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| 00:00.43 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 00:06.34 | hesco1 | OK, I wonder what this means I should investigate next: WARNING[31958]: app_queue.c:6131 queue_exec: Unable to join queue 'cf_support' |
| 00:12.53 | hesco1 | queue show tab-completes with default, giving me access to none of the other queues. |
| 00:14.39 | hesco1 | I moved my #include to the general context of queues.conf w/o changing that behavior. |
| 00:14.44 | hesco1 | ~pb |
| 00:14.44 | infobot | 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.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
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| 00:35.11 | hesco1 | never mind on that paste bin. composing it answered my question. |
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| 01:45.23 | MRH2_ | hi can anyone recommend a way of setting the channel field to a fixed entry |
| 01:46.48 | MRH2_ | for <10 simultaneous calls |
| 01:49.15 | MRH2_ | i'm thinking something like SIP -> Dial Zap channel out -> Zap channel in |
| 01:49.50 | WIMPy | "channel field"? |
| 01:50.04 | MRH2_ | then i know the channel will be Zap/2-1 to Zap/2-10 |
| 01:50.18 | *** join/#asterisk ectospasm (~ectospasm@unaffiliated/ectospasm) |
| 01:50.28 | MRH2_ | as in cdr field |
| 01:51.17 | WIMPy | What kind of interface? |
| 01:51.25 | MRH2_ | SIP |
| 01:51.37 | WIMPy | And BTW zaptel has been replaced by dahdi several years ago. |
| 01:51.49 | MRH2_ | yeah been a while since i messed with asterisk |
| 01:54.44 | MRH2_ | do you happen to know if asterisk 11 supports additional custom cdr fields |
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| 01:56.16 | WIMPy | Hasn't there always been the option? But I never tried it. |
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| 02:16.56 | MRH2_ | I think i'm going to loop it through a isdn card until i upgrade and sort things out properly |
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| 02:36.17 | WIMPy | Err, what's that good for? |
| 02:36.39 | MRH2_ | just will force the channel field |
| 02:36.53 | MRH2_ | anyone know a low end gateway that can be used for branch emergency calls SIP phones over a PSTN line (like a gateway with 1 ethernet and 1 FXO) |
| 02:37.26 | WIMPy | Why do you care about the channel name? |
| 02:38.07 | MRH2_ | another application needs it and i'm out of fields |
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| 02:48.17 | MRH2_ | maybe a dedicated context then a stored procedure on the db |
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| 03:07.00 | MRH2_ | gtg thanks anyway WIMPy it's useful to type stuff out even if i'm not at my best at 4am ;) |
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| 05:05.32 | AliRezaTaleghani | hi all |
| 05:06.03 | AliRezaTaleghani | i am trying to test the maximum call call be handled over asterisk |
| 05:06.36 | AliRezaTaleghani | due to test, i have to server (CentOS6.3) with Asterisk 1.8.16 (compiled from source) |
| 05:08.14 | AliRezaTaleghani | one is originateing call with asterisk cli originate command and drop the change into Wait application for a random of 300 to 600 soconds... |
| 05:08.49 | AliRezaTaleghani | the other sever is answering call and drop them into a MusicOnHold Application... |
| 05:09.51 | AliRezaTaleghani | after about 200 calls i face with some errors with tell there are some retransmission on critical sip messages.... |
| 05:10.22 | ChannelZ | That's packets disappearing |
| 05:15.56 | AliRezaTaleghani | ChannelZ: what dose it mean? |
| 05:16.30 | AliRezaTaleghani | there is just an SRX650 between servers... |
| 05:17.00 | ChannelZ | Either SIP packets aren't making it off your network, or the replies aren't making it back. Could be being dropped by anyone.. the interface, routers, not sure |
| 05:18.47 | AliRezaTaleghani | :-/ ooom |
| 05:19.06 | AliRezaTaleghani | ChannelZ: I will try to check it... |
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| 05:29.57 | AliRezaTaleghani | channelz: do u know what's the reason of this error: |
| 05:30.00 | AliRezaTaleghani | ERROR:[XXXX]: utils.c:1168 ast_ccarefullwrite: write() returened error Broken pipe |
| 05:31.45 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 05:51.13 | wdoekes | AliRezaTaleghani: AGI script that exits prematurely, I think |
| 05:51.46 | wdoekes | you need to read all data, not just the lines you think you need |
| 05:52.33 | wdoekes | weak |
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| 05:53.58 | wdoekes | oej: pinefrog is the one with the rtcp edits? |
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| 06:08.04 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:08.07 | schmidts | good morning |
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| 06:34.57 | oej | wdoekes: Yes! |
| 06:35.00 | oej | Good morning folks |
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| 06:50.13 | mcolombo | Good Morning! |
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| 07:15.49 | ChannelZ | good morning, blah! |
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| 07:22.57 | schmidts | hehe ChannelZ good night |
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| 07:26.53 | ChannelZ | What is it I'm thinking of.. was it the Count from the Muppets (Sesame street?) that would end every sentence with 'blah'? |
| 07:27.51 | ChannelZ | ohhhhhhhh |
| 07:27.52 | schmidts | i dont know narf |
| 07:27.54 | ChannelZ | Greg The Bunny |
| 07:27.59 | ChannelZ | https://www.youtube.com/watch?v=Rg0gOx5Iw5Y |
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| 07:35.56 | AliRezaTaleghani | i had face with such an error |
| 07:35.58 | AliRezaTaleghani | utils.c:1168 ast_carefulwrite: write() returned error Broken pipe |
| 07:36.05 | AliRezaTaleghani | after about 5K call |
| 07:36.08 | AliRezaTaleghani | :-| |
| 07:37.37 | schmidts | AliRezaTaleghani this could happens when you use AMI and didnt close the connection properbly for example |
| 07:37.45 | schmidts | there have been a lot of issues about this problem |
| 07:38.26 | AliRezaTaleghani | schmidts: tnx seem's smell |
| 07:38.35 | AliRezaTaleghani | I'll chk r-now |
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| 07:58.19 | AliRezaTaleghani | :-) how about this: |
| 07:58.24 | AliRezaTaleghani | channel.c:1485 __ast_queue_frame: Exceptionally long queue length queuing to SIP |
| 07:58.43 | AliRezaTaleghani | ofcourse i have too may packet drop on network interface! |
| 07:59.14 | AliRezaTaleghani | but i'm not sure that the packet drop course this error on asterisk console!? or |
| 07:59.41 | schmidts | AliRezaTaleghani which version do you use? |
| 07:59.48 | AliRezaTaleghani | or it's in revert! mean that asterisk drop some things out which then will droped out of network |
| 08:00.04 | AliRezaTaleghani | asterisk 1.8.16 compiled from source on CentOS 6.3 |
| 08:00.24 | AliRezaTaleghani | system load is about 1.2 in 5 min |
| 08:00.36 | *** join/#asterisk giany (~giany@shifu.x83.org) |
| 08:00.38 | AliRezaTaleghani | with 24 core and 32G mem |
| 08:00.43 | schmidts | no if asterisk would drop it which i dont think, then you wouldnt see it on the network side, only a missing reply or forward |
| 08:00.49 | schmidts | how many concurrent calls? |
| 08:01.11 | AliRezaTaleghani | 5k Conccurrnty |
| 08:01.22 | schmidts | ok no wonder |
| 08:01.37 | AliRezaTaleghani | 5000 I mean |
| 08:01.53 | schmidts | 1.8 could not easy handle that much, you should go for 10 which is better |
| 08:02.01 | AliRezaTaleghani | you mean it's normal to be dropped on network side? |
| 08:02.03 | schmidts | but btw do you only have one server or is this only part of a cluster |
| 08:02.30 | schmidts | dropps on your network side means the network buffer could handle the amount of packets |
| 08:02.37 | AliRezaTaleghani | to be honest i have 5 server |
| 08:02.41 | schmidts | s/could/couldnt/ |
| 08:02.57 | AliRezaTaleghani | 1 for generation of 5K SIP channel to be in Wait State |
| 08:03.12 | schmidts | 100mbit or gbit connection? |
| 08:03.17 | AliRezaTaleghani | 3 server which are behind an opensips proxy to handel media |
| 08:03.22 | giany | hi, i have a issue with Queue agents that for some reason are reporting as UNAVAILABLE. These agents are used by multiple Queues. Anyone know what could be the reason of this? |
| 08:03.31 | AliRezaTaleghani | and one other to terminate channels on MusicOnHold |
| 08:03.45 | AliRezaTaleghani | all are attached to SRX650 Gig interfacess |
| 08:03.48 | schmidts | ok |
| 08:04.04 | schmidts | so its not about normal traffic |
| 08:04.26 | schmidts | do you have problems with sip signaling or rtp ? |
| 08:04.40 | schmidts | could you pastebin the output of "sip show sched" |
| 08:05.07 | AliRezaTaleghani | schmidts: there was some issue about which was about Juniper SIP inspection feature |
| 08:05.15 | AliRezaTaleghani | which is disabled r-now |
| 08:06.20 | *** join/#asterisk jaxon007_ (~root@123.252.144.92) |
| 08:07.25 | AliRezaTaleghani | giany: did u check your agents side statuts? if you are faced with productive status! first check Client to not to exited there SIPclient. after that be sure about your LAN infrastructure! dose the problem happen in random? |
| 08:08.06 | AliRezaTaleghani | schmidts: I am going to pick the servers out of Juniper... |
| 08:08.14 | AliRezaTaleghani | maybe it's the reason |
| 08:08.34 | schmidts | juniper is a firewall right? |
| 08:08.39 | AliRezaTaleghani | yep |
| 08:09.06 | schmidts | but i still think that 5k calls on 1.8 is too much for it, you should really try 10 instead |
| 08:09.24 | schmidts | but an external firewall in front could be the problem |
