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| 01:22.53 | cusco | what is wanpipe? |
| 01:26.19 | [TK]D-Fender | cusco: Sangoma card driver |
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| 01:26.40 | cusco | ok |
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| 08:39.38 | bulkorok | hello |
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| 09:41.31 | BorjaGVO | Hi everyone, I want to check if a variable follows a certain pattern. If it doesn't, then go to label "return". That's what I'm trying with this dialplan snippet: is it right? Should I write quotes or it would work like this? GotoIf(${AMPUSER}!=38.}?return) |
| 09:44.53 | kaldemar | BorjaGVO: you're missing $[], curly braces don't match and you've made an erroneous assumption that . can be used as a wildcard. |
| 09:45.52 | BorjaGVO | kaldemar: yeah, that's right. It's an assumption. I don't know if in an if condition I can use that kind of wildcard |
| 09:46.09 | BorjaGVO | What wildcard can be used? |
| 09:51.38 | kaldemar | none. |
| 09:52.27 | kaldemar | what you should concentrate on is take two first characters of ${AMPUSER}. |
| 09:55.57 | BorjaGVO | kaldemar: alright..i got it! Thanks! |
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| 10:26.14 | m0sphere | anyone up for answering a bunch of probably retarded questions about asterisk and voip in general? |
| 10:28.06 | Chainsaw | Sure, we'll have a go. |
| 10:28.56 | m0sphere | great. so, i have a small company running off of old centrex services through the major provide here in canada |
| 10:29.50 | m0sphere | our customers dial a number for our switch which is local to them, they're then prompted for a 7 digit number, which would not be local to them. our switch then uses the centrex lines to connect the customer to the person they're trying to call |
| 10:30.20 | m0sphere | i want to get away from centrex and move to voip but i don't really know what it is I need |
| 10:30.54 | m0sphere | i also want the users to be able to dial a 10 digit number and allow them canada wide long distance |
| 10:31.52 | m0sphere | i would also like the server to become virtual/cloud based so i don't have to screw with hardware |
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| 10:32.21 | Chainsaw | m0sphere: So most of this is dial-plan magic. |
| 10:32.45 | Chainsaw | m0sphere: You basically want to take a number they dial, modify it, and then send it out through a paid VoIP telephony provider. |
| 10:32.57 | m0sphere | right |
| 10:33.12 | m0sphere | but still be able to verify they're valid and active customers |
| 10:33.13 | Chainsaw | m0sphere: How are we interfacing with the telephones in the small company? Analog phone lines? Digital ISDN phones? Some other system? |
| 10:33.25 | m0sphere | analog lines |
| 10:33.42 | Chainsaw | m0sphere: Okay. And what equipment are we using locally? An Asterisk box you provide, or are we buying something in? |
| 10:34.25 | m0sphere | currently our servers are red hat running some custom software that was made for this purpose |
| 10:34.34 | Chainsaw | m0sphere: (This is mostly about protocol options. SIP has the widest support, but you could talk IAX2 between two Asterisk-based solutions and bypass NAT just that little bit easier.) |
| 10:34.38 | m0sphere | but i'd like to go virtual/cloud based |
| 10:35.08 | Chainsaw | m0sphere: Well yes, you could virtualise your main Asterisk instance anywhere you like. But you can't connect far-away analog lines into that cloud without having some little box locally. |
| 10:35.43 | Chainsaw | m0sphere: I'm trying to find out whether that will be some "ATA" device or a full-fledged Asterisk server with PCI/PCI-X/PCIe cards. |
| 10:36.19 | m0sphere | currently our server's have dialogic pci cards in them |
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| 10:36.51 | Chainsaw | m0sphere: Okay. So you have a rackmount server at each customer location? |
| 10:37.04 | m0sphere | basically |
| 10:37.12 | Chainsaw | m0sphere: Did you wish to keep that, or are you scaling that down at the same time? |
| 10:37.33 | m0sphere | i'd like to get rid of phsysical hardware |
