[asterisk-users] Outbound fax issues
Hello all. I have the following setup: Fax machine | Sipura SPA-3120 | SIP 100BaseT | Asterisk 1.4 | IAX2 100BaseT | Asterisk 1.6 | ISDN PRI TE210P | Traditional Telco The fax lands on the Internal Asterisk 1.4 box, the sip config for this extension looks like: [35081] type=friend secret= qualify=yes port=5060 nat=no host=dynamic dtmfmode=rfc2833 dial=SIP/35081 context=fax-line canreinvite=no callerid=device 35081 disallow=all allow=ulaw allow=alaw Now, inbound faxing (from the other side of the Telco to me) is working and from what I can tell, never fails to receive. Sending though is a bit touch and go. Sometimes works, sometimes doesn't. It's about a 40% success rate, and does not seem to depend on what number dialed (ie, the problem has been basically isolated to my internal network). The symptoms are long handshake times with the fax trying to get carrier, then failing. Redialing 2-6 times eventually gets the fax through. The extensions.conf simply answers and dials out through the ISDN line. No special config here. I believe from what I have read via our friends at Google and voip forums around the place, that it is probably an echo or jitter problem, but what I have found so far has been a bit vague. Does anyone have any pointers on what I should be looking for to improve the outbound call? Thanks! Mikel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and Dabatase
Hi All; Anyone knows if there is an Asterisk version that setup can be stored in Database instead of the configuration files (.conf)? Any advise? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Dabatase
Why? Maybe I don't understand your needs. Is there some requirement you are trying to meet? Like tracking versions/changes who/what/when... .conf files are just text. Database files you would have to update by a database interface/program/query. Database are for relational data. Conf settings are NOT relational in this sense. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Sunday, December 21, 2008 01:52 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk and Dabatase Hi All; Anyone knows if there is an Asterisk version that setup can be stored in Database instead of the configuration files (.conf)? Any advise? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Good comparisons on cheaper VOIP phones
Hello list, I am doing some work for a non profit group. As part of this, I am going to be putting in a 30 handset Asterisk solution. We are trying to keep the costs down as much as possible, as this job includes cabling, I am looking at POE solutions. On the switch side, I am considering something like some Netgear ProSafe FS726TP 24 port switches, or maybe the equivalent Linksys SRW224MP 24 port switch. About 4 of these will run the phones and computers on the network connecting back to a gigabit switch handling the phone and other servers. On the phone side VOIP phones The price range sort of limits me to: * Aastra 9112i * Snom 300 * Polycom 320 * Cisco CP-7906G (But I believe this won't handle SIP out of the box?) Any good bad stories of the above? One thing I like about the Aastra is being able to go POE from a switch, to the Aastra, then out of the second port on the Aastra and into the PC. Regards Mikel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good comparisons on cheaper VOIP phones
Linksys SPA900 Series IP phones are good and run on POE w/ the built in switch. The have both just make sure you get the ones w/ the switch inclusive. They have some cheap 1, 2, 4, 6 line phones. We have this setup w/ SRW2008MP (This is only 8 port, but have up to 48 port POE). This way you can stick w/ one vendor for your VOIP. Makes support much easier when integrating and you don’t get finger pointing. Only other phone I could really recommend is the PolyCom they seem to be decent. DO NOT get phone w/ the power brick. Steve Wofford www.uctrlit.com From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mikel Lindsaar Sent: Sunday, December 21, 2008 02:00 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Good comparisons on cheaper VOIP phones Hello list, I am doing some work for a non profit group. As part of this, I am going to be putting in a 30 handset Asterisk solution. We are trying to keep the costs down as much as possible, as this job includes cabling, I am looking at POE solutions. On the switch side, I am considering something like some Netgear ProSafe FS726TP 24 port switches, or maybe the equivalent Linksys SRW224MP 24 port switch. About 4 of these will run the phones and computers on the network connecting back to a gigabit switch handling the phone and other servers. On the phone side VOIP phones The price range sort of limits me to: * Aastra 9112i * Snom 300 * Polycom 320 * Cisco CP-7906G (But I believe this won't handle SIP out of the box?) Any good bad stories of the above? One thing I like about the Aastra is being able to go POE from a switch, to the Aastra, then out of the second port on the Aastra and into the PC. Regards Mikel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Dabatase
On Sun, Dec 21, 2008 at 01:52:17AM -0800, bilal ghayyad wrote: Hi All; Anyone knows if there is an Asterisk version that setup can be stored in Database instead of the configuration files (.conf)? Err... this is basically supported as of 1.2 (Real-Time, ARA). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Dabatase
On Sun, Dec 21, 2008 at 9:52 AM, bilal ghayyad bilmar...@yahoo.com wrote: Anyone knows if there is an Asterisk version that setup can be stored in Database instead of the configuration files (.conf)? Yes all current versions can. Take a look at Asterisk Realtime http://www.voip-info.org/wiki-Asterisk+RealTime Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good comparisons on cheaper VOID phones
