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| 08:15.59 | babak | Hi, If I want to force outgoing call use "specified channel" number in one E1 or specified CIC (ss7) how I should fill Dial Application ? |
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| 08:33.18 | bounceman | This is somewhat of a stupid question perhaps. How can I see which module is in use? chan_sip or chan_pjsip, they are both loaded and their use count is 0. Even though I have made test calls to the system. |
| 08:34.23 | bounceman | Ah nvm chan_sip.c:4009 |
| 08:34.26 | bounceman | the CLI told me |
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| 11:11.37 | bounceman | Can this cause issues in the asterisk when handling SIP? http://codepad.org/z6yFSzv1 The spikes are the Recv-Q for UDP 5060 |
| 11:11.46 | bounceman | Does it even matter when they spike and shortly after go down to 0? |
| 11:12.18 | bounceman | Looks like it had 4024 bytes in queue for a whole second, lockup? |
| 11:19.02 | jkroon | bounceman, http://jkroon.blogs.uls.co.za/it/voip/asterisk-massively-speeding-up-those-register-requests |
| 11:19.21 | jkroon | yea, on astdb fsync/sync() call during a REGISTER frame that got received. |
| 11:19.46 | bounceman | We're not there yet jkroon. Will backtrace reveal anything interesting? |
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| 11:20.22 | jkroon | backtrace? |
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| 11:21.23 | jkroon | chan_sip has a thread dedicated to receiving packets and processing them. my measurements has shown that the far majority of frames gets processed in a matter of microseconds, except for REGISTER. after reading the code the astdb trying to sync to disk was the only viable answer. |
| 11:21.46 | jkroon | I had backlogs up to 256KB, after moving astdb into RAM instead of disk ... problem gone. |
| 11:23.00 | jkroon | oh, gdb bt ... been there ... doesn't actually lock up, and you'll need to get the btrace during the sync call to get anything useful. strace might provide more details but I don't know how to get strace to attach to that specific thread. |
| 11:26.54 | jkroon | strugling myself with a func_odbc issue. |
| 11:27.11 | jkroon | exten => s,n,Set(HASH(pin)=${ODBC_pin(${enteredpin})}) |
| 11:27.23 | jkroon | but I can't seem to find a reliable way to figure out if that has returned something. |
| 11:27.33 | jkroon | there is definitely no prior use of HASH(pin) before this. |
| 11:28.35 | bounceman | Maybe you can see anything in the odbc TraceFile if enabled |
| 11:29.12 | jkroon | wouldn't mind enabling it, but I need an in-dialplan solution. |
| 11:31.28 | bounceman | If tuning the udp buffers does not help us we will move the astdb into the RAM |
| 11:34.46 | jkroon | tuning those buffers will just prevent packets from going lost, so that might be adequate for you. |
| 11:35.00 | jkroon | is there even a way to configure in asterisk how big those buffers should be? |
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| 11:39.28 | bounceman | not * specific what I know. It is system wide |
| 11:40.48 | jkroon | there are ioctl()s to change the buffer sizes on a per-socket basis. fact is enlarging the buffer just prevents frames from going lost, doesn't ensure they get processed timeously. |
| 11:41.32 | bounceman | we're in a state where we are happy IF they are processed. |
| 11:45.26 | jkroon | net.ipv4.udp_mem = 378465 504622 756930 |
| 11:45.27 | jkroon | net.ipv4.udp_rmem_min = 4096 |
| 11:45.27 | jkroon | net.ipv4.udp_wmem_min = 4096 |
| 11:45.48 | jkroon | well, in that case those are probably the sysctl settings you want to tamper with. not sure how the udp_mem one works though. |
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| 12:00.50 | bounceman | jkroon: I still find it unclear how and when the astdb is being used. |
| 12:00.52 | bounceman | Care to fill me in? |
| 12:01.36 | jkroon | if you run "database show" you'll find a bunch of entries like /SIP/Registry/102 : 192.168.42.117:5060:3600:102:sip:102@192.168.42.117 |
