RE: [Asterisk-Users] A good HW
Hello Jan, It depends how many telephone lines you will connect with asterisk - the more lines you need the more expensive hardware you will need. For a single line the hardware can be as cheap as $20 (used) and for more lines the price increases rapidly. http://www.digium.com/index.php?menu=product_categorycategory=hardware Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jan Buchal Sent: Monday, September 05, 2005 7:35 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] A good HW Hello, I am working in non profit organisation Brailcom which develop Free Software for blind and visually impaired people. Now we think about a new switchboard for our current work and for better communication with our blind clients. If I good understand can be useful asterisk with some hw card for us. We have this requests: - linking with current analog provider - forwarding usually phone service to voip - forwarding voip to usually phone service - we will to use in Czech republic, Europe If I good understand for this we can use some HW which list I founded in asterisk documentation. However I do not know what kind will be good for us. The price is very important for us of course. Do you help me please and suggest some telephony card? thanks. -- Jan Buchal Tel: (00420) 224921679 Mob: (00420) 608023021 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] STUN on PAP2-NA 2.0.12(LS)
Hello, I'm having intermittent STUN trouble. Every one out of perhaps 5 reboots the PAP2 contacts STUN ... on the other attempts it just skips that step all together. I have been verifying this using ethereal which shows the distinctive STUN server DNS lookup followed by about 10 STUN queries (when it works - when it doesn't it skips all that including the initial DNS lookup ... apparently it doesn't even try) Has anyone else had this trouble? It's hard to find new firmware for PAP2-NA ... but maybe that's the problem ... is 2.0.12 very obsolete? The newest I saw on the web was 2.0.13 and even that one was hard to find. Thank you, Tomas ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] random beeps in MeetMe
Depends what the beep sounds like ... but I've been having this system on a busy system which has XP100 and the Ethernet cards or other devices sharing one IRQ. You might need to spread out the IRQs so that XP100 get's its own and Ethernet gets another one of its own. How fast of a system is it? Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Brown Sent: Friday, September 02, 2005 7:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] random beeps in MeetMe I have 3 users in a meetme conference. 2 of them are monitor only. I get a random beep in the audio during the conference. There appears to be no pattern. The 2 monitors are SIP softphones and the third is a POTS line on an XP100 card. disconnecting either of the monitors does not resolve the situation. This is currently a test box, so I would consider some sort of hardware issue a possibility, but I just want to make certain that there is not an asterisk issue here. Anyone have any thoughts on where I should start to look? Thanks BEN ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] canreinvite = yes with PAP2
Has anyone made this work? For me everything is fine until I switch canreinvite form no to yes. What happens is that asterisk hangs on attempting native bridge ... from what I understand attempting native bridge means that the RTP is routed through asterisk (just without any codec translation) But it shouldn't do that ... right? ... canreinvite is set to yes ... What's the best way to deal with this issue? I've also read that the only way to get the following situation ... UA --- NAT --- Internet --- NAT --- UA ... to work without passing the media path through asterisk is to use SER together with asterisk. Is that still true or was that because I was reading stuff from back in 2003? Some other discussions mention that canreinvite will simply not work with certain UAs .. is PAP2 one of those? .. Couple of other discussions that I've seen conclude that passing media stream UA-to-UA is just not practical when NAT is involved and is best to be avoided all together ... I'd like to make it work because it seems like a great way to save expensive server bandwidth. But if it will cause more trouble than it's worth then I will probably pass the media path through Asterisk and live with the fact that it will eat up my bandwidth. Also, IAX is superior when dealing with NATs , does it also handle UA-to-UA in NATed environment smoothly? What would be a good PAP2 alternative that uses IAX? This is my sip.conf: [1001] username=1001 type=friend secret= qualify=yes port=5060 nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=yes callerid=Test1 1001 ... My PAP2 is configured with: STUN=yes STUN=stun.xten.net NAT Keepalive = 15 Outbound proxy = blank Proxy = IP of asterisk Any suggestions? Thank you, Tomas ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Registrar only setup
Hello, Im having trouble figuring out how to setup Asterisk so that its only a registrar not passing any RTP data during phone calls. So far I got this far: Asterisk server holds registration information for phones Phones register with Asterisk giving it their ip+port where they can be currently contacted NAT doesnt seem to be a problem because STUN seems to take care of it nicely for me. The hard part that I dont understand is this: Phones can call each other BUT all the RTP traffic is passed through Asterisk I dont want this, I need that the phones call each other directly based on the registration info stored in Asterisk. Im having hard time wrapping my head around this I think Im missing some key part but the way I understand Asterisk is that it listens for requests on the SIP channel, when it gets a request it handles it appropriately using its dial plan. But in the dial plan the only thing that makes sense to use is dial and once I do that all the RTP is sent through asterisk (in-out) to the other phone right? Or maybe the problem is on the phone setup? I tried to make sure that Im not specifying any outbound proxy but I do have to specify proxy otherwise it will not know where to register right? Or maybe Im all messed up 8-P I thought I understood asterisk at least a *bit* until I came across this :-) Thanks for any clarification, Tomas ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Registrar only setup
