| 00:13.49 | CoffeeIV | my res_timing_dahdi is loaded, but chan_dahdi.so was not built when I made asterisk - the make menucofig is showing "XXX" and "Depends on: dahdi(E), tonezone(E)" for that, I'm trying to fix that and recompile now |
| 00:19.28 | *** join/#asterisk SeRi (~wtf@c-98-200-53-71.hsd1.tx.comcast.net) |
| 00:20.47 | SeRi | well I am officially on arch. |
| 00:20.54 | SeRi | left slackware behind for a bit |
| 00:20.59 | SeRi | lets see how this works out |
| 00:23.56 | *** join/#asterisk jsjc (~Adium@213.37.155.15) |
| 00:30.03 | *** join/#asterisk BrennoEmanuel (~Thunderbi@177.27.203.225) |
| 00:32.43 | *** join/#asterisk parasitodelsur (~wtf@c-98-200-53-71.hsd1.tx.comcast.net) |
| 00:35.25 | BrennoEmanuel | What do you use to make call from GSM network? |
| 00:36.51 | BrennoEmanuel | i mean hardware |
| 00:38.28 | jpsharp | Want to place GSM calls from Asterisk? |
| 00:38.54 | jpsharp | There's Asterisk-friendly GSM PCI cards. Or there's SIP gateways with GSM radios in them. |
| 00:40.40 | BrennoEmanuel | jpsharp: First option |
| 00:41.34 | jpsharp | OpenVOX makes Asterisk friendly GSM cards. |
| 00:42.10 | jpsharp | As does Junghanns.NET |
| 00:42.14 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 00:45.16 | *** join/#asterisk DaveCanoe (~Dave@2001:1928:0:ffff::2) |
| 00:46.12 | BrennoEmanuel | jpsharp: Thanks |
| 01:00.19 | *** part/#asterisk BrennoEmanuel (~Thunderbi@177.27.203.225) |
| 01:03.50 | CoffeeIV | I'm attempting to compile asterisk 1.8.15.1 on Ubuntu 10.4, I have the dahdi-dkms module installed and lsmod shows the module loaded. However when I compile asterisk, even with the --with-dahdi=/usr/src/dahdi-2.2.1+dfsg-1ubuntu2 on ./configure, make menuconfig shows the dahdi channel driver as XXX and you can't compile it. Is there a step I am missing ? |
| 01:09.37 | *** join/#asterisk KingDavidNYC (~IceChat77@pool-74-96-172-86.washdc.fios.verizon.net) |
| 01:09.56 | KingDavidNYC | Hello Everyone |
| 01:17.15 | KingDavidNYC | Can someone please help me with a polycom phone which for the life of me doesn't want to register? |
| 01:19.01 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 01:20.02 | *** join/#asterisk KingDavidNYC (~Chris1232@pool-74-96-172-86.washdc.fios.verizon.net) |
| 01:20.10 | KingDavidNYC | Hello Everyone |
| 01:22.00 | Mango45 | Does anyone have any idea why a macro would cause problems with direct media? |
| 01:26.25 | jpsharp | CoffeeIV: Look through config.log and see why it isn't detecting DAHDI |
| 01:27.19 | CoffeeIV | in regards to my quesiton about compiling on ubuntu, not using the ubuntu supplied modules for dahdi and downloading the dahdi-linux-complete tar file from asterisk.org seemed to make the difference |
| 01:27.47 | CoffeeIV | jpsharp: I did look in the config.log - there were some dahdi related tests that failed and most of them past |
| 01:28.09 | CoffeeIV | now, with the asterisk.org source, they all pass, I'm installing and restarting now to see if it works :) |
| 01:47.31 | *** join/#asterisk Kako (~Kako@ip-95-223-21-26.unitymediagroup.de) |
| 01:50.10 | KingDavidNYC | Can anyone please help me with a polycom phone which for the life of me I can not get to register? |
| 02:03.22 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 02:23.26 | KingDavidNYC | Hello |
| 02:23.32 | KingDavidNYC | anybody here? |
| 02:26.31 | Mango45 | Yes. |
| 02:32.26 | *** join/#asterisk YoMomma (~YoMomma@cpe-98-149-120-227.socal.res.rr.com) |
| 02:33.13 | *** join/#asterisk jsjc (~Adium@213.37.155.15) |
| 02:33.56 | *** join/#asterisk Kako (~Kako@ip-95-223-21-26.unitymediagroup.de) |
| 02:40.02 | KingDavidNYC | I am not sure if it is a problem with my IRC client software that doesn't show me all messages, or if just no one is talking :) |
| 02:40.16 | KingDavidNYC | what IRC software do you use? |
| 02:40.23 | KingDavidNYC | Mango45 |
| 02:40.38 | Mango45 | Trillian. |
| 02:42.12 | KingDavidNYC | I use mIRC, I seriously doubt it |
| 02:43.06 | KingDavidNYC | do you know Polycom/NAt? |
| 02:43.22 | Mango45 | Sorry, no. |
| 02:43.32 | KingDavidNYC | ok thanks |
| 02:53.11 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 02:55.20 | *** join/#asterisk jsjc (~Adium@213.37.155.15) |
| 03:06.51 | *** join/#asterisk deo (~dnepangue@222.127.13.226) |
| 03:11.30 | *** join/#asterisk volga629 (~volga629@76-10-130-18.dsl.teksavvy.com) |
| 03:13.46 | volga629 | Good Evening Everyone, I have this dial plan for app_konference.so and if understand correct it is for some old asterisk version is I am write and if yes how difficult will be adopt for asterisk 10.5.1 http://fpaste.org/TMKE/ any help and opinion thank you in advance |
| 03:14.17 | volga629 | right |
| 03:52.12 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 03:52.15 | fling | hello |
| 03:52.40 | fling | my pbx is behind the router |
| 03:55.43 | fling | -A PREROUTING -i metro -p udp -m multiport --dports 5060,10000:11122,11124:20000 -j DNAT --to-destination 10.0.1.101 |
| 03:55.56 | fling | it is accessible from the internets |
| 03:57.05 | fling | [asterisk] <- [router] <- (internets) <- sip clients |
| 03:57.09 | fling | ^ this works |
| 03:57.56 | fling | [asterisk] -> [router] -> (internets) -> [sip provider] |
| 03:57.59 | fling | ^ this is not |
| 03:58.11 | fling | so I can't register with asterisk |
| 03:59.21 | fling | sip show registry says 'Request Sent' |
| 03:59.41 | fling | sip show peers says 'UNREACHABLE' |
| 04:00.27 | fling | I've tried to run asterisk directly on router and it works, peer is OK and it is Registered |
| 04:01.20 | fling | what am I doing wrong? am I missing some packets I need to forward? or do I need a special config option? |
| 04:03.38 | volga629 | If you asterisk default gateway is router ? |
| 04:03.45 | volga629 | Is |
| 04:03.51 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-lityhhnxwvnadjwh) |
| 04:05.50 | fling | volga629: no; there are a lot of routers on the way to [router] which is nat for my lan |
| 04:07.10 | volga629 | Do you have forward rules between interfaces ? |
| 04:08.10 | fling | volga629: yes, I can wget sites directly from my pbx |
| 04:11.10 | volga629 | Did you tried use -A POSTROUTING -s pbx network -o eth1 -j MASQUERADE under nat ? |
| 04:12.28 | fling | volga629: no, I use this -A POSTROUTING -o metro -j SNAT --to-source <my-ip-addr> |
| 04:12.37 | fling | volga629: do I need masquerade? |
| 04:12.43 | volga629 | try it |
| 04:12.46 | fling | ok |
| 04:15.34 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 04:16.03 | fling | volga629: -A POSTROUTING -s 10.0.1.101/32 -o metro -j MASQUERADE |
| 04:16.28 | fling | volga629: 10.0.1.101 is my pbx address and metro is interface listening to internets :p |
| 04:17.13 | volga629 | yes |
| 04:17.27 | fling | nothing changed |
| 04:17.52 | *** join/#asterisk ziz212 (~chatzilla@203.115.2.202) |
| 04:17.56 | ziz212 | hi, |
| 04:18.00 | fling | ziz212: hello |
| 04:18.05 | volga629 | can you reach router |
| 04:18.16 | fling | volga629: reach |
| 04:18.18 | ziz212 | hi |
| 04:18.37 | volga629 | if you do traceroute |
| 04:19.12 | fling | volga629: http://dpaste.com/798793/ |
| 04:19.50 | volga629 | [asterisk] -> [router] -> (internets) as I see only one router on the way out right ? router is next hope ? |
| 04:20.28 | volga629 | If so asterisk should have point gateway to router internal ip |
| 04:20.42 | volga629 | default gateway |
| 04:20.57 | volga629 | if you do netstat -nr |
| 04:21.10 | ziz212 | I just need to find out better sip provider to get the service to asterisk box. My asterisk box behind broadband and with Static WAN ip and NAT. So which provider is good to choose? I know that some sip providers having some issues for NAT broadband connectivity. Pls help |
| 04:23.35 | fling | volga629: ok, there is a complete thing : [pbx 10.0.1.101] -> swithes -> [10.0.1.1 hatchery (default gw for pbx) 10.1.0.2] -gre-> (internets) -gre-> [10.1.0.1 mirror (my nat) <my-ip-addr>] -> internets -> [sip provider] |
| 04:24.25 | fling | ziz212: where are you from? |
| 04:24.29 | volga629 | so you ipsec tunnel GRE |
| 04:24.53 | fling | volga629: will use, now I'm just testing |
| 04:24.57 | ziz212 | my server is hosted in UAE |
| 04:25.10 | ziz212 | Berute |
| 04:25.29 | fling | ziz212: do you want to call/receive calls to/from pstn? |
| 04:25.35 | volga629 | Are you trying use VLAN ? |
| 04:25.39 | fling | volga629: no |
| 04:25.48 | ziz212 | yes |
