Re: [asterisk-users] sip register peer (the quest for near 100% availability)
On Wed, 2013-01-30 at 14:10 -0700, Carlos Alvarez wrote: On Wed, Jan 30, 2013 at 12:05 PM, XBrian bobo...@yahoo.co.uk wrote: Thanks - I was hoping there was some silver bullet to use out there. Thanks anyway. There is. If you build a reliable network, the phones will simply never have a problem. We've got customers with phones that have never lost contact for years. Re-registering is just a crutch for a network defect. -- Carlos Alvarez TelEvolve 602-889-3003 This is so true! -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip register peer (the quest for near 100% availability)
2013/1/31 Ishfaq Malik i...@pack-net.co.uk On Wed, 2013-01-30 at 14:10 -0700, Carlos Alvarez wrote: On Wed, Jan 30, 2013 at 12:05 PM, XBrian bobo...@yahoo.co.uk wrote: Thanks - I was hoping there was some silver bullet to use out there. Thanks anyway. There is. If you build a reliable network, the phones will simply never have a problem. We've got customers with phones that have never lost contact for years. Re-registering is just a crutch for a network defect. -- Carlos Alvarez TelEvolve 602-889-3003 This is so true! If you have no NAT or dynamic IP in your network, you can just remove the registration process and assign to each peer its IP address. Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip register peer (the quest for near 100% availability)
On Thu, Jan 31, 2013 at 3:08 AM, Leandro Dardini ldard...@gmail.com wrote: 2013/1/31 Ishfaq Malik i...@pack-net.co.uk On Wed, 2013-01-30 at 14:10 -0700, Carlos Alvarez wrote: On Wed, Jan 30, 2013 at 12:05 PM, XBrian bobo...@yahoo.co.uk wrote: Thanks - I was hoping there was some silver bullet to use out there. Thanks anyway. If you have no NAT or dynamic IP in your network, you can just remove the registration process and assign to each peer its IP address. This is the answer. If 100% availability is critical, your IP addresses shouldn't be changing anyway, so take the registration process out entirely. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip register peer (the quest for near 100% availability)
If you have no NAT or dynamic IP in your network, you can just remove the registration process and assign to each peer its IP address. This is the answer. If 100% availability is critical, your IP addresses shouldn't be changing anyway, so take the registration process out entirely. This advice is not valid for android / iphones though. You need the register to be able to have good battery life on those. If you use TCP, the softphone will go to sleep, OS will keep the stream alive. When a SIP packet comes in (INVITE, OPTIONS etc), the OS will wake up the softphone and the softphone will handle the packet. No register means no stream and the softphone will just sleep forever. Z -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip register peer (the quest for near 100% availability)
On Thu, Jan 31, 2013 at 9:45 AM, joachim zoach...@securax.org wrote: If you have no NAT or dynamic IP in your network, you can just remove the registration process and assign to each peer its IP address. This is the answer. If 100% availability is critical, your IP addresses shouldn't be changing anyway, so take the registration process out entirely. This advice is not valid for android / iphones though. That's absurd. Why would you use a battery-powered smartphone if you are trying to have 100% availability? -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip register peer (the quest for near 100% availability)
This is the answer. If 100% availability is critical, your IP addresses shouldn't be changing anyway, so take the registration process out entirely. This advice is not valid for android / iphones though. That's absurd. Why would you use a battery-powered smartphone if you are trying to have 100% availability? From what i understood from the original post, Xbrian is looking for a way to work around broken phones that fail to register when they should. I doubt his idea of 100% availability is the same as yours or he would/should be using a different brand/model of phones. + The mobile phone will survive a power outage, because of the register you could be behind NAT as it will open the bindings, you can take it to the bathroom etc. I'm just trying to illustrate the possible advantages of a register before XBbrian redoes his network config. Z -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip register peer (the quest for near 100% availability)
Hello. I am aware that 'sip show peers' will display my peers, and that 'sip unregister ' (where is the peer name) will unregister a peer - however, I want to force registration of a peer from the CLI. Is there any way to force this? I have several user agents and I want to achieve near 100% availability for all peers. I realise that the peer will be 'woken' up at my qualify intervals, but can I actually force registration from the CLI? A REGISTER request originates from the peer. How do you propose Asterisk ask the unregistered peers to REGISTER in a device agnostic fashion? Maybe it's possible to send a NOTIFY to a peer on the last IP it was seen at? I don't think I've seen anything that has a register command, but lots of devices can get a check your config or reboot command via SIP NOTIFY. I'm more wondering why the peer is unregistered but we still expect to communicate with it. Other than a network problem or the device being unplugged...neither of which could be fixed from the server. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip register peer (the quest for near 100% availability)
On Thu, Jan 31, 2013 at 9:26 AM, Adam Moffett adamli...@plexicomm.netwrote: Maybe it's possible to send a NOTIFY to a peer on the last IP it was seen at? I don't think I've seen anything that has a register command, but lots of devices can get a check your config or reboot command via SIP NOTIFY. If you can notify, you can call. This fixes nothing other than refreshing NAT if that's involved. I'm more wondering why the peer is unregistered but we still expect to communicate with it. Other than a network problem or the device being unplugged...neither of which could be fixed from the server. I have a feeling that some people in this discussion have a lack of understanding about the SIP protocol and the underlying networking that could affect it. The original post failed to say whether this was on a LAN without routing, on a LAN with routing, or a WAN. Each of those could result in totally different results and solutions. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip register peer (the quest for near 100% availability)
Another option would be a VPN between the phone and the LAN the Asterisk box is on. VPN software may handle IP address changes better than the Softphone. This way the IP of the softphone doesn't change. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of joachim Sent: Thursday, January 31, 2013 11:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip register peer (the quest for near 100% availability) This is the answer. If 100% availability is critical, your IP addresses shouldn't be changing anyway, so take the registration process out entirely. This advice is not valid for android / iphones though. That's absurd. Why would you use a battery-powered smartphone if you are trying to have 100% availability? From what i understood from the original post, Xbrian is looking for a way to work around broken phones that fail to register when they should. I doubt his idea of 100% availability is the same as yours or he would/should be using a different brand/model of phones. + The mobile phone will survive a power outage, because of the register you could be behind NAT as it will open the bindings, you can take it to the bathroom etc. I'm just trying to illustrate the possible advantages of a register before XBbrian redoes his network config. Z -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip register peer (the quest for near 100% availability)
Is there any way to force this? I have several user agents and I want to achieve near 100% availability for all peers. I realise that the peer will be 'woken' up at my qualify intervals, but can I actually force registration from the CLI? For those peers which are at known, fixed, predictable IP addresses (e.g. in-house phones which have statically-configured IP addresses or which get non-dynamic addresses from a DHCP server you control) you do not need to use registration at all. You can simply hard-code the peer's address into sip.conf (or, I presume, an equivalent realtime table). When you Dial() such a peer, Asterisk will start sending out the INVITE packets, regardless of whether it has heard anything at all from that peer in the last hour or fifty. No need for qualify although you can use this to keep track of whether the peers are actually alive or not. If you take this approach, you'll save yourself a great deal of heartburn if you can figure out an automated way of keeping the IP addresses synchronized, between Asterisk and whatever hand out the addresses mechanism the phones use (DHCP, TFTP-based provisioning files, etc.). Keep a master list of peers and addresses in a simple table or file somewhere, and use this to populate the other pieces of software which need to know. For peers which can move around to arbitrary IP addresses, and where your server system won't know what those addresses may be in advance, using REGISTER from the device is really the only good approach. If you've got a setup where devices change their IP address frequently and need to be on-line constantly, I'd say you have a fundamental problem with no easy solution. Using a short registration time limit (e.g. 30 seconds) is probably the least awful way to handle this, and if you're talking about a very large number of phones you may want to set up a dedicated SIP proxy to handle this registration burden and keep Asterisk from having to deal with it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip register peer (the quest for near 100% availability)
Hello. I am aware that 'sip show peers' will display my peers, and that 'sip unregister ' (where is the peer name) will unregister a peer - however, I want to force registration of a peer from the CLI. Is there any way to force this? I have several user agents and I want to achieve near 100% availability for all peers. I realise that the peer will be 'woken' up at my qualify intervals, but can I actually force registration from the CLI? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip register peer (the quest for near 100% availability)
On 01/30/2013 11:26 AM, XBrian wrote: Hello. I am aware that 'sip show peers' will display my peers, and that 'sip unregister ' (where is the peer name) will unregister a peer - however, I want to force registration of a peer from the CLI. Is there any way to force this? I have several user agents and I want to achieve near 100% availability for all peers. I realise that the peer will be 'woken' up at my qualify intervals, but can I actually force registration from the CLI? A REGISTER request originates from the peer. How do you propose Asterisk ask the unregistered peers to REGISTER in a device agnostic fashion? -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip register peer (the quest for near 100% availability)
I am aware that the direction is from peer to asterisk. Its a valid question. If a solution did exist, guarantees near 100 per cent availability. Especially if the device is actually there. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip register peer (the quest for near 100% availability)
You can just shorten the time the phone device register on the asterisk server. It is up to the peer to send the registration command. It cannot be triggered or forced in any way. Leandro 2013/1/30 XBrian bobo...@yahoo.co.uk I am aware that the direction is from peer to asterisk. Its a valid question. If a solution did exist, guarantees near 100 per cent availability. Especially if the device is actually there. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip register peer (the quest for near 100% availability)
Thanks - I was hoping there was some silver bullet to use out there. Thanks anyway. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip register peer (the quest for near 100% availability)
On Wed, Jan 30, 2013 at 12:05 PM, XBrian bobo...@yahoo.co.uk wrote: Thanks - I was hoping there was some silver bullet to use out there. Thanks anyway. There is. If you build a reliable network, the phones will simply never have a problem. We've got customers with phones that have never lost contact for years. Re-registering is just a crutch for a network defect. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP register refresh time
Hi all, question about register refresh time. One of our supplier had a maintenance work on sat 4 Aug which was replacing the production server for an Asterisk 1.4 running version. We have few Asterisk connected to them (version 1.6, 1.8 and 1.10) with register Username and Passwd. After the new server came up, no one of our Asterisks get registered back, which means no calls -incoming and outgoing- at all since this date :-( A sip show registry this morning (Mon 6 Aug) show us following: Hostdnsmgr Username Refresh State Reg.Time sip.domain.com:5060 N MyUser105 No Authentication Sat,04 Aug 2012 16:55:37 A simple sip reload made thinks working again. Why our Asterisks didn't get back for registration, refresh register time being the standard 120 seconds? Thanks for your explanation -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Register DOS attack
I'll check this option and see if it helps next time, just to clarify, there were no actual calls in place, just DOS register attack. On Wed, Jun 1, 2011 at 12:22 PM, Ira i...@extrasensory.com wrote: At 10:56 AM 6/1/2011, you wrote: Do you have: sip.conf [general] allowguest=no So because of this I decided to type sip show channels into my Asterisk and got this: Peer User/ANRCall ID Format Hold Last Message Expiry Peer 216.xxx.69.xxx (None) f2d8db55-0a7edd (nothing) NoRx: OPTIONS guest 216.xxx.69.xxx (None) 2ce0b9a5-6de7f4 (nothing) NoRx: OPTIONS guest 64.xxx.41.xxx6314098389 2a482e4b684a59a (nothing) No guest 192.168.233.xxx (None) ioh3fna2aw.n4mz (nothing) NoRx: REGISTER guest 4 active SIP dialogs I have allowguest=no and all of those IPs are either my providers or a SIP phone on my network so why would it show guest as the peer? I'm running Asterisk SVN-trunk-r319759M if that matters. Ira -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Register DOS attack
Also you guys may need to use: sip.conf [general] allowguest=no *alwaysauthreject = yes* On Thu, Jun 2, 2011 at 1:01 PM, Al lists asteris...@gmail.com wrote: I'll check this option and see if it helps next time, just to clarify, there were no actual calls in place, just DOS register attack. On Wed, Jun 1, 2011 at 12:22 PM, Ira i...@extrasensory.com wrote: At 10:56 AM 6/1/2011, you wrote: Do you have: sip.conf [general] allowguest=no So because of this I decided to type sip show channels into my Asterisk and got this: Peer User/ANRCall ID Format Hold Last Message Expiry Peer 216.xxx.69.xxx (None) f2d8db55-0a7edd (nothing) NoRx: OPTIONS guest 216.xxx.69.xxx (None) 2ce0b9a5-6de7f4 (nothing) NoRx: OPTIONS guest 64.xxx.41.xxx6314098389 2a482e4b684a59a (nothing) No guest 192.168.233.xxx (None) ioh3fna2aw.n4mz (nothing) NoRx: REGISTER guest 4 active SIP dialogs I have allowguest=no and all of those IPs are either my providers or a SIP phone on my network so why would it show guest as the peer? I'm running Asterisk SVN-trunk-r319759M if that matters. Ira -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Abdullah -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Register DOS attack
On 11-05-31 06:24 PM, Al lists wrote: Hi List Recently i have noticed this attack on couple of servers, usually a foreign IP starts sending tons of register request without any answer to authentication, if you type sip show channels in cli you will see tons of these: 1.2.3.4 (None) 2389603298 00101/1 0x0 (nothing)No Rx: REGISTER since there is no authentication in place, asterisk does not see any failed register attempt, so there wont be anything added to log file as failed attempt. thus fail2ban wont see any activity and wont block the IP. it simply brings down the internet link and the box due to too many sip channels. Do you have: sip.conf [general] allowguest=no -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Register DOS attack
At 10:56 AM 6/1/2011, you wrote: Do you have: sip.conf [general] allowguest=no So because of this I decided to type sip show channels into my Asterisk and got this: Peer User/ANR Call ID Format Hold Last Message Expiry Peer 216.xxx.69.xxx (None) f2d8db55-0a7edd (nothing) No Rx: OPTIONS guest 216.xxx.69.xxx (None) 2ce0b9a5-6de7f4 (nothing) No Rx: OPTIONS guest 64.xxx.41.xxx 6314098389 2a482e4b684a59a (nothing) No guest 192.168.233.xxx (None) ioh3fna2aw.n4mz (nothing) No Rx: REGISTER guest 4 active SIP dialogs I have allowguest=no and all of those IPs are either my providers or a SIP phone on my network so why would it show guest as the peer? I'm running Asterisk SVN-trunk-r319759M if that matters. Ira -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Register DOS attack
Hi List Recently i have noticed this attack on couple of servers, usually a foreign IP starts sending tons of register request without any answer to authentication, if you type sip show channels in cli you will see tons of these: 1.2.3.4 (None) 2389603298 00101/1 0x0 (nothing)No Rx: REGISTER since there is no authentication in place, asterisk does not see any failed register attempt, so there wont be anything added to log file as failed attempt. thus fail2ban wont see any activity and wont block the IP. it simply brings down the internet link and the box due to too many sip channels. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP register and contact header
Hello, I define SIP registrations as follow in sip.conf : register = number:passwd@sip-server example : register = 33:mypass@ip_sip_server But apparently the SIP 'contact' header in the SIP REGISTER looks like this : /Contact: sip:s@ip_my_asterisk/ How come ? And how to change this so it reads : /Contact: sip:/33/@ip_my_asterisk/ Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP register and contact header
On 4/4/11 5:13 PM, Jonas Kellens wrote: I define SIP registrations as follow in sip.conf : register = number:passwd@sip-server example : register = 33:mypass@ip_sip_server But apparently the SIP 'contact' header in the SIP REGISTER looks like this : /Contact: sip:s@ip_my_asterisk/ Change your register line into this: register = 33:mypass@ip_sip_server/33 -- Andreas Sikkema -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP REGISTER header not containing Allow-Events or Allow
The SIP trunking service that I am trying to set up keeps saying that my
registration from Asterisk is invalid.