| 08:09.41 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:10.11 | giany | AliRezaTaleghani: yes it happens randomly, agent status is like this : 1136*133 (Local/1136*133@queue/nj) with penalty 1 (Unavailable) |
| 08:10.53 | jaxon007_ | How can i increase the volume of a particular channel [not current channel] by using AMI + volume function..see my requirement is , i have one channel in conference on that an audio file is playing and i want to increase the volume of a that channel by using AMI ... |
| 08:12.57 | AliRezaTaleghani | schmidts: thanks, i will try both 10 and the new 11-Beta :-) |
| 08:14.12 | AliRezaTaleghani | giany: dose the clients use Internal VPN for internet access? |
| 08:14.59 | giany | AliRezaTaleghani: they are behind a NAT device |
| 08:16.09 | AliRezaTaleghani | giany: 1st check your network! suggest you to take the Asterisk behind NAT too! where the Clients are! |
| 08:16.09 | RiceCracker | does this mean no secret is set on my end or the other iax2 trunk end? "NOTICE[3011] chan_iax2.c: No secret associated with peer <trunkname>" |
| 08:16.25 | AliRezaTaleghani | giany: abit seem's to be network problem... |
| 08:17.10 | AliRezaTaleghani | jaxon007_: you should set TX on that specific channel |
| 08:17.16 | giany | AliRezaTaleghani: have you used queues until now? unavailable does not refer to the network status.. |
| 08:19.56 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 08:20.20 | jaxon007_ | <PROTECTED> |
| 08:23.08 | schmidts | AliRezaTaleghani i havent tested 11 so far so i dont know how good it is, but i have done 13k concurrent calls on 10 without any problems |
| 08:30.17 | RiceCracker | any help? does this mean no secret is set on my end or the other iax2 trunk end? "NOTICE[3011] chan_iax2.c: No secret associated with peer <trunkname>" |
| 08:31.06 | RiceCracker | because i did set one, and the trunk was working without issues, then about 2 weeks later just randomly decided it didnt want to connect anymore |
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| 08:38.19 | AliRezaTaleghani | schmidts: and with hony many servers? |
| 08:38.44 | AliRezaTaleghani | giany: a lot... |
| 08:39.05 | AliRezaTaleghani | giany: UnAv mean the client Session is not avail r-now |
| 08:39.15 | AliRezaTaleghani | so u see she that like in Q |
| 08:41.07 | AliRezaTaleghani | jaxon007_: use the AMI "Setvar" |
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| 08:46.24 | AliRezaTaleghani | schmidts: :-) mean how may servers? could you give me the spec of the Call Generator? plz |
| 08:46.25 | giany | AliRezaTaleghani: this does not help me at all |
| 08:46.26 | FluxiFlax2022 | hi all I am setting up a test environment, I have kamailio -> asterisk -> carrier ...this works fine ..now I am setting up a second environmnet with kamailio -> asterisk -> asteriskB where asteriskB is simulating a carrier, problem is asteriskB is requesting proxy authentication which is good but call is failing, although both Asterisks are using the same realtime db |
| 08:46.27 | AliRezaTaleghani | tnz |
| 08:48.19 | jaxon007_ | @AliRezaTaleghani: SetVariable will accept two parameter for AMI . one is variable name and another one is its value. So here what will b variable name and value |
| 08:50.07 | *** join/#asterisk smast (~chatzilla@seepzgw.mumbai.geodesic.com) |
| 08:50.10 | jaxon007_ | @AliRezaTaleghani@ I want to increase volume at any time while playing file when ever request sent by user.. |
| 08:53.06 | AliRezaTaleghani | jaxon007_: http://www.voip-info.org/wiki/view/Asterisk+func+volume |
| 08:53.08 | schmidts | AliRezaTaleghani 1 virtual server running on a HP DL380G7 with dual six cores and hyper threading, load was around 1 during the tests and call where generated with SIPP |
| 08:53.20 | AliRezaTaleghani | SetVar accept 3parameter |
| 08:53.38 | *** join/#asterisk Vince-0 (c4d7bcf4@gateway/web/freenode/ip.196.215.188.244) |
| 08:53.40 | AliRezaTaleghani | channel which is your MusicplaybackChan in Conf |
| 08:54.12 | AliRezaTaleghani | var name is TX and value can be increment! or reduced on the fly |
| 08:54.33 | AliRezaTaleghani | schmidts: perfect |
| 08:55.08 | jaxon007_ | @ AliRezaTaleghani: thank let me check |
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| 09:05.37 | jkroon | hi guys, whilst g723 can function in pass-through mode on asterisk - is there any module (licensed?) that will allow me to transcode to/from g723? |
| 09:09.36 | awk | hmm, I need to find gdb info |
| 09:10.43 | awk | what is the best way to gdb pid of asterisk to a > file with thread apply all |
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| 09:20.05 | jkroon | gdb /path/to/asterisk $(pidof asterisk) |
| 09:20.12 | jkroon | that will likely stop the running process though. |
| 09:23.42 | awk | jkroon then just > file.txt ? |
| 09:24.03 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
| 09:24.10 | sekil | hm |
| 09:24.17 | sekil | any blf problems on 1.8.17? |
| 09:24.21 | sekil | it doesn't seem to work here |
| 09:25.23 | jkroon | awk, various things to consider, you basically just want a trace of a running instance. In my experience it's always simpler to work with a core dump, so if you can get the running instance to dump core and continue (not sure how) then rather get the traces from the core file. |
| 09:25.30 | jkroon | that way it's a lot simpler |
| 09:26.43 | awk | gdb -ex "thread apply all bt" --batch /usr/sbin/asterisk `29912` > /tmp/asterisk-threads.txt |
| 09:26.46 | awk | that works |
| 09:27.02 | awk | im getting many auto destruct |
| 09:27.07 | awk | im loggin debug verbose etc to file |
| 09:27.24 | awk | and im going to try workout the code path that causes this autodestruct to reproduce |
| 09:30.35 | awk | no you right, keeps killing it |
| 09:30.36 | awk | grrr |
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| 09:51.41 | awk | hey anyone got a asterisk -rx "core show channels" on a 1.8 version over 90 to show me the output so I can prove something to my supplier chain... Also with a cat /var/log/messages | grep Autodestruct |
| 09:51.56 | awk | if no Autodestruct I can prove to them its the code path causing this |
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| 09:59.57 | hariom | What is the practical use of jbenable in sip.conf ? I have low bandwidth connection and I use IVR calls on it. Should I enable jbenable and force it? |
| 10:14.52 | bombev | any idea how to fix show queue XX, it shows all members as "INVALID", I tried with to add these two lines in modules.conf: preload => pbx_config.so , preload => chan_local.so and autoload=yes , but it didnt work.. |
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| 10:37.30 | hariom | I have to provide about 500 SIP accounts with different passwords to 500 users. How to configure sip.conf for that? All calls will come on IP channel |
| 10:38.10 | bulkorok | you can use templates |
| 10:38.19 | bulkorok | or realtime config in a database |
| 10:38.56 | bulkorok | https://wiki.asterisk.org/wiki/display/AST/Using+Templates |
| 10:41.27 | *** join/#asterisk kasanop (~kasanop@95-27-144-8.broadband.corbina.ru) |
| 10:45.11 | jaxon007_ | asterisk with meetme work well on amazon instances?.. |
| 10:45.43 | jaxon007_ | Ec2 |
| 10:51.35 | hariom | bulkorok: thanks. I have another query. Is it possible to use email id as username? eg: instead of [500](template) is it valid to write [abcd@example.com](template) ? |
| 10:54.04 | WIMPy | hariom: Using @ inside a username will get you in to trouble. |
| 10:54.28 | WIMPy | Probably anyway. |
| 11:10.23 | hariom | I had cli command: "sip set debug on" but now when I write this command * says no such command |
| 11:10.33 | hariom | Also sip reload doesn't seem to work |
| 11:10.44 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 11:13.41 | kaldemar | hariom: then you don't have the sip module loaded. "module load chan_sip.so" |
| 11:14.41 | hariom | kaldemar:module load chan_sip.so |
| 11:14.41 | hariom | Unable to load module chan_sip.so |
| 11:14.41 | hariom | Command 'module load chan_sip.so' failed. |
| 11:14.41 | hariom | [Oct 10 16:44:06] SIP channel loading... |
| 11:14.41 | hariom | [Oct 10 16:44:06] == Parsing '/etc/asterisk/sip.conf': [Oct 10 16:44:06] == Found |
| 11:15.34 | hariom | kaldemar: Looks like module is failing to load. Does that mean there is error in sip file? |
| 11:15.49 | WIMPy | probably |
| 11:16.06 | WIMPy | turn up verbose/debug and try again. Maybe it tells you what it doesn;t like. |
| 11:17.38 | hariom | WIMPy: I have verbose at level 5. asterisk -rvvvvvv |
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| 11:18.13 | WIMPy | That was one half of verbose and debug. |
| 11:26.13 | hariom | WIMpy: Not sure what is happening |
| 11:26.15 | hariom | module show like sip |
| 11:26.15 | hariom | Module Description Use Count |
| 11:26.15 | hariom | chan_sip.so Session Initiation Protocol (SIP) 0 |
| 11:26.15 | hariom | 1 modules loaded |
| 11:26.38 | hariom | WIMPy: but is neither load or unload chan_sip.so |
| 11:26.45 | hariom | Both times command failes |
| 11:30.31 | *** part/#asterisk TiJof (~TiJof@smtp.rabouin.es) |
| 11:30.33 | kaldemar | "core set debug 10" "module unload chan_sip.so" "module load chan_sip.so" |
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| 11:32.30 | bombev | any idea how to fix show queue XX, it shows all members as "INVALID", I tried with to add these two lines in modules.conf: preload => pbx_config.so , preload => chan_local.so and autoload=yes , but it didnt work.. |
| 11:34.18 | kaldemar | maybe your members are invalid. what does your queues.conf look like? |
| 11:34.25 | hariom | kaldemar: Same verbose output. No change |
