Re: [Asterisk-Users] Polycom IP600 - Flashing clock and date?
Billy Dunn wrote: This was a pain in the butt for me. In fact, I only was able to get it going by pointing the SNTP server to pool.ntp.org and making sure the DNS entries were correct. That works, but it's not a great solution. When the phone is flashing, that means it cannot contact the SNTP servers. Ideally it should talk to a local NTP server on your network, but I have yet to see that work (but I'm only two weeks into Asterisk too). Good luck. I'm using local NTP server (which, in turn, syncs from "close" NTP servers on Internet). Set for clients in DHCP: option ntp-servers x.x.x.x; Works great. Polycom 501. - Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller logging in to call out IAX line?
DISA - Dan Min Hwan Chang wrote: Hm, I'm wondering if its possible for someone to call in the POTS line, dial an extension, then be able to dial a number of their choosing out the IAX line? So let's say I'm here in california and I dial into the office. Dial which gets me a message saying please enter the number you'dl ike to call. At which point I dial 7983487 to dial someone in Austria over IAX. Is this possible? Regards, Min Chang ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension Lights Patch
I've got the 501, and have the presence subscription working. This is where you subscribe to a buddies presence. For more info go to: http://www.voip-info.org/tiki-index.php?page=Polycom+Phones last paragraph of Another sip.conf example. If that's not what you're talking about, then I'd love to be enlightened. - Dan Eric Rees wrote: Could you pass along the information you used to get the Polycom lights to work. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Hayden Sent: Wednesday, July 20, 2005 11:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Extension Lights Patch I've been using the extension lights on my polycoms before that patch, so I'm not sure what it fixed, but I've only seen the lights work on Polycoms and Snoms. Try using the hint priority and see if it works for your gxp2000, be sure to post your results! -- Tom Hayden Astoria Telecom, LLC www.astoriatelecom.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension Lights Patch
Matt Loretitsch wrote: I wish someone would just post a sample extensions.conf so I could FINALLY understand this. Could you post at least the hint portion of yours? I have tried this repeatedly without success and am starting to feel like a true idiot. Is it something like this? exten = 2352,hint,SIP/2352 That's what I've been trying. I also made sure the buddy watch was turned on. IP501 phone. Sip.conf has the sip channel defined as [2352] and not by user name. Thanks! -Matt Here's my hint stuff. exten = 501,hint,SIP/spa1 exten = 502,hint,SIP/sipura2000-1 exten = 503,hint,SIP/pc1 exten = 510,hint,SIP/pc2 spa1 is a Sipura 3000 FXS port. sipura2000-1 is a Sipura 2000 FXS port. pc1 and pc2 are Polycom 501 lines. Here's how I understand how it works. The phone requesting to be notified about another phone's status does a SIP SUBSCRIBE for that extension. Asterisk looks up that extension's hint priority to find out which device to monitor status for. When Asterisk senses the monitored device's status change, it sends a notify to the phone(s) that had subscribed to the device's presence. So basically, the hint priority ties together an extension and a device/channel. It looks like you have it right. Note that for the Polycom at least, in sip.conf you _have_ to have the Polycom's line 1 information _last_, if you have more than one line registered on the Polycom. This is because the Polycom uses the credentials for the first line when it authenticates the SIP SUBSCRIBE request. But Asterisk looks up the credentials for SIP SUBSCRIBE requests according to IP, and uses the first one it finds, which is the last one defined in sip.conf. At least it was in my case... and I banged my head against chan_sip.c for a number of hours getting this figured out. If you still have problems, what do you see on the Asterisk CLI at a decently high verbosity, when you set Watch Buddy to enabled? - Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best VoIP provider
I asked them a while ago (month or so?) about International rates. They responded that they got burned on International call fraud, and only allow International termination under special circumstances (or something to that effect). - Dan Jay Milk wrote: That's odd -- they used to be here: http://www.nufone.net/rates.csv Of course, you can't rely on that. -Original Message- From: Chris Mason (Lists) [mailto:[EMAIL PROTECTED]] Sent: Tuesday, July 19, 2005 6:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Best VoIP provider Madhawa Jayanath wrote: o Bernie, 1) best results www.nufone.net 2) low cost www.voipjet.com Anyone able to find NuFone's rates? I have been looking for them on their site. I need international rates and UK Mobile. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Quit Registering with Broadvoice
