Re: [Asterisk-Users] Grandstream Budgetone SIP registration fails
On Mon, 2004-03-08 at 18:50, Olle E. Johansson wrote: If you configure a static address, the PBX already know how to reach the client and no registration is therefore needed (and not allowed in asterisk). Enabling registration makes the SIP device mobile across the network. Configuring a static address locks it in. There is one more thing I do not understand : Asterisk logs messages such as Peer '6040' isn't dynamic; how does Asterisk determine wether the SIP client's IP address is statically or dynamically assigned ? Is that information contained in the message that the SIP client sends to initiate registration ? Does Asterisk do a DHCP request ? Is there another mechanism I have not imagined ? signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] ISDN BRI VoIP Internet
On Tue, 2004-03-09 at 00:52, David Uzzell wrote: Would the FRITZ! card be best suited to this or would something else be better suited? This pages contains the information you want : http://lists.digium.com/pipermail/asterisk-users/2003-March/008761.html http://www.telappliant.net/site2/basic_rate_cards.htm http://lists.digium.com/pipermail/asterisk-users/2003-November/028797.html signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Grandstream Budgetone SIP registration fails
On Tue, 2004-03-09 at 16:58, Olle E. Johansson wrote: Jean-Marc V. Liotier wrote: On Mon, 2004-03-08 at 18:50, Olle E. Johansson wrote: If you configure a static address, the PBX already know how to reach the client and no registration is therefore needed (and not allowed in asterisk). Enabling registration makes the SIP device mobile across the network. Configuring a static address locks it in. Asterisk logs messages such as Peer '6040' isn't dynamic; how does Asterisk determine wether the SIP client's IP address is statically or dynamically assigned ? The message Peer '6040' isn't dynamic is there to try to tell you that a peer you configured with a static IP is trying to register, and Asterisk will not let it register, since it's a static configuration. It's the host= field in sip.conf that determines if a peer is dynamic or static. Ok, so the default option is that the client is assumed to be statically addressed, and registration is therefore rejected because it would be redundant. In order to allow SIP registration with a login and a password, host=dynamic must be set. Please correct me if I'm wrong... Thanks for your answers. signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Grandstream Budgetone SIP registration fails
On Sun, 2004-03-07 at 13:03, Philipp von Klitzing answered off-list: [6040] defaultip=192.168.1.40 Replace this with host=dynamic and see what happens. That's it ! Thinking it was going to make things easier to diagnose, I had chosen to set the phones with a static IP. Apparently, wether defaultip is set or not, when the phone uses a static IP address, some assumptions are made about authentication. I do not understand what they are and which side assumes something, but it prevents SIP registration. Setting host=dynamic even though I use static IP addresses solved the problem. Many hours went into finding that out. If somebody has a clue about how the a static IP address influences the SIP registration process, I would be happy to know about it. reinvite=no That paramter does not exist - use canreinvite= instead of reinvite= This is a common error, and there are even examples on the web that include this faulty setup option. I also saw it in several examples. dtmfmode=rfc2833 For voicemail I had to switch to =info, but if =rfc works fine for you then stick with it. Thanks for all that advice, and thanks to others who contributed to this thread. Now that I have completed my first Grandstream to Grandstream call via the Asterisk server I feel much better and I can't wait for the QuadBRI to arrive so I can really begin to make useful stuff ! Here is my current working configuration : - In sip.conf : [6040] username=6040 secret=mysecret host=dynamic type=friend canreinvite=no dtmfmode=info disallow=all allow=ulaw allow=alaw - On the phone side : SIP User ID: 6040 Authenticate ID: 6040 Authenticate Password: mysecret SIP User ID is phone number: Yes (everything else is default) Nothing special about that configuration, so I guess it's not worthy of a wiki addition. What would be and should really should be looked into is the failure of SIP registration when a static IP is set without host=dynamic. signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Grandstream Budgetone SIP registration fails
