Re: [asterisk-users] Asterisk and OSX
Up, Does any body have solutions ? 1 Asterisk 12 is compiled, 2. No sound for ivr, voicemail, out calls in calls. 3. I can't use TTS from google, I want use say function on OsX. 4. Cli error Privilege escalation protection disabled! See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details. [0;37m[Apr 12 18:31:44] [1;33mNOTICE[0m[1287]: [1;37mcdr.c[0m:[1;37m1622[0m [1;37mdo_reload[0m: CDR simple logging enabled. [Apr 12 18:31:44] [1;33mNOTICE[0m[1287]: [1;37mloader.c[0m:[1;37m1208[0m [1;37mload_modules[0m: 202 modules will be loaded. [Apr 12 18:31:44] [1;31mWARNING[0m[1287]: [1;37mloader.c[0m:[1;37m439[0m [1;37mload_dynamic_module[0m: Error loading module 'app_queue.so': dlopen(/usr/asterisk/lib/asterisk/modules/app_queue.so, 5): Symbol not found: ___ast_monitor_setjoinfiles Referenced from: /usr/asterisk/lib/asterisk/modules/app_queue.so Expected in: flat namespace [Apr 12 18:31:44] [1;31mWARNING[0m[1287]: [1;37mloader.c[0m:[1;37m439[0m [1;37mload_dynamic_module[0m: Error loading module 'app_stack.so': dlopen(/usr/asterisk/lib/asterisk/modules/app_stack.so, 5): Symbol not found: ___ast_agi_register Referenced from: /usr/asterisk/lib/asterisk/modules/app_stack.so Expected in: flat namespace [Apr 12 18:31:44] [1;31mWARNING[0m[1287]: [1;37mloader.c[0m:[1;37m439[0m [1;37mload_dynamic_module[0m: Error loading module 'app_voicemail.so': dlopen(/usr/asterisk/lib/asterisk/modules/app_voicemail.so, 5): Symbol not found: ___ast_smdi_interface_find Referenced from: /usr/asterisk/lib/asterisk/modules/app_voicemail.so Expected in: flat namespace [Apr 12 18:31:44] [1;31mWARNING[0m[1287]: [1;37mloader.c[0m:[1;37m439[0m [1;37mload_dynamic_module[0m: Error loading module 'chan_agent.so': dlopen(/usr/asterisk/lib/asterisk/modules/chan_agent.so, 5): Symbol not found: ___ast_monitor_setjoinfiles Referenced from: /usr/asterisk/lib/asterisk/modules/chan_agent.so Expected in: flat namespace [Apr 12 18:31:44] [1;31mWARNING[0m[1287]: [1;37mloader.c[0m:[1;37m439[0m [1;37mload_dynamic_module[0m: Error loading module 'chan_iax2.so': dlopen(/usr/asterisk/lib/asterisk/modules/chan_iax2.so, 5): Symbol not found: ___ast_aes_decrypt Referenced from: /usr/asterisk/lib/asterisk/modules/chan_iax2.so Expected in: flat namespace [Apr 12 18:31:44] [1;31mWARNING[0m[1287]: [1;37mloader.c[0m:[1;37m439[0m [1;37mload_dynamic_module[0m: Error loading module 'chan_mgcp.so': dlopen(/usr/asterisk/lib/asterisk/modules/chan_mgcp.so, 5): Symbol not found: ___ast_pktccops_gate_alloc Referenced from: /usr/asterisk/lib/asterisk/modules/chan_mgcp.so Expected in: flat namespace [Apr 12 18:31:44] [1;31mWARNING[0m[1287]: [1;37mloader.c[0m:[1;37m439[0m [1;37mload_dynamic_module[0m: Error loading module 'chan_sip.so': dlopen(/usr/asterisk/lib/asterisk/modules/chan_sip.so, 5): Symbol not found: ___ast_websocket_add_protocol Referenced from: /usr/asterisk/lib/asterisk/modules/chan_sip.so Expected in: flat namespace [Apr 12 18:31:44] [1;31mWARNING[0m[1287]: [1;37mloader.c[0m:[1;37m439[0m [1;37mload_dynamic_module[0m: Error loading module 'func_aes.so': dlopen(/usr/asterisk/lib/asterisk/modules/func_aes.so, 5): Symbol not found: ___ast_aes_decrypt Referenced from: /usr/asterisk/lib/asterisk/modules/func_aes.so Expected in: flat namespace [Apr 12 18:31:44] [1;31mWARNING[0m[1287]: [1;37mloader.c[0m:[1;37m439[0m [1;37mload_dynamic_module[0m: Error loading module 'pbx_dundi.so': dlopen(/usr/asterisk/lib/asterisk/modules/pbx_dundi.so, 5): Symbol not found: ___ast_aes_decrypt Referenced from: /usr/asterisk/lib/asterisk/modules/pbx_dundi.so Expected in: flat namespace [Apr 12 18:31:44] [1;33mNOTICE[0m[1287]: [1;37mres_odbc.c[0m:[1;37m1889[0m [1;37mload_module[0m: res_odbc loaded. [Apr 12 18:31:44] [1;33mNOTICE[0m[1287]: [1;37mres_smdi.c[0m:[1;37m1418[0m [1;37mload_module[0m: No SMDI interfaces are available to listen on, not starting SMDI listener. [Apr 12 18:31:44] [1;31mWARNING[0m[1287]: [1;37mloader.c[0m:[1;37m439[0m [1;37mload_dynamic_module[0m: Error loading module 'chan_mgcp.so': dlopen(/usr/asterisk/lib/asterisk/modules/chan_mgcp.so, 5): Symbol not found: ___ast_pktccops_gate_alloc Referenced from: /usr/asterisk/lib/asterisk/modules/chan_mgcp.so Expected in: flat namespace [Apr 12 18:31:44] [1;31mWARNING[0m[1287]: [1;37mloader.c[0m:[1;37m918[0m [1;37mload_resource[0m: Module 'chan_mgcp.so' could not be loaded. [Apr 12 18:31:44] [1;33mNOTICE[0m[1287]: [1;37mconfig.c[0m:[1;37m2370[0m [1;37mast_config_engine_register[0m: Registered Config Engine curl [Apr 12 18:31:44] [1;33mNOTICE[0m[1287]: [1;37mconfig.c[0m:[1;37m2370[0m [1;37mast_config_engine_register[0m: Registered Config Engine odbc [Apr 12 18:31:44] [1;33mNOTICE[0m[1287]: [1;37mres_config_ldap.c[0m:[1;37m1639[0m [1;37mparse_config[0m: No directory user found, anonymous binding as default. [Apr 12