Jerry Geis wrote: Ken, Point to a different proxy. I had the same issue with chicago... Same here with DCA. Now using MIA. - Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs and bandwidth
If you include down + up, yes, it's actually about 150-160 using uLaw + IP/UDP/RTP/signaling overhead. But that's a little misleading, I think. 1/2 of that (~75-80) is down, 1/2 of that (~75-80) is up. So if you have, say, a 1.5Mbps down/384 up DSL connection, you can do up to 4 calls simultaneously. 80*4 = 320. You'd be using 320kbps down and 320kbps up, which is within your 1500kbps down / 384 kbps up. Someone please correct me if I'm wrong. - Dan Tim Pushor wrote: Of course - ISDN is bi-directional. I guess saying that ULAW takes 130K+ bandwidth depending on the framing type (local lan, w/1 hop, vlan, etc) is not very clear. Thats total bandwidth. With lots of us at home and small business using asynchronous connections - we need to keep that in consideration. Thanks for helping clear that up. Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 501 Configs
I've had one on my desk for a couple weeks now. What I've done: Used DHCP to get the IP address / gateway / ntp server / dns server. Not used DHCP to get the FTP server (rather than futz with my DHCP server settings). Manually set the FTP server IP/username/password on the phone (I didn't want to use mixed case username... I'm picky). I got my config files from what was posted on the wiki. At what point are you having problems? - Dan chris gamble wrote: I just received my first polycom 501, tried my best to follow all of the documentation and configs on the wiki ( though some conflicted in which case i tried both ways ), and at the end of the day can not connect my phone to asterisk. My Questsions: does the wiki information apply to the 501's? The images and config files at freedomphones, are those supposed to work with the 501? Has anyone gotten one of these updated phones to work with asterisk, and if so can you share your experience. Thanks, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 501 Configs
Kristian Kielhofner wrote: chris gamble wrote: I just received my first polycom 501, tried my best to follow all of the documentation and configs on the wiki ( though some conflicted in which case i tried both ways ), and at the end of the day can not connect my phone to asterisk. My Questsions: does the wiki information apply to the 501's? The images and config files at freedomphones, are those supposed to work with the 501? Has anyone gotten one of these updated phones to work with asterisk, and if so can you share your experience. Thanks, Chris, Have you tried my configs at: http://www.krisk.org/asterisk/pcom If you are using 1.5.2 or later you will want to use the configs in the 152 directory. Those updated phones are the same as the IP 500 only they can use 1.5.2 and later because they have 4mb of flash instead of 2mb like the 500's and 300's. Yeah... Those are the ones I used (Thanks, by the way!). - Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 600 phone
Just because the phone has the extra lines doesn't mean you are required to use them. Each line can handle 2 calls. The 600 has a working XML microbrowser, which the 50x does not. The Polycom 501 (not sure if the 500 is the same) doesn't have a place on the phone to plug power in. It gets it through the network cable one way or the other. As purchased, there is a power injector built into the cable they package with the phone. You plug the power from the wall wart into the cable. The special PoE cables they sell are for plugging the phone into the respective PoE standard. So if you're not using 802.3af or Cisco's PoE standard, then you won't need the special cables. Usable is a term relative to your requirements. I've used a Sipura SPA-841 and a Polycom 501. The Polycom is an excellent phone with excellent sound quality. The speaker phone is the best I've ever used (not that I've used alot). It has a nice solid look and feel. I only wish it had a back lit screen, since I keep my office a bit darker than most to reduce my eye strain, which makes reading the reflective LCD difficult. I'd say if you don't need the speaker phone, go with the 301. If you need the speaker phone, but don't need the XHTML microbrowser and/or the extra lines, go with the 501. - Dan Chris Gamble wrote: From their website, the key difference between the polycom 500 and 600 phones is the number of lines they support. What does this mean in terms of asterisk? Do I have to have a seperate extension for each of these lines or ? Also, slightly off-topic, how does the 500 POE optional cable work? Is this similar to have a power box on your desk, or is just a differently configured CAT5 cable? And finally, being poor as we are, are the cheaper (ie non-400$ each) Polycom's usable? Thanks, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any suggestions for an IP phone?
