[Asterisk-Users] Xten Lite Build 1079
I've just down loaded Xten Lite and it is now build 1079. It now finds the NAT firewall type and has loads more to configure. But it doesn't work on my poor W95 tablet PC. -- Dave Cotton Directeur Linux Autrement 193 rue Marcel Cerdan 84270 Vedene 04 90 23 30 81 Internet Sheriff Technology revendeur en France http://www.linuxautrement.com IAX 17004902330 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Web Admin - was:Re: [Asterisk-Users] CDR Web Search Frontend
- Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, October 01, 2003 12:09 PM Subject: Re: Web Admin - was:Re: [Asterisk-Users] CDR Web Search Frontend I think there is room for everyones ideas, the more the better.. The biggest problem I see with these things is that many people seem to end up developing in parallel streams and the result is 5 seperate projects all half baked and incomplete.. What is needed is for everyone to pool their efforts and come up with a definitave web application to run on top of Asterisk.. That's why i sent out this mail as the last thing i want to do is start on something where 5 others are starting on on their own too. We just need someone to take on the project and if someone is ready to do so then fine if not i'll be happy to keep track of features etc and people that are willing to put in their time and effore on this. But again i don't want to step on anyone's feet in case they are allready doing this. Maybe a php-dev mailinglist might be a good help here too ? Anyway I am rambling.. So I will stop now.. No you wheren't ;) Greetings, Tj ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR Web Search Frontend
- Original Message - From: Jamie Carl [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, October 02, 2003 1:58 AM Subject: Re: [Asterisk-Users] CDR Web Search Frontend As for the rest of this discussion, I have already started work on this Asterisk Web Interface. (visit http://astweb.sourceforge.net). The current release is still only the CDR section, but things are starting to evolve and I expect to have something usable in the next few weeks. It is being written in PHP and will attempt to use ZERO OS-DEPENDANT code. Is this a Jazz-inc copyright project or are you willing to just open it all up and make it an astweb team effort ? Like i and others have said in earlier posts would be good to join efforts, setup a roadmap and get a couple of php programmers working on something in joint effort. Greetings, Tj ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] error message 49159
Hi All I have that error message: WARNING[49159]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) What can be the problem? Thanks! miklos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with ISA PhoneJack.
Title: Message The device is seen in linux pnp: isapnp: Scanning for PnP cards...isapnp: Card 'Quicknet Internet PhoneJACK'isapnp: 1 Plug Play card detected total and I've installed the drivers from the openh323 dev... but I can't get * to see it. Does anyone have experience with this? THanks. -Dave.
Re: [Asterisk-Users] IAX and IAXTEL
The location of the guest / iaxtel section having to be at the end is, as it turns out, a configuration error on iaxtel. I hope to have it straightened out shortly. Mark On Thu, 2 Oct 2003, Bartosz Jozwiak wrote: Sometime yes sometimes no :) But thats the life :) Ok but I fixed it. Just put the guest section in iax.conf all the way on the end. And right now it works for me. :) -- Bart - Original Message - From: bill black [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, October 02, 2003 12:47 AM Subject: Re: [Asterisk-Users] IAX and IAXTEL Hello Bart: Did anyone ever follow up to your question? I have the same issue. thanks, Bill On Wednesday 01 October 2003 07:27, Bartosz Jozwiak wrote: Hello, Could somebody tell me what I should change in iax.conf file to be able to receive calls from iaxtel. I am already registered and I can make calls to IAXtel users but what I should do in iax.conf to be able to receive call also. -- Bart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR Web Search Frontend
Tjardick van der Kraan wrote: Is this a Jazz-inc copyright project or are you willing to just open it all up and make it an astweb team effort ? If you look at the sourceforge page its GPL.. http://sourceforge.net/projects/astweb/ Like i and others have said in earlier posts would be good to join efforts, setup a roadmap and get a couple of php programmers working on something in joint effort. Exactly right.. The mailing lists on SF for the project are probably the best place to debate and rationalise features for this project.. Although they are brand new so there isn't much activity right now.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk friendly IAX/SIP wholesalers in Australia
Title: Message its a fair question: does anyone know any? Bryan Nolen Lead Developer http://Arc.Net.AU http://cdonline.com.au
Re: [Asterisk-Users] Asterisk friendly IAX/SIP wholesalers in Australia
its a fair question: does anyone know any? I'm afraid this doesnt answer your question and is a bit of a shameless plug, but we have just started offering IAX (and SIP) termination in the UK, so if this helps anyone out, please feel free to contact me. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Web Admin - was:Re: [Asterisk-Users] CDR Web Search Frontend
Again, we need to seriously consider moving this to a separate mailing list and getting a 'Features' thread started, as well as a 'Mission' thread. These should get everyone's feet on the same path. I agree that the web administration application needs to be be something different than simply displaying the configuration file. By the time we're done, I think it would be ideal to have abstracted the entire * configuration and store it in some sort of organized fashion (flat-file, RDBMS, XML, whatever). -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tjardick van der Kraan Sent: Thursday, October 02, 2003 7:33 AM To: [EMAIL PROTECTED] Subject: Re: Web Admin - was:Re: [Asterisk-Users] CDR Web Search Frontend - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, October 01, 2003 12:09 PM Subject: Re: Web Admin - was:Re: [Asterisk-Users] CDR Web Search Frontend I think there is room for everyones ideas, the more the better.. The biggest problem I see with these things is that many people seem to end up developing in parallel streams and the result is 5 seperate projects all half baked and incomplete.. What is needed is for everyone to pool their efforts and come up with a definitave web application to run on top of Asterisk.. That's why i sent out this mail as the last thing i want to do is start on something where 5 others are starting on on their own too. We just need someone to take on the project and if someone is ready to do so then fine if not i'll be happy to keep track of features etc and people that are willing to put in their time and effore on this. But again i don't want to step on anyone's feet in case they are allready doing this. Maybe a php-dev mailinglist might be a good help here too ? Anyway I am rambling.. So I will stop now.. No you wheren't ;) Greetings, Tj ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] eBay Sip Phone Scam.
Ok, see, now you're confusing what I said. Nowhere did I say I had the 102D. I said he never mentioned that it was the 102, irregardless of the D. I *DO* have the 101, which is what he was talking about. No, it doesn't mention it's the 101. This argument has now proved silly, especially since you're confusing what I'm saying, with what he supposedly is. *I CLAIM END OF THREAD!* -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of costas Sent: Thursday, October 02, 2003 7:04 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] eBay Sip Phone Scam. Ok, in addition you are confusing the 102 with the 102D. If you had done your homework you would have noticed that the 102D (see the big D?) is a different model. Than one has the 16x2 LCD and 3 way conferencing. I spent a lot of time studying these phones. So no, you don't have that phone. check http://www.chagres.net -- Original Message -- From: Josh Roberson [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Thu, 2 Oct 2003 07:31:43 -0500 My bad... It's a .net, not a .com :P Oops... Sorry JMB (sheepish grin) http://www.chagres.net -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Low, Adam Sent: Thursday, October 02, 2003 6:22 AM To: '[EMAIL PROTECTED] ' Subject: RE: [Asterisk-Users] eBay Sip Phone Scam. Josh, Pls can you confirm that URL, www.chagres.com doesn't seem to mention the sale of any Grandstream phones Adam -Original Message- From: Josh Roberson To: [EMAIL PROTECTED] Sent: 02/10/03 13:04 Subject: RE: [Asterisk-Users] eBay Sip Phone Scam. Actually, had you taken the time to READ the auction details, He says (direct copy/paste from auction) -Begin Copy/Paste- Flash Based OS Easy to install and manage, Cost effective, Easy to use - Friendly GUI for 1st time user, Easy to learn - User's guide and on-line tutorial Big information and management LCD blue back light User friendly keypad Universal AC/DC adapter Ergonomic design 25-button keypad 12-digit caller ID LCD Universal Switching Power Adaptor Input: 100-240VAC Output: +5VDC, 400mA, 1. Auto-sensing 10/100 Base-TX Ethernet Port 2.UL/CE/FCC 3.Power Supply : Universal 90 ~ 264V Support all major Network Operating Systems (Windows, MAC, Linux/Unix) Web-Based Management TCP/IP Configuration with DHCP support Free Flash Firmware update No User Licenses System Restart/ Shutdown Password Access control 1 x 10/100Mbps Ethernet Port (RJ-45 Interface) Support STUN and SIMPLE extension Interoperable with 3d parties Proxy, Registrar and gateway products DSP technology for the best voice quality Support popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (alaw and u-law), G.726 (40K/32K/24K/16K), as well as G.728 (Model 102D) In and out-off-band DTMF Support 3-way conferencing (Model 102D), full duplex hands-free speakerphone, redial, call log, volume control, voice mail with indicator Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS) Remote software upgrade capability via TFTP Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control) -End Copy/Paste- Nowhere does he make the claims you're stating. He DOES, have (Model 102D) in one of the descriptions, but that is a direct quote from Grandstream's product brochure. Also, this phone *IS* out on the market.. I own one, and I'm quite happy with it.. I will tell you this though: Go order one from Chagres (http://www.chagres.com). They are an asterisk supporter/user on this list, and the price is MUCH better. ;) /rant -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of costas Sent: Thursday, October 02, 2003 4:50 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] eBay Sip Phone Scam. I did shop around. Nowhere does he say the phone is the 101. If you look at his ad he says the phone has 102D features and has 16x2 lines and 3 way conference. The starting price was $90. A reasonable opening price I thought. He also does not say the phone is not available until end of year. I only called Grandstream to find out some info on it after I placed the Bid. In a way Grandstream is also at fault. Nowhere do they say the phone is not available. I was suprised when they told me it wasnt even out. When I sent a message to this thief, he said its for the 101 and they are hard to get. Thats why he jacked up the price. He did cancel my bid after telling me what a bad
[Asterisk-Users] Call it Asterisk-Addons and let us go have some fun?