Asterisk registration:
REGISTER sip:{registration_ip} SIP/2.0
Via: SIP/2.0/UDP {asterisk_ip}:5060;branch=z9hG4bK5c2eb10c;rport
Max-Forwards: 70
From: sip:{registration_us...@{registration_ip};tag=as5579cc0c
To: sip:{registration_us...@{registration_ip}
Call-ID: 651194bd76e02f4d0126373c51568...@127.0.0.1
CSeq: 104 REGISTER
User-Agent: Asterisk PBX 1.6.2.5
Authorization: Digest username={registration_user},
realm=Registered_Subscribers, algorithm=MD5, uri=sip:broadsmart.net,
nonce=b47c87b5e93ba420a0cf25162fa29794,
response=98e4d21ca0f75497d7fb12a8a4914bcb, opaque=5adc3dd2
Expires: 3600
Contact: sip:{registration_us...@{asterisk_ip}
Content-Length: 0
Expected registration:
REGISTER sip:broadsmart.net SIP/2.0
Via: SIP/2.0/UDP
208.73.25.70:5060;branch=z9hG4bK-d87543-826b1b62b62ac91d-1--d87543-;rport
Record-Route: sip:2135997...@208.73.25.70;lr
From: 2135997816 sip:2135997...@broadsmart.net;tag=e944c233
To: 2135997816 sip:2135997...@broadsmart.net
Call-ID: e0576109f9699...@dgfjd3mxlmludc5uyxrlbgnvbw0uy29t
CSeq: 1 REGISTER
Contact: sip:2135997...@208.73.25.70:5060;rinstance=92c0558ad60f5de4
Max-forwards: 70
Expires: 3600
Supported: eventlist
User-agent: CounterPath eyeBeam release 3014w stamp 26275
*Allow-Events: BroadWorksSubscriberData
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO*
Content-Length: 0
They are saying that the Asterisk registration doesn't have an
Allow-Events and an Allow in the header. Would this cause any
problems and can this be set in Asterisk to send those in the header?
Thanks.
--
Mike A. Leonetti
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[asterisk-users] SIP REGISTER
Hi all, I'm facing an issue with my asterisk server when an extension (X-Lite softphone) tries to register on it...A huge amount of packets is exchanged between endpoint and asterisk server while the X-Lite is online...Even when I sign out from X-Lite, the asterisk server continues sending packets to my machine...Can Someone help me in that? Please find the SIP packets between asterisk and X-Lite on http://pastebin.com/d85f913e Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP REGISTER
These are requests where one endpoint pings the other to check if it is still alive. What is the problem? michel freiha wrote: Hi all, I'm facing an issue with my asterisk server when an extension (X-Lite softphone) tries to register on it...A huge amount of packets is exchanged between endpoint and asterisk server while the X-Lite is online...Even when I sign out from X-Lite, the asterisk server continues sending packets to my machine...Can Someone help me in that? Please find the SIP packets between asterisk and X-Lite on http://pastebin.com/d85f913e Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP REGISTER
Dear Alex, The problem is that the asterisk server is sending these packets continuously with no stop and with a negligible duration between packets for the same extension...My Asterisk server read the extensions from the database and not from extensions.conf...There is a field in the sip buddies table with name qualify and with type char...WHat do you suggest me to do? Put the value or qualify to no or ichange the type to int and put a numeric value inside? If I put the value to no what this the disadvantages? Regards On Thu, Oct 30, 2008 at 1:30 PM, Alex Balashov [EMAIL PROTECTED]wrote: These are requests where one endpoint pings the other to check if it is still alive. What is the problem? michel freiha wrote: Hi all, I'm facing an issue with my asterisk server when an extension (X-Lite softphone) tries to register on it...A huge amount of packets is exchanged between endpoint and asterisk server while the X-Lite is online...Even when I sign out from X-Lite, the asterisk server continues sending packets to my machine...Can Someone help me in that? Please find the SIP packets between asterisk and X-Lite on http://pastebin.com/d85f913e Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP REGISTER
By default, the interval at which the qualify pings are sent is, indeed quite low. There is no consequence to disabling it except for the obvious implication that Asterisk then has no way way of knowing if the peer is dead without first trying to reach it, every time and with every request. But there are disadvantages to using 'qualify' for that purpose, too; sometimes there is arbitrary latency in the network that can cause peers to become marked 'Unavailable' rather whimsically. The answer is basically: do whatever you want. No best practices here. Personally, I'd recommend a qualify setting like 2000. michel freiha wrote: Dear Alex, The problem is that the asterisk server is sending these packets continuously with no stop and with a negligible duration between packets for the same extension...My Asterisk server read the extensions from the database and not from extensions.conf...There is a field in the sip buddies table with name qualify and with type char...WHat do you suggest me to do? Put the value or qualify to no or ichange the type to int and put a numeric value inside? If I put the value to no what this the disadvantages? Regards On Thu, Oct 30, 2008 at 1:30 PM, Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: These are requests where one endpoint pings the other to check if it is still alive. What is the problem? michel freiha wrote: Hi all, I'm facing an issue with my asterisk server when an extension (X-Lite softphone) tries to register on it...A huge amount of packets is exchanged between endpoint and asterisk server while the X-Lite is online...Even when I sign out from X-Lite, the asterisk server continues sending packets to my machine...Can Someone help me in that? Please find the SIP packets between asterisk and X-Lite on http://pastebin.com/d85f913e Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP register problem
You changed your default SIP (bindport) port to 5061 at the server, so your client needs to look there. Try like these register = sipteszt:[EMAIL PROTECTED]:50/sipteszt bindport=5061 ; UDP Port to bind to (SIP standard port is 5060) Adrià Vidal ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP register problem
Hi all, We have two asterisk PBX. We would like to register it with SIP peer. The client sends the register request. It gets back: Jan 2 01:31:27 WARNING[186]: chan_sip.c:9760 handle_response_register: Got 404 Not found on SIP register to service [EMAIL PROTECTED], giving up server: Sip.conf [general] context=blackbox-in ; Default context for incoming calls realm=xxx bindport=5061 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls [sipteszt] type=user context=siporka username=sipteszt1 fromuser=sipteszt1 secret=password accountcode=sipteszt host=dynamic disallow=all ;allow=ilbc ;allow=speex allow=gsm allow=speex allow=alaw nat=yes notransfer=yes canreinvite=no qualify=no [sipteszt] type=peer fromuser=sipteszt1 username=sipteszt1 secret=password accountcode=sipteszt host=dynamic disallow=all ;allow=ilbc ;allow=speex allow=gsm allow=speex allow=alaw notransfer=yes canreinvite=no qualify=yes Client: sip.conf [general] context=default ; Default context for incoming calls accountcode=sip bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls callevents=yes ; generate manager events when sip ua performs events (e.g. hold) subscribecontext=ext-local-custom register = sipteszt:[EMAIL PROTECTED]/sipteszt [sipteszt] type=peer host=217.xxx.32.207 fromuser=sipteszt1 fromdomain= username=sipteszt1 secret=password dtmfmode=rfc2833 canreinvite=no disallow=all allow=alaw allow=gsm allow=ilbc nat=no qualify=no accountcode=12 trunktimestamps=no ; incoming peer from 217.xxx.32.207 [sipteszt-in] type=user host=217.xxx.32.207 username=sipteszt1 context=incoming dtmfmode=rfc2833 disallow=all allow=alaw allow=gsm allow=ilbc accountcode=12 trunktimestamps=no Kind regards Szolke ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP register question
I am trying to link an asterisk box up to a SIP server on the same subnet. The SIP server does not have a password (and is locked down by IP number 'allow'). How do I specify this on the register line? Based on the documentation, the line looks like this: register = user[:secret[:[EMAIL PROTECTED]:port][/extension] It looks like [EMAIL PROTECTED] is the minimum required. Is there anyway to specify a username of null, or something? Thanks in advance! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Register
I'm having trouble making calls over my VoIP provider. I do successfully
register, and when I try to establish a phone call Asterisk sends wrong
username and password. Instead of sending username and pass that I have
provided, he send username and pass of the SIP phone that is registered to *
(the phone from which I try to make a call).
What have I done wrong?
This is my sip.conf
[general]
context=sip
port=5060
bindaddr=0.0.0.0
srvlookup=no
tos=184
maxexpirey=3600
defaultexpirey=120
disallow=all
allow=ulaw
allow=alaw
allow=gsm
musicclass=default
useragent=PBX Lama
nat=no
externip = 200.200.200.200 ; my external IP
localnet = 10.0.0.0/255.255.255.0
realm=lama.hr
register = myusername:[EMAIL PROTECTED]
canreinvite=no
[iskon1]
type=friend
username=myusername
secret=mypass
host=sip.iskon.hr
nat=yes
canreinvite=no
[214]
callerid=Vice Lacmanovic 214
type=friend
username=214
secret=vice
host=dynamic
mailbox=214
canreinvite=no
dtmfmode=inband
And this is part of my extensions.conf - the line I use for calling out.
exten = _8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
Again, problem is that Asterisk to my VoIP provider sends username 214 and pass
vice (data of my SIP phone) and not the data that I have provide to it
(myusername and mypassword for that VoIP provider).
Thank you for your time.
--
Tomislav Parcina
tparcina#lama.hr
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RE: [Asterisk-Users] SIP Register
First impressions telling me you want to check your phone settings. What
phone are you using and what are the config settings?
Mark
-Original Message-
From: Tomislav Parèina [mailto:[EMAIL PROTECTED]
Sent: Tuesday, 14 February 2006 9:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] SIP Register
I'm having trouble making calls over my VoIP provider. I do successfully
register, and when I try to establish a phone call Asterisk sends wrong
username and password. Instead of sending username and pass that I have
provided, he send username and pass of the SIP phone that is registered to *
(the phone from which I try to make a call).
What have I done wrong?
This is my sip.conf
[general]
context=sip
port=5060
bindaddr=0.0.0.0
srvlookup=no
tos=184
maxexpirey=3600
defaultexpirey=120
disallow=all
allow=ulaw
allow=alaw
allow=gsm
musicclass=default
useragent=PBX Lama
nat=no
externip = 200.200.200.200 ; my external IP
localnet = 10.0.0.0/255.255.255.0
realm=lama.hr
register = myusername:[EMAIL PROTECTED]
canreinvite=no
[iskon1]
type=friend
username=myusername
secret=mypass
host=sip.iskon.hr
nat=yes
canreinvite=no
[214]
callerid=Vice Lacmanovic 214
type=friend
username=214
secret=vice
host=dynamic
mailbox=214
canreinvite=no
dtmfmode=inband
And this is part of my extensions.conf - the line I use for calling out.
exten = _8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
Again, problem is that Asterisk to my VoIP provider sends username 214 and
pass vice (data of my SIP phone) and not the data that I have provide to it
(myusername and mypassword for that VoIP provider).
Thank you for your time.
--
Tomislav Parcina
tparcina#lama.hr
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[Asterisk-Users] SIP register vs SIP with a fixed IP
Hi, Two questionsfor the gurus out here: 1) I recently asked, for a number of reasons, to have my provider changehis way of doing SIP wth me: instead of registering with his server, I know simply send my stuff to his IP without registration. I have always had two test numbers: one IAX and one SIP. When I get a call on either of those numbers, the call is bridged and send back over SIP to my land line. Since I made the switch to fixed IP (although it may have occured before without my knowledge) I get the following error ONLY when the call comes invia SIP "Got SIP response 488 "Not Acceptable Here" back from111.111.111.111", this being the IP I am sending my calls to. When it comes through IAX, the call is bridged without any problem. What could be the cause of this problem? 2) The way my contexts were setup before is that when a call came in through 555-555-, it landed in context_a. when it came in through 555-666-, it landed in context_b. This way I could have a different dial plan per customer (since I resell VoIP services). This was easy before I was using SIP registration. Now that I am using a fixed IP with no registration, how do I switch the call to a different context based on the number called? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP REGISTER
From the dump that I have attached It looks like the first attempt at register does not work then followd by a second register which then works. This is happening on all the SIP phone attach to asterisk. The version of asterisk here is 1.2.0b2. Here is sip.conf for ext 204 [204] username=204 type=friend secret= md5secret=356381525bb1969a32743b58db400342 auth=md5 record_out=Adhoc record_in=Adhoc port=5060 [EMAIL PROTECTED] host=dynamic context=from-sip canreinvite=no callerid=Library 204 Is there somethng that I am missing to have phone only reigster once and not get the 401 unauthorized on the first attempt which then get follow by the same register but get 200 Ok. No. TimeSourceDestination Protocol Info 3 0.494325192.168.3.70 192.168.3.28 SIP Request: REGISTER sip:192.168.3.28 Frame 3 (675 bytes on wire, 675 bytes captured) Ethernet II, Src: 192.168.3.70 (00:0e:08:ca:5f:2d), Dst: 192.168.3.28 (00:a0:c9:e7:9c:6e) Internet Protocol, Src: 192.168.3.70 (192.168.3.70), Dst: 192.168.3.28 (192.168.3.28) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:192.168.3.28 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.3.70:5060;branch=z9hG4bK-8c701711 From: Great Room sip:[EMAIL PROTECTED];tag=51baed127acfd9a6o0 SIP Display info: Great Room SIP from address: sip:[EMAIL PROTECTED] SIP tag: 51baed127acfd9a6o0 To: Great Room sip:[EMAIL PROTECTED] SIP Display info: Great Room SIP to address: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 79 REGISTER Max-Forwards: 70 Authorization: Digest username=204,realm=asterisk,nonce=6948ea10,uri=sip:192.168.3.28,algorithm=MD5,response=3c05b1dab053a739d7c8ad941ac98cee Contact: Great Room sip:[EMAIL PROTECTED]:5060;expires=60 Contact Binding: Great Room sip:[EMAIL PROTECTED]:5060;expires=60 User-Agent: Sipura/SPA3000-3.1.7(GWg) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura No. TimeSourceDestination Protocol Info 4 0.495379192.168.3.28 192.168.3.70 SIP Status: 100 Trying(1 bindings) Frame 4 (473 bytes on wire, 473 bytes captured) Ethernet II, Src: 192.168.3.28 (00:a0:c9:e7:9c:6e), Dst: 192.168.3.70 (00:0e:08:ca:5f:2d) Internet Protocol, Src: 192.168.3.28 (192.168.3.28), Dst: 192.168.3.70 (192.168.3.70) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 Trying Message Header Via: SIP/2.0/UDP 192.168.3.70:5060;branch=z9hG4bK-8c701711;received=192.168.3.70 From: Great Room sip:[EMAIL PROTECTED];tag=51baed127acfd9a6o0 SIP Display info: Great Room SIP from address: sip:[EMAIL PROTECTED] SIP tag: 51baed127acfd9a6o0 To: Great Room sip:[EMAIL PROTECTED] SIP Display info: Great Room SIP to address: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 79 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED] Contact Binding: sip:[EMAIL PROTECTED] Content-Length: 0 No. TimeSourceDestination Protocol Info 5 0.495662192.168.3.28 192.168.3.70 SIP Status: 401 Unauthorized(1 bindings) Frame 5 (555 bytes on wire, 555 bytes captured) Ethernet II, Src: 192.168.3.28 (00:a0:c9:e7:9c:6e), Dst: 192.168.3.70 (00:0e:08:ca:5f:2d) Internet Protocol, Src: 192.168.3.28 (192.168.3.28), Dst: 192.168.3.70 (192.168.3.70) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 401 Unauthorized Message Header Via: SIP/2.0/UDP 192.168.3.70:5060;branch=z9hG4bK-8c701711;received=192.168.3.70 From: Great Room sip:[EMAIL PROTECTED];tag=51baed127acfd9a6o0 SIP Display info: Great Room SIP from address: sip:[EMAIL PROTECTED] SIP tag: 51baed127acfd9a6o0 To: Great Room sip:[EMAIL PROTECTED];tag=as6a7c3473 SIP Display info: Great Room SIP to address: sip:[EMAIL PROTECTED] SIP tag: as6a7c3473 Call-ID: [EMAIL PROTECTED] CSeq: 79 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED] Contact Binding: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=3dbc57fe Content-Length: 0 No. TimeSource
Re: [Asterisk-Users] sip register incoming call contexts?
Hello,
I have set up 2 different fwd.pulver.com accounts on my Asterisk. One will ring all my phones through one context, while the other account was set up to fool Nigerian scam artists, and will go directly to a special voicemail (after a few rings to give the impression of ringing a real telephone). I will not even know somebody called until I get the voicemail in the mail.