That's just a marketing thing. Just because they are not supported under Asterisk doesn't mean they don't work under Asterisk. It just means don't call them (voipsupply or Polycom) if you have problems getting them to work under Asterisk. Otherwise myself and many others on this list wouldn't be recommending them. My USD0.02. - Dan List Receiver wrote: According to voipsupply.com http://www.voipsupply.com/product_info.php?cPath=95_112products_id=817 --Please Note: Polycom phones are not supported under Asterisk Open Source PBX. Polycom certified platform partners include Path Navigator, Broadsoft, Interactive Intelligence, Sphere, Sylantro, Vertical Networks, VocalData, Alcatel and 3COM. For more information on Polycom supported IP Communications platforms-- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP phone w/ XML browser
Brian Roy wrote: On 7/9/05, Dan Perik [EMAIL PROTECTED] wrote: PJ, You should check out the Polycom 500/501/600. I'm quite sure it has all that (although I don't use all of what you listed). IIRC, the 500's browser is crippled. I think you have to go up to the 600 to get that functionality. -Brian I should have tried it on my 501 before I went and opened my mouth. Sure enough, either it doesn't work, or I'm doing something wrong. The "Services" button is there, and the docs don't say anything about it not working, but even with it configured, it doesn't do anything. Seems to a be a "dead" button. Perhaps some firmware upgrade down the road will "turn it on". Looking through the archives I saw someone report that it did work on the 600, though. - Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP phone w/ XML browser
PJ, You should check out the Polycom 500/501/600. I'm quite sure it has all that (although I don't use all of what you listed). - Dan Pavel Jezek wrote: Still looking for cheaper (under $250,-) alternative to cisco 7940 with features needed for corporate use, mainly: - shared phone book (e.g. via LDAP or XML browser in phone) - in-line power - missed/dialed/received numbers - integrated switch (voice VLAN support) I found only aastara/sayson phone (and Intracom/Netphone in the past), that has xml services anounced, but still not available, so any other recommendation? Seems, that xml minibrowser isn't obvious even in high end phone, but I think that via this function can be phone very extensible... thanks PJ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Sipura SPA-841 Volume Oscillation Problem
Craig wrote: Greetings, We installed a number of SPA-841 in a office environment, originally firmware was 0.9 something and the audio in the headsets worked really well, with virtually no muting of received audio. When I first upgraded a couple of the phones to 3.1.2 the headset audio went to crap, muting incoming audio with any sort of background noise in the room. Speaker phone is totally useless (any sort of room noise-including incoming audio mutes the speaker) so I disconnected the internal microphone and just use the speakerphone mode as a monitor for off hook dialing. Recently upgraded to the new release of 3.1.3 didn't make the headset audio any better (but also didn't seem to make it any worse as some people on the list seem to have observed) We where using the couple of units with 0.9 something for the telephonists with headsets, but the phones crashed a number of times I had to update to the latest firmware now their headsets are pretty much un-usable. I was going to make up an adapter to hook the headset through the handset port as handset audio doesn't seem to go through the same muting problems, but this sort of defeats the purpose of having a simple phone, with speaker (monitor) mode, handset and headset modes built in. Suspect if the put in an option to turn off the audio muting for the headset it would fix our problem, but we are probably going to have to bite the bullet and put a couple of Cisco 7960 in for the headset wearers. cr Wow! This is all very interesting. I had a SPA-841 for about a few days. When I first got it, the headset port worked fine. A few days later it was like the headset port mic connection didn't work anymore. That and the handset mic was problematic, like what you and others have been describing. I sent it back for a refund. Now I wonder if it upgraded its firmware by itself (I didn't do it manually), which would explain why it worked fine in the first place, but then stop working properly. I now have a Polycom 501 on my desk. It's a beautiful phone. The audio quality is the best I've heard in a phone, both from the handset and the speaker phone! I haven't tried the headset port yet, since it's an RJ-11 style, and I only have the 2.5 style headset. I'd be interested to know if the 30x has the same audio quality. I would highly recommend the Polycom to anyone looking at a high-quality VoIP phone. - Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Emergency Asterisk Guru Help needed EMERGENCY