No I havent tried it but looks like exactly what Im missing. Thanks Ariel ! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ariel Batista Sent: Tuesday, August 30, 2005 7:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Registrar only setup have you tried in the sip.conf for the devices canreinvite=yes - Original Message - From: Tomas Florian To: asterisk-users@lists.digium.com Sent: Tuesday, August 30, 2005 8:48 PM Subject: [Asterisk-Users] Registrar only setup Hello, Im having trouble figuring out how to setup Asterisk so that its only a registrar not passing any RTP data during phone calls. So far I got this far: Asterisk server holds registration information for phones Phones register with Asterisk giving it their ip+port where they can be currently contacted NAT doesnt seem to be a problem because STUN seems to take care of it nicely for me. The hard part that I dont understand is this: Phones can call each other BUT all the RTP traffic is passed through Asterisk I dont want this, I need that the phones call each other directly based on the registration info stored in Asterisk. Im having hard time wrapping my head around this I think Im missing some key part but the way I understand Asterisk is that it listens for requests on the SIP channel, when it gets a request it handles it appropriately using its dial plan. But in the dial plan the only thing that makes sense to use is dial and once I do that all the RTP is sent through asterisk (in-out) to the other phone right? Or maybe the problem is on the phone setup? I tried to make sure that Im not specifying any outbound proxy but I do have to specify proxy otherwise it will not know where to register right? Or maybe Im all messed up 8-P I thought I understood asterisk at least a *bit* until I came across this :-) Thanks for any clarification, Tomas ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoIP Mythbusters Help!! - NATed phone-phone connection without proxy? Possible? Yes/No
Hello, All Im looking for is a yes/no answer here. I have heard that the following scenario is possible (reasonably easy to implement as well) but I just dont get it :-) if it is possible Ill go ahead and learn on my own, I just dont want to waste time on something that will not work. Scenario: 2x VoIP phones - Each phone is configured to register with SIP server 139.142.111.1 - Each phone is behind a standard NAT device (say regular home Linksys router with no ports manually forwarded its out of the box configuration) - Each phone is configured to use STUN to find out its external IP and the type of NAT its behind 1x Asterisk Server for SIP registration - 2 SIP peers defined with extensions 200 and 201 I already know I can make the phones call each other NP but the RTP data is routed over the Asterisk consuming bandwidth on that server (in+out). The real question is: Can I have no RTP bandwidth consumed by the Asterisk server? (SIP data allowed) Supposedly the 2 VoIP phones can talk to each other directly through the NAT once STUN and SIP do their *magic* to establish their RTP connection. So can this be done or did I pick up some myth somewhere? Also, if it can be done, how to I block the VoIP phones from sending their RTP over the Asterisk in case they cant negotiate a direct connection between each other? Thank you very much, Tomas ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Kind of Computer to use
I had a recent bad experience with Compaq with asterisk. For some reason it could not be forced to put my PCI cards on anything other than IRQ 11. This is a major problem because when all the cards (2 xnetwork,2x FXS) are active and generate interrupts it chops up the sound quality. I was trying to convince the bios to split it up for 2 days but then I gave up. Mind you, this is not a very powerful system at all so on a faster machine it might not be a problem (it could just handle the interrupts fast enough without making the sound choppy). But I would think that no matter how fast the machine is its always better to have the IRQs split up and not shared. I never tried on dell yet . Tomas From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B. Asterisk Users Sent: Wednesday, June 29, 2005 6:33 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Kind of Computer to use Hi, I am building PBX's for clients. I was thinking of using Dell computers. I was told that they do not work well with asterisk. Any one have any suggestions ? Any other brands that work well with asterisk ? Also any specific hardware to or not to use ? Finally does that Mac Mini work well with asterisk ? Thanks a lot. Dovid ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DID in AMP with 2+ incoming lines
Hello, I know that I can have DID on a single line, but will AMP support 2+ lines with DID? Has anyone tried this? Straight forward? Thank you, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive ring on BT100