As part of this, I am going to be putting in a 30 handset Asterisk solution. We are trying to keep the costs down as much as possible, as this job includes cabling, I am looking at POE solutions. On the switch side, I am considering something like some Netgear ProSafe FS726TP 24 port switches, or maybe the equivalent Linksys SRW224MP 24 port switch. About 4 of these will run the phones and computers on the network connecting back to a gigabit switch handling the phone and other servers. On the phone side VOIP phones The price range sort of limits me to: * Aastra 9112i * Snom 300 * Polycom 320 * Cisco CP-7906G (But I believe this won't handle SIP out of the box?) With the exception of mass market stuff like their ATA's (which are actually all right) stay clear of anything Linksys. It's over priced, over marketed and undelivered, poorly supported crap. Too add to your owes a lot of it is very proprietary and designed to lock you in to Linksys. Some of the above could also be said about Cisco though I do have a Cisco router I am very happy with and they are the gold standard. Having said that you are not a carrier or an ISP, and you are obviously on a limited budget, so I would not use any of their VoIP stuff in your situation. The only Snom phone I ever used was a total piece of s*. Having said that some people seem to like them - I don't know why. If you can get Netcomm where you are I'd recommend their V90S phone. Over all it's a nice phone, works well, quite classy and importantly not proprietary. It does have a couple of things that need improvement on - namely that you have to set up multiple SIP accounts on the phone to support conferencing (It's conferencing button doesn't function with server based conference) and it's web based GUI doesn't work with Firefox atm. MSIE needed for Windows or Konqurer on Linux works fine. If you want these phones and you can't find them locally drop me an email. Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good comparisons on cheaper VOIP phones
Anyone know if the Cisco 7940 would work?Ronny -- Original message from "Mikel Lindsaar" raasd...@gmail.com: -- Hello list, I am doing some work for a non profit group.As part of this, I am going to be putting in a 30 handset Asterisk solution. We are trying to keep the costs down as much as possible, as this job includes cabling, I am looking at POE solutions. On the switch side, I am considering something like some Netgear ProSafe FS726TP 24 port switches, or maybe the equivalent Linksys SRW224MP 24 port switch. About 4 of these will run the phones and computers on the network connecting back to a gigabit switch handling the phone and other servers. On the phone side VOIP phonesThe price range sort of limits me to:* Aastra 9112i* Snom 300* Polycom 320* Cisco CP-7906G (But I believe this won't handle SIP out of the box?) Any good bad stories of the above?One thing I like about the Aastra is being able to go POE from a switch, to the Aastra, then out of the second port on the Aastra and into the PC. RegardsMikel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Dabatase
In case I have a web based interface, so some configuration are going to be added via the web, I was looking to be inserted in the database. What do u suggest a solution? Regards Bilal --- On Sun, 12/21/08, Steve Wofford s...@uctrlit.com wrote: From: Steve Wofford s...@uctrlit.com Subject: RE: [asterisk-users] Asterisk and Dabatase To: bilmar...@yahoo.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sunday, December 21, 2008, 4:55 AM Why? Maybe I don't understand your needs. Is there some requirement you are trying to meet? Like tracking versions/changes who/what/when... .conf files are just text. Database files you would have to update by a database interface/program/query. Database are for relational data. Conf settings are NOT relational in this sense. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Sunday, December 21, 2008 01:52 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk and Dabatase Hi All; Anyone knows if there is an Asterisk version that setup can be stored in Database instead of the configuration files (.conf)? Any advise? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Dabatase
On Sun, Dec 21, 2008 at 03:19:42AM -0800, bilal ghayyad wrote: In case I have a web based interface, so some configuration are going to be added via the web, I was looking to be inserted in the database. While Asterisk can read configuration directly from a DB, it means that you have to cook some specific DB tables for it. It also means that if the DB is ever down, Asterisk is down. An alternative is to generate configuration files from Asterisk. A third option is to generate those tables and read them at configuration read time (static real time). Requires explicit load, but it also means that Asterisk knows what its configuration is. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good comparisons on cheaper VOIP phones
On Sun, 2008-12-21 at 21:00 +1100, Mikel Lindsaar wrote: Hello list, I am doing some work for a non profit group. As part of this, I am going to be putting in a 30 handset Asterisk solution. We are trying to keep the costs down as much as possible, as this job includes cabling, I am looking at POE solutions. On the switch side, I am considering something like some Netgear ProSafe FS726TP 24 port switches, or maybe the equivalent Linksys SRW224MP 24 port switch. About 4 of these will run the phones and computers on the network connecting back to a gigabit switch handling the phone and other servers. On the phone side VOIP phones The price range sort of limits me to: * Aastra 9112i * Snom 300 * Polycom 320 * Cisco CP-7906G (But I believe this won't handle SIP out of the box?) Mentioning costs, one might be tempted to mention grandsteam, but for some people, those have a bad reputation, although i have two of thos phone for over two years without any problem.. OTOH, why not consider the Siemens A580-IP? Recently i bought a package, containing the DECT-base-station (direct IP-interface) and two handsets, (each 6 sip-entries), two handset-chargers for about 100 Euro's. Audio-quality is good. Don't think you can buy SIP-phones any cheaper... hw ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Dabatase
Where this link? Did not see or receive any? In case of using database, how can I determine that my configuration be taken from DB or from .conf files? Where this can be setted? And is it possible to take confiruation from both? If any recorded added for DB, then I need to do a reload to become effected? Regards Bilal --- On Sun, 12/21/08, Steve Wofford s...@uctrlit.com wrote: From: Steve Wofford s...@uctrlit.com Subject: RE: [asterisk-users] Asterisk and Dabatase To: bilmar...@yahoo.com Date: Sunday, December 21, 2008, 6:23 AM I just didn't want you to get carried away as a database isn't typically required for base setups. They are good for certain situations. A database also offers the ability for multi user or process availability whereas a text file can become locked. You can also use different setting on the fly where that becomes difficult w/ .conf The previous posters sent a link to the database component written by someone that looks like that would be perfect fit. STeve -Original Message- From: bilal ghayyad [mailto:bilmar...@yahoo.com] Sent: Sunday, December 21, 2008 03:20 To: Asterisk Users Mailing List - Non-Commercial Discussion; Steve Wofford Subject: RE: [asterisk-users] Asterisk and Dabatase In case I have a web based interface, so some configuration are going to be added via the web, I was looking to be inserted in the database. What do u suggest a solution? Regards Bilal --- On Sun, 12/21/08, Steve Wofford s...