Jeffrey Starin wrote: 911 Help! I accidentially deleted all directories under /var/spool/asterisk I did use the backup facility not too long ago but cannot find the process for restore. However, I don't believe a full restore is needed -- I just need to know the names of the directories under /var/spool/asterisk and re-create them (I hope!). Can some kind soul give me some direction or tell me the directory structure under /var/spool/asterisk? Thanks, B. This is what I got. Note that this is a Gentoo system. Good luck!: gentoo ~ # ls -R /var/spool/asterisk/ -l /var/spool/asterisk/: total 0 drwxr-x--- 2 asterisk asterisk 48 Dec 18 2004 outgoing drwxr-x--- 2 asterisk asterisk 48 Dec 18 2004 qcall drwxr-x--- 2 asterisk asterisk 72 Jun 25 04:56 tmp lrwxrwxrwx 1 asterisk asterisk 37 Jun 25 04:56 vm - /var/spool/asterisk/voicemail/default drwxr-x--- 3 asterisk asterisk 72 Dec 17 2004 voicemail /var/spool/asterisk/outgoing: total 0 /var/spool/asterisk/qcall: total 0 /var/spool/asterisk/tmp: total 0 /var/spool/asterisk/voicemail: total 0 drwxr-x--- 8 asterisk asterisk 192 Mar 22 23:28 default /var/spool/asterisk/voicemail/default: total 0 drwxr-x--- 4 asterisk asterisk 216 Apr 8 22:36 1001 drwxr-x--- 3 asterisk asterisk 160 Mar 22 23:27 1002 drwxr-x--- 3 asterisk asterisk 160 Mar 22 23:27 1003 drwxr-x--- 3 asterisk asterisk 160 Mar 22 23:28 1004 drwxr-x--- 3 asterisk asterisk 160 Mar 22 23:28 1005 drwxr-x--- 3 asterisk asterisk 128 Jun 25 04:56 1234 /var/spool/asterisk/voicemail/default/1001: total 128 drwxr-x--- 2 asterisk asterisk48 Jul 6 18:57 INBOX drwxr-x--- 2 asterisk asterisk48 Jul 6 18:57 Old -rw-r- 1 asterisk asterisk 9306 Mar 22 23:27 busy.gsm -rw-r- 1 asterisk asterisk 6732 Mar 22 23:27 greet.gsm -rw-r- 1 asterisk asterisk 9306 Mar 22 23:27 unavail.gsm -rwxr-x--- 1 asterisk asterisk 96684 Mar 23 00:57 unavail.wav /var/spool/asterisk/voicemail/default/1001/INBOX: total 0 /var/spool/asterisk/voicemail/default/1001/Old: total 0 /var/spool/asterisk/voicemail/default/1002: total 32 drwxr-x--- 2 asterisk asterisk 48 Jul 1 23:24 INBOX -rw-r- 1 asterisk asterisk 9339 Mar 22 23:27 busy.gsm -rw-r- 1 asterisk asterisk 6765 Mar 22 23:27 greet.gsm -rw-r- 1 asterisk asterisk 9339 Mar 22 23:27 unavail.gsm /var/spool/asterisk/voicemail/default/1002/INBOX: total 0 /var/spool/asterisk/voicemail/default/1003: total 32 drwxr-x--- 2 asterisk asterisk 176 Apr 18 07:18 INBOX -rw-r- 1 asterisk asterisk 9471 Mar 22 23:27 busy.gsm -rw-r- 1 asterisk asterisk 6897 Mar 22 23:27 greet.gsm -rw-r- 1 asterisk asterisk 9471 Mar 22 23:27 unavail.gsm /var/spool/asterisk/voicemail/default/1003/INBOX: total 496 -rwxr-x--- 1 asterisk asterisk 41790 Apr 18 07:18 msg.WAV -rwxr-x--- 1 asterisk asterisk 42372 Apr 18 07:18 msg.gsm -rw-r- 1 asterisk asterisk254 Apr 18 07:18 msg.txt -rwxr-x--- 1 asterisk asterisk 410924 Apr 18 07:18 msg.wav /var/spool/asterisk/voicemail/default/1004: total 32 drwxr-x--- 2 asterisk asterisk 48 Mar 22 23:28 INBOX -rw-r- 1 asterisk asterisk 9273 Mar 22 23:28 busy.gsm -rw-r- 1 asterisk asterisk 6699 Mar 22 23:28 greet.gsm -rw-r- 1 asterisk asterisk 9273 Mar 22 23:28 unavail.gsm /var/spool/asterisk/voicemail/default/1004/INBOX: total 0 /var/spool/asterisk/voicemail/default/1005: total 32 drwxr-x--- 2 asterisk asterisk 48 Mar 22 23:28 INBOX -rw-r- 1 asterisk asterisk 9372 Mar 22 23:28 busy.gsm -rw-r- 1 asterisk asterisk 6798 Mar 22 23:28 greet.gsm -rw-r- 1 asterisk asterisk 9372 Mar 22 23:28 unavail.gsm /var/spool/asterisk/voicemail/default/1005/INBOX: total 0 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]
