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| 03:47.45 | atan | Can you have more than 1 register => in your [general] section of sip.conf? |
| 04:06.55 | p3nguin | atan: Of course. Since that is the only place register statements are valid, you'd be limited to registering to only one peer. |
| 04:08.33 | atan | p3nguin, okay so this must be where I am going wrong. I am trying to get a bunch of accounts registered each has a different incoming phone number. |
| 04:09.00 | atan | If I use a bunch of register => lines to connect to the accounts which context do the inbound calls get sent to? |
| 04:09.07 | p3nguin | None. |
| 04:09.15 | p3nguin | Register statements do not control that. |
| 04:09.54 | p3nguin | Well, I said none, but that's not right. The calls would go to the context set in the general section. |
| 04:10.20 | p3nguin | To control where your calls from a specific host go, you need to define a peer. |
| 04:10.55 | p3nguin | If the peer entry matches the call, the context set in the peer entry will be used. |
| 04:11.17 | atan | This must be where I'm getting confused. I have it registered using these lines but I can't seem to force the inbound calls into the correct contexts |
| 04:11.51 | atan | Could I just define context= in [general] and be done with it? =) |
| 04:12.03 | p3nguin | You either didn't define a peer, or your peer isn't matching the call. |
| 04:12.18 | p3nguin | You could do that, but that is not the right way to handle it. |
| 04:13.17 | atan | By way of example let's say I have register => providerusername:secret@host.com:5060 several times in the [general] section, each with different providerusernames |
| 04:13.23 | *** join/#asterisk KNERD (~KNERD@adsl-99-153-147-234.dsl.hrlntx.sbcglobal.net) |
| 04:13.42 | p3nguin | If you want help determining why calls are not going to the context you want them to go to, pastebin your sip.conf and a debug of a call intended to match the peer you defined. |
| 04:13.54 | atan | I also have [providerusername] type=peer context=providerusername-in, but that context isn't taking it |
| 04:14.14 | p3nguin | Did you set the host in that peer entry? |
| 04:14.29 | p3nguin | host=their-host-or-IP-address |
| 04:15.03 | atan | Let me craft a little pastebin of what I've got going on :-) |
| 04:15.14 | p3nguin | type=peer causes the entry to match on host. Without host= it cannot match. |
| 04:15.26 | p3nguin | I'm not interested in craftiness. Just paste what you have. |
| 04:15.35 | p3nguin | Hide only your passwords. |
| 04:16.48 | p3nguin | Two pastebins: sip.conf and a debug of a call which you expect to match the defined peer. |
| 04:19.19 | atan | http://pastebin.com/sedTvzmT is my sip & extension I am trying to use to catch the inbound call |
| 04:19.34 | atan | Now I must figure out how to get debug on for this call.. core set sip debug or something I presume. |
| 04:26.05 | p3nguin | Well, hang on... |
| 04:26.39 | p3nguin | With VoIP.ms, you'll never be able to match individual usernames on incoming calls. |
| 04:27.05 | p3nguin | They don't send the uid as part of the call. |
| 04:27.14 | atan | I'm not sure I follow but I am trying |
| 04:27.32 | atan | Are you suggesting I force all inbound voip.ms calls in via one registration not seperate accounts? |
| 04:28.00 | p3nguin | Maybe. What is your purpose for having individual accounts? |
| 04:28.16 | atan | For inbound calls it is pretty pointless. For outbound it was to track use. |
| 04:28.25 | atan | However inbounds I can base on the numbers alone so that would be fine |
| 04:28.35 | atan | Outbounds I could send via another SIP registration I think? |
| 04:28.47 | p3nguin | Okay, you can leave the separate accounts for CDR purposes. |
| 04:29.10 | p3nguin | You don't send calls via registration. Registration tells the other system how to reach you. |
| 04:30.26 | p3nguin | For incoming calls, because they do not send your uid as part of the call, you cannot discern a call via one account from a call via the other account. |
| 04:31.07 | atan | Okay so I'll use voip.ms to put all calls into the main account username and register with that to take the calls, but then how do I pick what context is matching for them? |
| 04:31.19 | p3nguin | So all incoming calls will match one peer entry. I can't remember if it is the first entry listed or the last, but all calls regardless of account will match only one of the peer entries. |
| 04:31.56 | atan | See when I put in like a list of 10 |
| 04:32.03 | atan | The first one would always take the call and I couldn't figure why |
| 04:32.30 | p3nguin | Also, with voip.ms, you will have to keep the registrations for any account you wish to send outbound calls through, because they require registration first. |
| 04:32.39 | atan | the second, third, whatnot were setup in a very similar way but they just didn't work despite having the same princial behind how I had them entered. No biggie though :-) |
| 04:32.56 | atan | Do I use type=peer or type=friend to register with them so I can make calls? |
| 04:33.15 | p3nguin | type=peer and type=friend have NOTHING to do with registering. |
| 04:33.41 | atan | In their example they show it as type=peer though, http://wiki.voip.ms/article/PBXs#Asterisk_.28SIP.29 |
| 04:33.51 | atan | Perhaps I'm not understanding the purpose the tutorial they posted there |
| 04:35.21 | p3nguin | Forget them. Let me give you a step-by-step. |
| 04:36.10 | p3nguin | You want to send calls via multiple accounts for CDR purposes. Leave your register statements in place -- they require you to register before they will accept calls from you. |
| 04:36.54 | p3nguin | Define a peer for each account you wish to make calls via. |
| 04:37.32 | p3nguin | Set the context for those accounts to a common inbound context, such as voipms-in. |
| 04:38.15 | p3nguin | You will use those peer entries for dialing outbound calls only. Only one will match inbound calls. |
| 04:39.08 | atan | So that takes care of the outbound pretty well |
| 04:39.15 | p3nguin | In your DID management, route the DIDs to any account(s) you want. It won't matter, because all the calls will go into one common voipms-in context. |
| 04:40.09 | p3nguin | The only real purpose for multiple accounts on you single asterisk is for CDR purposes on their portal. |
| 04:40.21 | p3nguin | s/you/your/ |
| 04:40.44 | atan | Okay just fixing up that part on their end here now for billing so I can see which line goes nuts hehe |
| 04:41.17 | p3nguin | In your voipms-in context, define an extension for every DID number you have routed to your system. |
| 04:41.55 | atan | Hey just quick question here, does Asterisk have some form of block comment? |
| 04:42.06 | atan | So I can comment out the crap I've messed up for now without deleting it |
| 04:42.41 | p3nguin | No. You have to comment out each line. |
| 04:43.41 | atan | Okay just so I don't keep you all night right now I have [voipms-inbound] inside my exntesions.conf. I have my two numbers defined and set to ring a SIP device. |
| 04:43.54 | atan | That device is registered and rings fine, now I've just got to get the calls over to it |
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| 04:45.03 | p3nguin | I don't understand your last statement. |
| 04:45.43 | atan | I have [voipms-inbound] defined in extensions.conf, and a pattern to match my voipms numbers |
| 04:45.47 | p3nguin | I got that part. |
| 04:45.57 | atan | I have both of them set to ring the same phone for now just so I know if it works :-) |
| 04:46.08 | p3nguin | I got that part. |
| 04:46.12 | atan | Actually, playback might be better right now. I suppose Playback(tt-monkeys) is default? |
| 04:46.25 | p3nguin | I don't know what default means in that sentence. |
| 04:46.55 | atan | sorry I mean like it's there |
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| 04:47.03 | p3nguin | If your extension runs Playback(tt-monkeys), that's what you should expect to hear. |
| 04:47.06 | atan | I can use Playback(tt-monkeys) and it'll work for me? or |
| 04:47.17 | atan | ok. tt-monkeys is included in a default install? |
| 04:48.16 | p3nguin | If you included it, sure. |
| 04:49.05 | atan | Was it in the optional sound pack or selected by default is what I mean? |