The first register goes like this:
register = 18469:[EMAIL PROTECTED]/89
while the number that goes directly to the answering machine is as follows:
register = 18336:[EMAIL PROTECTED]/36
Then I match the digits (36 and 89)within the contexts.
89 triggers the [inbound-fwd] context, while 36 triggers [boguscall]:
[boguscall]exten = 36,1,NoOp(This is context boguscall)exten = 36,2,Wait(0)exten = 36,3,Ringingexten = 36,4,Wait(15)exten = 36,5,Voicemail(su36)exten = 36,6,Hangup
[inbound-fwd]
exten = 89,2,Goto(ringall,${EXTEN},1) ; will go to context [ringall]
[ringall] ; Dial all telephones in the houseexten = _X.,1,Dial(SIP/30SIP/31SIP/32,35),t
Thor
On 10/10/05, Steve Gladden [EMAIL PROTECTED] wrote:
Sorry this is a bit of a newbie question, I've been at this for a fewmonths and still have not quite figured this one out.
I've been able to setup one itsp (incoming calls) (sip account) with aregister line like this:register = nnn:[EMAIL PROTECTED]
-or-register = nnn:[EMAIL PROTECTED]/nnnto come directly into an extension in the dialplanIt seems that this only works with the default context in the dialplan.
I have another sip account from another provider that I would likeall of it's incoming calls to come into the s, extension ofa new context but I have been unable to figure outhow to bring calls from a register line into an alternate context.
It seems that register lines are limited to only being used in thegeneral section of sip.conf and you are limited to one context=statement there.Is there a way to register a second account and have it's calls come into
another context in the dialplan?register lines only seem to work in [general] and it seems like youare limited to only one inbound context here.I would like the two inbound call accounts to be 'isolated' from each other
and not have to come in on the same incoming context in the dialplan.I'd also like to be able to have them have their own contexts with thierown s, (start) extension available.Thanks!Steve
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Re: [Asterisk-Users] sip register incoming call contexts?
I did try this and did get it to register as this peer. However inbound calls to that number are still coming into the context defined in [general] sip.conf I now have two numbers configured, the new peer as you sugested and my original that just has the register line without an associated peer section. BOTH numbers are still coming into the context defined in [general] THis is fine for the number of which I did not create a peer section for. The other number that I did indeed create a peer section for is not coming into the context that I set within the peer context= I of course am doing a full stop of asterisk and restart/reload for each test. Am I still doing something wrong here? Thanks! Steve Create a peer with a host= setting that matches the IP of the service provider's proxy. Set context for this peer. There are several examples out there, one is http://edvina.net/broadvoice/ /Olle Steve Gladden wrote: Sorry this is a bit of a newbie question, I've been at this for a few months and still have not quite figured this one out. I've been able to setup one itsp (incoming calls) (sip account) with a register line like this: register = nnn:[EMAIL PROTECTED] -or- register = nnn:[EMAIL PROTECTED]/nnn to come directly into an extension in the dialplan It seems that this only works with the default context in the dialplan. I have another sip account from another provider that I would like all of it's incoming calls to come into the s, extension of a new context but I have been unable to figure out how to bring calls from a register line into an alternate context. Create a peer with a host= setting that matches the IP of the service provider's proxy. Set context for this peer. There are several examples out there, one is http://edvina.net/broadvoice/ /Olle ___ --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
Re: [Asterisk-Users] sip register incoming call contexts?
Yep thanks for the reply, I figured out pretty quickly after one test that the /s did not work. The issue remains that I have been unsuccessful in getting an incoming call to come into any other context other than the one specified in sip.conf [general] section Anything I'm missing here? I have my context that I want incoming calls to come into setup as a peer (as I was instructed here context= inside of peer definition) and I have the context in extensions.conf yet the calls are still landing in the context defined in sip.conf [general] section. What the heck am I missing here? am completely restarting asterisk and reload and sip reload each try just to be sure. Steve OK, I'm starting to get somwhere with this, I'm at least registering now.. however My inbound calls are still coming into the context defined in [general] of sip.conf and not into the context I have defined in my peer and extensions.conf Here is what I have done: IN sip.conf: register = nnn:[EMAIL PROTECTED] ;also tried register = 2484987171:[EMAIL PROTECTED]/s The above statement with /s is not doing what you think it is. That statement would essentially tell your provider to dial s at your site for incoming calls, and their isn't such a thing as s. The register statement without the s (as in [EMAIL PROTECTED]) will have your provider send calls to you with no digits dialed by them. When a call comes into your machine with no digits dialed, the exten = s part of your dialplan will be executed. The s extension is a special case matching no dialed digits. You can't force it by registering with a /s at the end. exten = s,1,answer exten = s,2,wait(1) exten = s,3,Playback(testaudiofile) exten = s,4,wait(1) exten = s,5,hangup Verify that your register statement is doing what you expect by doing a 'sip show registry'. ___ --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] sip register incoming call contexts?
Sorry this is a bit of a newbie question, I've been at this for a few months and still have not quite figured this one out. I've been able to setup one itsp (incoming calls) (sip account) with a register line like this: register = nnn:[EMAIL PROTECTED] -or- register = nnn:[EMAIL PROTECTED]/nnn to come directly into an extension in the dialplan It seems that this only works with the default context in the dialplan. I have another sip account from another provider that I would like all of it's incoming calls to come into the s, extension of a new context but I have been unable to figure out how to bring calls from a register line into an alternate context. It seems that register lines are limited to only being used in the general section of sip.conf and you are limited to one context= statement there. Is there a way to register a second account and have it's calls come into another context in the dialplan? register lines only seem to work in [general] and it seems like you are limited to only one inbound context here. I would like the two inbound call accounts to be 'isolated' from each other and not have to come in on the same incoming context in the dialplan. I'd also like to be able to have them have their own contexts with thier own s, (start) extension available. Thanks! Steve ___ --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] sip register incoming call contexts?
Sorry this is a bit of a newbie question, I've been at this for a few months and still have not quite figured this one out. I've been able to setup one itsp (incoming calls) (sip account) with a register line like this: register = nnn:[EMAIL PROTECTED] -or- register = nnn:[EMAIL PROTECTED]/nnn to come directly into an extension in the dialplan It seems that this only works with the default context in the dialplan. I have another sip account from another provider that I would like all of it's incoming calls to come into the s, extension of a new context but I have been unable to figure out how to bring calls from a register line into an alternate context. It seems that register lines are limited to only being used in the general section of sip.conf and you are limited to one context= statement there. Is there a way to register a second account and have it's calls come into another context in the dialplan? register lines only seem to work in [general] and it seems like you are limited to only one inbound context here. I would like the two inbound call accounts to be 'isolated' from each other and not have to come in on the same incoming context in the dialplan. I'd also like to be able to have them have their own contexts with thier own s, (start) extension available. Try using something like: deny=0.0.0.0/0.0.0.0 permit=147.135.8.129/255.255.255.0 permit=147.135.0.129/255.255.255.0 permit=147.135.4.128/255.255.255.0 in each sip.conf itsp definition to limit which contexts will match. Obviously, replace the above permit's IP addresses with the correct ones for your provider. ___ --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] sip register incoming call contexts?
Ok :) -- From: Rich Adamson[SMTP:[EMAIL PROTECTED] Reply To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, October 10, 2005 11:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:Re: [Asterisk-Users] sip register incoming call contexts? Sorry this is a bit of a newbie question, I've been at this for a few months and still have not quite figured this one out. I've been able to setup one itsp (incoming calls) (sip account) with a register line like this: register = nnn:[EMAIL PROTECTED] -or- register = nnn:[EMAIL PROTECTED]/nnn to come directly into an extension in the dialplan It seems that this only works with the default context in the dialplan. I have another sip account from another provider that I would like all of it's incoming calls to come into the s, extension of a new context but I have been unable to figure out how to bring calls from a register line into an alternate context. It seems that register lines are limited to only being used in the general section of sip.conf and you are limited to one context= statement there. Is there a way to register a second account and have it's calls come into another context in the dialplan? register lines only seem to work in [general] and it seems like you are limited to only one inbound context here. I would like the two inbound call accounts to be 'isolated' from each other and not have to come in on the same incoming context in the dialplan. I'd also like to be able to have them have their own contexts with thier own s, (start) extension available. Try using something like: deny=0.0.0.0/0.0.0.0 permit=147.135.8.129/255.255.255.0 permit=147.135.0.129/255.255.255.0 permit=147.135.4.128/255.255.255.0 in each sip.conf itsp definition to limit which contexts will match. Obviously, replace the above permit's IP addresses with the correct ones for your provider. ___ --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
RE: [Asterisk-Users] sip register incoming call contexts?
Thank you for your reply and your help. I am still confused here and apologize. To some degree I still do not know what I am doing. We use 2 ITSP's and one of them we have multiple SIP accounts on so I will not be able to do this by IP address. For incoming calls we use a register line in the [general] section of sip.conf like: register = nnn:[EMAIL PROTECTED] We do not have an 'itsp section' for incoming calls. incoming calls come into the context defined in the general section of sip.conf. This is how we learned how to do it from the documentation my understanding is that anything else that involves sections like [itsp-provider out] yada= yada= yada= -or- [itsp-provider-in] yada= yada= yada= Works for permanent non-registered types of connections. I've experimented with trying to put register lines within anything else other than [general] in sip.conf and it does not work and causes a busy signal for an incoming caller. My further under(possibly-mis)undertanding is that with our type of itsp (sip) it requires us to register for incoming calls, and there may be no other way to accept incoming calls from our ITSP, It also seems that register lines only work in the [general] section of sip.conf which only allows me to define one single incoming context is this correct? So the matching by IP address is interesting but confusing and may not apply to what I am trying to do. I will not be able to match by ip with seeveral incoing sip (phone numbers) that I would like to come into their own context but come from the same IP address. Thanks!! Steve Ok :) -- From: Rich Adamson[SMTP:[EMAIL PROTECTED] Reply To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, October 10, 2005 11:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] sip register incoming call contexts? Sorry this is a bit of a newbie question, I've been at this for a few months and still have not quite figured this one out. I've been able to setup one itsp (incoming calls) (sip account) with a register line like this: register = nnn:[EMAIL PROTECTED] -or- register = nnn:[EMAIL PROTECTED]/nnn to come directly into an extension in the dialplan It seems that this only works with the default context in the dialplan. I have another sip account from another provider that I would like all of it's incoming calls to come into the s, extension of a new context but I have been unable to figure out how to bring calls from a register line into an alternate context. It seems that register lines are limited to only being used in the general section of sip.conf and you are limited to one context= statement there. Is there a way to register a second account and have it's calls come into another context in the dialplan? register lines only seem to work in [general] and it seems like you are limited to only one inbound context here. I would like the two inbound call accounts to be 'isolated' from each other and not have to come in on the same incoming context in the dialplan. I'd also like to be able to have them have their own contexts with thier own s, (start) extension available. Try using something like: deny=0.0.0.0/0.0.0.0 permit=147.135.8.129/255.255.255.0 permit=147.135.0.129/255.255.255.0 permit=147.135.4.128/255.255.255.0 in each sip.conf itsp definition to limit which contexts will match. Obviously, replace the above permit's IP addresses with the correct ones for your provider. ___ --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 ___ --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] sip register incoming call contexts?
Steve Gladden wrote: Sorry this is a bit of a newbie question, I've been at this for a few months and still have not quite figured this one out. I've been able to setup one itsp (incoming calls) (sip account) with a register line like this: register = nnn:[EMAIL PROTECTED] -or- register = nnn:[EMAIL PROTECTED]/nnn to come directly into an extension in the dialplan It seems that this only works with the default context in the dialplan. I have another sip account from another provider that I would like all of it's incoming calls to come into the s, extension of a new context but I have been unable to figure out how to bring calls from a register line into an alternate context. Create a peer with a host= setting that matches the IP of the service provider's proxy. Set context for this peer. There are several examples out there, one is http://edvina.net/broadvoice/ /Olle ___ --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] sip register incoming call contexts?
Long list of questions to follow: Short version: Does the register line mate to a peer or is a register line totally unrelated to a peer that is defined? When using a register line does it have to refer to actual hostnames? or can you refer to the peer name in the register line instead of an actual hostname/IP like this: register = username:password@peername/extension [peername] ; where peername does NOT = a hostname or ip address - Long version, Thanks, I have actually tried this and am still confused. Please bear with me... Also I was unclear on my first message and did not state that I also need to have a few seperate incoming contexts work from the same provider as well as two different providers. From the link you provided the confusion starts wit hthe very first line: It reads, register = accountid@sip.broadvoice.com:password:account id@sip.broadvoice.com/extension I'm used to: register = accountid@sip.broadvoice.com:password/extension This example has the accountid@sip.broadvoice.com:password/extension actually iterrated twice like: accountid@sip.broadvoice.com:password:account id@sip.broadvoice.com/extension (the same thing twice with a colon : in the middle. Is this an error? Next: You suggest to create a peer with the context= to what I need. This brings me back to the same problem of the [peer] name of [provider.net] I don't think I can make several peers called [my.provider.net] If I do.. register line will not know which one to use as they would all be called [my.provider.net] Do I need to make several different names and then do some tricks in DNS to make the different hostnames resolve to the same IP of my provider that I need? Please remember I am also trying to get several accounts from the same provider/IP to come into their own contexts. as well as an account or two from another provider. I get your point on what to do but I still do not undertand how to make several seperate peers that use the same IP/host yet bring their calls into separate contexts. The example shows using your providers hostname as the name of the peer. I guess you could name them something else but then how does the register line work with the peer if you do not use the hostname??? can you have something like this work? so far I have tried this and it has been unsuccessful. I'm still unclear as to if the registerline needs to use actual hostnames or can it use the name of the peer and get it from the peer section? And does the name of the peer somehow mate the register line with the peer if the name of the peer matches the ?hostname? in the register line, And does it even need to be an actual hostname? Can you only use the real hostname in the peer and NOT use the hostname in the register line? I hope you can undertand my confusion on this with all the examples using hostnames and none of the examples using two incoming contexts with the same hostname yet different sip accounts. Thank you for your help and your patience. I think I almost understand this but not quite yet! register = username:[EMAIL PROTECTED]/extension [testpeer] type=peer ;Enter your closest proxy server host=sip.myprovider.com fromdomain=sip.myprovider.com fromuser=accountid secret=password context=my-incoming-context-3 ;Disable canreinvite if you are behind a NAT canreinvite=no ;Don't try to authenticate on incoming calls insecure=very Steve Gladden wrote: Sorry this is a bit of a newbie question, I've been at this for a few months and still have not quite figured this one out. I've been able to setup one itsp (incoming calls) (sip account) with a register line like this: register = nnn:[EMAIL PROTECTED] -or- register = nnn:[EMAIL PROTECTED]/nnn to come directly into an extension in the dialplan It seems that this only works with the default context in the dialplan. I have another sip account from another provider that I would like all of it's incoming calls to come into the s, extension of a new context but I have been unable to figure out how to bring calls from a register line into an alternate context. Create a peer with a host= setting that matches the IP of the service provider's proxy. Set context for this peer. There are several examples out there, one is http://edvina.net/broadvoice/ /Olle ___ --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
Re: [Asterisk-Users] sip register incoming call contexts?