I just looked at their price page. Each package says setup is Free. Now, I do notice that the Price for the pay as you go doesn't have /mth. on it as the others do. So maybe there is a difference. I agree with you that it is not extremely clear and they could do a whole lot better job explaining it. - Dan Andrew Latham wrote: I think the $10 is setup, as you will notice all the others mention the monthly next to the rate. I was confused also. (Hint Teliax) On 6/27/05, John Goerzen [EMAIL PROTECTED] wrote: I'm looking for someone that sells minutes in bulk like LiveVoip used to. No monthly fee, just pay-as-you-go. It looks like Teliax charges a minimum of $10/mo, even if I use no minutes that month. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LiveVoip is Bankrupt
I'm not sure which is funnier... that someone would offer something like that for sale on ebay, or that someone would pay $10.56 + $4.50 shipping to buy it. rofl - Dan Steven Kalcevich wrote: I for one will not be using anymore live voip...I found my own provider. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemcategory=61840item=5783732903rd=1 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UTStarcom F1000 WiFi IP Phone Review
Not always. Some use a www capture page. When you log in through that page, it opens up that mac/ip for a specified length of time. We're doing that here using nocat (http://nocat.net) Without logging in, no traffic goes through from that mac/ip. - Dan Denis Galvão - iSolve wrote: Hi Steve. I think the proxy authorization is just for WWW access(tcp 80 and 443), if some VoIP port is open you will be able to access your provider without auth. Denis. On 25 de jun de 2005, at 02:22, Steve wrote: I keep getting asked by people if these types of wifi phones are capable at all of getting onto the type of wifi network where you have to login via http (web page) such as is typical at many hotels in the us. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Console ALSA Sound
In /etc/asterisk/modules.conf noload = chan_alsa.so noload = chan_oss.so - Dan Conrad Beckert wrote: Hi ... probably one of those RTFM kind of questions (while I'd be happy to know where a good reference FM is :-) ) Has anyone an idea on how to disable the console sound driver. My problem is that a running asterisk is muting my speakers. Thank you in advance for your help Conrad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP-INFO
Just got the same thing here. - Dan Chris Coulthurst wrote: Anyone else unable to get to www.voip-info.org? Site is returning 'connection refused' here. Chris Coulthurst [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *@home .conf files request
I'd be interesting in the same thing. Are they posted anywhere on the web, or anything (I've looked, but not found). Thanks, - Dan Luis Diaz wrote: hi all, can anyone emailme the .conf of asterisk at home, i cant download the full size tar or iso because of a network problem that fu*** every big file download and i just wanna learn not change my distro bye and thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0.7 on Gentoo
Waldo Rubinstein wrote: I installed Asterisk on Gentoo using emerge. At first, emerge tried installing version 0.9 but reading the wiki showed how to get the latest stable. I'm running Gentoo kernel 2.6.11-gentoo-r9. Asterisk seems to be working just fine, but I'm concerned that since I don't have any Digium hardware, I may need a timer source. When I executed emerge zaptel, it installed zaptel 1.0.7 as well. The problem is that I can't seem to be able to load ztdummy or any zaptel module. I'm running * on Gentoo. Just a shot in the dark here. Have you tried: /etc/init.d/zaptel start Then do your modprobe. Let us know what happens. - Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice - Customer feedback
I haven't been having any problems. Since it's for home use (and my daughters aren't teenagers yet :-) ), I don't have a lot of traffic, but I've had good success. - Dan Sean Kennedy wrote: Hi all, Can any broadvoice customers give me their opinions on the service recently? It's actually been pretty quiet on the list lately regarding them, so it seems to me that they're either getting things straightened out or everyone has dropped the service. Thank you in advance Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tools for effectively manage Asterisk