| 04:49.13 | KNERD | Is there a way to detect is imcoming call is from a public phone? |
| 04:50.37 | p3nguin | In the amount of time it is taking for you to understand that you only get what you ask for in your installation, you could have either A) looked for the file, or B) tried the Playback to see if the file played. |
| 04:51.00 | atan | Looked :-) it's there |
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| 04:51.34 | atan | Okay I think we got it up and working here now thanks to your help :D |
| 04:51.48 | atan | ty kindly p3nguin |
| 04:52.29 | p3nguin | I think incoming calls will match the first peer listed for the host, so you can then add your other peers for outbound calls after that primary one. |
| 04:53.22 | p3nguin | You can also use type=user for the entry for calls coming inbound from the provider. type=user does not permit calls going in the other direction. |
| 04:53.27 | atan | Mind if I toss in another totally stupid question? What is the idiot-friendly difference between peer & friend? |
| 04:53.54 | p3nguin | friend is peer+user |
| 04:54.28 | atan | Suffice to save friend is a trusted device, like a sip phone on my end? |
| 04:54.33 | atan | s/save/say/ |
| 04:54.45 | p3nguin | peer matches on IP/host and allows calls in both directions. user matches on username and allows calls from the other device inbound to asterisk. friend is a cross-bread of those two. |
| 04:55.20 | p3nguin | I use peer for all my phones. |
| 04:55.32 | p3nguin | I use friend very rarely. |
| 04:56.02 | p3nguin | When you need to match on a username rather than IP and port, you have to use friend or user. |
| 04:56.43 | p3nguin | An instance where this is important is a phone with multiple users from the same IP address and port on the phone. |
| 04:56.48 | atan | Now for the sub accounts I have setup within my sip.conf, I don't even need to put a context on those if I plan to just use them for outbound? |
| 04:57.31 | p3nguin | If for any reason your calls ever would match one of those peers and you do not have a context set, the call will fall into the context set in the general section. |
| 04:58.33 | p3nguin | If you don't trust that asterisk will always match that first entry for the host, set the same voip-in context on them all. |
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| 05:00.46 | p3nguin | I need to start looking at sip debug to determine if there is any way to match on username instead of host only. That is a very annoying characteristic, especially for configurations like yours and mine where we would like to manage them separately on asterisk. |
| 05:03.21 | atan | Now there is only one other pesky little thing I'd love to solve. From time to time when I place an outbound the other party will answer but there is a delay before audio is connected. |
| 05:03.40 | atan | The other person may have said "Hello?" but on my end it appears as though they are just not speaking. |
| 05:04.08 | atan | After a second or two it all connects and audio is normal. It's just the first second or so of the call that's finicky like this. Is this something I've setup wrong somewhere? |
| 05:06.37 | p3nguin | I can't really think of anything specific to configuration except a possible early media setting in the phone. |
| 05:06.38 | ChannelZ | Nat? |
| 05:07.05 | atan | The SIP phone is behind a router, so yes |
| 05:07.19 | atan | No DMZ or anything going on there |
| 05:10.24 | p3nguin | All I see is that calls are sent To: <sip:my-phone-number@my-IP-address:5060> ... |
| 05:11.05 | p3nguin | Would that be enough to match a call? That's where the username goes, isn't it? |
| 05:11.25 | p3nguin | The INVITE is also to that same URI. |
| 05:12.46 | p3nguin | Perhaps I can set type=friend and fromuser=my-phone-number and get a match. |
| 05:13.15 | p3nguin | Wait, no. |
| 05:13.27 | p3nguin | fromuser will match the From: not the To: |
| 05:14.02 | p3nguin | And the From: is the callerid number of the person calling, which might as well be totally random for this purpose. |
| 05:15.11 | p3nguin | This must be why I've left it the way it is for so long -- there is no suitable workaround. |
| 05:17.13 | atan | Would I be wrong to assume that Hangup() isn't needed after basic functions like Dial(), VoiceMailMain() and such as it's already included in that when the user disconnects? And when the context runs out of matching patterns it hangs up anyway? |
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| 05:21.17 | p3nguin | I think that depends on your fallthrough setting. I always define a Hangup() when I want to ensure hangup happens. |
| 05:21.58 | p3nguin | # grep Hangup /etc/asterisk/extensions.conf |wc -l |
| 05:22.00 | p3nguin | 161 |
| 05:22.22 | p3nguin | That doesn't count the included files. |
| 05:22.57 | *** part/#asterisk gonewage (~gonewage@c-68-54-124-223.hsd1.il.comcast.net) |
| 05:23.05 | p3nguin | # asterisk -rx 'dialplan show'|grep Hangup|wc -l |
| 05:23.06 | p3nguin | 211 |
| 05:24.48 | p3nguin | Also, it's a matter of extension running out of priority rather than a context running out of patterns. |
| 05:29.12 | din3sh | p3nguin: any idea why i could get on way audio on transfered call via a BRI gateway? |
| 05:29.22 | din3sh | one way audio* |
| 05:29.43 | p3nguin | Probably related to a reinvite. |
| 05:30.15 | p3nguin | One-way audio is usually a result of a reinvite where asterisk would have been better off retaining the media stream. |
| 05:31.24 | p3nguin | In other words, set directmedia=no everywhere and see what happens... or start pastebinning all relevant configs and debug info. |
| 05:32.08 | din3sh | sip debug :http://paste2.org/p/1976627 |
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| 05:35.12 | *** join/#asterisk Ruslan666 (~Ruslan666@46.32.171.162) |
| 05:36.38 | Ruslan666 | Hello. Need help about Wildcard TE122, Card. Can anyone help? |
| 05:37.14 | din3sh | Ruslan666: what help? |
| 05:37.14 | p3nguin | Hmm, weird. I don't really see anything like that in the debug. Only two devices, both on the same subnet. |
| 05:37.22 | din3sh | yes |
| 05:37.40 | p3nguin | No NAT, didn't notice any reinvites anyway. |
| 05:37.54 | p3nguin | It only happens on transfer? |
| 05:38.02 | p3nguin | direct calls work correctly? |
| 05:38.08 | din3sh | yes |
| 05:38.25 | din3sh | PSTN through BRI gateway to SIP extensions= ok |
| 05:38.28 | Ruslan666 | din3sh i have wcte12xp card. it works but i have noise problem :( dont know how to reslove it. |
| 05:39.13 | p3nguin | I'm only seeing asterisk and one SIP phone. |
| 05:39.18 | din3sh | once the call is transfered to another sip number on same box, the caller (PSTN) cannot hear anythg, the transferd SIP number can hear the PSTN num though |
| 05:39.27 | p3nguin | Where is the other device involved in the transfer? |
| 05:40.11 | p3nguin | I see .5 and .151 devices only. |
| 05:40.36 | din3sh | 4999493 calls 2729 |
| 05:40.43 | din3sh | 2729 transfers to 7662 |
| 05:41.00 | p3nguin | Okay, but where is the other device in this debug? |
| 05:41.07 | din3sh | 7662 is on another box via sip trunking |
| 05:41.16 | p3nguin | There is no such thing as sip trunking. |
| 05:41.43 | din3sh | whats is the proper term for that? |
| 05:41.47 | p3nguin | sip |
| 05:42.06 | din3sh | anyway even on the same box, i.e other 29xx numbers |
| 05:42.08 | p3nguin | I see only two devices involved: asterisk and one phone (or the gateway). |
| 05:42.15 | din3sh | the problem is same |
| 05:42.31 | p3nguin | For a transfer, I'm expecting to see at least one more device. |
| 05:42.31 | din3sh | 4999493 is from the gateway yes |
| 05:43.30 | p3nguin | Unless you are transferring one call on the gateway to another call also on the gateway. Then I would see the gateway's IP address twice as much. |
| 05:44.02 | din3sh | no one call is from gateway the 2 others internal extensions |
| 05:44.45 | Ruslan666 | Can anybody help me :( ? |
| 05:45.15 | p3nguin | There's only one SIP device involved in that call with asterisk. That is not a very good debug of a failed transfer. |
| 05:46.30 | din3sh | ok am trying to have another log |
| 05:51.31 | Ruslan666 | can somebody help me aboud dahdi boards? |
| 05:51.50 | Ruslan666 | can anobody help me about dahdi boards? |