| 11:35.18 | kaldemar | hariom: what version are you using? |
| 11:35.21 | hariom | kaldemar: Is it possible that chan_sip.so got corrupt? Any way to replace it with fresh copy (without again doing make install) |
| 11:35.37 | hariom | kaldemar: 1.8.16.0 |
| 11:36.54 | kaldemar | i's suspect a broken config first. you can always copy the compiled channels/chan_sip.so to /usr/lib/asterisk/modules/ if you've done something odd. |
| 11:37.59 | hariom | kaldemar: I have the minimalistic sip.conf . The same sip was working before |
| 11:38.09 | hariom | Will try with module replacement |
| 11:42.34 | hariom | kaldemar: Looks like it was problem with sip.conf only. I just removed the previous file and replaced working file from another system. It works now |
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| 12:05.49 | leifmadsen | hariom: probably too minimal. Did you enable callcounter=yes ? |
| 12:05.59 | *** part/#asterisk deo (~dnepangue@58.71.19.178) |
| 12:06.03 | hariom | NO |
| 12:06.08 | leifmadsen | YES |
| 12:13.46 | hariom | leifmadsen: didn't get you what you mean. I didn't enable callcounter to yes |
| 12:14.10 | leifmadsen | if you're using queues with sip members, then you should |
| 12:14.56 | hariom | Is it advisable to use SER before Asterisk if the number of SIP calls that will come are going to be large? With large I mean 10calls/min |
| 12:15.16 | seanbright | 10 calls/min is not large |
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| 12:15.32 | seanbright | 10 calls/sec would be |
| 12:15.38 | hariom | leifmadsen: queues with sip members I didn't get. I guess I should read about queues in this case |
| 12:15.59 | hariom | seanbright: So it doesn't make sense to use SER for 10calls/min? |
| 12:16.32 | seanbright | i don't think so, no. |
| 12:18.23 | leifmadsen | Asterisk can easily handle 10 calls per minute |
| 12:21.05 | beebeeep | asterisk can handle hundreds of concurrent calls |
| 12:21.11 | seanbright | millions |
| 12:21.14 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:21.17 | leifmadsen | right, but that's not what we're talking about :) |
| 12:21.19 | seanbright | 1s of 10s |
| 12:21.36 | seanbright | concurrent calls != calls per second |
| 12:21.40 | leifmadsen | that ^^ |
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| 12:28.34 | beebeeep | well, i've just checked my most loaded * in touch, today it handles 11 call at one second :) |
| 12:29.17 | leifmadsen | right, handle 11 calls concurrently is not the same as handling 11 call setups simultaneously |
| 12:29.37 | file | you gain no benefit and an extra point of failure |
| 12:31.37 | jaxon007_ | anyone have experience in installing asterisk with meetme on amazone virtual instances EC2 |
| 12:31.39 | jaxon007_ | ? |
| 12:32.18 | *** part/#asterisk giany (~giany@shifu.x83.org) |
| 12:32.33 | beebeeep | leifmadsen: it's not concurrent, it's just 11 Dial's at the same time at *'s log - http://pastebin.com/bRBZMbxA |
| 12:34.11 | [TK]D-Fender | jaxon007_, http://tinyurl.com/97p97cg |
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| 13:03.55 | mrhank | Hi! I have a few users logged into a queue with active calls to asterisk. I use call-files to auto-dial and when the call answers i put it in the queue. I want to be able to route specific calls to specific agents. The easiest way for me would be to create a call file and add info about which agent this call should go to if it picks up. This is probably not possible i can imagine, but is |
| 13:03.55 | mrhank | there a way to pause the agent, create a call file and just connect the call to the correct agent channel if the call answers? |
| 13:05.20 | leifmadsen | ummm..... there is, but you have to kind of build a state machine or some fancy dialplan that can determine the availability of an agent, pause them, place the call to them directly, then unpause them when the call is done |
| 13:05.50 | leifmadsen | I've done it, but it wasn't something you just enable in app_queue or anything. Had to build the dialplan to do some stuff to perform the checks, set the member correctly, etc |
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| 13:07.52 | mrhank | @leifmadsen: That is actually not a problem, as the agent will always be on pause and available when this happens. This is part of an automatic telemarketing system so the agent will pause, type the number and a call-file will be created based on that, and when the call is done they will have to unpause themselves.. how would i do the "bridging"-part? |
| 13:08.15 | leifmadsen | you just call the member |
| 13:08.24 | leifmadsen | via a direct channel outside of queue |
| 13:08.33 | leifmadsen | beyond that, I don't want to give any more help towards telemarketing |
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| 13:11.07 | mrhank | @leifmadsen: Hehe, i can see why! I want the call to connect to the currently active channel between the agent and asterisk, is that possible? |
| 13:11.14 | leifmadsen | yes. |
| 13:12.05 | [TK]D-Fender | mkJust DIAL them |
| 13:12.10 | [TK]D-Fender | mrhank, ^ |
| 13:12.35 | robl^ | same => Zapateller() ;-) |
| 13:13.24 | leifmadsen | I'd be surprised if that actually does anything anymore... |
| 13:14.30 | robl^ | still plays SIT at least.. but yeah, many outbound autodialers ignore it |
| 13:14.44 | mrhank | [TK]D-Fender: Just dial the Agent? Like Dial("Agent/agentname")? |
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| 13:15.04 | [TK]D-Fender | mrhank, Like whatever you're using for your queue members. |
| 13:15.30 | mrhank | Ok, thanks! Ill try it! |
| 13:20.16 | awk | hmm, i'm having a huge issue over 70 calls I get Autodestruct issues in 1.8 and in 1.4 I get calls stuck in a all-hangup.. I am having such issue re-producing this problem, however I have 2 systems both of them over 60+ calls and this happens. How would I workout the code path for this? |
| 13:20.33 | awk | I can't put a ticket in unless I can re-create the problem. I have tried everything to rid this issue |
| 13:21.20 | awk | Either its an issue where its fighting over SQL (writing) the call or a timout set or maybe just a+b+c = autodestruct or this is an I/O issue and happens when load on the server researches X calls |
| 13:21.57 | awk | anyonee here running some high load system I need the oputput of call show channels |
| 13:22.02 | awk | just the (number) at the bottom |
| 13:22.19 | awk | not the actual calls, just need to prove to this supplier of mine that asterisk can easily h andle more than 70 calls |
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| 13:25.12 | mirela666 | awk: When I was testing my 1.8 installation with sipp it easelly handled 300-400 simultanious calls, but didn't use odbc, just mysql writh for them |
| 13:26.15 | mirela666 | awk: you can try with sipp if you can stop one server |
| 13:32.18 | sekil | hm |
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| 13:32.38 | sekil | does anyone know why would a notify not be issued on a watched phone? |
| 13:32.40 | kaldemar | awk: 9840 active calls |
| 13:32.48 | sekil | I'm using 1.8.15 to 1.8.17 |
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| 14:12.42 | bombev | Any idea which Outbound CID has highest priority 1 Extensions, 2 Outbound routes , 3 Trunks |
| 14:14.28 | kaldemar | bombev: what GUI are you using? |
| 14:17.23 | *** join/#asterisk v4x (~v4x@unaffiliated/v4x) |
| 14:18.05 | [TK]D-Fender | FREEPBX ... like where he should leave that question. |
| 14:22.41 | *** join/#asterisk jaxon007_ (~jaxon007_@14.97.44.136) |
| 14:25.30 | awk | who here will do paid support to find an identiy my issues |
| 14:25.42 | awk | over 70 calls im getting auto destruct on 1.8 |
| 14:26.00 | awk | on 1.4 i get calls stuck in all-hangup |
| 14:26.21 | awk | and then I run out of g729 licenses, calls climb to eg: 150 then drop to 60 |
| 14:27.04 | awk | who can offer paid support to find where this fault lies so we can patch and fix this.. it is effecting many of our systems now |
| 14:29.25 | *** join/#asterisk elico (~Thunderbi@109.64.221.76) |
| 14:31.49 | awk | wow, thought people would jump at the chance to earn to identify the issue |
| 14:32.36 | Qwell | It's 1.4. |
| 14:32.41 | Qwell | ~upgrade asterisk |
| 14:32.41 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 14:32.43 | awk | 1.8 |
| 14:32.59 | awk | 1.8.17 or 16 I get auto destrudct |
| 14:33.05 | awk | once I reach 65+ calls |
| 14:33.15 | awk | we have built our software if someone can fiund the source of the problem |
| 14:33.19 | awk | you get paid |
| 14:33.19 | awk | we fix our software |
| 14:33.26 | awk | its probably just the code path. |
| 14:33.58 | Qwell | It would probably help to actually specify what's happening, what you're doing, etc. |
| 14:33.59 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 14:35.33 | awk | Qwell https://issues.asterisk.org/jira/browse/ASTERISK-20416 |
| 14:35.37 | awk | I have a ticket open with it |
| 14:35.55 | awk | And we msut update to state it isn't related to g729 as we have tested g711 |
| 14:36.12 | awk | this problem has gone on to long its becoming critical i'd like to pay someone to identify the fault |
| 14:42.20 | *** join/#asterisk bitwize (~bitwize@h87-96-213-2.dynamic.se.alltele.net) |
| 14:44.01 | bitwize | Hi guys, anybody know if it's possible to ask Asterisk through AMI if a specific channel exists? (ex. channel exists SIP/blablabla) |
| 14:45.45 | *** join/#asterisk paz (~paz@fwj00.e-fon.ch) |
| 14:45.48 | paz | hey :) |
| 14:45.50 | *** join/#asterisk kayatwork (~kayfox@orca.zerda.net) |
| 14:45.51 | bitwize | I can solve this issue by request a list of all active channels (core show channels) and then iterate the list of channels but I want a cleaner solution... |