| 04:26.03 | fling | volga629: I just have fine routes and a single snat |
| 04:26.14 | ziz212 | and i need to have asterisk to receive and cll |
| 04:26.43 | fling | ziz212: so just google for a services which will provide a number and pstn services in countries you want to call to/from |
| 04:28.11 | ziz212 | just find out sip service provider which gives sip trunks to my hosted asterisk which is behind the broadband NAT but with static ip for WAN |
| 04:28.26 | ziz212 | I ve used nymgo and itis not working |
| 04:28.29 | ziz212 | issues are there |
| 04:29.05 | *** join/#asterisk youjelly (~bwahahaha@182.177.108.29) |
| 04:29.53 | volga629 | on router where ipsec tunnel you need use MASQUERADE |
| 04:31.56 | volga629 | Do you see any traffic on tcp dump from pbx ? |
| 04:33.18 | volga629 | do you want replace this ip 10.0.1.101 to 10.0.1.1 ? |
| 04:33.41 | volga629 | that source will be 10.0.1.1 ? |
| 04:36.03 | volga629 | If replace source use something like this on pbx iptables -t nat -A POSTROUTING -o eth0 -j SNAT --to 10.0.1.1 |
| 04:37.49 | fling | volga629: it will not work |
| 04:39.21 | volga629 | did you tried use just static routes to reach network behind another end of tunnel ? |
| 04:40.30 | fling | volga629: [pbx] -> [hatchery] -gre-> [mirror] -> (internets) |
| 04:40.37 | fling | volga629: I use static routes everywhere |
| 04:40.45 | volga629 | and on asterisk set NAT properly |
| 04:40.52 | fling | I have a single snat on mirror |
| 04:41.07 | fling | how to set it properly? |
| 04:42.29 | volga629 | nat=yes externip= localnet= |
| 04:46.01 | fling | volga629: still nothing |
| 04:47.53 | volga629 | where in chain traffic stop going, if you trying generate traffic from pbx out to provider ? |
| 04:48.30 | fling | is starting tcpdump |
| 04:50.43 | fling | volga629: I see packets outgoing from gre on mirror '11:49:40.866789 IP 10.0.1.101.5060 > …' |
| 04:51.28 | fling | nothing on wan port :[ |
| 04:54.36 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 04:59.53 | fling | something is wrong |
| 05:02.42 | volga629 | So something not right where is tunnel is this hardware router ? |
| 05:03.20 | fling | Linux mirror 3.2.12-gentoo_fling #2 SMP Tue May 1 15:15:37 NOVT 2012 x86_64 AMD Phenom(tm) II X4 965 Processor AuthenticAMD GNU/Linux |
| 05:03.40 | volga629 | tunnel is openswan ? |
| 05:03.51 | fling | no, just gre |
| 05:04.08 | volga629 | ip add tunnel ? |
| 05:04.16 | volga629 | right ? |
| 05:04.27 | fling | http://dpaste.com/798803/ |
| 05:07.19 | ziz212 | hi, I have seen some options messages going from my asterisk to sip provider display as "From: "Unknown" <sip:Unknown@192.168.6.16>;tag=as2421ee06". What is this and how to solve this? |
| 05:08.09 | volga629 | and you can ping both ends of tunnel ? |
| 05:08.16 | volga629 | on local side |
| 05:08.45 | fling | volga629: yes |
| 05:10.51 | kaldemar | ziz212: that's the result of qualify=yes. solve what? stop the messages or change unknown to something else? |
| 05:12.48 | volga629 | Do you allow any outbound traffic from gre to pbx you can reached ? |
| 05:13.15 | ziz212 | yes I have commented out and now it is not there. But still cant take calls |
| 05:15.29 | ziz212 | How can i sort out error ---> "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 6 - failing through to other trunks" ... |
| 05:16.54 | fling | volga629: -A FORWARD -d 10.0.1.101/32 -p udp -j ACCEPT ; -A FORWARD -s 10.0.1.101/32 -p udp -j ACCEPT |
| 05:17.54 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 05:19.04 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 05:19.05 | volga629 | and do you have INPUT ? |
| 05:19.40 | fling | volga629: only -P INPUT ACCEPT on pbx |
| 05:21.51 | volga629 | example INPUT.... after we want forward from eth0 wan to eth 1 local and after all nat and -A FORWARD -i eth0 -o eth1 -m state --state ESTABLISHED,RELATED |
| 05:22.09 | volga629 | -j ACCEPT |
| 05:23.53 | kaldemar | ziz212: that's not an error from asterisk, but output from your dialplan. what exactly did you comment out? the whole peer for your provider? |
| 05:23.56 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 05:24.18 | fling | volga629: eth0 is wan and eth1 is gre? |
| 05:24.28 | volga629 | yes |
| 05:28.33 | ziz212 | I commented out qualify=yes line |
| 05:28.40 | fling | mirror ~ # iptables -I FORWARD -i metro -o to_hatchery -m state --state ESTABLISHED,RELATED -j ACCEPT |
| 05:28.43 | fling | hatchery ~ # iptables -I FORWARD -i to_mirror -o lan -m state --state ESTABLISHED,RELATED -j ACCEPT |
| 05:29.47 | volga629 | Did you tried ? |
| 05:30.06 | fling | nothing changed |
| 05:30.45 | volga629 | is tunnel up ? |
| 05:31.24 | fling | volga629: yes :p |
| 05:31.55 | volga629 | Is iptables have logs ? |
| 05:32.10 | ziz212 | the whole peer for your provider? --<--- means |
| 05:32.20 | ziz212 | host parameter or ?? |
| 05:32.34 | fling | everything works perfectly, I have a lot of things in that lan: http, ftp, rsync, a lot of workstations, only have problem with asterisk |
| 05:32.52 | fling | volga629: yes |
| 05:33.21 | volga629 | Do you see something related ? |
| 05:33.38 | fling | volga629: it is not really possible to see |
| 05:33.49 | ziz212 | Can I post my sip settings? |
| 05:34.09 | fling | volga629: watch -n,1 -d=cumulative iptables-save |
| 05:34.34 | volga629 | hmmm and nothing in files ? |
| 05:34.51 | fling | nothing |
| 05:35.04 | fling | iptables -c shows which rule is hit |
| 05:35.12 | *** join/#asterisk deo (~dnepangue@203.177.214.75) |
| 05:35.50 | ziz212 | kaldemar: which settings that i need to look for this error? Pls help |
| 05:35.50 | fling | so sometimes it is possible to detect something when you use watch's cumulative differenfe |
| 05:37.29 | volga629 | or just see packet increase |
| 05:38.13 | fling | I see rules related to pbx are counting. |
| 05:39.10 | ziz212 | Can some one help me to solve my issue ? I am getting "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 6 - failing through to other trunks" ...... |
| 05:39.11 | volga629 | when try use tcp dump on 5060 do you see sip negotiations ? |
| 05:39.38 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 05:40.17 | fling | volga629: nothing on wan interface |
| 05:40.33 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 05:41.46 | volga629 | So traffic coming out of gre tunnel and not use SNAT to WAN ? |
| 05:42.07 | fling | umm? |
| 05:42.19 | fling | mirror ~ # tcpdump udp port 5060 -i metro |
| 05:42.27 | fling | ^ this shows me nothing |
| 05:42.42 | fling | mirror ~ # tcpdump udp port 5060 -i to_hatchery |
| 05:42.50 | fling | ^ this shows outgoing packets |
| 05:52.22 | ziz212 | Can some one help me to solve my issue ? I am getting "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 6 - failing through to other trunks" ...... |
| 05:53.22 | kaldemar | ziz212: sip.conf and extensions.conf. your "error" is just a line of output that doesn't necessarily tell much about what happens. pastebin CLI output for the WHOLE call to give more information. |
| 05:55.32 | volga629 | fling check this http://www.linuxjournal.com/article/9399 I am think missing some DNAT, but I am trying guess setup and topology which make difficult to give you clear answer where the problem |
| 06:01.15 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 06:03.11 | ziz212 | kaldemar: Can i send that to your private channel becaus it contain sip server information and trunk information ?????? |
| 06:05.22 | kaldemar | why would you trust me any more than anyone else? don't send it to me in private, mask the information that you don't want to show. |
| 06:11.03 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:11.05 | schmidts | good morning |
| 06:11.11 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 06:13.39 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 06:14.51 | youjelly | good morning |
| 06:19.25 | *** join/#asterisk gurra (~gurra__@unaffiliated/gurra) |
| 06:20.54 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 06:29.06 | mirela666 | good morning :) |
| 06:29.12 | v0lZy | morning |
| 06:32.13 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 06:33.44 | ziz212 | ok cool ... here is my post --> http://pastebin.com/GeH7DZ3f I am trying to take a call to outside no 9611351400 from extension 4001 |
| 06:34.15 | ziz212 | Your valuable comments expected.. |
| 06:34.40 | ziz212 | kaldemar: Pls help me |