| 05:52.17 | p3nguin | I'm near falling asleep, so you may have to continue with someone else or I'll check the pastebin when I get back to the office in the morning (about 19:00 your time). |
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| 06:12.44 | Ruslan666 | i'm ready to may that |
| 06:12.59 | Ruslan666 | i'm ready to pay for that help |
| 06:17.05 | din3sh | p3nguin you still there? |
| 06:17.07 | din3sh | http://paste2.org/p/1977565 |
| 06:17.37 | din3sh | Ruslan666: does your card have echo cancel module? |
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| 06:33.05 | schmidts | good morning |
| 06:33.13 | din3sh | mrning |
| 06:37.15 | kleszcz | morning |
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| 06:44.20 | s[x] | sup peeps |
| 06:45.25 | bn-7bc | to any developer;: thanks for a great pace of sw and thanks to Digium for their support of the project |
| 06:47.33 | bn-7bc | has anyone tried video calling wit the grounwire app for iPhone and asterisk 10, dose it work well? |
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| 07:17.50 | Ruslan666 | din3sh hello. no do not have. |
| 07:24.54 | din3sh | Ruslan666: have u turned on echocancel in your dahdi settings? |
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| 08:52.06 | kaidranzer | how does asterisk send an audio file? what is the codec that it uses? |
| 08:53.03 | din3sh | send audio file where? |
| 08:53.36 | kaidranzer | for example when "hello-world" audio file is sent to a peer, what codec does it use? |
| 08:54.06 | Chainsaw | kaidranzer: Depends on what the peer supports. |
| 08:54.24 | Chainsaw | kaidranzer: Asterisk may have to transcode the audio file in some cases. |
| 08:55.04 | kaidranzer | assuming it sends using u-law g.711 to send, then how is the gsm file converted to the actual bytes that are sent in the packets? |
| 08:56.02 | din3sh | technically i dont think asterisk "sends" the file |
| 08:57.21 | kaidranzer | yes it sends bytes but how are those bytes actually created from the existing gsm file in the /var/lib/asterisk/sounds/en folder? |
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| 08:58.35 | din3sh | Chainsaw: any idea why a BRI ISDN gateway not able to bridge 2 calls on and attended-xfer? |
| 09:00.50 | din3sh | kaidranzer:no idea, i jst know that applications like Playback() , background() actually only render the requested file in the format they are |
| 09:00.55 | din3sh | i might be wrong |
| 09:02.40 | kaidranzer | i tried searching the bytes of the hello-world.gsm in the trace of my hello world call but couldn't find them so the format is something other than gsm |
| 09:02.42 | Chainsaw | din3sh: Hard to say without seeing a SIP trace of the transfer failing. |
| 09:03.27 | din3sh | http://paste2.org/p/1977565 |
| 09:03.42 | din3sh | sip debug of the transfer |
| 09:04.32 | Chainsaw | din3sh: Reliably Transmitting (no NAT) to 192.168.2.5 <- That looks suspicious. RFC1918 (LAN internal, unrouteable) addresses with NAT=no. |
| 09:04.50 | din3sh | am on same subnet |
| 09:04.57 | din3sh | no firewall no nat |
| 09:05.36 | din3sh | direct calls work fine, when i try to bridge 2 calls, the one coming through the gateway cannot hear anythg |
| 09:10.44 | din3sh | going nuts with this issue |
| 09:10.57 | *** join/#asterisk s[x] (~sx]@60-241-151-10.tpgi.com.au) |
| 09:14.59 | asr33 | Hello would BLACKLIST func work on alphabetic portion of the callerid? |
| 09:17.06 | asr33 | Example """Unavailable"" <Unavailable>" |
| 09:21.29 | *** join/#asterisk tapout (~tapout@unaffiliated/tapout) |
| 09:23.29 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 09:39.48 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 09:41.06 | qakhan | i want to setup an ext which will be used by supervisor to listen agents calls |
| 09:41.09 | qakhan | plz help |
| 09:42.20 | kaldemar | qakhan: app ChanSpy will help you with that, "core show application ChanSpy" |
| 09:42.54 | qakhan | yes i am using it |
| 09:43.10 | qakhan | exten => 100,1,ChanSpy(Agent,qw) |
| 09:44.11 | qakhan | but when i dial it. it goes to first agent. i want to enter the agent id to listen the call |
| 09:54.22 | *** join/#asterisk ks3 (ks3@tatooine.digitallotus.com) |
| 09:55.14 | *** join/#asterisk TriJetScud (~TriJetScu@testarossa.smb.curriegrad2004.ca) |
| 09:56.33 | qakhan | kaldemar u there? |
| 10:03.54 | kaldemar | qakhan: are you sure you want to use option w? |
| 10:04.56 | kaldemar | qakhan: as you can see from the application documentation, the first argument is a channel prefix. if you want the caller to be able to enter somethign, do it with app Read. |
| 10:07.20 | qakhan | u mean something like this |
| 10:07.22 | qakhan | 100,1,Read(digits) |
| 10:07.23 | qakhan | 100,n,ChanSpy(${digits},qw) |
| 10:07.24 | qakhan | ? |
| 10:08.01 | kaldemar | something like that, but ${digits} is surely not a valid channel prefix by itself. |
| 10:08.55 | qakhan | so what i need to do |
| 10:14.47 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 10:16.14 | kaldemar | qakhan: think about how you can put the digits as a part of the agent channel and have a valid channel prefix. for example ChanSpy(Agent/${digits},q) or something similar. depends on your agent names, which i do not know. |
| 10:16.39 | kaldemar | qakhan: and still why do you have the w option if the plan is to listen on a call? |
| 10:23.19 | *** join/#asterisk tapout (~tapout@unaffiliated/tapout) |
| 10:39.21 | *** join/#asterisk s[x] (~sx]@60-241-151-10.tpgi.com.au) |
| 10:43.59 | qakhan | kaldemar the plan is supervisor call listen agent call and if required then he can speak in that call |
| 10:45.19 | *** join/#asterisk wonderworld (~ww@dsdf-4db5f99a.pool.mediaWays.net) |
| 10:48.53 | din3sh | how to allow or forbid transfer for a sip peer in asterisk 1.8? |
| 10:49.53 | kaldemar | din3sh: allowtransfer in sip.conf |
| 10:59.47 | *** join/#asterisk bmg505 (~leon@196-209-123-205.dynamic.isadsl.co.za) |
| 11:04.11 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 11:19.41 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:23.45 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 11:23.54 | qakhan | kaldemar u there? |
| 11:25.51 | kaldemar | qakhan: don't ask if i'm here. if you have a question, just ask the question on channel. someone else might answer too. |
| 11:25.55 | qakhan | i want to save call answer date/time, call duration date/time and call end date/time in cdr table |
| 11:27.27 | kaldemar | have you disabled those somehow? |
| 11:27.49 | qakhan | no |
| 11:28.11 | kaldemar | they are logged by default by CDR modules. |
| 11:28.19 | qakhan | is it possible that i can save some other recorde in CDR? |
| 11:29.53 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 11:32.24 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 11:32.25 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 11:33.34 | din3sh | he wants to log the cdr in mysql backend |
| 11:33.52 | din3sh | ? |
| 11:36.50 | cusco | yes, in the dialplan you Set(CDR(newfield)=lalala); then use newfield in your cdr table |
| 11:42.02 | din3sh | still with my no audio on transfer problem |
| 11:42.04 | din3sh | :/ |
| 11:44.23 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 11:44.26 | cusco | nat? directmedia? how are you transfering? |
| 11:44.29 | qakhan | cusco there are 5 new fields i need to be saved in CDR |
| 11:45.04 | qakhan | is there any other way, any .conf file which contain these new fields and save in cdr after the call end |
| 11:45.22 | cusco | qakhan: what values will thos fields contain? |
| 11:45.25 | din3sh | i have a BRI gateway registered with sip |
| 11:46.13 | din3sh | caller from through the gateway when transferd by a sip extension to another, doesnt hear anythg, sip extension can hear the caller though |
| 11:46.28 | qakhan | call answer date/time, call duration date/time and call end date/time in cdr table |
| 11:48.20 | kaldemar | qakhan: those are logged by default. |
| 11:52.25 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 11:52.31 | din3sh | qakhan: thats found in /var/log/asterisk/cdr-csv |
| 11:52.38 | din3sh | in csv format |
| 11:53.09 | qakhan | kaldemar these are not in cdr |
| 11:53.16 | qakhan | din3sh let me check |