@uctrlit.com wrote: From: Steve Wofford s...@uctrlit.com Subject: RE: [asterisk-users] Asterisk and Dabatase To: bilmar...@yahoo.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sunday, December 21, 2008, 4:55 AM Why? Maybe I don't understand your needs. Is there some requirement you are trying to meet? Like tracking versions/changes who/what/when... .conf files are just text. Database files you would have to update by a database interface/program/query. Database are for relational data. Conf settings are NOT relational in this sense. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Sunday, December 21, 2008 01:52 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk and Dabatase Hi All; Anyone knows if there is an Asterisk version that setup can be stored in Database instead of the configuration files (.conf)? Any advise? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SMS text messaging capabilities
On Sat, 2008-12-20 at 19:27 +0200, Elliot Murdock wrote: Hello! What kind of sms text messaging capabilities does Asterisk have? I do not know very much about about SMS technology, but I am looking for the following features: 1. mobile SIP devices can send and receive SMS messages 2. Asterisk server be able to accept and send SMS messages through PRI lines and Internet connections. I noticed that Asterisk has an SMS function, but I am not farmiliar enough with that technology to make it useful. Any help with this would be great! Hi Elliot, sms-service is included in Asterisk since 1.2. Just tried it out, (from cli): smsq --motx-channel=mISDN/1/067364 061368506 testje =first number is the fixed-sms-provider (here, KPN) =second number is the target Just taken rightout from the wiki pages. I presume one could set the MSN-as well, hw ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good comparisons on cheaper VOIP phones
Mentioning costs, one might be tempted to mention grandsteam, but for some people, those have a bad reputation, although i have two of thos phone for over two years without any problem.. OTOH, why not consider the Siemens A580-IP? Recently i bought a package, containing the DECT-base-station (direct IP-interface) and two handsets, (each 6 sip-entries), two handset-chargers for about 100 Euro's. Audio-quality is good. Don't think you can buy SIP-phones any cheaper... My experience with Grandstream is that are one of the better 'cheap' ones, but cheap non the less. Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good comparisons on cheaper VOIP phones
On Mon, 22 Dec 2008, Michael wrote: Mentioning costs, one might be tempted to mention grandsteam, but for some people, those have a bad reputation, although i have two of thos phone for over two years without any problem.. OTOH, why not consider the Siemens A580-IP? Recently i bought a package, containing the DECT-base-station (direct IP-interface) and two handsets, (each 6 sip-entries), two handset-chargers for about 100 Euro's. Audio-quality is good. Don't think you can buy SIP-phones any cheaper... My experience with Grandstream is that are one of the better 'cheap' ones, but cheap non the less. I've deployed a few 100 Grandstreams and generally been happy with them. Maybe I've just been lucky. They do have their quirks though, but they are very easy to use and setup. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Dabatase
On Sun, Dec 21, 2008 at 11:57 AM, bilal ghayyad bilmar...@yahoo.com wrote: Where this link? Did not see or receive any? In case of using database, how can I determine that my configuration be taken from DB or from .conf files? Where this can be setted? And is it possible to take confiruation from both? If any recorded added for DB, then I need to do a reload to become effected? Getting Asterisk up and running with a database configuration (realtime) is not something that you will be able to do quickly or easily. If you are new to Asterisk I'd highly recommend getting a grip on running Asterisk with the configuration files before moving onto realtime. If you're not too interested in the inner workings and do just want to get something up and running pronto take a look at the AsteriskNow or TrixBox products. They tend to provide a higher level abstraction to running Asterisk and have a web page based configuration approach. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Dabatase
I am interested in the inner working and I would like to go deep, so how can I start with letting asterisk work based on the database and work realtime? About AsteriskNow, it is licensed as I know. Regards Bilal --- On Sun, 12/21/08, Grey Man greymanv...@gmail.com wrote: From: Grey Man greymanv...@gmail.com Subject: Re: [asterisk-users] Asterisk and Dabatase To: bilmar...@yahoo.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sunday, December 21, 2008, 7:33 AM On Sun, Dec 21, 2008 at 11:57 AM, bilal ghayyad bilmar...@yahoo.com wrote: Where this link? Did not see or receive any? In case of using database, how can I determine that my configuration be taken from DB or from .conf files? Where this can be setted? And is it possible to take confiruation from both? If any recorded added for DB, then I need to do a reload to become effected? Getting Asterisk up and running with a database configuration (realtime) is not something that you will be able to do quickly or easily. If you are new to Asterisk I'd highly recommend getting a grip on running Asterisk with the configuration files before moving onto realtime. If you're not too interested in the inner workings and do just want to get something up and running pronto take a look at the AsteriskNow or TrixBox products. They tend to provide a higher level abstraction to running Asterisk and have a web page based configuration approach. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Dabatase
On 21 Dec 2008, at 12:39, bilal ghayyad wrote: I am interested in the inner working and I would like to go deep, so how can I start with letting asterisk work based on the database and work realtime? About AsteriskNow, it is licensed as I know. Regards Bilal http://tinyurl.com/6t9x9b ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good comparisons on cheaper VOIP phones