No one else would step up to the plate so I did. Here is your list. HAVE SOME FUN! need any more just ask. Troy Settle wrote: With all the discussion about licensing issues and the sort, I think it's time for a full blown 3rd party application to work with Asterisk while at the same time not causing Asterisk to become encumbered. For such a project, I'm license neutral. While I prefer the BSD license, the GPL would work just as well for such a project. I'd say the first order of business, is to move this discussion to a separate list so as not to annoy the purists. Perhaps Digium would be willing to host it? Call it Asterisk-Addons and let us go have some fun?If this is going to become a full blown third party app, I am willing to organize any documentation that needs to be done for it. I have been attempting to get a better handle of some of the more undocumented features of asterisk lately in an attempt to start documenting that as well, but since this "addon" is in the beginning stages, it might be nice to have documentation thought about right from the beginning (as opposed to an after thought)Just my 0.02 cents CDN.Leif Madsen. This message was checked by MailScan for WorkgroupMail. www.workgroupmail.com
RE: [Asterisk-Users] IAX and IAXTEL
Well, that's odd.. Can you, then, with IAX, determine in which section (first, second, last, etc...) you read your configuration in iax.conf, rather than matching up with passwords? -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Spencer Sent: Thursday, October 02, 2003 7:16 AM To: ASTERISK USERS Subject: Re: [Asterisk-Users] IAX and IAXTEL The location of the guest / iaxtel section having to be at the end is, as it turns out, a configuration error on iaxtel. I hope to have it straightened out shortly. Mark On Thu, 2 Oct 2003, Bartosz Jozwiak wrote: Sometime yes sometimes no :) But thats the life :) Ok but I fixed it. Just put the guest section in iax.conf all the way on the end. And right now it works for me. :) -- Bart - Original Message - From: bill black [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, October 02, 2003 12:47 AM Subject: Re: [Asterisk-Users] IAX and IAXTEL Hello Bart: Did anyone ever follow up to your question? I have the same issue. thanks, Bill On Wednesday 01 October 2003 07:27, Bartosz Jozwiak wrote: Hello, Could somebody tell me what I should change in iax.conf file to be able to receive calls from iaxtel. I am already registered and I can make calls to IAXtel users but what I should do in iax.conf to be able to receive call also. -- Bart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.516 / Virus Database: 313 - Release Date: 9/1/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.516 / Virus Database: 313 - Release Date: 9/1/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fw: Call it Asterisk-Addons and let us go have some fun?
the link is at www.pawbell.com - Original Message - From: sip To: [EMAIL PROTECTED] Sent: Thursday, October 02, 2003 8:57 AM Subject: Call it Asterisk-Addons and let us go have some fun? No one else would step up to the plate so I did. Here is your list. HAVE SOME FUN! need any more just ask. Troy Settle wrote: With all the discussion about licensing issues and the sort, I think it's time for a full blown 3rd party application to work with Asterisk while at the same time not causing Asterisk to become encumbered. For such a project, I'm license neutral. While I prefer the BSD license, the GPL would work just as well for such a project. I'd say the first order of business, is to move this discussion to a separate list so as not to annoy the purists. Perhaps Digium would be willing to host it? Call it Asterisk-Addons and let us go have some fun?If this is going to become a full blown third party app, I am willing to organize any documentation that needs to be done for it. I have been attempting to get a better handle of some of the more undocumented features of asterisk lately in an attempt to start documenting that as well, but since this "addon" is in the beginning stages, it might be nice to have documentation thought about right from the beginning (as opposed to an after thought)Just my 0.02 cents CDN.Leif Madsen. This message was checked by MailScan for WorkgroupMail. www.workgroupmail.com
Re: Web Admin - was:Re: [Asterisk-Users] CDR Web Search Frontend
Troy Settle wrote: Again, we need to seriously consider moving this to a separate mailing list and getting a 'Features' thread started, as well as a 'Mission' thread. These should get everyone's feet on the same path. I agree that the web administration application needs to be be something different than simply displaying the configuration file. By the time we're done, I think it would be ideal to have abstracted the entire * configuration and store it in some sort of organized fashion (flat-file, RDBMS, XML, whatever). -- You can get onto the mailing list here http://sourceforge.net/mail/?group_id=91371 Probably a better place to take the discussion further.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GSM player for Windows
Does anyone know if there is a GSM player for windows? Dante
[Asterisk-Users] WINXP Messenger SIP Client (Good News, Bad News)
I found this information on how to make XP have a dialpad in Windows Messenger which was awesome news HKEY_CURRENT_USER\Software\Microsoft\MessengerService\CorpPC2PHone (change it from 0 to 1 and a magic new choice to make phone calls appears) only to be crushed hours later when I realized It doesnt seem to do dtmf right. If i make an ext lead to AgentLogin for instance and press my extension when I hit # i get like 4 overlapping incorrect errs because it must be sending like 4 # digits instead of 1. I dont see a place to change the dtmf in xp and only rfc mode works any other method is ignored.. Did anyone ever get this to work right? estara works right in the same dtmf mode so i'm inclined to blame windows. It at least works as a phone gateway cos you can dial the inital ext properly when placing the call. Do you Yahoo!? The New Yahoo! Shopping - with improved product search
Re: [Asterisk-Users] WINXP Messenger SIP Client (Good News, Bad News)
Anthony Minessale wrote: I found this information on how to make XP have a dialpad in Windows Messenger which was awesome news HKEY_CURRENT_USER\Software\Microsoft\MessengerService\CorpPC2PHone (change it from 0 to 1 and a magic new choice to make phone calls appears) only to be crushed hours later when I realized It doesnt seem to do dtmf right. If i make an ext lead to AgentLogin for instance and press my extension when I hit # i get like 4 overlapping incorrect errs because it must be sending like 4 # digits instead of 1. I dont see a place to change the dtmf in xp and only rfc mode works any other method is ignored.. Did anyone ever get this to work right? estara works right in the same dtmf mode so i'm inclined to blame windows. It at least works as a phone gateway cos you can dial the inital ext properly when placing the call. Do you Yahoo!? The New Yahoo! Shopping http://shopping.yahoo.com/?__yltc=s%3A15443%2Cd%3A22708228%2Cslk%3Atext%2Csec%3Amail - with improved product search Some more crushing news is if you upgrade MSN messenger past ver 4.x it no longer uses SIP.. (so I have been told).. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Has anyone got * working with Xten soft phones
I use mine all the time. Things to check or set: Under System SettingsNetwork 1- Set the IP of you * box in Outbound SIP Proxy Under System SettingsSip Proxy 1- Enable yes 2- Username (the name or number in your SIP.CONF [brackets] 3- Leave Authorized User blank (and remark out in SIP.CONF if you have it in there.) 4- Obviously set the password 5- Domain/Realm: the IP of your * box. 6- Sip Proxy: the IP of your * box. 7- Send Internal IP: ON Under System SettingsAudio Device 1- Make sure it's your mic and not soemthing else. Under Advanced System SettingsFeature Settings DTMF Force Send In Band: Yes (is you use inband in your sip.conf) That should do it. Mine works fine. Regards, Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fats Neutron Sent: Thursday, October 02, 2003 9:18 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Has anyone got * working with Xten soft phones I have tried loads of configurations but I cannot get it to work. I basically have three computers and want to use soft phones on two of them to connect to asterisk so I can use them instead of dialling into my X100P card as the phone bill is getting bigger. I assume that I can configure * to accept calls from the soft phones and then route them according to my dial plan. I am basically trying to write some extensions to asterisk and need to debug the code but need to phone in to do it. I could route in from the net if it wasn't for the fact that I am behind a NAT and I've decided to give up on getting it to work as I have just got stuck. So this is another option to attempt to get debugging. So to paraphrase: Has anyone got a setup working with xten soft phones behind a NAT and if so could you share your setup with me. Any help would be appreciated. Thanks in advance. Fats. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR Web Search Frontend
On Thu, Oct 02, 2003 at 09:58:37AM +1000, Jamie Carl wrote: ... As for the rest of this discussion, I have already started work on this Asterisk Web Interface. (visit http://astweb.sourceforge.net). The current release is still only the CDR section, but things are starting to evolve and I expect to have something usable in the next few weeks. It is being written in PHP and will attempt to use ZERO OS-DEPENDANT code. ... Sorry, preview selection generates: Your DNS2Go account has been disabled. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WINXP Messenger SIP Client (Good News, Bad News)
So where do or can you get older version? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut Sent: Thursday, October 02, 2003 10:50 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] WINXP Messenger SIP Client (Good News, Bad News) Anthony Minessale wrote: I found this information on how to make XP have a dialpad in Windows Messenger which was awesome news HKEY_CURRENT_USER\Software\Microsoft\MessengerService\CorpPC2PHone (change it from 0 to 1 and a magic new choice to make phone calls appears) only to be crushed hours later when I realized It doesnt seem to do dtmf right. If i make an ext lead to AgentLogin for instance and press my extension when I hit # i get like 4 overlapping incorrect errs because it must be sending like 4 # digits instead of 1. I dont see a place to change the dtmf in xp and only rfc mode works any other method is ignored.. Did anyone ever get this to work right? estara works right in the same dtmf mode so i'm inclined to blame windows. It at least works as a phone gateway cos you can dial the inital ext properly when placing the call. -- -- Do you Yahoo!? The New Yahoo! Shopping http://shopping.yahoo.com/?__yltc=s%3A15443%2Cd%3A22708228%2Cslk%3Atext %2Csec%3Amail - with improved product search Some more crushing news is if you upgrade MSN messenger past ver 4.x it no longer uses SIP.. (so I have been told).. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] eBay Sip Phone Scam.
