and other is standby ... Is this what you want ...? Regard/chui king man 寄件人︰ Danny Nicholas da...@debsinc.com 收件人︰ 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com 傳送日期︰ 2012年11月15日 (週四) 11:31 PM 主題︰ Re: [asterisk-users] Simple failover configuration You can actually configure at least some Polycom phones to 3 or more SIP servers. Your problem is going to be that when one of your servers is down for whatever reason, the line key attached to that server will be off. In a Dual Server environment, I would lean toward putting something like Kamailio (sp) in line so it can determine which server is the active one. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Thursday, November 15, 2012 9:27 AM To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Simple failover configuration Polycom phones after firmware 2.x register to BOTH the primary and backup servers. On Thu, Nov 15, 2012 at 8:59 AM, Chris Nighswonger cnighswon...@foundations.edu wrote: Would the simplest approach to failover be to just configure my primary asterisk server as the first SIP server and my backup as the second? -- _ -- 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 -- _ -- 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 -- _ -- 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] 回覆︰ On SIP REGISTER event trigger a AGI script
I try before , I connect AMI by telnet localhost .. In this session , if I register a softphone or unregiser from softphone , There is event message and show peer no : XXX ,and show register or unregister . You can capture this message and do further what you want ... Regad/chui king man 寄件人︰ Face falaz...@gmail.com 收件人︰ Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com 傳送日期︰ 2012年11月16日 (週五) 9:55 AM 主題︰ Re: [asterisk-users] On SIP REGISTER event trigger a AGI script On Fri, Nov 16, 2012 at 12:32 AM, Danny Nicholas da...@debsinc.com wrote: Check for Status on these commands. If it comes back OK the peer is registered. If not, it should return UNKNOWN. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Face Sent: Thursday, November 15, 2012 12:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] On SIP REGISTER event trigger a AGI script On Wed, Nov 14, 2012 at 7:42 PM, Steve Edwards asterisk@sedwards.com wrote: On Wed, 14 Nov 2012, Face wrote: Is there a way I can trigger a AGI script On SIP REGISTER event. On Wed, 14 Nov 2012, Danny Nicholas wrote: What you will need to do is to monitor for the SIP REGISTER in AMI, then launch a local channel call to run your AGI process. If the process is not time-critical, you could monitor the logs to cut down on AMI traffic. Just for clarity... If the OP's script does not interact with Asterisk using the AGI protocol, it is not an 'AGI' and does not need any channel. If the OP's 'goal' is something like 'when Asterisk receives a REGISTER, I want to update a counter in my MySQL database' then no channel is needed. Hey, OP, if we know what you are trying to accomplish, maybe we can suggest a solution rather than answering a 'feature' question. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 I have a related question. I need to know if a SIP peer is registered or unregistered. I try : CLI -- sip show peer 6000 AMI -- Action: SIPShowPeer Peer: 6000 AMI -- Action: ExtensionState Exten: 1 noun of those return anything about the SIP peer is registered or unregistered! -- Sincerely, falazemi -- _ -- 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 -- _ -- 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 I try all of those -- Action: Status Peer: 600 Action: Status Exten: 600 Action: Status SIP: 600 the return is this -- Response: Success Message: Channel status will follow Event: StatusComplete Items: 0 Response: Success Message: Channel status will follow Event: StatusComplete Items: 0 Response: Success Message: Channel status will follow Event: StatusComplete Items: 0 -- Sincerely, falazemi -- _ -- 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 -- _ -- 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:
[asterisk-users] 回覆︰ Asterisk SIP authenticate using Radius / LDAP
HI, I have connect to Radius with asterisk 1.8.11 before. For CDR, I use the cdr_radius and the cdr can write to radius server. For auth with radius server, I use php-radius to write php script and use agi in dialplan to auth the account . It is work .. Regard/chui king man 寄件人︰ qasimak...@gmail.com qasimak...@gmail.com 收件人︰ Samira Hosseini samiramhosse...@yahoo.com; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com 傳送日期︰ 2012年11月12日 (週一) 3:50 PM 主題︰ Re: [asterisk-users] Asterisk SIP authenticate using Radius / LDAP You can use Radius Agi developed by PortaOne from following link. http://www.voip-info.org/wiki/view/PortaOne+Radius+auth Regards, Qasim On Mon, Nov 12, 2012 at 11:24 AM, Samira Hosseini samiramhosse...@yahoo.com wrote: Hi all, based on the following link, I am going to authenticate SIP asterisk users via Radius client that is installed on my Asterisk then the radius client connect to asterisk using the radius and ldap: https://who.rocq.inria.fr/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html#AEN237 So I want to know for implementing the mentioned authentication method I need to use the patched asterisk as follow : https://issues.asterisk.org/jira/browse/ASTERISK-5278?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel Thanks. -- _ -- 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 -- _ -- 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 -- _ -- 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] 回覆︰ 回覆︰ Asterisk SIP authenticate using Radius / LDAP