| 12:02.05 | jkroon | on REGISTER the information is written to the DB in order to persist registrations over SIP reloads and even asterisk restarts. |
| 12:02.47 | jkroon | ast_db_get("SIP/Registry", peer->name, data, sizeof(data)) <-- during peer load. |
| 12:03.04 | jkroon | ast_db_put("SIP/Registry", peer->name, data) <-- in parse_register_contact |
| 12:09.27 | bounceman | Good info |
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| 15:17.09 | Haris | guys, how can work work-around for browsers going unreachable on register with asterisk for ws/wss traffic ? |
| 15:17.45 | jkroon | fix the network? |
| 15:18.13 | jkroon | if they are unreachable it means asterisk can't communicate with them ... so you need to find why asterisk can't communicate with them and fix that. |
| 15:21.28 | Haris | I'v turned off windows firewall on my win7 laptop. I'm not sure what else could be blocking it |
| 15:24.51 | Samot | Your router. |
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| 15:29.09 | luminblade | i'm using AGI to Dial, is it possible to not ACK the 200/Answer on the Called Channel until the Calling Channel also ACK's the 200/Answer? |
| 15:29.47 | Samot | Why would you ACK something that actually hasn't sent back an ACK? |
| 15:30.52 | Samot | If Asterisk is sending back an ACK to an endpoint, it's acknowledging whatever SIP message the endpoint sent. |
| 15:32.58 | luminblade | Asterisk is ACK'ing the outbound channel before relaying the 200 to the inbound channel. |
| 15:33.28 | luminblade | i just want to delay that ack until i'm sure the inbound channel also ack's. |
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| 15:35.33 | luminblade | i thought the 'a' option on Dial might do it, but it doesn't "This option can be used to answer the calling channel before doing anything on the called channel." |
| 15:39.58 | [TK]D-Fender | that forcible does the opposite of what you just asked |
| 15:40.21 | [TK]D-Fender | * won't 200 the current channel unless it has a reason to force the audio channel up |
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| 15:49.05 | Haris | router should be edge network device. it should not be filtering LAN based traffic ? |
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| 15:57.36 | [TK]D-Fender | What do you think? |
| 15:58.29 | luminblade | the issue is asterisk is a b2bua, so it's answering half of the call without waiting for the original caller to acknowledge the answer. |
| 15:59.13 | [TK]D-Fender | huh? |
| 15:59.53 | [TK]D-Fender | Originally caller doesn't ACK. They are the caller. * ACKS. |
| 16:00.04 | [TK]D-Fender | or at least 220 OK. |
| 16:00.08 | [TK]D-Fender | Show us the call. |
| 16:08.29 | luminblade | SRC (INVITE) -> AST, AST (100) -> SRC, AST (INVITE) -> DST, DST (100) -> AST, DST (183) -> AST, DST (200) -> AST, AST (ACK) -> DST (Delay this ACK, until after SRC (ACK), AST (200) -> SRC, SRC (ACK) -> AST, ..., people are talking, ... END |
| 16:08.37 | luminblade | wow, that looks bad, sorry |
| 16:09.19 | [TK]D-Fender | It does. |
| 16:09.21 | [TK]D-Fender | Show us the actual call |
| 16:09.30 | [TK]D-Fender | * CLI |
| 16:09.33 | [TK]D-Fender | Not external debug |
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| 16:11.09 | luminblade | calls are working fine, i'm not trying to solve a sip signaling issue, i just had a configuration question about delaying the ACK to the 200. |
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| 16:15.53 | [TK]D-Fender | Show us the call |
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| 16:56.15 | ntt | Hi, I'm using asterisk 13.9.1 and freepbx. Randomly I have problems with outgoing calls: I see "Got SIP response 503 "No more gateways" back from " ....someone can help me please? |
| 16:57.31 | file | that would be coming from a remote SIP server, not from Asterisk itself |
| 16:58.06 | ntt | file: thank you.... yes, I know.... but why this happens randomly? |
| 16:58.28 | ntt | connection is good.... I have a ping of 50ms |
