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| 07:01.25 | benngard | hmm, how do u remove/unset a global variable from fialplan? |
| 07:03.04 | benngard | Set(x=) will do the trick? |
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| 07:17.13 | benngard | Set(GLOBAL(x)=) did the trcik for me :) |
| 07:24.26 | p3nguin | I guess if you used clearglobalvars=yes you wouldn't have that problem, huh? |
| 07:25.12 | benngard | maybe not |
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| 07:26.40 | benngard | but anyway, ${BLINDTRANSFER} works over qsig/ooh323 now :) |
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| 09:30.09 | sigius | e |
| 09:31.40 | sigius | Question: I would like to play some audio at the start of a SIP call to both caller and callee (the same audio to both at the same time). How would I do that ? |
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| 10:03.39 | antoasla | Hello all, I have created some extension files in order to create a 2 computer asterisk based network (1 pc asterisk server and 1 caller from laptop with windows xp). For the pc that runs asterisk i use twinkle softphone and i register it successfully but when i try to register a softphone from my laptop i absolutely cant. Here are my conf files http://pastebin.ubuntu.com/588587/ plz take a look and any ideas are welcome, thank you! |
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| 10:48.36 | antoasla | does anyone knows what is the diference running the asterisk -r and asterisk -gc? |
| 11:19.12 | benngard | antoasla: is it phone2 that cant register? |
| 11:36.51 | antoasla | yes |
| 11:45.15 | benngard | what softphone are u using? |
| 11:52.43 | antoasla | twinkle in my fedora and tried with several in the windows xp laptop |
| 11:53.41 | benngard | in the softphone, what "username" are u trying to register? |
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| 11:54.48 | benngard | phone2 or tel2? |
| 11:58.32 | antoasla | the problem is not in the configuration. There is problem with the network propably cause both computers are in the same home network. Now i use these configurations they are even simpler http://pastebin.ubuntu-gr.org/m2ced4f3c and i use user:2001 and pass 4321 |
| 11:59.47 | benngard | in asterisk cli, do u se any register attempts? |
| 12:00.18 | antoasla | No nothing. |
| 12:01.15 | antoasla | these accounts are registrable when i do it from the server pc and twinkle softphone but nothing hapens when i use the laptop |
| 12:01.42 | benngard | can the laptop reach "ping" the server? |
| 12:02.47 | benngard | and u have typed "sip set debug on" from clid? |
| 12:02.53 | benngard | cli |
| 12:03.02 | antoasla | yes ping is ok |
| 12:03.41 | benngard | and u typed the correct address in softphone, i mean address of sip server? |
| 12:04.51 | antoasla | i havent done the debug you mentioned before. the configurations in softphones are correct yes |
| 12:05.14 | benngard | are u using ip address or dns name? |
| 12:05.20 | benngard | in softphone i mean |
| 12:05.22 | antoasla | ip only |
| 12:05.33 | benngard | then i cant be a dns error |
| 12:05.49 | benngard | type the debug command i sent u |
| 12:06.09 | antoasla | i did |
| 12:06.37 | benngard | still nothing in cli? |
| 12:07.23 | benngard | hang on maby i got an idea |
| 12:07.50 | antoasla | maybe you want to try connect to me? |
| 12:08.03 | benngard | sec |
| 12:08.15 | benngard | what do u have in [general] in sip.conf |
| 12:08.38 | antoasla | :/ nothing |
| 12:08.39 | benngard | allowguest=no |
| 12:10.51 | sigius | Question: I would like to play some audio at the start of a SIP call to both caller and callee (the same audio to both at the same time). How would I do that ? |
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| 16:31.15 | voila | hello guys . |