OK, I'm starting to get somwhere with this, I'm at least registering now.. however My inbound calls are still coming into the context defined in [general] of sip.conf and not into the context I have defined in my peer and extensions.conf Here is what I have done: IN sip.conf: register = nnn:[EMAIL PROTECTED] ;also tried register = 2484987171:[EMAIL PROTECTED]/s [ptest] type=peer ;Enter your closest proxy server host=sip.myprovider.net fromdomain=sip.myprovider.net fromuser=nnn secret=:ppp context=incoming2test ;Disable canreinvite if you are behind a NAT canreinvite=no ;Don't try to authenticate on incoming calls insecure=very nat=yes In extensions.conf: [incoming2test] exten = s,1,answer exten = s,2,wait(1) exten = s,3,Playback(testaudiofile) exten = s,4,wait(1) exten = s,5,hangup ; It never makes it into here but gets grabbed by s, defined in [general] section of sip.conf Any ideas?? Thanks! Steve -- Long list of questions to follow: Short version: Does the register line mate to a peer or is a register line totally unrelated to a peer that is defined? When using a register line does it have to refer to actual hostnames? or can you refer to the peer name in the register line instead of an actual hostname/IP like this: register = username:password@peername/extension [peername] ; where peername does NOT = a hostname or ip address - Long version, Thanks, I have actually tried this and am still confused. Please bear with me... Also I was unclear on my first message and did not state that I also need to have a few seperate incoming contexts work from the same provider as well as two different providers. From the link you provided the confusion starts wit hthe very first line: It reads, register = accountid@sip.broadvoice.com:password:account id@sip.broadvoice.com/extension I'm used to: register = accountid@sip.broadvoice.com:password/extension This example has the accountid@sip.broadvoice.com:password/extension actually iterrated twice like: accountid@sip.broadvoice.com:password:account id@sip.broadvoice.com/extension (the same thing twice with a colon : in the middle. Is this an error? Next: You suggest to create a peer with the context= to what I need. This brings me back to the same problem of the [peer] name of [provider.net] I don't think I can make several peers called [my.provider.net] If I do.. register line will not know which one to use as they would all be called [my.provider.net] Do I need to make several different names and then do some tricks in DNS to make the different hostnames resolve to the same IP of my provider that I need? Please remember I am also trying to get several accounts from the same provider/IP to come into their own contexts. as well as an account or two from another provider. I get your point on what to do but I still do not undertand how to make several seperate peers that use the same IP/host yet bring their calls into separate contexts. The example shows using your providers hostname as the name of the peer. I guess you could name them something else but then how does the register line work with the peer if you do not use the hostname??? can you have something like this work? so far I have tried this and it has been unsuccessful. I'm still unclear as to if the registerline needs to use actual hostnames or can it use the name of the peer and get it from the peer section? And does the name of the peer somehow mate the register line with the peer if the name of the peer matches the ?hostname? in the register line, And does it even need to be an actual hostname? Can you only use the real hostname in the peer and NOT use the hostname in the register line? I hope you can undertand my confusion on this with all the examples using hostnames and none of the examples using two incoming contexts with the same hostname yet different sip accounts. Thank you for your help and your patience. I think I almost understand this but not quite yet! register = username:[EMAIL PROTECTED]/extension [testpeer] type=peer ;Enter your closest proxy server host=sip.myprovider.com fromdomain=sip.myprovider.com fromuser=accountid secret=password context=my-incoming-context-3 ;Disable canreinvite if you are behind a NAT canreinvite=no ;Don't try to authenticate on incoming calls insecure=very Steve Gladden wrote: Sorry this is a bit of a newbie question, I've been at this for a few months and still have not quite figured this one out. I've been able to setup one itsp (incoming calls) (sip account) with a register line like this: register = nnn:[EMAIL PROTECTED] -or- register = nnn:[EMAIL PROTECTED]/nnn to come directly into an extension in the dialplan It seems that this only
Re: [Asterisk-Users] sip register incoming call contexts?
OK, I'm starting to get somwhere with this, I'm at least registering now.. however My inbound calls are still coming into the context defined in [general] of sip.conf and not into the context I have defined in my peer and extensions.conf Here is what I have done: IN sip.conf: register = nnn:[EMAIL PROTECTED] ;also tried register = 2484987171:[EMAIL PROTECTED]/s The above statement with /s is not doing what you think it is. That statement would essentially tell your provider to dial s at your site for incoming calls, and their isn't such a thing as s. The register statement without the s (as in [EMAIL PROTECTED]) will have your provider send calls to you with no digits dialed by them. When a call comes into your machine with no digits dialed, the exten = s part of your dialplan will be executed. The s extension is a special case matching no dialed digits. You can't force it by registering with a /s at the end. exten = s,1,answer exten = s,2,wait(1) exten = s,3,Playback(testaudiofile) exten = s,4,wait(1) exten = s,5,hangup Verify that your register statement is doing what you expect by doing a 'sip show registry'. ___ --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] SIP register More then 1 by one userid
Hi I tested i can able to register 2 sip phone by same user id and same phone number. I need help to view there IP . i just find one . not two of them, is there any command i can view both registration IP. Thanks. Bashir ___ 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] SIP register more then 1 with same username
Hi all * user I did connected with * from 2 sip-softphone and i registered with asterisk under same username and password and working both fine. but * shows only one. is there any way to find them both by using any tips. Bashir ___ 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 register more then 1 with same username
Asterisk does not support multiple SIP registrations with same username. -Matthew From: Bashir Ullah - www.Lamsre.Com [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sat, 2 Apr 2005 04:04:16 -0800 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP register more then 1 with same username Hi all * user I did connected with * from 2 sip-softphone and i registered with asterisk under same username and password and working both fine. but * shows only one. is there any way to find them both by using any tips. Bashir ___ 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 register problem
Karl Brose wrote: Is the SIPquest server sending the 401 Unauthorized message verbatim as you printed it here? I.e. is the WWW-Authentcate header broken up into several lines like that? If so, how man spaces are actually at the beginning of each new line? Continuation lines are allowed in SIP, but I think it's only one space at the beginning of each new line. To make Asterisk parse this correctly you need to turn on pedantic=yes It's silly that Asterisk doesn't turn this header parsing on by default, no reason not to. I agree. /O ___ 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] SIP register problem
Hi,
Thank you, it's working now!
Do you think that this patch will be included in the next cvs versions?
Sincerely,
Cyrille Demaret
- Original Message -
From: Karl Brose [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Tuesday, November 16, 2004 8:29 PM
Subject: Re: [Asterisk-Users] SIP register problem
Just checked the RFC, and it does say that a tab is acceptable.
SIP header field values can be folded onto multiple lines if the
continuation line begins with a space or horizontal tab. All linear
white space, including folding, has the same semantics as SP. A
recipient MAY replace any linear white space with a single SP before
interpreting the field value or forwarding the message downstream.
This is intended to behave exactly as HTTP/1.1 as described in RFC
2616 [8]. The SWS construct is used when linear white space is
optional, generally between tokens and separators.
LWS = [*WSP CRLF] 1*WSP ; linear whitespace
SWS = [LWS] ; sep whitespace
This means that Asterisk is broken.
Please try the patch attached to this e-mail
filename is lws2sws.patch.txt
Just in case, here is how to:
copy the patch to your Asterisk source code channels directory:
(assuming that /usr/src/asterisk is your base)
cp lws2sws.patch.txt /usr/src/asterisk/channel
cd /usr/src/asterisk/channels
It would be safer to make a backup copy of chan_sip.c before patching.
patch chan_sip.c lws2sws.patch.txt
cd ..
then recompile asterisk and install the new chan_sip.so file
test with pedantic=yes
unless you removed the if clause
--- chan_sip.c 2004-11-16 14:14:42.0 -0500
+++ chan_sip.c_lws 2004-11-16 14:18:44.0 -0500
@@ -2499,33 +2499,32 @@
}
/* Check for end-of-line */
if (msgbuf[h] == '\n') {
- /* Check for end-of-message */
+ /* Check for end-of-message */
if (h + 1 == len)
- break;
- /* Check for a continuation line */
- if (msgbuf[h + 1] == ' ') {
- /* Merge continuation line */
- h++;
+ break;
+ /* Check for a continuation line */
+ if (msgbuf[h + 1] == ' ' || msgbuf[h + 1] == '\t') {
+ /* Merge continuation line */
+ h++;
+ continue;
+ }
+ /* Propagate LF and start new line */
+ msgbuf[t++] = msgbuf[h++];
+ lws = 0;
continue;
}
- /* Propagate LF and start new line */
- msgbuf[t++] = msgbuf[h++];
- lws = 0;
- continue;
- }
-
- if (msgbuf[h] == ' ' || msgbuf[h] == '\t') {
- if (lws) {
- h++;
+ if (msgbuf[h] == ' ' || msgbuf[h] == '\t') {
+ if (lws) {
+ h++;
+ continue;
+ }
+ msgbuf[t++] = msgbuf[h++];
+ lws = 1;
continue;
}
msgbuf[t++] = msgbuf[h++];
- lws = 1;
- continue;
- }
- msgbuf[t++] = msgbuf[h++];
- if (lws)
- lws = 0;
+ if (lws)
+ lws = 0;
}
msgbuf[t] = '\0';
return t;
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Re: [Asterisk-Users] SIP register problem
You're welcome. It's been submitted. Cyrille Demaret wrote: Hi, Thank you, it's working now! Do you think that this patch will be included in the next cvs versions? Sincerely, Cyrille Demaret ___ 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] SIP register problem
Hi, I'm trying to register Asterisk with my sip provider but I've a problem. Here's the log file : REGISTER sip:sip.aquanta.com SIP/2.0 Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK2810ee43 From: sip:[EMAIL PROTECTED];tag=as2e43c573 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 105 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 (no NAT) to 212.3.244.8:5060 ziki5*CLI Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK2810ee43 From: sip:[EMAIL PROTECTED];tag=as2e43c573 To: sip:[EMAIL PROTECTED]; tag=lPaMk9ATvmCVvvGpOPJFkg Call-ID: [EMAIL PROTECTED] Cseq: 105 REGISTER Date: Tue, 16 Nov 2004 17:23:15 GMT Server: SIPquest-SIP-Server/2.2 Content-Length: 0 WWW-Authenticate: Digest realm=sip.aquanta.com, nonce=hDeaQQR5yOCHgx8YjKqI5zOcB1w=, stale=FALSE, algorithm=MD5, qop=auth 14 headers, 0 lines 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.aquanta.com SIP/2.0 Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK06fcb9f0 From: sip:[EMAIL PROTECTED];tag=as2e43c573 To: sip:[EMAIL PROTECTED];tag=lPaMk9ATvmCVvvGpOPJFkg Call-ID: [EMAIL PROTECTED] CSeq: 106 REGISTER User-Agent: Asterisk PBX Authorization: Digest username=0032123456789, realm=sip.aquanta.com, algorithm=MD5, uri=sip:sip.aquanta.com, nonce=, response=854c3960f84b454af9d25fcfdb0aaee4, opaque= Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 (no NAT) to 212.3.244.8:5060 ziki5*CLI Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK06fcb9f0 From: sip:[EMAIL PROTECTED];tag=as2e43c573 To: sip:[EMAIL PROTECTED];tag=lPaMk9ATvmCVvvGpOPJFkg Call-ID: [EMAIL PROTECTED] Cseq: 106 REGISTER Date: Tue, 16 Nov 2004 17:23:15 GMT Server: SIPquest-SIP-Server/2.2 Content-Length: 0 WWW-Authenticate: Digest realm=sip.aquanta.com, nonce=hDeaQQR5yOCHgx8YjKqI5zOcB1w=, stale=TRUE, algorithm=MD5, qop=auth 14 headers, 0 lines 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.aquanta.com SIP/2.0 Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK05c0559b From: sip:[EMAIL PROTECTED];tag=as2e43c573 To: sip:[EMAIL PROTECTED];tag=lPaMk9ATvmCVvvGpOPJFkg Call-ID: [EMAIL PROTECTED] CSeq: 107 REGISTER User-Agent: Asterisk PBX Authorization: Digest username=0032123456789, realm=sip.aquanta.com, algorithm=MD5, uri=sip:sip.aquanta.com, nonce=, response=854c3960f84b454af9d25fcfdb0aaee4, opaque= Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 (no NAT) to 212.3.244.8:5060 ziki5*CLI Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK05c0559b From: sip:[EMAIL PROTECTED];tag=as2e43c573 To: sip:[EMAIL PROTECTED];tag=lPaMk9ATvmCVvvGpOPJFkg Call-ID: [EMAIL PROTECTED] Cseq: 107 REGISTER Date: Tue, 16 Nov 2004 17:23:15 GMT Server: SIPquest-SIP-Server/2.2 Content-Length: 0 WWW-Authenticate: Digest realm=sip.aquanta.com, nonce=hDeaQQR5yOCHgx8YjKqI5zOcB1w=, stale=TRUE, algorithm=MD5, qop=auth 14 headers, 0 lines Nov 16 18:19:49 NOTICE[26352]: chan_sip.c:6768 handle_response: Failed to authenticate on REGISTER to 'sip:[EMAIL PROTECTED];tag=as2e43c573' Destroying call '[EMAIL PROTECTED]' It's seems that the nonce is not sended back to the server with de REGISTER packet. I don't know if it's due to the \n in the WWW-Authenticate response. Do you think it's possible? Thank you, Sincerely, Cyrille Demaret ___ 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] SIP register problem