| 10:38.04 | Chainsaw | m0sphere: Well, we can likely scale down to a little "ATA". |
| 10:38.07 | m0sphere | if i have to keep them, thats fine, but they're so old and having to drive to the other 2 cities if something goes wrong is not ideal |
| 10:38.30 | Chainsaw | m0sphere: By which I mean a little box like this: http://www.voipmechanic.com/images/spa2102_210.jpg |
| 10:39.00 | Chainsaw | m0sphere: You could go for something more Telco-grade though, like a Patton Smartnode. 4118 comes to mind. |
| 10:39.42 | m0sphere | so I would need 1 of those in each city? |
| 10:39.46 | Chainsaw | m0sphere: In which case the 4118 speaks SIP to your "cloud" Asterisk instance and provides 8 analog telephone lines, complete with ringing voltage and caller ID, to your customer equipment. |
| 10:40.12 | Chainsaw | m0sphere: Well, I can't answer that yet. You haven't told me enough about your topology. |
| 10:40.55 | m0sphere | hmm 1 sec |
| 10:41.41 | Chainsaw | m0sphere: But just as a plan of action, I would only make one change at a time. So I would deploy say... a 4118 on the customer site to push the telephony out towards a physical server elsewhere. And then in phase 2, I would virtualise that server. |
| 10:42.09 | Chainsaw | m0sphere: That way, my customers can never conflate the change of equipment with the virtualisation of the core server. Because they can, and they will. And it will be your fault. |
| 10:42.15 | m0sphere | oh i think i didn't explain something properly |
| 10:42.30 | Chainsaw | m0sphere: Very well. Let's add that clarification and see if it changes the picture. |
| 10:42.36 | m0sphere | my customers are the general public, they use their own phones, analog, digital, cellular, whatever |
| 10:42.40 | m0sphere | they dial an access number basically |
| 10:42.53 | m0sphere | they don't get any equipment from me |
| 10:42.56 | Chainsaw | m0sphere: That is rather different a picture, yes. |
| 10:43.15 | m0sphere | http://i.imgur.com/1WXFIax.png i made this a little while ago |
| 10:43.17 | Chainsaw | m0sphere: So you have a telco terminating those access numbers on analog telephone lines. |
| 10:43.32 | m0sphere | ya |
| 10:43.57 | Chainsaw | m0sphere: Which you take elsewhere over SIP, and then terminate on other telco-provided analog telephone lines. |
| 10:44.56 | Chainsaw | m0sphere: What I said about replacing servers with smaller equipment, like that 4118 I mentioned, continues to apply. |
| 10:45.33 | Chainsaw | m0sphere: You could use a single core server, be it "cloud" virtualised, part of a virtual cluster you host yourself, or a plain old physical server... to route the traffic between those boxes. |
| 10:46.09 | m0sphere | so in my diagram, it makes sense right? |
| 10:46.30 | Chainsaw | m0sphere: It does, yes. |
| 10:46.50 | Chainsaw | m0sphere: I could use smaller, simpler equipment in most places, but I'd have to place a core server in the middle. |
| 10:47.35 | m0sphere | are you in north america? |
| 10:47.37 | Chainsaw | m0sphere: The edge routers, currently in your diagram as 4 big servers, can be replaced with simpler equipment if you place that one core server in the middle. |
| 10:47.46 | Chainsaw | m0sphere: No, I am in the United Kingdom. |
| 10:48.13 | Chainsaw | m0sphere: I can think of two people that could set this up for you, but one is also in the United Kingdom and the other is in South Africa. |
| 10:48.58 | m0sphere | does voip.ms offer what I need? |
| 10:49.00 | m0sphere | http://voip.ms |
| 10:49.28 | Chainsaw | m0sphere: That would replace all of your equipment though. |
| 10:49.44 | Chainsaw | m0sphere: It depends on whether they'd be able to take over your current analog line termination points. |
| 10:50.36 | m0sphere | ok so i need those boxes to connect the analog lines to a virtual pbx? |
| 10:51.03 | Chainsaw | m0sphere: Well, this is highly dependent upon your ambitions. |
| 10:51.22 | m0sphere | what do you mean? |
| 10:51.31 | Chainsaw | m0sphere: The way I'm looking at it, is by providing the equipment yourself. So as to maximise your ability to rectify faults and your profit. |
| 10:52.04 | Chainsaw | m0sphere: If you're looking for a minimum hassle approach where your profit margin is less, but where you can then expand so much further to make equal profit... |