| 16:32.14 | voila | well i have landline phone , and asterisk on my ubuntu , is it possible to divert all calls on my landline phone to asterisk on my computer ?? |
| 16:34.35 | wdoekes2 | sure.. but you'll need a card in your computer to connect the landline to |
| 16:35.45 | voila | wdoekes2: hi .. just one thing , do you understand what i meant by landline (old school phones wired network) ?? |
| 16:36.04 | voila | i said this , becuase i m home user , creating some application . |
| 16:36.51 | wdoekes2 | google says this: http://onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 16:38.02 | voila | okies .. wdoekes2: |
| 16:40.42 | wdoekes2 | btw, zap and zaptel is now known as DAHDI |
| 16:41.15 | wdoekes2 | and there my knowledge on the subject of analog phone lines ends |
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| 17:42.55 | *** join/#asterisk Aut0Exec (~Joe@65.75.65.130) |
| 17:43.12 | Aut0Exec | hi guys... is gtalk able to tie in multiple accounts? |
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| 17:50.45 | voila | Aut0Exec: by default nope , |
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| 17:53.05 | voila | in asterisk , [] , holds some special meaning ?? may be like calculation ?? |
| 17:56.29 | kaldemar | $[] denotes an expression |
| 17:57.34 | voila | kaldemar: do u mean , set(i=$[$var + 1]) , will ste i=1 .. if var =0 ... ?? |
| 17:57.40 | voila | ste* |
| 17:57.43 | voila | set* |
| 18:04.30 | Aut0Exec | whats a good prodiver for free 800 numbers |
| 18:04.39 | Kobaz | free? |
| 18:04.40 | Kobaz | haha |
| 18:05.03 | voila | Kobaz: haha is not the good prodiver |
| 18:05.05 | voila | :P |
| 18:05.07 | Aut0Exec | not really free |
| 18:05.14 | Aut0Exec | but a provider |
| 18:05.33 | Kobaz | ~itsplist-us |
| 18:05.33 | 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 |
| 18:06.12 | Aut0Exec | ok thanks m8 |
| 18:08.35 | kaldemar | voila: try it |
| 18:09.38 | drmessano | Aut0Exec, are you asking if you can connect to multiple gtalk/gvoice accounts with Asterisk? |
| 18:09.43 | drmessano | If so, the answer is YES |
| 18:09.50 | Aut0Exec | drmessano: yes... |
| 18:09.54 | Aut0Exec | drmessano: how do I do that? |
| 18:10.14 | drmessano | You just add them to the jabber.conf file.. Just as you did the first one |
| 18:10.21 | Aut0Exec | ahhh ok |
| 18:10.39 | Aut0Exec | gatcha |
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| 18:31.21 | koffel | i have asterisk 1.6 and i seem to get this wierd error has anyone seen this error before? chan_sip.c: Maximum retries exceeded on transmission d4097689-d8c2-122e-08b7-003048c90586 for seqno 10591550 (Critical Response) -- See doc/sip-retransmit.txt |
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| 18:41.02 | manji | koffel, missing ACK ? |
| 18:48.35 | kaldemar | koffel: you're most likely suffering from a NAT setup issue. see the NAT parameters in sip.conf. |
| 18:49.44 | koffel | i don't have nat |
| 18:51.12 | manji | koffel, use ngrep and check the sip messages |
| 18:53.51 | koffel | okay so it a nat problem |
| 18:54.24 | voila | hi , well Dial(ZAP/1/9899792117) , willl it dial the mobile with mobile number 9899792117 on channel 1 ?? |
| 18:54.39 | koffel | i thinking i gota have nat because i am on business class with comcast and 5 static ips |
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| 19:11.29 | p3nguin | koffel: You can have static IPs and have NAT, or you can have static IPs and not have NAT. The fact that you are on Comcast Business with static IPs has nothing to do with it. |
| 19:13.02 | p3nguin | This is really starting to piss me off... we've got 30 MPH winds (which isn't really all that bad) and the power keeps going up and down and up and down, and it seems like it is down more than it is up. |
| 19:13.05 | *** join/#asterisk linuxgecko (~gecko@99-182-113-98.lightspeed.clmboh.sbcglobal.net) |