| 06:35.17 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 06:42.02 | *** join/#asterisk xnixan (~xnixan@unaffiliated/xnixan) |
| 06:46.22 | kaldemar | ziz212: externip and localnet are not peer paramaters, they belong under [general]. |
| 06:46.41 | *** join/#asterisk jsjc (~Adium@225.Red-83-49-190.dynamicIP.rima-tde.net) |
| 06:46.51 | kaldemar | dtmfmode=RFC2833 => dtmfmode=rfc2833 |
| 06:48.10 | kaldemar | your dial via nymgo-out fails due to some reason. check with "sip show peers" if asterisk has an ip address for it. |
| 06:53.48 | ziz212 | Thanks doing that |
| 06:55.58 | kaldemar | the second dial via DAHDI (g0) also fails due to hangup cause 6. what is g0? analog? PRI? |
| 06:58.50 | ChannelZ | shoe string |
| 07:02.01 | kaldemar | with cans of nanya. |
| 07:03.44 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 07:06.33 | *** join/#asterisk pbxMan (c335d9a4@gateway/web/freenode/ip.195.53.217.164) |
| 07:06.37 | pbxMan | hello |
| 07:10.24 | *** join/#asterisk coppice (~chatzilla@m121-202-33-96.smartone.com) |
| 07:15.14 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 07:16.23 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 07:16.34 | gavimobile | some of my softphones are showing that they are connected with sip show peers. they receive incomming calls, but I cannot make a call rfom them, sip debug doesn't show anything |
| 07:16.38 | *** join/#asterisk Vince-0 (c4d7bcf4@gateway/web/freenode/ip.196.215.188.244) |
| 07:17.48 | gavimobile | nothing, not even dead air |
| 07:18.20 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:cdf9:5975:d4f4:6cd3) |
| 07:25.52 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 07:26.17 | ChannelZ | then they are confused |
| 07:26.31 | gavimobile | oops, did I miss something |
| 07:26.37 | gavimobile | I just tried rebooting my router |
| 07:26.49 | ChannelZ | no just commenting your softphones are confused |
| 07:26.54 | gavimobile | could you repost |
| 07:27.12 | ChannelZ | Or your router has gone to sleep and is bit-bucketing the incoming traffic from them, |
| 07:27.32 | gavimobile | ChannelZ: did I say softphones? my bad, two of my hardphones are making outgoing calls. my softphone can make calls. |
| 07:27.52 | ChannelZ | either way |
| 07:28.23 | gavimobile | what type of diagnosis can I run to do further testing? |
| 07:29.48 | ChannelZ | Are they set to qualify in sip.conf? |
| 07:30.29 | ChannelZ | Actually it makes no sense that they'd be able to receive calls but not make them. Usually it's the other way around if it's firewall issues. |
| 07:30.35 | kaldemar | gavimobile: the phones might be misconfigured to use a proxy that is not your asterisk. |
| 07:31.03 | ChannelZ | Yeah you gotta figure out if they're trying to send their traffic to the right place or not first |
| 07:31.35 | gavimobile | the domain didn't change |
| 07:32.33 | ChannelZ | Do they work for a time if you reboot them and then stop working? |
| 07:34.11 | gavimobile | ChannelZ: no |
| 07:34.21 | gavimobile | ChannelZ: I wouldn't say not working, cause they register |
| 07:34.34 | ChannelZ | Not working in whatever manner they are not working |
| 07:34.42 | ziz212 | kaldemar: I ve applied the changes but still the same pls see the output of a call ...--> http://pastebin.com/P6fz0crn sip show peers--> http://pastebin.com/zhTmEjuT |
| 07:34.57 | gavimobile | just my polycom and my ata device won't report any outbound info |
| 07:35.05 | ziz212 | g0 is a mistake and i ve removed it |
| 07:35.21 | gavimobile | would you care for an explination on what I did? |
| 07:35.54 | kaldemar | ziz212: sip debug will tell you more. |
| 07:38.18 | ziz212 | yes I ve gone through it and difficult to find out the error |
| 07:38.26 | ChannelZ | gavimobile: if you're not even getting a SIP message from them when you try to call, it's something external to asterisk - your routers/firewalls on either end, or the config of the device its self |
| 07:39.28 | kaldemar | ziz212: maybe you should pastebin the sip debug... |
| 07:39.28 | ziz212 | Do you need any other information .... regarding this? |
| 07:39.48 | kaldemar | ziz212: sip debug is what is needed. |
| 07:39.49 | gavimobile | the hardphone settings never changed since it was working yesterday.. only the port forwarding in my router changed.. maybe my router is a shyt router and I need to completely unplug it |
| 07:40.08 | ChannelZ | like I said |
| 07:40.15 | ChannelZ | What changed on the port forwarding? |
| 07:40.28 | gavimobile | the forwarding address |
| 07:40.32 | ziz212 | ok ... debug information before call dialing to call faild ? |
| 07:40.56 | kaldemar | ziz212: for the whole call. |
| 07:41.01 | gavimobile | ChannelZ: I changed it back and rebooted the router both ther router and server |
| 07:43.15 | ChannelZ | well as I said, if sip debug shows 0 traffic, you've got to troubleshoot your end(s) - either the phones are sending INVITEs to the wrong place, or they are getting blocked either leaving their network or entering yours. |
| 07:44.28 | ChannelZ | If you see them register it implies the port forwarding is working |
| 07:44.35 | ziz212 | sip debug ---> http://pastebin.com/PUNFHhNf |
| 07:44.41 | kaldemar | gavimobile: the fact that a phone sends registers to your asterisk box does not mean it even tries to send invites (calls) to the same place. |
| 07:45.28 | gavimobile | what does it mean that they won't even make a local call to another extention |
| 07:45.39 | gavimobile | but can receive calls from outside or from another ext |
| 07:46.16 | ChannelZ | Because in one case Asterisk is sending packets to the phone, and in the other the phone should be sending packets to Asterisk |
| 07:46.49 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85e0.bcn.adamo.es) |
| 07:46.55 | kaldemar | ziz212: the required information is not in your paste. "sip set debug on", not only for the phone that you use to dial. |
| 07:47.28 | ChannelZ | And as he's pointed out config-wise, most devices can register to one place but send calls to another (like a proxy) so one success doesn't necessarily imply the other should work as well |
| 07:49.02 | gavimobile | unpdate: I just tested another softphone and I tried from both my internal ip and my extenal ip. they both seem to be working |
| 07:49.02 | ChannelZ | Registering simply tells the remote end (asterisk in this case) "Here I am, this is my IP" so asterisk knows where to send calls for it to. It has no other real bearing on where that phone might send its own calls to. |
| 07:49.28 | gavimobile | ChannelZ: good explination |
| 07:51.29 | ChannelZ | you should check they don't have some other proxy host entered into them.. and/or it could even be the device's dialplan not even attempting to place the call in the first place. |
| 07:52.02 | ziz212 | taking call, sip show peers ,,, sip reload .... all are there in sip debug ------>> http://pastebin.com/hCZqsQdG |
| 07:55.58 | kaldemar | still no sip traces to your provider in the paste. |
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| 07:57.27 | ziz212 | bottom portion it is there. Pls see it in last portion. |
| 07:58.45 | kaldemar | no it is not. there is no SIP traffic to/from the address that "sip show peers" shows for nymgo-out. |
| 07:59.15 | kaldemar | other than a registration, that is. |
| 08:02.32 | ziz212 | this is not still in production. so what i can do is " sip set debug on" take call, sip reload, wait one minit or two (otherwise log is getting bigger) , sip show peers( only one peer is there, 4001) what else .. you name it I do .." |
| 08:02.39 | ziz212 | pls let me know |
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| 08:17.21 | kaldemar | ziz212: your asterisk does not seem to be dialing the provider. at least that does not show up in sip debug. |
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| 09:15.09 | ziz212 | kaldemar: Again I have past the output sip reload ---> http://pastebin.com/1m5bQ9Df taking call to out side from sip trunk --> http://pastebin.com/1JLjXYLq pls check??? |
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| 09:23.21 | kaldemar | ziz212: still no. |
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| 09:24.03 | ziz212 | What kind of information your reffering |
| 09:24.13 | ziz212 | pls let me know |
| 09:27.58 | kaldemar | ziz212: there is no SIP traffic between asterisk and the provider in your paste. |
| 09:31.39 | ziz212 | 0k I am sorry. I got that in putty log. I will directly copy and past the full output. |