| 11:55.03 | kaldemar | qakhan: i don't believe you until you show it. |
| 11:56.07 | qakhan | din3sh /var/log/asterisk/cdr-csv contain only data |
| 11:56.13 | qakhan | not table field |
| 11:57.49 | qakhan | kaldemar look here |
| 11:57.50 | qakhan | http://pastebin.com/tkQjsacA |
| 11:58.56 | din3sh | as i said, thats a mysql-backend cdr |
| 12:00.51 | din3sh | create your table |
| 12:01.00 | din3sh | and connect using cdr_mysql.conf |
| 12:01.28 | *** join/#asterisk timahvo1 (~rogue@197.177.82.49) |
| 12:01.38 | din3sh | if asterisk 1.4.x , you have to compile asterisk-addons for cdr database support |
| 12:02.02 | qakhan | i have setup cdr in asterisk 1.4.38 |
| 12:02.18 | din3sh | so whats the issue? |
| 12:02.19 | qakhan | but i just want to add 3 firlds |
| 12:04.09 | qakhan | din3sh i m not getting the point where to start |
| 12:05.31 | kaldemar | qakhan: what have you set up? |
| 12:06.12 | kaldemar | qakhan: so what you're really asking is for help with your database usage? |
| 12:06.42 | qakhan | noooo |
| 12:07.09 | qakhan | listen if we need a ext we add it in sip.conf |
| 12:07.36 | qakhan | which file is containing the CDR conf? |
| 12:10.09 | kaldemar | which CDR module are you using? |
| 12:11.45 | din3sh | grrrrr |
| 12:12.05 | din3sh | would take a feww hours to understand what qakhan wants |
| 12:12.39 | kaldemar | din3sh: don't be so optimistic. |
| 12:12.55 | din3sh | haha |
| 12:14.27 | qakhan | i installed asterisk addons |
| 12:15.11 | kaldemar | qakhan: that is not what i asked. which CDR module are you using? |
| 12:15.44 | qakhan | who i can check it? |
| 12:16.57 | kaldemar | if you think you need to add fields to CDR, one would assume you know where you are checking the records. |
| 12:18.02 | qakhan | kaldemar sorry i didnt get you |
| 12:20.50 | kaldemar | qakhan: why do you think you don't have that information in your CDR fields? how did you determine that and from where? |
| 12:20.58 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:21.03 | Foxi352 | Hmm … I have an issue with incoming calls on SIP trunk if using realtime DB. No answer here (some month ago) and no reply to my post on forum.asterisk.org (http://forums.asterisk.org/viewtopic.php?f=1&t=82136). Am i really the only one using SIP trunks with asterisk realtime db ? |
| 12:21.35 | qakhan | kaldemar are you asking about CDR fields? |
| 12:21.58 | Foxi352 | Has anyone in here Asterisk 1.8 with a SIP trunk and SIP table in realtime db ? |
| 12:22.42 | *** join/#asterisk gonewage (~gonewage@c-68-54-124-223.hsd1.il.comcast.net) |
| 12:22.56 | kaldemar | qakhan: no, i'm asking about the midsummer night northern lights and flowers in your backyard. of course i'm asking about CDR fields, i said "CDR fields". |
| 12:23.54 | qakhan | sir i sent you link http://pastebin.com/tkQjsacA |
| 12:24.18 | [TK]D-Fender | Foxi352, Dump your DB, and show us a call with SIP debug enabled, and a a peer dump priori and post. |
| 12:24.19 | [TK]D-Fender | ~pb |
| 12:24.19 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 12:24.20 | [TK]D-Fender | ^^^ |
| 12:24.57 | Foxi352 | [TK]D-Fender: Ok, give me some mins, will ping you back |
| 12:25.20 | kaldemar | qakhan: which is not of much use by itself. |
| 12:26.09 | qakhan | kaldemar ? |
| 12:26.22 | kaldemar | qakhan: give more information. |
| 12:26.40 | kaldemar | qakhan: you have been asked for it. |
| 12:27.25 | *** part/#asterisk gonewage (~gonewage@c-68-54-124-223.hsd1.il.comcast.net) |
| 12:27.32 | qakhan | there fields i have in cdr. which more info you required? |
| 12:28.26 | kaldemar | qakhan: i won't waste my time repeating stuff over and over again. |
| 12:28.52 | qakhan | kaldemar seriously tell me what you want |
| 12:29.20 | qakhan | i sent all info. which info you need more ? |
| 12:30.12 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 12:32.07 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 12:36.48 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 12:41.36 | Foxi352 | [TK]D-Fender: Want a real mysql sql dump or a select * copy&paste ? |
| 12:41.41 | *** join/#asterisk chuckf_ (~chuckf@ubuntu/member/chuckf) |
| 12:42.04 | [TK]D-Fender | Either/both |
| 12:43.26 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 12:44.44 | Foxi352 | k |
| 12:45.09 | fprior | Hi all, here again with the case of the spa400 and * 1.8 . Info on http://pastebin.com/fVC9JtV2 |
| 12:46.31 | din3sh | kaldemar: qakhan still at it, and its been more than an hour |
| 12:47.15 | din3sh | instead of dumping sql, dump the server itself |
| 12:51.28 | din3sh | [TK]D-Fender: can you take a look at this and tell me what might be wrong? http://paste2.org/p/1977565 |
| 12:54.13 | Foxi352 | [TK]D-Fender: Here is SIP peers before / after. My COMMENTS ARE IN CAPS to quickly identify. This is with high verbose level to see dialplan details. SIP debug of misleaded call follows in some minutes. Fo readability it did not turn it on @ the same time with high verbose … : http://pastebin.com/DscDe07M |
| 12:54.35 | [TK]D-Fender | din3sh, [Apr 13 10:01:34] VERBOSE[8314] chan_sip.c: Peer audio RTP is at port 0.0.0.0:10008 |
| 12:54.40 | [TK]D-Fender | din3sh, That UA is retarded. |
| 12:55.03 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:55.14 | [TK]D-Fender | din3sh, It seems to have no clue about its IP. I'm wondering if it's attempting to parse some sort of host name out for this. |
| 12:55.38 | [TK]D-Fender | din3sh, You could probably get away with setting "nat=yes" to solve this for that peer. |
| 12:55.50 | din3sh | oh |
| 12:56.02 | Foxi352 | [TK]D-Fender: In Résumé: After core restart SIP peers table is obviously empty. Incoming calls are redirected to context set in general section of sip.conf instead of context set in trunk definition in sip_device table. After first outgoing call SIP peer is in SIP peer table because it's cached, and then incoming calls are redirected to the right context. So basically do an outgoing call on every line after a core restart, and incoming works |
| 12:56.10 | [TK]D-Fender | din3sh, SIP/2729 <- this peer |
| 12:56.34 | din3sh | actually 2729 triggers the transfer |
| 12:56.42 | [TK]D-Fender | Foxi352, So basically it isn't doing the first lookup on incoming? |
| 12:57.05 | [TK]D-Fender | din3sh, doesn't look sane. |
| 12:57.09 | Foxi352 | No, as long as the SIP peer is not in the peer table (cache) ... |
| 12:57.11 | [TK]D-Fender | din3sh, Try what I've suggested |
| 12:57.20 | *** join/#asterisk jkroon (~jkroon@dsl-244-29-126.telkomadsl.co.za) |
| 12:58.29 | [TK]D-Fender | Foxi352, What ver are you on? |
| 12:59.13 | Foxi352 | Asterisk 1.8.11.0-1digium1~lucid |
| 12:59.27 | [TK]D-Fender | Foxi352, Hrm.... |
| 12:59.32 | Foxi352 | I work for an OSS project and we are atm bound to Lucid... |
| 12:59.41 | Foxi352 | This is latest lucid package available on digium servers |
| 13:00.00 | [TK]D-Fender | Foxi352, No, your release is quite current.... not sure on what to do at this point if it's a first-load issue... |
| 13:00.04 | din3sh | [TK]D-Fender: no luck, problem still same |
| 13:00.12 | din3sh | here is whats hapening |
| 13:00.26 | [TK]D-Fender | din3sh, I am not seeing that as a "transfer". What exactly is it doing? |
| 13:00.50 | Foxi352 | [TK]D-Fender: I was the one porting our OSS project from .conf files to realtime DB. Everything works perfectly except this. I googled for nearly 3 month now, impossible to find somthing ... |
| 13:00.52 | [TK]D-Fender | din3sh, And if it is attempting to pass on a 3rd party IP then we don't have it and I guess we're screwed |
| 13:01.00 | Foxi352 | I fear that i will have to revert at least trunks back to .conf files ... |
| 13:01.12 | [TK]D-Fender | Foxi352, I'd post this up on the tracker to see if its a bug... |
| 13:01.42 | [TK]D-Fender | Foxi352, What happens f you don't cache? |
| 13:01.56 | din3sh | 4999493(PSTN) calls 2729(SIP extension) via a BRI gateway, 2729 then hits attended transfer to 2713 (SIP extension), MOH starts off |
| 13:01.59 | [TK]D-Fender | Foxi352, Aside from showing up in "sip show peers", do you see it hitting the DB when the call comes in regardless? |
| 13:02.32 | Foxi352 | [TK]D-Fender: Did not yet debug the sql, good idea … Will do that …. |