Hello, Is it possible to make BT100 phones ring in different ways based on where the call is coming from? The general idea is that I need the BT100 ring in 2 different ways depending on whether the call come from Zap1 or Zap2. It's because this system is for a receptionist answering two different phone lines for two separate companies and she needs to know how to greet the person on the other side ... one way that could be useful for her to recognize which line is ringing is by having a different ring tone for each. If BT100 cannot do it .. which phone can? Or is there some alternative way of helping the receptionist in this situation distinguish between the two lines? (Flash Operator Panel would not work well since she would not have it on all the time) Thanks, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP registration behind Linksys WRT54G - FIXED :-)
I finally figured it out ... working with BT100 you need to make a little voodoo ritual first :-) ... so follow the steps --exactly-- if you have trouble This is my working configuration behind Linksys WRT54G router: - Upgrade firmware 1.0.5.23 - Reset BT100 to factory defaults - SIP Server: asterisk.mydomain.com - Outgoing Proxy: asterisk.mydomain.com - DTMF: SIP INFO - Reboot BTW ... this is exactly what I tried 100x before but without the exact order of steps. I think especially step #2 about resetting to factory defaults before you do any re-configuration is critical. Don't trust the web interface always start fresh. Strangely, I had no problems whenever I was behind any other router than Linksys ... didn't have to do all this voodoo stuff ... makes me uncomfortable since I feel like I'll plug the phones in tomorrow and I'll be back where I started. Maybe the secret was not changing my underwear in the morning :-) LOL On the Asterisk side it's just the usual: Nat = yes Qualify = yes Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomas Florian Sent: Saturday, April 23, 2005 11:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SIP registration behind Linksys WRT54G - URI? I think I'm getting closer to figuring this out ... I just tried Linksys PAP2 and it registered just fine. I looked at the SIP packets captured by ethereal and I discovered that the real problem will probably be the uri in the authorization. For the working Linksys PAP2 and X-Lite I get: Authorization: DIGEST ... uri=sip:asterisk.mydomain.com ... For the BT100 which doesn't register (403 Forbidden) I get: Authorization: DIGEST ... uri=sip:wan-ip-of-the-router ... ... this kind of makes sense ... that looks like the wrong uri to send. So for some reason BT100 sends the wrong URI ... how can I fix this?? Again the weird thing is that if I plug in the BT100 behind any other router then Linksys WRT54G everything works fine. I'm trying my BT100 with the following config: - SIP Server: asterisk.mydomain.com - Outgoing Proxy: asterisk.mydomain.com - Nat travelsal: no - Local sip port: 5060 - Use NAT ip: no - Proxy require: no And in my sip.conf I have Nat=yes Qualify=yes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomas Florian Sent: Saturday, April 23, 2005 11:04 PM To: 'Pedro'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SIP registration behind Linksys WRT54G Yes that's the first thing I tried ... I'm able to make it work (using different routers than Linksys) in the following ways: - Set outgoing proxy and no STUN OR - No outgoing proxy and set STUN But once I put it behind Linksys everything registration does not work any more. Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pedro Sent: Saturday, April 23, 2005 10:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Have you tried to enable NAT translation on the Grandstream? On 4/23/05, Tomas Florian [EMAIL PROTECTED] wrote: I'm trying to register BT100s ... (doesn't work) X-Lite seems to work though Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo-Jojo Sent: Saturday, April 23, 2005 8:48 PM To: Luki; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Oh yeah, duh.. Forgot.. I also have an SPA-2000 and a Cisco ATA-186 running behind my Linksys WTR43GS with no issues. This is at home registering to an external * box and to vonage. - Original Message - From: Luki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 9:41 PM Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G The WRT54G work fine... I have a Sipura 1000 and a Grandstream 286, both nated through a WRT54G on a single public IP. Worked out of the box -- no special settings needed. I was even surprised that I did not need to turn on the NAT handling in the Sipura ATA. Then I have a WRT54G running as a wireless client, and a Sipura 1001 connected to it, essentially behind two NAT's. Works fine too. --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http
[Asterisk-Users] SIP registration behind Linksys WRT54G
Hello, I'm having some major problems getting SIP phones to register whenever I put them behind a Linksys router. The same phones will register behind any other NAT (I've tried 3 others without problems) I've been debugging using Ethereal and these are the differences that I found between Linksys WRT54G and a Monowall Router as an example (Monowall router is one of the many that work fine for me): REGISTER sip:asterisk.mydomain.com Monowall (good registration) - Via: SIP/2.0/UDP 192.168.10.199;branch=... - Authorization: DIGEST ..., uri=sip:asterisk.mydomain.com, ... - Contact sip: [EMAIL PROTECTED];user=phone Linksys WRT54G (Bad registration - 403 Forbidden) - Via: SIP/2.0/UDP 66.x.x.166;branch=... - Authorization: DIGEST ..., uri=sip 66.x.x.166:5060, ... - Contact * As you can see the difference seems to be that with the Linksys the SIP request has it's WAN IP + port (66.x.x.166) whereas the request from behind a monowall has the LAN IP of the phone What is the explanation for this difference? Needless to say - I don't have any special port forwarding enabled on either one of these routers and I'm using the identical phone with identical configuration for both tests. I have outgoing proxy in my phone's configuration but it almost looks like it's disregarding that option when behind the Linksys router. Another interesting thing to note is that I have tried connecting to some other proxy from behind Linksys (not my own asterisk but some other provider - I don't know what they are running) I was able to register without a problem. Interestingly, the registration request looked identical to the monowall one (Via: LocalIP , uri: FQDN ) ... unfortunately because I am not the system admin on that VoIP server I can't login to see what configuration they have in order to copy it. I'm really out of ideas ... if anyone has any hints of what else I could check out I would really appreciate that. Thank you, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP registration behind Linksys WRT54G