| 09:35.36 | ziz212 | from line 147 to 315 call logged here---> http://pastebin.com/1hdCyjuZ |
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| 09:37.08 | kaldemar | still no sip debug of any call attempt to your provider. |
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| 09:47.40 | ziz212 | sip debug menas .... sip set debug on command in astersik -r cli interface? |
| 09:48.44 | ziz212 | I would like you to provide the teamviewer access to putty machine |
| 09:50.28 | ziz212 | kaldemar: I would like you to provide the teamviewer access to putty machine |
| 09:50.38 | ziz212 | Can you help me in like that |
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| 09:54.53 | kaldemar | ziz212: no. |
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| 10:24.28 | ziz212 | ok kaldemar: thanks for the help that given to me for this. :) |
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| 10:42.30 | gavimobile | ChannelZ: are you still awake? |
| 10:43.30 | gavimobile | well at least your sleeing! that's a good sign |
| 10:57.23 | Chainsaw | Yay, sleeing. |
| 10:57.24 | Chainsaw | Wait, what? |
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| 11:15.37 | gavimobile | if my itsp is located in the states and I need to call a 1800 number in my country with country code 972 so I basiclly dial 9721800..? |
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| 11:20.34 | Chainsaw | gavimobile: Highly dependent on your dial plan. |
| 11:21.11 | Chainsaw | gavimobile: You may find that 1800 free numbers are simply not reachable internationally. |
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| 11:22.55 | WIMPy | And if they are reachable, they are probably not free. |
| 11:23.20 | Chainsaw | nods in agreement |
| 11:23.40 | Chainsaw | So if your ITSP offers line termination in Israel, you may want to take them up on that offer |
| 11:23.54 | Chainsaw | Or, you may want to open another account with another ITSP and route 1800 calls there. |
| 11:25.33 | gavimobile | Chainsaw: I don't see a reason why my itsp won't allow me to call an isreal 1800 number |
| 11:25.44 | gavimobile | my goal is to add 1800 support to my dialplan |
| 11:26.08 | Chainsaw | gavimobile: Because, judging by the x800 prefix, it is supposed to be free or otherwise in a different billing class. |
| 11:26.30 | Chainsaw | gavimobile: More often then not, such numbers simply can't be reached internationally. Consider say... 0845 in the UK. +44 845? Not a chance! |
| 11:26.35 | gavimobile | Chainsaw: so does that mean I should get a local itsp? |
| 11:27.04 | Chainsaw | gavimobile: There are other ways out of this, such as one of those GSM-SIP gateways. You just get a pay as you go SIM and a device like that, and plug that into your Asterisk server. |
| 11:27.54 | gavimobile | but then I need a card |
| 11:27.54 | WIMPy | you good lode land line isn;t good enought for that? |
| 11:28.06 | Chainsaw | WIMPy: Land lines? In Israel? |
| 11:28.24 | WIMPy | Where does the internet come from? |
| 11:28.33 | Chainsaw | WIMPy: The local WiMax tower? |
| 11:29.02 | WIMPy | And that doesn't offer voice? |
| 11:29.12 | Chainsaw | WIMPy: Not at the price you want! |
| 11:32.45 | gavimobile | Chainsaw: better than gsm sip gateways, how about asterisk with bluetooth support |
| 11:32.49 | gavimobile | :-) |
| 11:33.09 | WIMPy | Ther are many ways to link somethign GSM to Asterisk. |
| 11:33.17 | WIMPy | And yes, that's one of them. |
| 11:33.23 | Chainsaw | gavimobile: Not the most... reliable. |
| 11:33.34 | Chainsaw | gavimobile: That'd be worse than an analog line without disconnect supervision. |
| 11:34.08 | WIMPy | Huh? That should work via BT, shouldn;t it? |
| 11:35.13 | gavimobile | really? how bout this http://gigaset.com/us/en/product/GIGASETL700MLINK.html |
| 11:37.49 | Chainsaw | WIMPy: BT doesn't operate in Israel, last I checked. |
| 11:38.00 | WIMPy | If you want it cheap, try with USB dongles. |
| 11:38.09 | WIMPy | BT as in Bluetooth |
| 11:38.28 | gavimobile | Chainsaw: I've tested this device with a digital call center |
| 11:38.39 | gavimobile | not with asterisk cause I don't have a digium card |
| 11:38.52 | WIMPy | BTW: Did someone save the link to that picture with the HUBs full of USB dongles? |
| 11:39.23 | gavimobile | WIMPy: I would like to see that |
| 11:39.37 | gavimobile | why isn't the bluetooh driver no good? |
| 11:39.50 | gavimobile | I can't believe I need to do all this for 1800 support |
| 11:40.04 | gavimobile | what about if I don't mind being charged for it, or will enum work for isreali 1800 numbers |
| 11:42.21 | WIMPy | Does ENUM work in any noteworty way anywhere except Austria? |
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| 11:48.56 | gavimobile | WIMPy: it works in the us |
| 11:49.33 | gavimobile | maybe someone can help me with something else.... I want to make a virtual extention that when my queue rings it forwards the call to an external cell phone |
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| 11:57.54 | gavimobile | folks? |
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| 12:02.57 | kaldemar | gavimobile: nothing virtual there. just add a member that dials your phone. |
| 12:03.35 | gavimobile | kaldemar: are you refering to the member=xxx in queues.conf? |
| 12:03.52 | kaldemar | yes |
| 12:04.27 | gavimobile | kaldemar: that works (member=SIP/1234567@my-trunk) however I want to controll the caller id of the call. also after I receive the call the first time I don't get any more calls |
| 12:04.47 | gavimobile | meaning it will only ring my cellphone for the first caller. any caller after that, it won't ring my celll |
| 12:04.54 | gavimobile | let me try again to verify this |
| 12:06.00 | kresp0 | WIMPy, I've tried that looking for that pic, but no luck: |
| 12:06.01 | kresp0 | site:ibot.rikers.org intitle:"#asterisk" HUB USB jpg gsm |
| 12:06.06 | kaldemar | member=Local/exten@context and in that extension first set the caller id and then dial. |
| 12:06.10 | kresp0 | s/that/this |
| 12:06.41 | kresp0 | maybe if you remember something about the context (date, some keyword..) |
| 12:06.46 | gavimobile | I can confirm that it won't ring |
| 12:06.48 | gavimobile | I have the output |
| 12:07.43 | gavimobile | have a look, it doesn't ring the second cal http://pastebin.com/jnbsxhRF |
| 12:08.38 | WIMPy | kresp0: I have no idea what the filename might have contained. But it must have been a few months ago. |
| 12:09.09 | gavimobile | how do I set the caller id in extension? I don't know how to apply Set(CALLERID(num)=${OUTCID2}) to the member= in queues.conf |
| 12:09.34 | WIMPy | use a local channel |
| 12:10.13 | gavimobile | WIMPy: im lost |
| 12:10.17 | gavimobile | this isn't working for me |
| 12:10.38 | gavimobile | I not have 1 member= in my queues.conf looks like this member=Local/972524353816@rapidvox-trunk |
| 12:10.51 | gavimobile | 972524353816 is the number I want it forwarded to |
| 12:11.01 | gavimobile | rapidvox-trunk is the name of my trunk |
| 12:11.26 | WIMPy | What you write there is the name of a dialplan context. |
| 12:12.15 | gavimobile | WIMPy: I see |
| 12:12.17 | gavimobile | let me try that |
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| 12:12.43 | dandate2 | is there a way to turn skype into a softphone for asterisk? |
| 12:13.05 | dandate2 | to receive incoming calls from queue |
| 12:13.26 | gavimobile | WIMPy: still not working for me |
| 12:13.34 | WIMPy | dandate2: Too lat. far too late. |
| 12:13.38 | kaldemar | gavimobile: the Set() goes into extensions.conf, not queues.conf. |
| 12:13.38 | WIMPy | ~sfa |
| 12:13.53 | kaldemar | gavimobile: < kaldemar> member=Local/exten@context and in that extension first set the caller id and then dial. |
| 12:14.14 | WIMPy | So much too late that infobot doesn't even know it any more. |
| 12:14.48 | gavimobile | so kaldemar exten is not my telephone number, it's a "virtual" extention I want to dial, which in my dialplan id will set the caller id than answer() than dial my number? |
| 12:15.28 | kaldemar | gavimobile: there is nothing virtual about this. the exten is an extension in your dialplan just like any other. |
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| 12:15.53 | kaldemar | gavimobile: and no, it will not answer. i told you the extension should DIAL your cell phone. |
