Re: [asterisk-users] How to Busy signals on DAHDI
Il 05/02/2014 8.42, Olivier ha scritto: channel then it depends upon what you have the priindication option set to. With priindication=outofband then a busy cause code is sent to the network and the call is hung up. With priindication=inband then a busy tone is sent after a possible PROGRESS message. I'm gonna check what's really happening from caller's perspective when such parameter is set. I would expect a busy tone (and the opportunity for automatic redialing). Sending a busy signalling on D channel (priindication=outofband) will go back into the telephone network exactly the same way the called party was really busy... so no charge, busy tone, not completed call on the caller side. Just a couple of notes about priindication=inband. - Usually sending in-band audio on a not connected state is not allowed unless a special setup is done by the isdn provider. - While sending in-band audio or busy signalling on D channel will make no difference (from the caller perspective) if the call is originated from an analog pstn line anything else (gsm-isdn, sip-isdn...) will fail to get a proper state for that call. Definitely I will never use it unless it is required, like the mandatory price information message to be sent **before** connection if you run a value added incoming service billed to the caller. -- TeeBX VoIP communication platform (coming soon) http://code.google.com/p/teebx/ --- Lightweight++ Business Friendly++ Open++ -- _ -- 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] Looking for some guidance with the Asterisk 12 ARI/API
Hi - I figured this was probably the best place to ask this question Is there anyone that has done anything practical with the API and/or Real Time Database config? If so, I would like to pick your brains if I may. Thanks - G -- _ -- 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] Looking for some guidance with the Asterisk 12 ARI/API
On 14-02-06 11:10 AM, James Wystead wrote: Hi - I figured this was probably the best place to ask this question Is there anyone that has done anything practical with the API and/or Real Time Database config? We actually have a new mailing list[1] that is focused on discussion about the new Asterisk REST Interface and application development itself. Is this what you are looking to talk about? It's a bit confusing because the REST interface really has nothing to do with the realtime architecture for databases. [1] http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.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] Looking for some guidance with the Asterisk 12 ARI/API
We run a multi node, multi tenanted hosted VoIP service using centralised databases for sip/extensions/voicemail configuration allowing resellers and end users to make updates to their walled garden themselves. We're using asterisk 1.8 but Realtime is no different on asterisk 12 (with the exception of PJSIP). Not done anything with the ARI. On 6 February 2014 15:10, James Wystead szilvertho...@gmail.com wrote: Hi - I figured this was probably the best place to ask this question Is there anyone that has done anything practical with the API and/or Real Time Database config? If so, I would like to pick your brains if I may. Thanks - G -- _ -- 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 -- 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] auto-answer call
hi when i try to this with page() exten = 506,1,SIPAddHeader(Call-Info:__\; answer-after=0) exten = 506,n,page(SIP/105) CLIAccepting call from '0661xx' to '506' on channel 1/13, span 1 -- Executing [506@default:1] SIPAddHeader(DAHDI/13-1, Call-Info:__; answer-after=0) in new stack -- Executing [506@default:2] Page(DAHDI/13-1, SIP/105) in new stack -- Called 105 -- DAHDI/13-1 Playing 'beep' (language 'en') -- SIP/105-00c7 is ringing -- SIP/105-00c7 is ringing -- SIP/105-00c7 is ringing -- Created MeetMe conference 1023 for conference '1894843837d' -- SIP/105-00c7 is ringing -- Span 1: Channel 1/13 got hangup, cause -1 -- Hungup 'DAHDI/pseudo-358137724' == Spawn extension (default, 506, 2) exited non-zero on 'DAHDI/13-1' -- Hungup 'DAHDI/13-1' and the call hungup when i use the Dial the sip/105 still ringing thanks and regards 2014-02-05 Larry Moore lmo...@omninet.net.au: On 6/02/2014 2:21 AM, Salaheddine Elharit wrote: thanks for your response , i test this solution but the issue still the same any other solution thanks and regards 2014-02-04 Steve Edwards asterisk@sedwards.com mailto:asterisk@sedwards.com: On Tue, 4 Feb 2014, Salaheddine Elharit wrote: i have asterisk 1.4.43 installed and i want to configure the auto-answer exten = 506,1,SIPAddHeader(Call-Info:__\; answer-after=0) I'm just a 1.2 Luddite... I have this for a Sipura: exten = _!.,n,sipaddheader(Call-Info:\__;answer-after=0) Maybe the quotes or the space after the semi-colon? Maybe wireshark would yield a clue? -- Thanks in advance, Here is a list of headers used for various vendors, I can't remember which one is for Polycom. SIPAddHeader(Alert-Info: Ring Answer); SIPAddHeader(Alert-Info: Info=Alert-Autoanswer); SIPAddHeader(Call-Info:\;Answer-After=0); SIPAddHeader(P-Auto-Answer: normal); SIPAddHeader(Answer-Mode: Auto); Larry. -- _ -- 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] multicastRTP source interface