| 13:02.37 | din3sh | then 2729 transfers the call, 7662 can hear 4999493, but 4999493 cannot hear anythg |
| 13:02.56 | Foxi352 | [TK]D-Fender: If i remember correctly it does never pick up if i don't cache … I think that was the major reason for me using cache |
| 13:03.13 | Foxi352 | I mean it picks up, but sends to the wrong context... |
| 13:03.24 | din3sh | [TK]D-Fender:4999493(PSTN) calls 2729(SIP extension) via a BRI gateway, 2729 then hits attended transfer to 2713 (SIP extension), MOH starts off,then 2729 transfers the call, 7662 can hear 4999493, but 4999493 cannot hear anythg |
| 13:04.28 | din3sh | same setup was working fine on asterisk 1.4, its not working in 1.8.11 |
| 13:05.27 | *** join/#asterisk chazzam (~chazz@50-81-150-34.client.mchsi.com) |
| 13:06.51 | *** join/#asterisk darkskiez (~mhb@fsf/member/darkskiez) |
| 13:07.08 | [TK]D-Fender | From: 2729 <sip:2729@192.168.2.0 |
| 13:07.16 | [TK]D-Fender | No sure what that IP is supposed to be later on... |
| 13:07.24 | din3sh | [TK]D-Fender: I have tried to isolate problem by calling a pstn num (from 2729) and transfering to another pstn num via same gateway, when the call is bridged/transfered by 2729, the 2 connected pstn numbers cannot hear anythg even being connected |
| 13:07.58 | din3sh | the problem is the 192.168.2.5 which is the gateway |
| 13:08.04 | [TK]D-Fender | din3sh, And yeah there's a lot going on and I'm not sure I can tell exactly what the isue is here yet... I'd start by disabling reinvites everywhere. Don't forget the new directive is called "directmedia=on" instead of "canreinvite=no" |
| 13:08.15 | [TK]D-Fender | dineWhat "gateway"? |
| 13:08.41 | [TK]D-Fender | Foxi352, If it needs cache, then that sounds like a bug... |
| 13:08.45 | [TK]D-Fender | Foxi352, I'd report it... |
| 13:09.17 | Foxi352 | [TK]D-Fender: Ok, will do … thanks |
| 13:10.05 | din3sh | i have tried that also |
| 13:10.14 | din3sh | [TK]D-Fender:4999493(PSTN) calls 2729(SIP extension) via a BRI gateway, 2729 then hits attended transfer to 2713 (SIP extension), MOH starts off,then 2729 transfers the call, 7662 can hear 4999493, but 4999493 cannot hear anythg |
| 13:10.37 | *** join/#asterisk Bullmoose (~Bullmoose@12.50.16.66) |
| 13:10.55 | din3sh | direct calls from/to pstn numbers work fine, but when a transfer occurs, the pstn number cannot hear anythg |
| 13:12.02 | [TK]D-Fender | din3sh, Did you test the extension you're forwarding to direct with that gateway? |
| 13:12.20 | [TK]D-Fender | din3sh, to validate the taget by itself (without a transfer involved) |
| 13:14.46 | *** join/#asterisk MrTelephone (~MrTelepho@h697179-171.picriverisp.net) |
| 13:15.27 | din3sh | http://paste2.org/p/1977892 my sip.conf |
| 13:15.29 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 13:15.48 | MrTelephone | When are you going to add authentication support for multiple peer entries. I keep getting hash mismatches? Is there a way around this issue? |
| 13:15.49 | din3sh | yes direct calls work without problem |
| 13:15.52 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-gstvtgylfwwcockz) |
| 13:15.52 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:17.21 | din3sh | I have tried to isolate problem by calling a pstn num (from 2729) and transfering to another pstn num via same gateway, when the call is bridged/transfered by 2729, the 2 connected pstn numbers cannot hear anythg even being connected |
| 13:17.35 | [TK]D-Fender | allowtransfer=yes <_? |
| 13:17.40 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 13:17.47 | din3sh | should be no? |
| 13:17.57 | [TK]D-Fender | Don't know this one at all... I'd say just remove it |
| 13:19.36 | din3sh | doesnt help, i commented it and just tried again...no luck |
| 13:19.37 | din3sh | :S |
| 13:20.47 | din3sh | no attended transfer from pstn on the prod box since yesterday |
| 13:20.56 | *** join/#asterisk slingr (santas@will.one.day.hack-the-pla.net) |
| 13:21.15 | MrTelephone | Most phone devices use the same port for sip messaging so it conflicts with other peer entries. I was waiting patiently to have this fixed since version 1.2 and I'm running 1.8. Maybe I'm doing something wrong but shouldn't the logic be checking all hashes from peers with the same connection info before it denies the authentication? |
| 13:21.35 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 13:21.45 | jaytee | has anyone here ever setup Asterisk behind a Cisco ASA 5505 (asa v8.2)? |
| 13:21.48 | [TK]D-Fender | peer = auth by IP. use "type=friend, or run a proxy for a multi-phone site |
| 13:22.24 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 13:22.30 | MrTelephone | if you use friend for everything when you reinvite to antoher asteirsk box it wants to authenticate the peer not the asterisk box |
| 13:22.35 | MrTelephone | fix the logic |
| 13:24.55 | *** join/#asterisk serafie (~erin@nat/digium/x-nnzvzbkdvdorltpm) |
| 13:25.21 | [TK]D-Fender | Good reason to not allow reinvites. |
| 13:25.30 | MrTelephone | for clustering |
| 13:25.33 | [TK]D-Fender | Which is what I advise rather consistently |
| 13:25.47 | MrTelephone | it's not even reinvite |
| 13:25.56 | [TK]D-Fender | Lots of things are good for clustering, especially "fucks". |
| 13:26.43 | MrTelephone | there is all types of clustering tools but really you shouldn't need any with the complex dialplan options |
| 13:27.53 | MrTelephone | you can just use dial() in sequence to hit all your other machines. Doesn't that seem easy? |
| 13:28.44 | MrTelephone | but then I run into this quirky authentication problem that occurs only when there are more than 2 endpoints configured on one ata |
| 13:31.55 | din3sh | throws his phone out of the window.. >:/ |
| 13:32.32 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 13:32.48 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 13:35.05 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:35.19 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 13:40.14 | [TK]D-Fender | din3sh, Have you tried with other phones? Perhaps a model issue? |
| 13:44.02 | din3sh | tried atcom,cisco,grandstream |
| 13:44.06 | din3sh | no luck |
| 13:45.18 | [TK]D-Fender | So tranfers work fine when it's just phones but as soon as that BRI gateway is involved ht audio gets lost? |
| 13:45.19 | din3sh | actually same hardware was working on asterisk 1.4.22, i installed another box 1.8.11 yesterday, evrythg is working fine(almost) except for the attended transfer where pstn numbers are concerned |
| 13:45.26 | din3sh | yes |
| 13:45.34 | [TK]D-Fender | din3sh, Very odd... |
| 13:46.09 | din3sh | and blind transfers via the gateway also works |
| 13:46.18 | [TK]D-Fender | din3sh, If you're stuck with that gateway you might try to use a 1.4 system as a gateway between it and your 1.8 syerver |
| 13:46.25 | [TK]D-Fender | server* |
| 13:46.25 | din3sh | so the trouble is with bridged calls |
| 13:46.46 | din3sh | yes might try that |
| 13:46.55 | din3sh | 1.4 being an interface between the two |
| 13:47.13 | [TK]D-Fender | din3sh, On the cheap side you could do this with an OpenWRT compatible router running 1.4 for about $50 USD tops |
| 13:47.28 | [TK]D-Fender | as for as "low power" options goes. |
| 13:47.37 | WIMPy | Or just replace the GW with a card? |
| 13:47.38 | [TK]D-Fender | din3sh, Or there is always VM's.... |
| 13:47.44 | din3sh | yeah might give it a try |
| 13:48.03 | [TK]D-Fender | WIMPy, If his server supports it, yes. |
| 13:48.28 | [TK]D-Fender | There are many very inexpensive BRI cards out there that might do the job just fine as well |
| 13:48.33 | WIMPy | There is always the USB option. |
| 13:48.54 | [TK]D-Fender | WIMPy, Also viable |
| 13:48.55 | din3sh | this one is pretty expensive |
| 13:49.03 | din3sh | parlay voxip gateway |
| 13:49.04 | WIMPy | I've read here that USB dongles can be uese in production environments successfully. |
| 13:49.30 | WIMPy | Ho many ports do you need? And do you have a spare PCI slot? |
| 13:49.31 | din3sh | around 1400$ for 4BRI lines |
| 13:49.44 | *** join/#asterisk tuxx- (tuxx@pantoff0l.nl) |
| 13:49.46 | din3sh | 4 bri lines coming in |
| 13:50.19 | tuxx- | hi guys, is it possible to record (monitor) every call of one sip peer without adjusting my dialplan? |
| 13:50.23 | WIMPy | Ok, new 4 BRI cards are not that cheap, but the do have quite some advantages. |