Is your problem on the same model of Linksys? WRT54G? I haven't had a chance to try some other Linksys routers so I'm curious. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Henderson Sent: Saturday, April 23, 2005 7:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Please make sure you post any solution you find to this issue to the list I have been frustrated by this as well. Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Tomas Florian wrote: Hello, I'm having some major problems getting SIP phones to register whenever I put them behind a Linksys router. The same phones will register behind any other NAT (I've tried 3 others without problems) I've been debugging using Ethereal and these are the differences that I found between Linksys WRT54G and a Monowall Router as an example (Monowall router is one of the many that work fine for me): REGISTER sip:asterisk.mydomain.com Monowall (good registration) - Via: SIP/2.0/UDP 192.168.10.199;branch=... - Authorization: DIGEST ..., uri=sip:asterisk.mydomain.com, ... - Contact sip: [EMAIL PROTECTED];user=phone Linksys WRT54G (Bad registration - 403 Forbidden) - Via: SIP/2.0/UDP 66.x.x.166;branch=... - Authorization: DIGEST ..., uri=sip 66.x.x.166:5060, ... - Contact * As you can see the difference seems to be that with the Linksys the SIP request has it's WAN IP + port (66.x.x.166) whereas the request from behind a monowall has the LAN IP of the phone What is the explanation for this difference? Needless to say - I don't have any special port forwarding enabled on either one of these routers and I'm using the identical phone with identical configuration for both tests. I have outgoing proxy in my phone's configuration but it almost looks like it's disregarding that option when behind the Linksys router. Another interesting thing to note is that I have tried connecting to some other proxy from behind Linksys (not my own asterisk but some other provider - I don't know what they are running) I was able to register without a problem. Interestingly, the registration request looked identical to the monowall one (Via: LocalIP , uri: FQDN ) ... unfortunately because I am not the system admin on that VoIP server I can't login to see what configuration they have in order to copy it. I'm really out of ideas ... if anyone has any hints of what else I could check out I would really appreciate that. Thank you, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP registration behind Linksys WRT54G
I'm trying to register BT100s ... (doesn't work) X-Lite seems to work though Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo-Jojo Sent: Saturday, April 23, 2005 8:48 PM To: Luki; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Oh yeah, duh.. Forgot.. I also have an SPA-2000 and a Cisco ATA-186 running behind my Linksys WTR43GS with no issues. This is at home registering to an external * box and to vonage. - Original Message - From: Luki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 9:41 PM Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G The WRT54G work fine... I have a Sipura 1000 and a Grandstream 286, both nated through a WRT54G on a single public IP. Worked out of the box -- no special settings needed. I was even surprised that I did not need to turn on the NAT handling in the Sipura ATA. Then I have a WRT54G running as a wireless client, and a Sipura 1001 connected to it, essentially behind two NAT's. Works fine too. --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP registration behind Linksys WRT54G
Yes that's the first thing I tried ... I'm able to make it work (using different routers than Linksys) in the following ways: - Set outgoing proxy and no STUN OR - No outgoing proxy and set STUN But once I put it behind Linksys everything registration does not work any more. Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pedro Sent: Saturday, April 23, 2005 10:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Have you tried to enable NAT translation on the Grandstream? On 4/23/05, Tomas Florian [EMAIL PROTECTED] wrote: I'm trying to register BT100s ... (doesn't work) X-Lite seems to work though Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo-Jojo Sent: Saturday, April 23, 2005 8:48 PM To: Luki; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Oh yeah, duh.. Forgot.. I also have an SPA-2000 and a Cisco ATA-186 running behind my Linksys WTR43GS with no issues. This is at home registering to an external * box and to vonage. - Original Message - From: Luki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 9:41 PM Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G The WRT54G work fine... I have a Sipura 1000 and a Grandstream 286, both nated through a WRT54G on a single public IP. Worked out of the box -- no special settings needed. I was even surprised that I did not need to turn on the NAT handling in the Sipura ATA. Then I have a WRT54G running as a wireless client, and a Sipura 1001 connected to it, essentially behind two NAT's. Works fine too. --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP registration behind Linksys WRT54G - URI?