I have an asterisk 11.4.0 server with two interfaces, eth0 and eth1. Eth0 has a default gateway on it, eth1 is connected the subnet that has my phones registered. I'd like to use the multicastRTP driver to do paging. However, when a phone dials an extension with multicastRTP, the multicast stream goes to the primary interface (eth0) and it really needs to go to eth1. Is there a way to specify which interface the rtp is sourced from? Matt Hoskins | NPG Corp | Systems Architect 816.749.2815 -- _ -- 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] SPA112 Won't stay up
Hi all, I have an SPA112 that in sitting behind a Ubee cable modem. The internet link is solid, but the device becomes unreachable within a day or so of being rebooted. Then the customer goes to reboot the device, they report that all 4 lights are lit. The ISP reports that the device does respond to ping, so it's not completely dead. I've had the same symptoms with SPA303's sitting behind Ubee modems. So, is there some configuration setting on the SPA that I can set to make this device more stable? Mike. -- _ -- 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] SPA112 Won't stay up
How long is the registration timeout? If the device is behind a router/firewall, then you need to set a registration timeout lower than the state table life in the router/firewall. I usually set my devices to just 2 minutes and it works almost all the time. Most Cisco devices have a very long timeout of 3600 seconds. Leandro 2014-02-06 17:18 GMT+01:00 Mike Diehl mdiehlena...@gmail.com: Hi all, I have an SPA112 that in sitting behind a Ubee cable modem. The internet link is solid, but the device becomes unreachable within a day or so of being rebooted. Then the customer goes to reboot the device, they report that all 4 lights are lit. The ISP reports that the device does respond to ping, so it's not completely dead. I've had the same symptoms with SPA303's sitting behind Ubee modems. So, is there some configuration setting on the SPA that I can set to make this device more stable? Mike. -- _ -- 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] SPA112 Won't stay up
I've got the registration period set to 15 minutes. However, I've got similar devices all over the place that don't seem to have this unreliability issue. The way I solved it with the SPA303 that I had in the office was to replace the Ubee modem with a different make/model. That's not an option in this particular case, though. Mike. On Thu, Feb 6, 2014 at 11:27 AM, Leandro Dardini ldard...@gmail.com wrote: How long is the registration timeout? If the device is behind a router/firewall, then you need to set a registration timeout lower than the state table life in the router/firewall. I usually set my devices to just 2 minutes and it works almost all the time. Most Cisco devices have a very long timeout of 3600 seconds. Leandro 2014-02-06 17:18 GMT+01:00 Mike Diehl mdiehlena...@gmail.com: Hi all, I have an SPA112 that in sitting behind a Ubee cable modem. The internet link is solid, but the device becomes unreachable within a day or so of being rebooted. Then the customer goes to reboot the device, they report that all 4 lights are lit. The ISP reports that the device does respond to ping, so it's not completely dead. I've had the same symptoms with SPA303's sitting behind Ubee modems. So, is there some configuration setting on the SPA that I can set to make this device more stable? Mike. -- _ -- 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
Re: [asterisk-users] SPA112 Won't stay up