Hi Josh, Costas, Adam, et all We do sell the phones. http://www.chagres.net/products/voip/phones.html and digium cards http://www.chagres.net/products/voip/cards.html plus new things real soon :) and if anyone ever has a problem, go yell at me and I'll try like crazy to fix it. john brown, ceo chagres technologies, inc ps: Chagres.com (Chagres River) is where our company got its name from. I used to live there ;) On Thu, Oct 02, 2003 at 02:21:55PM +0200, Low, Adam wrote: Josh, Pls can you confirm that URL, www.chagres.com doesn't seem to mention the sale of any Grandstream phones Adam -Original Message- From: Josh Roberson To: [EMAIL PROTECTED] Sent: 02/10/03 13:04 Subject: RE: [Asterisk-Users] eBay Sip Phone Scam. Actually, had you taken the time to READ the auction details, He says (direct copy/paste from auction) -Begin Copy/Paste- Flash Based OS Easy to install and manage, Cost effective, Easy to use - Friendly GUI for 1st time user, Easy to learn - User's guide and on-line tutorial Big information and management LCD blue back light User friendly keypad Universal AC/DC adapter Ergonomic design 25-button keypad 12-digit caller ID LCD Universal Switching Power Adaptor Input: 100-240VAC Output: +5VDC, 400mA, 1. Auto-sensing 10/100 Base-TX Ethernet Port 2.UL/CE/FCC 3.Power Supply : Universal 90 ~ 264V Support all major Network Operating Systems (Windows, MAC, Linux/Unix) Web-Based Management TCP/IP Configuration with DHCP support Free Flash Firmware update No User Licenses System Restart/ Shutdown Password Access control 1 x 10/100Mbps Ethernet Port (RJ-45 Interface) Support STUN and SIMPLE extension Interoperable with 3d parties Proxy, Registrar and gateway products DSP technology for the best voice quality Support popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (alaw and u-law), G.726 (40K/32K/24K/16K), as well as G.728 (Model 102D) In and out-off-band DTMF Support 3-way conferencing (Model 102D), full duplex hands-free speakerphone, redial, call log, volume control, voice mail with indicator Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS) Remote software upgrade capability via TFTP Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control) -End Copy/Paste- Nowhere does he make the claims you're stating. He DOES, have (Model 102D) in one of the descriptions, but that is a direct quote from Grandstream's product brochure. Also, this phone *IS* out on the market.. I own one, and I'm quite happy with it.. I will tell you this though: Go order one from Chagres (http://www.chagres.com). They are an asterisk supporter/user on this list, and the price is MUCH better. ;) /rant -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of costas Sent: Thursday, October 02, 2003 4:50 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] eBay Sip Phone Scam. I did shop around. Nowhere does he say the phone is the 101. If you look at his ad he says the phone has 102D features and has 16x2 lines and 3 way conference. The starting price was $90. A reasonable opening price I thought. He also does not say the phone is not available until end of year. I only called Grandstream to find out some info on it after I placed the Bid. In a way Grandstream is also at fault. Nowhere do they say the phone is not available. I was suprised when they told me it wasnt even out. When I sent a message to this thief, he said its for the 101 and they are hard to get. Thats why he jacked up the price. He did cancel my bid after telling me what a bad person I am for wasting his time. -- Original Message -- From: Andrew Kohlsmith [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Wed, 1 Oct 2003 23:32:51 -0400 Some guy on eBay is trying to sell the Grandstream Budgetone Phone 101 as the 102D. And to make matters worse he starts the bid at $90.00 Beware. There's no need to beware -- anyone who doesn't shop around deserves to get suckered. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Costas Menico Meezon Software Corp 201-224-8111 [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version:
Re: [Asterisk-Users] Has anyone got * working with Xten soft phones
On 2/10/03 3:51 pm, Joseph Finley [EMAIL PROTECTED] wrote: I use mine all the time. Things to check or set: Under System SettingsNetwork 1- Set the IP of you * box in Outbound SIP Proxy Under System SettingsSip Proxy 1- Enable yes 2- Username (the name or number in your SIP.CONF [brackets] 3- Leave Authorized User blank (and remark out in SIP.CONF if you have it in there.) 4- Obviously set the password 5- Domain/Realm: the IP of your * box. 6- Sip Proxy: the IP of your * box. 7- Send Internal IP: ON Under System SettingsAudio Device 1- Make sure it's your mic and not soemthing else. Under Advanced System SettingsFeature Settings DTMF Force Send In Band: Yes (is you use inband in your sip.conf) That should do it. Mine works fine. Regards, Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fats Neutron Sent: Thursday, October 02, 2003 9:18 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Has anyone got * working with Xten soft phones I have tried loads of configurations but I cannot get it to work. I basically have three computers and want to use soft phones on two of them to connect to asterisk so I can use them instead of dialling into my X100P card as the phone bill is getting bigger. I assume that I can configure * to accept calls from the soft phones and then route them according to my dial plan. I am basically trying to write some extensions to asterisk and need to debug the code but need to phone in to do it. I could route in from the net if it wasn't for the fact that I am behind a NAT and I've decided to give up on getting it to work as I have just got stuck. So this is another option to attempt to get debugging. So to paraphrase: Has anyone got a setup working with xten soft phones behind a NAT and if so could you share your setup with me. Any help would be appreciated. Thanks in advance. Fats. I've tried those settings and so far so good. At least it seems to be talking to asterisk. What are your settings in sip.conf? Also how do you get the soft phone to dial an internal extension? Thanks for your help. Fingers crossed I figure this out. Fats. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with Dutch PSTN-line on X100P
Yo all, I have a problem with a Dutch (KPN) PSTN-line, connected to an X100P. The call wil sound OK at first, but after 10-20 minutes, the audio will start to crackle. Soon after that, this crackle turns into a continuous noise and the parties won't be able to hear eachother anymore. It also sometimes happens that the party on the TDM400P hears a very loud, short-delay echo of themselves, best described as talking in a bathroom. It very much sounds like very bad feedback, most of the time. The system concerned is a PIII-750 with an X100P and a TDM400P. The problem occurs when a call is bridged between the X100P and a port on the TDM400P. Calls from the TDM400P to a remote IAX2- connected box seem OK. rxgain / txgain are at their default (0) for all interfaces and I'm using the default echo-cancellation settings, with echocancelwhenbridged enabled. The PSTN-line concerned is a standard analog line from KPN, which also has ADSL coming in over it. The X100P is, ofcourse, connected behind the splitter. When I connect the X100P to an internal analog port of a small legacy home-switch, the problem doesn't seem to occur, either. I'm using opermode=1 for the X100P, but read on this list somewhere that the chip used on the X100P is actually not the international version, so this setting might not have any effect. In this case, it doesn't seem to matter if I use it or not. There's no notable difference. I'll probably investigate this problem further and play with the echo- canceller and rxgain / txgain a bit. Just curious if anyone else is experiencing something similar and might have found a solution already. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] eBay Sip Phone Scam.
Grandstream 102D won't be available until December, at the earliest. I have 101's in stock now and can ship same day as the order is funded. 102 (not the D model) are on backorder and I expect inventory by the end of next week. Transfering and other functions are really going to be a matter of what your SIP server does, feature wise. GAPS is not free. It starts at $3000 and goes up to 18,000 We have built our own system that we use internally to configure phones for our customers. On Thu, Oct 02, 2003 at 12:14:42PM -0400, costas wrote: Thanks! Do you have the 102D. Thats what started this. How fast can you ship any of the phones? Also I have question on the phones. 1) If you have 2 callers can you transfer one of them, or you can only transfer if you have the second line available? 2) Automated Provisioning System (GAPS). Free? -- Original Message -- From: John Brown (CV) [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Thu, 2 Oct 2003 09:06:00 -0600 Hi Josh, Costas, Adam, et all We do sell the phones. http://www.chagres.net/products/voip/phones.html and digium cards http://www.chagres.net/products/voip/cards.html plus new things real soon :) and if anyone ever has a problem, go yell at me and I'll try like crazy to fix it. john brown, ceo chagres technologies, inc ps: Chagres.com (Chagres River) is where our company got its name from. I used to live there ;) On Thu, Oct 02, 2003 at 02:21:55PM +0200, Low, Adam wrote: Josh, Pls can you confirm that URL, www.chagres.com doesn't seem to mention the sale of any Grandstream phones Adam -Original Message- From: Josh Roberson To: [EMAIL PROTECTED] Sent: 02/10/03 13:04 Subject: RE: [Asterisk-Users] eBay Sip Phone Scam. Actually, had you taken the time to READ the auction details, He says (direct copy/paste from auction) -Begin Copy/Paste- Flash Based OS Easy to install and manage, Cost effective, Easy to use - Friendly GUI for 1st time user, Easy to learn - User's guide and on-line tutorial Big information and management LCD blue back light User friendly keypad Universal AC/DC adapter Ergonomic design 25-button keypad 12-digit caller ID LCD Universal Switching Power Adaptor Input: 100-240VAC Output: +5VDC, 400mA, 1. Auto-sensing 10/100 Base-TX Ethernet Port 2.UL/CE/FCC 3.Power Supply : Universal 90 ~ 264V Support all major Network Operating Systems (Windows, MAC, Linux/Unix) Web-Based Management TCP/IP Configuration with DHCP support Free Flash Firmware update No User Licenses System Restart/ Shutdown Password Access control 1 x 10/100Mbps Ethernet Port (RJ-45 Interface) Support STUN and SIMPLE extension Interoperable with 3d parties Proxy, Registrar and gateway products DSP technology for the best voice quality Support popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (alaw and u-law), G.726 (40K/32K/24K/16K), as well as G.728 (Model 102D) In and out-off-band DTMF Support 3-way conferencing (Model 102D), full duplex hands-free speakerphone, redial, call log, volume control, voice mail with indicator Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS) Remote software upgrade capability via TFTP Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control) -End Copy/Paste- Nowhere does he make the claims you're stating. He DOES, have (Model 102D) in one of the descriptions, but that is a direct quote from Grandstream's product brochure. Also, this phone *IS* out on the market.. I own one, and I'm quite happy with it.. I will tell you this though: Go order one from Chagres (http://www.chagres.com). They are an asterisk supporter/user on this list, and the price is MUCH better. ;) /rant -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of costas Sent: Thursday, October 02, 2003 4:50 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] eBay Sip Phone Scam. I did shop around. Nowhere does he say the phone is the 101. If you look at his ad he says the phone has 102D features and has 16x2 lines and 3 way conference. The starting price was $90. A reasonable opening price I thought. He also does not say the phone is not available until end of year. I only called Grandstream to find out some info on it after I placed the Bid. In a way Grandstream is also at fault. Nowhere do they say the phone is not available. I was suprised
Re: [Asterisk-Users] Problem with Dutch PSTN-line on X100P
Check /proc/interrupts to make sure the cards are not shareing IRQs with anything. On Thu, 2003-10-02 at 11:43, The Traveller wrote: Yo all, I have a problem with a Dutch (KPN) PSTN-line, connected to an X100P. The call wil sound OK at first, but after 10-20 minutes, the audio will start to crackle. Soon after that, this crackle turns into a continuous noise and the parties won't be able to hear eachother anymore. It also sometimes happens that the party on the TDM400P hears a very loud, short-delay echo of themselves, best described as talking in a bathroom. It very much sounds like very bad feedback, most of the time. The system concerned is a PIII-750 with an X100P and a TDM400P. The problem occurs when a call is bridged between the X100P and a port on the TDM400P. Calls from the TDM400P to a remote IAX2- connected box seem OK. rxgain / txgain are at their default (0) for all interfaces and I'm using the default echo-cancellation settings, with echocancelwhenbridged enabled. The PSTN-line concerned is a standard analog line from KPN, which also has ADSL coming in over it. The X100P is, ofcourse, connected behind the splitter. When I connect the X100P to an internal analog port of a small legacy home-switch, the problem doesn't seem to occur, either. I'm using opermode=1 for the X100P, but read on this list somewhere that the chip used on the X100P is actually not the international version, so this setting might not have any effect. In this case, it doesn't seem to matter if I use it or not. There's no notable difference. I'll probably investigate this problem further and play with the echo- canceller and rxgain / txgain a bit. Just curious if anyone else is experiencing something similar and might have found a solution already. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] error message 49159
Martin Pycko wrote: We send SIP messages to that device up to 6-7 times and then we stop and this message shows on the console. WARNING[49159]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) So it isn't really an error then, but an artifact of something asterisk is trying to do? I have seen these messages pretty much since the beginning of time, and I figured something was out of spec with my phones. I can't tell from what you say whether it is normal or not to see those messages? Thanks. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with Dutch PSTN-line on X100P
Yo Eric, On Thu, Oct 02, 2003 at 11:56:44 -0500, Eric Wieling wrote: Check /proc/interrupts to make sure the cards are not shareing IRQs with anything. Sorry, forgot to mention it. All Zaptel-cards in that machine already have their own unique interrupts. I will try moving the cards to different slots, though. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] eBay Sip Phone Scam.