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| 12:16.16 | gavimobile | kaldemar: ok, so say 105 http://pastebin.com/1hGFPXDz |
| 12:16.41 | kaldemar | gavimobile: something like that. |
| 12:16.55 | gavimobile | kaldemar: I've tried this... |
| 12:17.08 | gavimobile | let me try again |
| 12:17.21 | gavimobile | I had this already created but it didn't work when I tried it |
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| 12:18.04 | gavimobile | nice |
| 12:18.07 | gavimobile | worked on round one |
| 12:18.09 | gavimobile | now round 2 |
| 12:18.35 | gavimobile | bingo! |
| 12:18.36 | gavimobile | cool thanks |
| 12:18.39 | gavimobile | got that sorted |
| 12:20.20 | gavimobile | next question.. to make a confrence call, say I want to call company a 1234567 and company b 7654321 (both external calls). I first call company a, than I put him on hold and call company b. than I join them. my question is how many channels are required, 2 right? |
| 12:21.18 | dandate2 | as far as the outbound provider goes its 2 channels, but your asterisk will be using like 3 or 4 |
| 12:22.45 | beebeeep | unless you're using meetme asterisk will handle 4 channels - 2 outbound and 2 channels to your phone |
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| 12:24.57 | gavimobile | dandate2: ok, so if I have a itsp which promises me with the did trunk I purchased that I have 2 channels, and with another itsp, I am promised to have 1 channel. is there a way I can verify how many total channels my asterisk server has for outbound? |
| 12:25.24 | gavimobile | I should have 3 channels |
| 12:25.35 | gavimobile | I've tried sip show channels |
| 12:25.45 | gavimobile | I don't think that's what I want |
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| 12:27.00 | gavimobile | in short I would like to know the total amount of channels I have without guessing |
| 12:27.03 | gavimobile | can I do this from the cli |
| 12:27.03 | gavimobile | ?" |
| 12:27.08 | aliasbex | can so tell me how to increase some counter to bypass the infinite loop detection in a macro? |
| 12:27.32 | aliasbex | (generated macro content has multiple identical lines of code within the generated macro) |
| 12:28.52 | aliasbex | otherwise a plan is to use the while statement in dialplan..... but the counter setup would be easier if possible. |
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| 12:29.25 | aliasbex | gavimobile: "core show channels"? |
| 12:29.46 | gavimobile | aliasbex: Channel Location State Application(Data) |
| 12:29.46 | gavimobile | 0 active channels |
| 12:29.46 | gavimobile | 0 active calls |
| 12:29.46 | gavimobile | 4 calls processed |
| 12:29.51 | gavimobile | does that mean I have 4 channels total? |
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| 12:30.49 | aliasbex | a channel is generated in runtime. used in dialplan etc. there is only a restriction at your providers side. (SIP) |
| 12:31.32 | gavimobile | so how can I find out how many concurrent calls I can have at 1 time? |
| 12:32.04 | aliasbex | with "core show channels" youd see those with current format "some codec" |
| 12:32.28 | aliasbex | otherwise if the sip dialog is not really a call then there is "nothin" in "format" column |
| 12:32.57 | gavimobile | I don't have a format coloum |
| 12:33.13 | gavimobile | as shown above I have channel location state and application |
| 12:36.05 | aliasbex | sip show channels |
| 12:36.20 | aliasbex | my error |
| 12:37.26 | gavimobile | aliasbex: sip show channel does not show me that |
| 12:37.41 | gavimobile | 0 active SIP dialogs |
| 12:38.02 | aliasbex | and while making a call? |
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| 12:46.47 | gavimobile | aliasbex: when I make a call it shows stuff there, but I want to know how many concurent calls I can have at one time max |
| 12:46.51 | gavimobile | is there a way to do this? |
| 12:47.43 | carrar | yes |
| 12:48.35 | carrar | check out GROUP, GROUP_COUNT |
| 12:49.13 | carrar | set it as calls come in so you can keep a running total |
| 12:49.40 | carrar | perhaps GROUP(in) for inbound calls |
| 12:49.46 | carrar | and GROUP(out) for outbound calls |
| 12:50.20 | carrar | GROUP_COUNT(in) will then have your total inbound calls |
| 12:50.28 | carrar | etc.. |
| 12:52.36 | tompaw | Guys, I need a top quality US termination with CLI - can you please recommend someone? It's not for wholesale, but phone center termination. |
| 12:52.56 | carrar | ~itsp |
| 12:52.56 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 12:53.18 | tompaw | ~itsplist-us |
| 12:53.18 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 12:53.40 | tompaw | Anyone can recommend some of these from their personal experience? |
| 12:54.55 | [TK]D-Fender | "core show channels" will tell you how many channels you have up.... |
| 12:55.36 | carrar | thats too simple! |
| 12:55.37 | [TK]D-Fender | tompaw, voip.ms , flowroute, les.net |
| 12:58.02 | tompaw | [TK]D-Fender: thanks! |
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| 13:00.56 | tompaw | voip.ms looks good, but they're doing A-Z, I was hoping for someone specialised in US with their own PSTN interconnections |
| 13:01.58 | carrar | Verizon does SIP |
| 13:02.31 | tompaw | carrar: and I believe I even have an exchange with them in London! thanks!! |
| 13:02.32 | gavimobile | wow thoese rates are expensive |
| 13:02.38 | gavimobile | I just chaned all thoese sites |
| 13:02.59 | carrar | might look at siproute.com |
| 13:02.59 | gavimobile | didlogic and rapidvox seem to be cheaper for calls to my country |
| 13:03.24 | gavimobile | carrar: that link doesn't work |
| 13:03.46 | carrar | yeah I don't think thats their host name |
| 13:04.02 | carrar | siproutes.com |
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| 13:18.01 | LemensTS | While in a call, how can i press 1 on my phone to make it play a pre recorded message? |
| 13:18.45 | LemensTS | I was hoping to do this via dialplan, I've done it in phpagi before. |
| 13:26.01 | blitzrage | LemensTS: check the Asterisk Cookbook on ofps.oreilly.com for a solution I wrote |
| 13:26.13 | blitzrage | something like Injecting Audio into the call |
| 13:26.27 | blitzrage | that is dynamic data, but you probably just want an [application_map] in features.conf |
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| 13:29.33 | LemensTS | blitzrage: thank you, that is cool! |
| 13:30.33 | blitzrage | LemensTS: check leifmadsen.com for the PDF and I think I linked to the astricon video from last year Cooking With Asterisk |
| 13:30.39 | blitzrage | where I go through the solution |
| 13:31.15 | *** join/#asterisk TheCompWiz (~TheCompWi@63.214.236.169) |
| 13:31.16 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 13:31.33 | TheCompWiz | anyone know a good way to get asterisk to re-send MWI notifications anytime a peer reconnects? |
| 13:32.01 | TheCompWiz | ... or modify the time between re-transmissions? |
| 13:32.32 | *** join/#asterisk alsuren (~dlaban@80.169.133.251) |
| 13:33.19 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:33.20 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:34.19 | blitzrage | TheCompWiz: just enable polling and then it'll update periodically |
| 13:34.22 | blitzrage | voicemail.conf |
| 13:34.41 | blitzrage | once the peer registers, then the MWI will be updated for the peers |
| 13:36.37 | alsuren | is currently pegging your svn server thanks to eclipse's default for checking out all branches. Poke me if you want me to stop |
| 13:37.00 | blitzrage | alsuren: ya that's not a problem :) |
| 13:37.10 | blitzrage | you're not the only one accessing it |
| 13:39.08 | alsuren | would be using git if it were an option </plug> |
| 13:40.41 | edge | With the Cisco SPA(502g) phone. When a user is part of a call queue. That user dials a number into their phone without picking up the receiever. If that user hits dial at or about the time when a call enters the queue, then hitting dial on the phone picks up the call and blasts digits into that incoming caller's ear. How can I stop this behavior? |
| 13:41.26 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 13:44.53 | blitzrage | alsuren: that conversation has come up a few times -- doesn't look to be in the cards |
| 13:45.05 | blitzrage | alsuren: it is available on git mirrors though |
| 13:48.13 | *** join/#asterisk alsuren_ (~dlaban@80.169.133.251) |