Is there another router in the mix? Put the cable modem in bridge mode and attAch a real router. http://401stblow.wordpress.com/2012/10/21/fixing-time-warner-cable-ubee-modem-connectivity-issues/ On Thursday, February 6, 2014, Mike Diehl mdiehlena...@gmail.com wrote: I've got the registration period set to 15 minutes. However, I've got similar devices all over the place that don't seem to have this unreliability issue. The way I solved it with the SPA303 that I had in the office was to replace the Ubee modem with a different make/model. That's not an option in this particular case, though. Mike. On Thu, Feb 6, 2014 at 11:27 AM, Leandro Dardini ldard...@gmail.comjavascript:_e(%7B%7D,'cvml','ldard...@gmail.com'); wrote: How long is the registration timeout? If the device is behind a router/firewall, then you need to set a registration timeout lower than the state table life in the router/firewall. I usually set my devices to just 2 minutes and it works almost all the time. Most Cisco devices have a very long timeout of 3600 seconds. Leandro 2014-02-06 17:18 GMT+01:00 Mike Diehl mdiehlena...@gmail.comjavascript:_e(%7B%7D,'cvml','mdiehlena...@gmail.com'); : Hi all, I have an SPA112 that in sitting behind a Ubee cable modem. The internet link is solid, but the device becomes unreachable within a day or so of being rebooted. Then the customer goes to reboot the device, they report that all 4 lights are lit. The ISP reports that the device does respond to ping, so it's not completely dead. I've had the same symptoms with SPA303's sitting behind Ubee modems. So, is there some configuration setting on the SPA that I can set to make this device more stable? Mike. -- _ -- 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 -- -- www.ringfree.biz 828-575-0030 -- _ -- 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] SPA112 Won't stay up
On 02/06/2014 09:25 AM, Mike Diehl wrote: I've got the registration period set to 15 minutes. However, I've got similar devices all over the place that don't seem to have this unreliability issue. The way I solved it with the SPA303 that I had in the office was to replace the Ubee modem with a different make/model. That's not an option in this particular case, though. More then likely, the replacement router/modem had a different timeout and it was a luck of the draw that it worked. In routers that allow me to set the UDP timeout, I normally set the timeout to 90 seconds. Most routers that done offer a setting for this are usually set to 90 or 120 seconds Then I usually set my registration time on the ATA's to 60 seconds. The devices I seem to have most issues with are SonicWall routers. Jim Mike. On Thu, Feb 6, 2014 at 11:27 AM, Leandro Dardini ldard...@gmail.com wrote: How long is the registration timeout? If the device is behind a router/firewall, then you need to set a registration timeout lower than the state table life in the router/firewall. I usually set my devices to just 2 minutes and it works almost all the time. Most Cisco devices have a very long timeout of 3600 seconds. Leandro 2014-02-06 17:18 GMT+01:00 Mike Diehl mdiehlena...@gmail.com: Hi all, I have an SPA112 that in sitting behind a Ubee cable modem. The internet link is solid, but the device becomes unreachable within a day or so of being rebooted. Then the customer goes to reboot the device, they report that all 4 lights are lit. The ISP reports that the device does respond to ping, so it's not completely dead. I've had the same symptoms with SPA303's sitting behind Ubee modems. So, is there some configuration setting on the SPA that I can set to make this device more stable? Mike. -- _ -- 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 -- Jim Lucas http://www.cmsws.com/ http://www.cmsws.com/examples/ -- _ -- 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] SPA112 Won't stay up
Unfortunately, we plug straight into the Ubee and the ISP will not support any other modem. GRRr.. Mike. On Thu, Feb 6, 2014 at 12:34 PM, David Wessell da...@ringfree.biz wrote: Is there another router in the mix? Put the cable modem in bridge mode and attAch a real router. http://401stblow.wordpress.com/2012/10/21/fixing-time-warner-cable-ubee-modem-connectivity-issues/ On Thursday, February 6, 2014, Mike Diehl mdiehlena...@gmail.com wrote: I've got the registration period set to 15 minutes. However, I've got similar devices all over the place that don't seem to have this unreliability issue. The way I solved it with the SPA303 that I had in the office was to replace the Ubee modem with a different make/model. That's not an option in this particular case, though. Mike. On Thu, Feb 6, 2014 at 11:27 AM, Leandro Dardini ldard...@gmail.comwrote: How long is the registration timeout? If the device is behind a router/firewall, then you need to set a registration timeout lower than the state table life in the router/firewall. I usually set my devices to just 2 minutes and it works almost all the time. Most Cisco devices have a very long timeout of 3600 seconds. Leandro 2014-02-06 17:18 GMT+01:00 Mike Diehl mdiehlena...@gmail.com: Hi all, I have an SPA112 that in sitting behind a Ubee cable modem. The internet link is solid, but the device becomes unreachable within a day or so of being rebooted. Then the customer goes to reboot the device, they report that all 4 lights are lit. The ISP reports that the device does respond to ping, so it's not completely dead. I've had the same symptoms with SPA303's sitting behind Ubee modems. So, is there some configuration setting on the SPA that I can set to make this device more stable? Mike. -- _ -- 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 -- -- www.ringfree.biz 828-575-0030 -- _ -- 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] Fax buffer overflow detected