Dustin Wildes wrote: I feel there is nothing wrong with having a web-based configuration utility, if set up correctly. Look at the WRT54G Linksys router, plus other countless devices that use an embedded browser for configurations. Just a nitpick, if I may. They have embedded http servers, not browsers. But I'm sure that's what you meant. Having said that, I agree that putting streamlined apache/php on an * box isn't going to cause grief. Heck, I'm breaking lots of rules, and haven't running into problems (yet). I run _everything_ on my Athlon 3000+/1GB Gentoo machine. Apache, postfix, named, mysql, courier-imap, firebird / avg tcp server, nagios, samba, X/Gnome, and vncserver/Gnome! I even (gasp) play some games on it. I'm sure that slows down some of the server functions, but I haven't noticed any problems (yet). I'm hoping to get my own dedicated server box soon to offload all the non-client stuff, but until then, it all goes on this one machine. Yes, this is a home setup, but with ties to work functions. - Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CRM integration (was RE: CallerID)
Rusty Shackleford wrote: It doesnt seem to be complicated but for example, the things that bother me are refreshes, I dont want to use meta refreshes for this monitoring webpage every X seconds, rather, use something more realtime... Any ideas? And that's the real trick. Web browsers, unless they are instructed to do otherwise, don't DO anything once they've completed loading a page. So without instructing them to refresh, they aren't going to be aware of a server-side change, such as an incoming call. For that, you're going to have to have some way of sending a message TO the client machine, have it received by that machine, and have that client machine take the desired action (pop up an incoming call dialog, load a contact record, etc.). http://wp.netscape.com/assist/net_sites/pushpull.html Would that work? Especially the "server push". Not sure if current browsers like it or not. I've never tried it, but came across this document, and thought it may be something useful. - Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Traffic shaping for IAX and SIP calls through Asterisk?
Mike Dent wrote: Hi, Is it possible to put some kind of bridge which will do traffic shaping/prioritising between my 6 external IP addresses and my PPPoA modem interface? My other option is to put some kind of device at the edge of all my networks to shape the traffic in/out. I'd rather do it in one box if possible? thanks Mike http://lartc.org/wondershaper/ That's a good place for starting with traffic shaping, in addition to Luki's link. - Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Other memory stuff
Wiley Siler wrote: Other memory stuff On a similar note, I have a server with 1GB of memory that seems to never release the memory back to system use. Is it just Linux using it for buffers/caching. My system always shows lots of memory being used.: total used free shared buffers cached Mem: 904752 881060 23692 0 94384 413448 -/+ buffers/cache: 373228 531524 Swap: 524280 3600 520680 But most of it is being used for buffers/cache. - Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A@H Email Relay
Patrick M. Gray, Jr. wrote: My [EMAIL PROTECTED] box sits behind a firewall and needs to use an internal host to relay all email (voicemail notifications). I cant for the life of me find where to make this setting, as Im used to postfix MTAs. Use something like ssmtp to route outgoing email for your * machine through your mail hub. Asterisk doesn't have it's own smtp client, it just uses the local machines MTA. - Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Broadvoice is finally starting to give answers
trixter http://www.0xdecafbad.com wrote: They paid 100% of the *UNDISPUTED* charges but nothing is said about the disputed ones. Typo or intentional? It also sounds to me like its an access charge issue, but I may be reading too much into this. Sounds like BroadVoice paid their bill according to their interpretation of their contract with the carrier. And that the carrier interpreted the contract differently and billed them a significantly larger amount (thus the use of the word undisputed). It also sounds like this dispute went on for quite a while. The carrier finally pulled the plug. It also sounded like the carrier is initiating a lawsuit against BroadVoice. So to sum up, it seems like a basic contract dispute. - Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AAH lockup