Can someone post their experiences with these phones together with asterisk, and give an impartial listing of what features they find indispensible, and others that are a pain to have missing. What codec configurations do people use these phones in currently with asterisk ? Can someone clarify the transfer questions, and any other related topics as well please. For example do they support callwaiting callerid ? (either in or out of band in conjunction with asterisk ?) At 10:52 AM 10/2/2003 -0600, you wrote: Grandstream 102D won't be available until December, at the earliest. I have 101's in stock now and can ship same day as the order is funded. 102 (not the D model) are on backorder and I expect inventory by the end of next week. Transfering and other functions are really going to be a matter of what your SIP server does, feature wise. GAPS is not free. It starts at $3000 and goes up to 18,000 We have built our own system that we use internally to configure phones for our customers. On Thu, Oct 02, 2003 at 12:14:42PM -0400, costas wrote: Thanks! Do you have the 102D. Thats what started this. How fast can you ship any of the phones? Also I have question on the phones. 1) If you have 2 callers can you transfer one of them, or you can only transfer if you have the second line available? 2) Automated Provisioning System (GAPS). Free? -- Original Message -- From: John Brown (CV) [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Thu, 2 Oct 2003 09:06:00 -0600 Hi Josh, Costas, Adam, et all We do sell the phones. http://www.chagres.net/products/voip/phones.html and digium cards http://www.chagres.net/products/voip/cards.html plus new things real soon :) and if anyone ever has a problem, go yell at me and I'll try like crazy to fix it. john brown, ceo chagres technologies, inc ps: Chagres.com (Chagres River) is where our company got its name from. I used to live there ;) On Thu, Oct 02, 2003 at 02:21:55PM +0200, Low, Adam wrote: Josh, Pls can you confirm that URL, www.chagres.com doesn't seem to mention the sale of any Grandstream phones Adam -Original Message- From: Josh Roberson To: [EMAIL PROTECTED] Sent: 02/10/03 13:04 Subject: RE: [Asterisk-Users] eBay Sip Phone Scam. Actually, had you taken the time to READ the auction details, He says (direct copy/paste from auction) -Begin Copy/Paste- Flash Based OS Easy to install and manage, Cost effective, Easy to use - Friendly GUI for 1st time user, Easy to learn - User's guide and on-line tutorial Big information and management LCD blue back light User friendly keypad Universal AC/DC adapter Ergonomic design 25-button keypad 12-digit caller ID LCD Universal Switching Power Adaptor Input: 100-240VAC Output: +5VDC, 400mA, 1. Auto-sensing 10/100 Base-TX Ethernet Port 2.UL/CE/FCC 3.Power Supply : Universal 90 ~ 264V Support all major Network Operating Systems (Windows, MAC, Linux/Unix) Web-Based Management TCP/IP Configuration with DHCP support Free Flash Firmware update No User Licenses System Restart/ Shutdown Password Access control 1 x 10/100Mbps Ethernet Port (RJ-45 Interface) Support STUN and SIMPLE extension Interoperable with 3d parties Proxy, Registrar and gateway products DSP technology for the best voice quality Support popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (alaw and u-law), G.726 (40K/32K/24K/16K), as well as G.728 (Model 102D) In and out-off-band DTMF Support 3-way conferencing (Model 102D), full duplex hands-free speakerphone, redial, call log, volume control, voice mail with indicator Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS) Remote software upgrade capability via TFTP Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control) -End Copy/Paste- Nowhere does he make the claims you're stating. He DOES, have (Model 102D) in one of the descriptions, but that is a direct quote from Grandstream's product brochure. Also, this phone *IS* out on the market.. I own one, and I'm quite happy with it.. I will tell you this though: Go order one from Chagres (http://www.chagres.com). They are an asterisk supporter/user on this list, and the price is MUCH better. ;) /rant -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of costas Sent: Thursday, October 02, 2003 4:50 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] eBay Sip Phone Scam. I did shop around. Nowhere does he say the phone is the 101. If you
Re: [Asterisk-Users] (still) channel problems
Two IDENTICAL MACHINES (same motherboard, same RH 7.2, same *) communicate through IAX2. Everything works ok on machine 1. On machine 2, if I try to use 4 fxo's from a TDM400 card, sound gets lousy. If I manually destroy one of the zap channels (e.g. zap destroy channel 4), sound gets good again. I had a similar problem once with a mb with bad processor caps. The noise from the power supply screwed up every analog card in the machine. Didn't figure it out until the caps leaked all over the mb. Probably does not apply in this case. Maybe if the machines are identical you can try swapping hardware back and forth.. I'd start by swapping harddrives and see if the problem follows the drive or the system. Mark Farver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WINXP Messenger SIP Client (Good News, Bad News)
WipeOut wrote: Olle E. Johansson wrote: I still can't get Windows messenger to register with a secret to Asterisk. Anthony - do you connect without registering or does Windows messenger register properly with your * ? /O Have you tried forcing Asterisk to use plain text authentication for that SIP account?? The default is MD5 which is not supported by M$.. Can't find an SIP.conf option for doing that. Looked in the Wiki, in the Handbook and in the sample config. ??? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] error message 49159
Hi Martin Please explain, why did you send the messages? miklos - Original Message - From: Brian Capouch [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, October 02, 2003 2:04 PM Subject: Re: [Asterisk-Users] error message 49159 Martin Pycko wrote: We send SIP messages to that device up to 6-7 times and then we stop and this message shows on the console. WARNING[49159]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) So it isn't really an error then, but an artifact of something asterisk is trying to do? I have seen these messages pretty much since the beginning of time, and I figured something was out of spec with my phones. I can't tell from what you say whether it is normal or not to see those messages? Thanks. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WINXP Messenger SIP Client (Good News, Bad News)
auth=plain On Thu, 2003-10-02 at 12:36, Olle E. Johansson wrote: WipeOut wrote: Olle E. Johansson wrote: I still can't get Windows messenger to register with a secret to Asterisk. Anthony - do you connect without registering or does Windows messenger register properly with your * ? /O Have you tried forcing Asterisk to use plain text authentication for that SIP account?? The default is MD5 which is not supported by M$.. Can't find an SIP.conf option for doing that. Looked in the Wiki, in the Handbook and in the sample config. ??? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Has anyone got * working with Xten softphones
On 2/10/03 5:59 pm, Joseph Finley [EMAIL PROTECTED] wrote: Very basic: Sip.conf ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = sip-phones; Default for incoming calls bandwidth=low disallow=all allow=ulaw allow=alaw [3010] type=friend secret=testx host=dynamic qualify=400 callerid=Joseph 3010 dtmfmod=inband mailbox=3010,1234 canreinvite=no reinvite=no nat=yes I've tried those settings and so far so good. At least it seems to be talking to asterisk. What are your settings in sip.conf? Also how do you get the soft phone to dial an internal extension? Thanks for your help. Fingers crossed I figure this out. Fats. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users That worked a treat. I did not have dtmfmod=inband set in the sip file hence why I could not get it working. Thanks a lot for your help it's great to get it working. Regards Fats -- Fats Neutron [EMAIL PROTECTED] direct +44 (0)20 7274 0386 mobile +44 (0)79 7045 9548 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk friendly IAX/SIP wholesalers in Australia
Bryan, IP Telephonics is developing a VoIP gateway service in Australia. It is not yet operational. If you want to discuss anything please email me offlist. Peter Brown At 23:23 2/10/2003 +1000, you wrote: its a fair question: does anyone know any? Bryan Nolen Lead Developer http://Arc.Net.AU http://cdonline.com.au ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] error message 49159
It's a WARNING, so if you want to know why your phone doesn't work you can read it or ignore it. regards Martin On Thu, 2 Oct 2003, Brian Capouch wrote: Martin Pycko wrote: We send SIP messages to that device up to 6-7 times and then we stop and this message shows on the console. WARNING[49159]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) So it isn't really an error then, but an artifact of something asterisk is trying to do? I have seen these messages pretty much since the beginning of time, and I figured something was out of spec with my phones. I can't tell from what you say whether it is normal or not to see those messages? Thanks. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Any way to get out of a remote console without stopping *
This probably has an easy solution, but I found it yet. How can I get out of a remote console after using ssh to get into the box, making changes, reload etc. without stopping *? Thanks in advance. Sincerely, Andy Hester Consero ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problem w/ musiconhold mpg123
I'm trying to get musiconhold to work w/ my Asterisk system, and I'm not having much success yet. First, I noticed that nothing happened even after I had enabled all of the options in zapata.conf setup a sample extension in extensions.conf. Then I read something about how Asterisk uses mpg123 to play the files. I discovered that this had not been installed on my system, so I used apt to install it. That install when successfully. But now, instead of the silence I used to get during holds (why would Asterisk have not indicated that it was missing mpg123 to me?), I get this very strange sound that is certainly not the sample mp3 that's in the music on hold directory. It's possible it's that file w/ the pitch and or speed way out of adjustment, I guess, but why would that be happening? As a side note, I've always seen this error message on startup in Asterisk, even though I doubt it'd be critical to play music on hold, since normal messages (like in the included 'demo' context) play fine (GSM, WAV, I suppose): Oct 2 13:23:46 WARNING[1074402464]: File chan_oss.c, Line 423 (soundcard_init): Unable to open /dev/dsp: No such device Anyone have any ideas on this? Thanks, John Lawler ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] eBay Sip Phone Scam.
On Thu, 2 Oct 2003, Michael T Farnworth wrote: The people at chagres.net appear to sell the phones. They do in fact sell the phones, as I bought one from them :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any way to get out of a remote console without stopping *
On Thu, 2003-10-02 at 14:53, Andy Hester wrote: This probably has an easy solution, but I found it yet. How can I get out of a remote console after using ssh to get into the box, making changes, reload etc. without stopping *? Not that it is clean or neat, but control-c is what I use. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Any way to get out of a remote console without stopping *
Simply run the /usr/src/asterisk/safe_asterisk And then type /usr/sbin/asterisk -vvvgcr ^ r being remote console and then you can do everything as if you ran it directly and exit as you wish or STOP NOW to kill it. Regards, Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Hester Sent: Thursday, October 02, 2003 3:53 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Any way to get out of a remote console without stopping * This probably has an easy solution, but I found it yet. How can I get out of a remote console after using ssh to get into the box, making changes, reload etc. without stopping *? Thanks in advance. Sincerely, Andy Hester Consero ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any way to get out of a remote console without stopping *
use quit or ctrl-D Martin On Thu, 2 Oct 2003, Andy Hester wrote: This probably has an easy solution, but I found it yet. How can I get out of a remote console after using ssh to get into the box, making changes, reload etc. without stopping *? Thanks in advance. Sincerely, Andy Hester Consero ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Gastman working in W2Kp.
