| VJFROMGT | Darkness Source server http://pastebin.ca/495218 destination server http://pastebin.ca/495221 |
| tuan_modulis | is there a way to dynamically change the timeout of a call initiated by the Dial app? like during the call |
| [TK]D-Fender | tuan_modulis: Nope. |
| tuan_modulis | oh well :( |
| stoffell | [TK]D-Fender, i'm outta here, just wanted to let you know the app_page works perfect with the polycom's for paging. cheers ;) |
| [TK]D-Fender | stoffell: np |
| flujan | hi guys... I am using asterisk with X-lite... but it is not working with NAT. When I put the nat=yes on the sip.conf file, the softphone register... but I have no audio from both sides of the line. I research a bit about NAT and there is a lot of configurations... did you guys could point me a specific documentation that works? |
| [TK]D-Fender | ~sipnat |
| jbot | i heard sipnat is for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| [TK]D-Fender | flujan: Many things need to be done depending on WHO is behind NAT |
| LeddyHM | nats are bad, mmmmkay |
| flujan | [TK]D-Fender, the server with asterisk is behind NAT and the softphone is also behind a NAT. both sides have a router. |
| Qwell[] | [TK]D-Fender: hey |
| [TK]D-Fender | Qwell[]: y0 |
| Qwell[] | [TK]D-Fender: polycom.xml - what happens if you omit voIpProt.server.1.address? |
| shapr | Hi, I've started asterisk with debug, and I get REGISTER attempt 1 to username@hostname.com \n [May 18 12:27:32] NOTICE[4472]: chan_sip.c:12157 handle_response_peerpoke: Peer 'hostname' is now Reachable. (166ms / 2000ms). Does that mean I should be able to call through asterisk now? |
| shido6 | :) |
| [TK]D-Fender | flujan: you need to specify under [general] "nat=yes" , "canreinvite=no" , "localnet=" , "externip=" (or EXTERNHOST + EXTERNREFRESH). For the phone's entry you'll just need "nat=yes" |
| shapr | Also, can someone suggest softphones that should work with asterisk? Hopefully something that's already a debian/unstable package? |
| [TK]D-Fender | Qwell[]: That is not a standard polycom config file name format, and it would depend wherelse a server may be defined |
| Qwell[] | and reg.1.server.1.address |
| [TK]D-Fender | Qwell[]: I usually specify my server in sip.cf and jsut specif the user/pass in the "phoneXXX.cfg" for each phone |
| Qwell[] | what happens if you omit it though? |
| flujan | thanks [TK]D-Fender I will check this options now... :) |
| sweeper | hey, what are those short sms-only numbers called that all those ringtones/googlesearch/whatever services get called? and how do you go about getting one? |
| [[blah]asfd | I am trying to set queue_prio in a macro, but it does not seem to be working. all of my calls coming in do not get the prio set unless i do it right from the line before the queue it's self. here is how i do it: http://pastebin.ca/495252. can anyone suggest a better way to make this work? or point out what I am doing wrong? |
| saftsack | hi, if i originate a call over the manager api i get no voice in this call. is this a common bug/configuration error? |
| sweeper | saftsack: no |
| [TK]D-Fender | QwellAs in your phone's configs (however many you have) don't specify ANY server amongst them? |
| savaticus | it means you prolly didnt properly spec the channels |
| saftsack | Dial("Local/52@von-intern-e9de,1", "SIP/0506101@patton") Sweeper this dial string should be ok ... :( |
| Qwell[] | [TK]D-Fender: yeah |
| [TK]D-Fender | Qwell[]: Next I presum that it'll fallback to DHCP supplied server (No, NOT Opt 66), and barring even that... Dunno :) |
| sweeper | saftsack: what is this "Local" business? |
| Qwell[] | dhcp supplied server? |