| 10:52.13 | Chainsaw | m0sphere: I suppose you can involve others like voip.ms |
| 10:52.35 | Chainsaw | m0sphere: This goes more into "running a business" territory then "setting up a VoIP infrastructure" though. |
| 10:54.47 | m0sphere | assuming i keep phsyical boxes in each city |
| 10:55.26 | m0sphere | how do they allow multiple callers to use them at the same time |
| 10:56.10 | Chainsaw | m0sphere: By having multiple analog lines (difficult way), or by terminating on ISDN instead (easy way). |
| 10:56.24 | m0sphere | with centrex, we only need 2 lines going into each server, from how i understand it, it connects the caller and the callee together and drops our servers involvement |
| 10:57.56 | Chainsaw | m0sphere: Doesn't sound like analog lines to me. |
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| 10:59.16 | m0sphere | hmm |
| 11:02.39 | m0sphere | what kind of costs would i be looking at |
| 11:03.32 | m0sphere | our monthly costs are currently $400/city for centrex lines |
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| 11:06.26 | Chainsaw | m0sphere: Unfortunately I do not deal with the commercial side here. |
| 11:07.23 | m0sphere | ok |
| 11:08.19 | m0sphere | the asterisk software can do what I need it to do though right? with the routing based on the callers number and their input |
| 11:18.42 | Chainsaw | m0sphere: Yes, I see no problem with that requirement. |
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| 12:44.08 | BorjaGVO | I want to give my dialplan some custom behaviour (I'm using FreePBX, but I considered that this is an Asterisk issue as I don't really understand what is happening within Asterisk Dialplan. For now I'm just trying that it does Noop correctly. You can see both contexts in here: http://pastebin.com/xvxRk94g I can not make it work. What am I doing wrong? |
| 12:44.12 | BorjaGVO | Tnaks in advance |
| 12:47.09 | Chainsaw | You need to speak to the FreePBX people. |
| 12:47.15 | Chainsaw | They are in #freepbx |
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| 13:16.32 | BorjaGVO | If you use "include =>" inside a context and you include the same extensions? What would have priority? Which extensions would be executed? |
| 13:17.50 | [TK]D-Fender | BorjaGVO: Answered in #freepbx |
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| 13:25.42 | Sophira | Hiya. |
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| 14:16.04 | m0sphere | if you guys had to guess, what is the average length of a phone call these days? would there be a significant difference between the average length of a business call vs a residential call? |
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| 14:16.44 | Chainsaw | m0sphere: I don't think Asterisk keeps those statistics by default, you'd have to extract it from CDR data. |
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| 14:17.31 | m0sphere | i just meant in general |
| 14:17.37 | m0sphere | not related to asterisk |
| 14:19.07 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:d058:1520:8dcd:78bc) |
| 14:19.10 | jacobw | Hi, I'm getting "chan_sip.c:32856 setup_srtp: No SRTP module loaded, can't setup SRTP session." when I try to call an extension, but I don't have an SRTP configuration in the extension or server in general |
| 14:20.22 | [TK]D-Fender | jacobw: Perhaps the other side is OFFERING it |
| 14:20.57 | puzzled | hi |
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| 14:27.43 | jacobw | [TK]D-Fender: could be I guess |
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| 14:35.57 | [TK]D-Fender | jacobw: "could be" is often confirmed by looking at the call. |
| 14:36.58 | BorjaGVO | what cmd can I use to grab the two first digits of a variable? I want to get just 32 from 3249 |
| 14:37.21 | jacobw | [TK]D-Fender: sorry, checking now |
| 14:38.05 | [TK]D-Fender | BorjaGVO: there is no "command". You can return the RESULT of chopping the first 2 off by ${VARIABLE:2} |
| 14:38.17 | [TK]D-Fender | BorjaGVO: Go read your DIALPLAN BASICS |
| 14:38.23 | [TK]D-Fender | BorjaGVO: This is all in the book... |
| 14:38.25 | [TK]D-Fender | ~book |