I try before . I use asterisk 1.8.11 , I cannot compile the patch under asterisk 1.8.11 . So, it isn ot work in asterisk 1.8.11 an Use php-radius to write ph p script to auth radius server Please advice other method to auth with radius server under asterisk 1.8.11 if you know .. Thank Regard/chui king man 寄件人︰ s...@yahoo.com samiramhosse...@yahoo.com 收件人︰ kingman chui chuiking...@yahoo.com.hk 傳送日期︰ 2012年11月12日 (週一) 6:22 PM 主題︰ Re: 回覆︰ [asterisk-users] Asterisk SIP authenticate using Radius / LDAP Hello, thanks for your help, but do you think I will able to connect asterisk(that is installed radius client on it) to the radius server by the following link? https://issues.asterisk.org/jira/browse/ASTERISK-5278?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel From: kingman chui chuiking...@yahoo.com.hk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; Samira Hosseini samiramhosse...@yahoo.com Sent: Monday, 12 November 2012, 12:26:32 Subject: 回覆︰ [asterisk-users] Asterisk SIP authenticate using Radius / LDAP HI, I have connect to Radius with asterisk 1.8.11 before. For CDR, I use the cdr_radius and the cdr can write to radius server. For auth with radius server, I use php-radius to write php script and use agi in dialplan to auth the account . It is work .. Regard/chui king man 寄件人︰ qasimak...@gmail.com qasimak...@gmail.com 收件人︰ Samira Hosseini samiramhosse...@yahoo.com; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com 傳送日期︰ 2012年11月12日 (週一) 3:50 PM 主題︰ Re: [asterisk-users] Asterisk SIP authenticate using Radius / LDAP You can use Radius Agi developed by PortaOne from following link. http://www.voip-info.org/wiki/view/PortaOne+Radius+auth Regards, Qasim On Mon, Nov 12, 2012 at 11:24 AM, Samira Hosseini samiramhosse...@yahoo.com wrote: Hi all, based on the following link, I am going to authenticate SIP asterisk users via Radius client that is installed on my Asterisk then the radius client connect to asterisk using the radius and ldap: https://who.rocq.inria.fr/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html#AEN237 So I want to know for implementing the mentioned authentication method I need to use the patched asterisk as follow : https://issues.asterisk.org/jira/browse/ASTERISK-5278?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel Thanks. -- _ -- 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 -- _ -- 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 -- _ -- 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] 回覆︰ problem on LDAP (Invalid credential)
I use res_ldap.conf to connect lync server by ldap ebfore. For auth in lync server . The username in res_lap.conf cannot use format cn=xx,dc=xxx . It should use format username=@x and passwd. Then lync server can auth you . I try before and it work in asterisk 1.8.11 Hope it can help you .. Regard/chui king man 寄件人︰ Samira Hosseini samiramhosse...@yahoo.com 收件人︰ asterisk-users@lists.digium.com asterisk-users@lists.digium.com 傳送日期︰ 2012年11月9日 (週五) 1:39 AM 主題︰ [asterisk-users] problem on LDAP (Invalid credential) Hello all, I am going to register asterisk sip users through active directory accounts LDAP (that is a separated server with ip : 192.168.11.17) So I have followed the below link as well: https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html http://ensiwiki.ensimag.fr/index.php/Asterisk's_external_configuration_(LDAP) Server:192.168.14.90 = asterisk server:192.168.11.17 = ActiveDirectory Finally, this is my configuration file : [root@PBX ~]# telnet 192.168.11.17 389 Trying 192.168.11.17... Connected to 192.168.11.17 (192.168.11.17). Escape character is '^]'. [_general] host=192.168.11.17 ; LDAP host port=389 protocol=3 ; Version of the LDAP protocol to use; default is 3. url=ldap://192.168.11.17:389 basedn=dc=example,dc=com ;User=cn=,dc=example,dc=com ;User=cn=join_lan,dc=example,dc=com ;User=cn=sa_hosseini,dc=rasana,dc=ir User=cn=lan,cn=technical,cn=xyz,cn=join_lan,dc=example,dc=com Pass=123456 --- vim /etc/asterisk/extconfig.conf sipusers = ldap,dc=example,dc=com,sip vim /etc/asterisk/sip.conf [general] callevents=yes rtcachefriends=yes but i got the follwoing error : PBX*CLI module reload res_config_ldap.so -- Reloading module 'res_config_ldap.so' (LDAP realtime interface) == Parsing '/etc/asterisk/res_ldap.conf': == Found [Nov 8 09:38:06] WARNING[8687]: res_config_ldap.c:1750 ldap_reconnect: bind failed: Invalid credentials [Nov 8 09:38:06] WARNING[8687]: res_config_ldap.c:1598 reload: Couldn't establish connection to your directory server. Check debug. == LDAP RealTime driver reloaded. Then i have registered with user:join_lan;pass:123456 domain:192.168.14.90 and get the following error on CLI: Verbosity is at least 15 [Nov 8 09:41:42] NOTICE[8674]: chan_sip.c:25005 handle_request_register: Registration from 'join_lansip:join_lan@192.168.14.90' failed for '192.168.19.21:38968' - No matching peer found [Nov 8 09:41:42] NOTICE[8674]: chan_sip.c:25005 handle_request_register: Registration from 'join_lansip:join_lan@192.168.14.90' failed for '192.168.19.21:38968' - No matching peer found -- _ -- 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 -- _ -- 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] ldap realtime function do not work in asterisk 1.8.11
[code]
res_ldap.conf
[_general]
;
; Specify one of either host and port OR url. URL is preferred, as you can
; use more options.