| 13:48.33 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 13:55.51 | edge | Does anybody have a general direction i could may look? I was thinking it had something to do with the way the phone was picking the line up, but i'm not sure how to approach this issue |
| 13:56.05 | TheCompWiz | East. That's a good direction. |
| 13:56.28 | [TK]D-Fender | edge, Go read your phone's admin guide. I would bet that this behaviour is not optional however |
| 13:56.46 | edge | [TK]D-Fender: So its more than likely with the phone and not the queue or something |
| 14:01.25 | alsuren_ | just FYI. I'm trying to get ODBC storage to store multiple formats in the database, but looking at the code, it seems that there is a do{}while(0) that is preventing it from happening |
| 14:02.02 | *** join/#asterisk bchia (~Adium@nat/digium/x-bhcwofngnbdtrwyy) |
| 14:02.29 | alsuren_ | current candidates for blame are markster and tilghman. Will update you when I have a better idea |
| 14:02.54 | TheCompWiz | weeeeeeee... everyone loves the blame-game :D |
| 14:03.01 | alsuren_ | also: where should I submit patches when I have it fixed? |
| 14:04.03 | Chainsaw | alsuren: The JIRA instance please, http://issues.asterisk.org/jira/ |
| 14:06.07 | [TK]D-Fender | edge, This has nothing to do with Asterisk |
| 14:07.09 | *** join/#asterisk tm1000 (tm1000@2600:3c01::f03c:91ff:fe93:fa45) |
| 14:07.57 | Corydon76-home | alsuren: ODBC has _never_ supported storing multiple formats |
| 14:08.33 | edge | [TK]D-Fender: does it have to do with how the phone uses SIP to dial? I'm guessing other phones don't do this? |
| 14:09.09 | alsuren | Corydon76-home: this is what I'm starting to suspect |
| 14:09.29 | alsuren | imap storage is documented as not supporting it, but not odbc |
| 14:10.21 | [TK]D-Fender | edge, the phone is responsible for when the phone sends a SIP call. Or accepts one. Or shoves digits it has saved up onto a call it's answering when you were doing that before the call came in. |
| 14:10.49 | edge | [TK]D-Fender: Thanks for helping me clear up where i should be looking |
| 14:11.21 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:11.37 | TheCompWiz | edge: It's a simple timing issue. If Phone A rings... and user A picks up the phone and dials digits to make a call... Caller A is going to be connected to the line and hear the digits. You may want to consider setting up a wrap-up time... or let agents pause themselves so that they won't get calls when trying to make a call... or whatever to suit your needs. |
| 14:13.07 | edge | TheCompWiz: The user is question, actually dials without picking up the phone which stores digits on the display, and then his "dial" on the soft button and it picks up an incoming call and dials into it. I'm not sure how I'd work around this, short of telling them not to do it. Intead pick up the phone and dial |
| 14:15.12 | TheCompWiz | that's something you'll have to work out. |
| 14:15.32 | TheCompWiz | asterisk doesn't control how your endpoint behaves. |
| 14:15.54 | edge | TheCompWiz: Now I know, i can save time not troubleshooting asterisk |
| 14:16.06 | TheCompWiz | asking never hurts. |
| 14:17.25 | edge | TheCompWiz: and i learn a lot here. |
| 14:18.27 | TheCompWiz | I wonder if there is a way to prevent asterisk from translating your key-presses into DTMF. That would be a bit of a dirty-hack... but it would keep your customers from being blasted by DTMF. |
| 14:20.55 | [TK]D-Fender | TheCompWiz, there isn't |
| 14:21.04 | edge | TheCompWiz: its also that the user is waiting to hear ringing (like they're actually calling somewhere) not a "uh..hello" |
| 14:21.24 | TheCompWiz | [TK]D-Fender: no way to stop processing sip-info packets? ... darn. |
| 14:21.36 | TheCompWiz | edge: yeah... |
| 14:21.59 | [TK]D-Fender | TheCompWiz, No way to have Asterisk be PSYCHIC and say "oh I expect this guy to be a twit for THIS call" |
| 14:22.13 | [TK]D-Fender | TheCompWiz, Or are you goign to break all DTMF for a phone? |
| 14:22.28 | [TK]D-Fender | The phone is dumb. Adjust it or replace it |
| 14:22.43 | TheCompWiz | [TK]D-Fender: not entirelly... perhaps just stop processing for calls in a queue. |
| 14:23.06 | TheCompWiz | I don't think it's the phones... just user-error. |
| 14:23.23 | TheCompWiz | if your phone is ringing... don't blindly try and make a call. |
| 14:23.33 | [TK]D-Fender | it's both. The phone for acing that way, and the user for not dealing with it |
| 14:24.38 | edge | TheCompWiz: it isn't rining when they hit dial, its just a unfortunate coincidence that one comes in when they hit dial |
| 14:26.08 | TheCompWiz | that should be a fairly rare situation... |
| 14:26.25 | TheCompWiz | unless your phone *is* ringing... but your ring-tone has a bit of silence before it actually plays some tone. |
| 14:27.07 | edge | TheCompWiz: i should check the ring tone the user is using. This user reports this kind of event more than 3 times a week. which means i "should" fix it |
| 14:27.59 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 14:29.27 | TheCompWiz | lol... sometimes... you can't fix stupid. |
| 14:30.05 | [TK]D-Fender | "Don't tase me bro!" <- behavioural modification |
| 14:31.00 | TheCompWiz | the taser didn't make him less-stupid... just reminded him that it's easier to do what you're told than getting tased. |
| 14:33.18 | edge | [TK]D-Fender: TheCompWiz : I opened a ticket with Cisco to see if i can get some assistance fixing it. |
| 14:33.19 | [TK]D-Fender | "Stupid" that is conditioned to the point where the body won't let the brain commit stupid acts = fixed. |
| 14:33.34 | [TK]D-Fender | I don't care what goes on in their head so long as it stays in there. |
| 14:33.37 | TheCompWiz | LOL good point [TK]D-Fender. |
| 14:33.58 | TheCompWiz | Another example of stupid... http://autos.aol.com/gallery/unsafe-ford-pinto-minnesota/?ncid=txtlnkusauto00000020 |
| 14:34.06 | [TK]D-Fender | edge, And did you go over every inch of the administrators guide first? |
| 14:35.00 | [TK]D-Fender | TheCompWiz, that cr self-regulates stupidity by natural selection. I fail to see the "problem" ;) |
| 14:35.02 | edge | [TK]D-Fender: I read it once, and went through the SIP configuration pages for the device and its configuration. I don't see anything that jumps out at me. There are a few more things inside the configure i'm not sure of (old RFC compatibility modes) so I've a bit more work to do |
| 14:35.03 | [TK]D-Fender | car* |
| 14:37.39 | TheCompWiz | ... Darwin sometimes doesn't work fast enough. |
| 14:39.06 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
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| 14:56.28 | *** join/#asterisk Mozillion (~paul@37.72.100.21) |
| 14:56.35 | *** join/#asterisk fireman_biff (~biff@65.48.222.203) |
| 14:57.42 | Mozillion | hey all.. I've been struggling with FIELDNUM.. it works for me with example=ex-amp-le and ${FIELDNUM(example,-,amp)}... but not for commas as suggested by the docs.. i.e. example=ex,amp,le and ${FIELDNUM(example,\054,amp)} (always returns 0) |
| 14:57.46 | Mozillion | am I overlooking something? |
| 14:59.57 | fireman_biff | Hi, with a PRI I can use the "pri set debug" commands to troubleshoot when calls are failing. Is there a similar command for a non-PRI T1? (asterisk 1.8, no alarms) |
| 15:00.41 | Mozillion | hm, same actually seems to hold for ${REPLACE(example,\054,-)} |
| 15:01.26 | TheCompWiz | use \44 instead of \054. I think it's hex ... not octal. |
| 15:02.05 | Mozillion | hex for comma is \x2C.. also doesn't work |
| 15:02.13 | Mozillion | both are suggested in the docs |
| 15:02.32 | Mozillion | I found out that "\\054" instead of \054 works! |
| 15:02.48 | TheCompWiz | doh 44 = deciaml /facepalm. |
| 15:03.23 | wdoekes | Mozillion: what's the entire line? |
| 15:03.47 | Mozillion | ok, for full context.. I'm testing membership of a queue, so: |
| 15:03.59 | Mozillion | atm I have: exten => *2000,n,GotoIf(${FIELDNUM(QUEUE_MEMBER_LIST(default),"\\054",member)}?leave:join) |
| 15:05.44 | wdoekes | it's likely that there is some double decoding going on. consider it a feature |
| 15:06.41 | Mozillion | that always returns 0 and thus goes to "join" |
| 15:09.55 | *** join/#asterisk krotos (~d3v1l@95.237.101.175) |
| 15:10.09 | krotos | hi all |
| 15:10.56 | krotos | someone here have used a Cisco 1751v or similar with ios 12.4 |
| 15:11.16 | Mozillion | ok now it starts to drive me insane.. why does it work with the example and not with that line |