| 19:15.40 | linuxgecko | if i just need to read more, that's fine, but i'm trying to connect my google voice account with my asterisk server, and all the single-DID ways for doing that seem to all point to FREPBX. i'd rather not use freepbx unless there is no way around it. ( which i find hard to believe). is there a"good" reason everyone says to use freebpx? |
| 19:34.54 | koffel | does this look like right |
| 19:34.58 | koffel | http://pastebin.com/Ms4yaQxZ |
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| 20:19.16 | wonderworld | has someone used Patton media gateways (and would reccomend them?) |
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| 20:20.38 | koffel | does this look like right |
| 20:20.40 | koffel | http://pastebin.com/Ms4yaQxZ |
| 20:25.41 | linuxgecko | koffel: unless i recal incorrectly, the only ting marked ly wrong with that is that it's for a freepbx setup, and last i recalled, we don't support that here. |
| 20:26.25 | linuxgecko | unless like 25 means someting other than what i think. |
| 20:30.30 | koffel | okay linuxgecko |
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| 20:35.20 | drmessano | Why would one install of Asterisk 1.8 from SVN show Asterisk SVN-branch-1.8-r312509 and the other show Asterisk SVN-branch-1.8-r312509M (note the M at the end of the second) |
| 20:38.35 | benngard | svn version will differ |
| 20:38.54 | benngard | i have for axample a Asterisk SVN--r atm :) |
| 20:40.03 | drmessano | Yes, but why the M |
| 20:40.16 | drmessano | I just did an svn update on two boxes at the same time.. got two different strings |
| 20:40.25 | drmessano | Asterisk SVN-branch-1.8-r312509 and Asterisk SVN-branch-1.8-r312509M |
| 20:40.34 | drmessano | same revision |
| 20:42.03 | wdoekes2 | locally modified? |
| 20:42.10 | drmessano | Nope |
| 20:43.31 | wdoekes2 | really really sure? |
| 20:45.01 | drmessano | I have patched in the past.. but there shouldn't be anything that's not a direct mirror of SVN now. |
| 20:45.21 | wdoekes2 | svn revert . -R and retry |
| 20:47.32 | drmessano | (in progress..) |
| 20:49.37 | drmessano | AH HA |
| 20:50.09 | drmessano | Thank you.. that was it |
| 20:50.14 | wdoekes2 | :) |
| 20:52.41 | jaytee | drmessano, doing upgrades? what, no honeymoon? |
| 20:52.59 | *** join/#asterisk zkn (~zkn@82.131.93.186.cable.starman.ee) |
| 20:54.14 | drmessano | She's 15 hours across the country :( |
| 20:55.47 | drmessano | I did wake her up earlier and we had a smoke together.. It gets no better than that lol |
| 20:56.03 | jaytee | wow, now that's a long-distance relationship |
| 20:56.31 | drmessano | Won't be for long, thank goodness |
| 20:56.44 | jaytee | well that's good |
| 20:56.51 | jaytee | she moving or are you? |
| 20:57.05 | drmessano | I will be.. to Missouri of all places |
| 20:57.22 | jaytee | so you're leaving the job at Clear Channel? |
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| 20:57.40 | drmessano | Yep.. new roads ahead |
| 20:57.53 | jaytee | what part of Missouri? |
| 20:58.02 | drmessano | Branson/Springfield area |
| 20:58.39 | Nivex | jeez, seems like everyone's on the move lately. |
| 20:58.57 | jaytee | maybe you'll get a chance for a trip near Cape Giradeau and you could meet up with Katty for lunch |
| 20:59.24 | drmessano | Yep.. I already have plans to show up on her doorstep one day soon |
| 20:59.31 | jaytee | :-) |
| 20:59.58 | drmessano | In the great spirit of #asterisk, "Voluntarily or against her will" |
| 21:13.24 | p3nguin | She was supposed to be moving up this way (closer to STL). |
| 21:13.58 | p3nguin | I haven't heard from her for a while, so I don't know where those plans stand. |
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| 21:27.56 | koffel | kaldemar i configured my nat and still same message |