On Mon, 2004-03-08 at 18:50, Olle E. Johansson wrote: Jean-Marc V. Liotier wrote: On Sun, 2004-03-07 at 13:03, Philipp von Klitzing answered off-list: [6040] defaultip=192.168.1.40 Replace this with host=dynamic and see what happens. That's it ! Thinking it was going to make things easier to diagnose, I had chosen to set the phones with a static IP. Apparently, wether defaultip is set or not, when the phone uses a static IP address, some assumptions are made about authentication. I do not understand what they are and which side assumes something, but it prevents SIP registration. Setting host=dynamic even though I use static IP addresses solved the problem. Many hours went into finding that out. If somebody has a clue about how the a static IP address influences the SIP registration process, I would be happy to know about it. Registration is simply a way for a SIP device to tell the PBX I'm here, place any calls to this IP address. To be able to do so, most configurations require the client to authenticate. If you configure a static address, the PBX already know how to reach the client and no registration is therefore needed (and not allowed in asterisk). Enabling registration makes the SIP device mobile across the network. Configuring a static address locks it in. Thanks for enlightening me. I'll put that in the wiki if it is not already there. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream Budgetone SIP registration fails
This is a very basic problem, and I feel stupid to resort to the list to solve it, but after three hours pulling my hair trying all combinations of a handful of parameters and getting nowhere I fail to see the path leading to a solution. I just got a pair of Budgetones. I have played a little with Asterisk before, for example using Gnophone to call the talking clock, to leave voicemail and receive it by email or to call the Digium IVR - basic stuff but this is hust to point out that I am not completely lost with Asterisk. Although I am new to the SIP part I have probably read all that there is to read about configuring a SIP phone with Asterisk : it seems like a very simple process and it makes not succeeding even more frustrating... Whatever solution I find I will add to the wiki ! Here is the Asterisk console output of what the phone initially sends when it attempts SIP registration : Sip read: REGISTER sip:192.168.1.30 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.40;branch=z9hG4bK4f3dc531eb4f14dc From: sip:[EMAIL PROTECTED];tag=eb4f14dc11187288 To: sip:[EMAIL PROTECTED] Contact: * Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER Expires: 0 User-Agent: Grandstream SIP UA 1.0.4.23 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE Content-Length: 0 Here is the relevant section of sip.conf : [6040] username=6040 secret=mysecret type=friend The phone is on a static IP address. I have tried various possibly useful additions I picked along my readings such as : defaultip=192.168.1.40 auth=md5 reinvite=no nat=no qualify=1000 dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw But it makes no difference to my authentication problem, so I stuck to the most basic possible set of parameters. Now the worst part is that at some point I got a working setup, but I changed something afterward to try to make it better, but I neglected taking note of the working setup and I broke it again. I never found the working setup again... Next time I'll make a backup copy of the working setup before trying to ameliorate it... On a successful attempt, here is the Asterisk console output of what the phone initially sends when it attempts SIP registration : Sip read: REGISTER sip:192.168.1.30 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.40;branch=z9hG4bK43770e17dbdb1738 From: sip:[EMAIL PROTECTED];tag=1c6ad3084cb9ac64 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Proxy-Authorization: DIGEST username=poste40, realm=asterisk, algorithm=MD5, uri=sip:192.168.1.30, nonce=4c7355bd, response=7c3304ec9ffa7069de64ed17ef72f14d Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER Expires: 3600 User-Agent: Grandstream SIP UA 1.0.4.23 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE Content-Length: 0 It is different from what the phone sent before a failed attempt. The main difference is that the 'Contact: *' line of the failed attempt is replaced by : Contact: sip:[EMAIL PROTECTED] Proxy-Authorization: DIGEST username=poste40, realm=asterisk, algorithm=MD5, uri=sip:192.168.1.30, nonce=4c7355bd, respo Since this is the initial packet in the SIP session and it is emitted by the client, I infer that the solution of my problem certainly lies in the configuration of the phone. Admin password, IP configuration and SIP server address being proved correct since I access the phone's administrative interface and the phones reaches the , the only remaining parameters that have been changed are : SIP User ID: 6040 Authenticate ID: 6040 Authenticate Password: mysecret Now what ? I tried countless combinations of those parameters with the basic configuration on the server, but I can't find the working one. The answer is probably very simple and very obvious... Someone on the list certainly has a working setup with Asterisk and Grandstream Budgetone phones, I would be grateful if their SIP configuration was posted to the list. Quite unexpectedly I found no complete example of such working setup on the Web, maybe because it was so simple that no one thought that posting it would be useful to anyone. One I get mine working I shall post the parameters ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] how to disable zap debug!!!