I think I'm getting closer to figuring this out ... I just tried Linksys PAP2 and it registered just fine. I looked at the SIP packets captured by ethereal and I discovered that the real problem will probably be the uri in the authorization. For the working Linksys PAP2 and X-Lite I get: Authorization: DIGEST ... uri=sip:asterisk.mydomain.com ... For the BT100 which doesn't register (403 Forbidden) I get: Authorization: DIGEST ... uri=sip:wan-ip-of-the-router ... ... this kind of makes sense ... that looks like the wrong uri to send. So for some reason BT100 sends the wrong URI ... how can I fix this?? Again the weird thing is that if I plug in the BT100 behind any other router then Linksys WRT54G everything works fine. I'm trying my BT100 with the following config: - SIP Server: asterisk.mydomain.com - Outgoing Proxy: asterisk.mydomain.com - Nat travelsal: no - Local sip port: 5060 - Use NAT ip: no - Proxy require: no And in my sip.conf I have Nat=yes Qualify=yes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomas Florian Sent: Saturday, April 23, 2005 11:04 PM To: 'Pedro'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SIP registration behind Linksys WRT54G Yes that's the first thing I tried ... I'm able to make it work (using different routers than Linksys) in the following ways: - Set outgoing proxy and no STUN OR - No outgoing proxy and set STUN But once I put it behind Linksys everything registration does not work any more. Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pedro Sent: Saturday, April 23, 2005 10:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Have you tried to enable NAT translation on the Grandstream? On 4/23/05, Tomas Florian [EMAIL PROTECTED] wrote: I'm trying to register BT100s ... (doesn't work) X-Lite seems to work though Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo-Jojo Sent: Saturday, April 23, 2005 8:48 PM To: Luki; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Oh yeah, duh.. Forgot.. I also have an SPA-2000 and a Cisco ATA-186 running behind my Linksys WTR43GS with no issues. This is at home registering to an external * box and to vonage. - Original Message - From: Luki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 9:41 PM Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G The WRT54G work fine... I have a Sipura 1000 and a Grandstream 286, both nated through a WRT54G on a single public IP. Worked out of the box -- no special settings needed. I was even surprised that I did not need to turn on the NAT handling in the Sipura ATA. Then I have a WRT54G running as a wireless client, and a Sipura 1001 connected to it, essentially behind two NAT's. Works fine too. --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk acting as PBX + SIP Proxy ... possible?
Hello, I'm in the process of implementing the following setup External SIP phones at another location(s) (nat = yes) | | Analog phone line | | |-- |ext if 142.x.x.41 | |Asterisk | |int if 192.168.0.1 |-- | Internal SIP Phones (nat=no) Excuse my ASCII art ... if you cant see the diagram I'm basically doing the following: - There are some phones on the LAN, and some other phones on the internet side - Both sets of phones use Asterisk to make calls between each other as if they were all on LAN and to the phone line. Is something like this going to work reliably? Or will I need a second central server to act as a proxy. The reason I'm asking this is that I have been able to make this setup work but am having some strange registration issues whenever my external sip phones sit behind Linksys router (I get 403 forbidden) ... when I use some other router the stuff seems to work. But I'm worried about reliability since I read recently that Asterisk is not a proxy and I'm definitely using it as an outgoing proxy in this case. http://www.voip-info.org/wiki-Asterisk+SIP+not-proxy Has anyone successfully created this kind of setup before? (having Asterisk pass calls on both LAN and WAN side?) Do you have any hints for me to get this 403 forbidden error figured out? I think it might have something to do with FQDN - but the strange thing is that it happens only behind Linksys And if I do need an outgoing proxy which proxy do you recommend? Thank you, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Annoying SIP registration problem behind ?Linksys?
Hello, I'm trying to use some VoIP phones behind a Linksys WRT54G router but can't get them to register. The annoying thing is that I've taken the phones to 3 other locations with non-Linksys NAT routers and the phones work immediately without any problems. I've tried STUN, outgoing proxy . everything works immediately and reliably whenever I put the phone at the location that doesn't have the Linksys router there , but as soon as I bring it over it refuses to register through SIP. All the locations that I have tried have very similar setups LAN 192.168.x.x addresses . The only other difference is that the location with the Linksys has static WAN IP from a datacenter whereas the other locations have their WAN side from a regular DSL. The most annoying thing is that I can see that phones provisioned for a SIP proxy that is not managed by me and I don't have access to the logs or config work immediately at the location with Linksys . I even tried setting everything up the same as they have on the phone and just changing the IP of the proxy and it failed again . of course once I brought the phone to some other location it worked immediately like always. . so its not like the Linksys is not capable of doing it .. it just doesn't want to work with my asterisk setup. Am I missing some trick to get Linksys to cooperate with my asterisk setup? [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 142.x.x.91;(I also had 0.0.0.0 - Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown [204] username=204 type=friend secret=*** qualify=1000 port=5060 pickupgroup= nat=yes ;(I also had no) mailbox= host=dynamic dtmfmode=info disallow= context=from-internal canreinvite=no callgroup= callerid=Some user 204 allow= On the phone I'm using BT100 .. and I just have outgoing