I have 2 Snom 200 and would like to get them to work properly with
Asterisk. With the Firmware 2.02t I am able to use the phone. But only
one line configured. With there newer firmware 2.03o it will not allow
me to make calls. But I can get calls on the unit. Again the 2nd line
is not able to be registered. Is this an issue with Asterisk or Snom?
I could use some example configuration files. I have followed the Snom
FAQ step by step. But it's still not working.
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\____\_ Ariel Batista
/ /
/ Red-Fone Communications, Inc.
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abatista@red-fone.com
Ph: 786-544-1114
Fx: 305-574-0212
I am using 2.03o with a Snom200 with two lines registered without a problem: However, I am not using any authorization, whih could be where your problem is. from my sip.conf: [5117] type=friend host=dynamic context=sip-gb mailbox=5117 callerid="Name <5117>" On Thursday, 22 January, 2004 16:04, Ariel Batista wrote:> I have 2 Snom 200 and would like to get them to work properly with > Asterisk. With the Firmware 2.02t I am able to use the phone. But only > one line configured. With there newer firmware 2.03o it will not allow > me to make calls. But I can get calls on the unit. Again the 2nd line > is not able to be registered. Is this an issue with Asterisk or Snom? > > I could use some example configuration files. I have followed the Snom > FAQ step by step. But it's still not working. > ----------------------------------------------------------------- > \ > \____\_ Ariel Batista > / / > / Red-Fone Communications, Inc. > ------------------------------------------------------------------ > abatista@red-fone.com > Ph: 786-544-1114 > Fx: 305-574-0212 > > _______________________________________________ > 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
Ariel Batista wrote:>I have 2 Snom 200 and would like to get them to work properly with >Asterisk. With the Firmware 2.02t I am able to use the phone. But only >one line configured. With there newer firmware 2.03o it will not allow >me to make calls. But I can get calls on the unit. Again the 2nd line >is not able to be registered. Is this an issue with Asterisk or Snom? > >I could use some example configuration files. I have followed the Snom >FAQ step by step. But it's still not working. > >I had the same problem, so I emailed SNOM. After a quick and clear reaction from SNOM, the following turns out: I have an SRV record set for Asterisk using both TCP and UDP, because I was first experimenting with SER and that SIP proxy DOES support TCP. Asterisk does not. So in firmware 2.02t the phone tried udp automatically, and in firmware 2.03o the phone tried to use tcp, which will not work in Asterisk. Removing the TCP SRV entry solved my problem Maybe this will solve your problem. However, I still have other problems with the SNOM phones: - All sound stops working sometimes (also ringtone) - Speech sometimes not working (not sure if it is RTP problem or SNOM firmware problem) - Sometimes the phone returns BUSY when not busy in firmware 2.02t. Resetting the phone or adding a new SIP line solves this. In firmware 2.03o, this is different. The phone does not respond and Asterisk gives the error: phone CIRCUIT BUSY. I have seen other posts here from people having the same problems. Other people having the same problems? It makes my case a bit more clear at SNOM when I can point them to a thread with a lot of people having the same problems:-) I am emailing with SNOM about these issues. Kind regards, Geert Nijpels
Maybe you have to update Asterisk (see http://bugs.digium.com/bug_view_page.php?bug_id=0000732). snom is now a little bit picky about line-ID (see the discussion in the bug). CS> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Ariel Batista > Sent: Thursday, January 22, 2004 10:04 PM > To: Asterisk User List > Subject: [Asterisk-Users] Snom 200 phones not working. > > I have 2 Snom 200 and would like to get them to work properly with > Asterisk. With the Firmware 2.02t I am able to use the phone. But only > one line configured. With there newer firmware 2.03o it will not allow > me to make calls. But I can get calls on the unit. Again the 2nd line > is not able to be registered. Is this an issue with Asterisk or Snom? > > I could use some example configuration files. I have followed the Snom > FAQ step by step. But it's still not working. > ----------------------------------------------------------------- > \ > \____\_ Ariel Batista > / / > / Red-Fone Communications, Inc. > ------------------------------------------------------------------ > abatista@red-fone.com > Ph: 786-544-1114 > Fx: 305-574-0212 > > _______________________________________________ > 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