| 13:50.54 | WIMPy | tuxx-: No, that's exactely what your dialplan is there for. |
| 13:50.59 | tuxx- | mkay |
| 13:51.12 | din3sh | a strange issue though |
| 13:51.38 | din3sh | good think i've shaved my head, else i'd pull out all my hair |
| 13:52.20 | tuxx- | WIMPy, how about if the sip peer is an agent of a queue, is it possible to record only that agent? I cant seem to find an option like this in the queue.conf |
| 13:52.34 | [TK]D-Fender | tuxx-, because it isn't an option |
| 13:52.53 | tuxx- | so i just have to record all calls of the queue and filter out the peer that i nee |
| 13:52.56 | [TK]D-Fender | tuxx-, Queue records all, or none. Put them as a Local channel memeber and do it in dialplan. |
| 13:52.56 | tuxx- | d |
| 13:52.58 | tuxx- | mkay |
| 13:53.02 | tuxx- | right |
| 13:53.04 | tuxx- | thanks :) |
| 13:57.25 | fprior | [TK]D-Fender, in your opinion is correct to use a * 1.4 as a proxy between an * 1.8 and "something that does not work with 1.8" , as din3sh' case ? |
| 13:57.41 | [TK]D-Fender | ... |
| 13:57.48 | [TK]D-Fender | I just SUGGESTED it to him.... |
| 13:58.13 | [TK]D-Fender | What color was Napoleon's white horse? |
| 13:59.52 | fprior | [TK]D-Fender, I ask this in relation to my problem with * 1.8 and spa400. Time ago I suggest solve this problem putting an * 1.4 between gateway and * production 1.8 |
| 14:00.47 | *** join/#asterisk serafie (~erin@nat/digium/x-knqtonhhslglkaui) |
| 14:01.05 | fprior | [TK]D-Fender, the 1.4 would be a mere proxy to redirect calls to 1.8 |
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| 14:38.04 | Katty | stabs things |
| 14:39.35 | Chainsaw | dives for cover |
| 14:44.11 | jaytee | so no one here has setup Asterisk behind a Cisco ASA 5505? |
| 14:50.37 | Chainsaw | jaytee: You contrain the RTP ports to a specific range, or you allow all UDP to the relevant subnet. |
| 14:53.27 | Chainsaw | jaytee: constrain, even. There isn't a magic Asterisk keyword I'm afraid. Just SIP, RTP, etc. |
| 14:54.03 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:11ee:decb:5fde:e77f) |
| 14:54.05 | fprior | GXW4108 + Asterisk 1.8: very bad clipping problem, I can't hear other's party, last hundred milliseconds are clipped. Any idea ? (echo cancellation enabled, silence suppression disabled.) |
| 14:56.21 | [TK]D-Fender | Moving DOWN to GrandSuck I see.... |
| 14:58.09 | fprior | [TK]D-Fender, I'm testing another brand, this one is not good ? |
| 14:58.32 | [TK]D-Fender | You are jumping from one class of shit, to shit with no class whatsoever |
| 15:00.13 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 15:00.17 | cusco | hi |
| 15:00.59 | cusco | I'm trying to help out a friend with asterisk, he is trying to place a call from x-lite to ekiga .. he says he has no audio and sent me his sip call log: |
| 15:01.10 | cusco | http://ovh.tretas.eu/a.log |
| 15:01.35 | Chainsaw | cusco: No audio? NAT. Sort out STUN. Don't use stun.ekiga.net, it's broken. Try stun.xten.com |
| 15:01.48 | cusco | thei're on the same network |
| 15:01.51 | fprior | [TK]D-Fender, delicious. is what exists in this country. Please, tell me on brand / model to connect * to PSTN lines (excluding shit, please :) |
| 15:02.44 | [TK]D-Fender | Mediatrix, AudioCodes, for SIP,. Sangoma or Digium cards. |
| 15:03.47 | cbdev | what would you guys suggest for a 24 fxs ATA device? preferrably rack mountable. |
| 15:04.05 | [TK]D-Fender | cbdev, Mediatrix 4124 |
| 15:04.11 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:7921:a9ca:f84e:9cd4) |
| 15:04.50 | cbdev | thanks, seems like a perfect fit |
| 15:07.05 | [TK]D-Fender | cbdev, or AudioCodes MP124 FXS. |
| 15:07.40 | [TK]D-Fender | cbdev, Comparable, but AC's configuration is considerably more cryptic. But once you get one down, deploying an army of them is fast by exporting configs. |
| 15:08.11 | cbdev | i'll be needing 4 of those for a medium-sized hotel |
| 15:08.27 | [TK]D-Fender | AC might be a viable plan. Shop around. |
| 15:08.44 | cbdev | well be replacing our existing telephony system with an asterisk-based solution and im looking for legacy analog phone support during transition |
| 15:08.44 | _Corey_ | I've had good luck with Adtran for similar situations |
| 15:09.28 | cbdev | seems like you could get the AC's for a bit less |
| 15:09.52 | [TK]D-Fender | Yup, they are very reputable as a company as well... |
| 15:09.52 | [TK]D-Fender | cbdev, A good bit less at VoIPSupply.com yes... |
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| 15:26.59 | luke-jr | any recommendations for per-minute termination? flowroute seems to be unreliable |
| 15:27.56 | upb | have you tried JesusRouting? |
| 15:31.38 | [TK]D-Fender | Telephony is NOT faith-based |
| 15:34.20 | autofsckk | i want to learn about callcenters with asterisk, can somebody please tell me what should i read to learn all i can about it, maybe a book, pdf, specialized forum or blog? thanks a lot |
| 15:35.13 | [TK]D-Fender | autofsckk, queues.conf.sample. "core show application queue" |
| 15:36.01 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v010-227.mobile.uci.edu) |
| 15:36.48 | autofsckk | im reading this too http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf and thanks for the advice, i will take a look at what you are telling me |
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| 16:22.19 | catphish | is it possible / likely that if a call were hung up during a brief network outage, and rtptimeout were set, the call might carry in indefinitely (perhaps because of a handset stupidly sending rtp traffic for a call it already tried and failed to terminate)? |
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| 16:26.07 | *** join/#asterisk paolosupino (~paolo-sup@net-2-38-113-188.cust.dsl.vodafone.it) |
| 16:26.18 | paolosupino | hi |
| 16:28.36 | Katty | HAI HOW ARE YOU TODAY |
| 16:29.38 | paolosupino | a question about phrasing: When it comes to call transfers there are 2 types: 1. a call is transferred immediately… 2. the caller is put on hold while the 2 extensions talk and when the done talking the caller is trasnfered. how are the 2 options referred to in Asterisk's documentation? |
| 16:30.15 | p3nguin | Blind transfer and attended transfer |
| 16:30.45 | paolosupino | thanx p3nguin :-) |
| 16:40.41 | Katty | i want.... |
| 16:40.45 | Katty | a chihuahua |
| 16:41.15 | p3nguin | Why? They don't really taste all the good. |
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| 16:49.07 | Naikrovek | :) |
| 16:49.27 | Naikrovek | chihuahuas are annoying little shits of animals. they're the little punk of the canine kingdom. |
| 16:49.31 | Naikrovek | you don't want a chihuahua |
| 16:50.12 | Naikrovek | my HR manager brought her chihuahua in yesterday and it went around biting everyone. |
| 16:50.19 | Naikrovek | bit about 8 people. |
| 16:50.29 | Naikrovek | because my HR manager is an HR person none of this was her dog's fault. |
| 16:50.40 | paolosupino | what is the difference between the transfer application and the Tt options of the dial application? |
| 16:50.48 | Naikrovek | and we all got written up for instigating the dog. (not kidding.) |
| 16:51.03 | [TK]D-Fender | paolosupino, Transwer tells the dialplan to throw the caller completely off your server |
| 16:51.33 | [TK]D-Fender | paolosupino, "Tt" are dial() options. that says the caller can indicate they want to transfer the other party WITHIN the system |
| 16:51.47 | paolosupino | [TK]D-Fender: interesting features… |
| 16:52.20 | paolosupino | can the Tt be used to transfer to an extension on another PBX? |
| 16:52.28 | [TK]D-Fender | paolosupino, Any decent SIP/IAX2 device will have its own built-in feature and you should never have to use DTMF based options as configured in your dialplan |
| 16:52.36 | citywok | paolosupino: it can be used to tranfer to any extension. how you handle that extension is up to you. |
| 16:53.05 | citywok | paolosupino: the Tt options use the feature codes to execute the transfer, like [TK]D-Fender said you should nevern ed them unless you're using some funky analog device on an ATA |
| 16:53.51 | citywok | s/n ed/ need/ |
| 16:55.26 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v010-227.mobile.uci.edu) |