I did a google search and did not come up with anything on this. I loaded Gastman on a Windows 2000 pro PC and it will not work. It says the following. gastman.exe has generated errors and will be closed by Windows. You will need to restart the program. I have tried to set the compatability down to Windows 98 mode. And same problem. I installed it on a Windows XP pro and it works fine. But the customer does not want to migrate to XP at this time! Is there a fix for this problem? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem w/ musiconhold mpg123
That sound you hear is the sound of mpg321 running. Do an ls -l /usr/bin/mpg123 if it's a symlink to mpg321 then you have found your problem. On Thu, 2003-10-02 at 14:54, john lawler wrote: I'm trying to get musiconhold to work w/ my Asterisk system, and I'm not having much success yet. First, I noticed that nothing happened even after I had enabled all of the options in zapata.conf setup a sample extension in extensions.conf. Then I read something about how Asterisk uses mpg123 to play the files. I discovered that this had not been installed on my system, so I used apt to install it. That install when successfully. But now, instead of the silence I used to get during holds (why would Asterisk have not indicated that it was missing mpg123 to me?), I get this very strange sound that is certainly not the sample mp3 that's in the music on hold directory. It's possible it's that file w/ the pitch and or speed way out of adjustment, I guess, but why would that be happening? As a side note, I've always seen this error message on startup in Asterisk, even though I doubt it'd be critical to play music on hold, since normal messages (like in the included 'demo' context) play fine (GSM, WAV, I suppose): Oct 2 13:23:46 WARNING[1074402464]: File chan_oss.c, Line 423 (soundcard_init): Unable to open /dev/dsp: No such device Anyone have any ideas on this? Thanks, John Lawler ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any way to get out of a remote console without stopping *
on Thu, Oct 02, 2003 at 02:53:00PM -0500, Andy Hester wrote: This probably has an easy solution, but I found it yet. How can I get out of a remote console after using ssh to get into the box, making changes, reload etc. without stopping *? Thanks in advance. Looks like exit will release you from the * console but not stop * from running when I start * with asterisk -vvvc. Then asterisk -r to reconnect. I like to use the screen command to preface other console grabing prgs. screen -A -m -d -S asterisk asterisk -vvvc then screen -r asterisk connects you to the screen you just called -S asterisk. You use key binding similar to minicom to do things... so to release your screen from running * while in a screen session (and not kill the *), just ctrl+a then hit d to detatch. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] eBay Sip Phone Scam.
Hmm, impartial, well I sell them but I'll try :) :) Bad things: Base color, white... new colors coming soon Call waiting ring is a bit annoying The normal ring is really annoying, they are changing this good things: they work cheap easy to setup sound quality very good multi codec selection just make sure you are running 1.0.3.81 or NEWER Yes they support callerID and call waiting callerID transfer works. push the button dial the number hope this helps. its not flowery because I don't want to sound like a sales man :) john brown, ceo chagres technologies, inc http://www.chagres.net/products/voip/phones.html On Thu, Oct 02, 2003 at 01:08:58PM -0400, Jon Pounder wrote: Can someone post their experiences with these phones together with asterisk, and give an impartial listing of what features they find indispensible, and others that are a pain to have missing. What codec configurations do people use these phones in currently with asterisk ? Can someone clarify the transfer questions, and any other related topics as well please. For example do they support callwaiting callerid ? (either in or out of band in conjunction with asterisk ?) At 10:52 AM 10/2/2003 -0600, you wrote: Grandstream 102D won't be available until December, at the earliest. I have 101's in stock now and can ship same day as the order is funded. 102 (not the D model) are on backorder and I expect inventory by the end of next week. Transfering and other functions are really going to be a matter of what your SIP server does, feature wise. GAPS is not free. It starts at $3000 and goes up to 18,000 We have built our own system that we use internally to configure phones for our customers. On Thu, Oct 02, 2003 at 12:14:42PM -0400, costas wrote: Thanks! Do you have the 102D. Thats what started this. How fast can you ship any of the phones? Also I have question on the phones. 1) If you have 2 callers can you transfer one of them, or you can only transfer if you have the second line available? 2) Automated Provisioning System (GAPS). Free? -- Original Message -- From: John Brown (CV) [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Thu, 2 Oct 2003 09:06:00 -0600 Hi Josh, Costas, Adam, et all We do sell the phones. http://www.chagres.net/products/voip/phones.html and digium cards http://www.chagres.net/products/voip/cards.html plus new things real soon :) and if anyone ever has a problem, go yell at me and I'll try like crazy to fix it. john brown, ceo chagres technologies, inc ps: Chagres.com (Chagres River) is where our company got its name from. I used to live there ;) On Thu, Oct 02, 2003 at 02:21:55PM +0200, Low, Adam wrote: Josh, Pls can you confirm that URL, www.chagres.com doesn't seem to mention the sale of any Grandstream phones Adam -Original Message- From: Josh Roberson To: [EMAIL PROTECTED] Sent: 02/10/03 13:04 Subject: RE: [Asterisk-Users] eBay Sip Phone Scam. Actually, had you taken the time to READ the auction details, He says (direct copy/paste from auction) -Begin Copy/Paste- Flash Based OS Easy to install and manage, Cost effective, Easy to use - Friendly GUI for 1st time user, Easy to learn - User's guide and on-line tutorial Big information and management LCD blue back light User friendly keypad Universal AC/DC adapter Ergonomic design 25-button keypad 12-digit caller ID LCD Universal Switching Power Adaptor Input: 100-240VAC Output: +5VDC, 400mA, 1. Auto-sensing 10/100 Base-TX Ethernet Port 2.UL/CE/FCC 3.Power Supply : Universal 90 ~ 264V Support all major Network Operating Systems (Windows, MAC, Linux/Unix) Web-Based Management TCP/IP Configuration with DHCP support Free Flash Firmware update No User Licenses System Restart/ Shutdown Password Access control 1 x 10/100Mbps Ethernet Port (RJ-45 Interface) Support STUN and SIMPLE extension Interoperable with 3d parties Proxy, Registrar and gateway products DSP technology for the best voice quality Support popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (alaw and u-law), G.726 (40K/32K/24K/16K), as well as G.728 (Model 102D) In and out-off-band DTMF Support 3-way conferencing (Model 102D), full duplex hands-free speakerphone, redial, call log, volume control, voice mail with indicator Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS) Remote software upgrade capability via TFTP Support Silence Suppression, VAD (Voice Activity Detection), CNG
Re: [Asterisk-Users] eBay Sip Phone Scam.
Colors other than white seem to be hard to get from GS. I've been asking for multiple weeks for something other than white. I don't have solid ship dates on black or any other color. I'll let folks know when we have something other than white. On Thu, Oct 02, 2003 at 12:45:50PM -0400, Steve Totaro wrote: do you have the other colors besides white? - Original Message - From: costas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, October 02, 2003 12:14 PM Subject: Re: [Asterisk-Users] eBay Sip Phone Scam. Thanks! Do you have the 102D. Thats what started this. How fast can you ship any of the phones? Also I have question on the phones. 1) If you have 2 callers can you transfer one of them, or you can only transfer if you have the second line available? 2) Automated Provisioning System (GAPS). Free? -- Original Message -- From: John Brown (CV) [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Thu, 2 Oct 2003 09:06:00 -0600 Hi Josh, Costas, Adam, et all We do sell the phones. http://www.chagres.net/products/voip/phones.html and digium cards http://www.chagres.net/products/voip/cards.html plus new things real soon :) and if anyone ever has a problem, go yell at me and I'll try like crazy to fix it. john brown, ceo chagres technologies, inc ps: Chagres.com (Chagres River) is where our company got its name from. I used to live there ;) On Thu, Oct 02, 2003 at 02:21:55PM +0200, Low, Adam wrote: Josh, Pls can you confirm that URL, www.chagres.com doesn't seem to mention the sale of any Grandstream phones Adam -Original Message- From: Josh Roberson To: [EMAIL PROTECTED] Sent: 02/10/03 13:04 Subject: RE: [Asterisk-Users] eBay Sip Phone Scam. Actually, had you taken the time to READ the auction details, He says (direct copy/paste from auction) -Begin Copy/Paste- Flash Based OS Easy to install and manage, Cost effective, Easy to use - Friendly GUI for 1st time user, Easy to learn - User's guide and on-line tutorial Big information and management LCD blue back light User friendly keypad Universal AC/DC adapter Ergonomic design 25-button keypad 12-digit caller ID LCD Universal Switching Power Adaptor Input: 100-240VAC Output: +5VDC, 400mA, 1. Auto-sensing 10/100 Base-TX Ethernet Port 2.UL/CE/FCC 3.Power Supply : Universal 90 ~ 264V Support all major Network Operating Systems (Windows, MAC, Linux/Unix) Web-Based Management TCP/IP Configuration with DHCP support Free Flash Firmware update No User Licenses System Restart/ Shutdown Password Access control 1 x 10/100Mbps Ethernet Port (RJ-45 Interface) Support STUN and SIMPLE extension Interoperable with 3d parties Proxy, Registrar and gateway products DSP technology for the best voice quality Support popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (alaw and u-law), G.726 (40K/32K/24K/16K), as well as G.728 (Model 102D) In and out-off-band DTMF Support 3-way conferencing (Model 102D), full duplex hands-free speakerphone, redial, call log, volume control, voice mail with indicator Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS) Remote software upgrade capability via TFTP Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control) -End Copy/Paste- Nowhere does he make the claims you're stating. He DOES, have (Model 102D) in one of the descriptions, but that is a direct quote from Grandstream's product brochure. Also, this phone *IS* out on the market.. I own one, and I'm quite happy with it.. I will tell you this though: Go order one from Chagres (http://www.chagres.com). They are an asterisk supporter/user on this list, and the price is MUCH better. ;) /rant -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of costas Sent: Thursday, October 02, 2003 4:50 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] eBay Sip Phone Scam. I did shop around. Nowhere does he say the phone is the 101. If you look at his ad he says the phone has 102D features and has 16x2 lines and 3 way conference. The starting price was $90. A reasonable opening price I thought. He also does not say the phone is not available until end of year. I only called Grandstream to find out some info on it after I placed the Bid. In a way Grandstream is also at fault.
