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| 06:31.38 | floyd20183 | Good day. Could someone perhaps assist in pointing me to a resource that will help me understand what this error means : ERROR[3039] chan_dahdi.c: PRI Span: 4 Unable to receive TEI from network in state 3(Establish awaiting TEI)! |
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| 07:30.49 | WIMPy | floyd20183: Either your connection is broken or you are on a ptp line. |
| 07:31.42 | floyd20183 | ok. Thanks. This appears in the logs about 10 to 20 times a day and we are able to make and receive calls. Sometimes it just drops an incoming call or will not allow us to answer so not sure if this is related |
| 07:32.49 | WIMPy | So you have a bad connection somewhere. |
| 07:33.12 | floyd20183 | Thanks. I will get them to have the telecoms company check out their end |
| 07:34.23 | floyd20183 | The phones are on a local lan so am happy with that end |
| 07:45.56 | TandyUK | anyone know of good apps (free or paid) that allow full featured CTI with asterisk, kinda like the Avaya Phone Manager app |
| 07:46.21 | TandyUK | this does call popups, for answer, send to voicemail, etc, call history, phonebook searching, and outlook integration with click to call |
| 07:52.18 | Rasputin3711 | may be phonerlite |
| 07:54.30 | TandyUK | phonerlite is a softphone though, i just want the CTI, we have proper phones on desks (yealink T41's mainly) |
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| 09:17.21 | polysics | hello! question: can specifying a rport in the INVITE change what codecs Asterisk offers? |
| 09:19.30 | polysics | I have an odd situation where one port works and the other is rejected on no codecs |
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| 09:43.18 | Samot | rport has nothing to do with codecs |
| 09:43.23 | Samot | It's the Received Port. |
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| 09:53.02 | refeaime | Hello! |
| 09:53.43 | refeaime | Could someone help me with creating trunk via freepbx? |
| 09:54.19 | WIMPy | Ask in #freepbx |
| 09:54.31 | refeaime | WIMPy: Thank you. |
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| 10:04.56 | Samot | Thanks. |
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| 10:13.38 | rwsq1 | in pjsip show channelstats should I believe what the codec says? I have strong reason to doubt it. |
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| 10:30.21 | skrusty | anyone here used res_chan_stats? |
| 10:31.14 | Samot | Nope |
| 10:32.15 | rwsq1 | also what does it mean if codec is empty? |
| 10:32.29 | rwsq1 | surely it must know which codec is being used |
| 10:42.42 | polysics | Samot: right, sorry. I took a second look at sip.conf and I had a space too much. |
| 10:43.51 | TandyUK | so, asterisk 11, and RTCP-XR and/or HEP |
| 10:44.42 | TandyUK | we have a supplier provided platform, with centos 5 :(, and we use TLS/SRTP so things like sipgrep, sngrep, etc wont compile with TLS support on it |
| 10:45.22 | TandyUK | what other options can anyone think of for getting our realtime data out of asterisk (even by doing a pcap>netcat feed from higher up in the network) |
| 10:50.27 | Samot | Wouldn't a sip debug work as well? |
| 10:50.34 | Samot | Or a tcpdump? |
| 10:51.29 | TandyUK | for tcpdump to work, we'd need to feed the pcap out to another machine (running centos6 /debian/etc) with sngrep/sipgrep/etc on it along with our keys |
| 10:51.46 | TandyUK | i can capture non TLS traffic no problem on the C5 box |
| 10:52.11 | TandyUK | im looking to feed this data into sipcapture/homer |
| 10:52.12 | polysics | I don't see many ways around that that do not include another box with the TLS tools installed |
| 10:52.27 | Samot | a pcap file is a text file. |
| 10:52.34 | Samot | You can read the pcap file straight on the machine. |
| 10:52.36 | TandyUK | yeah pcap is just tcpdump > file |
| 10:52.51 | TandyUK | yes, but the machine cant compile the TLS compnents needed for sngrep/sipgrep/etc |
| 10:52.59 | TandyUK | so it _has_ to be done on another machine |
| 10:53.27 | TandyUK | and i really want this realtime, hence the pcap>netcat option to feed it out to the other machine in realtime |
| 10:53.40 | Samot | Realtime to view or real time to log? |