All; I'm running Asterisk 1.8.15-cert3 with the newest version of spandsp. I've even tried unloading that and using Digium's FFA module but I receive the same error on an outbound transmission: [2014-02-06 14:35:14] ERROR[19066]: udptl.c:294 encode_open_type: UDPTL (SIP/XXX_outbound-): Buffer overflow detected (59 + 127 175) I only get this with one specific upstream provider. Has anyone seen this before? Any help at all would be greatly appreciated. Regards; John -- _ -- 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] SPA112 Won't stay up
On 2/6/14, 11:18 AM, Mike Diehl wrote: Hi all, I have an SPA112 that in sitting behind a Ubee cable modem. The internet link is solid, but the device becomes unreachable within a day or so of being rebooted. Then the customer goes to reboot the device, they report that all 4 lights are lit. The ISP reports that the device does respond to ping, so it's not completely dead. I've had the same symptoms with SPA303's sitting behind Ubee modems. So, is there some configuration setting on the SPA that I can set to make this device more stable? I have seen similar behavior before on the SPA122. I could ping it, open the web page, etc... but it would not register until I rebooted it. Upon closer examination I could see that the SPA122 was only working partially. The voice modules appeared to be dead thus it would not register. You could see this by looking at the stats page and the lines would not show any stats at all or even if they were ON or OFF hook. A reboot would fix it for a few days. The solution was to get a new SPA122. My take on this is that it was a hardware issue, not a software one that could be fixed with configuration settings. What I hate about these units is that they take more that 1 minute to boot and register. The SPA2102 only took about 15 seconds. That really sucks when you have a customer on the line and are troubleshooting an issue that requires a reboot. Mike. -- Technical Support http://www.cellroute.net -- _ -- 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] SPA112 Won't stay up
Even if they don't support another modem, you can usually use another router. Just put the Ubee in bridge mode and slap a router behind it. This is our typical best practice install. You just can't trust the piece of shit routers that ISP's send out these days.. dw On Thu, Feb 6, 2014 at 12:54 PM, Mike Diehl mdiehlena...@gmail.com wrote: Unfortunately, we plug straight into the Ubee and the ISP will not support any other modem. GRRr.. Mike. On Thu, Feb 6, 2014 at 12:34 PM, David Wessell da...@ringfree.biz wrote: Is there another router in the mix? Put the cable modem in bridge mode and attAch a real router. http://401stblow.wordpress.com/2012/10/21/fixing-time-warner-cable-ubee-modem-connectivity-issues/ On Thursday, February 6, 2014, Mike Diehl mdiehlena...@gmail.com wrote: I've got the registration period set to 15 minutes. However, I've got similar devices all over the place that don't seem to have this unreliability issue. The way I solved it with the SPA303 that I had in the office was to replace the Ubee modem with a different make/model. That's not an option in this particular case, though. Mike. On Thu, Feb 6, 2014 at 11:27 AM, Leandro Dardini ldard...@gmail.com wrote: How long is the registration timeout? If the device is behind a router/firewall, then you need to set a registration timeout lower than the state table life in the router/firewall. I usually set my devices to just 2 minutes and it works almost all the time. Most Cisco devices have a very long timeout of 3600 seconds. Leandro 2014-02-06 17:18 GMT+01:00 Mike Diehl mdiehlena...@gmail.com: Hi all, I have an SPA112 that in sitting behind a Ubee cable modem. The internet link is solid, but the device becomes unreachable within a day or so of being rebooted. Then the customer goes to reboot the device, they report that all 4 lights are lit. The ISP reports that the device does respond to ping, so it's not completely dead. I've had the same symptoms with SPA303's sitting behind Ubee modems. So, is there some configuration setting on the SPA that I can set to make this device more stable? Mike. -- _ -- 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 -- -- www.ringfree.biz 828-575-0030 -- _ -- 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 -- -- www.ringfree.biz 828-575-0030 -- _ -- 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] Rejecting a call as if the extension does not exist.