Sounds like a kernel lock up. After you've rebooted, check out /var/log/messages to see what happened. - Dan [EMAIL PROTECTED] wrote: I don't know if this is related, but the last two mornings I've come in, the newer AAH 1.0 computer has been locked-up. The Caps Lock and Scroll Lock lights on the keyboard are flashing (apparently in a specific pattern). The computer is a HP 7960 w/ ASUS mobo, P4, 1.3Ghz, 256MB RDram. Not being a Linux person, I don't know if this is a Linux issue and/or a hardware issue. Is there a specific log I can look at that might tell me what happened? Thanks, Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice Issues
I look on their site, and the only thing is to sign up or sign in. No other marketing material. Rate plans, rates, products, services, etc. Am I missing something? - Dan Derek Whitten wrote: nufone has been rock solid http://www.nufone.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO ATA?
Joseph wrote: snip The AG-168 supports IAX2 and the FXO port is pass though type. The difference is that SPA-3000 answer the phone and rings asterisk (the phone at this moment has been answered the ringing party is incurring the charges before asterisk answered the phone), the AG-168 is ringing the asterisk directly, so I think the pass through port is a benefit in this case for asterisk users. It is possible to pass through an incoming call to Asterisk without the Sipura answering it, although it does take some contortions. Basically, you have the FXO port add a character to the beginning of the CIDNumber (I picked Z). Then, for the FXS port, have it conditionally forward a call to * if it has that character at the beginning of the CIDNumber. Since all calls coming in from the FXO port would have that character, but no other calls would, it effectively makes the call pass through to * without answering it. See: http://www.voip-info.org/wiki-Sipura+3000 It took me a few tries to get the settings right, but in the end it works well. Also, in addressing the post about the Handytone 488... I had one for a week. Either I had a bad one, or the item needs a bit more work to be marketed as something anyone would want to rely on. I ended up returning mine. But it did seem to only pass the call to Asterisk after about 4 rings. - Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Timeout
Joseph wrote: On Mon, 2005-05-02 at 12:40 -0400, Dan Levine wrote: The Box Itself doesn't get a new IP address, the router does. What I'm looking to do is have the IAX connection re-register every hour or so. Is this the right idea? Why not get a static IP? If that is not available, some routers have a capabilities to update your DNS setting and send you an email. Nest, you can scan your email for an IP from certain provider if it doesn't match lunch/execute a script that will reload asterisk. I've never done it but I know it is possible to do it. Better yet. There are client programs out there which can read your ip off many routers. I've used ddclient specifically. You could hack that to reregister your IAX when the IP changes. (Although a static IP would come in mighty handy.) - Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fedora Core 3 Shorewall Install
Not sure about FC3 issues, but I use Shorewall on the border. My * server is a 1-1 NAT inside. Here's all I need in /etc/shorewall/rules: # for SIP, IAX2, IAX, RTP, MGCP ACCEPT net loc:192.168.1.5 udp 5060,4569,5036,1:2,2727 - - - - I probably don't need all that, since I'm not running some of it, but it works for me. If your firewall is your * machine, you probably will need something like $FW instead of loc:192.168.1.5, and maybe all instead of net. HTH, (De nada) - Dan Anonymous Account wrote: Dear asterisk-users, Allow me to preface this newbie's question with a statement: 1. I searched the archives the Wiki 2. I Googled until I couldn't Google anymore My questions concern the installation of the latest/greatest Asterisk on Fedora Core 3 with a Shorewall (Shoreline) Firewall installed. I haven't been able to find a step-by-step howto that is CURRENT that addresses this particular configuration. Does anyone have a link they could point me to? Please keep in mind the word current and by that I mean something that takes into account that I am using a Kernel that is 2.6+ and that Shorewall is version 2.2+ Mucho Gracias, amigos! 031547 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need info : lspci