| 15:11.46 | krotos | and try to register to asterisk? The registration is ok, but when i call the cisco i've got "Invalid Host" |
| 15:13.02 | *** join/#asterisk navaismo (~navaismo@189.144.249.74) |
| 15:16.25 | Katty | is trying not to kill people this morning. |
| 15:17.19 | Mozillion | because it should be ${member} instead of member.. kill me now |
| 15:17.28 | Mozillion | oops, poor choice of words :P |
| 15:17.56 | TheCompWiz | snuggles up to a warm Katty... and brings hot-chocolate and cookies. |
| 15:20.49 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v012-133.mobile.uci.edu) |
| 15:21.40 | chuckf | Katty: if you kill them today, they won't bother you tomorrow |
| 15:24.14 | Katty | theoretically |
| 15:34.10 | [TK]D-Fender | Katty, #zombieapocalypse |
| 15:38.36 | Katty | i'm in too many channels as it is. |
| 15:41.56 | [TK]D-Fender | Was agging that Twitter-wise |
| 15:42.10 | [TK]D-Fender | not IRC though that confusion point dawned on me quickly.. |
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| 15:45.36 | davlefouAMD | hi, my sflphone have an anchorage option, do you know what is it? |
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| 16:01.12 | shadebob | Hi. Astdb have a limitation size ? |
| 16:06.06 | *** part/#asterisk fireman_biff (~biff@65.48.222.203) |
| 16:07.23 | shadebob | I make some changes and the size is always 339968 oct |
| 16:14.58 | alsuren | Chainsaw: thanks for the link. I created an issue to track my progress at: https://issues.asterisk.org/jira/browse/ASTERISK-20400 |
| 16:16.41 | Chainsaw | alsuren: Cool. Hope you get some answers. |
| 16:21.19 | *** join/#asterisk thebitguru (~thebitgur@97-87-12-42.dhcp.mdsn.wi.charter.com) |
| 16:22.24 | thebitguru | Hi, does anyone have any recommendation for a hosted Asterisk system? I am looking for a provider where I can reliably host an asterisk system. I have been unsuccessful with Linode. |
| 16:22.31 | *** join/#asterisk _Corey_ (~chatzilla@64.215.11.114) |
| 16:22.42 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 16:27.15 | blitzrage | thebitguru: are you building the system, or can you use an existing infrastructure? |
| 16:27.23 | blitzrage | thebitguru: www.coredial.com does hosted PBX systems for example |
| 16:27.24 | *** join/#asterisk MLNoah (~MLNoah@noc.metalink.net) |
| 16:27.41 | blitzrage | thebitguru: otherwise you can just use Amazon EC2 instances (need at least m1.medium in my experience) |
| 16:29.45 | MLNoah | Does anyone have experience with an XMPP server that works well with Asterisk that they would recommend? |
| 16:30.01 | TheCompWiz | openfire works for me. |
| 16:30.02 | thebitguru | blitzrage: I would prefer a managed system, but can maintain if necessary. Checking coredial.com |
| 16:30.09 | TheCompWiz | it all depends on what you're trying to accomplish. |
| 16:30.29 | blitzrage | thebitguru: ya coredial.com is a managed system -- you just use the web interface provided to you |
| 16:31.12 | blitzrage | MLNoah: I know at least one person who uses Tigase in production |
| 16:31.59 | MLNoah | CompWiz > I'm trying to get a system by which my users can send an XMPP message to Asterisk and interact with our timeclock (we already have a IVR interface to the timeclock on the system). I'm running an OpenFire 3.6.4 server, with Asterisk 11-SVN on the phone system, and my test box is crashing like mad while I try to send messages to it. |
| 16:32.00 | *** join/#asterisk twodogs (~twodogs@telok.vitiate.me) |
| 16:32.41 | TheCompWiz | what exactly is crashing... and what are you doing? |
| 16:32.44 | blitzrage | MLNoah: use Tigase because it's the only one I know of that has really been tested with Asterisk |
| 16:32.47 | MLNoah | I've messed with every setting I can think of, but not much luck. I also tried setting up a separate OpenFire 3.7.1 server to test, no users except me and the Asterisk test box, still bombs out. |
| 16:32.59 | thebitguru | blitzrage: do you have any experience with a hosted pbx on Amazon? |
| 16:33.05 | blitzrage | thebitguru: yes |
| 16:33.18 | thebitguru | how was your experience? |
| 16:33.40 | blitzrage | works well enough |
| 16:33.44 | MLNoah | Basic flow is: XMPP user sends message to *, * processes message to determine how to handle it (is the user requesting the timeclock, etc), sends a message back to XMPP user to request authentication, and then does the requested step post-authentication. |
| 16:34.03 | thebitguru | anyone here running Asterisk on Linode? |
| 16:34.14 | drmessano | I've used ejabberd, but I had Asterisk stability issues at times. I couldn't narrow down which end was the problem |
| 16:34.36 | TheCompWiz | MLNoah: you still haven't said *what* is crashing... |
| 16:34.40 | drmessano | MLNoah: That should be easy to do with a stable system |
| 16:34.45 | MLNoah | two places I'm seeing seg-faults are either when I open a window with my chat client to start sending the message (Asterisk receives a (null) message, which my dialplan filters and discards) -- or right after Asterisk sends the request for authentication and starts listening for the response. |
| 16:34.53 | MLNoah | Asterisk itself completely bombs out and restarts. |
| 16:35.05 | MLNoah | Drops any active anything on the system - calls, etc. Dumps out a core. file. |
| 16:35.32 | TheCompWiz | sounds like asterisk is the problem... not your choice of xmpp server. |
| 16:35.55 | TheCompWiz | (or rather... your build/configuration of asterisk) |
| 16:36.27 | MLNoah | any suggestions on what to troubleshoot in my build/configuration? |
| 16:36.44 | TheCompWiz | look at logs... see what errors are thrown up... |
| 16:37.02 | TheCompWiz | monitor the situation & see if you can get some indications as to what the system looks like before dying... |
| 16:37.24 | TheCompWiz | what sort of data is sent to/from wherever when it crashes... |
| 16:37.29 | TheCompWiz | etc... |
| 16:40.07 | MLNoah | the biggest problem i'm having is that the crashes seem to be somewhat intermittent -- i'll have stretches where doing anything at all with XMPP will crash asterisk (even just opening a window to get ready to send a message to asterisk). but then i'll have stretches where the system works as intended and I can send 10-20 messages without a crash. |
| 16:40.31 | MLNoah | not seeing anything different in verbose/debug console output in the cases where it is crashing versus the cases where it isn't. |
| 16:40.47 | TheCompWiz | the logs should still persist. Try adding noops & get some additional debug info where possible. |
| 16:41.20 | TheCompWiz | i.e. what was recieved... what was sent... etc... |
| 16:41.34 | TheCompWiz | what version of asterisk? |
| 16:41.46 | MLNoah | SVN-branch-11-r372473 |
| 16:42.04 | MLNoah | which is the up-to-date as of last friday version. |
| 16:42.09 | *** join/#asterisk frawd (~francois@221.red-80-28-139.adsl.static.ccgg.telefonica.net) |
| 16:42.38 | TheCompWiz | sighs. Bleeding edge is bleeding. |
| 16:44.22 | MLNoah | yeah. one of the other things i'm doing with the system has been greatly improved by the hangup handler changes in 11, so I don't really want to go back if I can avoid it. though I guess if 10 handles XMPP better, i'll have to look at it. |
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| 17:18.57 | *** join/#asterisk Unchained (~Unchained@surft.nackt-im-garten.de) |
| 17:19.02 | Unchained | good evening |
| 17:20.43 | *** join/#asterisk wonderworld (~ww@dsdf-4db5e9a2.pool.mediaWays.net) |
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| 17:36.25 | Unchained | I've a small question about the dialplan |
| 17:36.39 | Unchained | xten => _ZX.,1,Dial(SIP/prefix${EXTEN}@sipaccount) |
| 17:36.46 | Unchained | exten |
| 17:36.47 | Unchained | sorry |
| 17:37.17 | Unchained | thats part of my dialplan. and my wish is to put that mixmonitor before. |
| 17:37.43 | Unchained | i'm a little confused about the pattern. |
| 17:38.03 | Unchained | can someone explain/help me with the mixmonitor command? |
| 17:38.31 | Unchained | exten => ????,n,MixMonitor(${MONITOR_FILENAME}.wav,b) |
| 17:39.23 | Unchained | there are to sip commands for outbound calls |
| 17:39.30 | Unchained | exten => _ZX.,1,Dial(SIP/prefix${EXTEN}@sipaccount) |
| 17:40.00 | Unchained | exten => _0Z.,1,Dial(SIP/${EXTEN}@sipaccount) |
| 17:40.14 | *** join/#asterisk diatonic (~diatonic_@mail.clearwater-research.com) |