Is the SIPquest server sending the 401 Unauthorized message verbatim as you printed it here? I.e. is the WWW-Authentcate header broken up into several lines like that? If so, how man spaces are actually at the beginning of each new line? Continuation lines are allowed in SIP, but I think it's only one space at the beginning of each new line. To make Asterisk parse this correctly you need to turn on pedantic=yes It's silly that Asterisk doesn't turn this header parsing on by default, no reason not to. If this fixes your problem, you should edit the sip source code and remove the if clause before calling lws2sws(blah blah) there is only one instance in the code. pedantic=yes may break other things for you. Cyrille Demaret wrote: Hi, I'm trying to register Asterisk with my sip provider but I've a problem. Here's the log file : REGISTER sip:sip.aquanta.com SIP/2.0 Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK2810ee43 From: sip:[EMAIL PROTECTED];tag=as2e43c573 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 105 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 (no NAT) to 212.3.244.8:5060 ziki5*CLI Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK2810ee43 From: sip:[EMAIL PROTECTED];tag=as2e43c573 To: sip:[EMAIL PROTECTED]; tag=lPaMk9ATvmCVvvGpOPJFkg Call-ID: [EMAIL PROTECTED] Cseq: 105 REGISTER Date: Tue, 16 Nov 2004 17:23:15 GMT Server: SIPquest-SIP-Server/2.2 Content-Length: 0 WWW-Authenticate: Digest realm=sip.aquanta.com, nonce=hDeaQQR5yOCHgx8YjKqI5zOcB1w=, stale=FALSE, algorithm=MD5, qop=auth 14 headers, 0 lines 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.aquanta.com SIP/2.0 Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK06fcb9f0 From: sip:[EMAIL PROTECTED];tag=as2e43c573 To: sip:[EMAIL PROTECTED];tag=lPaMk9ATvmCVvvGpOPJFkg Call-ID: [EMAIL PROTECTED] CSeq: 106 REGISTER User-Agent: Asterisk PBX Authorization: Digest username=0032123456789, realm=sip.aquanta.com, algorithm=MD5, uri=sip:sip.aquanta.com, nonce=, response=854c3960f84b454af9d25fcfdb0aaee4, opaque= Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 (no NAT) to 212.3.244.8:5060 ziki5*CLI Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK06fcb9f0 From: sip:[EMAIL PROTECTED];tag=as2e43c573 To: sip:[EMAIL PROTECTED];tag=lPaMk9ATvmCVvvGpOPJFkg Call-ID: [EMAIL PROTECTED] Cseq: 106 REGISTER Date: Tue, 16 Nov 2004 17:23:15 GMT Server: SIPquest-SIP-Server/2.2 Content-Length: 0 WWW-Authenticate: Digest realm=sip.aquanta.com, nonce=hDeaQQR5yOCHgx8YjKqI5zOcB1w=, stale=TRUE, algorithm=MD5, qop=auth 14 headers, 0 lines 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.aquanta.com SIP/2.0 Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK05c0559b From: sip:[EMAIL PROTECTED];tag=as2e43c573 To: sip:[EMAIL PROTECTED];tag=lPaMk9ATvmCVvvGpOPJFkg Call-ID: [EMAIL PROTECTED] CSeq: 107 REGISTER User-Agent: Asterisk PBX Authorization: Digest username=0032123456789, realm=sip.aquanta.com, algorithm=MD5, uri=sip:sip.aquanta.com, nonce=, response=854c3960f84b454af9d25fcfdb0aaee4, opaque= Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 (no NAT) to 212.3.244.8:5060 ziki5*CLI Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK05c0559b From: sip:[EMAIL PROTECTED];tag=as2e43c573 To: sip:[EMAIL PROTECTED];tag=lPaMk9ATvmCVvvGpOPJFkg Call-ID: [EMAIL PROTECTED] Cseq: 107 REGISTER Date: Tue, 16 Nov 2004 17:23:15 GMT Server: SIPquest-SIP-Server/2.2 Content-Length: 0 WWW-Authenticate: Digest realm=sip.aquanta.com, nonce=hDeaQQR5yOCHgx8YjKqI5zOcB1w=, stale=TRUE, algorithm=MD5, qop=auth 14 headers, 0 lines Nov 16 18:19:49 NOTICE[26352]: chan_sip.c:6768 handle_response: Failed to authenticate on REGISTER to 'sip:[EMAIL PROTECTED];tag=as2e43c573' Destroying call '[EMAIL PROTECTED]' It's seems that the nonce is not sended back to the server with de REGISTER packet. I don't know if it's due to the \n in the WWW-Authenticate response. Do you think it's possible? Thank you, Sincerely, Cyrille Demaret ___ 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 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] SIP register problem
Hi, Thank you for your answer. Unfortunately, pedantic does not help. Here's an ngrep dump of the corresponding packet. U 212.3.244.8:5060 - 123.123.123.123:5060 53 49 50 2f 32 2e 30 2034 30 31 20 55 6e 61 75SIP/2.0 401 Unau 74 68 6f 72 69 7a 65 640d 0a 56 69 61 3a 20 53thorized..Via: S 49 50 2f 32 2e 30 2f 5544 50 20 32 31 33 2e 31IP/2.0/UDP 213.1 38 36 2e 35 34 2e 37 353a 35 30 36 30 3b 62 7286.54.75:5060;br 61 6e 63 68 3d 7a 39 6847 34 62 4b 32 34 62 64anch=z9hG4bK24bd 32 34 32 31 0d 0a 46 726f 6d 3a 20 3c 73 69 702421..From: sip 3a 30 30 33 32 32 34 3033 30 37 39 36 30 40 73:[EMAIL PROTECTED] 69 70 2e 61 71 75 61 6e74 61 2e 63 6f 6d 3e 3bip.aquanta.com; 74 61 67 3d 61 73 32 3338 65 31 66 32 39 0d 0atag=as238e1f29.. 54 6f 3a 20 3c 73 69 703a 30 30 33 32 32 34 30To: sip:0032240 33 30 37 39 36 30 40 7369 70 2e 61 71 75 61 6e[EMAIL PROTECTED] 74 61 2e 63 6f 6d 3e 3b20 74 61 67 3d 39 6c 21ta.com; tag=9l! 68 43 32 5a 71 38 5a 6657 47 39 21 70 57 47 63hC2Zq8ZfWG9!pWGc 4a 46 41 0d 0a 43 61 6c6c 2d 49 44 3a 20 36 62JFA..Call-ID: 6b 38 62 34 35 36 37 33 3237 62 32 33 63 36 36 348b4567327b23c664 33 63 39 38 36 39 36 3633 33 34 38 37 33 40 32[EMAIL PROTECTED] 31 33 2e 31 38 36 2e 3534 2e 37 35 0d 0a 43 7313.186.54.75..Cs 65 71 3a 20 31 30 32 2052 45 47 49 53 54 45 52eq: 102 REGISTER 0d 0a 44 61 74 65 3a 2054 75 65 2c 20 31 36 20..Date: Tue, 16 4e 6f 76 20 32 30 30 3420 31 38 3a 33 38 3a 33Nov 2004 18:38:3 39 20 47 4d 54 0d 0a 5365 72 76 65 72 3a 20 539 GMT..Server: S 49 50 71 75 65 73 74 2d53 49 50 2d 53 65 72 76IPquest-SIP-Serv 65 72 2f 32 2e 32 0d 0a43 6f 6e 74 65 6e 74 2der/2.2..Content- 4c 65 6e 67 74 68 3a 2030 0d 0a 57 57 57 2d 41Length: 0..WWW-A 75 74 68 65 6e 74 69 6361 74 65 3a 20 44 69 67uthenticate: Dig 65 73 74 20 72 65 61 6c6d 3d 22 73 69 70 2e 61est realm=sip.a 71 75 61 6e 74 61 2e 636f 6d 22 2c 0d 0a 09 6equanta.com,...n 6f 6e 63 65 3d 22 4c 306d 61 51 56 75 67 48 54once=L0maQVugHT 6e 7a 66 33 42 61 56 4958 57 72 74 77 58 39 4anzf3BaVIXWrtwX9J 6b 3d 22 2c 0d 0a 09 7374 61 6c 65 3d 46 41 4ck=,...stale=FAL 53 45 2c 0d 0a 09 61 6c67 6f 72 69 74 68 6d 3dSE,...algorithm= 4d 44 35 2c 0d 0a 09 716f 70 3d 22 61 75 74 68MD5,...qop=auth 22 0d 0a 0d 0a The characters at the beginning of the WWW-Authenticate lines are CR LF and TAB. Sincerely, Cyrille Demaret -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Karl Brose Envoyé : mardi 16 novembre 2004 19:05 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] SIP register problem Is the SIPquest server sending the 401 Unauthorized message verbatim as you printed it here? I.e. is the WWW-Authentcate header broken up into several lines like that? If so, how man spaces are actually at the beginning of each new line? Continuation lines are allowed in SIP, but I think it's only one space at the beginning of each new line. To make Asterisk parse this correctly you need to turn on pedantic=yes It's silly that Asterisk doesn't turn this header parsing on by default, no reason not to. If this fixes your problem, you should edit the sip source code and remove the if clause before calling lws2sws(blah blah) there is only one instance in the code. pedantic=yes may break other things for you. Cyrille Demaret wrote: Hi, I'm trying to register Asterisk with my sip provider but I've a problem. Here's the log file : REGISTER sip:sip.aquanta.com SIP/2.0 Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK2810ee43 From: sip:[EMAIL PROTECTED];tag=as2e43c573 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 105 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 (no NAT) to 212.3.244.8:5060 ziki5*CLI Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK2810ee43 From: sip:[EMAIL PROTECTED];tag=as2e43c573 To: sip:[EMAIL PROTECTED]; tag=lPaMk9ATvmCVvvGpOPJFkg Call-ID: [EMAIL PROTECTED] Cseq: 105 REGISTER Date: Tue, 16 Nov 2004 17:23:15 GMT Server: SIPquest-SIP-Server/2.2 Content-Length: 0 WWW-Authenticate: Digest realm=sip.aquanta.com, nonce=hDeaQQR5yOCHgx8YjKqI5zOcB1w=, stale=FALSE, algorithm=MD5, qop=auth 14 headers, 0 lines 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.aquanta.com SIP/2.0 Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK06fcb9f0 From: sip:[EMAIL PROTECTED];tag=as2e43c573 To: sip:[EMAIL PROTECTED];tag=lPaMk9ATvmCVvvGpOPJFkg Call-ID: [EMAIL PROTECTED] CSeq: 106 REGISTER User-Agent: Asterisk PBX
Re: [Asterisk-Users] SIP register problem
Just checked the RFC, and it does say that a tab is acceptable.
SIP header field values can be folded onto multiple lines if the
continuation line begins with a space or horizontal tab. All linear
white space, including folding, has the same semantics as SP. A
recipient MAY replace any linear white space with a single SP before
interpreting the field value or forwarding the message downstream.
This is intended to behave exactly as HTTP/1.1 as described in RFC
2616 [8]. The SWS construct is used when linear white space is
optional, generally between tokens and separators.
LWS = [*WSP CRLF] 1*WSP ; linear whitespace
SWS = [LWS] ; sep whitespace
This means that Asterisk is broken.
Please try the patch attached to this e-mail
filename is lws2sws.patch.txt
Just in case, here is how to:
copy the patch to your Asterisk source code channels directory:
(assuming that /usr/src/asterisk is your base)
cp lws2sws.patch.txt /usr/src/asterisk/channel
cd /usr/src/asterisk/channels
It would be safer to make a backup copy of chan_sip.c before patching.
patch chan_sip.c lws2sws.patch.txt
cd ..
then recompile asterisk and install the new chan_sip.so file
test with pedantic=yes
unless you removed the if clause
--- chan_sip.c 2004-11-16 14:14:42.0 -0500
+++ chan_sip.c_lws 2004-11-16 14:18:44.0 -0500
@@ -2499,33 +2499,32 @@
}
/* Check for end-of-line */
if (msgbuf[h] == '\n') {
- /* Check for end-of-message */
+ /* Check for end-of-message */
if (h + 1 == len)
- break;
- /* Check for a continuation line */
- if (msgbuf[h + 1] == ' ') {
- /* Merge continuation line */
- h++;
+ break;
+ /* Check for a continuation line */
+ if (msgbuf[h + 1] == ' ' || msgbuf[h + 1] == '\t') {
+ /* Merge continuation line */
+ h++;
+ continue;
+ }
+ /* Propagate LF and start new line */
+ msgbuf[t++] = msgbuf[h++];
+ lws = 0;
continue;
}
- /* Propagate LF and start new line */
- msgbuf[t++] = msgbuf[h++];
- lws = 0;
- continue;
- }
-
- if (msgbuf[h] == ' ' || msgbuf[h] == '\t') {
- if (lws) {
- h++;
+ if (msgbuf[h] == ' ' || msgbuf[h] == '\t') {
+ if (lws) {
+ h++;
+ continue;
+ }
+ msgbuf[t++] = msgbuf[h++];
+ lws = 1;
continue;
}
msgbuf[t++] = msgbuf[h++];
- lws = 1;
- continue;
- }
- msgbuf[t++] = msgbuf[h++];
- if (lws)
- lws = 0;
+ if (lws)
+ lws = 0;
}
msgbuf[t] = '\0';
return t;
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[Asterisk-Users] SIP Register with Huawei equipment HELP
Dear all, I'm trying to perform a register = userid:[EMAIL PROTECTED] for incoming SIP calls from a provider and is not going too well. Pls refer to the attached debug log and hopefully someone can help me out. I believe that they're using Huawei equipment and the same userid/password pair is working fine with X-lite. What I noticed is that the nonce field on the reply is blank and I'm not sure if that's the problem and how I can fix it. I've included the register line with userid:[EMAIL PROTECTED]/context in the sip.conf already Please help Reliably Transmitting: REGISTER sip:***.***.***.*** SIP/2.0 Via: SIP/2.0/UDP ***.***.84.142:5060;branch=z9hG4bK3dc4c27c From: sip:[EMAIL PROTECTED];tag=as3e3d1bc2 To: sip:[EMAIL PROTECTED];tag=1159d164 Call-ID: [EMAIL PROTECTED] CSeq: 110 REGISTER User-Agent: X-Lite release 1103m Authorization: Digest username=21257343, realm=huawei.com, algorithm=MD5, uri=sip:***.*.***.***, nonce=, response=5b36c5c5997a2948cc0c8cb6c6d65711, opaque= Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 (no NAT) to ***.*.***.***:5060 alansip3*CLI Sip read: SIP/2.0 401 Unauthorized From: sip:[EMAIL PROTECTED];tag=as3e3d1bc2 To: sip:[EMAIL PROTECTED];tag=1159d164 CSeq: 110 REGISTER Call-ID: [EMAIL PROTECTED] Via: SIP/2.0/UDP ***.***.**.142:5060;branch=z9hG4bK3dc4c27c WWW-Authenticate: Digest realm=huawei.com, nonce=f343463ed7c33c6f233646513b01fa54,stale=false,algorithm=MD5 Content-Length: 0 9 headers, 0 lines Destroying call '[EMAIL PROTECTED]' 11 headers, 0 lines Reliably Transmitting: REGISTER sip:***.*.***.*** SIP/2.0 Via: SIP/2.0/UDP ***.***.**.***:5060;branch=z9hG4bK0f47ded1 From: sip:[EMAIL PROTECTED];tag=as752ce90a To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 111 REGISTER User-Agent: X-Lite release 1103m Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 (no NAT) to ***.*.***.226:5060 alansip3*CLI Sip read: SIP/2.0 401 Unauthorized From: sip:[EMAIL PROTECTED];tag=as752ce90a To: sip:[EMAIL PROTECTED];tag=2271772e CSeq: 111 REGISTER Call-ID: [EMAIL PROTECTED] Via: SIP/2.0/UDP ***.***.**.142:5060;branch=z9hG4bK0f47ded1 WWW-Authenticate: Digest realm=huawei.com, nonce=4bd6abe2e1266bbae8075eb1f7f733e7,stale=false,algorithm=MD5 Content-Length: 0 ___ 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] SIP REGISTER -- Via 0.0.0.0:5060 -- Oooops?!
On a newly built Asterisk 1.02 system I am getting a rather strange SIP register message ... REGISTER sip:ispvoip-.ocn.ne.jp SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK3d1d0887 and further down ... Contact: sip:[EMAIL PROTECTED] Event: registration the register directive in sip.conf looks like this: register = 05012345678:passwd:[EMAIL PROTECTED]/05012345678 What's this 0.0.0.0 address doing the SIP message? Is this another bug? -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ 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] SIP REGISTER -- Asterisk non-compliant or is it the provider?
On Sun, 07 Nov 2004 09:47:33 +0200, Gilad Ben-Yossef
[EMAIL PROTECTED] wrote:
I don't claim to understand the code at all, but what little I think I
understand from it makes me believe this is not the change you're
looking for.
The differences between chan_sip.c of the version on the production
system and today's CVS are so major and plenty, that I felt it was a
better idea to try to make sense of the older code.
I put in ast_log calls all over the place to see what various
variables are at which stage and that way I managed to track down the
problem to function build_reply_digest, which come to think of it
makes a lot of sense just by looking at the name of it ;-)
here are the changes I made ...
static int build_reply_digest(struct sip_pvt *p, char* orig_header,
char* digest, int digest_len)
{
char a1[256];
char a2[256];
char a1_hash[256];
char a2_hash[256];
char resp[256];
char resp_hash[256];
char uri[256] = ;
char cnonce[80];
char *uname;
- if (strlen(p-username))
- uname = p-username;
-else
-uname = p-peername;
+
+uname = p-peername; /* uname should always be peername */
and this fixes the problem.
The reason being that sip_transmit_register already contains the code
that checks which value to put into p-peername and it also populates
p-username. Consequently the else branch is never executed and that
is why the authuser value is never used in the digest.
Unfortunately, but not unexpected, Asterisk's attempts to register
with NTT's SIP service still fail despite the REGISTER message and the
digest now conforming to what NTT's staff claims to be proper. But
what else would you expect of NTT?!
rgds
benjk
--
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Tokyo, Japan.
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Re: [Asterisk-Users] SIP REGISTER -- Asterisk non-compliant or is it the provider?