SIP proxy set and no STUN no NAT IP ... it works at all locations but not Linksys (same results when I tried doing STUN, NAT IP etc ... no luck on Linksys but worked everywhere else) I know that the best way to isolate this problem is to switch the Linksys router to some that I have seen working (or borrow their Linksys and try to see if it works at some other location) . I'll do that but the trouble is I didn't have a chance yet because the client sees VoIP phones registering with another proxy without problems and is skeptical that the router has anything to do with it . I'm skeptical also but I'm just out of ideas on what else could be the problem . Another thing that I'm not sure about is how it works with the SIP registry on asterisk . is it possible that it's somehow remembering the locations / configurations or that the entries for that location were corrupted somehow??? Any ideas anyone? Thank you very much, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BT100 wrong NAT detection
Hello, I'm having trouble getting BT100 to identify NAT type reliably for Asterisk. My setup is as follows: - Asterisk is on the open internet 142.x.x.41 - BT100 phones are behind NATs - I use STUN for my BT100 : 142.x.x.41 (same server as Asterisk) - BT100 firmware (tried .16,.18,.23 same results) When I go to the BT100 setup page I can see the following: - detected NAT type is symmetric NAT OR (sometimes) - detected NAT type (blank) Both of these are wrong as my NAT type should be: Port restricted NAT ... if I'm lucky sometimes BT100 comes back with port restricted answer and in that case I'm ready to go .. but it rarely works after a reboot ... sometimes yes sometimes no .. I tested the STUN server and my actual NAT type by running the WinSTUN ... it always answers correctly 100% of the time. I also tried setting the BT100 STUN server to some public STUN servers .. no luck. ... so why is BT100 so unreliable??? I even did ./sever -v to watch my STUN server in action and it does actually talk to the BT100s on every phone reboot .. but the weird thing is that between BT100 and STUN there are only 3 messages sent whereas between XLite and STUN or WinStunClient and STUN server there are about 8+ ... it's almost as though BT100 gives up .. is BT100 compatible only with certain STUN servers? Is there some trick to this? What else can watch to troubleshoot this situation? Thank you, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BT100 wrong NAT detection
This answers a lot of questions - I am in fact using Vovida STUN (so I have to find a replacement) - I don't have 2IPs on the Asterisk server - so that's wrong too Thanks for your help!! :-) Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Vishnev Sent: Saturday, April 16, 2005 12:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] BT100 wrong NAT detection Tomas, Yes, BT100 is a little picky on the use of Stun Servers. For example, it will not work at all with Vovida Stun server. Also, Stun negotiation takes some time. So if you rebooted the phone, I would suggest waiting 15-30 seconds until phone syncs up with Stun server and requests binding. You can also run ethereal on your LAN and monitor the packets coming from Bt100. Then you can compare them to Xlite or other phones to see how they differ. I would also suggest contacting grandstream and getting the latest firmware for granstream. Another thing that made we wonder is when you said you are running Stun on the same system as asterisk. Normally Stun requires 2 systems, system with 2 NIC cards, or at minimum 2 IP addresses that stun can bind to. Is that what you are doing? Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomas Florian Sent: Saturday, April 16, 2005 1:17 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] BT100 wrong NAT detection Hello, I'm having trouble getting BT100 to identify NAT type reliably for Asterisk. My setup is as follows: - Asterisk is on the open internet 142.x.x.41 - BT100 phones are behind NATs - I use STUN for my BT100 : 142.x.x.41 (same server as Asterisk) - BT100 firmware (tried .16,.18,.23 same results) When I go to the BT100 setup page I can see the following: - detected NAT type is symmetric NAT OR (sometimes) - detected NAT type (blank) Both of these are wrong as my NAT type should be: Port restricted NAT ... if I'm lucky sometimes BT100 comes back with port restricted answer and in that case I'm ready to go .. but it rarely works after a reboot ... sometimes yes sometimes no .. I tested the STUN server and my actual NAT type by running the WinSTUN ... it always answers correctly 100% of the time. I also tried setting the BT100 STUN server to some public STUN servers .. no luck. ... so why is BT100 so unreliable??? I even did ./sever -v to watch my STUN server in action and it does actually talk to the BT100s on every phone reboot .. but the weird thing is that between BT100 and STUN there are only 3 messages sent whereas between XLite and STUN or WinStunClient and STUN server there are about 8+ ... it's almost as though BT100 gives up .. is BT100 compatible only with certain STUN servers? Is there some trick to this? What else can watch to troubleshoot this situation? Thank you, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BT100 wrong NAT detection