| 14:48.05 | *** part/#asterisk keycruncher (~Adium@c-174-59-233-118.hsd1.pa.comcast.net) |
| 14:48.23 | [TK]D-Fender | bitwize, Ask for a status dump on that specific channel and watch an ERROR come back |
| 14:48.41 | paz | what is the problem, when i restart asterisk, i cant unpause a qeueue-member bevor i deal out to a number. after i call a number i can unpause and pause... the errormessage is: "[Oct 10 16:47:40] WARNING[15293]: app_queue.c:4626 upqm_exec: Attempt to unpause interface SIP/6230a9007, not found" why asterisk can't find interface in realtime before call out? (sorry, man text ^^) |
| 14:49.10 | paz | man/many |
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| 14:50.30 | bitwize | [TK]D-Fender: Thanks, which cli-command can I use to get the status dump? |
| 14:50.53 | [TK]D-Fender | bitwize, Have you looked through the entire AMI command list? |
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| 14:55.20 | bitwize | [TK]D-Fender: Actually I did not, sorry.. I have now found the command, thank you! |
| 14:55.33 | [TK]D-Fender | \o/ |
| 14:55.38 | bitwize | ;) |
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| 14:59.04 | Dovid | hi oej |
| 14:59.59 | oej | Hi Dovid! |
| 15:00.17 | bombev | kaldemar what do you mean what GUI i am using |
| 15:02.12 | [TK]D-Fender | bombev, You are asking a FREEPBX question, which is not supported here |
| 15:02.25 | bombev | sorry |
| 15:04.48 | paz | my train <o> see you tomorrow! |
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| 15:37.20 | jaxon007_ | in my conferece there is one leg which is plying an audio file , i want to increase volume when send api request to web server . in web server i use AMI setVar to set the TX variable to increase the Volume .but it is not increasing what i missed ... |
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| 15:42.11 | [TK]D-Fender | jaxon007_, Maybe having gotten it backwards.... |
| 15:42.21 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 15:42.27 | [TK]D-Fender | jaxon007_, Or having done it right |
| 15:45.38 | jaxon007_ | $asm->send_request('SetVar') |
| 15:47.51 | jaxon007_ | @fender: array('Channel' => 'local/s@meetme','Variable'=>'TX','Value'=>'3') i set this in AMI |
| 15:51.05 | [TK]D-Fender | jaxon007_, Who said that variable meant anything? |
| 15:52.15 | jaxon007_ | @fender : i got this info from IRC, its not correct? |
| 15:52.19 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 15:53.00 | [TK]D-Fender | "from IRC"? WHO? When? |
| 15:53.11 | [TK]D-Fender | facepalms |
| 15:55.39 | Katty | you. |
| 15:55.41 | Katty | now. |
| 15:55.44 | *** join/#asterisk whtsup (~whtsup@180.92.141.172) |
| 15:55.51 | Katty | are you being nice today fender? |
| 15:56.17 | [TK]D-Fender | Katty, Does that maen chewing before swallowing after I bite their head off? ;) |
| 15:56.29 | Katty | no that means put the nice man down. |
| 15:57.35 | [TK]D-Fender | Well I'm decapitation free today so far .... |
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| 15:58.08 | jaxon007_ | Fender@: today indian time around 1 0 clock .. |
| 15:58.57 | [TK]D-Fender | jaxon007_, who? |
| 15:59.37 | jaxon007_ | @fender: i dont remember name begins with a... |
| 16:00.11 | [TK]D-Fender | jaxon007_, ibot's logs show nothing of the sort. |
| 16:01.32 | jaxon007_ | @fender: actualli i asked those from my office machine now that i am home...otherwise i can give name.. |
| 16:05.04 | jaxon007_ | @fender: i had detailed asked then they told TX is variable number is Value... |
| 16:05.55 | whtsup | how to install dahdi on grsecurity kernel plz help me |
| 16:06.28 | zamba | i have a question about the nat setting.. i have a separate voip network where most of my sip phones are connected and the asterisk has a leg in this network as well.. should all these peers be configured with nat=no? |
| 16:07.22 | *** join/#asterisk Vince-0 (~Vincent@41-133-16-59.dsl.mweb.co.za) |
| 16:09.52 | zamba | because they are directly connected in the same network as the asterisk box? |
| 16:10.27 | [TK]D-Fender | jaxon007_, It isn't, and I've wasted my time telling you what you needed to do for this REPEATEDLY. |
| 16:11.08 | [TK]D-Fender | zamba, Correct |
| 16:12.14 | zamba | the same goes with peers that are on public addresses and can be directly routed to, right? |
| 16:12.45 | [TK]D-Fender | zamba, Wouldn't trust those. I'd leave them as "yes" |
| 16:13.29 | zamba | [TK]D-Fender: but i KNOW they are 1:1 connected |
| 16:13.37 | jaxon007_ | @fender: sorry |
| 16:13.38 | zamba | they're my devices |
| 16:13.43 | zamba | just located somewhere else |
| 16:15.26 | [TK]D-Fender | zamba, Still no point |
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| 16:43.28 | *** join/#asterisk rbowles (~bb@75-147-63-201-NewEngland.hfc.comcastbusiness.net) |
| 16:51.13 | *** join/#asterisk af_ (~getsmart@78.134.61.252) |
| 16:55.42 | *** join/#asterisk elico (~Thunderbi@109.64.221.76) |
| 16:57.25 | *** join/#asterisk Galen (~Galen@rrcs-24-43-17-237.west.biz.rr.com) |
| 16:58.04 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 17:02.18 | jaxon007_ | @fender: i am veru confused , can u please expalain once more for increasing volume |
| 17:02.32 | Qwell | jaxon007_: core show function VOLUME |
| 17:03.37 | Katty | very cranky. |
| 17:03.52 | Qwell | turns the crank the other direction |
| 17:04.40 | Katty | breathes. |
| 17:06.17 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CMb89bc9d2e1a5.cpe.net.cable.rogers.com) |
| 17:07.55 | jaxon007_ | @Qwell: hw can use volume function + AMI for increasing the volume of a specified channel.. |
| 17:10.16 | Qwell | ~book |
| 17:10.16 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 17:13.21 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.111) |
| 17:19.49 | jaxon007_ | @Qwell: in my conferece there is one leg which is plying an audio file , i want to increase volume when send api request to web server . how can i use volume function with AMI |
| 17:20.25 | Qwell | Forget the conference. You care about the channel. |
| 17:21.04 | jaxon007_ | @qwell: but how its possible... |
| 17:21.23 | Qwell | By reading documentation that exists for such things. |
| 17:21.36 | [TK]D-Fender | SetVar + VOLUME.... like I've told you FOR DAYS |
| 17:24.22 | Katty | throws popcorn at Qwell |
| 17:24.29 | Katty | Qwell: have we said hi today? |
| 17:25.22 | jaxon007_ | @SetVar require three parameter 1) channel name , 2) Variable name and 3) Variable value.. |
| 17:25.35 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:25.36 | jaxon007_ | What will b variable name? |
| 17:26.08 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 17:26.22 | *** join/#asterisk l2cached (~gregg@c-50-129-68-241.hsd1.il.comcast.net) |
| 17:26.27 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:27.17 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 17:27.41 | n3hxs | wonders why there is popcorn on the floor. |
| 17:27.48 | ChannelZ | whatev U want it 2 B |
| 17:27.51 | l2cached | Hi, I have a problem with an allworx system trunked to my asterisk system. The allworx phones are redirected to register to asterisk, and their is a trunk registered from allworx to the asterisk. In sip show peers all of the extensions and the trunk share the same IP. When a call comes in the trunk to asterisk, it matches against one of the extensions (same ip). Is there any way to give priority instead of going by IP first, then us |
| 17:27.51 | l2cached | ername? |
| 17:28.55 | [TK]D-Fender | l2cached, type=friend |
| 17:29.51 | l2cached | That's how it is setup. I've tried peer as well. Almost every setting I can think of. But it seems to go by the order of the bottom list in the sip.conf file upwards. |
| 17:30.20 | jaxon007_ | @fender: SetVar require three parameter 1) channel name , 2) Variable name and 3) Variable value.. What will b variable name? |
| 17:31.02 | *** join/#asterisk donnib (~donnib@2.105.191.170) |
| 17:32.44 | ChannelZ | jaxon007_: The VOLUME function presumably |
| 17:32.54 | *** join/#asterisk keith4 (~keith@unaffiliated/keith4) |
| 17:33.14 | ChannelZ | VOLUME(TX) or whatev |
| 17:35.28 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 17:36.02 | *** join/#asterisk donnib_ (~donnib@2.105.191.170) |
| 17:38.10 | *** join/#asterisk qlok (~jereminsh@210.186.243.127) |
| 17:40.00 | *** join/#asterisk fukuda_ (~IceChat9@86.70.181.183) |
| 17:40.06 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 17:40.18 | fukuda_ | hi ! |
| 17:40.38 | fukuda_ | i'm a problem |
| 17:40.53 | [TK]D-Fender | 11 steps to go. |
| 17:42.11 | fukuda_ | i'm installed iaxmodem and asterisk |
| 17:42.23 | *** join/#asterisk keycruncher (~Adium@c-174-59-233-118.hsd1.pa.comcast.net) |
| 17:43.24 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 17:43.28 | Katty | i am the problem. |
| 17:43.33 | Katty | and i'm ok with this. |
| 17:43.36 | Katty | pesters [TK]D-Fender |
| 17:44.49 | fukuda_ | but, on asterisk the port of my iaxmodem is not stable.It alternates between a random port and port inscript in the configuration file. |
| 17:46.16 | l2cached | any ideas? |
| 17:46.19 | Katty | qlok: please stop sending me private messages. if you need help with asterisk, please ask in the channel where they are many experts to help. thanks. |
| 17:46.37 | l2cached | Why asterisk can't separate out calls to auth, say if you had 3 trunks registered to the same IP |
| 17:46.38 | fukuda_ | in the process, I see two that are iaxmodem execution |
| 17:47.06 | [TK]D-Fender | l2cached, Perhaps you should actually SHOW us what you've done and the debug of a call you think should have processed differently... |
| 17:48.01 | qlok | oh ok Katty |