| 16:57.49 | [TK]D-Fender | Or unless the ATA is dumb. Some cheap crap softphone either cripple the feature or never have a version offering such standard functionality. |
| 17:03.20 | paolosupino | [TK]D-Fender, citywok: I looked at the documentation for the balk of the phones that we're going to use (unidata WPU-7800) and there it says that you have to press the send button to put the call on hold, call the extension to transfer to, then menu and choose transfer… Does this mean I need to setup phone features or is that DTMF based? |
| 17:03.48 | citywok | paolosupino: the phone will do it in SIP |
| 17:04.03 | citywok | you won't need to worry about it. |
| 17:04.22 | paolosupino | citywok: nothing to do… Me like it very much ;-) |
| 17:04.23 | citywok | although, that sounds like a horrible transfer process. lol |
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| 17:04.38 | [TK]D-Fender | paolosupino, Those phones look like cheap crap... |
| 17:04.43 | [TK]D-Fender | ~wifisip |
| 17:04.43 | infobot | [~wifisip] Please refer to ~wifivoip for info on this. |
| 17:04.47 | paolosupino | for my protection I can nay say that I didn't choose the phones… |
| 17:04.49 | [TK]D-Fender | ~wifivoip |
| 17:04.49 | infobot | [~wifivoip] Wi-Fi (802.11a/b/g) VoIP phones on the market may possess any of the following drawbacks : poor battery life, limited range, lack of call features (many lack transfer, etc), poor or no NAT traversal, shoddy construction, etc. Because of this they are usually not recommended. Some alternatives are : VoIP ATA + cordless, DECT w/ VoIP base, etc |
| 17:04.54 | citywok | yea... it sounds like it based on their process for transferring calls |
| 17:05.04 | p3nguin | paolosupino: That sounds like a regular phone transfer to me. |
| 17:05.18 | citywok | paolosupino: well, when 20% of them fail each year you're going to be tired of dealing with them :p |
| 17:05.44 | paolosupino | I'm not the hardware admin in the company either ;-) |
| 17:06.05 | citywok | p3nguin: i like the transfer button being right on the phone, it makes training users so much easier |
| 17:06.23 | p3nguin | infobot: no, wifisip is <reply> see wifivoip |
| 17:06.23 | infobot | p3nguin: okay |
| 17:06.30 | p3nguin | ~wifisip |
| 17:06.30 | infobot | [~wifivoip] Wi-Fi (802.11a/b/g) VoIP phones on the market may possess any of the following drawbacks : poor battery life, limited range, lack of call features (many lack transfer, etc), poor or no NAT traversal, shoddy construction, etc. Because of this they are usually not recommended. Some alternatives are : VoIP ATA + cordless, DECT w/ VoIP base, etc |
| 17:06.39 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 17:07.09 | paolosupino | p3nguin: transfer button is stupid proof as long as it's red… |
| 17:07.25 | p3nguin | I don't know what that means. |
| 17:08.04 | paolosupino | p3nguin: if they are of a different color people will never understand what they are for even if they have the title clearly written (I know that from experience). |
| 17:08.06 | citywok | has anybody used the panasonic tgp500? our receptionists love them, after having aastra cordless phones at least. lol |
| 17:09.28 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 17:20.08 | paolosupino | (I feel stupid asking) since I am going to use DTMF to transfer calls is the only difference between blind xfer and attendant xfer the button sequence? |
| 17:21.20 | pabelanger | paolosupino, blind transfer are transferring a call and not caring who or what answers the other side, attended transfer you usually wait for somebody at the far side to answer the call before doing the transfer |
| 17:22.39 | Kobaz | (usually) |
| 17:22.44 | Kobaz | unless it's a blond transfer |
| 17:22.50 | paolosupino | pabelanger: that I understood… |
| 17:23.37 | paolosupino | Kobaz: I prefer brunettes… In which case I'd rather have her walking across the office and pick me up personally ;-) |
| 17:23.50 | Kobaz | hehe |
| 17:24.17 | Kobaz | blond transfer is when you attended transfer, and then complete the call before the other side picks up |
| 17:24.35 | Kobaz | i forgot who coined that term, i think it was russelb |
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| 17:30.19 | paolosupino | Kobaz: learned a new term… can any one coin a term with brunettes (I still prefer them to blonds)? ;-) |
| 17:32.56 | p3nguin | Try pressing the brunette transfer key. |
| 17:33.35 | paolosupino | p3nguin I only have a redhead transfer key ;-) |
| 17:37.09 | _Corey_ | "blond transfer"... that's great! haha |
| 17:37.27 | [TK]D-Fender | I accept blond transfers.... |
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| 17:41.54 | Kobaz | heh |
| 17:42.33 | Katty | hmm |
| 17:42.35 | paolosupino | [TK]D-Fender: at what exchange rates from redheads and brunettes? |
| 17:42.46 | Katty | maybe an italian greyhound then |
| 17:42.48 | Katty | or a corgi |
| 17:42.49 | [TK]D-Fender | I don't exchange, I collect ;) |
| 17:42.49 | Katty | with a tail |
| 17:42.52 | Katty | i forget which breed that is |
| 17:43.06 | Qwell | glomps Katty |
| 17:43.13 | Katty | ohai |
| 17:43.17 | [TK]D-Fender | And as they say, "Gentlemen prefer blonds". |
| 17:43.25 | [TK]D-Fender | Of course ... I'm no gentleman ;) |
| 17:43.27 | Katty | they do not |
| 17:43.37 | Katty | brunettes are way cooler. |
| 17:43.42 | paolosupino | [TK]D-Fender: well if you want blonds I'm think you should exchange the brunettes and redheads you have… |
| 17:43.56 | Katty | redheads are fiesty |
| 17:44.00 | Katty | bit too fiesty for me |
| 17:44.54 | tzanger | you got that right |
| 17:44.54 | tzanger | my wife's a redhead |
| 17:45.00 | tzanger | I'm not sure if I would make the same choice if I did it again |
| 17:45.57 | Qwell | tzanger: I am so telling. |
| 17:46.04 | mchou | haha |
| 17:46.13 | tzanger | heh |
| 17:46.30 | tzanger | it's okay, she's not so sure she'd make the same mista^Wchoice again either |
| 17:46.51 | mchou | tzanger: the question is whether you're a redhead |
| 17:47.25 | *** join/#asterisk timahvo1 (~rogue@197.177.99.180) |
| 17:47.35 | tzanger | me? no |
| 17:47.41 | tzanger | I'm german. I think they eradicated redheads well before Hitler |
| 17:47.44 | Katty | that's part of the problem. |
| 17:47.49 | Katty | tzanger: agreed. |
| 17:47.53 | Katty | tho i'm more welsh than german |
| 17:47.54 | Katty | close enough |
| 17:48.41 | mchou | tzanger: I'm not faniliar with german history. did they have a redhead pogrom? |
| 17:48.50 | mchou | familiar* |
| 17:48.51 | tzanger | it's all europe and england's like a suburb of berlin, isn't it? ;-) |
| 17:48.58 | Katty | *hee* |
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| 17:53.35 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 17:54.40 | paolosupino | tzanger: I don't know about the rest of Europe Italy and Italian's (me included) are protected from the Germanic tribes of norther europe by the Alps… so we're not a suburb of Berlin :-) |
| 17:56.21 | tzanger | :-) |
| 17:56.27 | tzanger | I think most people should have a mountain range between them and the germans |
| 17:56.30 | tzanger | ... including the germans |
| 17:57.20 | paolosupino | well you have the |
| 17:58.18 | paolosupino | tzanger: than how do you explain the idea of putting the black forest (and the mountains they're on) in the corner of the country? |
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| 18:15.50 | paolosupino | another nice phone that I have to setup: can anyone make my life simple and direct me to XMLs for SNOM360 phones? |
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| 18:22.55 | citywok | paolosupino: http://www.voip-info.org/wiki/view/snom+360 |
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| 18:29.04 | paolosupino | citywok: crap, |
| 18:29.26 | paolosupino | citywok: I looked it up in google and it gave me a different link. |
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| 18:43.42 | bent_screwdriver | anyone know if the newer polycom firmware make the phone presense show differently for ringing/vs on call? or do they still show steady red for both? (IP650's) |
| 18:47.18 | _Corey_ | bent_screwdriver: It's not the firmware |
| 18:47.36 | bent_screwdriver | ok w/e sip software, etc. |
