Hi everybody, I am testing Optipoint 400 Standard SIP (Firmware 2.3.14) with Asterisk. It is posible to dial from another Phone (x-lite) to the Optipoint, but when I try to dial from the Optipoint there is no dialtone and there is only a short beep when I dial Numbers. The Optipoint shows "no Server..." (Registrar?) in Display. Sip debug shows no unusual (to me) Messages. Sip show peers: Name/username Host Dyn Nat ACL Mask Port Status 1006/1006 (Unspecified) D 255.255.255.255 0 Unmonitored 1005/1005 (Unspecified) D 255.255.255.255 0 Unmonitored 1004/1004 192.168.1.98 D 255.255.255.255 5060 Unmonitored ---This is the Optipoint 400 sipgate/wendys 217.10.79.9 255.255.255.255 5060 Unmonitored Optipoint Config: Registrar: 192.168.1.99 SIP-Server: 192.168.1.99 Realm: 192.168.1.99 Routing = Server register by Name (Tested also register by ID doesn't matter since they are the same) SIP conf: [1004] type=friend username=1004 host=dynamic dtmfmode=rfc2833 callerid="1004" <1004> mailbox=1000 context=sip Sip debug peer 1004: SIP Debugging Enabled for IP: 192.168.1.98:5060 Sending to 192.168.1.98 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKa956fdf98 From: 1004 <sip:1004@192.168.1.99>;tag=e4a7266f4d69369;epid=SC2026F5 To: 1004 <sip:1004@192.168.1.99>;tag=as50ba5e89 Call-ID: 8003812aded555fef6f5827f4a12298b@192.168.1.99 CSeq: 847678061 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:1004@192.168.1.99> Content-Length: 0 to 192.168.1.98:5060 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKa956fdf98 From: 1004 <sip:1004@192.168.1.99>;tag=e4a7266f4d69369;epid=SC2026F5 To: 1004 <sip:1004@192.168.1.99>;tag=as50ba5e89 Call-ID: 8003812aded555fef6f5827f4a12298b@192.168.1.99 CSeq: 847678061 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: <sip:1004@192.168.1.99>;expires=3600 Date: Sun, 27 Jun 2004 18:26:39 GMT Content-Length: 0 to 192.168.1.98:5060 Scheduling destruction of call '8003812aded555fef6f5827f4a12298b@192.168.1.99' in 15000 ms 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:1004@192.168.1.98 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.99:5060;branch=z9hG4bK57fa7f3b;rport From: "asterisk" <sip:asterisk@192.168.1.99>;tag=as5071967c To: <sip:1004@192.168.1.98> Contact: <sip:asterisk@192.168.1.99> Call-ID: 2d9992ad78c69f110a8557a74745c333@192.168.1.99 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 36 Messages-Waiting: no Voicemail: 0/0 (no NAT) to 192.168.1.98:5060 Scheduling destruction of call '2d9992ad78c69f110a8557a74745c333@192.168.1.99' in 15000 ms Destroying call '2d9992ad78c69f110a8557a74745c333@192.168.1.99' Destroying call '8003812aded555fef6f5827f4a12298b@192.168.1.99' There is no event on hookoff, but there is still no event at the Softphone that workes fine! Could anybody help? With best regards Marco Wendenburg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040627/a968a5e3/attachment.htm
Hi, nobody got any Idea? ;-( ----- Original Message ----- From: wendys To: Asterisk-Users Sent: Sunday, June 27, 2004 8:43 PM Subject: [Asterisk-Users] Optipoint 400 Standard Sip Hi everybody, I am testing Optipoint 400 Standard SIP (Firmware 2.3.14) with Asterisk. It is posible to dial from another Phone (x-lite) to the Optipoint, but when I try to dial from the Optipoint there is no dialtone and there is only a short beep when I dial Numbers. The Optipoint shows "no Server..." (Registrar?) in Display. Sip debug shows no unusual (to me) Messages. Sip show peers: Name/username Host Dyn Nat ACL Mask Port Status 1006/1006 (Unspecified) D 255.255.255.255 0 Unmonitored 1005/1005 (Unspecified) D 255.255.255.255 0 Unmonitored 1004/1004 192.168.1.98 D 255.255.255.255 5060 Unmonitored ---This is the Optipoint 400 sipgate/wendys 217.10.79.9 255.255.255.255 5060 Unmonitored Optipoint Config: Registrar: 192.168.1.99 SIP-Server: 192.168.1.99 Realm: 192.168.1.99 Routing = Server register by Name (Tested also register by ID doesn't matter since they are the same) SIP conf: [1004] type=friend username=1004 host=dynamic dtmfmode=rfc2833 callerid="1004" <1004> mailbox=1000 context=sip Sip debug peer 1004: SIP Debugging Enabled for IP: 192.168.1.98:5060 Sending to 192.168.1.98 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKa956fdf98 From: 1004 <sip:1004@192.168.1.99>;tag=e4a7266f4d69369;epid=SC2026F5 To: 1004 <sip:1004@192.168.1.99>;tag=as50ba5e89 Call-ID: 8003812aded555fef6f5827f4a12298b@192.168.1.99 CSeq: 847678061 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:1004@192.168.1.99> Content-Length: 0 to 192.168.1.98:5060 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKa956fdf98 From: 1004 <sip:1004@192.168.1.99>;tag=e4a7266f4d69369;epid=SC2026F5 To: 1004 <sip:1004@192.168.1.99>;tag=as50ba5e89 Call-ID: 8003812aded555fef6f5827f4a12298b@192.168.1.99 CSeq: 847678061 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: <sip:1004@192.168.1.99>;expires=3600 Date: Sun, 27 Jun 2004 18:26:39 GMT Content-Length: 0 to 192.168.1.98:5060 Scheduling destruction of call '8003812aded555fef6f5827f4a12298b@192.168.1.99' in 15000 ms 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:1004@192.168.1.98 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.99:5060;branch=z9hG4bK57fa7f3b;rport From: "asterisk" <sip:asterisk@192.168.1.99>;tag=as5071967c To: <sip:1004@192.168.1.98> Contact: <sip:asterisk@192.168.1.99> Call-ID: 2d9992ad78c69f110a8557a74745c333@192.168.1.99 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 36 Messages-Waiting: no Voicemail: 0/0 (no NAT) to 192.168.1.98:5060 Scheduling destruction of call '2d9992ad78c69f110a8557a74745c333@192.168.1.99' in 15000 ms Destroying call '2d9992ad78c69f110a8557a74745c333@192.168.1.99' Destroying call '8003812aded555fef6f5827f4a12298b@192.168.1.99' There is no event on hookoff, but there is still no event at the Softphone that workes fine! Could anybody help? With best regards Marco Wendenburg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040702/83b1f67e/attachment.htm
Hi, I'm kind of a newbie myself. I've had similar problems and it can be very frustrating. I did get them all resolved so I'll share some of what I did in hopes that it will fix your issue. To get some of my phones to work (Grandstream BT100) I had to add a line "nat = yes" in my sip.conf under each phone config. I also had to set the bindaddr, externip, and localnet (which is a network address not a host address) in the general section of sip.conf. I'll put a plug in for Grandstream here. Their phones aren't nearly so expensive as some others and they work very well. Check to make sure they have the features you need, if they do, I definitely recommend them. I believe the issues you are having are NAT related. SIP uses one set of rules for routing and RTP uses a different set of rules. You may see other things - like one way audio - in addition once you are getting the config close. If you have the equipment, one way to isolate NAT issues with non-routeable addresses (192.168.x.x or 10.x.x.x) is to create a VPN tunnel between the network where your server is located and the network where your clients are located. If the system works until you remove the tunnel, you are definitely having NAT problems (the tunnel masks the problem because it will actually send the non-routeable packets to the other side). My personal preference is IPSec, but PPTP or L2TP should work fine for testing. I keep reading everything I can. The wiki is very helpful, even though you have to search for a while to find some answers. I also have pretty good luck by searching in Google for examples of other people's files (which an amazing number of people are kind enough to post). I have posted several issues to this bulletin board and I have gotten very good answers that way too. Don't give up. Once you get your configuration correct, Asterisk works amazingly well. I prefer it over every commercial product I've seen. Regards, John
Roland.Knoerl@student.fh-nuernberg.de
2004-Jul-05 05:57 UTC
[Asterisk-Users] Optipoint 400 Standard Sip
Hi Marco, I faced the same problems with the optipoints. I tried nearly everything, up and downgraded firmware, but nothing worked. I phoned with a siemens engineer, and he told me, that this version is not 100% sip confirm. But there will be a workaround. Hopefully that will not take too long ! greetings Roland, nuermberg, germany
Roland.Knoerl@student.fh-nuernberg.de
2004-Jul-06 04:27 UTC
[Asterisk-Users] Help, plz! Sip:roland.knoerl@company should use a specific outgoing Tel-number. so that the called party is able to see who is calling.
Hi there,
my * is running very well in standard mode, but
I would like to implement some more features.
that=B4s why I hope that someone could help me !
For example:
I have Asterisk working behind a Cisco Gateway.
This is working well, but I would like to
have the ID translated, so that the right telephone-number is transfered.
e.g.: SIP-user roland.knoerl should have the outgoing number 772 .
I=B4m sure that it would be easier to change the username into 772, but=20
I have to implement it that way.
Maybe it has something to do with the line:
exten =3D> _0.,1,Dial(SIP/${EXTEN:0}@mycisco,30,r)
Put something in front of the ${EXTEN:0} ???
For incoming calls, I already did a few lines that makes the user roland.knoerl
available by dialing 772.
Thanks in advance !
Roland Knoerl , Nuremberg , Germany
Roland.Knoerl@student.fh-nuernberg.de
2004-Jul-28 01:54 UTC
[Asterisk-Users] Optipoint 400 Standard Sip
Hello Wendys & Steffen. I tried to change chan_sip.c that way, you told us. but my optiset isn?t working properly. may you be so kind enough to send me your configuration of an working optipoint ? Mine is working as Sip routing: GATEWAY. An outgoing call can be established, but due to not registering to asterisk an incoming call could not be delivered. HELP PLZ. ! ;-) Kind regards & thx for help in advance Roland / Nuermberg / Germany P.S. Wendys k?nnen wir mal mails auf deutsch austauschen. wenn du schon aus n?rnberg bist :-)