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| 00:49.06 | knightr | Hi! Is there a way I can make sure I have the right sound prompts for my Asterisk version? |
| 00:50.01 | knightr | I have a problem with sayunixtime because it asks for a file which is not present on my server where it looks for it... |
| 00:51.02 | knightr | it's present elsewhere mind you, just not where it looks for it... |
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| 10:05.45 | _abc_ | Hello. Two questions: 1) is there a generic dsp filter plugin for * core or any channel which can filter audio frequencies in a tighter way? i.e. restrict to 500-3500Hz with quite steep skirts? I have trouble with hum and low frequency noises from cheap headsets. No, chaning headset is not going to happen, the noise is industrial in nature. 2) what's the situation with low cost "personal" iax2 trunk and te |
| 10:05.51 | _abc_ | rmination providers? I have been reading ... |
| 10:05.54 | _abc_ | ... some threads and it seems to be really low volume, I mean, there are >1100 known SIP providers but <400 IAX2 and most of those are no longer around (2012 data). I see * is QUITE popular in various rather commercial settings, yet no providers? |
| 10:06.23 | _abc_ | I read in one thread that *'s core model does not make use of multiple cpus (~2010 thread), is this still true? I.e. iax call routing through a pbx may not be able to use several present cores? |
| 10:07.05 | _abc_ | (this in the context of why providers prefer to provide SIP with direct rtp reinvite probably, vs iax2 without direct reinvite) |
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| 11:00.37 | _abc_ | ?? |
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| 11:51.15 | davlefou | Hi, how can i use Mixmonitor to create an mp3 file? |
| 12:43.51 | _0x5eb_ | hi, how could I avoid Asterisk to replace my FQDN by a hard-coded IP address in the Contact: header of a register SIP request? |
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| 13:00.42 | _0x5eb_ | I have lots of trouble trying to connect Asterisk to Cirpack: both servers seem to have fun randomly replacing their FQDN by an IP address in their From: and Contact: headers |
| 13:01.41 | _0x5eb_ | this of course breaks the SIP connections as soon as Asterisk replies to a Cirpack keep-alive OPTIONS request with a 404 Not Found |
| 13:03.05 | _abc_ | nat |
| 13:03.12 | _abc_ | nat + stun + etc |
| 13:09.52 | _0x5eb_ | _abc_, yes NAT is probably involved, but that's why I would like Asterisk to stop showing its LAN IP address in its Contact: headers and instead put its FQDN instead, is it feasible? |
| 13:10.48 | _0x5eb_ | I tried externhost=my.fqdn, fromdomain=my.fqdn and domain=my.fqdn, all without success |
| 13:12.58 | _0x5eb_ | (well domain=my.fqdn,sip-context to be more accurate) |
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| 13:28.41 | _abc_ | http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html _0x5eb_ see 1.4.7 -- you can't make it send a fqdn there iirc, but you can do servlookup=no or such |
| 13:36.18 | _0x5eb_ | _abc_, well thanks, but I found that actually my issue was finally not caused by the IP address/FQDN mismatch (as long as I have domain= for both my FQDN and my public IP address) |
| 13:38.19 | _0x5eb_ | I just found that I always put a wrong SIP context in my domain= in my [general], and for some reason it was overridden with my context= in my peer in Asterisk 1.6 |
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| 13:39.49 | _0x5eb_ | now that I upgraded to Asterisk 1.11, it seems to use the [general] domain= instead of the context= of the peer (which is actually more convenient in my case... once I set the correct SIP context in my domain= of my [general]) |
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| 14:53.33 | _0x5eb_ | any idea where we could find the documentation for "nat=force_rport"? |
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| 17:05.34 | davlefou | If i have an mixmonitor before an voicemail, is the mixmonitor ll start? |
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| 22:32.04 | _0x5eb_ | hi, with asterisk 11.13.1, I try to register an SIP account (register in [general] of my sip.conf) with defaultexpiry = 1800, hoping that it would be reflected in the REGISTER SIP requests |
| 22:32.45 | _0x5eb_ | however, I always see "Expires: 120" when analyzing the SIP packets sent by Asterisk to my peer |
| 22:33.45 | _0x5eb_ | as a result, the SIP registration is valid only 2 mn every 30 mn (since Asterisk does not even try to re-register after the 120s, but actually waits 1800s) |
| 22:34.28 | _0x5eb_ | how could I set the "Expires: " field of the outgoing REGISTER requests sent by Asterisk 1.11? |
| 22:38.26 | _0x5eb_ | OK, I found the solution, but it is tricky and I only found a reference to this syntax in http://lists.digium.com/pipermail/asterisk-dev/2010-May/043906.html : |
| 22:38.44 | _0x5eb_ | I should append ~1800 to my register lines in my sip.conf |
| 22:39.37 | lvlinux | _0x5eb_: defaultexpiry should do it if the file is being loaded. |
| 22:39.55 | lvlinux | _0x5eb_: you can also explicitly specify it on the register string |
| 22:43.32 | lvlinux | _0x5eb_: like register => user:pass@sipprovider.com~1800 |
| 22:44.39 | _0x5eb_ | lvlinux, well for some reason defaultexpiry is being ignored, but it works with ~1800, thanks |
| 22:45.07 | lvlinux | _0x5eb_: np :-) |
| 22:45.25 | WIMPy | has never seen that ~ thing mentioned anywhere. |
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