| 14:38.26 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 14:38.36 | BorjaGVO | [TK]D-Fender: yes, thanks |
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| 15:01.51 | Histerisk | hi! I need help to properly BLF-monitor an spa3102 PSTN gateway from some of my phones. Someone who could help? |
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| 15:03.25 | [TK]D-Fender | Histerisk: set callcounter for you peer. make a dialplan hint pointing to the device. use the hint. Done/. |
| 15:07.56 | Histerisk | D-fender, thanks, this already works. However when the analog line is seized externally by an analog phone connected to it, asterisk doesn't seems to get informed of the busy status, and it doesn't notify the BLF subscribed phones. Any hint about how this could be fixed? |
| 15:09.21 | [TK]D-Fender | Histerisk: There is no detection that the "line" is in use, only if the SPA is claiming to have seized it |
| 15:09.42 | [TK]D-Fender | Histerisk: It is not made to "share" a line. |
| 15:10.13 | [TK]D-Fender | Histerisk: There is no normal means of getting that info |
| 15:12.14 | *** join/#asterisk SushiB (~Thunderbi@200.77.217.106) |
| 15:15.46 | Histerisk | D-Fender: so I understand from your answer that although it knows the line is busy (it is shown in its info tab, near the voltage and current levels), the firmware does not notify asterisk about the seizure unless the analog line has an incoming call or an outgoing call placed by the spa itself |
| 15:16.35 | [TK]D-Fender | Histerisk: There is no mothod for it to tell * this |
| 15:16.48 | WIMPy | Maybe you can subsribe directly to the GW and get that info, i.e. without Asterisk. |
| 15:16.48 | [TK]D-Fender | Histerisk: * does not receive presence info from devices. |
| 15:16.59 | [TK]D-Fender | Histerisk: * reports ITS usage of devices |
| 15:19.47 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:19.47 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:20.21 | Histerisk | D-Fender: I understand. So maybe the only way is making the phones subscribe to the GW, as WIMPy pointed out |
| 15:23.36 | [TK]D-Fender | If it even allows it... |
| 15:24.39 | *** join/#asterisk gtjoseph (~gtj@unaffiliated/gtj) |
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| 15:26.28 | Histerisk | D-Fender: well, the spa allows the SUBSCRIBE method in its headers, so something must be able to be done! :-/ |
| 15:28.46 | [TK]D-Fender | Histerisk: Go try.... |
| 15:29.24 | file | just because it has SUBSCRIBE does not mean it will do what you want. |
| 15:32.33 | Histerisk | D-Fender: yep, I'll try. Thanks a lot! |
| 15:33.20 | Histerisk | @file: I know, I know. I doubt it can do it, but I'll give it a try. Thanks! |
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| 15:46.05 | *** join/#asterisk orn (~orn@2a01:8280:101:0:a17e:8f1b:bca:b6db) |
| 15:47.45 | orn | Hello. Does anyone know if I can catch a call file that has given up (retried and failed too many times) in the dialplan somehow, e.g. send a text message if this is the case? If not in the dialplan, it would be fine if I could specify an action within the call file. |
| 15:50.59 | *** part/#asterisk Histerisk (501cfaa4@gateway/web/freenode/ip.80.28.250.164) |
| 15:53.10 | [TK]D-Fender | orn: http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out#Example |
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| 15:54.44 | orn | thank you! |
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| 16:05.24 | orn | D-Fender: Do you know if there's anything else I can access other than an archived call file to determine whether the outgoing call failed completely? That is, all the retry attempts failed too? This goes to failed for each failure. |
| 16:06.15 | *** join/#asterisk pietro (~pietro@78-134-1-229.v4.ngi.it) |
| 16:06.22 | pietro | hello. |
| 16:06.52 | [TK]D-Fender | orn: Don't see it telling you the reason. At that point, if you really care I'd recommend doing the actual dialout in a local channel and monitoring it yourself... |
| 16:07.30 | pietro | I need to grab the Call-ID header used in the 2nd leg of call(SIP/${EXTEN}). I need it before the call answer (so no M() ). Is there a way ? |