Clone here as well. :00:0a.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface - DAn Marco Supino wrote: Hi, I need some info from people with the x100p card (digium or clone), please send me the output of lspci and lspci -n from your linux machine, i am tring to find out something on my * server. Thanks. Marco. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BYOD provider other than broadvoice
Michael Lyszczek wrote: I have broadvoice and they suck lately. Can you elaborate? - Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] US$200 bounty for * paging feature
Michael D Schelin wrote: Ok you guys enough. The debate will go on forever. Agreed! At the risk of wasting bandwidth myself Please, guys stop wasting my precious bandwidth. If you want to private message your flames, great but leave this list to Asterisk, please. Thanks! - Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream HandyTone-488, * - FXO problems
Pardon my answering myself (and for the long post). But I do have it sort of working, and I come back with information on the GS HT-488, as well as questions related to SIP / DTMF issues. The GS HT-488 acts as a PSTN pass through device for 4 rings. If the phone attached to the FXS port hasn't picked up by 4 rings, it will by default answer, and you're at an internal (*) dial tone. You can also configure the HT-488 to dial a specific extention, which it will then do instead of dropping you at an internal dial tone. From there you can obviously do what ever you want with the call. (It would be nice if you could configure and/or disable the # rings before it switches over to VoIP. Maybe that will be something they will add to a firmware update someday.) For dialing out, you set up an extention for the FXO port, and dial that. It will ring once, and then present you with the PSTN line, dial tone and all. From there you (should be) are able to dial out. Now, here is my problem and question. Both the FXS and FXO ports are set up to use SIP INFO for DTMF. You would think that when you have dialed the FXO port, and are at the PSTN dial tone, the HT-488 will translate the SIP DTMF INFO passed through to the FXO port as audible DTMF on the PSTN line. This is not the case. So I really can't make outgoing calls yet. Now, I can change the FXS line to send DTMF in audio, which works, but I figure that sending DTMF in audio is not ideal. So I'm trying to translate the SIP DTMF INFO to DTMF in-audio. I've tried a few combinations of SipDTMFMode(inband) (trying to do a DTMF style translation, I guess), and Dial(SIP/gs1-FXO,10,D(PSTNnumber) ), but can't get it to work. Should I just suck it up and keep the FXS port using DTMF in-audio, or is there a way to get SIP DTMF INFO translated to DTMF tones in audio in the Dial settings for the FXO extension? Thanks! Dan Dan Perik wrote: I just got my shiny new Grandstream HandyTone-488 today. My goal is to use it to allow incoming/outgoing calls to PSTN using my normal ole' phone as usual. I will be switching over to using BroadVoice as my main phone #, but want that to be as seemless of a switchover as possible (for the wife and kids, and for people needing to call us). I've got the following working: FXS - * ( and then - BroadVoice ) ( BroadVoice - ) * - FXS FXO - * ( and then - FXS ) I don't have this working: ( FXS - ) * - FXO In other words, I can't seem to call out on my PSTN line from Asterisk. snip ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk - Altigen
Hi, If this belongs on a different list, please let me know. I oversee an Altigen IP-based PBX. We're wanting to make VoIP calls through the Internet out to PSTN via a service like BroadVoice or similar. I think Asterisk is the ticket of this. I have successfully configured Asterisk to dialout/dialin on BroadVoice, FWD, etc. to/from X-Lite softphone. Altigen uses H.323, and can be configured for IP-based trunk access. Does anyone know if it would work to have the Altigen system trunk out calls via Asterisk as a gateway, and then Asterisk can connect them out via BroadVoice. Has anyone successfully tied together an Altigen system to an Asterisk system using VoIP (ie. not using hardware (FXO/FXS cards, etc.))? Thanks, Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