Michael wrote: My experience with Grandstream is that are one of the better 'cheap' ones, but cheap non the less. I am yet to run into a worse IP phone than the Grandstreams - although having said that, I should say that I've always steered clear of most of the Chinese no-name brand phones. They're unstable, temperamental and upgrading the firmware is a crapshoot half the time since you never know what new bugs will be introduced and quite often you can't downgrade the firmware if you don't like the newer firmware. My suggestion would be to look at the Snom 300 (although they are very simplistic phones), the Polycom IP330 (I have a feeling the 320s don't support PoE) or the Linksys phones. I noted an earlier post saying that these phones were overpriced and designed to lock you in to Linksys gear - my experience has been completely different. The SPA-942 is quite cheap and integrates nicely with Asterisk. The SPA-962 is considerably more expensive - but considering the size of the colour LCD screen, they're not that badly priced. (as an aside, the button banks for the SPA-962 are one of the /cheapest/ available!) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good comparisons on cheaper VOIP phones
On Dec 21, 2008, at 8:27 AM, Rob Hillis wrote: Michael wrote: My experience with Grandstream is that are one of the better 'cheap' ones, but cheap non the less. I am yet to run into a worse IP phone than the Grandstreams - although having said that, I should say that I've always steered clear of most of the Chinese no-name brand phones. They're unstable, temperamental and upgrading the firmware is a crapshoot half the time since you never know what new bugs will be introduced and quite often you can't downgrade the firmware if you don't like the newer firmware. My suggestion would be to look at the Snom 300 (although they are very simplistic phones), the Polycom IP330 (I have a feeling the 320s don't support PoE) or the Linksys phones. I noted an earlier post saying that these phones were overpriced and designed to lock you in to Linksys gear - my experience has been completely different. The SPA-942 is quite cheap and integrates nicely with Asterisk. The SPA-962 is considerably more expensive - but considering the size of the colour LCD screen, they're not that badly priced. (as an aside, the button banks for the SPA-962 are one of the cheapest available!) One person's trash is another's treasure. I've used many linksys phones, including the SPA962 and found the sound quality and usage to be simply sub-par. In several set-up's I found the sound quality of a Grandstream 286 ATA to be much better than a SPA962 IP phone. But, I agree in I find the polycom to be amazing phones — and you end up paying for it (although on long term cost of ownership it might not be that bad. I bought a Polycom 601 years ago, use it heavily, and it sounds just as good today as the day I bought it. The SPA962 went on ebay within 3 months of me buying it. I have a few grandstream 286's I like to use for traveling and placing in remote areas of an installation. Fred Posner ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good comparisons on cheaper VOIP phones
On Mon, 22 Dec 2008 02:27:29 you wrote: Michael wrote: My experience with Grandstream is that are one of the better 'cheap' ones, but cheap non the less. I am yet to run into a worse IP phone than the Grandstreams - although having said that, I should say that I've always steered clear of most of the Chinese no-name brand phones. They're unstable, temperamental and upgrading the firmware is a crapshoot half the time since you never know what new bugs will be introduced and quite often you can't downgrade the firmware if you don't like the newer firmware. +1 I STRONGLY recommend to the O.P. that whatever they do, whatever path they decide to take, that they *only* buy one or two units to test, and test them fully, until they are absolutely sure the item is not a POS. Nothing worse then being stuck with 30x POS. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good comparisons on cheaper VOIP phones
One person's trash is another's treasure. I've used many linksys phones, including the SPA962 and found the sound quality and usage to be simply sub-par. In several set-up's I found the sound quality of a Grandstream 286 ATA to be much better than a SPA962 IP phone. But, I agree in I find the polycom to be amazing phones — and you end up paying for it (although on long term cost of ownership it might not be that bad. I bought a Polycom 601 years ago, use it heavily, and it sounds just as good today as the day I bought it. The SPA962 went on ebay within 3 months of me buying it. I have a few grandstream 286's I like to use for traveling and placing in remote areas of an installation. Fred Posner 3 months... that long? I have a Linksys SPA9000 IP PBX I want to quit. Mint condition with all packing etc. Nothing 'wrong' with it (except that it's a Linksys) I just hate proprietary stuff which is what the Linksys is. Still it may well suit someone present on the list who doesn't mind that they will have to buy Linksys brand phones to work with it, and they want it because of the nifty any n00b can use this set up utility. Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good comparisons on cheaper VOIP phones
Michael wrote: I bought it. The SPA962 went on ebay within 3 months of me buying it. I have a few grandstream 286's I like to use for traveling and placing in remote areas of an installation. 3 months... that long? Again I'm surprised. I've had no problems at all with the Linksys phones connected to an Asterisk system. My list of irritants with the phone is pretty low - you can't use the line buttons as BLF buttons and localising tones is rather painful. They're not in the same class as Polycoms when it comes to hands-free (but then again, basically nothing else is) but the hands-free is quite usable. I have a Linksys SPA9000 IP PBX I want to quit. Mint condition with all packing etc. Nothing 'wrong' with it (except that it's a Linksys) I just hate proprietary stuff which is what the Linksys is. Ahh... that's a bit different. Yes, the SPA-9000s are an overpriced pain in the ass, but the phones certainly don't fall into the same category. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Dabatase