| 17:48.11 | l2cached | I was hoping for a revelation. Because google shows many many people with similar issues. |
| 17:49.23 | [TK]D-Fender | l2cached, We don't know how many of those issues you are responsible for creating, and failure to look at what you're doing is a waste of everyone's time. |
| 17:49.32 | qlok | first of all, hi |
| 17:49.41 | l2cached | 0. :) |
| 17:49.44 | l2cached | let me put something together |
| 17:50.14 | qlok | i'm new to asterisk. i need help |
| 17:50.35 | drmessano | ~book |
| 17:50.35 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 17:51.36 | qlok | i've gone through that book but theres no annswer to my question |
| 17:52.12 | Katty | hi danny |
| 17:52.18 | qlok | i'm thinking of integrating a gsm sim bank with asterisk. is it possible? |
| 17:52.43 | [TK]D-Fender | qlok, I've seen some |
| 17:53.39 | qlok | how is it possible |
| 17:54.05 | [TK]D-Fender | How is it NOT possible? |
| 17:54.15 | [TK]D-Fender | Get a device that * can speak to that you plug SIM's into. |
| 17:54.21 | [TK]D-Fender | Done. |
| 17:55.22 | *** part/#asterisk aruntomar (~Thunderbi@49.248.154.166) |
| 17:55.44 | qlok | do we need a specific channel driver for that to happen? |
| 17:56.11 | [TK]D-Fender | Of course. You need the channel drive based on the kind of DEVICE you are going to talk to. |
| 17:57.09 | qlok | Is it provided by the manufacturer. for example portech sbk-32 |
| 17:59.29 | qlok | hello..anyone? |
| 17:59.55 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 18:03.26 | *** join/#asterisk blee (~blee@72.188.117.219) |
| 18:04.53 | drmessano | qlok: What sort of interface does it have? |
| 18:05.07 | qlok | lan |
| 18:05.21 | drmessano | You need to be more specific.. |
| 18:06.06 | drmessano | What protocols? How does the manufacturer expect you to interface it with other equipment.. ? and don't say with an ethernet switch |
| 18:06.49 | [TK]D-Fender | qlok, And whoat does that manufacturer say about what it talks out the other side? |
| 18:07.10 | [TK]D-Fender | qlok, And if you didn't get the message already STOP PRIVATE MESSAGING PEOPLE DIRECTLY |
| 18:07.14 | keith4 | drmessano: 802.3 ;-) |
| 18:07.28 | qlok | wait |
| 18:08.14 | drmessano | The SBK-32 is a SIM BANK.. That box doesn't appear to handle any sort of transport. |
| 18:08.46 | qlok | it doesnt |
| 18:09.02 | qlok | what i understand it doesnt trancode |
| 18:09.40 | drmessano | Again, that box has nothing to do with Asterisk |
| 18:10.05 | qlok | i just want to pass an already transcoded call to gsm to an available gsm card |
| 18:10.33 | drmessano | They do appear to have companion GSM gateways, which are what is actually used to make calls. Those speak SIP, and it looks like they mention Asterisk specifically |
| 18:10.42 | qlok | to eliminate the use of voip-gsm gateway |
| 18:10.48 | [TK]D-Fender | qlok, Go get a device that Asterisk can TALK to. |
| 18:10.56 | drmessano | qlok: That device you mentioned is not a communication device |
| 18:11.04 | drmessano | It doesn't not handle calls |
| 18:11.08 | [TK]D-Fender | qlok, Because that is meant to speak only to THIER special devices. |
| 18:13.12 | qlok | i'm actually looking for an alternative way to direct a call to gsm rather than using multiple phones connected with bluetooth |
| 18:13.24 | drmessano | Great, you need a GSM GATEWAY |
| 18:13.40 | drmessano | http://www.portech.com.tw/p3-product1_1.asp?Pid=13 <-- They sell a few models |
| 18:14.25 | drmessano | There's dozens of GSM Gateways on the market that speak SIP. Pick one |
| 18:14.27 | qlok | yes that the one i'm looking for |
| 18:15.11 | qlok | thanks |
| 18:15.40 | drmessano | I still don't understand what all the transcoding business was about. |
| 18:15.49 | jmetro | Hm.. why would you want to transfer SIP to GSM. Just out of curiosity and lurking. |
| 18:16.25 | qlok | i was looking at the wrong device |
| 18:16.31 | drmessano | Sometimes a GSM gateway is your only way out |
| 18:16.36 | drmessano | Mobile, for example |
| 18:16.54 | drmessano | You may also want to use it for SMS |
| 18:17.27 | qlok | thats what i wanna do |
| 18:19.22 | drmessano | We had some GSM channel banks with analog ports on them in our mobile operations center. I scrapped them and moved to SIP over Verizon 4G broadband or satellite, and it's worked really well |
| 18:19.31 | drmessano | More channels, less BS.. lower cost |
| 18:20.23 | qlok | are there softphones which support gsm codec |
| 18:21.22 | qlok | does sip support gsm |
| 18:21.38 | qlok | just to make sure |
| 18:21.43 | l2cached | [TK]D-Fender, sorry about that. Here is a more detailed description: http://pastebin.com/3N8ggm5i |
| 18:22.33 | [TK]D-Fender | qlok, ASTERISK supports GSM 6.10. |
| 18:23.29 | [TK]D-Fender | l2cached, I see part of ONE config, and incomplete debug from a call. You've got one last chance at this.... |
| 18:23.41 | l2cached | ... |
| 18:23.58 | l2cached | So friendly...chillax bro |
| 18:24.12 | qlok | say for an example someone call from a softphone to an asterisk server using gsm and then pass the call to the gsm gateway |
| 18:24.36 | qlok | is it what asterisk do |
| 18:24.43 | rbowles | off-topic: Current favorite SIP clients for Linux workstations? (I get the BEST PCMU audio results from telepathy/sofiasip but it is a really poor UI) |
| 18:26.06 | artyx | whats wrong with linphone? |
| 18:27.35 | rbowles | actually linphone has been good, but crashes with really long URIs :-) |
| 18:27.45 | drmessano | qlok: The codec is irrelevant here |
| 18:27.59 | artyx | Well, there is that.. but you didnt say SIP client that supports REALLY long uri's ;) |
| 18:28.24 | artyx | uses blackvoib on his phone |
| 18:28.40 | artyx | hardly a linux workstation i know.. but it gets the job done |
| 18:28.52 | qlok | why is that. let say i have an E1 line that can trasport more than 140 gsm call simultaneously right? |
| 18:28.56 | drmessano | qlok: Asterisk is going to communicate with your GSM gateway over SIP, likely using G711 by default. You don't need to have asterisk use the GSM codec in order to make calls that will pass over a GSM gateway |
| 18:29.17 | Qwell | GSM gateways wouldn't even use GSM. |
| 18:29.18 | rbowles | heh :-), yeah I had testing linphone for dialing without registration to a bunch of places and was happy with it |
| 18:29.40 | Qwell | GSM protocol != GSM codec |
| 18:29.45 | drmessano | ^^^^^ |
| 18:30.20 | qlok | sip protocol with gsm codec. is that make sense or it just a stupid question |
| 18:30.31 | drmessano | Read what I said |
| 18:30.34 | drmessano | qlok: Maybe you're confused here. You don't shoot calls at asterisk using GSM and they just pipe out a GSM gateway like water. |
| 18:30.45 | [TK]D-Fender | qlok, E1 DATA running SIP for those? No chance |
| 18:31.28 | qlok | i am confused drmessano |
| 18:31.32 | drmessano | Seems so |
| 18:31.38 | qlok | can u explain |
| 18:32.50 | *** join/#asterisk AviMarcus (~avi@bzq-79-176-236-51.red.bezeqint.net) |
| 18:32.55 | AviMarcus | Hello. Is anyone here in the calling card biz? |
| 18:33.02 | drmessano | Your GSM Gateway is a box that speaks to the GSM NETWORK <> and to another device using SIP over ethernet. The codec used between the PHONE and ASTERISK can be whatever you want.. Its unimportant. The codec between ASTERISK and the GSM GATEWAY can be whatever you want, its unimportant. |
| 18:34.10 | drmessano | qlok: There is NO CORRELATION between the codecs used on the SIP transport and what is spoken over the GSM Connection. You dont control that. |
| 18:34.32 | drmessano | qlok: Chances are you are going to be using G711 by default. No reason to change it |
| 18:36.48 | drmessano | http://www.gsmgatewaysdirect.co.uk/how-does-a-gsm-gateway-work/ |
| 18:38.06 | qlok | does it mean every calls made using a softphone to asterisk are all using G.711 |
| 18:38.32 | jmetro | depending on the softphones configuration |
| 18:38.34 | drmessano | It can be.. Doesnt matter |
| 18:39.50 | qlok | so if i have an E1 line to accept incoming calls from softphones, the line can handle 147 call simultaneously, correct? |
| 18:40.03 | drmessano | No |
| 18:40.23 | qlok | if gsm onlt takes 13kb |
| 18:40.43 | drmessano | You're talking about a DATA connection, right? |
| 18:40.52 | qlok | yes |
| 18:40.56 | drmessano | E1 is overgeneral |
| 18:41.21 | drmessano | You can't just say E1 without providing some additional info. An E1 PRI is going to handle 31 calls. There is a difference |
| 18:42.03 | qlok | E1 pri that can handle 30 calls using G711 64kb |
| 18:42.30 | qlok | that means 147 calls gsm 13kb |
| 18:42.37 | drmessano | Negative |
| 18:42.43 | qlok | ??? |
| 18:42.56 | jmetro | not to mention a 13kb phone call is going to sound like garbage |
| 18:43.15 | qlok | with echo cancellation software installed too? |
| 18:43.22 | drmessano | 31 channels on a PRI circuit |
| 18:43.47 | n3hxs | It is not a data circuit but a 31 channel circuit. |
| 18:43.51 | drmessano | or is it 30? I forget if E1 is 32-1 or 31-1 |
| 18:43.59 | qlok | its 30 |
| 18:44.02 | n3hxs | or 30 |
| 18:44.05 | drmessano | Ok, so you can pass 30 calls |
| 18:44.45 | qlok | with G711 and 147 gsm, right> |
| 18:44.58 | drmessano | ..... |
| 18:45.00 | drmessano | NO! |
| 18:45.04 | jmetro | channel dependent, not size dependent. |
| 18:45.13 | drmessano | You have 30 CHANNELS |
| 18:45.15 | drmessano | 30 CALLS |
| 18:45.22 | drmessano | This isn't SIP |
| 18:45.38 | drmessano | It isn't about BANDWIDTH.. It's about using a CHANNELIZED CIRCUIT |
| 18:46.11 | qlok | only for u-law a-law right? |
| 18:46.21 | qlok | what about sip |
| 18:46.43 | [TK]D-Fender | PRI != SIP |
| 18:46.44 | drmessano | Do you know what any of these terms mean, or are you just throwing shit out there? |