I'm trying to address a problem with users transferring to invalid
destinations.
In my sip peer I'm setting both __FORWARD_CONTEXT and __TRANSFER_CONTEXT to
a context with a extension defined below to set some CDR variables before
the call is transferred.
[customer-forward]
exten = _X.,1,Progress()
exten = _X.,n,Gosub(do-billing,s,1${EXTEN}))
exten = _X,n,Goto(customer-internal,${EXTEN},1)
Now if my user Dials an invalid extension, Say '9595' from their sip phone
they get back an 'Address Incomplete' message from their phone because it's
not a valid extension defined in my dialplan.
If my user Transfers a call to '9595' the call gets transferred and then
hung up on as there's no '9595' destination.
I'd like to reject the calls in my customer-foward context that do not
exist in my customer-internal context.
I've tried doing something like:
exten = _X.,1,Progress()
same = n,ExecIf($[${DIALPLAN_EXISTS(customer-internal,${EXTEN},1)} =
0]?Hangup(28))
But that's still accepting the call as the _X makes it a Valid extension.
I'm looking for suggestions or ideas on a better way to do this.
--
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
--
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Re: [asterisk-users] Rejecting a call as if the extension does not exist.
You could have the call immediately return to the transferer -Original Message- From: John Kiniston johnkinis...@gmail.com Sender: asterisk-users-bounces@lists.digium.comDate: Thu, 6 Feb 2014 17:14:02 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Rejecting a call as if the extension does not exist. -- _ -- 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] Fax buffer overflow detected
On 7/02/2014 3:38 AM, Tech Support wrote: All; I’m running Asterisk 1.8.15-cert3 with the newest version of spandsp. I’ve even tried unloading that and using Digium’s FFA module but I receive the same error on an outbound transmission: [2014-02-06 14:35:14] ERROR[19066]: udptl.c:294 encode_open_type: UDPTL (SIP/XXX_outbound-): Buffer overflow detected (59 + 127 175) I only get this with one specific upstream provider. Has anyone seen this before? Any help at all would be greatly appreciated. Not sure if this relates to T38FaxMaxDatagram, here is an extract from a sample sip.conf file. ; In some cases, T.38 endpoints will provide a T38FaxMaxDatagram value (during T.38 setup) that ; is based on an incorrect interpretation of the T.38 recommendation, and results in failures ; because Asterisk does not believe it can send T.38 packets of a reasonable size to that ; endpoint (Cisco media gateways are one example of this situation). In these cases, during a ; T.38 call you will see warning messages on the console/in the logs from the Asterisk UDPTL ; stack complaining about lack of buffer space to send T.38 FAX packets. If this occurs, you ; can set an override (globally, or on a per-device basis) to make Asterisk ignore the ; T38FaxMaxDatagram value specified by the other endpoint, and use a configured value instead. ; This can be done by appending 'maxdatagram=value' to the t38pt_udptl configuration option, ; like this: ; ; t38pt_udptl = yes,fec,maxdatagram=400 ; Enables T.38 with FEC error correction and overrides ; ; the other endpoint's provided value to assume we can ; ; send 400 byte T.38 FAX packets to it. ; Larry. -- _ -- 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