Ariel,> I have 2 Snom 200 and would like to get them to work properly with > Asterisk. With the Firmware 2.02t I am able to use the phone. But only > one line configured. With there newer firmware 2.03o it will not allow > me to make calls. But I can get calls on the unit. Again the 2nd line > is not able to be registered. Is this an issue with Asterisk or Snom? > > I could use some example configuration files. I have followed the Snom > FAQ step by step. But it's still not working.I just upgraded my 200 to v2.03o and its working fine with two extns defined. I happen to be using * CVS-12/04/03-14:24:40 on the same wire (no nat, etc). My sip.conf entries look like: [3007] type=friend host=dynamic username=3007 secret=mypassword context=from-sip [3008] type=friend host=dynamic username=3008 secret=mypassword context=from-sip Using your web browser to config the phone, verify: Settings/SIP/Lines Account = 3007 (to match above sip.conf def) Registrar = ip address of asterisk box Action = proxy Account = 3008 (to match above sip.conf def) Registrar = ip address of asterisk box Action = proxy Settings/SIP/Stack Outbound Proxy, Registrar is outbound proxy = yes Settings/SIP/Authentication Line 1, Realm = asterisk, Username = 3007, Pasword = mypassword Line 2, Realm = asterisk, Username = 3008, Pasword = mypassword Settings/Key Mapping P1 = Line, Number = 3007 P2 = Line, Number = 3008 After ensuring your phone settings actually match the sip.conf settings and that you've properly selected "Save" after changing each of the above entries in the phone, then reboot the phone. If the phone prompts you to download another firmware image, simply press ESC. (Seems some config changes don't take effect until after a phone reboot.) The above config has been working fine with the last several (estimate about 10) firmware versions, however the "user" interaction with several of the keys are rather non-intuitive (or even backwards) for US users. For example, if you answer an incoming call on Line 1 (x3007 above) and place that call on hold using the "Hold" key, then select Line 2 (x3008) to do a consultive call to a different extn, you have to press the ESC key to hang up that second consultive call. If instead of pressing the ESC key you simply press Line 1 to return to the original call, Line 2 is automatically put on hold (instead of dropping the line as it does in the US). If you're not paying attention to the LEDs, you've now tied up the second line/extn until such time as you muck around to release it. If that second (consultive) call happens to be to a pstn user and your Central Office supports calling-party line supervision, you've probably tied up that person's telephone line as well. (Email comments to snom resulted in push-back, suggesting the ESC key is the proper way to drop that second line. I'd guess US users (not techie's) will object to using the phone in any form of production telephony.) I've not tried the 200 with the later CVS versions, so don't have a clue as to what you're milage may be. Rich
Geert Nijpels wrote:> Ariel Batista wrote: >> > I had the same problem, so I emailed SNOM. After a quick and clear > reaction from SNOM, the following turns out: > > I have an SRV record set for Asterisk using both TCP and UDP, because > I was first experimenting with SER and that SIP proxy DOES support > TCP. Asterisk does not. So in firmware 2.02t the phone tried udp > automatically, and in firmware 2.03o the phone tried to use tcp, which > will not work in Asterisk. Removing the TCP SRV entry solved my > problem Maybe this will solve your problem.I found my main problem. It was the way I was setting it up. I was putting the servers IP address in the authentication area. This you have to put asterisk instead. And the IP address goes in the line area.> However, I still have other problems with the SNOM phones: > - All sound stops working sometimes (also ringtone) > - Speech sometimes not working (not sure if it is RTP problem or SNOM > firmware problem) > - Sometimes the phone returns BUSY when not busy in firmware 2.02t. > Resetting the phone or adding a new SIP line solves this. In firmware > 2.03o, this is different. The phone does not respond and Asterisk > gives the error: phone CIRCUIT BUSY. I have seen other posts here > from people having the same problems.I have upgraded to the 2.03o and I have both phones working just fine. I had the choppy sound and found out it was a network cable that was bad. I have it working with great sound now on gsm which before it would only work with alaw. ulaw seems to still have some what I call hickcups. You hear it sound but every now and then you hear a skip.> Other people having the same problems? It makes my case a bit more > clear at SNOM when I can point them to a thread with a lot of people > having the same problems:-) > > I am emailing with SNOM about these issues. > > Kind regards, > > Geert Nijpels > > _______________________________________________ > 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