| 10:53.53 | Samot | Because sip set debug will view the TLS traffic. |
| 10:53.57 | TandyUK | realtime to feed into sipcapture in realtime |
| 10:54.06 | Samot | Then you need tcpdum. |
| 10:54.11 | TandyUK | so we can see call quality stats for in-progess calls |
| 10:54.16 | Samot | Then you need tcpdump. Which will also get the traffic. |
| 10:54.24 | TandyUK | yeah tcpdump / pcap same thing ;) |
| 10:54.33 | Samot | Then you have an option. |
| 10:54.36 | TandyUK | id actually capture from our perimiter router |
| 10:54.41 | TandyUK | not the asterisk box |
| 10:55.12 | TandyUK | effectively port-mirror the asterisk box to a 'logger' machine of some sort |
| 10:55.30 | TandyUK | was just wondering if theres any easier ways anyone could think of :P |
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| 13:32.09 | Ahmadi | Hi |
| 13:33.14 | Ahmadi | When my softphone try to register in Asterisk, and failed, the asterisk ban my IP ! then i can't retry any more. |
| 13:33.23 | Ahmadi | How can i disable this feature of Asterisk! |
| 13:33.32 | Ahmadi | i don't want bane any IP |
| 13:33.55 | Ahmadi | i don't want asterisk bane any IP, even if entered wrong username/password for a extension, |
| 13:34.01 | Ahmadi | Please help me |
| 13:35.42 | Rasputin3711 | Freepbx/Elastix ? |
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| 13:36.04 | Ahmadi | freepbx |
| 13:36.12 | Ahmadi | asterisk now 13 |
| 13:36.38 | Rasputin3711 | channel #freepbx |
| 13:39.16 | *** join/#asterisk avb (~avb@45.55.203.176) |
| 13:39.36 | avb | hey guys, can I check what codec is used on a channel from the dialplan? |
| 13:41.12 | [TK]D-Fender | avb, "core show function CHANNEL" |
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| 13:43.13 | avb | [TK]D-Fender: huh, forgot about this one. thank you buddy |
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| 15:26.32 | Phil-Work | can you play an audio file on a given channel with AMI? |
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| 15:50.06 | mjordan | Phil-Work: No. AMI can only do that by redirecting to a dialplan application that plays back audio files. |
| 15:50.26 | mjordan | Phil-Work: there are AMI commands for controlling a currently playing audio file, but not for arbitrarily starting it. |
| 15:50.43 | Phil-Work | or presumably, if it's an ongoing call, you create a conf bridge? |
| 15:50.47 | mjordan | ARI does have that capability, as your remote application is essentially the dialplan application |
| 15:51.33 | mjordan | If you have a two-party call, you can redirect all the parties into a confbridge, then play back an audio file into the confbridge. Or, if you wanted to play back to a single party, you could originate a Local channel that ChanSpys on one of the participants |
| 15:51.43 | mjordan | basically, you can use the dialplan :-) |
| 15:51.56 | Phil-Work | ok, thanks Matt :) |
| 15:51.59 | mjordan | np :-) |
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| 16:25.03 | *** join/#asterisk topro (~prousa@p578af414.dip0.t-ipconnect.de) |
| 16:27.40 | topro | hi there, someone could please give me a hint on how to further investigate my (SIP-NAT ?!?) issues? I got the following topology: |
| 16:28.24 | topro | Provider-SIP-Peer <---> FW/NAT <---> Asterisk <----> local SIP Clients |
| 16:28.43 | topro | you guess it, I got audio issues... |
| 16:29.00 | topro | call initialisation works in both directions (incoming and outgoing calls) |
| 16:30.00 | topro | the strange thing is that when getting the call from provider throug FW/NAT audio is working in both directions. but when calling from inside to provider i get the call established but no audio in either direction |
| 16:30.28 | Mango45 | Do you have directmedia=no and directrtpsetup=no configured in your sip.conf? Either in the [general] context, or for the peers in question? |
| 16:30.56 | topro | in that case "rtp set debug on" gives me rtp messages that it receives from my local client and sens to the provider peer, but no rtp messages coming back from the provider peer. |
| 16:30.57 | Mango45 | Actually, put it in [general] if it's not already there. |
| 16:30.59 | topro | any idea? |
| 16:31.18 | topro | let me see.. |
| 16:31.38 | Mango45 | Let's start with that because that is easiest. If you want to improve upon that, you can work your way up. |