| 21:28.18 | koffel | [Apr 3 17:25:22] WARNING[29225] chan_sip.c: Maximum retries exceeded on transmission aa419f43-d8db-122e-3f84-003048c64ece for seqno 10596884 (Critical Response) -- See doc/sip-retransmit.txt. |
| 21:28.18 | koffel | [Apr 3 17:25:22] WARNING[29225] chan_sip.c: Hanging up call aa419f43-d8db-122e-3f84-003048c64ece - no reply to our critical packet (see doc/sip-retransmit.txt). |
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| 22:35.29 | p3nguin | traderz: What part of IL are you in? Chicago area, by chance? |
| 22:36.14 | koffel | i don't think it a nat problem |
| 22:36.36 | p3nguin | If you're getting retransmission failure notices, that's usally related to NAT. |
| 22:37.01 | koffel | well i used dmz |
| 22:37.05 | koffel | on my modem |
| 22:37.09 | p3nguin | That's half the problem. |
| 22:37.37 | p3nguin | Do you have a "router" appliance that connects to your modem? |
| 22:37.54 | koffel | it a gateway |
| 22:37.58 | p3nguin | Do you have a "router" appliance that connects to your modem? |
| 22:38.01 | koffel | router/modem |
| 22:38.07 | p3nguin | Do you have a "router" appliance that connects to your modem? |
| 22:38.25 | p3nguin | What part of my question are you having problems understanding? |
| 22:38.41 | koffel | well router is part of the modem |
| 22:38.45 | p3nguin | Do you have a "router" appliance that connects to your modem? |
| 22:38.46 | koffel | it is all in one |
| 22:38.48 | koffel | yes |
| 22:39.22 | p3nguin | So you have a "router" and also a modem. That's two (2) pieces of equipment. |
| 22:39.26 | p3nguin | Right? |
| 22:39.31 | koffel | wrong |
| 22:39.32 | koffel | 1 |
| 22:39.49 | koffel | it a gateway |
| 22:39.51 | p3nguin | Next time, tell me accurate information. |
| 22:39.52 | jaytee | "stay with Charlie Babbitt or go back to Wallbrook?" |
| 22:39.58 | p3nguin | (1738.44) <p3nguin> Do you have a "router" appliance that connects to your modem? |
| 22:40.01 | p3nguin | (1738.48) <koffel> yes |
| 22:40.17 | p3nguin | router, connected to modem... That's two pieces of equipment. |
| 22:40.28 | koffel | no |
| 22:40.31 | koffel | listen |
| 22:40.33 | p3nguin | That's what you told me. |
| 22:40.43 | koffel | it a bussiness class gateway |
| 22:40.48 | p3nguin | If you aren't willing to cooperate with me, I have no desire to help you. |
| 22:41.13 | koffel | p3 |
| 22:41.22 | p3nguin | Are you going to cooperate? |
| 22:41.27 | koffel | yes |
| 22:41.39 | koffel | it a bussiness class gateway |
| 22:41.50 | p3nguin | Let's try it again, then... do you have a router appliance which is connected to your modem (business class gateway)? |
| 22:41.58 | koffel | no |
| 22:42.20 | p3nguin | Okay. Does your gateway have a built-in switch where you can connect multiple computers? |
| 22:42.28 | koffel | yes |
| 22:42.38 | koffel | 4 ports |
| 22:43.03 | p3nguin | Okay, good. So your gateway will have adequate port forwarding settings built in so we can forward ports to your Asterisk computer. |
| 22:43.16 | koffel | okay |
| 22:43.23 | p3nguin | Did you configure the Asterisk computer to have a static IP address on your LAN? |
| 22:43.27 | koffel | i did do port 5060-5061 |
| 22:43.33 | koffel | yes |
| 22:43.38 | p3nguin | Okay, good. |
| 22:44.00 | p3nguin | Now go in and disable any of the DMZ bullshit that you previously configured. |
| 22:44.15 | koffel | just did |
| 22:44.52 | p3nguin | Go to the port forwarding section and forward UDP port 5060 and the UDP port range found in rtp.conf to your Asterisk IP address. That's UDP 5060 and normally UDP 10000-20000. |
| 22:45.06 | p3nguin | What version of Asterisk are you using? |
| 22:45.13 | koffel | 1.6.2 |
| 22:45.19 | p3nguin | That's not a version. |
| 22:45.58 | p3nguin | But knowing that you are using branch 1.6.2 is actually what I needed to know anyway. |
| 22:46.10 | koffel | Asterisk 1.6.2.16.1 |