| 16:58.34 | file | that would be a question for the other side... |
| 16:59.11 | ntt | it is difficult for me to catch a log because the problem happens in an unpredictable manner |
| 16:59.28 | file | it's not a problem in Asterisk itself, it sent a call request and that is what it got back |
| 16:59.49 | ntt | ok.... I will try to solve with my provider |
| 16:59.56 | ntt | btw, thank you |
| 17:01.01 | file | you're welcome |
| 17:03.51 | ntt | file: stupid question: could be related to codec? |
| 17:04.06 | file | I doubt it, it looks to be related to routing by your provider |
| 17:04.07 | ntt | actually I'm using only alaw. no other. |
| 17:04.35 | file | they are trying to least cost route your call and they have nowhere to send it, but that's just a guess |
| 17:10.15 | ntt | ok... could be the reason.... but in the past my provider told me that this problem happens only with asterisk/freepbx. They have a lot of ATA with analog phones and don't have this kind of problem.... for this reason I thought that the problem is asterisk |
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| 18:05.18 | Samot | ntt: "503 No More Gateways" is generally the response sent back when you've maxed out your channels with them. |
| 18:05.44 | Samot | ntt: It happens with Asterisk/FreePBX because there's nothing limiting your "lines" |
| 18:05.57 | ntt | Samot: can you explain me ? |
| 18:06.13 | ntt | I can use "maximum channels" in the trunk if needed |
| 18:06.28 | Samot | Providers may or may not limit you on concurrent calling. |
| 18:06.38 | drmessano | ATA = 1 call.. so yeah, no limit |
| 18:06.44 | Samot | Or 2 |
| 18:06.44 | drmessano | Asterisk allows MANY calls |
| 18:07.11 | Samot | If it's a Resi service, you are probably on a tight limit. |
| 18:07.12 | robmal | That is >3 |
| 18:07.46 | Samot | Providers that don't want PBXes on their network will limit concurrent calls. |
| 18:08.21 | Samot | Probably with the logic that the service they are providing shouldn't be used on a PBX system. |
| 18:08.58 | drmessano | Providers that intend for the user to make 1 call (ok, plus CW) that simulates a POTS line HAVE to limit calls or else people will put a PBX on there and abuse the hell out of a flat rate |
| 18:09.26 | drmessano | So you basically need to limit your calls in Asterisk if don't want that to happen |
| 18:09.36 | Samot | Some providers just limit "channels" to control costs and revenue. |
| 18:09.59 | Samot | "Oh you want to make more than 2 calls, you have to buy extra channels" |
| 18:10.37 | drmessano | Samot: Im thinking more along the lines "This service/rate is intended for 1/2 calls, ALA ATA" |
| 18:10.40 | Samot | Unlimited service for VoIP is inherently abused. |
| 18:11.09 | Samot | drmessano: I'm giving the benefit of the doubt it's not Resi level service. |
| 18:11.47 | drmessano | ntt: Who is the provider? |
| 18:11.57 | ntt | Convergenze (www.convergenze.it) |
| 18:12.55 | drmessano | Do you have the 14.64 a month tier? |
| 18:13.14 | ntt | yes |
| 18:13.18 | drmessano | Yep |
| 18:13.29 | drmessano | Thats what I like to call a "Simulated POTS" bundle |
| 18:13.59 | drmessano | Basically unlimited, it's self limiting because they expect the user to drop an ATA on there |
| 18:14.13 | drmessano | If you try to use a PBX, you're going to hit the 1/2 call limit |
| 18:14.42 | drmessano | If you want more concurrent calls, you need one of their other packages |
| 18:15.26 | Samot | Like the PBX package. |
| 18:15.34 | Samot | That says "This is what you want if you want more than 2 calls" |
| 18:15.40 | ntt | mmm |
| 18:15.53 | ntt | so this happens when there are more than 2 calls active on my pbx? |
| 18:16.01 | Samot | ntt: Either upgrade your package or limit Asterisk to 2 max channels. |
| 18:16.06 | drmessano | ntt: 1 or 2.. yes |
| 18:16.14 | Samot | You get a 503 No More Gateways. |
| 18:16.40 | Samot | Inbound calls will most likely get the European version of a busy signal. |
| 18:17.10 | drmessano | Samot: Basically, Bono? |