[asterisk-users] Webrtc and adventures with Asterisk 11
Hi, I spent the past week experimenting with webrtc + asterisk 11.9.0-rc1 + opus/vb8 codec patch. This is interesting technology and I try to find out how to connect all the moving parts. Firefox: Neither sipml5 or jssip works with calls to asterisk, audio/video doesn't matter. WARNING[977][C-0005] chan_sip.c: Rejecting secure audio stream without encryption details: audio 35684 RTP/SAVPF 109 0 8 101 -- Asterisk sends SIP/2.0 488 Not acceptable here Chrome: I've tried both sipml5 and jssip softphones and they both work. Even video + confbridge works with some minor quirks (lost connections sometimes, I guess plain old nat issues). Just relaying audio+video with confbridge to a handful of participants seems to use quite a bit of cpu thought. Screen-share: This works, but Confbridge is not very happy about a channel with video (vp8) and not audio and is printing this 80 times a second: WARNING[8919][C-] channel.c: Unable to find a codec translation path from (vp8) to (slin) WARNING[8919][C-] chan_sip.c: Asked to transmit frame type slin, while native formats is (vp8) read/write = unknown/unknown WARNING[8919][C-] channel.c: Don't know any of (vp8) formats How do you think about adding webrtc to a existing Asterisk/Kamailio environment? Do you use kamailio (websockets) as a front, a dedicated webrtc asterisk or something like webrtc2sip? How do you use / plan to implement webrtc in your environment? Any feedback is welcome. Thanks! -- Johan Wilfer -- _ -- 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] Webrtc and adventures with Asterisk 11
Hello, I was able to use webrtc2sip and connect audio calls in g729 passthrough and ulaw modes over a callus webpage js. However not tested Video. and it worked good even on AST 1.8.XX Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-71967196 Cell: +91-9820332422 On Mon, Apr 14, 2014 at 2:26 PM, Johan Wilfer li...@jttech.se wrote: Hi, I spent the past week experimenting with webrtc + asterisk 11.9.0-rc1 + opus/vb8 codec patch. This is interesting technology and I try to find out how to connect all the moving parts. Firefox: Neither sipml5 or jssip works with calls to asterisk, audio/video doesn't matter. WARNING[977][C-0005] chan_sip.c: Rejecting secure audio stream without encryption details: audio 35684 RTP/SAVPF 109 0 8 101 -- Asterisk sends SIP/2.0 488 Not acceptable here Chrome: I've tried both sipml5 and jssip softphones and they both work. Even video + confbridge works with some minor quirks (lost connections sometimes, I guess plain old nat issues). Just relaying audio+video with confbridge to a handful of participants seems to use quite a bit of cpu thought. Screen-share: This works, but Confbridge is not very happy about a channel with video (vp8) and not audio and is printing this 80 times a second: WARNING[8919][C-] channel.c: Unable to find a codec translation path from (vp8) to (slin) WARNING[8919][C-] chan_sip.c: Asked to transmit frame type slin, while native formats is (vp8) read/write = unknown/unknown WARNING[8919][C-] channel.c: Don't know any of (vp8) formats How do you think about adding webrtc to a existing Asterisk/Kamailio environment? Do you use kamailio (websockets) as a front, a dedicated webrtc asterisk or something like webrtc2sip? How do you use / plan to implement webrtc in your environment? Any feedback is welcome. Thanks! -- Johan Wilfer -- _ -- 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
Re: [asterisk-users] Asterisk and OSX