[Asterisk-Users] RE: WINXP Messenger SIP Client (Good News, Bad News)
WipeOut wrote: Anthony Minessale wrote: I found this information on how to make XP have a dialpad in Windows Messenger which was awesome news Some more crushing news is if you upgrade MSN messenger past ver 4.x it no longer uses SIP.. (so I have been told).. MSN messenger does not use SIP. Windows messenger (another product) use SIP. I still can't get Windows messenger to register with a secret to Asterisk. Anthony - do you connect without registering or does Windows messenger register properly with your * ? /O After just deciding to guess imaginary config options not documented anywhereI took out secret= and made it password= instead and tada!my reasoning was that secret reminded me of radius where secret was not a password but a common shared string to base aencryption algorythm against so i just tried password cos it made more sense and i'm not sure if that is why it is like that or not but it worked for me.[fred]type=friendusername=fredpassword=fredspasshost=dynamicP.S. I have 4.7 from 2 days ago and it is still doing the sip (although you gotta go set the reg key again)nonetheless I still cant log into agentlogin properlyon it. Do you Yahoo!? The New Yahoo! Shopping - with improved product search
Re: [Asterisk-Users] Front end
Look at www.pawbell.com they have the frontend. They even have the NAT problem fixed! - Original Message - From: 23 To: [EMAIL PROTECTED] Sent: Thursday, October 02, 2003 12:01 PM Subject: [Asterisk-Users] Front end Hi, Can anyone help mewith a few links to sites that have either open source or low cost software to run a voip business front end on the internet over linux servers? I have been watching this list for about a month and am interested in building an offshore phone solution, but so far have not found any front end software for interfacing with clients. thanks Scott This message was checked by MailScan for WorkgroupMail. www.workgroupmail.com
RE: [Asterisk-Users] Any way to get out of a remote console without stopping *
Wow look at the choices :) Thanks everyone for the info. I'll try them out. Sincerely, Andy Hester Consero (817)375-1244 (817)937-7977 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Any way to get out of a remote console without stopping *
-Original Message- From: Martin Pycko Sent: Thursday, October 02, 2003 4:13 PM use quit or ctrl-D Martin From what I can tell, * doesn't honor EOF, at least I've had no luck with it. -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any way to get out of a remote console without stopping *
Or you can use safe_asterisk to start * then asterisk -r to connect bkw On Thu, 2 Oct 2003, PJ Welsh wrote: on Thu, Oct 02, 2003 at 02:53:00PM -0500, Andy Hester wrote: This probably has an easy solution, but I found it yet. How can I get out of a remote console after using ssh to get into the box, making changes, reload etc. without stopping *? Thanks in advance. Looks like exit will release you from the * console but not stop * from running when I start * with asterisk -vvvc. Then asterisk -r to reconnect. I like to use the screen command to preface other console grabing prgs. screen -A -m -d -S asterisk asterisk -vvvc then screen -r asterisk connects you to the screen you just called -S asterisk. You use key binding similar to minicom to do things... so to release your screen from running * while in a screen session (and not kill the *), just ctrl+a then hit d to detatch. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: WINXP Messenger SIP Client (Good News, Bad News)
Title: Message I was able to get it to register just fine, but I get no sound. It connects fine, no sound. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony MinessaleSent: Thursday, October 02, 2003 4:31 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] RE: WINXP Messenger SIP Client (Good News, Bad News) WipeOut wrote: Anthony Minessale wrote: I found this information on how to make XP have a dialpad in Windows Messenger which was awesome news Some more crushing news is if you upgrade MSN messenger past ver 4.x it no longer uses SIP.. (so I have been told).. MSN messenger does not use SIP. Windows messenger (another product) use SIP. I still can't get Windows messenger to register with a secret to Asterisk. Anthony - do you connect without registering or does Windows messenger register properly with your * ? /O After just deciding to guess imaginary config options not documented anywhereI took out secret= and made it password= instead and tada!my reasoning was that secret reminded me of radius where secret was not a password but a common shared string to base aencryption algorythm against so i just tried password cos it made more sense and i'm not sure if that is why it is like that or not but it worked for me.[fred]type=friendusername=fredpassword=fredspasshost=dynamicP.S. I have 4.7 from 2 days ago and it is still doing the sip (although you gotta go set the reg key again)nonetheless I still cant log into agentlogin properlyon it. Do you Yahoo!?The New Yahoo! Shopping - with improved product search
[Asterisk-Users] SIP and DSL Bandwidth queries.
Here is my setup 7960(A)--Firewall/PAT--dsl-Internetdsl--Firewall/NAT---7960(B) | | | | 7960(C)--NAT--cable- -dsl -- Asterisk (A) can communicate with (C) only when C is configured with canreinvite=no. The call gets dropped in few seconds if canreinvite is set to yes for C. (A) and (B) can communicate fine when both sides have canreinvite=yes. Since (C) is not working with canreinvite, traffic goes thru Asterisk server. This brings the Dsl connection to asterisk to a crawl. It is so bad that even a idle ssh connection gets disconnected. Is it possible to configure C so that reinvite works. If not what kind of a bandwidth should I have for Asterisk server. Currently it has a upload of 128K. The codec currently getting used is ULAW. Even if I configure 7960's to use g729, show sip channel reports as using ULAW. Thanks, ==ratnakar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP and DSL Bandwidth queries.
Here is my setup 7960(A)--Firewall/PAT--dsl-Internetdsl--Firewall/NAT---7960(B) | | | | 7960(C)--NAT--cable- -dsl -- Asterisk (A) can communicate with (C) only when C is configured with canreinvite=no. The call gets dropped in few seconds if canreinvite is set to yes for C. (A) and (B) can communicate fine when both sides have canreinvite=yes. Since (C) is not working with canreinvite, traffic goes thru Asterisk server. This brings the Dsl connection to asterisk to a crawl. It is so bad that even a idle ssh connection gets disconnected. Is it possible to configure C so that reinvite works. If not what kind of a bandwidth should I have for Asterisk server. Currently it has a upload of 128K. The codec currently getting used is ULAW. Even if I configure 7960's to use g729, show sip channel reports as using ULAW. Thanks, ==ratnakar If you are moving your traffic from behind a NAT, your Asterisk server must have a G.729 license to terminate the traffic, since Asterisk must be the media proxy for the stream. As you are connecting endpoints together that are behind NAT, you would need multiple G.729 licenses - one for every device that would be concurrently talking to the Asterisk server. I do not believe that it is possible to configure C so that reinvite works, though I would be interested in how you do it if you are able to make that function without Asterisk being a media channel proxy (quasi-border session controller.) You should have at least 56kbps for G.729, in my experience, unless you have no other traffic on the end legs of the diagram. G.729 uses less than 32kbps during normal circumstances, but other TCP traffic needs to squeeze in (as you have discovered.) Your Asterisk server will of course need to have N*(leg bandwidth) capacity, where N is the number of legs active at any one time. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP and DSL Bandwidth queries.
Hi guys, Don't want to ruffle feathers, but did I see Ratnakar's email address as being @cisco.com. Is Cisco thinking of using Asterisk? Just a thought. Welcome Ratnakar Peter From: [EMAIL PROTECTED] At 14:50 2/10/2003 -0700, you wrote: Here is my setup 7960(A)--Firewall/PAT--dsl-Internetdsl--Firewall/NAT---7960 (B) | | | | 7960(C)--NAT--cable- -dsl -- Asterisk (A) can communicate with (C) only when C is configured with canreinvite=no. The call gets dropped in few seconds if canreinvite is set to yes for C. (A) and (B) can communicate fine when both sides have canreinvite=yes. Since (C) is not working with canreinvite, traffic goes thru Asterisk server. This brings the Dsl connection to asterisk to a crawl. It is so bad that even a idle ssh connection gets disconnected. Is it possible to configure C so that reinvite works. If not what kind of a bandwidth should I have for Asterisk server. Currently it has a upload of 128K. The codec currently getting used is ULAW. Even if I configure 7960's to use g729, show sip channel reports as using ULAW. Thanks, ==ratnakar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iaxtel fixes
okay, you no longer have to have [iaxtel] as the last entry. It was a config error on x... mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Predictive Dialer
Hi James-- I got a dialer working without too many hiccups about two months ago. It relies on changes to chan_agent, app_queue, a PostgreSQL backend, a Tcl-* manager interface, a bunch of Tcl glue, and some cron jobs. The results for each call are logged in right through the phone key pad, and the algorithm for prediction looks at number of agents logged in, average length of calls, and a magic number the boss man can set to speed it up or slow it down, plus a couple others I forget. Although it relies on some bastardization of the Caller-ID (who doesn't), it is in compliance with all the latest FCC rules. A key to making it stable was the recent placement of extra locks in the queue and agent code. It still gets some frozen lines, but I blame it on the Zhone, and they seem to thaw out when you power cycle the POS channel bank. I know there was a separate list setup for discussions about a predictive dialer, and I would like to contribute my code there but don't remember who made the list or if it has ever seen any traffic. Not to make a meta-comment on this thread, but whenever the discussion of a predictive dialer does arise, it seems to get spit on by those who aren't fans of the technology. I think that's a real shame as it represents a huge market for *. I had some moral qualms about it, too, but they pale in comparison to those I would have if, say, I was hacking on voicemail for the Pentagon or rolling out a PBX at Fox News. --Chris On Thu, 2 Oct 2003, James Coberly waxed: Hi, Some time ago there were posts about Predictive dialing. Has anyone seen or made any forward progress on this ability? I would be very interested in any further info regarding the ability. Thanks, James- -- Chris Maj cmaj_hat_freedomcorpse_hot_info 0xC0051F6A 5EBB 2035 F07B 3B09 5A31 7C09 196F 4126 C005 1F6A ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP and DSL Bandwidth queries.