;host=192.168.1.1 ; LDAP host
host=lync-demo.local ; LDAP host
port=389
url=ldap://ad.lync-demo.local:389
protocol=3 ; Version of the LDAP protocol to use;
default is 3.
basedn=dc=lync-demo,dc=local ; Base DN
;user=cn=administrator,dc=lync-demo,dc=local ; Bind DN
user=cn=administrator,cn=users,dc=lync-demo,dc=local ; Bind DN
;user=dc=lync-demo,dc=local ; Bind DN
pass=Esi88
[extensions]
;context = AstExtensionContext
;context = givenname
;exten = AstExtensionExten
attribute=exten=givenname
;priority = AstExtensionPriority
;attribute=priority=givenname
;app = AstExtensionApplication
;appdata = AstExtensionApplicationData
additionalFilter=(objectClass=user)
[/code]
[code]
extconfig.conf
extensions = ldap,dc=lync-demo,dc=local,extensions
[/code]
[code]
[from-internal]
include = from-internal-xfer
include = bad-number
switch = Realtime/@extensions
exten= William,1,Set(CHANNEL(secure_bridge_media)=1)
exten= William,2,Set(_SIP_SRTP_SDES=1)
exten= William,3,Set(_SIPSRTP=optional)
exten= William,4,Set(_SIPSRTP_CRYPTO=enable)
exten = William,5,Set(b=${REALTIME(extensions,givenname,William)})
exten = William,6,NoOp(${b})
exten = William,7,Set(pair=${CUT(b,|,1)});
exten = William,8,Set(col_name=${CUT(pair,=,2)});
exten= William,n,Hangup()
[/code]
I use realtime to connect ldap server at lync
But When I query the ldap , I get below error in full log .
I expect the ldap query will get back something according input
givename=William .
The REALTiME function cannot retrevie the givename from lync and output null.
There is this key/attribute in lync server .
There is openration error .
The lync ldap server is working and I can use the filter
((objectClass=user)(givenname=William)) to get the result by
php ldap_Search .. it is work .
Please advice what is wrong in asterisk I use asterisk 1.8.11 ...
[Oct 30 00:42:48] DEBUG[9260] app_queue.c: Device 'SIP/3200' changed to state
'2' (In use) but we don't care because they're not a member of any queue.
[Oct 30 00:42:48] WARNING[9264] res_config_ldap.c: Failed to query directory.
Error: Operations error.
[Oct 30 00:42:48] WARNING[9264] res_config_ldap.c: Query:
((objectClass=user)(givenname=William))
[Oct 30 00:42:48] DEBUG[9264] pbx.c: Function result is '(null)'
[Oct 30 00:42:48] DEBUG[9264] pbx.c: Launching 'Set'
[Oct 30 00:42:48] VERBOSE[9264] pbx.c: -- Executing
[William@from-internal:5] Set(SIP/3200-, b=) in new stack
Log for asterisk full
https://dl.dropbox.com/u/68357652/full.rar--
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[asterisk-users] Anyobe help: call leg do not exist error
Dear Sir, I use asterisk 1.8.11 (192.168.100.202)to connect lync server .I use tls port 5068 to connect to this lync server . The tls is ok to establish and I make call from softphone 3200 (register to Asterisk) and dial 9XXX (9+85225082162) , this prefix will dial to trunk lync_trunk and pass to lync server(192.168.100.14) using tls . But the lync client in opposite side ringing and they recevie the call , but when they answer the call , the call drop and hang up immediately .In sip trace I see there is call leg not exits error . what is wrong ...Below is the related setting and trace ... I attach the trace and config setting in attach file . Regard/Chui king man trace.rar Description: Binary data -- _ -- 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] Anyone help: call leg do not exist err
Dear Sir, I use asterisk 1.8.11 (192.168.100.202)to connect lync server .I use tls port 5068 to connect to this lync server . The tls is ok to establish and I make call from softphone 3200 (register to Asterisk) and dial 9XXX (9+85225082162) , this prefix will dial to trunk lync_trunk and pass to lync server(192.168.100.14) using tls . But the lync client in opposite side ringing and they recevie the call , but when they answer the call , the call drop and hang up immediately .In sip trace I see there is call leg not exits error . what is wrong ...Below is the related setting and trace ... I attach the trace and config setting in attach file . Regard/Chui king man trace.rar Description: Binary data -- _ -- 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