| 16:33.06 | topro | what value would you suppose for nat= in sip.conf? |
| 16:34.11 | Mango45 | For now we will assume that the peers know their IP addresses and Asterisk should route audio to and from the IP specified in the SDP. Let's start with nat=no. |
| 16:34.16 | topro | <PROTECTED> |
| 16:34.23 | Mango45 | did you do sip reload? |
| 16:34.41 | topro | just reload which reloads whole config AFAIK |
| 16:34.48 | [TK]D-Fender | "sip show settings" <- |
| 16:35.22 | Mango45 | Since [TK]D-Fender is here I'll let him take over since he's more experienced than I am. |
| 16:35.46 | topro | any help very welcome :) |
| 16:36.32 | topro | http://pastebin.com/T1ycP4hM |
| 16:37.22 | topro | btw i got a lot of settings in specific section, not global so here is the relevant part of sip.conf... |
| 16:37.49 | Mango45 | Double check that 87.138.244.20 is your IP address. |
| 16:38.03 | [TK]D-Fender | your trunk peer should be "nat=no", "directmedia=no" |
| 16:38.19 | [TK]D-Fender | Also you need to have SIP + RTP forwarded to your server from your FW |
| 16:38.26 | topro | http://pastebin.com/GZyBxL8r |
| 16:38.51 | Mango45 | I would change your passwords for those SIP accounts. |
| 16:39.14 | topro | forwarding RTP means I have to do UDP port forwarding for the whole range of RTP ports specified, i.e. 10000-20000 ?!? |
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| 16:39.37 | [TK]D-Fender | yes |
| 16:39.40 | topro | Mango45: ?!? what passwords? |
| 16:39.43 | Phil-Work | tompaw, yes - but you can limit that range in rtp.conf |
| 16:40.22 | Phil-Work | eh, topro not tompaw |
| 16:40.46 | topro | Mango45: you think 7654321 is the real thing? hint: it isn't :) |
| 16:40.55 | Mango45 | Okay, good. :) |
| 16:41.21 | [TK]D-Fender | That's like the kind of password an idiot would use on their luggage... |
| 16:41.47 | topro | ok, I tried to get around without port forwarding, hoped sip_conntrack module might do the job |
| 16:42.16 | topro | what I still don't get is why does audio work in both directions when getting the call from outside? |
| 16:42.16 | [TK]D-Fender | your GW should NOT be screwing with SIP at all |
| 16:42.28 | [TK]D-Fender | turn off ALL SIP ALG's, helper, proxy's, etc |
| 16:43.07 | topro | yes, I read about that. so why is that stuff there in the first place if its more pain than helping? |
| 16:44.12 | [TK]D-Fender | it helps some situations. NOT this |
| 16:45.02 | [TK]D-Fender | Penicillin helps some people too ... except those who are deathly allergic to it... |
| 16:53.57 | topro | [TK]D-Fender: I'll give it a try. thanks for the pointer. I'll report... |
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| 18:28.41 | drmessano | Isn't VOIP illegal in Dubai? |
| 18:32.42 | thiagoc | digium.com and asterisk.org down? |
| 18:33.01 | *** join/#asterisk karelk (~karel@31.10.149.26) |
| 18:33.50 | karelk | I am getting these errors, when calling via WebRTC: |
| 18:33.52 | karelk | res_srtp.c:406 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10 |
| 18:34.15 | karelk | any idea what that means? |
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| 18:46.29 | *** join/#asterisk rolledgold (47c09026@gateway/web/freenode/ip.71.192.144.38) |
| 18:46.40 | rolledgold | Afternoon all. |
| 18:46.50 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:5038:5cf8:1e42:88f4) |
| 18:47.34 | rolledgold | I upgraded from 11 to 13 today from source but rasterisk says both 13 and 11 on the welcome screen |
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| 18:48.41 | rolledgold | http://pastebin.ca/3737434 Any thoughts? |
| 18:57.11 | *** join/#asterisk lordvadr (~lordvadr@unaffiliated/lordvadr) |
| 18:58.18 | Samot | Did you restart Asterisk? |
| 18:59.09 | lordvadr | Am I the only one seeing name resolution failures for essentially all of asterisk.org and digium.com? |
| 18:59.53 | *** join/#asterisk klow (~textual@c-98-247-49-57.hsd1.wa.comcast.net) |
| 19:00.51 | Samot | I'm having an issue as well. |
| 19:01.56 | Samot | The digium NS's must be having an issue. |
| 19:02.47 | lordvadr | I have routes in BGP for all of the IP spaces that their name servers live on but none of them respond to ICMP or DNS. |