| 16:12.53 | orn | ah, the channel does have a REASON=5 variable |
| 16:13.00 | orn | after it gives up |
| 16:13.07 | orn | whatever the =5 means :D |
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| 16:13.21 | orn | ahh |
| 16:13.22 | orn | ok |
| 16:13.25 | orn | means busy |
| 16:13.26 | orn | thanks a lot! |
| 16:14.05 | orn | oh, never mind. it does that for all the calls :) oh well |
| 16:14.28 | [TK]D-Fender | pietro: Not in that leg.. you'd have to use an external process to monitor AMI. |
| 16:16.15 | pietro | [TK]D-Fender: as I suspected .. thanks |
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| 16:33.43 | SushiB | Hi people, there is anyone can help me with this problem.... Signalling requested on channel 13 is MFC/R2 but line is in ISDN PRI signalling???? |
| 16:36.00 | [TK]D-Fender | So change your signaling parameters in your DAHDI configs... |
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| 16:41.08 | SushiB | when I change /etc/asterisk/dahdi-channels.conf and I set signalling = mfcr2 and I reload the module, it show me this message, WARNING attempt to configure channel 12 with signalling MFC/R2 ignored because it is already configured to be MFC/R2, and ERROR, signalling requested on channel 16 is MFC/R2 but line is in unknown signalling 896 signalling |
| 16:42.51 | Penguin | The word is spelled "signaling" so could that have an effect on the setting? Or is the setting actually misspelled? |
| 16:43.22 | SushiB | my bad, is signalling |
| 16:45.03 | SushiB | I set the mfcr2_variant and all the other mfcr2 parameters on the /etc/asterisk/dahdi-channels.conf file |
| 16:46.21 | SushiB | and I also set on the chan_dahdi.conf over [channels] signalling=mfcr2 |
| 16:48.16 | Penguin | No, signaling. |
| 16:50.23 | SushiB | no, the file is spelled as: signalling |
| 16:50.34 | file | grins |
| 16:50.44 | Penguin | So they spell it wrong in the configuration setting? |
| 16:50.52 | Penguin | That's not very intuitive. |
| 16:51.01 | file | it will accept either |
| 16:51.34 | SushiB | jejee,,, nooo,, it was my bad... I typed wrong! on the chat |
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| 17:36.34 | mitchrodrigues | Is their an easy way to enable ref debug |
| 17:36.37 | mitchrodrigues | accorss the entire core |
| 17:36.51 | mitchrodrigues | im about tojust stick it in asterisk.h |
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| 17:38.16 | mitchrodrigues | nvm i foudn the docs :) |
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| 19:28.35 | zpotoloom | evening |
| 19:29.10 | zpotoloom | has anyone encountered a problems with asterisk and Cisco spa79XX series phones ? |
| 19:29.30 | zpotoloom | phone is registered to asterisk, but on INVITE it answers with SIP/2.0 404 Not Found |
| 19:30.51 | navaismo | cisco 79xx phones sucks |
| 19:30.59 | navaismo | use spa series |
| 19:31.05 | navaismo | or another sip phone |
| 19:31.27 | zpotoloom | yeah... works on lan, but if NAT comes to play then problems start |
| 19:31.34 | zpotoloom | 79XX series then |
| 19:32.28 | navaismo | maybe you can correct the nat issues on the asterisk side |
| 19:33.14 | zpotoloom | currently i'm not even sure what's causing the phone to respond with 404 for invites |
| 19:33.48 | zpotoloom | cause they were working for some time without problems, and now just stopped |
| 19:33.58 | zpotoloom | without changing the configuration and everything |
| 19:33.58 | SushiB | I have those phones working with my asterisk, I'm using cisco 7940 phones. and on every extension I set nat=no |
| 19:34.08 | zpotoloom | reloaded the fw and configuration and still |
| 19:36.18 | slav3_kitten | happy veterans day to all our current or former armed services in here :) |
| 19:36.31 | zpotoloom | SushiB I have 7940 and 7960 on LAN also working fine without NAT |
| 19:37.27 | zpotoloom | nop, no use, even with nat=no still answering with 404 |
| 19:38.57 | Penguin | nat=no is not a setting for extensions. |
| 19:39.07 | Penguin | It's a setting for sip devices in sip.conf. |
| 19:39.16 | zpotoloom | yes, i know :) |
| 19:39.30 | Penguin | sushib didn't. |
