00:19.26 | qakhan | i was getting message No audio format found to offer. Cancelling call |
00:19.26 | qakhan | [TK]D-Fender asked me to check sip show pree preename |
00:19.26 | qakhan | hen i checked there was Codecs : (nothing) in sip show peer |
00:19.26 | qakhan | but in sip config there was disallow = all and allow = ulaw |
00:19.26 | qakhan | i reloaded the asterisk many time, reloaded the sip. but still Codecs : (nothing) |
00:19.26 | qakhan | then i removed the [vitelity-outbound] config (from line 1165 in sip.conf) and recreate after some lines (1178 in sip.conf) |
00:19.27 | qakhan | then it started working |
00:19.27 | qakhan | i am curious how these changes made differnece ? |
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00:56.14 | [TK]D-Fender | You never showed proper or full configs your PB didn't match the debug and I don't trust the tiny pieces it took you forever to provide |
01:12.26 | qakhan | [TK]D-Fender when you asked me to send sip show peer, i sent you PB but you left that time. i noticed that Codec : (noting) in sip show peer |
01:12.55 | qakhan | but in sip.conf there was allow=ulaw |
01:13.42 | qakhan | i reloaded the asterisk, reloaded sip but it was not taking the allow=ulaw |
01:13.53 | qakhan | then i removed the config |
01:14.25 | qakhan | recreated the config in sip then asterisk took allow=ulaw |
01:14.42 | qakhan | which make me curious what happened |
01:15.38 | qakhan | just removed the config, reload asterisk, then make same config and reload the asterisk, booom its working |
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08:17.15 | wasanzy | Hello |
08:18.13 | wasanzy | I have queue members set. I want to add to my diaplan such that, if a call comes in and the calling phone number matches a phone number in the queue, an actions should be taken. is this possible? |
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08:55.38 | ixyd | wasanzy, should be no problem by using the queue_member_list() function |
08:56.18 | wasanzy | ixyd: ok, but that will return a list of all members right? |
08:56.38 | ixyd | i think so, yes |
08:56.57 | ixyd | try to combine it with the regex function?! |
08:57.10 | wasanzy | ok |
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09:35.10 | andycol_500 | hi guys |
09:35.26 | andycol_500 | how do i set the dialplan to go to the next priority if a queue has no answer |
09:35.38 | andycol_500 | so for example ring a queue for 10seconds and if no agents answer go to s,2 |
09:39.57 | andycol_500 | as ,8 doesnt seem to work |
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11:33.54 | wasanzy | is the CHANNEL(peername) same us CALLERID? |
11:37.21 | Samot | No, one would be the name of the channel peer and the other would be the caller id. |
11:37.40 | wasanzy | ok |
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12:16.21 | FarhaadN | hi every one |
12:16.34 | FarhaadN | can i set queue maxlen = 0 |
12:17.04 | FarhaadN | i want to fail call, when all agents in queue are busy |
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12:56.22 | ixyd | FarhaadN have a look at the joinempty option |
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13:35.38 | FarhaadN | ixyd: joinempty is for when extentions unavailable |
13:35.48 | FarhaadN | i want to use this for when all agents busy |
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13:46.50 | Samot | Busy = Unavailable. |
13:46.54 | Samot | Oh. |
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17:30.10 | polysics | hello! anyone is a SIPp user, by chance? |
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21:27.29 | mkresin | hey folks, are you aware that it is impossible to view an gerrit issue as anon user? I've tracked the issue down to the installed emoticons plugin which seams to tries to load resources from an uri that is only available for authenticated users |
21:29.08 | file | silly |
21:29.08 | file | let me remove it |
21:29.22 | file | I thought it was just the last Gerrit upgrade that did it |
21:29.47 | danielyk | Hi since a few days I receive following error after hangup of a follow me call: wrong response code 0 http://pastebin.com/6Fj5QK6H |
21:30.37 | file | mkresin: fixed! thanks |
21:31.53 | mkresin | file: confirmed! works again. thanks |
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