On Fri, 2004-03-05 at 12:18, atif wrote: how to disable this DEBUG information... I would have intuitively said 'zap no debug' but apparently the 'no debug' is not implemented for zap although it exist for sip, iax, h323, skinny and mgcp. Should we consider this absence as a bug worthy of a wishlist item ? signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Connecting an ISDN (1 BRI) DECT multi phone base. NO ONE ???
On Fri, 2004-02-27 at 11:28, Frederic Olivie wrote: I own a Siemens 3070 DECT system. It's a simple DECT base which allows the connection of a few DECT phones. It's a very basic PBX. It's connected to the public network using an ISDN bri (2B + D) plug. According to the doc, it can also be connected to a PBX. Is there a way to connect this to Asterisk ? [..] Is there an ISDN hardware card that would handle such a connection ? In order to connect an user device, you need an ISDN adapter that supports NT mode. From http://www.isdn4linux.de/faq/i4lfaq-29.html : When multiple devices are connected to the ISDN connection, then all user device behave as slaves, where the network terminator (NT) behaves as master and synchronizes the communication on the S0 bus. The special behavior of the network terminator is called NT mode. User devices are normally not capable of running in NT mode. As a result, user devices can not communicate with each other even when they are connected via a crossed cable. Only some special ISDN cards (HFC chipset) are capable of running in NT mode, and can directly communicate with other ISDN user devices via a crossed cable. The QuadBRI from Junghanns does it and I'm about to get one, both to connect to the public ISDN and to connect ISDN DECT base stations : http://www.junghanns.net/asterisk/page17.html The Eicon Diva server cards do it too and they seem to be an industry reference, but they are twice as expensive as the Junghanns QuadBRI for about the same functions. As soon as I get my QuadBRI, I'll report my experience with it. signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Simple Front Gate Intercom
On Thu, 2004-02-26 at 19:43, Greg Kedrovsky wrote: On Thu, Feb 26, 2004 at 09:12:08AM -0800, TC wrote: I have these hooked to a fxo port FXO? I've been thinking fxs. ?? I have a 4-port fxs card, and one port goes to the gate. Am I (still?) confused about fxo/fxs?? Probably not. Terminals usually connect to an FXS port, but I have seen door bell intercoms that actually connect to the PSTN, hence the FXO port. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Analogical FXO vs. BRI dialing speed
When dialing out, will a call be established significantly faster by an ISDN adapter such as an Eicon Diva server compared to an analogical FXO such as Digium's X100P ? signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] ISDN BRI card
On Thu, 2003-10-16 at 14:35, Tomica Crnek wrote: Anyone knows of a good ISDN BRI card to use with Asterisk? Take a look at the list archives, there has been much discussion on this subject in the recent past. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: ISDN BRI active adapters with NT mode - any alternatives ?
On Tue, 2003-09-16 at 22:22, Klaus-Peter Junghanns wrote: Am Die, 2003-09-16 um 18.05 schrieb Louis-David Mitterrand: Now I have no idea if * supports plugging ISDN phones in the Diva. AFAIK it's not supported by chan_capi, but that may change. Yes, that may change. I will check with Eicon headquarters what the NT mode support in the BRI cards is about. Please keep us posted : I'm about to buy a Diva Server 4BRI and connecting ISDN phones (actually DECT base stations) is a critical capability; my project is a no go without it. And if somebody out there feels like sponsoring an Eicon BRI, so i can add support for it to chan_capi i wouldnt mind taking it ;-) (a place in the capi hall of fame will be yours) ;-) I had a hard time convincing management to fund the purchase of just one card for an experiment with Asterisk, so I guess I'm not going to be the one to sponsor one. But the development server is going to get a public IP address and an excellent symmetrical DSL connectivity, so if you want to play with that card I will gladly open an account with root privileges for you on that machine and do the local testing that you need. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN BRI active adapters with NT mode - any alternatives ?