| 19:40.14 | zpotoloom | i thought maybe Cisco doesn't like letters in extension name as I have |
| 19:40.49 | Penguin | Phones don't care what extensions are named, but it is hard to dial letters from the keypad. |
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| 19:43.44 | zpotoloom | my bad, not extension, sip device name :) |
| 19:44.14 | Penguin | Letters are perfectly valid in the device names. And they are encouraged. |
| 19:44.44 | Penguin | Pastebin your device definition for the phone giving you trouble. Mask ONLY the secret. |
| 19:51.32 | zpotoloom | Penguin, http://pastebin.com/irQKbvXb added SIP trace also |
| 19:52.24 | Penguin | You've mangled the sip debug, so it's useless. |
| 19:52.24 | zpotoloom | at first it seemed like some SIP ALG proxy doing something, but I've changed the port from 5060 to 5080 and still the same |
| 19:52.33 | zpotoloom | i-ve only removed the IP address |
| 19:52.58 | Penguin | Always make sure SIP ALG is disabled. |
| 19:53.10 | Penguin | Is the phone behing a NAT? |
| 19:53.31 | Penguin | Asterisk is in a remote location while the phone is on your LAN? |
| 19:54.03 | zpotoloom | nop, i have asterisk access, it's on the public network, phone is behind NAT, no access to there, nor idea what router is in use |
| 19:54.45 | Penguin | So you have the phone behind a NAT, with asterisk on the outside, but you still set nat=no for the phone? |
| 19:55.25 | zpotoloom | nat was set to yes, tried with nat=no also |
| 19:55.37 | Penguin | If the phone is behind a nat, the correct setting is not nat=no. |
| 19:57.10 | Penguin | So let's change that setting to a more appropriate setting and change the port back to the normal port. |
| 19:57.34 | zpotoloom | ok, I'll try this |
| 19:58.02 | Penguin | In your SIPDefault.cnf or SIP<MAC>.cnf, make sure you did not enable the nat settting. nat_enable: 0 |
| 19:58.07 | Penguin | Let asterisk handle the nat stuff. |
| 20:02.26 | Sophira | I know this channel is for Asterisk, but I can't find any IRC channel applicable for this. Is there anybody here who has experience with the Linksys SPA3102? I'm setting one up and I'm running into an odd issue. |
| 20:02.46 | Sophira | I'm happy to take it to /msg if the conversation is too off-topic for this channel. |
| 20:02.51 | Penguin | What's the problem? |
| 20:03.55 | Sophira | Well, I have a phone connected to the SPA's FXS socket and have got the SPA to route my calls okay, but the dial tone I get from the SPA after going off-hook only lasts for about 2.5 seconds before giving me the reorder tone. |
| 20:04.39 | Sophira | Syslog doesn't show any messages other than going off-hook, and my dial tone specification seems to imply it should last for 10 seconds. ("350@-19,480@-22;10(*/0/1+2)") |
| 20:05.08 | Sophira | (yes, I realise 480Hz is non-standard; that's deliberate) |
| 20:05.25 | Sophira | (although hmm, I wonder if my phoneis detecting that or something. Hang on) |
| 20:05.35 | Sophira | puts it back to 440. |
| 20:06.06 | Sophira | Nope, it's definitely the SPA doing it. |
| 20:07.31 | zpotoloom | Penguin, no change, except client port which is now 5060, thanks for input |
| 20:07.43 | zpotoloom | i'll try to get access to the router to see what's going on there |
| 20:08.01 | Penguin | zpotoloom: Did you check the cnf files to be sure nat_enable is set to 0? |
| 20:08.05 | Sophira | I can't find any settings that might do it. The only one close to a value that seems relevant is "CWT8 Cadence", which is "2.3(.3/2)". Changing the 2.3 to a higher number makes no difference though, as expected. |
| 20:10.46 | Penguin | Is the phone even registering to asterisk? It looked like you were trying to make a call to the phone. |
| 20:11.23 | Sophira | Penguin: (are you talking to me or to zpotoloom?) |
| 20:12.02 | Penguin | Oh, I was talking to zpotoloom about his Cisco phone. |
| 20:12.13 | Sophira | Okay. |
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| 20:14.35 | zpotoloom | Penguin, nop, cause that will not work, but I tried still setting cisco nat_enable to 0, now it's time to go to sleep :) |
| 20:14.49 | zpotoloom | cause it's sending register with internal IP REGISTER sip:XX.XX.XX.XX:5060 SIP/2.0 |