| 18:46.52 | carrar | 1,000,000,000,000,000 SIP CALLS |
| 18:46.53 | n3hxs | channelized circuit. no sip can run on it. |
| 18:46.56 | [TK]D-Fender | Yes, he is just throwing shit around |
| 18:46.57 | drmessano | PRI is a fucking channelized circuit. |
| 18:47.31 | drmessano | You do not pass SIP over it, you do not choose your codec in some config file, you do not pass go, you do not collect $200 |
| 18:47.48 | file | I collect $200, as I withdrew that much from Paypal earlier this week. |
| 18:47.55 | file | drmessano, so HA! |
| 18:48.09 | drmessano | :( |
| 18:48.16 | drmessano | I have $1 in my paypal account |
| 18:48.18 | drmessano | I guess you win |
| 18:48.46 | qlok | thats why i wanted to talk in private |
| 18:48.53 | drmessano | We dont talk in private |
| 18:48.59 | drmessano | If you want to talk, you talk here |
| 18:49.13 | carrar | private talk makes me HOT++ |
| 18:49.17 | drmessano | Private is for sexting and links to kittens with captions |
| 18:49.21 | drmessano | Channel is for talking |
| 18:49.32 | drmessano | Seems like you dont get the CHANNEL concept in any context |
| 18:50.08 | carrar | qlok, a great book is T1 & Sonets |
| 18:50.19 | drmessano | I wrote a Sonnet |
| 18:51.23 | carrar | err: Sonet and T1 |
| 18:51.34 | carrar | looks like they are in their second edition now |
| 18:51.50 | rbowles | i have the hardcover, I'll sell it for $10 plus shipping |
| 18:52.01 | carrar | any version will help you |
| 18:52.30 | drmessano | This may help too --> http://web.mit.edu/adorai/www/seuss-technical-writing.html |
| 18:52.33 | drmessano | I learned a lot from it |
| 18:52.48 | qlok | hmm |
| 18:52.50 | rbowles | 10 year ago I had 14 DS3s of PRIs, today I have none :-) |
| 18:53.51 | *** join/#asterisk keycruncher (~Adium@c-174-59-233-118.hsd1.pa.comcast.net) |
| 18:53.59 | drmessano | I had 2 PRI's at the office for 15 days. $1400 to install them, 2 hours of install time, and I never turned them up |
| 18:54.14 | *** join/#asterisk salviadud (~dude@187.192.112.101) |
| 18:54.34 | n3hxs | qlok, your E1 PRI is the same as 30 copper pairs from the telco to your office.... |
| 18:54.36 | *** join/#asterisk nanoha-sama (~nanoha-sa@nanoha-sama.freenode.bouncers.smb.curriegrad2004.ca) |
| 18:54.39 | drmessano | PRI sucks, AT&T makes it worse |
| 18:54.45 | qlok | you mean pri only connects to the pstn not the internet |
| 18:54.53 | n3hxs | right |
| 18:55.03 | cmendes0101 | drmessano: haha that sounded way to close to that poem you posted |
| 18:55.06 | n3hxs | or what ever the other end is connected to. |
| 18:56.53 | qlok | is that mean we choose whether to connect to the pstn or to the internet |
| 18:56.56 | *** join/#asterisk navaismo (~navaismo@189.191.10.146) |
| 18:58.04 | *** join/#asterisk Nemus (~Nemus@c-67-161-251-38.hsd1.ut.comcast.net) |
| 18:58.16 | *** part/#asterisk keycruncher (~Adium@c-174-59-233-118.hsd1.pa.comcast.net) |
| 18:58.35 | drmessano | It's not an ethernet interface |
| 18:58.37 | *** join/#asterisk droemel (~droemel@p4FCADC39.dip.t-dialin.net) |
| 18:59.05 | qlok | i thought i could have an isdn line to receive incoming calls over internet |
| 18:59.23 | drmessano | ... |
| 18:59.41 | drmessano | A PRI is not ethernet.. It's not IP |
| 18:59.49 | drmessano | It does not connect to the internet |
| 19:00.14 | qlok | You want high speed and error-free quality, and you want it for all aspects of your communication needs Voice, Fax, Data, Video, Internet -- all simultaneously? |
| 19:00.15 | drmessano | It a specialized interface providing 30 channels of voice using a specialized PRI interface on your equipment |
| 19:00.15 | qlok | Then ISDN is the solution for you. |
| 19:00.15 | qlok | ISDN (Integrated Services Digital Network) is a system of digital phone connections designed for sending voice, video, and data simultaneously over digital or ordinary phone lines, with a much faster speed and higher quality than an analog system can provide. |
| 19:00.15 | qlok | With speeds of up to 2 Mbps, it transmits information four times faster than a normal phone line. That means no slowdown, no waiting and, most of all, higher efficiency. |
| 19:00.36 | qlok | You want high speed and error-free quality, and you want it for all aspects of your communication needs Voice, Fax, Data, Video, Internet -- all simultaneously? |
| 19:00.36 | qlok | Then ISDN is the solution for you. |
| 19:00.36 | qlok | ISDN (Integrated Services Digital Network) is a system of digital phone connections designed for sending voice, video, and data simultaneously over digital or ordinary phone lines, with a much faster speed and higher quality than an analog system can provide. |
| 19:00.36 | qlok | With speeds of up to 2 Mbps, it transmits information four times faster than a normal phone line. That means no slowdown, no waiting and, most of all, higher efficiency. |
| 19:00.38 | drmessano | This is turning into networking 101 |
| 19:00.41 | drmessano | STOP PASTING |
| 19:00.42 | [TK]D-Fender | You can channelize DATA over the E1 carrier circuit. |
| 19:00.44 | n3hxs | Your ISDN line may have an Internet connection, but it is not an E1 |
| 19:01.08 | qlok | i just want to show a promotion from an isdn provider |
| 19:01.16 | carrar | Go read that book |
| 19:01.18 | carrar | then come back |
| 19:01.28 | drmessano | WE KNOW WHAT ISDN IS.. YOU are the one that DOESNT |
| 19:01.28 | n3hxs | And An E1 is not Internet, data or network, but a digital phone line which is not SIP. |
| 19:01.30 | [TK]D-Fender | qlok, Before you go throwing around even vague concepts, go find out what your TELCO even offers you. You don't even understand the technology yet and we could be explaining 50 things they DON'T offer you. |
| 19:01.39 | [TK]D-Fender | qlok, So go find out so we don't waste our time. |
| 19:02.04 | jmetro | d-fender always seems to be on the o-fense |
| 19:02.36 | qlok | i dont mean to waste your time |
| 19:03.13 | qlok | in fact i really thank you guys for answering my stupid questions |
| 19:03.37 | qlok | :) |
| 19:04.18 | drmessano | Someone licked all the cheddar off my rice cakes before they were packaged |
| 19:04.28 | n3hxs | And An E1 is not Internet,data or network, but a digital phone line which is not SIP. Your telco might give you an ISDN line with Internet. |
| 19:04.58 | qlok | TK u said we can channelize data over e1 |
| 19:06.16 | Katty | hello my asterisk dose not work at all how to fix??? answer plz it is urgent thx. |
| 19:06.33 | [TK]D-Fender | qlok, No I said it IS possible. Maybe not for YOU. |
| 19:06.38 | carrar | install msft bob katty |
| 19:06.55 | [TK]D-Fender | qlok, This is the part where you stop and then ask your TELCO what services they can actually offer you. |
| 19:07.08 | Katty | i see fender is being cranky, as usual. |
| 19:07.17 | qlok | like i just paste |
| 19:07.40 | [TK]D-Fender | Katty, I haven't started writing in >50% all-caps or swearing yet..... |
| 19:08.00 | [TK]D-Fender | qlok, Less theory, more actual results.... |
| 19:08.01 | carrar | caps day is next month |
| 19:08.06 | Katty | let's hope you don't start. either. |
| 19:08.06 | [TK]D-Fender | \o/ |
| 19:08.11 | carrar | Oh sorry, 2 weeks |
| 19:08.15 | Katty | instead, let's hope that it's all sunshine and rainbows today. |
| 19:08.16 | carrar | Monday the 22nd |
| 19:08.17 | n3hxs | Katty, wax the string and use double knots in each can. Much better audio. |
| 19:08.32 | Katty | n3hxs: FIRST!time i've heard that. |
| 19:08.33 | Katty | applauds |
| 19:08.40 | drmessano | qlok: http://www.mainlinecom.com/t1vspri.html <-- Read all of that, NOW.. BEFORE you continue on |
| 19:08.52 | Katty | don't tell me danny is going to get cranky now. |
| 19:09.01 | drmessano | this conversation is becoming ridiculously circular |
| 19:09.04 | Katty | i hate it when people get cranky in here. |
| 19:09.22 | Katty | drmessano: look! beer! OVER THERE!!! |
| 19:09.26 | Katty | shoves drmessano out of room ehhehe |
| 19:09.26 | qlok | so what i learned isdn has two types: e1 to connect to pstn, the other one to the internet |
| 19:09.37 | n3hxs | Waxed string keeps the string dry between your bedroom and his. |
| 19:09.57 | Katty | asuming we have seperate bedrooms. |
| 19:10.02 | drmessano | ;) |
| 19:10.12 | n3hxs | Well, then you can use the string for other fun stuff. |
| 19:10.23 | Katty | saw that coming!!! |
| 19:10.36 | Katty | if you hadn't said it, drmessano would have :P |
| 19:10.42 | n3hxs | ;) |
| 19:10.49 | leifmadsen | tompaw: my filter arrived yesterday! thanks! |
| 19:10.54 | drmessano | String is good for strangling hookers and tying small things to other things. I prefer rope |
| 19:11.12 | Katty | we really don't need to hear about drmessano preferences >.< |
| 19:11.35 | Katty | let's not disturb my brain today, yes? drmessano :P |
| 19:12.24 | drmessano | lol |
| 19:12.36 | n3hxs | qlok, I think that E1 is not ISDN... from what I know. |
| 19:12.51 | Katty | days i happy to NOT be a telepath. |
| 19:13.17 | n3hxs | but would the path understand? |
| 19:14.22 | [TK]D-Fender | <qlok> so what i learned isdn has two types: e1 to connect to pstn, the other one to the internet <- There is CAS, PRI, BRI, and no doubt others as far as "voice" goes. Not just ONE form. |
| 19:14.40 | [TK]D-Fender | qlok, Stop wasting time on theories that your provider may not offer you. |
| 19:17.49 | qlok | what i need is an isdn which provide access to packet switched networks |
| 19:18.04 | [TK]D-Fender | qlok, Who says you actually need that? |
| 19:20.40 | qlok | for asterisk to accept incomings from softphones and route them to the gsm gateway. |
| 19:21.15 | jpsharp | Then you just need an internet connection, not necessarily an ISDN connection. |