Hello, --- Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] wrote: The provider's support staff says that the userid in 'From: sip:[EMAIL PROTECTED] ...' should be the phone number while the userid in 'Authorization: Digest username=userid...' of the same REGISTER message should be the account name. I am not sure if this can be right. At least, whether compliant or not, it would seem that such a REGISTER message cannot be constructed by Asterisk. The From header field in a SIP message is used to identify the initiator of the request. AFAIK, the 050 in the From header acts as a display name. It can be used to determine the processing rules by other SIP entities. For example, PSTN gateways can use it to determine if it is a valid callerid or not (For INVITEs). The Auth credentials in the Request can be different. It should be the username and password for the account that the provider has given you. I hope others here will give a better explanation on this... - is it in compliance with RFC3261 to have different values in the From and the Digest username fields? I think yes. Our UACs register with SIP Express Router with different values in these fields. Attached below is an ngrep trace of such a request processing. - can Asterisk construct such a REGISTER message? Sorry. I am not sure about this. Regards, Girish Here's the trace: REGISTER sip:XXX.XXX.XXX.XXX SIP/2.0. Via: SIP/2.0/UDP 192.168.68.24:12894. Max-Forwards: 70. From: sip:[EMAIL PROTECTED];tag=699fdcaedde144a68097a86c5ec00655;epid=7de641515e. To: sip:[EMAIL PROTECTED]. Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER. Contact: sip:192.168.68.24:12894;methods=INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER. User-Agent: RTC/1.2.4949 Event: registration. Allow-Events: presence. Content-Length: 0. . SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.68.24:12894;rport=4061;received=XX.XX.XX.XXX. From: sip:[EMAIL PROTECTED];tag=699fdcaedde144a68097a86c5ec00655;epid=7de641515e. To: sip:[EMAIL PROTECTED];tag=b27e1a1d33761e85846fc98f5f3a7e58.d1fd. Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER. WWW-Authenticate: Digest realm=XXX.XXX.XXX.XXX, nonce=418cd4b94a2ea94004191fd618b7bbb7041f8f40, qop=auth. Server: SIP EXpress Router (0.8.14 (i386/linux)). Content-Length: 0. . . REGISTER sip:XXX.XXX.XXX.XXX SIP/2.0. Via: SIP/2.0/UDP 192.168.68.24:12894. Max-Forwards: 70. From: sip:[EMAIL PROTECTED];tag=699fdcaedde144a68097a86c5ec00655;epid=7de641515e. To: sip:[EMAIL PROTECTED]. Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER. Contact: sip:192.168.68.24:12894;methods=INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER. User-Agent: RTC/1.2.4949 Authorization: Digest username=girish-smarttest-com, realm=XXX.XXX.XXX.XXX, qop=auth, algorithm=md5, uri=sip:XXX.XXX.XXX.XXX, nonce=418cd4b94a2ea94004191fd618b7bbb7041f8f40, nc=0001, cnonce=15420645451543562242791578613325, response=89d8531e598629b022230df475b5bb65. Event: registration. Allow-Events: presence. Content-Length: 0. . . SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.68.24:12894;rport=4061;received=XX.XX.XX.XXX. From: sip:[EMAIL PROTECTED];tag=699fdcaedde144a68097a86c5ec00655;epid=7de641515e. To: sip:[EMAIL PROTECTED];tag=b27e1a1d33761e85846fc98f5f3a7e58.d1fd. Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER. Expires: 120. Contact: sip:XX.XX.XX.XXX:4061;q=0.00;expires=3600. Server: SIP EXpress Router (0.8.14 (i386/linux)). Content-Length: 0. __ Do you Yahoo!? Check out the new Yahoo! Front Page. www.yahoo.com ___ 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] SIP REGISTER -- Asterisk non-compliant or is it the provider?
On Sat, 6 Nov 2004 06:33:15 -0800 (PST), Girish Gopinath [EMAIL PROTECTED] wrote: AFAIK, the 050 in the From header acts as a display name. It can be used to determine the processing rules by other SIP entities. [SNIP] The Auth credentials in the Request can be different. Thanks, this was important to know because I couldn't seem to verify this using various SIP devices. I hope others here will give a better explanation on this... More important than explanations is to know the fact that the two can in fact be different. The question now is how can Asterisk's sip implementation be fixed so that it can construct such REGISTER messages. If this remains unsupported, providers will actively use it to lock Asterisk out. The Japanese telecom monopoly NTT already does so as this example shows. And since most ISPs over here act as satellites for NTT, even if you change providers, you are likely to end up with NTT again. - can Asterisk construct such a REGISTER message? Sorry. I am not sure about this. given the very limited syntax for the register directlive, register = [EMAIL PROTECTED] OR register = [EMAIL PROTECTED]/contactid I can't really see how to change the From field independently of the digest username. but I would be more than happy to learn of any workarounds or patches that would allow this. thanks rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ 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] SIP REGISTER -- Asterisk non-compliant or is it the provider?
The syntax for the register command is register=username:secret:[EMAIL PROTECTED]:port/extension Benjamin on Asterisk Mailing Lists wrote: On Sat, 6 Nov 2004 06:33:15 -0800 (PST), Girish Gopinath [EMAIL PROTECTED] wrote: AFAIK, the 050 in the From header acts as a display name. It can be used to determine the processing rules by other SIP entities. [SNIP] The Auth credentials in the Request can be different. Thanks, this was important to know because I couldn't seem to verify this using various SIP devices. I hope others here will give a better explanation on this... More important than explanations is to know the fact that the two can in fact be different. The question now is how can Asterisk's sip implementation be fixed so that it can construct such REGISTER messages. If this remains unsupported, providers will actively use it to lock Asterisk out. The Japanese telecom monopoly NTT already does so as this example shows. And since most ISPs over here act as satellites for NTT, even if you change providers, you are likely to end up with NTT again. - can Asterisk construct such a REGISTER message? Sorry. I am not sure about this. given the very limited syntax for the register directlive, register = [EMAIL PROTECTED] OR register = [EMAIL PROTECTED]/contactid I can't really see how to change the From field independently of the digest username. but I would be more than happy to learn of any workarounds or patches that would allow this. thanks rgds benjk ___ 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] SIP REGISTER -- Asterisk non-compliant or is it the provider?
On Sat, 06 Nov 2004 12:00:57 -0500, Karl Brose [EMAIL PROTECTED] wrote: The syntax for the register command is register=username:secret:[EMAIL PROTECTED]:port/extension Trouble is though that this does not have any effect on the username in the digest. Whatever it is intended for, it's not doing anything to untie the From field from the Digest username field. but thanks anyway rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ 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] SIP REGISTER -- Asterisk non-compliant or is it the provider?
Benjamin on Asterisk Mailing Lists wrote: On Sat, 06 Nov 2004 12:00:57 -0500, Karl Brose [EMAIL PROTECTED] wrote: The syntax for the register command is register=username:secret:[EMAIL PROTECTED]:port/extension Trouble is though that this does not have any effect on the username in the digest. Whatever it is intended for, it's not doing anything to untie the From field from the Digest username field. Well, it looks like the digest is being built with authname in build_reply_digest(). It's using sip_pvt.authname which get's initialized to the *username* in create_addr, but is then being copied over by the authuser field from the SIP registery which looks like it should get initalized by sip_register(). But obviously, it doesn't :-) Hope this helps in any way, Gilad ___ 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] SIP REGISTER -- Asterisk non-compliant or is it the provider?
On Sat, 06 Nov 2004 20:32:05 +0200, Gilad Ben-Yossef [EMAIL PROTECTED] wrote: Well, it looks like the digest is being built with authname in build_reply_digest(). This seems to indicate that it was indeed the intend of the implementor to use authname in the digest's username field and consequently that this is a bug. I have therefore filed a bug report for this now ... http://bugs.digium.com/bug_view_page.php?bug_id=0002802 let's see what the response will be and at the very least we will know for sure what the intend of this actually is. It's using sip_pvt.authname which get's initialized to the *username* in create_addr, but is then being copied over by the authuser field from the SIP registery which looks like it should get initalized by sip_register(). But obviously, it doesn't :-) Hope this helps in any way I wish I could just change this in the code myself but the code is just so all over the place, I can't seem to get a grip on it and so I don't know where to make changes. Then again, I am not really a developer (or at least not anymore), so it may just be me. Mind you, back in the day when I was doing this sort of thing, code looked much more structured, much cleaner and an outside party was able to apply simple fixes, like this one would seem to be. I guess in another 20 years from now, it will require a PhD to change a filename ;-) anyway, thanks for your assistance. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ 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] SIP REGISTER -- Asterisk non-compliant or is it the provider?
On Sun, 7 Nov 2004 10:00:25 +0900, Benjamin on Asterisk Mailing Lists
[EMAIL PROTECTED] wrote:
http://bugs.digium.com/bug_view_page.php?bug_id=0002802
Mark has fixed this in the most recent CVS (Wow! that was fast!).
However, I will need this for a production system that cannot be
upgraded (Zaptel drivers have become extremely unstable recently when
used on any kind of Japanese analog line, PSTN or otherwise), so I
would like to apply the changes to the version running on that system.
In order to do so, I needed to indentify what exactly has changed and
so I ran a diff on the latest CVS versus a few hours back.
It would seem that there is only a single line which has changed in
respect of SIP reigstration ...
*** static int transmit_register(struct sip_
*** 4054,4059
--- 4055,4061
if (!ast_strlen_zero(r-username)) {
strncpy(p-peername, r-username,
sizeof(p-peername)-1);
strncpy(p-authname, r-username,
sizeof(p-authname)-1);
+ strncpy(p-fromuser, r-username,
sizeof(p-fromuser)-1);
}
... and I am trying to make sense of this so as to be confident to
apply the change to the earlier version. Is this likely to be what
fixed this bug or did I mess up with the diff? I'd appreciate if
somebody who understands the code could comment on this, please.
thanks
rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.
NB: Spam filters in place. Messages unrelated to the * mailing lists
may get trashed.
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Re: [Asterisk-Users] SIP REGISTER -- Asterisk non-compliant or is it the provider?
Benjamin on Asterisk Mailing Lists wrote:
It would seem that there is only a single line which has changed in
respect of SIP reigstration ...
*** static int transmit_register(struct sip_
*** 4054,4059
--- 4055,4061
if (!ast_strlen_zero(r-username)) {
strncpy(p-peername, r-username,
sizeof(p-peername)-1);
strncpy(p-authname, r-username,
sizeof(p-authname)-1);
+ strncpy(p-fromuser, r-username,
sizeof(p-fromuser)-1);
}
... and I am trying to make sense of this so as to be confident to
apply the change to the earlier version. Is this likely to be what
fixed this bug or did I mess up with the diff? I'd appreciate if
I don't claim to understand the code at all, but what little I think I
understand from it makes me believe this is not the change you're
looking for.
Cheers,
Gilad
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence. A name you can trust(tm)
Web: http://codefidence.com | SIP: [EMAIL PROTECTED]
Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643
I am Jack's Overwritten Stack Pointer
-- Hackers Club, the movie
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[Asterisk-Users] SIP REGISTER -- Asterisk non-compliant or is it the provider?
I am trying to get Asterisk to register with a SIP provider who officially only supports ATAs of the incumbent telephone monopolist over here. I have so far been lucky enough to get them to ***respond*** to my requests for information on what parameters need changing in the REGISTER messages in order to successfully register. Confusion arises from the fact that there are two different user IDs, one is a telephone number associated with the SIP account, say 050, the other is an account name, say DUDEDUDE. The provider's support staff says that the userid in 'From: sip:[EMAIL PROTECTED] ...' should be the phone number while the userid in 'Authorization: Digest username=userid...' of the same REGISTER message should be the account name. I am not sure if this can be right. At least, whether compliant or not, it would seem that such a REGISTER message cannot be constructed by Asterisk. If I use register = [EMAIL PROTECTED] then then Asterisk will construct a REGISTER message with the phone number in both the From field and the username field in the digest. If I use register = [EMAIL PROTECTED] then Asterisk uses the account name in both places. The optional parameter at the end of the register directive only seems to have an effect on the contact field, ie register = [EMAIL PROTECTED]/123456789 will still put fred into both the From field and the username field of the digest while 123456789 will show up in the Contact field only. Can somebody comment on this? - is it in compliance with RFC3261 to have different values in the From and the Digest username fields? - can Asterisk construct such a REGISTER message? To illustrate this a bit further, here is an excerpt from a session transscript ... REGISTER sip:ispvoip-.ocn.ne.jp SIP/2.0 Via: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK6920ea27 From: sip:[EMAIL PROTECTED];tag=as040e1159 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 961 REGISTER Authorization: Digest username=050, realm=ocn.ne.jp, algorithm=MD5, uri=sip:210.9.9.9, nonce=1099640598, response=a9d877017f24bb624cdc1a39a8a73b4c, opaque= Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration SIP/2.0 401 Unauthorized v: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK6920ea27 From: sip:[EMAIL PROTECTED];tag=as040e1159 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 961 REGISTER Expires: 120 Event: registration Date: Fri, 05 Nov 2004 08:17:48 GMT WWW-Authenticate: Digest realm=ocn.ne.jp, domain=sip:210.9.9.9, nonce=1099640598, opaque=, stale=FALSE, algorithm=MD5 The provider's support staff suggested that their server expected to see a REGISTER message like the following: REGISTER sip:ispvoip-.ocn.ne.jp SIP/2.0 Via: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK6920ea27 From: sip:[EMAIL PROTECTED];tag=as040e1159 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 961 REGISTER Authorization: Digest username=DUDEDUDE, realm=ocn.ne.jp, algorithm=MD5, uri=sip:210.9.9.9, nonce=1099640598, response=a9d877017f24bb624cdc1a39a8a73b4c, opaque= Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Unfortunately, I have not been able to get Asterisk to construct a message like the above. any hints appreciated. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ 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] SIP register and unregister events via Manager API
And I would very much like feedback on those - are they useful? If they are, I'll backport to chan_sip. I find them useful (for the HTML based block LED field display of DeStar). Today I even thought about writing a patch that sends the available/unavailable messages from the quality=... code to the Manager. ___ 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] SIP register and unregister events via Manager API
And I would very much like feedback on those - are they useful? Oh, I just found out by looking at the source code that there are database entries SIP/Registry. I think the used database entries is something that is currently under-documented ... :-) ___ 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] SIP register and unregister events via Manager API
At 9:41 AM +0200 on 7/23/04, Holger Schurig wrote: And I would very much like feedback on those - are they useful? If they are, I'll backport to chan_sip. I find them useful (for the HTML based block LED field display of DeStar). Today I even thought about writing a patch that sends the available/unavailable messages from the quality=... code to the Manager. Would it be just as useful to actually have the numeric values readable to the Manager, as well as the binary (available/unavailable) status of results from quality= statements? You could (theoretically) color code an icon through some type of good-to-bad spectrum depending on how fast it was responding to the quality queries. More eye candy for manager-types. JT ___ 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] SIP register and unregister events via Manager API
Holger Schurig wrote: And I would very much like feedback on those - are they useful? Oh, I just found out by looking at the source code that there are database entries SIP/Registry. I think the used database entries is something that is currently under-documented ... :-) If you simply type database show (and nothing more) on the CLI you'll discover even more if you have IAX clients registered... :-) Cheers, Philipp ___ 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] SIP register and unregister events via Manager API