| 19:03.20 | lordvadr | All three of them? |
| 19:03.25 | Samot | Not being able to ping an IP is not definitive. |
| 19:03.26 | lordvadr | That's a pretty bad screwup. |
| 19:03.32 | lordvadr | Work for a carrier. I'm aware. |
| 19:05.09 | Samot | So you can't dig through any of the servers via IP? |
| 19:07.19 | lordvadr | Their NS1 and 2 are on a /23 originating AS14793, APID, appears to die somewhere on APID's network, their 3rd one does at apparently cogent's edge in San Fran or San Diego...not sure. |
| 19:07.43 | lordvadr | and no, none of them respond to tcp or UDP dns requests |
| 19:08.01 | Samot | How are you making the requests? |
| 19:08.08 | rolledgold | @Samot Thanks again stop/start fixed it. |
| 19:08.25 | lordvadr | Samot: dig +norecurse digium.com |
| 19:08.43 | Samot | So no, you haven't done it via IP yet. |
| 19:09.00 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 19:09.14 | lordvadr | That'll give you the IP's of their name servers. From there, dig @..., +tcp, telnet 53, ping, traceroute all you want. They don't go anywhere. |
| 19:09.46 | Samot | OK, so they do not respond to lookups directly. |
| 19:10.25 | lordvadr | Nope. Problem is dig will pull pull the IP's from the recursive resolver and then try to verify it from the actual host. Since they're not reachable, you have to +norecurse to get dig to spit out something useful. |
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| 19:13.35 | rolledgold | wiki.asterisk.org responds now |
| 19:13.36 | lordvadr | OOh, they're back. Somebody screwed up though. |
| 19:15.16 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-rwsxijkqycylcshf) |
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| 19:21.52 | lordvadr | kharwell, rmudgett: What happened? |
| 19:22.39 | rmudgett | Internet outage? |
| 19:23.07 | lordvadr | All of Digium's name servers went offline. |
| 19:23.34 | kharwell | I am not sure either what it was |
| 19:27.31 | lordvadr | Fair enough. Just curious. |
| 19:30.03 | rolledgold | quit |
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| 21:20.16 | Akari | Any of you guys messed with calendar integration with gmail? I'm trying to figure out how to set the alarm. |
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| 21:57.09 | drmessano | Google is EVIL |
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| 22:25.57 | igcewieling | Alarmingly so, apparently. |
| 22:34.46 | mmlj4 | hey igcewieling, long time... |
| 22:35.39 | igcewieling | hey, mmlj4 |
| 22:36.04 | mmlj4 | been OK? |
| 22:38.07 | igcewieling | well enough. you? |
| 22:59.19 | drmessano | Nobody ever answered me |
| 22:59.29 | drmessano | About VoIP being illegal in Dubai |
| 23:00.55 | Akari | How could you make VoIP illegal |
| 23:01.00 | drmessano | I got that email about Digium providing AsteriskTraining in Dubai |
| 23:01.14 | drmessano | Im hoping they leave off the parts about SIP |
| 23:01.24 | drmessano | Or else hands will come flying off |
| 23:01.25 | Akari | That would be pointless. |
| 23:01.38 | drmessano | Actually, I just verified it |
| 23:01.49 | drmessano | It's still not legal in Saudi Arabia |
| 23:02.21 | drmessano | Including Voice and Video apps like Facebook Messenger |
| 23:02.44 | igcewieling | Other countries have banned voip/pstn service, but have nothing which makes voip within a company between offices illegal |
| 23:02.45 | Akari | They ban SIP trunking |
| 23:03.07 | igcewieling | I've not looked at SA's requirements' |
| 23:03.34 | drmessano | It's all about Government run telecoms |
| 23:03.50 | drmessano | I think Vietnam is the same |
| 23:05.47 | drmessano | igcewieling: I've noticed an incredible downswing of visitors to IRC asking about Asterisk over a VPN, using GSM gateways, and the other standard wares of the clandestine VoIP operator that we used to see years ago |
| 23:06.05 | drmessano | I dont know if that's just IRC or what.. but it made me curious if it was still an issue.. and I guess it is |
| 23:06.16 | igcewieling | I put most of those on /ignore 8-| |
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| 23:52.18 | drmessano | I guess in Dubai they have disclaimers about putting SIP on the wire and hand choppage |
| 23:52.40 | drmessano | Probably spend an extra hour on proper firewalls :) |