So few people use Asteisk on OSX that I doubt anyone will answer. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Manu Sent: Monday, April 14, 2014 4:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk and OSX Up, Does any body have solutions ? -- _ -- 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] Asterisk and OSX
On Apr 14, 2014, at 10:19 AM, Eric Wieling ewiel...@nyigc.com wrote: So few people use Asteisk on OSX that I doubt anyone will answer. Yup. I’d love for someone to say otherwise, but the state of Asterisk on OS X appears to be “can be configured to barely work for development”, and if that’s your goal, it’s much simpler to just run it in Vagrant. You could use my Vagrantfile from octothorpe to get started: https://github.com/zigg/octothorpe/blob/master/Vagrantfile I think you’ve got a huge uphill battle ahead of you if you want to actually run a production service there. signature.asc Description: Message signed with OpenPGP using GPGMail -- _ -- 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] how to configure callcentric peer
On 11.9, trying to set up a callcentric peer: sip debug: --- SIP read from UDP:204.11.192.161:5060 --- INVITE sip:1777myccid@10.10.11.180:5060 SIP/2.0 v: SIP/2.0/UDP 204.11.192.161:5060;branch=z9hG4bK-6104e46ef4249814d16a2ffb990d f: sip:calling number@66.193.176.35;tag=3606475083-968127 t: sip:1914mycctelnum@ss.callcentric.com i: 18075985-3606475083-968...@msw2.telengy.net CSeq: 1 INVITE Max-Forwards: 8 m: sip:f1eb8ab7586b3f2b72742b5e4d43d78d@204.11.192.161:5060;transport=udp Supported: timer c: application/sdp l: 350 v=0 o=NexTone-MSW 2147483647 2147483647 IN IP4 204.11.192.161 s=sip call c=IN IP4 204.11.192.161 t=0 0 m=audio 50960 RTP/AVP 18 0 8 101 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - a=setup:actpass - --- (11 headers 16 lines) --- Sending to 204.11.192.161:5060 (NAT) Sending to 204.11.192.161:5060 (NAT) Using INVITE request as basis request - 18075985-3606475083-968...@msw2.telengy.net No matching peer for 'calling number' from '204.11.192.161:5060' --- Reliably Transmitting (NAT) to 204.11.192.161:5060 --- SIP/2.0 401 Unauthorized asterisk is trying to find a peer based on the _calling number_! Here's the callcentric peer based on its support pages: [callcentric] type=peer context=from-callcentric host=callcentric.com defaultuser=1777myccid ;defaultuser=1914mycctelnum secret=mysecret fromuser=1777myccid ;fromuser=1914mycctelnum fromdomain=callcentric.com ;fromdomain=ss.callcentric.com insecure=port,invite disallowed_methods=UPDATE directmedia=no videosupport=no disallow=all allow=ulaw As you can see I also tried matching to the callcentric telephone number instead of the id. No luck. The only number I can't match is the calling number. Any help appreciated. sean -- _ -- 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] AMI and pyst
On Thu, Apr 10, 2014 at 9:14 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Does anyone on this list use pyst for AMI purposes? If so, can you point me in the direction of some simple examples. There seems to be none anywhere online. Probably doesn't help that I'm not that experienced at python but not insurmountably so. Thanks in Advance Ish Hey Ish - This isn't directly answering your question, but I noticed no one chimed in. At Digium we don't use pyst for Python integration with Asterisk, so I don't have any experience with it. We do, however, use starpy (https://github.com/asterisk/starpy) extensively in the Asterisk Test Suite. It does lock you into using twisted (https://twistedmatrix.com/trac/) - which has both pros and cons - but it may be a viable alternative for you if pyst doesn't work out. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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] AMI and pyst