At 08:54 AM 10/3/2003 +1000, you wrote: Hi guys, Don't want to ruffle feathers, but did I see Ratnakar's email address as being @cisco.com. Is Cisco thinking of using Asterisk? Just a thought. Well if I was a large hardware manufacturer I would certainly be testing compatibility of my hardware with other popular stuff, since only a fool would think people are going to buy open standards based equipment all from one manufacturer. If cisco is doing some testing, great !, but I doubt they are actually planning to deploy asterisk corporate wide. Welcome Ratnakar Peter From: [EMAIL PROTECTED] At 14:50 2/10/2003 -0700, you wrote: Here is my setup 7960(A)--Firewall/PAT--dsl-Internetdsl--Firewall/NAT---7960 (B) | | | | 7960(C)--NAT--cable- -dsl -- Asterisk (A) can communicate with (C) only when C is configured with canreinvite=no. The call gets dropped in few seconds if canreinvite is set to yes for C. (A) and (B) can communicate fine when both sides have canreinvite=yes. Since (C) is not working with canreinvite, traffic goes thru Asterisk server. This brings the Dsl connection to asterisk to a crawl. It is so bad that even a idle ssh connection gets disconnected. Is it possible to configure C so that reinvite works. If not what kind of a bandwidth should I have for Asterisk server. Currently it has a upload of 128K. The codec currently getting used is ULAW. Even if I configure 7960's to use g729, show sip channel reports as using ULAW. Thanks, ==ratnakar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxtel fixes
so does that mean I can now have multiple iaxtel numbers? Doug On Thu, 2 Oct 2003 17:56:43 -0500 (CDT), Mark Spencer [EMAIL PROTECTED] wrote: okay, you no longer have to have [iaxtel] as the last entry. It was a config error on x... mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Using M2, Opera's revolutionary e-mail client: http://www.opera.com/m2/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Predictive Dialer
Chris are you willing to post the code? Peter At 19:09 2/10/2003 -0400, you wrote: Hi James-- I got a dialer working without too many hiccups about two months ago. It relies on changes to chan_agent, app_queue, a PostgreSQL backend, a Tcl-* manager interface, a bunch of Tcl glue, and some cron jobs. The results for each call are logged in right through the phone key pad, and the algorithm for prediction looks at number of agents logged in, average length of calls, and a magic number the boss man can set to speed it up or slow it down, plus a couple others I forget. Although it relies on some bastardization of the Caller-ID (who doesn't), it is in compliance with all the latest FCC rules. A key to making it stable was the recent placement of extra locks in the queue and agent code. It still gets some frozen lines, but I blame it on the Zhone, and they seem to thaw out when you power cycle the POS channel bank. I know there was a separate list setup for discussions about a predictive dialer, and I would like to contribute my code there but don't remember who made the list or if it has ever seen any traffic. Not to make a meta-comment on this thread, but whenever the discussion of a predictive dialer does arise, it seems to get spit on by those who aren't fans of the technology. I think that's a real shame as it represents a huge market for *. I had some moral qualms about it, too, but they pale in comparison to those I would have if, say, I was hacking on voicemail for the Pentagon or rolling out a PBX at Fox News. --Chris On Thu, 2 Oct 2003, James Coberly waxed: Hi, Some time ago there were posts about Predictive dialing. Has anyone seen or made any forward progress on this ability? I would be very interested in any further info regarding the ability. Thanks, James- -- Chris Maj cmaj_hat_freedomcorpse_hot_info 0xC0051F6A 5EBB 2035 F07B 3B09 5A31 7C09 196F 4126 C005 1F6A ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Predictive Dialer
On Thu, 2003-10-02 at 17:09, C. Maj wrote: I know there was a separate list setup for discussions about a predictive dialer, and I would like to contribute my code there but don't remember who made the list or if it has ever seen any traffic. That list was set up by me back in April. There wasn't much traction at that time, but for those interested in the subject of predictive dialing and Asterisk the subscription page for the list is http://www.putland.linux-site.net/mailman/listinfo/astdialer-dev The best thing I can say for contribution is either to post it to the dialer list or open a sourceforge project for it. --Karl -- Karl Putland [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SIP and DSL Bandwidth queries.
yes i work for cisco. But playing around with asterisk is purely personal. It is in no way related to my work at cisco. I tried using another email id yesterday, but the post never showed up. Even though I got a mail from the news server that it was posted. Thanks, =ratnakar Jon Pounder wrote: At 08:54 AM 10/3/2003 +1000, you wrote: Hi guys, Don't want to ruffle feathers, but did I see Ratnakar's email address as being @cisco.com. Is Cisco thinking of using Asterisk? Just a thought. Well if I was a large hardware manufacturer I would certainly be testing compatibility of my hardware with other popular stuff, since only a fool would think people are going to buy open standards based equipment all from one manufacturer. If cisco is doing some testing, great !, but I doubt they are actually planning to deploy asterisk corporate wide. Welcome Ratnakar Peter From: [EMAIL PROTECTED] At 14:50 2/10/2003 -0700, you wrote: Here is my setup 7960(A)--Firewall/PAT--dsl-Internetdsl--Firewall/NAT---7960 (B) | | | | 7960(C)--NAT--cable- -dsl -- Asterisk (A) can communicate with (C) only when C is configured with canreinvite=no. The call gets dropped in few seconds if canreinvite is set to yes for C. (A) and (B) can communicate fine when both sides have canreinvite=yes. Since (C) is not working with canreinvite, traffic goes thru Asterisk server. This brings the Dsl connection to asterisk to a crawl. It is so bad that even a idle ssh connection gets disconnected. Is it possible to configure C so that reinvite works. If not what kind of a bandwidth should I have for Asterisk server. Currently it has a upload of 128K. The codec currently getting used is ULAW. Even if I configure 7960's to use g729, show sip channel reports as using ULAW. Thanks, ==ratnakar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323 Ringing Congestion causes * segfault
We have an odd problem, where inbound H323 (chan_h323) calls will sometimes cause a Ringing Congestion that appears to keep the channels open and never release it until we kill and restart asterisk. These Ringing Congestions start to pile up, which eventually crashes Asterisk. H323 Gateway - Asterisk (chan_h323) - Tor2/PRI - PSTN Has anyone ran into this problem or know how to resolve it? The H323 device making the calls doesn't seem to have a problem calling other H323 gateways or gatekeepers, this problem only appears in Asterisk. Again this problem is intermittent and occurs once a day. I have included a paste of the Ringing Congestions below as well as the GDB dump. Thanks --- H323/ip$61.33.231.34:24585/5 (h323 17704703893 4 ) Ringing Congestion(Empty) H323/ip$61.33.231.34:24581/2 (h323 12128665244 4 ) Ringing Congestion(Empty) H323/ip$61.33.231.34:24596/15 (h323 12128665244 4 ) Ringing Congestion(Empty) H323/ip$61.33.231.34:24592/11 (h323 12128665244 4 ) Ringing Congestion(Empty) H323/ip$61.33.231.34:24591/10 (h323 12128665244 4 ) Ringing Congestion(Empty) H323/ip$61.33.231.34:24589/8 (h323 12128665244 4 ) Ringing Congestion(Empty) H323/ip$61.33.231.34:24581/1 (h323 12128665244 4 ) Ringing Congestion(Empty) H323/ip$61.33.231.34:24647/67 (h323 12128665244 4 ) Ringing Congestion(Empty) H323/ip$61.33.231.34:24644/64 (h323 14349230857 4 ) Ringing Congestion(Empty) H323/ip$61.33.231.34:24643/63 (h323 14349230857 4 ) Ringing Congestion(Empty) H323/ip$61.33.231.34:24641/61 (h323 19788482664 4 ) Ringing Congestion(Empty) H323/ip$61.33.231.34:24640/60 (h323 19788482994 4 ) Ringing Congestion(Empty) H323/ip$61.33.231.34:24634/54 (h323 18586380364 4 ) Ringing Congestion(Empty) H323/ip$61.33.231.34:24608/28 (h323 12062233600 4 ) Ringing Congestion(Empty) - (gdb) bt #0 connection_made (call_reference=1106240992) at chan_h323.c:1188 #1 0x41ef7973 in MyH323EndPoint::OnConnectionEstablished(H323Connection, PString const) ( this=0x814c1a8, [EMAIL PROTECTED], [EMAIL PROTECTED]) at ast_h323.cpp:294 #2 0x482985f5 in H323Connection::OnEstablished() () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #3 0x482a215e in H323Connection::InternalEstablishedConnectionCheck() () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #4 0x48297d28 in H323Connection::HandleSignalPDU(H323SignalPDU) () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #5 0x48297902 in H323Connection::HandleSignallingChannel() () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #6 0x482a8795 in H225CallThread::Main() () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #7 0x47b750a7 in PThread::PX_ThreadStart(void*) () from /usr/src/pwlib/lib/libpt_linux_x86_r.so.1 #8 0x40031332 in start_thread () from /lib/tls/libpthread.so.0 (gdb) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WINXP Messenger SIP Client (Good News, Bad News)
I think there's a confusion here. There're 2 different products: 1. MSN Messenger 4.6/4.7 (Windows 9x, ME, 2K) 2. Windows Messenger 4.7/5.0 (Windows XP) I was told MSN Messenger 4.7 works with the registry hack. Have never tested this myself, though I'm very certain 5.0 doesn't work. Windows Messenger is a different animal. It's a SIP UA which happens to understand MSN protocol. It's using Windows latest RTC API (Real Time Communication). It's meant to connect to the new M$ SIP server (codename Greenwich). Both version 4.7 and 5.0 works with SIP servers. You can use the RTC API to build you own SIP UA. I've written a simple VB client for use with SIP servers. If anyone's interested, just drop me a mail. Bear in mind, it only works in Windows XP (which has built-in RTC). WipeOut wrote: Anthony Minessale wrote: I found this information on how to make XP have a dialpad in Windows Messenger which was awesome news HKEY_CURRENT_USER\Software\Microsoft\MessengerService\CorpPC2PHone (change it from 0 to 1 and a magic new choice to make phone calls appears) only to be crushed hours later when I realized It doesnt seem to do dtmf right. If i make an ext lead to AgentLogin for instance and press my extension when I hit # i get like 4 overlapping incorrect errs because it must be sending like 4 # digits instead of 1. I dont see a place to change the dtmf in xp and only rfc mode works any other method is ignored.. Did anyone ever get this to work right? estara works right in the same dtmf mode so i'm inclined to blame windows. It at least works as a phone gateway cos you can dial the inital ext properly when placing the call. Do you Yahoo!? The New Yahoo! Shopping http://shopping.yahoo.com/?__yltc=s%3A15443%2Cd%3A22708228%2Cslk%3Atext%2Csec%3Amail - with improved product search Some more crushing news is if you upgrade MSN messenger past ver 4.x it no longer uses SIP.. (so I have been told).. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Does gnophone 0.2.5 work? Other god sftphones?