Bilal, I've explored two ways of working with the database, as far as dialplan goes. The first was using realtime, as described in the wiki, to store the dialplan. This became very messy very quickly, as an SQL database isn't a great support for the dialplan. It's (relatively) hard to modify, and if your goal is to build your own Web portal (which was my goal) you're stuck making the portal code very complexe to accommodate the messy dialplan. What I went with was the complete opposite. I build the dialplan in the conf files, but everytime I refered to something dynamic, I called an SQL function (see MYSQL cmd for Asterisk). This made the dialplan more complexe (lot's of call to MYSQL) but it's more easily changed (since it's still text) and the Web portal is much readable since everything relies a well-designed relational DB instead of the dialplan realtime database. But there are donwsides to that too: MYSQL down means my VoIP server is also down (which never actually happened, but it could). And, as I said, the .conf file is a lot more complexe As for the other parts of Asterisk (SIP accounts, queue data, etc) I did use Asterisk realtime DB for that. Be aware that, as of 1.4.22, there are limitations with using SIP accounts in a realtime DB (for example, the SIPPEER func doesn't seem to work, which is a darn shame, and hints don't quite work too unless they are in a conf file. That last issue seems to be fixed in the latest 1.6, but I haven't tried it)... Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Sunday, December 21, 2008 7:40 To: Asterisk Users Mailing List - Non-Commercial Discussion; Grey Man Subject: Re: [asterisk-users] Asterisk and Dabatase I am interested in the inner working and I would like to go deep, so how can I start with letting asterisk work based on the database and work realtime? About AsteriskNow, it is licensed as I know. Regards Bilal --- On Sun, 12/21/08, Grey Man greymanv...@gmail.com wrote: From: Grey Man greymanv...@gmail.com Subject: Re: [asterisk-users] Asterisk and Dabatase To: bilmar...@yahoo.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sunday, December 21, 2008, 7:33 AM On Sun, Dec 21, 2008 at 11:57 AM, bilal ghayyad bilmar...@yahoo.com wrote: Where this link? Did not see or receive any? In case of using database, how can I determine that my configuration be taken from DB or from .conf files? Where this can be setted? And is it possible to take confiruation from both? If any recorded added for DB, then I need to do a reload to become effected? Getting Asterisk up and running with a database configuration (realtime) is not something that you will be able to do quickly or easily. If you are new to Asterisk I'd highly recommend getting a grip on running Asterisk with the configuration files before moving onto realtime. If you're not too interested in the inner workings and do just want to get something up and running pronto take a look at the AsteriskNow or TrixBox products. They tend to provide a higher level abstraction to running Asterisk and have a web page based configuration approach. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good comparisons on cheaper VOIP phones
My suggestion would be to look at the Snom 300 (although they are very simplistic phones), the Polycom IP330 (I have a feeling the 320s don't support PoE) or the Linksys phones. Polycom 320 does support PoE (I have a few deployed) but don't include the 10/100 switch. That is the only diff between the 330 and 320. So if you're sharing a wall jack between a VoIP phone and a PC, you're better off with the 330. If not, the 320 is noticeably cheaper. That being said, unless money is tight (which happens), go with a Polycom 550 or even a 650.The bigger screen makes juggling calls easier. Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good comparisons on cheaper VOIP phones
On Sun, Dec 21, 2008 at 10:08 AM, Mike l...@virtutel.ca wrote: My suggestion would be to look at the Snom 300 (although they are very simplistic phones), the Polycom IP330 (I have a feeling the 320s don't support PoE) or the Linksys phones. Polycom 320 does support PoE (I have a few deployed) but don't include the 10/100 switch. That is the only diff between the 330 and 320. So if you're sharing a wall jack between a VoIP phone and a PC, you're better off with the 330. If not, the 320 is noticeably cheaper. That being said, unless money is tight (which happens), go with a Polycom 550 or even a 650.The bigger screen makes juggling calls easier. I would personally call VoIPSupply and other vendors. I mention VoIPSupply because they often post to the biz list with extra inventory or refurbs and a good price. I picked up 10 Polycom 600s I believe for $150 each. End of fiscal year or quarter, they need to make numbers.. Mike -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Dabatase
Steve Howes schrieb: On 21 Dec 2008, at 12:39, bilal ghayyad wrote: I am interested in the inner working and I would like to go deep, so how can I start with letting asterisk work based on the database and work realtime? http://tinyurl.com/6t9x9b LOL Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Dabatase
On Sun, Dec 21, 2008 at 04:39:51AM -0800, bilal ghayyad wrote: I am interested in the inner working and I would like to go deep, so how can I start with letting asterisk work based on the database and work realtime? An example PBX that is based on Asterisk and uses Real-Time (from a PostgreSQL database) is Druid: http://voiceroute.org/ -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Dabatase
On Sunday 21 December 2008 04:14:58 Grey Man wrote: On Sun, Dec 21, 2008 at 9:52 AM, bilal ghayyad bilmar...@yahoo.com wrote: Anyone knows if there is an Asterisk version that setup can be stored in Database instead of the configuration files (.conf)? Yes all current versions can. Take a look at Asterisk Realtime http://www.voip-info.org/wiki-Asterisk+RealTime This is true, but there are multiple files that cannot be stored in a database. These include asterisk.conf, extconfig.conf, and modules.conf. Additionally, storing the extensions in a database is among the worst interfaces currently for Asterisk. It's basically the same as static configuration, with a performance penalty and less versatility. I don't know if I could come up with a worse interface if I was trying to drive someone insane. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Dabatase
On Sun, 21 Dec 2008, Steve Howes wrote: On 21 Dec 2008, at 12:39, bilal ghayyad wrote: I am interested in the inner working and I would like to go deep, so how can I start with letting asterisk work based on the database and work realtime? http://tinyurl.com/6t9x9b Cute. Much more polite than the standard RTFM that came to mind. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Dabatase
On Sunday 21 December 2008 09:03:29 Mike wrote: hints don't quite work too unless they are in a conf file. That last issue seems to be fixed in the latest 1.6, but I haven't tried it)... No, due to a technical limitation of how device states work, they will never work out of a database. What is new in 1.6 is the ability to define a pattern match for a hint (it expands automatically, based on subscriptions to the various hints). -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Dabatase