The patch tries to send the time as well, but it fails. There are some problems currently: I applied the path from bug 2117. After this I got some events: Event: SIPPeerStatus Peer: weckhardt Status: reachable Time: 55 Event: SIPPeerRegistration Peername: dnarotam Status: Offline I think we have several problems here. Once it's Peer:, the other time it's Peername. Also, I don't like the name of the event. It should just be PeerStatus and PeerRegistration, because we might add something to IAX2 as well. So I'd suggest to do it this way: Event: PeerStatus Peer: SIP/weckhardt PeerStatus: reachable Time: 55 Event: PeerRegistration Peer: SIP/dnarotam Channel: Offline This way, other channels can send the same events, just with PeerStatus: IAX2/qtiax. I guess I'll redo the path and re-submit it to bugs.digium.com. ___ 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] SIP register and unregister events via Manager API
Okay, I have finished my patch. With qualify=yes in sip.conf it looks like this: - # telnet 127.0.0.1 5038 Trying 127.0.0.1... Connected to localhost.localdomain (127.0.0.1). Escape character is '^]'. Asterisk Call Manager/1.0 Action: Login Username: destar Secret: secret Response: Success Message: Authentication accepted Event: PeerStatus Peer: SIP/weckhardt PeerStatus: Registered Event: PeerStatus Peer: SIP/weckhardt PeerStatus: Reachable Time: 81 Event: PeerStatus Peer: SIP/weckhardt PeerStatus: Registered Event: PeerStatus Peer: SIP/weckhardt PeerStatus: Unreachable Time: -1 Event: PeerStatus Peer: SIP/weckhardt PeerStatus: Unregistered Cause: Expired - Without the quality, you still get the PeerStatus: Registered and PeerStatus: Unegistered events. John, you can do your color-coding :-) And while I was at this patch, I also changed the Event: SIPRegistry Domain: ... Status: ... to Event: Register Channel: SIP Domain: ... Status: ... I don't have a register= line in my sip.conf, so I didn't see this event. But again I hope that this makes the event usable for other channel types as well. Because I did not see the event, I am not sure if I need the Channel: SIP at all. If the Domain is actually an sip:// - URL, it would be superfluous. I'll post the patch to bugs.digium.com when someone enlightens me on this point :-) ___ 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] SIP register and unregister events via Manager API
Holger Schurig wrote: The patch tries to send the time as well, but it fails. There are some problems currently: Seems like you are getting Time - how does it fail? Please explain... /O I applied the path from bug 2117. After this I got some events: Event: SIPPeerStatus Peer: weckhardt Status: reachable Time: 55 Event: SIPPeerRegistration Peername: dnarotam Status: Offline I think we have several problems here. Once it's Peer:, the other time it's Peername. Also, I don't like the name of the event. It should just be PeerStatus and PeerRegistration, because we might add something to IAX2 as well. So I'd suggest to do it this way: Event: PeerStatus Peer: SIP/weckhardt PeerStatus: reachable Time: 55 Event: PeerRegistration Peer: SIP/dnarotam Channel: Offline This way, other channels can send the same events, just with PeerStatus: IAX2/qtiax. I guess I'll redo the path and re-submit it to bugs.digium.com. ___ 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 -- Olle E. Johansson, Edvina.net AB, [EMAIL PROTECTED] - Phone +46 8 594 788 10, Cell phone: +46 70 593 68 51 - IP phone: sip:[EMAIL PROTECTED] - Address: Runbovägen 10, SE-192 48 Sollentuna, Sweden - Web: http://edvina.net ___ 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] SIP register and unregister events via Manager API
Holger Schurig wrote: I think we have several problems here. Once it's Peer:, the other time it's Peername. That's clearly a bug. Also, I don't like the name of the event. It should just be PeerStatus and PeerRegistration, because we might add something to IAX2 as well. So I'd suggest to do it this way: I used SIP as a prefix for testing in chan_sip2. But if it's possible to add similar events in IAX, I agree. Good to get feedback that it was a useful addition :-) /O ___ 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] SIP register and unregister events via Manager API
Holger Schurig wrote: And while I was at this patch, I also changed the Event: SIPRegistry Domain: ... Status: ... to Event: Register Channel: SIP Domain: ... Status: ... I still believe it would be better to call this Registry since that's a common term across IAX and SIP for outbound registrations. The event reflects changes in our registry, like SIP show registry and Iax2 show registry. Because I did not see the event, I am not sure if I need the Channel: SIP at all. If the Domain is actually an sip:// - URL, it would be superfluous. Domain is just the domain we register to. It might be a good thing to add the host IP we resolve this to, since that may change. Channel: makes sense, to separate registries. /O ___ 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] SIP register and unregister events via Manager API
At 10:58 AM +0200 on 7/23/04, Holger Schurig wrote: Okay, I have finished my patch. With qualify=yes in sip.conf it looks like this: - # telnet 127.0.0.1 5038 Trying 127.0.0.1... Connected to localhost.localdomain (127.0.0.1). Escape character is '^]'. Asterisk Call Manager/1.0 Action: Login Username: destar Secret: secret Response: Success Message: Authentication accepted Event: PeerStatus Peer: SIP/weckhardt PeerStatus: Registered Event: PeerStatus Peer: SIP/weckhardt PeerStatus: Reachable Time: 81 Event: PeerStatus Peer: SIP/weckhardt PeerStatus: Registered Event: PeerStatus Peer: SIP/weckhardt PeerStatus: Unreachable Time: -1 Event: PeerStatus Peer: SIP/weckhardt PeerStatus: Unregistered Cause: Expired - Without the quality, you still get the PeerStatus: Registered and PeerStatus: Unegistered events. John, you can do your color-coding :-) [snip] Not me! :-) I'd point a finger at Nicolás Gudiño and have him include it in the Asterisk Flash Operator panel, which seems to be one of the appropriate places that this could create a graphical representation of registration status and quality= response time. Maybe a red-to-green spectrum of colors on the button background. I'd expecet that each button would need to have probably independent configurations, since some devices may be very far away and thus have different numeric values mapped to different colors. If the device falls out of registration, then perhaps have thin black lines diagonally through the button, and dim it slightly? JT ___ 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] SIP register and unregister events via Manager API
Hi John, John Todd wrote: At 10:58 AM +0200 on 7/23/04, Holger Schurig wrote: Okay, I have finished my patch. With qualify=yes in sip.conf it looks like this: output snip Event: PeerStatus Peer: SIP/weckhardt PeerStatus: Reachable Time: 81 some more snip - Without the quality, you still get the PeerStatus: Registered and PeerStatus: Unegistered events. John, you can do your color-coding :-) [snip] Not me! :-) I'd point a finger at Nicolás Gudiño and have him include it in the Asterisk Flash Operator panel, which seems to be one of the appropriate places that this could create a graphical representation of registration status and quality= response time. Maybe a red-to-green spectrum of colors on the button background. I'd expecet that each button would need to have probably independent configurations, since some devices may be very far away and thus have different numeric values mapped to different colors. If the device falls out of registration, then perhaps have thin black lines diagonally through the button, and dim it slightly? Thats me... :) Well, we already have in the panel dimmed buttons for SIP peers that are unreachable, and really dimmed ones for not registered ones. Now I will have code the color shift based on the round trip time. Maybe I can zoom out the button instead of color coding? If the latency is high display the button far far away :) -- Nicolás Gudiño House Internet S.R.L. Buenos Aires - Argentina ___ 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] SIP register and unregister events via Manager API
Nicolas Gudino wrote: On Fri, 2004-07-16 at 18:28, Matthias Endler wrote: is it possible to receive SIP/IAX register and unregister events via the manager API (like in CLI)? I do receive all kinds of call events (Hangup|Join|Leave|Link|Newchannel|Newexten|Newstate|Rename|Unlink). chan_sip2 supports manager notifications: http://bugs.digium.com/bug_view_page.php?bug_id=759 And I would very much like feedback on those - are they useful? If they are, I'll backport to chan_sip. first, sorry to answer so late. I find them very useful, especially for my current project written in Java, which heavily makes use of the Asterisk Call Manager Interface. I started to backport to chan_sip.c already, it will be done by tomorrow. If you are interested I can send you a diff (stable version 0.9.1). As promised yesterday: Anybody interrested can download the patch for Asterisk 0.9.1 at http://matthiasendler.net/asterisk/patch/. Best regards Matt ___ 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] SIP register and unregister events via Manager API
Matthias Endler wrote: As promised yesterday: Anybody interrested can download the patch for Asterisk 0.9.1 at http://matthiasendler.net/asterisk/patch/. Great! Please add it to the bugtracker in a .txt file created with cvs diff -u channels/chan_sip.c The diff has to be for CVS HEAD, that is 1.0rc1 /O ___ 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] SIP register and unregister events via Manager API
Hi Olle, Nicolas Gudino wrote: On Fri, 2004-07-16 at 18:28, Matthias Endler wrote: is it possible to receive SIP/IAX register and unregister events via the manager API (like in CLI)? I do receive all kinds of call events (Hangup|Join|Leave|Link|Newchannel|Newexten|Newstate|Rename|Unlink). chan_sip2 supports manager notifications: http://bugs.digium.com/bug_view_page.php?bug_id=759 And I would very much like feedback on those - are they useful? If they are, I'll backport to chan_sip. first, sorry to answer so late. I find them very useful, especially for my current project written in Java, which heavily makes use of the Asterisk Call Manager Interface. I started to backport to chan_sip.c already, it will be done by tomorrow. If you are interested I can send you a diff (stable version 0.9.1). Best regards Matt ___ 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] SIP register and unregister events via Manager API
Nicolas Gudino wrote: On Fri, 2004-07-16 at 18:28, Matthias Endler wrote: is it possible to receive SIP/IAX register and unregister events via the manager API (like in CLI)? I do receive all kinds of call events (Hangup|Join|Leave|Link|Newchannel|Newexten|Newstate|Rename|Unlink). chan_sip2 supports manager notifications: http://bugs.digium.com/bug_view_page.php?bug_id=759 And I would very much like feedback on those - are they useful? If they are, I'll backport to chan_sip. /O ___ 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] SIP register and unregister events via Manager API
Hi all, is it possible to receive SIP/IAX register and unregister events via the manager API (like in CLI)? I do receive all kinds of call events (Hangup|Join|Leave|Link|Newchannel|Newexten|Newstate|Rename|Unlink). My manager.conf looks like this: [general] enabled = yes port = 5038 bindaddr = 0.0.0.0 [manage] secret = mysecret deny=0.0.0.0/0.0.0.0 permit=192.168.1.0/255.255.255.0 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user Thanks for any hints in advance. Matt ___ 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] SIP register and unregister events via Manager API
Hi Matthias, On Fri, 2004-07-16 at 18:28, Matthias Endler wrote: Hi all, is it possible to receive SIP/IAX register and unregister events via the manager API (like in CLI)? I do receive all kinds of call events (Hangup|Join|Leave|Link|Newchannel|Newexten|Newstate|Rename|Unlink). chan_sip2 supports manager notifications: http://bugs.digium.com/bug_view_page.php?bug_id=759 Best regards, -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ 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] sip register and nat
As I understand it, if I understand you correctly, the register parameter is for the client side. The nat=yes parameter is for the server side, so it has nothing to do with your register statement. The sip debug displays no nat because sip.broadvoice.com is not behind the nat, it's in front of it. -g On Tue, 2004-06-15 at 17:29, Kubat, Philip wrote: This may be a newbie SIP/NAT question. If so I am sorry. But any help would be appreciated. My Asterisk server is behind an ipchains box and I am trying to connect to Broadvoice. All works fine without the NAT. I have a global nat=yes prior to my register, but the sip debug allows shows no nat). Is this label issue, and am I barking up the wrong tree? Sip.conf nat=yes register = 1235551234: password @sip.broadvoice.com:5060/1235551234 sip debug Retransmitting #5 (no NAT): REGISTER sip:sip.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4:5060 ... Event: registration Content-Length: ___ 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 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] sip register and nat
This may be a newbie SIP/NAT question. If so I am sorry. But any help would be appreciated. My Asterisk server is behind an ipchains box and I am trying to connect to Broadvoice. All works fine without the NAT. I have a global nat=yes prior to my register, but the sip debug allows shows no nat). Is this label issue, and am I barking up the wrong tree? Sip.conf nat=yes register = 1235551234: password @sip.broadvoice.com:5060/1235551234 sip debug Retransmitting #5 (no NAT): REGISTER sip:sip.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4:5060 ... Event: registration Content-Length: ___ 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] SIP REGISTER
We have many Asterisk systems connected to us - we provider them with worldwide origination and termination and we have no problems. It could be provider configuration. If you can find out what kind of GW is in use at provider and version I can probably tell you how to configure it properly. Aram Ter-Martirosyan Senior Account Manager Hi-Tech Gateway, Inc. http://www.hi-teck.com 1225 Grand Central Ave. Glendale, CA 91201 [EMAIL PROTECTED] tel 818.546.4601 fax 818.546.4617 Turning Technology Into Business Solutions -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Micke Andersson Sent: Tuesday, February 17, 2004 9:05 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP REGISTER Hiyas.. I have a little problem .. I try to register my Asterisk at a sip provider.. but it just wont work. It works fine with eg xlite or Grandstream.. .but not with Asterisk. I think it is in the Register process: This is the difference I cen tell in the sip headers between Xlite and Asterisk ( I have removed IPs and numbers and replaces them with text) First Xlite: (this works) -snip SEND provider.ip.ip.ip:5060 REGISTER sip:provider.com SIP/2.0 Via: SIP/2.0/UDP ip.ip.ip.ip:5060;rport;branch=z9hG4bK06595964B0AE46CF9271267AD534E632 From: pstn-number sip:[EMAIL PROTECTED] To: pstn-number sip:[EMAIL PROTECTED] Contact: pstn-number sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 8823 REGISTER Expires: 1800 Authorization: Digest username=pstn-number,realm=provider.com,nonce=MTA3NzAyOTUwMjk2NWI2Y jg4MjcxOGNlZWRkODRhYzg4NmEyZWE5NTYwN2Y0,response=f833201fd4a8719ea9a2e c505debbd56,uri=sip:provider.com,opaque=dd5d790f90d0307c7390cdb8f6e9 4cc8,qop=auth,cnonce=4B86525A67C646469656D90AD4C1273C,nc=0002 Max-Forwards: 70 User-Agent: X-Lite build 1101 Content-Length: 0 RECEIVE provider.ip.ip.ip:5060 SIP/2.0 200 OK - end snip - This is Asterisk (does not work) --snip Reliably Transmitting: REGISTER sip:provider.com SIP/2.0 Via: SIP/2.0/UDP ip.ip.ip:5060;branch=z9hG4bK56158c1f From: sip:[EMAIL PROTECTED];tag=as017cdd56 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER User-Agent: Asterisk PBX Expires: 1200 Contact: sip:[EMAIL PROTECTED] Event: registration Content-length: 0 (no NAT) to provider.ip.ip.ip:5060 pbx1*CLI Sip read: SIP/2.0 403 Forbidden --- end snip --- The difference as I can tell is in the From: and to: lines xlite says From: number [EMAIL PROTECTED] asterisk only says From: [EMAIL PROTECTED] How do I tell my Asterisk to send the registration as xlite ? /Mike ___ 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 attachment: winmail.dat
[Asterisk-Users] SIP register and externip
I put an externip=xxx.xxx.xxx.xxx in my sip.conf so I can register with FWD from behind a NAT With this entry my PSTN calls have a problem in that the other party cannot hear me - I can hear them. It does not matter whether I make the call or the other party does. Any ideas ? TIA Simon Brown - This mail was content checked for malicious code and viruses by GFI MailSecurity. ___ 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] Sip Register Fail - NAT