| 22:47.41 | p3nguin | In Asterisk's sip.conf, in the general section, set nat=yes, canreinvite=no, localnet= to whatever your LAN network is including subnet mask (example: localnet=192.168.1.0/255.255.255.0), and externip= to whatever your public (outside) IP address is that you are using for Asterisk. |
| 22:48.58 | p3nguin | Were you having trouble with phones or with an ITSP? |
| 22:49.28 | koffel | voice mail |
| 22:49.45 | p3nguin | that wasn't one of the two choices. |
| 22:49.58 | p3nguin | phones or ITSP |
| 22:50.08 | koffel | itsp |
| 22:50.58 | p3nguin | So you are calling in from an outside phone on a non-VoIP telephone company to your DID number that you got from an ITSP? |
| 22:51.05 | p3nguin | ...trying to access voicemail. |
| 22:51.29 | koffel | yes |
| 22:52.18 | p3nguin | In sip.conf, you should have defined a peer entry for your ITSP. Check that it says nat=no in that peer definition. |
| 22:53.10 | koffel | nat = no |
| 22:53.31 | p3nguin | After you've made the above changes to sip.conf and checked your ITSP's peer entry, save any changes, and then run "sip reload" on the asterisk CLI, or optionally restart asterisk. |
| 22:55.35 | koffel | did everything u said |
| 22:55.39 | p3nguin | That should be a *normal* configuration for an Asterisk behind a NAT with an ITSP that isn't behind NAT. |
| 22:55.56 | koffel | still same error |
| 22:55.58 | p3nguin | I wanted to start with a *normal* configuration. |
| 22:56.34 | koffel | [Apr 3 18:54:33] WARNING[26205] chan_sip.c: Maximum retries exceeded on transmission 25b0c82e-d8e8-122e-3f84-003048c64ece for seqno 10599565 (Critical Response) -- See doc/sip-retransmit.txt. |
| 22:57.10 | p3nguin | Now that you've got things to a good starting point, it's time to troubleshoot. I can't promise I can figure out the problem, so maybe someone else will be willing to help. I'd like to see a sip debug of your failed call. |
| 22:57.38 | p3nguin | You know how to capture a sip debug of a failed call, right? |
| 22:58.01 | koffel | in asterisk -r do debug |
| 22:58.04 | koffel | right |
| 22:58.35 | p3nguin | asterisk -r will connect you to the console, "sip set debug on" will turn on sip debug. |
| 22:59.16 | p3nguin | Don't run the command until you are ready to make the call. Run the command, make the call, and then run "sip set debug off" to turn it back off as soon as the call has failed and you're done. |
| 22:59.18 | koffel | did that |
| 22:59.26 | p3nguin | Then copy the sip debug and pastebin it. |
| 23:00.36 | p3nguin | This is the part where we'll needs all hands on deck. Sometimes sip debugs don't tell me anything because of my lack of experience with sip debugs. |
| 23:01.05 | koffel | http://pastebin.com/VVaBUjsK |
| 23:01.09 | p3nguin | I (thankfully) don't usually run into problems, so I don't have to do a lot of troubleshooting. |
| 23:02.12 | p3nguin | I don't see any sip debug there. |
| 23:02.26 | jaytee | me neither |
| 23:03.19 | koffel | i did sip set debug on |
| 23:05.14 | koffel | http://pastebin.com/A6PTxW60 |
| 23:05.42 | p3nguin | That's better. |
| 23:07.35 | p3nguin | What is operating on port 5061, and why? |
| 23:07.46 | p3nguin | I'm not really seeing any fatal problems in that sip debug. |
| 23:07.51 | koffel | http://pastebin.com/Be9YhtEt |
| 23:08.01 | koffel | 5061 |
| 23:10.06 | koffel | looks like i got noisy feedback |
| 23:10.58 | p3nguin | What is operating on port 5061, and why? |
| 23:11.10 | koffel | sip phone 1 |
| 23:11.23 | koffel | should i change it |
| 23:11.26 | koffel | to none |
| 23:12.28 | p3nguin | I don't see any reason that every phone on the LAN can't operate on the standard port (5060). |
| 23:13.08 | koffel | okay all set on 5060 |
| 23:19.23 | koffel | http://pastebin.com/sE86bsMa |
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| 23:32.51 | koffel | p3n any suggestions |
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