On Sunday 21 December 2008 09:03:29 Mike wrote: I've explored two ways of working with the database, as far as dialplan goes. The first was using realtime, as described in the wiki, to store the dialplan. This became very messy very quickly, as an SQL database isn't a great support for the dialplan. It's (relatively) hard to modify, and if your goal is to build your own Web portal (which was my goal) you're stuck making the portal code very complexe to accommodate the messy dialplan. What I went with was the complete opposite. I build the dialplan in the conf files, but everytime I refered to something dynamic, I called an SQL function (see MYSQL cmd for Asterisk). This made the dialplan more complexe (lot's of call to MYSQL) but it's more easily changed (since it's still text) and the Web portal is much readable since everything relies a well-designed relational DB instead of the dialplan realtime database. This is exactly what I recommend to people, and I'm happy to see that others are heading in this direction, even if it's not with the tools that I've developed for this purpose. But there are donwsides to that too: MYSQL down means my VoIP server is also down (which never actually happened, but it could). And, as I said, the .conf file is a lot more complexe I've tried to address several of the complaints about func_odbc in the 1.6.x version. Specifically, func_odbc supports multirow queries, separation of read and write operations, and automatic failover between database handles. One of the other advantages that func_odbc has over the MYSQL command is that resources are automatically deallocated when a channel hangs up. When the MYSQL command was originally written, it did not have the advantage of some of the channel infrastructure that we now have, and therefore it could not automatically destroy resources when they were no longer relevant. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Dabatase
Thanks, I appreciate getting the update on 1.6 with regards to the DB functions (and, for that matter, on the hints too). Wouldn't using MYSQL Disconnect handle the ressources as well (if, of course, called when needed)? About 1.6, would you know if the SIPPEER func now support sip realtime? It's a shame Asterisk is (was?) developped with mismatches between .conf functionality and realtime ones. Regards, Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Sunday, December 21, 2008 12:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk and Dabatase On Sunday 21 December 2008 09:03:29 Mike wrote: I've explored two ways of working with the database, as far as dialplan goes. The first was using realtime, as described in the wiki, to store the dialplan. This became very messy very quickly, as an SQL database isn't a great support for the dialplan. It's (relatively) hard to modify, and if your goal is to build your own Web portal (which was my goal) you're stuck making the portal code very complexe to accommodate the messy dialplan. What I went with was the complete opposite. I build the dialplan in the conf files, but everytime I refered to something dynamic, I called an SQL function (see MYSQL cmd for Asterisk). This made the dialplan more complexe (lot's of call to MYSQL) but it's more easily changed (since it's still text) and the Web portal is much readable since everything relies a well-designed relational DB instead of the dialplan realtime database. This is exactly what I recommend to people, and I'm happy to see that others are heading in this direction, even if it's not with the tools that I've developed for this purpose. But there are donwsides to that too: MYSQL down means my VoIP server is also down (which never actually happened, but it could). And, as I said, the .conf file is a lot more complexe I've tried to address several of the complaints about func_odbc in the 1.6.x version. Specifically, func_odbc supports multirow queries, separation of read and write operations, and automatic failover between database handles. One of the other advantages that func_odbc has over the MYSQL command is that resources are automatically deallocated when a channel hangs up. When the MYSQL command was originally written, it did not have the advantage of some of the channel infrastructure that we now have, and therefore it could not automatically destroy resources when they were no longer relevant. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Dabatase
On Sun, Dec 21, 2008 at 12:48 PM, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: On Sunday 21 December 2008 09:03:29 Mike wrote: I've explored two ways of working with the database, as far as dialplan goes. The first was using realtime, as described in the wiki, to store the dialplan. This became very messy very quickly, as an SQL database isn't a great support for the dialplan. It's (relatively) hard to modify, and if your goal is to build your own Web portal (which was my goal) you're stuck making the portal code very complexe to accommodate the messy dialplan. What I went with was the complete opposite. I build the dialplan in the conf files, but everytime I refered to something dynamic, I called an SQL function (see MYSQL cmd for Asterisk). This made the dialplan more complexe (lot's of call to MYSQL) but it's more easily changed (since it's still text) and the Web portal is much readable since everything relies a well-designed relational DB instead of the dialplan realtime database. This is exactly what I recommend to people, and I'm happy to see that others are heading in this direction, even if it's not with the tools that I've developed for this purpose. But there are donwsides to that too: MYSQL down means my VoIP server is also down (which never actually happened, but it could). And, as I said, the .conf file is a lot more complexe I've tried to address several of the complaints about func_odbc in the 1.6.x version. Specifically, func_odbc supports multirow queries, separation of read and write operations, and automatic failover between database handles. One of the other advantages that func_odbc has over the MYSQL command is that resources are automatically deallocated when a channel hangs up. When the MYSQL command was originally written, it did not have the advantage of some of the channel infrastructure that we now have, and therefore it could not automatically destroy resources when they were no longer relevant. -- Tilghman MySQL admin which I usually use Webmin to access is good. I also had a very bright, but too bright customer decide to base his call center on 1.2 using ODBC connected to an access DB. I worked great, too great, when he changed from one cell to another, the change was immediate. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A method to determine PSTN Call Provider?