I checked out gnophone from CVS and I'm trying to build it. I got as far as getting a ./configure built and that to build the makefiles and then I find compile poblems in the source. Leads me to thing maybe 0.2.5 is still a work in progress. true? One more question. What software phones are people likeing for Linux/w2k/Solaris I want to build a voip-only system and of course need good quality softphones = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX and IAXTEL
Mark, The location of the guest / iaxtel section having to be at the end is, as it turns out, a configuration error on iaxtel. I hope to have it straightened out shortly. Ok but I fixed it. Just put the guest section in iax.conf all the way on the end. And right now it works for me. :) Since many are having a tough time with documentation, would it be possible to add at least a few words with some of these explanations to give users a slight clue (more then ...an error...) so we have some idea as to whether it might be important to react to cvs updates (etc)? Just a simple sentence like 'should have no effect on anyone other then those experiencing new iax link issues' would have an entirely different impact then 'the error may allow anomymous access to the full dialplan' would be nice. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New TDM cards--driver won't load
I've searched the site with google, but can't think of the magic words I guess. I got a swap out TDM30 today to replace my buzzy one. I swapped it with the older one, swapped out the FXS modules, hooked it up to the computer's power supply, and booted, but the wcfxs driver won't load--it gives me the standard Cannot init module types of errors that happen when one tries to load drivers but they can't find the hardware. I have checked, and there aren't any interrupt conflicts. Is there some change that I need to know about wrt the drivers? Thanks. B. -- This message has been scanned for viruses and is believed to be clean. Scan engine v4.2.40 for Linux. Virus data file v4294 created Sep 18 2003 Scanning for 80178 viruses, trojans and variants. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX and IAXTEL
there is no security risk, actually it reduces access. The bug meant that a [iaxtel] section had to be the last section in the file, otherwise it would be ignored. If you aren't having a problem with authenication in iax (people who have access but are getting rejected), you won't need the update. Mark, The location of the guest / iaxtel section having to be at the end is, as it turns out, a configuration error on iaxtel. I hope to have it straightened out shortly. Ok but I fixed it. Just put the guest section in iax.conf all the way on the end. And right now it works for me. :) Since many are having a tough time with documentation, would it be possible to add at least a few words with some of these explanations to give users a slight clue (more then ...an error...) so we have some idea as to whether it might be important to react to cvs updates (etc)? Just a simple sentence like 'should have no effect on anyone other then those experiencing new iax link issues' would have an entirely different impact then 'the error may allow anomymous access to the full dialplan' would be nice. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] THE NAT-MARE IS OVER test volunteers needed
5volunteers needed to test NAT Transversal software in realtime enviroment. Must be behind a firewall. Reply to [EMAIL PROTECTED] if you would like to join the test. This message was checked by MailScan for WorkgroupMail. www.workgroupmail.com
Re: [Asterisk-Users] New TDM cards--driver won't load
Is it showing up on /proc/pci? It should be a tigerjet. Does dmesg report anything unusual? There are *some* machines which have no no 3.3V supply. If that's the story with yours, send me your machine and I'll try an experimental fix on it. Mark On Thu, 2 Oct 2003, Brian Capouch wrote: I've searched the site with google, but can't think of the magic words I guess. I got a swap out TDM30 today to replace my buzzy one. I swapped it with the older one, swapped out the FXS modules, hooked it up to the computer's power supply, and booted, but the wcfxs driver won't load--it gives me the standard Cannot init module types of errors that happen when one tries to load drivers but they can't find the hardware. I have checked, and there aren't any interrupt conflicts. Is there some change that I need to know about wrt the drivers? Thanks. B. -- This message has been scanned for viruses and is believed to be clean. Scan engine v4.2.40 for Linux. Virus data file v4294 created Sep 18 2003 Scanning for 80178 viruses, trojans and variants. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New TDM cards--driver won't load
Mark Spencer wrote: Is it showing up on /proc/pci? It should be a tigerjet. Yes. I put the other card back in (production machine) but over the weekend I'll get the card in there and capture the output of lspci. Does dmesg report anything unusual? Nope. Doesn't show any sign of seeing the card at all. It does see the FXO card that's in the same machine. There are *some* machines which have no no 3.3V supply. If that's the story with yours, send me your machine and I'll try an experimental fix on it. I'm going to play a bit first. . . I have a bunch of machines and tomorrow the class I am using this gear with has lab day. We'll spend some time playing and see what we can find out. Thanks for the quick reply. The machine is old--an HP Vectra 400, but not *that* old. I haven't yet had any problems with it other than apparently it's PCI version I, and so a PCI/PCMCIA adapater I got doesn't work with it. Thanks. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] echo for 15 seconds
Shaun == Shaun Ewing [EMAIL PROTECTED] writes: Shaun - Original Message - Shaun From: Chad R. Graham For the first 15 seconds of a call I get echo on the ata 186 side only. I assume after that the echo canceller kicks in but is there any way to make it happen faster? Shaun Same thing here - except we're using Cisco 7960 and 7940 IP Shaun phones. Shaun We're getting used to it, the main thing is that the remote Shaun caller doesn't hear it (which they don't). A person visiting our Shaun office and using the phone may get a bit of a surprise though. [...] I'm also hearing this, with an analog phone (connected to an S100U). Rather annoying. Incoming calls have an entirely different problem for me, a disastrous 5-8 second crackling/clicking sound, which seems to go quiet a while after you start speaking. The other side doesn't hear it, but it makes you miss the beginning of a call, e.g. you usually don't know who's calling :-/ This happens in a phone - S100U - * - * - X100P - PSTN setup, when somebody is calling from the PSTN. --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Date: header
So, a quick look through a full session of a call between two SIP phones doesn't show that there is a Date: header being inserted anywhere in the SIP headers. I _swear_ I saw that earlier, and in fact, I recall watching Mark fix some syntax this spring on the floor of the VON show to make the SNOM phones work correctly for call timestamps. I even found the Date: header code block in chan_sip, but no evidence of it in a packet dump with CVS from this morning. Is it only inserted under special circumstances, or is it on any transmission? JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] the g729 situation
LDM == Louis-David Mitterrand [EMAIL PROTECTED] writes: LDM Having purchased a license for 5 g729 channels on Digium's web LDM shop I thought registration and installation would be a snap. NOT. LDM I followed registration instructions to the letter but it failed LDM with that message: LDM ERROR! Your Internet connection is probably behind a proxy and the LDM Registration program can't communicate with our server LDM Which is stupid as my * box is a firewall and sits directly on the LDM Internet whith no restrictions from in-out. I must say I'm impressed that people are brave enough to (1) accept the long, restrictive and sometimes outright scary (did you read the parts about credit card charges, or the definition of G.729 software in connection with Improvement by Licensee?) licensing agreement and (2) run a binary module that touches strange parts of the machine and communicates that information over the network to a third party. I also feel sorry for Digium, because they have to take the heat from unhappy users. IMHO this codec should be avoided at all cost. --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configs for IAX IAX trunk
Brian == Brian West [EMAIL PROTECTED] writes: Brian Just a heads up.. you can't loop switch statements ie Brian BOX A switch = BOX B BOX B switch = BOX A [...] I was actually wondering -- why? This is something I very naturally wanted to do the first time I configured two *'s. I wanted them to exchange dialplans, so that I don't have to replicate this information. I have some extensions on one of them, and others on the other, they are all unique and I want them all to be globally callable. So, why can't one do something like this? Is this a valid feature request? --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with GPL license of Asterisk
Mark == Mark Spencer [EMAIL PROTECTED] writes: [...] Mark No problem, it's easy to get confused :) I would, however, take Mark issue with the GPL being evil. It's not my *ideal* license, Mark but it certainly is good enough. Just for the reference, while we're at it. GPL does have an issue, which can cause problems to some people or companies. It is often overlooked, because the open source issues seem much more controversial. Having worked with GPL software quite a bit, also in the commercial world, and having gotten some legal advice, I believe that the anti-patent clauses in the GPL and LGPL are quite possibly the biggest problem preventing the use of GPL'd software by commercial entities, much bigger than the pass on the source and the rights requirement. An excerpt from the GPL: 7. If, as a consequence of a court judgment or allegation of patent infringement or for any other reason (not limited to patent issues), conditions are imposed on you (whether by court order, agreement or otherwise) that contradict the conditions of this License, they do not excuse you from the conditions of this License. If you cannot distribute so as to satisfy simultaneously your obligations under this License and any other pertinent obligations, then as a consequence you may not distribute the Program at all. For example, if a patent license would not permit royalty-free redistribution of the Program by all those who receive copies directly or indirectly through you, then the only way you could satisfy both it and this License would be to refrain entirely from distribution of the Program. [...] 8. If the distribution and/or use of the Program is restricted in certain countries either by patents or by copyrighted interfaces, the original copyright holder who places the Program under this License may add an explicit geographical distribution limitation excluding those countries, so that distribution is permitted only in or among countries not thus excluded. In such case, this License incorporates the limitation as if written in the body of this License. As I understand it (and as my legal counsel advises me) this effectively means that if I distribute GPL/LGPL code, I have to make sure that its distribution and re-distribution is not restricted by patents (or other restrictions). If the code in question contains parts which some patents lay claim to, restricting distribution, then I must not distribute the code at all. Furthermore, by distributing the code I breach the GPL and expose myself to legal threat of a lawsuit from the FSF. It is needless to mention that it is impossible to me to verify that no patents (worldwide!) lay claim to the code I'm distributing and impose restrictions upon its distribution. Sooner or later I'm going to find out that I do not comply with the GPL, because I distribute GPLd code even though there are patent restrictions that apply to it. An example of a particularly clear case of this problem is the XviD code (http://www.xvid.org/), which is GPL-licensed. It seems to me that the authors (copyright holders, to be precise) may distribute the software under any license they choose, but nobody else is allowed to re-distribute it, because they would be violating section 7 of the GPL, as the MPEG-4 compression is (in some countries) covered by patents requiring royalties to be paid. This is an issue which is very often overlooked in the hot GPL debates. However, in the commercial world, it is possibly the most important one. Conclusion (IMHO of course): if you have the choice, use a license that is OSI-compliant but does not have the anti-patent clause. Or has it phrased differently. --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] echo for 15 seconds 002401c38308$2e05e0a0$0102010a@JUPITER
On Thu, 2003-10-02 at 12:04, Jan Rychter wrote: I'm also hearing this, with an analog phone (connected to an S100U). Rather annoying. Incoming calls have an entirely different problem for me, a disastrous 5-8 second crackling/clicking sound, which seems to go quiet a while after you start speaking. The other side doesn't hear it, but it makes you miss the beginning of a call, e.g. you usually don't know who's calling :-/ This happens in a phone - S100U - * - * - X100P - PSTN setup, when somebody is calling from the PSTN. The first server that I set up asterisk on had the same problem. I was using BudgeTones and a couple X100P's. Internal calls had no echo, etc, but calls over the X100P's had tons of echo for 10-15 sec. We also got a beeping sound. However, since the problem didn't seem widespread among X100P users, we decided it might be our server hardware, which while decent spec wise, was on the cheap end quality wise. We got some nicer hardware, and the problem went away. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] eBay Sip Phone Scam.
On Thu, 2003-10-02 at 07:51, Josh Roberson wrote: Ok, see, now you're confusing what I said. Nowhere did I say I had the 102D. I said he never mentioned that it was the 102, irregardless of the D. I *DO* have the 101, which is what he was talking about. No, it doesn't mention it's the 101. This argument has now proved silly, especially since you're confusing what I'm saying, with what he supposedly is. Actually, when this was first posted to the list, I looked at the eBay listing. It specifically said that the phone had a 16x2 display, which is only found on the 102D. It seems that the listing has been changed since then, which would explain the confusion between you two. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice detection
Does anyone know if there's public voice detection algorithms available? I've scoured the net for the last hour or so, and I can't come up with anything except a few proprietary or embedded solutions. I know dsp.c uses goertzel algorithms for DTMF detection, but how does one detect voice? I dunno, maybe detecting voice isn't the way to go. I want to begin playback of a file after a phone/answering machine has answered. Suggestions? Brad Waite ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help-to start Asterik PBX
Hi I am trying to get started with asterik PBX.I need to establish a call between an H.323 terminal (example : Netmeeting) and a SIP terminal. I would like to know : 1)What are the configuration to be done Asterik PBX (I coild build the source on Redhat Linux 7.3). 2)How to configure the extensions 3)How to make call from H.323 terminal (Do we need to register with PBX like registering with Gatekeeper). Can any body provide some pointers for the above clarifications. Thanks for your time. venkateswaran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