One more question ... I did a search on Google for STUN servers and didn't find any other open source server other than Vovida's What other open source Stun servers are there? And if there are none, what commercial one have you found to work well with BT100? Thanks again, Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Vishnev Sent: Saturday, April 16, 2005 12:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] BT100 wrong NAT detection Tomas, Yes, BT100 is a little picky on the use of Stun Servers. For example, it will not work at all with Vovida Stun server. Also, Stun negotiation takes some time. So if you rebooted the phone, I would suggest waiting 15-30 seconds until phone syncs up with Stun server and requests binding. You can also run ethereal on your LAN and monitor the packets coming from Bt100. Then you can compare them to Xlite or other phones to see how they differ. I would also suggest contacting grandstream and getting the latest firmware for granstream. Another thing that made we wonder is when you said you are running Stun on the same system as asterisk. Normally Stun requires 2 systems, system with 2 NIC cards, or at minimum 2 IP addresses that stun can bind to. Is that what you are doing? Alex -Original Message- rom: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomas Florian Sent: Saturday, April 16, 2005 1:17 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] BT100 wrong NAT detection Hello, I'm having trouble getting BT100 to identify NAT type reliably for Asterisk. My setup is as follows: - Asterisk is on the open internet 142.x.x.41 - BT100 phones are behind NATs - I use STUN for my BT100 : 142.x.x.41 (same server as Asterisk) - BT100 firmware (tried .16,.18,.23 same results) When I go to the BT100 setup page I can see the following: - detected NAT type is symmetric NAT OR (sometimes) - detected NAT type (blank) Both of these are wrong as my NAT type should be: Port restricted NAT ... if I'm lucky sometimes BT100 comes back with port restricted answer and in that case I'm ready to go .. but it rarely works after a reboot ... sometimes yes sometimes no .. I tested the STUN server and my actual NAT type by running the WinSTUN ... it always answers correctly 100% of the time. I also tried setting the BT100 STUN server to some public STUN servers .. no luck. ... so why is BT100 so unreliable??? I even did ./sever -v to watch my STUN server in action and it does actually talk to the BT100s on every phone reboot .. but the weird thing is that between BT100 and STUN there are only 3 messages sent whereas between XLite and STUN or WinStunClient and STUN server there are about 8+ ... it's almost as though BT100 gives up .. is BT100 compatible only with certain STUN servers? Is there some trick to this? What else can watch to troubleshoot this situation? Thank you, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Strange intermittent NAT problem with BT100s
Hello, I have a strange problem whenever I have 2 or more BT100s behind NAT. I am not able to reproduce this error reliably, but it happens every 2-5 minutes. The general setup is that there is Asterisk server sitting at a central location. Some peers connect directly (206,205,201) but some (204,203,200) connect through NAT. This all works fine ...but it is extremely unreliable ... I get UNREACHABLE and then OK again ... UNREACHABLE and OK again .. unpredictably. When it's OK I can make phone calls no problem of course when it goes UNREACHABLE there is trouble. I tried to replace one of the BT100 phones with X-Lite and that one is OK (~40ms) rock solid - or seems to be so far. So it seems that there is something weird going on with BT100 My configuration of BT100 is as follows: - firmware 1.0.5.23 (I've noticed similar problems with .16 also) - detected NAT type is symmetric NAT - STUN stun.xten.net (I'm using Xtens ... or do I have to use my own???) - no outbound proxy - register expiration = 1 - keep alive interval = 20 sec (I also tried as low as 1 sec) My sip configuration uses: - nat = yes - qualify = yes (I also tried longer qualify 1 with no luck) This is what I get with sip show peers ... the 204 and 200 are BT100 and sometimes one or both go UNREACHABLE for a while ... 203 is X-Lite and didn't go UNREACHABLE yet. 206/206 (Unspecified)D 255.255.255.255 0 Unmonitored 205/205 (Unspecified)D 255.255.255.255 0 Unmonitored 204/204 209.x.x.125 D N 255.255.255.255 38340OK (40 ms) 203/203 209.x.x.125 D N 255.255.255.255 1548 OK (38 ms) 201/201 192.168.2.112D 255.255.255.255 5060 Unmonitored 200/200 209.x.x.125 D N 255.255.255.255 37838 UNREACHABLE Any ideas? Is there some trick to get BT100 to cooperate? Thanks, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk not recognizing key beeps
Hello, So far everything that I'm trying with asterisk is working except for this weird thing. When I try to call voicemail and it asks me for the password I enter it in but from the debug message I can see that it thinks I didn't enter anything in. Also when I'm leaving a message it sais press pound to end, but even if I press it 10 times it keeps on recording until I hang up. It just doesn't seem to recognize my key presses. I can dial, talk and do everything else ... but I just can't press keys during the call. I'm using a very simple setup from some quickstart with SIP and voicemail - nothing more than that. I remember that this used to work for me but then it stopped. I have no idea why, I couldn't find anything on the net about this problem. Any ideas? Thanks, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk not recognizing key beeps - DTMF on BT100
Thanks, this is what I found out so far: I have a Grandstream BT100, that is capable of doing both out of band and in band DTMF. But it doesn't work with either setting (I changed my sip.conf and the BT100 client phone accordingly of course) X-Lite works fine. I also upgraded the BT100 to have the newest firmware but that didn't help either. Are there some issues with BT100 phones and DTMF? Can I turn on DTMF debugging in asterisk somehow? Thanks, Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yair Hakak Sent: Wednesday, January 19, 2005 2:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk not recognizing key beeps what endpoints are you using? You probably have a DTMF type mismatch between asterisk and your endpoint (IP phone or softphone) -yair On Wed, 19 Jan 2005 01:49:46 -0700, Tomas Florian [EMAIL PROTECTED] wrote: Hello, So far everything that I'm trying with asterisk is working except for this weird thing. When I try to call voicemail and it asks me for the password I enter it in but from the debug message I can see that it thinks I didn't enter anything in. Also when I'm leaving a message it sais press pound to end, but even if I press it 10 times it keeps on recording until I hang up. It just doesn't seem to recognize my key presses. I can dial, talk and do everything else ... but I just can't press keys during the call. I'm using a very simple setup from some quickstart with SIP and voicemail - nothing more than that. I remember that this used to work for me but then it stopped. I have no idea why, I couldn't find anything on the net about this problem. Any ideas? Thanks, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk not recognizing key beeps - DTMF onBT100