| 18:17.26 | Samot | This is nothing more but "traditional" phone service but with a different delivery method. |
| 18:17.55 | Samot | drmessano: Barf. |
| 18:18.36 | drmessano | Samot: IS this the european busy signal now? |
| 18:18.38 | drmessano | https://www.youtube.com/watch?v=UJXqnYCWW7Y |
| 18:18.40 | Samot | drmessano: Let's be honest, the EU isn't going to let an Irishman be their voice. Even if it is Bono. |
| 18:18.54 | Samot | They still have some standards. |
| 18:19.25 | drmessano | I just saw that EP the other night. They ripped on Bono so hard. It was beautiful |
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| 18:23.48 | Samot | Yes. Its a good one. |
| 18:25.54 | sikun | I'm very new to PBX/SIP/etc... I'm a system admin and I've been tasked with moving a trunk off of a currently virtualized TrixBox PBX to a physical machine on which I have installed AsterisksNOW. I've been reading the manual I found for Asterisks, but I'm just trying some stuff out today. |
| 18:26.10 | sikun | What would be the correct trunk type to link both PBX's |
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| 18:31.17 | Samot | SIP or IAX. They will both do what you need. |
| 18:35.23 | sikun | ok |
| 18:35.24 | sikun | thanks |
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| 20:19.46 | ntt | I'm trying to simulate a call from the cli with the originate command, but I have "Failed to authenticate on INVITE to '"Anonymous" <sip:anonymous@anonymous.invalid:5160>;tag=as5fb3337b'" ...someone can help me pleasE? |
| 20:20.11 | ntt | first: it is possible to call an external number (not an extension, for example a mobile) in this way? |
| 20:24.56 | [TK]D-Fender | You don't call "an external number" |
| 20:25.05 | [TK]D-Fender | that is not a magical thing |
| 20:25.21 | [TK]D-Fender | You call a number to some service provider or out a physical line you have |
| 20:25.40 | [TK]D-Fender | Your originate is clearly using SIP and the auth isn't good for the format you're using |
| 20:26.11 | [TK]D-Fender | CLI originate has no callerid because there is on original channel to base it on, and it isn'tt a parameter you can set |
| 20:26.36 | [TK]D-Fender | So if you pass that as-is our a peer, or direct tbased on a URI it's going to be very ... "dislked" most likely |
| 20:27.33 | [TK]D-Fender | So if you are not forcing a CID on a peer then you'll definately have to at the very least use a call-file or AMI Originate instead, or use a Local channel as your "Channel:" and set that there |
| 20:27.41 | [TK]D-Fender | before doing an actual dial |
| 20:28.01 | ntt | basically I need to originate more than 2 calls using my trunk. I asked on the channel for a problem with my asterisk/freepbx that return 503(no more gateways) randomly when I try to call , and one possible reason for this is that my provider blocks the number of concurrent call to 1 or 2 |
| 20:28.25 | ntt | I'd like to check this |
| 20:28.31 | [TK]D-Fender | We should be looking at the actual call before guessing what the problems are |
| 20:28.34 | [TK]D-Fender | ^ |
| 20:28.50 | [TK]D-Fender | but for now ... I'm heading home. BBIAB |
| 20:29.06 | drmessano | ntt: Why are you doing this the hardest way possible? |
| 20:29.12 | drmessano | Make 1 call |
| 20:29.19 | drmessano | Make #2 call |
| 20:29.22 | drmessano | Make #3 call |
| 20:29.49 | drmessano | I would imagine 2 will work and 3 wont |
| 20:30.07 | ntt | I'm not near the pbx/phones... |
| 20:30.11 | drmessano | There is another more complex option. Call the provider and ask |
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| 20:31.35 | ntt | mmm....ok ... Tomorrow I will give a try (now it's late - 10PM) |
| 20:35.34 | Samot | Woo. |
| 20:35.36 | Samot | Late. |
| 20:35.40 | Samot | 10pm. |
| 20:35.46 | drmessano | IKR |
| 20:36.39 | Samot | He should try GV. |
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| 21:01.15 | drmessano | Totally |
| 21:01.35 | drmessano | GV is Totes MaGotes |
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