On Mon, 2003-09-15 at 11:52, Klaus-Peter Junghanns wrote: i dont think that the Eicon Diva Server 4BRI's NT mode feature will work with linux/capi. I think the feature in the driver is for their PRI cards (where everything is always P2P). i may be wrong, though. I just had a chat with Eicon's French representatives and they confirm that the Eicon Diva Server 4BRI supports NT mode with Eicon's binary drivers that Eicon supports for Red Hat and Suse. They have no idea what the capabilities of the standard kernel drivers are : they do not support software libre. I took a look at the source and there may be something relevant at line 341 of drivers/isdn/eicon/bri.c but actually understanding what this code is supposed to do is far beyond my technical abilities. It is quite strange that Eicon's representatives have no clue about the standard kernel driver's capabilities : drivers/isdn/eicon/bri.c is copyright Eicon. But maybe the people who actually know are at Eicon's headquarters, not in France. Anyone here knows who the relevant contacts may be ? We are working on an alternative, a passive multiport ISDN card that supports TE and NT mode with zaptel drivers for asterisk. Interesting. Got any time to market estimates ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: ISDN BRI active adapters with NT mode - any alternatives ?
On Tue, 2003-09-16 at 18:05, Louis-David Mitterrand wrote: I am using the Diva 4BRI daily with our * and indeed it does support NT mode on a port by port basis Good news : I was not 100% sure about that. is with the open-source Melware drivers from http://mmm.melware.de. Very nice. I did not know about that one. Is that an isolated branch of the Diva drivers or do the modifications normally end up merged in the standard kernel ? Now I have no idea if * supports plugging ISDN phones in the Diva. Isn't it what NT mode is all about ? I thought it was merely an issue between the two ISDN devices with * having nothing to do with that. Now I'm troubled... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN BRI active adapters with NT mode - any alternatives ?
Having ISDN FXOs and a bunch of ISDN DECT base stations, I went looking for a CAPI compatible active BRI adapter capable of NT mode. I have found that the 'Eicon Diva Server 4BRI' fulfills theses requirements. However, at EUR 1387 for four BRI interfaces, it does not come cheap. Are there other users of similar hardware that can point me to alternatives ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN BRI active adapters with NT mode - any alternatives ?
On Mon, 2003-09-15 at 15:02, Matthew Enger wrote: Not sure if it will do what you want, but have you looked at the AVM C2 or C4 ISDN cards (www.avm.de)? They work under capi on linux from what I have read I read the spec sheets a few days ago and I did find no mention of NT mode, so AWM's cards do not fit my needs unless NT mode is present but not mentioned in the documentation, which I believe is extremely unlikely. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail notification email with no attachment despite attach=yes
On Wed, 2003-09-10 at 19:25, Tilghman Lesher wrote: in voicemail.conf : 1234 = 4242,Test mailbox,[EMAIL PROTECTED] 6004 = 4242;Other test mailbox,[EMAIL PROTECTED] It's probably the semicolon (;) in the second line, instead of a comma (,). Thanks to Tilghman, Troy and Paul for that answer and sorry for asking the list for something so trivial. I wonder how I missed something that obvious. I guess I spent too long staring at the files : maybe I would have noticed after having taken a pause... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail notification email with no attachment despite attach=yes
The demo 1235 extension that Asterisk ships with works fine and the messages are sent to the address I set in voicemail.conf. I guess that means that my configuration is working perfectly so far. But when I set up another extension with a voicemailbox, no mail is sent when a message is left, although I can dial voicemail and listen to the message just fine which I guess rules out voicemailbox misconfiguration. The strange thing is that both extensions and mailboxes are configured exactly the same : in extensions.conf : exten = 1235,1,Voicemail(u1234); Right to voicemail exten = 6004,1,Voicemail(u6004) in voicemail.conf : 1234 = 4242,Test mailbox,[EMAIL PROTECTED] 6004 = 4242;Other test mailbox,[EMAIL PROTECTED] I don't understand why these two seemingly identical configuration yield different results. I guess that I must have missed something that was included in the example and not in my new mailbox. Could somebody give me a hint about what it could be ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