| 18:48.00 | _Corey_ | check out the "attendant" option in the phone config file and you might have more luck if you want your contacts to ring |
| 18:49.07 | bent_screwdriver | yeah, tried that and it worked ok, but i did not like lots of other stuff it did. maybe i can only use piceces of it....i'll take another look. its been a while |
| 18:56.49 | Naikrovek | yeah polycom firmware is intimidating a bit at first. |
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| 20:14.05 | din3sh | p3nguin you there? |
| 20:14.43 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 20:16.07 | p3nguin | Yes. |
| 20:19.48 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
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| 20:29.58 | din3sh | have almost given up on the failed attended transfer |
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| 20:33.07 | p3nguin | Did you ever pastebin a call where you were transferring with audio failure? |
| 20:34.13 | *** join/#asterisk timahvo1 (~rogue@197.177.99.180) |
| 20:35.07 | din3sh | yeah i did, no luck |
| 20:35.08 | din3sh | http://paste2.org/p/1978331 |
| 20:35.11 | din3sh | here it is |
| 20:36.05 | din3sh | 4999493(pstn number via BRI gateway)------->calls 2729(SIP extension)-------->transfers to 2713(SIP extension) |
| 20:36.49 | p3nguin | .5 is the VoXip gateway, .151 is asterisk? |
| 20:36.55 | din3sh | moh starts off, after attended xfer, 4999493 cannot hear anythg, 2713 can hear 4999493 |
| 20:37.02 | din3sh | yes |
| 20:38.11 | din3sh | i have tried to call 2 pstn numbers via the gateway and bridge them with attended transfer, both cannot hear anythg even being connected |
| 20:38.40 | p3nguin | .52 is what? |
| 20:38.58 | din3sh | the sip phone 2729 |
| 20:43.10 | p3nguin | Okay, so you now have one SIP phone and one call on a gateway appliance. That makes a complete call. The other phone you are involving in the transfer is at .33? |
| 20:43.11 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 20:43.20 | din3sh | yes |
| 20:43.48 | din3sh | also blind xfers work correctly |
| 20:44.17 | p3nguin | What is this Voip Phone 1.0 |
| 20:45.56 | din3sh | atcom phone |
| 20:46.15 | p3nguin | Can you try another user agent? |
| 20:46.25 | p3nguin | Do you have other phones? |
| 20:48.00 | p3nguin | Even a soft phone would be okay. Maybe zoiper or something else with a good reputation. |
| 20:48.43 | din3sh | have tried cisco 7911 |
| 20:48.52 | din3sh | grandstream 1450 |
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| 20:49.08 | p3nguin | "good reputation" |
| 20:49.09 | din3sh | even softphone |
| 20:49.21 | *** part/#asterisk asterisk-Tester (~ramy@80.79.159.42) |
| 20:49.34 | din3sh | the thing is the same setup worked few days back |
| 20:49.44 | p3nguin | Peer audio RTP is at port 0.0.0.0:10008 |
| 20:49.45 | din3sh | and still does with asterisk 1.4.xx |
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| 20:50.08 | din3sh | have installed another box with 1.8.11 |
| 20:50.10 | p3nguin | That is a concern. I want to eliminate your Voip Phone 1.0 UA. |
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| 20:50.38 | p3nguin | I want to use something with a good reputation and see if the problem persists. |
| 20:50.47 | din3sh | i tried cisco |
| 20:50.53 | p3nguin | (1549.05) <p3nguin> "good reputation" |
| 20:50.59 | din3sh | they have been the most stable for me |
| 20:51.00 | din3sh | loll |
| 20:51.01 | p3nguin | You're not catching on. |
| 20:51.22 | p3nguin | If you don't have a good hard phone, use zoiper soft phone. |
| 20:51.54 | din3sh | you think it might be the phone? |
| 20:52.00 | p3nguin | Cisco 7900 series with SIP does not have a good reputation. |
| 20:52.25 | p3nguin | I can't say for sure until you eliminate it from the equation. This is called troubleshooting. |
| 20:52.54 | din3sh | have tested snom/polycom/cisco/linksys/atcom/grandstream |
| 20:53.00 | p3nguin | Show me. |
| 20:53.12 | din3sh | am not at office currently |
| 20:53.17 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:53.20 | din3sh | its past midnight here |
| 20:53.23 | p3nguin | Test with the Polycom and show me the debug. |
| 20:53.30 | p3nguin | We know they work. |
| 20:53.35 | din3sh | ok will do once at office |
| 20:54.24 | p3nguin | That Voip Phone 1.0 UA is the only one I see with RTP audio at 0.0.0.0 instead of the real IP address. |
| 20:54.58 | din3sh | with rtp debug on |
| 20:55.01 | din3sh | i get these |
| 20:55.03 | din3sh | [Apr 13 09:54:49] DEBUG[10683] res_rtp_asterisk.c: No remote address on RTP instance '0x7fdc100135d8' so dropping frame |
| 20:55.16 | din3sh | you think this might be causing it? |
| 20:55.26 | p3nguin | Right. 0.0.0.0 is not a valid remote address. |
| 20:56.33 | p3nguin | So you need to use a phone that you know works correctly. Without having the data to show a good phone, you have to rely on reputation. Polycom has a good reputation, and therefore should be suitable for testing in place of a possibly busted phone. |
| 20:56.48 | din3sh | like i told u, i have same hardware setup working with no problem but only on asterisk 1.4.22 |
| 20:57.07 | p3nguin | If the Polycom shows the same thing, the phone probably isn't the problem. |
| 20:57.33 | p3nguin | Lots of things changed between 1.4.22 and 1.8.11.0. |
| 20:57.52 | jsmith | 0.0.0.0 is a way of telling the phone not to send audio |
| 20:57.58 | jsmith | (like when it's receiving hold music) |
| 20:57.59 | din3sh | also attended transfers between hard phones work well |
| 20:58.17 | din3sh | 0.0.0.0 is only when the gateway comes into play |
| 20:59.44 | din3sh | like [TK]D-Fender suggested, i'll try to hook asterisk 1.4.22 in the middle to interface with the gateway and ast 1.8.11 |
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| 21:01.07 | din3sh | my sip.conf http://paste2.org/p/1977892 |
| 21:01.17 | p3nguin | So the 0.0.0.0 is normal when the call goes on hold? |
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| 21:02.13 | din3sh | i dont think so |
| 21:02.25 | din3sh | have to run sip debug on ast 1.4.22 |
| 21:02.28 | din3sh | and compare |
| 21:02.50 | p3nguin | No, you have to use a good phone on the current system. |
| 21:03.15 | p3nguin | You can't go changing out major components to troubleshoot minor components. |
| 21:03.38 | p3nguin | Buy a new car because you got a flat tire? |
| 21:03.56 | p3nguin | The car must have been bad, because this new car I got doesn't have a flat tire. |
| 21:04.14 | din3sh | i get your point |
| 21:04.43 | din3sh | will try the polycom or zoiper |
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| 21:05.18 | p3nguin | If all you care about is making it go, proxy it through 1.4. If you care about finding out what the problem is and possibly fixing it, you have to effectively troubleshoot it. |
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| 21:07.05 | din3sh | the proxy thing cannot be a viable solution |
| 21:07.29 | din3sh | i have a few client setups with that similar gateway |
| 21:07.49 | din3sh | if i want to move to 1.8.xx on all my systems i have to find the cause and the solution |
| 21:09.10 | p3nguin | I'm glad you agree. I hate workarounds when legitimate solutions have not been attempted. |
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| 21:12.02 | din3sh | if the polycom does the same thing, what next? |
| 21:12.34 | [TK]D-Fender | get another BRI interface |
| 21:14.47 | din3sh | yes [TK]D-Fender, i'll be ordering the digium bri card shortly |
| 21:15.51 | din3sh | i guess for the few installations with the gateway, i'll have to stick with ast 1.4.xx (at least till i find a solution to the issue) |
| 21:17.33 | din3sh | its 01:15 saturday mrning here. am still talking work! |
| 21:17.35 | din3sh | :o |
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| 23:34.52 | bluregard | good evening everyone |
| 23:36.56 | bluregard | when using asterisk realtime for extensions, if in extensions.conf you have a [test] context that has switch => Realtime when you do a dialplan show test should you see all of the extensions in the database? Or are they not read until an attempt to access them is made? |
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