That is exactly the first thing I did, didn't work :-| -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Illmayer Sent: Wednesday, January 19, 2005 7:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk not recognizing key beeps - DTMF onBT100 Set the grandstream to RFC2883 in your phone, this will work with asterisk. also define DTMFMODE=RFC2883 in sip.conf under the phone definition. Pete -- Open WebMail Project (http://openwebmail.org) -- Original Message --- From: Tomas Florian [EMAIL PROTECTED] To: [EMAIL PROTECTED], 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wed, 19 Jan 2005 15:27:44 -0700 Subject: RE: [Asterisk-Users] Asterisk not recognizing key beeps - DTMF on BT100 Thanks, this is what I found out so far: I have a Grandstream BT100, that is capable of doing both out of band and in band DTMF. But it doesn't work with either setting (I changed my sip.conf and the BT100 client phone accordingly of course) X-Lite works fine. I also upgraded the BT100 to have the newest firmware but that didn't help either. Are there some issues with BT100 phones and DTMF? Can I turn on DTMF debugging in asterisk somehow? Thanks, Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yair Hakak Sent: Wednesday, January 19, 2005 2:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk not recognizing key beeps what endpoints are you using? You probably have a DTMF type mismatch between asterisk and your endpoint (IP phone or softphone) -yair On Wed, 19 Jan 2005 01:49:46 -0700, Tomas Florian [EMAIL PROTECTED] wrote: Hello, So far everything that I'm trying with asterisk is working except for this weird thing. When I try to call voicemail and it asks me for the password I enter it in but from the debug message I can see that it thinks I didn't enter anything in. Also when I'm leaving a message it sais press pound to end, but even if I press it 10 times it keeps on recording until I hang up. It just doesn't seem to recognize my key presses. I can dial, talk and do everything else ... but I just can't press keys during the call. I'm using a very simple setup from some quickstart with SIP and voicemail - nothing more than that. I remember that this used to work for me but then it stopped. I have no idea why, I couldn't find anything on the net about this problem. Any ideas? Thanks, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- End of Original Message --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk not recognizing key beeps - DTMF onBT100
By the way ... this is my sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) allow=all ; Allow all codecs context = bogon-calls ; Send SIP callers that we don't know about here dtmfmode=rfc2883; I added this here too just in case [2000] type=friend ; This device takes and makes calls username=2000 ; Username on device secret=xx ; Password for device host=dynamic ; This host is not on the same IP addr every time context=from-sip ; Inbound calls from this host go here mailbox=100 ; Activate the message waiting light if this canreinvite=no dtmfmode=rfc2883 allow=ulaw -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomas Florian Sent: Wednesday, January 19, 2005 9:33 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk not recognizing key beeps - DTMF onBT100 That is exactly the first thing I did, didn't work :-| -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Illmayer Sent: Wednesday, January 19, 2005 7:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk not recognizing key beeps - DTMF onBT100 Set the grandstream to RFC2883 in your phone, this will work with asterisk. also define DTMFMODE=RFC2883 in sip.conf under the phone definition. Pete -- Open WebMail Project (http://openwebmail.org) -- Original Message --- From: Tomas Florian [EMAIL PROTECTED] To: [EMAIL PROTECTED], 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wed, 19 Jan 2005 15:27:44 -0700 Subject: RE: [Asterisk-Users] Asterisk not recognizing key beeps - DTMF on BT100 Thanks, this is what I found out so far: I have a Grandstream BT100, that is capable of doing both out of band and in band DTMF. But it doesn't work with either setting (I changed my sip.conf and the BT100 client phone accordingly of course) X-Lite works fine. I also upgraded the BT100 to have the newest firmware but that didn't help either. Are there some issues with BT100 phones and DTMF? Can I turn on DTMF debugging in asterisk somehow? Thanks, Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yair Hakak Sent: Wednesday, January 19, 2005 2:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk not recognizing key beeps what endpoints are you using? You probably have a DTMF type mismatch between asterisk and your endpoint (IP phone or softphone) -yair On Wed, 19 Jan 2005 01:49:46 -0700, Tomas Florian [EMAIL PROTECTED] wrote: Hello, So far everything that I'm trying with asterisk is working except for this weird thing. When I try to call voicemail and it asks me for the password I enter it in but from the debug message I can see that it thinks I didn't enter anything in. Also when I'm leaving a message it sais press pound to end, but even if I press it 10 times it keeps on recording until I hang up. It just doesn't seem to recognize my key presses. I can dial, talk and do everything else ... but I just can't press keys during the call. I'm using a very simple setup from some quickstart with SIP and voicemail - nothing more than that. I remember that this used to work for me but then it stopped. I have no idea why, I couldn't find anything on the net about this problem. Any ideas? Thanks, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- End of Original Message --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