| 17:40.27 | Unchained | so, which is the correct pattern to monitor these bose sip commands? |
| 17:42.12 | pabelanger | ~book |
| 17:42.12 | 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 |
| 17:42.16 | pabelanger | Unchained: ^ |
| 17:42.21 | pabelanger | read the chapter about dialplans |
| 17:42.32 | pabelanger | it will help you understand how it works |
| 17:43.18 | [TK]D-Fender | It isn't the pattern... its the PRIORITY.... |
| 17:43.31 | diatonic | I have a strange issue where inbound calls are failing fram a sip provider when outbound calls are in progress through that provider. Asterisk 1.8.5.0 behind pfsense NAT. With sip debug on the peer I don't even see the call try to come in if outbound calls are in progress. |
| 17:43.47 | diatonic | Not sure what to look at. |
| 17:43.58 | pabelanger | sounds like a route issue |
| 17:44.03 | pabelanger | setup tcpdump and see if you can see traffic |
| 17:45.08 | diatonic | It's weird that inbound only fails when outbound is in progress |
| 17:46.25 | pabelanger | no |
| 17:46.42 | pabelanger | 2 different paths via IP |
| 17:46.49 | pabelanger | you know the IP address of the server for outbound |
| 17:47.01 | pabelanger | the ITSP does not know your IP address and calls are failing |
| 17:47.06 | pabelanger | with inbound |
| 17:48.18 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 17:48.24 | raden | is it possible to change ringer on polycoms ? |
| 17:48.35 | pabelanger | yes |
| 17:53.37 | [TK]D-Fender | raden, Clearly. |
| 17:54.42 | raden | hmmm |
| 17:54.50 | raden | i dont care for these 301's |
| 17:55.04 | Unchained | pabelanger: thanks, I have it now. _X. was the right pattern |
| 17:55.15 | [TK]D-Fender | Nice phones. Solid, can juggle 10 calls at a time w/ 3-way, splits, etc |
| 17:55.23 | Unchained | that I have to set the priority, that was clear for me. |
| 17:55.29 | [TK]D-Fender | Unchained, there was no need for a new pattern |
| 17:55.40 | Unchained | why? |
| 17:56.15 | [TK]D-Fender | You wanted it to do 1 more thing to what you already had. That doesn't mean "new pattern needed", that says "one more PRIORITY to add for them |
| 17:56.50 | Unchained | exten => _x.,1,Set(MONITOR_FILENAME=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${EXTEN}-outbound) |
| 17:56.50 | Unchained | exten => _X.,2,MixMonitor(${MONITOR_FILENAME}.wav,b) |
| 17:56.50 | Unchained | ;local 44444 lokales Ortsnetz; |
| 17:56.50 | Unchained | exten => _ZX.,3,Dial(SIP/44444${EXTEN}@sip) |
| 17:56.50 | Unchained | ;DE 0xxx |
| 17:56.51 | Unchained | exten => _0Z.,3,Dial(SIP/${EXTEN}@sip) |
| 17:56.57 | Unchained | thats what I did now |
| 17:57.30 | pabelanger | kinda ugly |
| 17:57.36 | Unchained | :) |
| 17:57.45 | [TK]D-Fender | those patterns are not the same. do not assume a safew overlap |
| 17:57.46 | Unchained | because? |
| 17:57.50 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:58.18 | [TK]D-Fender | You are allowing almost ALL numbers to hit those first priority... not just ones hitting the other patterns as * continues to try to process priorities |
| 17:59.27 | Unchained | that was my wish, to record all outbound calls (pri 1 +2 ) |
| 18:00.06 | Unchained | next command ist to add the missing prefix |
| 18:00.09 | [TK]D-Fender | then make proper priorities for each of your patterns. |
| 18:06.22 | Nephfl | anybody know if Godaddy is having major issues atm? |
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| 18:09.46 | rillian | Nephfl: yes |
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| 18:15.20 | *** part/#asterisk LemensTS (~Administr@h20.246.23.98.dynamic.ip.windstream.net) |
| 18:16.27 | Nephfl | Would be nice if they updated their system alerts page |
| 18:17.40 | *** part/#asterisk rillian (~giles@mf4-xiph.osuosl.org) |
| 18:50.05 | jpsharp | Yeah, they're busy being Godaddy. |
| 18:50.32 | dandate2 | i'm glad i moved my stuff off there recently |
| 18:51.45 | dandate2 | i still have my webmail with them though, will this result in incomming emails being lost int he void? |
| 18:51.59 | jaytee | hmmm, interesting. I've been dealing with dropped calls through Flowroute due to it being unreachable on our system plus several clients here in Indiana. Flowroute was still routing calls to the failover numbers if an account was unreachable. Thought it might be Comcast. |
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| 19:09.44 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 19:09.50 | eXcAliBuR | Hello all my friends :] |
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| 19:10.28 | eXcAliBuR | I have a digium d50 phone, and 2 analog lines, I'd like to know if it's capable to showing when a line is in use? |
| 19:12.04 | [TK]D-Fender | eXcAliBuR, Set up a hint in your dialplan. Watch it. |
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| 19:15.36 | eXcAliBuR | TK can you tell me what term to search for? |
| 19:15.38 | eXcAliBuR | hints? |
| 19:15.40 | eXcAliBuR | maybe |
| 19:16.23 | [TK]D-Fender | that is what I told you you needed to set |
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| 19:20.50 | eXcAliBuR | TK..ummm |
| 19:21.05 | eXcAliBuR | can i maybe borrow your brain for a little bit??? I promise to give it right back :] |
| 19:23.26 | [TK]D-Fender | eXcAliBuR, "asterisk dialplan hints" <--- 3 magic words and Google will point you places. |
| 19:28.42 | eXcAliBuR | thats what i type ;D |
| 19:28.50 | eXcAliBuR | typed* |
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| 19:36.12 | eXcAliBuR | I need to find my big book :/ |
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| 19:49.39 | rbowles | anyone seeing godaddy effects on Asterisk dns lookups? I'm hanging on getonsip.com and ipcomms.net |
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| 19:51.35 | jaytee | rbowles, I've been getting lagged and unreachable due to latency for my Flowroute ITSP on our system and several clients I've setup. |
| 19:52.06 | jaytee | seems to have quieted down since about 2:45 EDT |
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| 20:17.06 | *** join/#asterisk CrazyTux[m] (~Brandon@wsip-68-15-80-34.oc.oc.cox.net) |
| 20:17.15 | CrazyTux[m] | Anyone know why Asterisk would strip Route/Record Route headers? |
| 20:17.20 | CrazyTux[m] | on reply (ACK) |
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| 22:09.36 | navaismo | Using Fax for asterisk to send multipages pds I need to open one channel for each page, or wait for end the first page then send another or so on? |
| 22:13.09 | CrazyTux[m] | navaismo: one channel, one stream ? |
| 22:14.31 | navaismo | CrazyTux[m], sorry, I dont understand |
| 22:15.54 | CrazyTux[m] | navaismo: why would you open up multiple channels for a fax? The line will already be in use / busy |
| 22:16.09 | CrazyTux[m] | for an *outbound* |
| 22:16.23 | WIMPy | Maybe you could send the pages in parrallel :-) |
| 22:19.26 | navaismo | ok I dont have idea on how to do that i use the normal behavior exten => send,n,SendFAX(/home/dwayne/faxout/${FAXFILE},d) |
| 22:19.28 | navaismo | <PROTECTED> |
| 22:19.33 | navaismo | will see |
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| 23:45.24 | jpsharp | CrazyTux[m]: Create multipage TIFF files. That's the only way to do it other than dial, send 1 page, hangup, wait, dial ad nauseum |
| 23:46.16 | CrazyTux[m] | jpsharp: ah it wasn't me asking - navaismo, but yea I was thinking the same thing (to either catenate the "documents") |
| 23:46.47 | CrazyTux[m] | or multiple calls, since a modem will be busy by default when communicating with it for fax |
| 23:46.57 | *** part/#asterisk rbowles (~bb@75-147-63-201-NewEngland.hfc.comcastbusiness.net) |
| 23:50.12 | dandate2 | why do people always say that skype proprietary codec sounds better than ulaw? |
| 23:52.46 | [TK]D-Fender | Have you considered the possibility that it IS? |
| 23:53.55 | dandate2 | is anyone working on an open source codec that would also be better than ulaw? |
| 23:55.48 | [TK]D-Fender | You seem to have an elevated opinion on ULAW. I think your sense of scal is skewed |
| 23:55.58 | [TK]D-Fender | scale* |
| 23:56.45 | jpsharp | I'm not sure why people want better than ULAW. Do they want CD quality audio for their phone calls? |
| 23:58.59 | [TK]D-Fender | "CD Quality" is "good"? I've heard Eddie Van Halen beg to differ about 2 decades ago.... |
| 23:59.32 | WIMPy | G.722 is good enough for me for phone calls. |
| 23:59.44 | WIMPy | And I do agree on CD quality. |