Re: [asterisk-users] UDP miss a hangup on SIP
2012-08-16 02:13, Jerry Geis skrev: Is it possible to miss a UDP SIP packet to hangup a call? Using 1.4.43 I had a call from on asterisk box (server) to a low end client (chan_alsa) not hangup. Could this be due to missed UDP SIP packet to hangup? Is there anyway for a client asterisk (chan_alsa again) to monitor the connection and if the channel is not there to hangup? In sip.conf you could use rtp-timers to hangup a call if the media-stream stops to flow. Look at these options in sip.conf: rtptimeout=60 rtpholdtimeout=300 rtpkeepalive=0 -- 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
[asterisk-users] Fax Detect on Demand
Using Asterisk 1.8.mumble. We would like to use fax detect on demand. Both chan_dahdi and chan_sip support setting fax detetect on a static basis, but no way I've been able to find to enable/disable it on demand in the dialplan. In 1.4 we used the NVFaxDetect 3rd party app, but that no longer appears to be maintained. Does anyone have any suggestions? -- _ -- 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] Requiring agent to confirm queue calls only when forwarded to external device
I'd like to allow my users to forward their calls using the forwarding feature on their SIP handsets and continue to receive Queue() calls. Currently I set the 'i' option in Queue() so that if a user forwards to their cell phone, or any other extension that has voicemail, the voicemail doesn't eat all the calls to the queue. I'm aware that I can configure the queue to require agents to acknowledge the call. However, most of the calls go to internal devices where confirmation isn't necessary, so I'd like to avoid the extra inconvenience in that most common case. What I'd like to do is somehow detect that a handset has responded with a SIP 302 response, and only when this is the case, require the agent to confirm humanness before answering the call from the queue. Any ideas on how this could be implemented? -- _ -- 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] TDM Fax
Has anyone experimented with increasing the DAHDI chunk size in improve fax reliability? If so, did it help, hurt, or not make any difference? -- _ -- 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] Requiring agent to confirm queue calls only when forwarded to external device
forward to a Local extension that has dialplan requiring the acknowledgement? On 16 August 2012 21:12, Phil Frost p...@macprofessionals.com wrote: I'd like to allow my users to forward their calls using the forwarding feature on their SIP handsets and continue to receive Queue() calls. Currently I set the 'i' option in Queue() so that if a user forwards to their cell phone, or any other extension that has voicemail, the voicemail doesn't eat all the calls to the queue. I'm aware that I can configure the queue to require agents to acknowledge the call. However, most of the calls go to internal devices where confirmation isn't necessary, so I'd like to avoid the extra inconvenience in that most common case. What I'd like to do is somehow detect that a handset has responded with a SIP 302 response, and only when this is the case, require the agent to confirm humanness before answering the call from the queue. Any ideas on how this could be implemented? -- __**__**_ -- 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-**usershttp://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] Grandstream VoIP phones
Who else on the list is using them, particularly in a hosted environment? We've just decided to transition to them as our primary recommendation instead of the Cisco SPA series. We did it because of the value and feature set, like having an inexpensive phone with a small BLF, which a lot of customers asked for. I'm wondering if others have tips they've learned along the way, or any advice they want to offer. Also anyone using the advanced features like the browser for anything useful? For those who haven't tried them, or who like us, didn't like their older models, take another look. We have been surprised at the value they give us. The prices are low, but the functionality and quality are high. They aren't Polycom 600s to be sure, but they are nice phones that have a huge set of features for a great price. Customers are liking them a lot. Has anyone used the new DECT phone? We currently use the Panasonic DECT phones but they are a nightmare to configure. If anyone wants to get in touch with them, our Grandstream contact is Dennis Ryan, dr...@grandstream.com . -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Requiring agent to confirm queue calls only when forwarded to external device
On Aug 16, 2012, at 6:25 PM, Tiago Geada wrote: forward to a Local extension that has dialplan requiring the acknowledgement? On 16 August 2012 21:12, Phil Frost p...@macprofessionals.com wrote: I'd like to allow my users to forward their calls using the forwarding feature on their SIP handsets and continue to receive Queue() calls. Currently I set the 'i' option in Queue() so that if a user forwards to their cell phone, or any other extension that has voicemail, the voicemail doesn't eat all the calls to the queue. I'd think that would require teaching all the users to forward to a different extension if they thought they could be receiving queue calls. My users probably aren't that good at following directions ;) Ultimately, I'm sure I could solve this problem by taking management of forwarding off the phone and into Asterisk, since then I'd absolutely have some flag indicating if forwarding is active or not. However, I was just hoping there was an easier way. I'm really happy with the forwarding interface on our current handsets, and I'd rather not go through the effort of changing their configuration, or changing the user experience if I can avoid it. -- _ -- 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