I'm looking for a solution to determine if a PSTN call to a zaptel channel was originated from a VoIP provider or not in real time. I'd like to use the callerid(num) to reverse match to the provider. Does anyone have a clue how I could do this? TIA Bart___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Dabatase
On Sunday 21 December 2008 12:08:17 Mike wrote: Thanks, I appreciate getting the update on 1.6 with regards to the DB functions (and, for that matter, on the hints too). Wouldn't using MYSQL Disconnect handle the ressources as well (if, of course, called when needed)? The Disconnect command only frees the resources associated with the connection. The Clear command is needed to destroy resources associated with any outstanding query. Of course, if you get a hangup prior to completing either of these cleanup commands (and you aren't explicitly handling those in the h extension), you'll get a resource leak. Func_odbc does not need any explicit command to cleanup its resources; when the channel is destroyed, the resource-cleanup routines are automatically called within the destruction sequence. About 1.6, would you know if the SIPPEER func now support sip realtime? It's a shame Asterisk is (was?) developped with mismatches between .conf functionality and realtime ones. 1.6 works identically to 1.4, in this regard, although some improvements have been made to the rtcachefriends setting, which now allows expired cached realtime peers to exit memory gracefully. This was broken at some point (which may be the issue in your installed version), but the current release of both 1.4 and 1.6.0 should now both support using SIPPEER, as long as the peer is cached in memory. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A method to determine PSTN Call Provider?
With number portability? I would think that would require a very large and dynamic database that would get obsolete rather quickly. I guess somewhere in the deep innards of the network that database has to exist, but are mere mortals allowed to access it? Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A method to determine PSTN Call Provider?
Bart: Please contact me off-list. I seem to get a security rejection from your mail server when I tried to e-mail you. -- Alex Barton Fisher wrote: I'm looking for a solution to determine if a PSTN call to a zaptel channel was originated from a VoIP provider or not in real time. I'd like to use the callerid(num) to reverse match to the provider. Does anyone have a clue how I could do this? TIA Bart ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Dabatase
Tilghman Lesher schrieb: On Sunday 21 December 2008 12:08:17 Mike wrote: Thanks, I appreciate getting the update on 1.6 with regards to the DB functions (and, for that matter, on the hints too). Wouldn't using MYSQL Disconnect handle the ressources as well (if, of course, called when needed)? The Disconnect command only frees the resources associated with the connection. The Clear command is needed to destroy resources associated with any outstanding query. Of course, if you get a hangup prior to completing either of these cleanup commands (and you aren't explicitly handling those in the h extension), you'll get a resource leak. That's why the MySQL functions should either be fixed or removed. They make it too easy to shoot yourself in the foot. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good comparisons on cheaper VOIP phones
I don't know if Thomson ST2030 SIP phones are distributed where you live but those have the best feature set-price ratio. They integrate smoothly with Asterisk (one touch pickup, BLF, MWI, ...) with up to 5 simultaneous calls. Here in France, those are selected everywhere ... I would recommend them without any hesitation. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A method to determine PSTN Call Provider?
On Sun, 21 Dec 2008, Barton Fisher wrote: I'm looking for a solution to determine if a PSTN call to a zaptel channel was originated from a VoIP provider or not in real time. I'd like to use the callerid(num) to reverse match to the provider. Does anyone have a clue how I could do this? What country are you in/want to tell? In the UK it's realtively easy, but it's a big database, and it won't tell you anything about ported numbers eg. from a PSTN to VoIP operator) That's assuming the caller doesn't withold... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A method to determine PSTN Call Provider?