| 20:14.50 | zpotoloom | Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK0de06ced |
| 20:15.04 | Penguin | That's because of a fucked up nat setting. |
| 20:15.21 | Penguin | If you don't ever register, asterisk will never be able to send a call to the phone. |
| 20:16.02 | *** join/#asterisk dxd828 (~dxd828@host-92-24-127-29.ppp.as43234.net) |
| 20:19.30 | zpotoloom | damn, there was also nat_received_processing: "1" in SIPDefault.cnf, missed that |
| 20:20.30 | Penguin | Always turn off all nat crap on phones. Let asterisk handle the nat stuff. |
| 20:21.13 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.81) |
| 20:31.13 | kasanop | zpotoloom: cisco 79xx phones use random source ports for sending sip register, and 5060 for all other messages, that's why nat=yes in sip.conf doesn't work |
| 20:33.22 | Penguin | I've never had a problem with NAT and my Cisco phones, but I did switch to SCCP because the SIP firmware is extremely limited. |
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| 20:53.43 | Sophira | Hmm. Nobody knows the answer to my query, then? |
| 20:54.03 | Sophira | (not that I expect anybody to in an Asterisk channel, given my question isn't actually about Asterisk...) |
| 20:56.05 | Penguin | Lots of people use that device with asterisk. Give people time to read and decide if they want to answer. |
| 20:57.50 | ChannelZ-Wk | Sophira: Sounds like your dialplan in the SPA has a weird timeout or globally as a setting it has a short timeout |
| 20:59.33 | Sophira | ChannelZ-Wk: That's what you'd think, but I can't find the timeout value anywhere. |
| 20:59.51 | Sophira | Oh, hang on. |
| 20:59.56 | Sophira | I just realised something. |
| 21:00.46 | ChannelZ-Wk | I'm looking in mine, there might not be a default as a setting.. but it is something you can change within the dialplan string so make sure that doesn't have something going on |
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| 21:01.15 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 21:02.57 | Sophira | ChannelZ-Wk: I just found the problem. As a test, I was using the dial plan "(x.<:@gw0>)". As it turns out, though, the "." operator means repetition zero or more times, meaning that no entry at all was turning out to be a completed sequence and thus the short interdigit timeout was used. |
| 21:03.15 | Sophira | The solution was using "(xx.<:@gw0>)" instead. |
| 21:03.32 | Sophira | (obviously in my real usage, I wouldn't use a dial plan like that; it was just a test) |
| 21:03.33 | ChannelZ-Wk | The default Interdigit timers are on the Regional tab BTW |
| 21:04.29 | Sophira | Yeah, and that's set to 3 seconds for the short one. The thing is, I was assuming that that wouldn't be the issue. Heh. |
| 21:05.10 | Sophira | Also, my measurements showed the tone was only on for 2.5 seconds or so. I guess the beginning .5 seconds weren't coming through because of my FXS settings. |
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| 21:49.17 | ChannelZ-Wk | Well glad you found it |
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| 22:09.40 | bluehawk | So I'm testing and playing around with an event socket and trying to initiate a call through the php example from the wiki (http://wiki.freeswitch.org/wiki/PHP_Event_Socket) I have two SIP agents registered, 1003 and 1004. What $cmd to I send to connect a call between those two users? Something like "originate user/1003, user/1004" isn't working, so I must be doing something stupid. |
| 22:13.19 | navaismo | wrong channel? |
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| 22:14.34 | bluehawk | ha |
| 22:14.36 | bluehawk | yes |
| 22:14.37 | bluehawk | sorry |
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| 23:11.09 | elcontrastador | Can I get some opinions on the very best desktop sip softphone for osx and Asterisk? I've been using Bria but there's just got to be something better.... |
| 23:11.54 | navaismo | BLINK |
| 23:13.59 | elcontrastador | sweet...checking it out now |
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| 23:21.22 | elcontrastador | navaismo: Thanks so much! Blink is so much better than Bria so far. |
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