On 14 April 2014 16:34, Matthew Jordan mjor...@digium.com wrote: On Thu, Apr 10, 2014 at 9:14 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Does anyone on this list use pyst for AMI purposes? If so, can you point me in the direction of some simple examples. There seems to be none anywhere online. Probably doesn't help that I'm not that experienced at python but not insurmountably so. Thanks in Advance Ish Hey Ish - This isn't directly answering your question, but I noticed no one chimed in. At Digium we don't use pyst for Python integration with Asterisk, so I don't have any experience with it. We do, however, use starpy (https://github.com/asterisk/starpy) extensively in the Asterisk Test Suite. It does lock you into using twisted (https://twistedmatrix.com/trac/) - which has both pros and cons - but it may be a viable alternative for you if pyst doesn't work out. Matt Hi Matt Thanks for the reply. I actually chose pyst as Billy Chia said that's what you guys used when I was at Astricon last year... Anyway, I overcame my initial hurdle but do want to try out the alternatives before I commit to one library -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] how to configure callcentric peer
wild guess would be a conflict on host= setting. there might be another entity on your sip.conf which have type=friend and host=callcentric.com or host=204.11.192.161 Kelvin Chua On Mon, Apr 14, 2014 at 8:01 AM, Sean Darcy seandar...@gmail.com wrote: On 11.9, trying to set up a callcentric peer: sip debug: --- SIP read from UDP:204.11.192.161:5060 --- INVITE sip:1777myccid@10.10.11.180:5060 SIP/2.0 v: SIP/2.0/UDP 204.11.192.161:5060;branch=z9hG4bK-6104e46ef4249814d16a2ffb990d f: sip:calling number@66.193.176.35;tag=3606475083-968127 t: sip:1914mycctelnum@ss.callcentric.com i: 18075985-3606475083-968...@msw2.telengy.net CSeq: 1 INVITE Max-Forwards: 8 m: sip:f1eb8ab7586b3f2b72742b5e4d43d78d@204.11.192.161:5060;transport=udp Supported: timer c: application/sdp l: 350 v=0 o=NexTone-MSW 2147483647 2147483647 IN IP4 204.11.192.161 s=sip call c=IN IP4 204.11.192.161 t=0 0 m=audio 50960 RTP/AVP 18 0 8 101 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - a=setup:actpass - --- (11 headers 16 lines) --- Sending to 204.11.192.161:5060 (NAT) Sending to 204.11.192.161:5060 (NAT) Using INVITE request as basis request - 18075985-3606475083-968...@msw2.telengy.net No matching peer for 'calling number' from '204.11.192.161:5060' --- Reliably Transmitting (NAT) to 204.11.192.161:5060 --- SIP/2.0 401 Unauthorized asterisk is trying to find a peer based on the _calling number_! Here's the callcentric peer based on its support pages: [callcentric] type=peer context=from-callcentric host=callcentric.com defaultuser=1777myccid ;defaultuser=1914mycctelnum secret=mysecret fromuser=1777myccid ;fromuser=1914mycctelnum fromdomain=callcentric.com ;fromdomain=ss.callcentric.com insecure=port,invite disallowed_methods=UPDATE directmedia=no videosupport=no disallow=all allow=ulaw As you can see I also tried matching to the callcentric telephone number instead of the id. No luck. The only number I can't match is the calling number. Any help appreciated. sean -- _ -- 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] Alembic - Asterisk 11
I've had years of experience using ODBC for CDR, SIP, and extensions with Asterisk. One thing that has been problematic in the past is with documentation as far as database tables changing between versions (even within minor releases, though that was back in the 1.4 days). I was excited to see there is a plan for better managing that on Asterisk 12 via Alembic. All that being said, are there any plans to implement that with Asterisk 11, since that is the current LTS release? Or are we pretty sure the table structure won't be changing within that version through the rest of its lifespan, making such an effort a waste? Thanks, Josh -- _ -- 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] Archlinux dahdi start script
I asked a similar question yesterday but unfortunately I somehow got disconnected from this group and I may have missed a response. Wanting a systemd start script for dahdi for archlinux. The install does not seem to make one. Has anyone created one or is there a way to create one from the make config scripts? I really don't need to load any modules other than dahdi itself for my application so I guess it oculd be as simple as 'modprobe dahdi' for the load. Also I have not built asterisk yet in archlinux. Does it create a systemd script with its make config? Doug-- _ -- 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