| 19:21.43 | qlok | not reliable enough to handle 100 of calls simultaneously |
| 19:21.50 | drmessano | ..... |
| 19:21.58 | drmessano | An E1 won't do that |
| 19:22.09 | qlok | yes i finally understand that |
| 19:22.16 | *** join/#asterisk droemel (~droemel@p4FCADC39.dip.t-dialin.net) |
| 19:22.41 | qlok | i was actually confused. i thought e1 was isdn vice versa |
| 19:23.14 | qlok | i thought e1 was the isdn to transmit packet data |
| 19:23.17 | jpsharp | You can run ISDN over an E1, but E1 != ISDN. |
| 19:23.22 | jpsharp | No. |
| 19:23.35 | [TK]D-Fender | You CAN use E1 for data |
| 19:24.00 | [TK]D-Fender | Then agin, that's just 2mbps bi-directional ..... |
| 19:24.22 | [TK]D-Fender | qlok, and then there is the question of who it's going to talk to on the other side |
| 19:25.04 | *** join/#asterisk Galen (~Galen@rrcs-24-43-20-118.west.biz.rr.com) |
| 19:26.13 | qlok | people who gets the call |
| 19:26.36 | [TK]D-Fender | ... |
| 19:27.07 | [TK]D-Fender | No.. what TELCO do you think is going to be on the other end and what protocol do you think THEY are going to support? You are assuming what your TELCO offers you. |
| 19:27.24 | [TK]D-Fender | qlok, Stop. Right now. Call them up and go find out what services they even offer you. |
| 19:27.32 | [TK]D-Fender | qlok, Anything more is a waste of time. |
| 19:28.13 | qlok | mobile telco |
| 19:28.21 | [TK]D-Fender | NO, the ISD SIDE |
| 19:28.24 | drmessano | Oh god |
| 19:28.39 | [TK]D-Fender | You keep trying to connect GSM to SOMETHING ELSE. |
| 19:28.43 | [TK]D-Fender | the OTHER side. |
| 19:28.45 | [TK]D-Fender | ^ |
| 19:28.46 | qlok | dont worry about that |
| 19:28.53 | drmessano | Are we leveraging emerging technologies? |
| 19:28.59 | [TK]D-Fender | That's what we have wasted our time talking about. |
| 19:28.59 | qlok | im not in america |
| 19:29.13 | [TK]D-Fender | qlok, Yes, we knew that. |
| 19:29.18 | [TK]D-Fender | qlok, And that doesn't amtter. |
| 19:29.40 | qlok | the gsm will act as pstn |
| 19:29.41 | [TK]D-Fender | GSM wireless is ONE side and then you are going to the PSN ont he OTHER. HOW you get there is what we've wasted all this time debating. |
| 19:30.17 | *** join/#asterisk artyx (U2FsdGVkX1@junction.googleplex.net) |
| 19:30.20 | [TK]D-Fender | qlok, Are you talking about going FROM GSM, TO GSM using that gateway? |
| 19:30.56 | [TK]D-Fender | qlok, As in no other connection to the PSTN? |
| 19:31.02 | drmessano | qlok: Also, can I point out that even if you did have transport for a couple hundred calls TO your GSM gateway, you're going to need a gateway that can handle the couple hundred calls? |
| 19:31.04 | qlok | softphone-->asterisk--->gsm |
| 19:31.29 | qlok | thats why i need multiple 32 channels gsm gateways |
| 19:31.34 | [TK]D-Fender | Fine. And what is your connection between the softphone & Asterisk? |
| 19:32.14 | drmessano | I've not seen a readily available 32 channel GSM gateway.. Most of what I have had exposure to, at a non-carrier level, is 4 and 8 port |
| 19:32.45 | qlok | ok multiple 8 ports gsm gateway.. i thought i've seen one |
| 19:33.19 | [TK]D-Fender | qlok, please answer that last question of mine... |
| 19:33.37 | qlok | softphone--->sip<---asterisk--->gsm |
| 19:33.52 | jpsharp | So you need an IP connection internet connection |
| 19:34.02 | qlok | yes i do |
| 19:34.22 | jpsharp | So order one. |
| 19:34.36 | qlok | so is the codec thing now rellevant to ask |
| 19:35.01 | chuckf | thinks if Katty had allowed qlok to pm we wouldn't have had all this entertainment.... |
| 19:35.21 | Katty | :> |
| 19:35.29 | *** join/#asterisk ChanServ (ChanServ@services.) |
| 19:35.29 | *** mode/#asterisk [+o ChanServ] by pratchett.freenode.net |
| 19:35.31 | jmetro | Qlok, i think it would be best if you better describe where you are, what you have, and what you are trying to do, instead of asking questions that might be wrong or poorly phrased. |
| 19:35.32 | drmessano | Katty failed us again on the sexting front |
| 19:35.42 | qlok | softphone--->sip/gsm codec<---asterisk---->sip/gsm--->gsm gateway |
| 19:35.51 | Katty | i know. i'm a terrible sexter. |
| 19:35.59 | Katty | i should practice. |
| 19:36.02 | drmessano | Why would you want to use GSM to the gateway? Its over a fucking LAN |
| 19:36.29 | drmessano | softphone--->sip/gsm codec<---asterisk---->sip/WHOCARESITSONALAN--->gsm gateway |
| 19:36.29 | Katty | you need more colorful words dear. |
| 19:36.31 | [TK]D-Fender | qlok, How is the softphone reaching Asterisk. I'm asking about the networking between the two. not about the codec of the SIP call. |
| 19:36.44 | qlok | what so suggestion drmessano |
| 19:37.34 | drmessano | I think I mentioned like a dozen times that the CODEC between Asterisk and the GSM gateway is unimportant. I certainly wouldn't use GSM.. its a horrible codec |
| 19:38.03 | qlok | i just thought if the incoming calls is using gsm codec so asterisk dont have to transcoding job. saving the cpu |
| 19:38.08 | drmessano | Again, it's a discussion not worth having. The default on the device is likely g711. Dont mess with it |
| 19:38.14 | drmessano | .... |
| 19:38.56 | qlok | so what codec is the best one |
| 19:38.59 | drmessano | [14:29] <@Qwell> GSM protocol != GSM codec |
| 19:39.06 | drmessano | We went over this earlier |
| 19:39.07 | qlok | yes i know that |
| 19:39.11 | drmessano | No you do not |
| 19:39.20 | qlok | different between the protocol and codec |
| 19:39.44 | drmessano | Or you wouldnt keep bringing up transcoding and the GSM gateway. You dont need to send a call to the GSM gateway using the GSM codec |
| 19:39.51 | drmessano | Apples and Bananas |
| 19:39.58 | qlok | what say you TK |
| 19:40.30 | drmessano | Jeez.. Just ask for a second opinion right in front of me |
| 19:40.40 | qlok | TK the softphone will connect to asterisk using SIP |
| 19:41.08 | chuckf | Wow, that actually made sense |
| 19:43.27 | qlok | if i follow what drmessano said, asterisk will end up transcode the incoming call codec to what preffered by drmessano |
| 19:43.29 | [TK]D-Fender | qlok, I didn't ask for the protocol.... I asked what networking was there. |
| 19:43.34 | drmessano | As a matter of fact, the few GSM Gateways I just looked at don't even speak GSM |
| 19:43.35 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 19:43.48 | drmessano | g711, g723, g729 |
| 19:44.20 | qlok | ok what ever it is its doesnt matter right? cause its on lan |
| 19:44.52 | *** join/#asterisk lvlinux (~n1gg@c-50-147-64-9.hsd1.tn.comcast.net) |
| 19:45.10 | qlok | so asterisk----G711/G723/G729--->gsm gateway |
| 19:45.17 | [TK]D-Fender | qlok, The answer was LAN. that's what took 5 requests to get. So in this you asked a lot of questions about ISDN that really don't factor into your needs at all. |
| 19:45.37 | [TK]D-Fender | qlok, And so far the codec isn't even important. |
| 19:45.53 | drmessano | We should go over than 3 or 4 more times |
| 19:45.58 | Katty | there hasn't been much conversation in here for at least an hour |
| 19:46.04 | [TK]D-Fender | qlok, If you are looking at transcoding 32 channels .... my watch can do that ... and it's ANALOG. |
| 19:46.11 | qlok | i just want to find answer to squeeze hundred of calls into a single isdn simultaneously |
| 19:46.15 | qlok | its bandwidth issue |
| 19:46.23 | drmessano | You can't |
| 19:46.27 | jpsharp | You're not going to get there. |
| 19:46.38 | [TK]D-Fender | qlok, You're LAN between the softphone and your GSM gatway. There is no ISDN there. At all. Anywhere. |
| 19:46.41 | Katty | hi mister sharp. |
| 19:47.35 | *** join/#asterisk _Corey_ (~chatzilla@64.215.11.114) |
| 19:47.36 | qlok | softphones from around the world |
| 19:47.45 | qlok | not installed locally on my computer |
| 19:48.09 | jpsharp | So you need an internet connection > a 2mbps ISDN circuit. |
| 19:48.16 | jpsharp | s/ISDN/E1/ |
| 19:50.23 | qlok | softphones(from around the world)---->asterisk---->gsm gateway |
| 19:51.18 | drmessano | Look, you're not the first guy who decided they wanted to start a telco and needed some way to shove 10,000 calls over a 56k dialup connection. I have a guy that messages me every few weeks insisting on VPN'ing 23 calls over a 300kb link, if only he can find the right codec |
| 19:51.38 | jpsharp | LPC10! |
| 19:51.58 | [TK]D-Fender | qlok, Still no ISDN anywhere in there.... |
| 19:52.00 | qlok | i didnt say over a stupid 56k modem.. |
| 19:52.14 | qlok | i said isdn 2mbps |
| 19:52.18 | [TK]D-Fender | qlok, And it took another 5 questions to find out it's "softphones around the world" and not LAN. |
| 19:52.32 | [TK]D-Fender | qlok, And you still haven't answered how THOSE are going to reach Asterisk... |
| 19:52.38 | artyx | drmessano: Hmm, voice to text and back again? |
| 19:52.49 | jpsharp | IP over carrier pigeon |
| 19:52.52 | artyx | granted all his callers will sound like stephen hawking |
| 19:52.59 | artyx | IP over avian, i remember that RFC |
| 19:53.31 | [TK]D-Fender | qlok, the most you're going to get over even a RAW E1 is 30 calls using SIP+GSM6.10 |
| 19:53.32 | artyx | hell, you could pitch that, a good speech to text converter, and back again, true anonymity! never get blackmailed with a recording of YOUR voice |
| 19:53.55 | *** join/#asterisk keycruncher (~Adium@c-174-59-233-118.hsd1.pa.comcast.net) |
| 19:54.02 | jmetro | Artyx: Stephen HAWKing.. voice over pigeon.. HAH! |
| 19:54.02 | [TK]D-Fender | 60 rather. |
| 19:54.16 | artyx | jmetro: But the problem is the strings get tangeld up after a few calls... |
| 19:54.30 | artyx | one, two, after that the poor birds just fly in circles |
| 19:55.34 | qlok | softphones---sip+gsm--->asterisk |