I am having an issue with registering SIP client w/ NAT. I have set this up before on other boxes... But for some reason this one is not acting the same... I have attached a sip debug from the registration... For what ever reason it does not appear to be setting up the nat session correctly Am I seeing something wrong or even doing something wrong -gcc SIP CONFIG ## ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to externip = nat ip ; Address that we're going to put in SIP messages if we're behind a NAT localnet = 10.100.254.0; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask context=default ; Default for incoming calls ;srvlookup = yes; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=ilbc allow=alaw [4003] type=friend username=4003 secret=4003 host=dynamic qualify=500 context=local nat=yes mailbox=4003 ## SIP DEBUG #3 Sip read: REGISTER sip:10.100.254.21 SIP/2.0 Content-Length: 0 Contact: sip:192.168.1.10 Call-ID: [EMAIL PROTECTED] From: sip:[EMAIL PROTECTED];tag=10990022 CSeq: 87 REGISTER To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.1.10:5060 8 headers, 0 lines Using latest request as basis request Sending to 192.168.1.10 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.10:5060;received=69.132.68.17 From: sip:[EMAIL PROTECTED];tag=10990022 To: sip:[EMAIL PROTECTED];tag=as138021c1 Call-ID: [EMAIL PROTECTED] CSeq: 87 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 69.132.68.17:5060 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.10:5060;received=69.132.68.17 From: sip:[EMAIL PROTECTED];tag=10990022 To: sip:[EMAIL PROTECTED];tag=as138021c1 Call-ID: [EMAIL PROTECTED] CSeq: 87 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=267e89bd Content-Length: 0 to 69.132.68.17:5060 ^Dtnevoip*CLI Sip read: REGISTER sip:10.100.254.21 SIP/2.0 Content-Length: 0 Contact: sip:192.168.1.10 Call-ID: [EMAIL PROTECTED] From: sip:[EMAIL PROTECTED];tag=10990413 CSeq: 88 REGISTER To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.1.10:5060 Proxy-Authorization: Digest username=4003,realm=asterisk,nonce=267e89bd,uri=sip:10.100.254.21 ,response=fb30e53fffc30ea15fc97acf7d82322f 9 headers, 0 lines Using latest request as basis request Sending to 192.168.1.10 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.10:5060;received=69.132.68.17 From: sip:[EMAIL PROTECTED];tag=10990413 To: sip:[EMAIL PROTECTED];tag=as138021c1 Call-ID: [EMAIL PROTECTED] CSeq: 88 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 69.132.68.17:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.10:5060;received=69.132.68.17 From: sip:[EMAIL PROTECTED];tag=10990413 To: sip:[EMAIL PROTECTED];tag=as138021c1 Call-ID: [EMAIL PROTECTED] CSeq: 88 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 69.132.68.17:5060 Feb 22 19:33:23 [1;33;40mNOTICE[0;37;40m[-1147384912]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m5577[0;37;40m [1;37;40mhandle_request[0;37;40m: Registration from 'sip:[EMAIL PROTECTED]' failed for '69.132.68.17' ^Dtnevoip*CLI Sip read: REGISTER sip:10.100.254.21 SIP/2.0 Content-Length: 0 Contact: sip:192.168.1.10 Call-ID: [EMAIL PROTECTED] From: sip:[EMAIL PROTECTED];tag=10990613 CSeq: 89 REGISTER To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.1.10:5060 8 headers, 0 lines Using latest request as basis request Sending to 192.168.1.10 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.10:5060;received=69.132.68.17 From: sip:[EMAIL PROTECTED];tag=10990613 To: sip:[EMAIL PROTECTED];tag=as42b62c4b Call-ID: [EMAIL PROTECTED] CSeq: 89 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content- ___ Asterisk-Users mailing list [EMAIL PROTECTED]
RE: [Asterisk-Users] Sip Register Fail - NAT
I was able to resolve the issue... Me being stupid... Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of AstGrp Posted At: Sunday, February 22, 2004 7:45 PM Posted To: Asterisk User Group Conversation: Sip Register Fail - NAT Subject: [Asterisk-Users] Sip Register Fail - NAT I am having an issue with registering SIP client w/ NAT. I have set this up before on other boxes... But for some reason this one is not acting the same... I have attached a sip debug from the registration... For what ever reason it does not appear to be setting up the nat session correctly Am I seeing something wrong or even doing something wrong -gcc SIP CONFIG ## ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to externip = nat ip ; Address that we're going to put in SIP messages if we're behind a NAT localnet = 10.100.254.0; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask context=default ; Default for incoming calls ;srvlookup = yes; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=ilbc allow=alaw [4003] type=friend username=4003 secret=4003 host=dynamic qualify=500 context=local nat=yes mailbox=4003 ## SIP DEBUG #3 Sip read: REGISTER sip:10.100.254.21 SIP/2.0 Content-Length: 0 Contact: sip:192.168.1.10 Call-ID: [EMAIL PROTECTED] From: sip:[EMAIL PROTECTED];tag=10990022 CSeq: 87 REGISTER To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.1.10:5060 8 headers, 0 lines Using latest request as basis request Sending to 192.168.1.10 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.10:5060;received=69.132.68.17 From: sip:[EMAIL PROTECTED];tag=10990022 To: sip:[EMAIL PROTECTED];tag=as138021c1 Call-ID: [EMAIL PROTECTED] CSeq: 87 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 69.132.68.17:5060 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.10:5060;received=69.132.68.17 From: sip:[EMAIL PROTECTED];tag=10990022 To: sip:[EMAIL PROTECTED];tag=as138021c1 Call-ID: [EMAIL PROTECTED] CSeq: 87 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=267e89bd Content-Length: 0 to 69.132.68.17:5060 ^Dtnevoip*CLI Sip read: REGISTER sip:10.100.254.21 SIP/2.0 Content-Length: 0 Contact: sip:192.168.1.10 Call-ID: [EMAIL PROTECTED] From: sip:[EMAIL PROTECTED];tag=10990413 CSeq: 88 REGISTER To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.1.10:5060 Proxy-Authorization: Digest username=4003,realm=asterisk,nonce=267e89bd,uri=sip:10.100.254.21 ,response=fb30e53fffc30ea15fc97acf7d82322f 9 headers, 0 lines Using latest request as basis request Sending to 192.168.1.10 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.10:5060;received=69.132.68.17 From: sip:[EMAIL PROTECTED];tag=10990413 To: sip:[EMAIL PROTECTED];tag=as138021c1 Call-ID: [EMAIL PROTECTED] CSeq: 88 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 69.132.68.17:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.10:5060;received=69.132.68.17 From: sip:[EMAIL PROTECTED];tag=10990413 To: sip:[EMAIL PROTECTED];tag=as138021c1 Call-ID: [EMAIL PROTECTED] CSeq: 88 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 69.132.68.17:5060 Feb 22 19:33:23 [1;33;40mNOTICE[0;37;40m[-1147384912]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m5577[0;37;40m [1;37;40mhandle_request[0;37;40m: Registration from 'sip:[EMAIL PROTECTED]' failed for '69.132.68.17' ^Dtnevoip*CLI Sip read: REGISTER sip:10.100.254.21 SIP/2.0 Content-Length: 0 Contact: sip:192.168.1.10 Call-ID: [EMAIL PROTECTED] From: sip:[EMAIL PROTECTED];tag=10990613 CSeq: 89 REGISTER To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.1.10:5060 8 headers, 0 lines Using latest request as basis request Sending to 192.168.1.10 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.10:5060;received=69.132.68.17 From: sip
RE: [Asterisk-Users] SIP REGISTER
John Fraizer wrote on the Tuesday, February 17, 2004 7:14 PM Try sending us the registry line and context information from your sip.conf. It is much easier to figure out what you're doing wrong from there. I register my Asterisk server with 6 different SIP providers with no problems at all. John this is the line: register = pstn-number:passwd:[EMAIL PROTECTED]/pstn-number /Mike Micke Andersson wrote: Hiyas.. I have a little problem .. I try to register my Asterisk at a sip provider.. but it just wont work. It works fine with eg xlite or Grandstream.. .but not with Asterisk. I think it is in the Register process: This is the difference I cen tell in the sip headers between Xlite and Asterisk ( I have removed IPs and numbers and replaces them with text) First Xlite: (this works) -snip SEND provider.ip.ip.ip:5060 REGISTER sip:provider.com SIP/2.0 Via: SIP/2.0/UDP ip.ip.ip.ip:5060;rport;branch=z9hG4bK06595964B0AE46CF9271267AD534E632 From: pstn-number sip:[EMAIL PROTECTED] To: pstn-number sip:[EMAIL PROTECTED] Contact: pstn-number sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 8823 REGISTER Expires: 1800 Authorization: Digest username=pstn-number,realm=provider.com,nonce=MTA3NzAyOTUwMjk2NWI 2Y jg4MjcxOGNlZWRkODRhYzg4NmEyZWE5NTYwN2Y0,response=f833201fd4a8719ea9a 2e c505debbd56,uri=sip:provider.com,opaque=dd5d790f90d0307c7390cdb8f6 e9 4cc8,qop=auth,cnonce=4B86525A67C646469656D90AD4C1273C,nc=0002 Max-Forwards: 70 User-Agent: X-Lite build 1101 Content-Length: 0 RECEIVE provider.ip.ip.ip:5060 SIP/2.0 200 OK - end snip - This is Asterisk (does not work) --snip Reliably Transmitting: REGISTER sip:provider.com SIP/2.0 Via: SIP/2.0/UDP ip.ip.ip:5060;branch=z9hG4bK56158c1f From: sip:[EMAIL PROTECTED];tag=as017cdd56 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER User-Agent: Asterisk PBX Expires: 1200 Contact: sip:[EMAIL PROTECTED] Event: registration Content-length: 0 (no NAT) to provider.ip.ip.ip:5060 pbx1*CLI Sip read: SIP/2.0 403 Forbidden --- end snip --- The difference as I can tell is in the From: and to: lines xlite says From: number [EMAIL PROTECTED] asterisk only says From: [EMAIL PROTECTED] How do I tell my Asterisk to send the registration as xlite ? /Mike ___ 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 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 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] SIP REGISTER
On Wed, 18 Feb 2004, Micke Andersson wrote: John Fraizer wrote on the Tuesday, February 17, 2004 7:14 PM Try sending us the registry line and context information from your sip.conf. It is much easier to figure out what you're doing wrong from there. I register my Asterisk server with 6 different SIP providers with no problems at all. John this is the line: register = pstn-number:passwd:[EMAIL PROTECTED]/pstn-number /Mike You've got a syntax problem. It SHOULD be: register = pstn-number:[EMAIL PROTECTED]/pstn-number John ___ 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] SIP REGISTER
John Fraizer wrote on the Wednesday, February 18, 2004 6:16 PM You've got a syntax problem. It SHOULD be: register = pstn-number:[EMAIL PROTECTED]/pstn-number Tried that too, no go.. I thought the syntax were: register = username:passwd:[EMAIL PROTECTED]/local number /Mike ___ 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] SIP REGISTER
You've got a syntax problem. It SHOULD be: register = pstn-number:[EMAIL PROTECTED]/pstn-number Tried that too, no go.. I thought the syntax were: register = username:passwd:[EMAIL PROTECTED]/local number Don't know about all the possible variations, but I'm using register=61890:[EMAIL PROTECTED] without any trailing /local number, and its working fine. My inbound fwd calls go to an ivr though. Rich ___ 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] SIP REGISTER
Hiyas.. I have a little problem .. I try to register my Asterisk at a sip provider.. but it just wont work. It works fine with eg xlite or Grandstream.. .but not with Asterisk. I think it is in the Register process: This is the difference I cen tell in the sip headers between Xlite and Asterisk ( I have removed IPs and numbers and replaces them with text) First Xlite: (this works) -snip SEND provider.ip.ip.ip:5060 REGISTER sip:provider.com SIP/2.0 Via: SIP/2.0/UDP ip.ip.ip.ip:5060;rport;branch=z9hG4bK06595964B0AE46CF9271267AD534E632 From: pstn-number sip:[EMAIL PROTECTED] To: pstn-number sip:[EMAIL PROTECTED] Contact: pstn-number sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 8823 REGISTER Expires: 1800 Authorization: Digest username=pstn-number,realm=provider.com,nonce=MTA3NzAyOTUwMjk2NWI2Y jg4MjcxOGNlZWRkODRhYzg4NmEyZWE5NTYwN2Y0,response=f833201fd4a8719ea9a2e c505debbd56,uri=sip:provider.com,opaque=dd5d790f90d0307c7390cdb8f6e9 4cc8,qop=auth,cnonce=4B86525A67C646469656D90AD4C1273C,nc=0002 Max-Forwards: 70 User-Agent: X-Lite build 1101 Content-Length: 0 RECEIVE provider.ip.ip.ip:5060 SIP/2.0 200 OK - end snip - This is Asterisk (does not work) --snip Reliably Transmitting: REGISTER sip:provider.com SIP/2.0 Via: SIP/2.0/UDP ip.ip.ip:5060;branch=z9hG4bK56158c1f From: sip:[EMAIL PROTECTED];tag=as017cdd56 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER User-Agent: Asterisk PBX Expires: 1200 Contact: sip:[EMAIL PROTECTED] Event: registration Content-length: 0 (no NAT) to provider.ip.ip.ip:5060 pbx1*CLI Sip read: SIP/2.0 403 Forbidden --- end snip --- The difference as I can tell is in the From: and to: lines xlite says From: number [EMAIL PROTECTED] asterisk only says From: [EMAIL PROTECTED] How do I tell my Asterisk to send the registration as xlite ? /Mike ___ 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] SIP REGISTER
Try sending us the registry line and context information from your sip.conf. It is much easier to figure out what you're doing wrong from there. I register my Asterisk server with 6 different SIP providers with no problems at all. John Micke Andersson wrote: Hiyas.. I have a little problem .. I try to register my Asterisk at a sip provider.. but it just wont work. It works fine with eg xlite or Grandstream.. .but not with Asterisk. I think it is in the Register process: This is the difference I cen tell in the sip headers between Xlite and Asterisk ( I have removed IPs and numbers and replaces them with text) First Xlite: (this works) -snip SEND provider.ip.ip.ip:5060 REGISTER sip:provider.com SIP/2.0 Via: SIP/2.0/UDP ip.ip.ip.ip:5060;rport;branch=z9hG4bK06595964B0AE46CF9271267AD534E632 From: pstn-number sip:[EMAIL PROTECTED] To: pstn-number sip:[EMAIL PROTECTED] Contact: pstn-number sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 8823 REGISTER Expires: 1800 Authorization: Digest username=pstn-number,realm=provider.com,nonce=MTA3NzAyOTUwMjk2NWI2Y jg4MjcxOGNlZWRkODRhYzg4NmEyZWE5NTYwN2Y0,response=f833201fd4a8719ea9a2e c505debbd56,uri=sip:provider.com,opaque=dd5d790f90d0307c7390cdb8f6e9 4cc8,qop=auth,cnonce=4B86525A67C646469656D90AD4C1273C,nc=0002 Max-Forwards: 70 User-Agent: X-Lite build 1101 Content-Length: 0 RECEIVE provider.ip.ip.ip:5060 SIP/2.0 200 OK - end snip - This is Asterisk (does not work) --snip Reliably Transmitting: REGISTER sip:provider.com SIP/2.0 Via: SIP/2.0/UDP ip.ip.ip:5060;branch=z9hG4bK56158c1f From: sip:[EMAIL PROTECTED];tag=as017cdd56 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER User-Agent: Asterisk PBX Expires: 1200 Contact: sip:[EMAIL PROTECTED] Event: registration Content-length: 0 (no NAT) to provider.ip.ip.ip:5060 pbx1*CLI Sip read: SIP/2.0 403 Forbidden --- end snip --- The difference as I can tell is in the From: and to: lines xlite says From: number [EMAIL PROTECTED] asterisk only says From: [EMAIL PROTECTED] How do I tell my Asterisk to send the registration as xlite ? /Mike ___ 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 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] SIP register/auth with Grandstream BudgeTone-100
Hi,Key Aavoja, Have you successfully registed to * with secret specificated? Regards. bfrac - Original Message - From: Key Aavoja [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, January 23, 2004 2:00 AM Subject: [Asterisk-Users] SIP register/auth with Grandstream BudgeTone-100 Hello, I have a problem with asterisk and Grandstream BudgeTone-100. With default configuration everything works (in anonymous mode and fixed IP), but if Im trying to enable registering, it dos not work. I used 'sip debug' and verbose level 10, nothing happens if I switch telephone on (no messages about bad auth etc). As I understood, after switching phone on at first it will try to register in asterisk if Im trying to call somewhere. I searched in list-archive and I didnt found that anybody else has this kind of problem. I read also: http://lists.digium.com/pipermail/asterisk-users/2003-June/013288.html and I did so. sip.conf - [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls disallow=all; Disallow all codecs allow=g729 [cisco] context=in type=friend insecure=yes host=removed dtmfmode=rfc2833 [grandstream1] type=friend secret=grandstream1 host=dynamic context=class1 dtmfmode=rfc2833 [grandstream2] type=friend secret=grandstream2 nat=yes host=dynamic context=class1 dtmfmode=rfc2833 Asterisk ver: Asterisk CVS-01/22/04-18:13:23 Grandstream ver: Program--1.0.3.81Bootloader--1.0.0.7 HTML--1.0.0.18 * And as I mentioned before, without registration and with static IP everything works, it seems, that something is misconfigured in my setup for authentication or this phone firmware is buggy? (but its latest, I checked www.grandstream.com) Best Regards: Key Aavoja /* Never argue with an idiot. They drag you down to their level, then beat you with experience.*/ ___ 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 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] SIP register/auth with Grandstream BudgeTone-100
Hello, I have a problem with asterisk and Grandstream BudgeTone-100. With default configuration everything works (in anonymous mode and fixed IP), but if Im trying to enable registering, it dos not work. I used 'sip debug' and verbose level 10, nothing happens if I switch telephone on (no messages about bad auth etc). As I understood, after switching phone on at first it will try to register in asterisk if Im trying to call somewhere. I searched in list-archive and I didnt found that anybody else has this kind of problem. I read also: http://lists.digium.com/pipermail/asterisk-users/2003-June/013288.html and I did so. sip.conf - [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls disallow=all; Disallow all codecs allow=g729 [cisco] context=in type=friend insecure=yes host=removed dtmfmode=rfc2833 [grandstream1] type=friend secret=grandstream1 host=dynamic context=class1 dtmfmode=rfc2833 [grandstream2] type=friend secret=grandstream2 nat=yes host=dynamic context=class1 dtmfmode=rfc2833 Asterisk ver: Asterisk CVS-01/22/04-18:13:23 Grandstream ver: Program--1.0.3.81Bootloader--1.0.0.7HTML--1.0.0.18 * And as I mentioned before, without registration and with static IP everything works, it seems, that something is misconfigured in my setup for authentication or this phone firmware is buggy? (but its latest, I checked www.grandstream.com) Best Regards: Key Aavoja /* Never argue with an idiot. They drag you down to their level, then beat you with experience.*/ ___ 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