- Original Message - From: Gordon Henderson gordon+aster...@drogon.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, December 21, 2008 11:30 AM Subject: Re: [asterisk-users] A method to determine PSTN Call Provider? On Sun, 21 Dec 2008, Barton Fisher wrote: I'm looking for a solution to determine if a PSTN call to a zaptel channel was originated from a VoIP provider or not in real time. I'd like to use the callerid(num) to reverse match to the provider. Does anyone have a clue how I could do this? What country are you in/want to tell? In the UK it's realtively easy, but it's a big database, and it won't tell you anything about ported numbers eg. from a PSTN to VoIP operator) That's assuming the caller doesn't withold... Gordon Sorry, I'm in the US Bart ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Dabatase
On Sunday 21 December 2008 13:07:05 Philipp Kempgen wrote: Tilghman Lesher schrieb: On Sunday 21 December 2008 12:08:17 Mike wrote: Thanks, I appreciate getting the update on 1.6 with regards to the DB functions (and, for that matter, on the hints too). Wouldn't using MYSQL Disconnect handle the ressources as well (if, of course, called when needed)? The Disconnect command only frees the resources associated with the connection. The Clear command is needed to destroy resources associated with any outstanding query. Of course, if you get a hangup prior to completing either of these cleanup commands (and you aren't explicitly handling those in the h extension), you'll get a resource leak. That's why the MySQL functions should either be fixed or removed. They make it too easy to shoot yourself in the foot. They do, but they also allow something that I currently do not permit in func_odbc: retrieval and destruction of results and handles by a channel other than the one which created them. So you could plausibly use the MYSQL command for something like a serialization operation, whereby each channel fetches a row in turn and uses the values therein for a coordination procedure of some kind. I'm not quite sure what the value of that is, but it is possible with the current model. Like every other powerful utility, you may choose to use the rope to make a bridge or a noose. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Dabatase
Tilghman Lesher schrieb: On Sunday 21 December 2008 13:07:05 Philipp Kempgen wrote: Tilghman Lesher schrieb: The Disconnect command only frees the resources associated with the connection. The Clear command is needed to destroy resources associated with any outstanding query. Of course, if you get a hangup prior to completing either of these cleanup commands (and you aren't explicitly handling those in the h extension), you'll get a resource leak. That's why the MySQL functions should either be fixed or removed. They make it too easy to shoot yourself in the foot. They do, but they also allow something that I currently do not permit in func_odbc: retrieval and destruction of results and handles by a channel other than the one which created them. So you could plausibly use the MYSQL command for something like a serialization operation, whereby each channel fetches a row in turn and uses the values therein for a coordination procedure of some kind. I'm not quite sure what the value of that is, but it is possible with the current model. Like every other powerful utility, you may choose to use the rope to make a bridge or a noose. So if that's a good thing because it is powerful that raises the question why you don't allow it in func_odbc then. I guess nobody needs it. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 module hung and unable to unload chan_iax2.so
Hello everybody, This is a problem disturb me for long. I run asterisk Asterisk 1.4.14 and A2Billing 1.3 in the same Debian 4.0 ETCH server. And there is also FoneBridge for TDM over Ethernet with E1 to make as well as receive calls from mobiles or PSTN. And IAX2 trunk runs between Asterisk and A2Billing. For most of the time they really do a good job. But some hours or days laterI mean it happens indefinitely, they both strike. When it happens, no outgoing calls or incoming calls from mobiles or PSTN can get through, but the SIP extensions can call each other without issues. I am sure the IAX2 trunk and IAX2 modules hung because the SIP calls work but IAX2 related calls does not work. So I issue 'unload chan_iax2.so' doesn't work. It will give 'unable to unload chan_iax2.so' at CLI and 'WARNING[20750] loader.c: Soft unload failed, 'chan_iax2.so' has use count 1' at the Asterisk log file /var/log/asterisk/full. I tried that 'amportal restart' to restart it, but the asterisk and safe_asterisk processes just keep alive. Even kill and pkill also does not work. The only works way I've ever tried is 'kill -9 PID' and run it again. Thank GOD! It will do a good job again. But hours or days later it happens again. Daily restart Asterisk periodically tried too but it happened just hours ago. Please any idea or help will be highly appreciated. Regards Jason Hu Regards Jason Hu ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Dabatase
On Sunday 21 December 2008 15:14:18 Philipp Kempgen wrote: Tilghman Lesher schrieb: On Sunday 21 December 2008 13:07:05 Philipp Kempgen wrote: Tilghman Lesher schrieb: The Disconnect command only frees the resources associated with the connection. The Clear command is needed to destroy resources associated with any outstanding query. Of course, if you get a hangup prior to completing either of these cleanup commands (and you aren't explicitly handling those in the h extension), you'll get a resource leak. That's why the MySQL functions should either be fixed or removed. They make it too easy to shoot yourself in the foot. They do, but they also allow something that I currently do not permit in func_odbc: retrieval and destruction of results and handles by a channel other than the one which created them. So you could plausibly use the MYSQL command for something like a serialization operation, whereby each channel fetches a row in turn and uses the values therein for a coordination procedure of some kind. I'm not quite sure what the value of that is, but it is possible with the current model. Like every other powerful utility, you may choose to use the rope to make a bridge or a noose. So if that's a good thing because it is powerful that raises the question why you don't allow it in func_odbc then. Because while func_odbc is admittedly an overengineered solution to a problem I once had to solve, I did not have a need for such functionality. If someone had a use case for why that would be a useful feature within func_odbc, I would probably find a way to do it. I guess nobody needs it. I wouldn't go that far. In fact, there's probably very few people who share it amongst multiple channels. However, if I make the assumption that people will have no need for a particular feature, I usually find out that I'm wrong about six months too late. And then I get to spend time figuring out how to meld two different feature sets together, despite having removed one to make the other more easily written. So my reticence on removal is based upon past experience. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Supermicro and onboard Intel e1000 Ethernet controllers ... no longer an issue?
From googling around, I see people refer to http://www.digium.com/en/docs/misc/compatibility_notes.php which used to say: Some server motherboards utilize an onboard Intel e1000 Ethernet controller that can interfere with the operation of Digium's cards. The recommended action for this server is to disable the onboard Ethernet controller and use a PCI-based solution. Also, the MS-7032 (K8T Neo-V/K8M Neo-V) motherboard is incompatible with the TE4XXP using the firmware ending in 164. The problem is that the card will randomly receive interrupts. ...but currently has been removed. Does this imply the issue is resolved? Of course, I can always try it out, I just want to verify if there is a definitive answer. Or I can email Digium - but they haven't been too responsive to a couple past emails I've sent about other issues. Any thoughts? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