| 19:55.40 | drmessano | artyx: I am not sure where he is from, but VoIP is illegal in his country, and he originally wanted to pay for Asterisk install assistance. Then it became "My boss said he knows you can pass 32 calls over 300kb GSM data line with SSL VPN. How to do?" Then it went into wanting a DISTRO built to roll Asterisk + VPN with this magical codec |
| 19:57.13 | qlok | asterisk---sip+whatever codec supported--->gsm gateway |
| 19:57.31 | jmetro | DrMessano: and i thought we had it bad with cable companies locking down on Net Neutrality to keep their business since everyone is flocking to streaming video. |
| 19:57.42 | *** part/#asterisk hariom (~hariom@117.195.188.80) |
| 19:58.42 | [TK]D-Fender | qlok, Where do you see that your telco offers you internet over E1? Where do you see that this would be better than any other service they could offer you? And then you're STILL not getting even the 60 I quoted earlier... that's best case, and you never get that... |
| 20:00.21 | [TK]D-Fender | (60 simultaneous calls). |
| 20:00.51 | qlok | why not? |
| 20:01.40 | *** join/#asterisk Emmanuel18 (AcidBurn@per18-2-88-165-49-175.fbx.proxad.net) |
| 20:01.41 | artyx | well, yum is a pretty slick trick, i'm sure freepbx centos would have handled hte rollout requirement, but im not sure about this magic codec he wanted... 32 calls on 300kb/ps ... It'll sound like roger-wilco by the time hes done |
| 20:01.43 | qlok | if a 64k(G711) can make 30, why dont gsms |
| 20:01.54 | Emmanuel18 | Bonsoir |
| 20:02.28 | Emmanuel18 | je viens d'installer freepbx tout configurer comme il faut |
| 20:02.44 | Emmanuel18 | mais j'ai un message d'erreur et je ne peux pas passer de communications |
| 20:02.54 | Emmanuel18 | pourrai je avoir de l'aide s'il vous plait? |
| 20:02.57 | [TK]D-Fender | qlok, Because that is using PRI for VOICE. |
| 20:03.02 | jpsharp | Cause a G711 VOIP call takes about 80k with IP overhead. |
| 20:03.13 | [TK]D-Fender | qlok, SIP over TCP/IP has PACKET overhead |
| 20:03.29 | qlok | what about gsm over ip |
| 20:03.30 | [TK]D-Fender | qlok, E1 VOICE != E1 DATA |
| 20:03.49 | qlok | isdn not e1 |
| 20:03.54 | [TK]D-Fender | And you do not understand SIP or RTP, or much else at this point |
| 20:04.51 | [TK]D-Fender | qlok, You aare trying to do math on things you do not understand at all yet. |
| 20:06.29 | Emmanuel18 | http://pastebin.com/Pi0zV120 |
| 20:06.42 | *** join/#asterisk elico (~Thunderbi@109.64.221.76) |
| 20:07.35 | [TK]D-Fender | Emmanuel18, t'as pas faite un <<outbound route>> pour accomoder votre # 0951107239 |
| 20:07.56 | [TK]D-Fender | Emmanuel18, Il ne sait pas quoi faire avec... |
| 20:08.27 | Emmanuel18 | ok |
| 20:08.29 | [TK]D-Fender | Emmanuel18, et tu devrait etre dans #freepbx pour tous ces questions.... |
| 20:08.37 | Emmanuel18 | ok |
| 20:09.35 | qlok | just give me a rough figure how many calls a single isdn which connected to packet switched network can handle simultaneously to connect to asterisk server before asterisk route the call to the gsm gateway |
| 20:10.23 | qlok | assuming all calls are made using gsm codec over ip |
| 20:10.51 | jpsharp | 24kbps per call. |
| 20:11.04 | jpsharp | 24kbps / 2 mbps of E1 |
| 20:11.08 | [TK]D-Fender | jpsharp, not for GSM6.10\ |
| 20:11.39 | [TK]D-Fender | qlok, 13kbps PAYLOAD, 20kbpbs OVERHEAD. Per CALL. |
| 20:11.39 | qlok | so roughly 80 concurrent calls |
| 20:12.18 | [TK]D-Fender | qlok, **60** like I've told you repeatedly. Best case. And yuo aren't going to GET best case because of toehr factors like SIP keep-alives, and other random traffic. |
| 20:12.31 | [TK]D-Fender | qlok, And assuming your telco will even OFFER it. |
| 20:12.34 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 20:12.50 | qlok | yes they do offer it |
| 20:13.05 | rrittgarn | Anybody know of a SIP based attendant console that can do 100 BLFs? |
| 20:13.29 | qlok | finally |
| 20:14.26 | drmessano | http://www.asteriskguru.com/tools/bandwidth_calculator.php <--- There you go |
| 20:14.42 | [TK]D-Fender | rrittgarn, iPad :) |
| 20:15.19 | _Corey_ | rrittgarn: Digium D70 Phone |
| 20:15.22 | [TK]D-Fender | rrittgarn, or any PC |
| 20:15.28 | drmessano | OMG.. I keep forgetting about the best codec of all.. LPC10 |
| 20:15.43 | [TK]D-Fender | drmessano, Do..mo..ar...i..ga....TO!! |
| 20:15.47 | qlok | at least its double the capacity it can handle compare to 64k calls over 2mbps isdn |
| 20:16.06 | drmessano | Shhh.. nobody mention LPC10 and it's bandwidth usage |
| 20:16.19 | kaldemar | 600 bit/s MELP would suit him. |
| 20:16.21 | rrittgarn | fender: Which app(s) are you thinking? |
| 20:16.39 | [TK]D-Fender | drmessano, Consdiering that the payload at GSm6.10 is SMALLER than the overhead ... LOL <- |
| 20:17.04 | [TK]D-Fender | rrittgarn, I'm sure one could be wtieen rather easily.. or browser-based w/ HTML/5 |
| 20:17.07 | drmessano | What about FOP2? You can resize everything |
| 20:17.23 | [TK]D-Fender | drmessano, FLASH ....on ..an iPad <- |
| 20:17.50 | drmessano | I didn't suggest the iPad. |
| 20:17.54 | [TK]D-Fender | drmessano, http://cdn.walyou.com/wp-content/uploads//2010/04/the-flash-iphone-ipad-spoof.jpg |
| 20:18.12 | [TK]D-Fender | http://www.guiadophp.yoonix.com.br/wp-content/uploads/2010/09/flash-ipad.png |
| 20:18.33 | drmessano | Sadly though, Steve won |
| 20:19.14 | drmessano | Flash is going bye-bye, in part due to it's poor scaling on the mobile platform, and the fact its crap |
| 20:19.14 | tzanger | nah, I'm happy to see flash go away |
| 20:19.24 | tzanger | HTML5 is pretty versatile |
| 20:19.33 | drmessano | HTML5 has a way to go |
| 20:19.37 | tzanger | sure |
| 20:19.45 | tzanger | but it's not locked behind one company |
| 20:20.07 | drmessano | Poor Facebook for betting the farm on HTML5. Ended up rewriting their iOS app in native code |
| 20:20.34 | jpsharp | And there was much rejoicing. |
| 20:20.46 | drmessano | That's not the word for it |
| 20:20.48 | artyx | (Yay) |
| 20:21.11 | drmessano | I still have a 3GS. It went from 'FROCKING UNUSABLE' to 'NOT BAD' |
| 20:22.10 | drmessano | My wife is a Facebook junkie, and I had to hear like 3x a week about how I didn't see, or even worse that I didn't 'like' some post she made. "My phone sucks!" "Doesn't matter.. SAD FACE" |
| 20:23.35 | *** part/#asterisk qlok (~jereminsh@210.186.243.127) |
| 20:24.22 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:31.23 | [TK]D-Fender | checkout time, BBIAB |
| 20:38.15 | *** join/#asterisk vlad_starkov (~vlad_star@77.41.88.210) |
| 20:44.30 | *** part/#asterisk keycruncher (~Adium@c-174-59-233-118.hsd1.pa.comcast.net) |
| 20:54.01 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:54.27 | Dovid | anyone here use mplayer with asterisk? |
| 21:01.08 | drmessano | No, I think it's been discouraged over mpg123 |
| 21:24.06 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 21:39.09 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 21:47.46 | *** join/#asterisk Zelitors (~Zelitors@189.89.153.98) |
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| 21:50.38 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 21:51.24 | artyx | So i dial 7 digit, it works, i dial 10 digit, it works, i dial 11 digit, it works, i dial 4 digit with handset off cradle, it works, but 4 digit with handset on cradle fails/times out/goes back to primary phone screen. 4 digit and # with handset in cradle works |
| 21:51.54 | artyx | Is this a phone configurational issue, or a pbx configuration problem |
| 21:54.00 | *** join/#asterisk aross42 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 21:58.51 | *** join/#asterisk bmg505 (~leon@196-209-7-243.dynamic.isadsl.co.za) |
| 22:03.37 | *** join/#asterisk droemel (~droemel@p4FCADC39.dip.t-dialin.net) |
| 22:05.25 | *** join/#asterisk TSM (~the_softw@fw-lon1.wenn.com) |
| 22:09.58 | leifmadsen | artyx: dialplan on the phone itself |
| 22:10.18 | artyx | Thanks, looking there then |
| 22:10.31 | artyx | This is the ONLY problem ive had dialign with the yealinks so far |
| 22:20.01 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 22:20.02 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 22:23.26 | paulc | SBC sends call to Asterisk, which answers, interacts a bit, then Dials back to SBC, where the media gets bridged via a re-invite. Calls drop after 1800 seconds. Change sip.conf session-expires to a big number and calls drop after 3600 (the SBC's expiry timer) |
| 22:24.16 | paulc | First time I've come across this kind of issue and I'm feeling a bit lost.. would a different value of session-timers (like refuse?) help, or session-refresher (because some change was made between 10.8 and 10.9?) |
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| 22:52.48 | Rahoul | Hi I am having problems trying to place calls over google talk. jabber show connections shows my username is connected and this is my dial plan: http://pastebin.com/agz91tU2 the message I get on my sip phone is Service Not Available |
| 22:53.41 | leifmadsen | Rahoul: you should be using Asterisk 11 + chan_motif + res_xmpp |
| 22:53.46 | leifmadsen | old methods are unreliable |
| 22:55.47 | Rahoul | hi Leif, I was actually working with your book :-) I think I am using asterisk 1.8 atm can you point me in the right direction (is there a tutorial)? |
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| 22:57.47 | Rahoul | I found the following: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google I will try to get these modules loaded on my version |
| 22:58.16 | Rahoul | Great book by the way :D |
| 22:58.25 | Rahoul | bought the Kindle version |
| 23:00.41 | Rahoul | leifmadsen: Thanks |
| 23:10.03 | jpsharp | RV works great for me on Asterisk 10.3.blah |
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