Re: [asterisk-users] On Register, run a script, validate source IP

Hi Jöran > for me it sounds like you need an SBC. > We use Kamailio in order to check users IP Addresses. There are modules > like "permissions" in kamailio what could do this. As well there are pike > checks, sanity checks and a bunch of other useful tools. You are absolutely right. We are on a

Re: [asterisk-users] On Register, run a script, validate source IP

Hi, for me it sounds like you need an SBC. We use Kamailio in order to check users IP Addresses. There are modules like "permissions" in kamailio what could do this. As well there are pike checks, sanity checks and a bunch of other useful tools. If you want to secure and protect your Asterisk

Re: [asterisk-users] On Register, run a script, validate source IP

Hello, Have you tried with ACL (acl.conf) ? Cheers Le lun. 18 nov. 2019 à 13:22, Benoit Panizzon a écrit : > Hi Gang > > To increase security against phished passwords and similar attacks, we > consider offering customers to define IP ranges (or GeoIP locations) > from which their dynamic

Re: [asterisk-users] On Register, run a script, validate source IP

Hi Sebastian > That would require your script to update sip.conf dynamically and reload the > config for each time user wants to update their accepted location. Hmm, maybe using asterisk realtime and attempting to put the config into a database would be worth an approach. Until now we only use

Re: [asterisk-users] On Register, run a script, validate source IP

18 november 2019 13:23 Till: asterisk-users@lists.digium.com Ämne: [asterisk-users] On Register, run a script, validate source IP Hi Gang To increase security against phished passwords and similar attacks, we consider offering customers to define IP ranges (or GeoIP locations) from which

[asterisk-users] On Register, run a script, validate source IP

Hi Gang To increase security against phished passwords and similar attacks, we consider offering customers to define IP ranges (or GeoIP locations) from which their dynamic registrations are being accepted. I can already look at the source IP in the dial plan, so no issue with validate an INVITE

[asterisk-users] No register between Asterisk 15 and 13 running pjsip

Hello, we have 4 asteriks, 2 in office on one server (wazo and mobydick), and 2 in DC (self compiled) each on his own server. All of them are VMs under Debian Stretch. We used OpenVPN to connect the machines together in TAP mode, everything was running well. Setup is following: the 2

Re: [asterisk-users] Received REGISTER response 401(Unauthorized 1103003032F)

Possibly the realm? Thanks, Steve On Sat, Sep 2, 2017 at 3:58 AM, O. Hartmann wrote: > > It might sound stupid and a kind of "noobish", but I have serious trouble > with > registering one of my ITSP to Asterisk 13, running on a FreeBSD 12-CURRENT > box. > > The

Re: [asterisk-users] Received REGISTER response 401(Unauthorized 1103003032F)

On 09/02/2017 06:36 PM, O. Hartmann wrote: Is this question to "blunt" for this forum? No, But it is a holiday weekend in the States. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check

Re: [asterisk-users] Received REGISTER response 401(Unauthorized 1103003032F)

Am Sat, 2 Sep 2017 09:58:09 +0200 "O. Hartmann" schrieb: Is this question to "blunt" for this forum? The background is, that I have two ITSP providing VoIP. One works with Asterisk 13 like a charme, but the other one not. This specific ITSP claims that they've provided

[asterisk-users] Received REGISTER response 401(Unauthorized 1103003032F)

It might sound stupid and a kind of "noobish", but I have serious trouble with registering one of my ITSP to Asterisk 13, running on a FreeBSD 12-CURRENT box. The following is seen in the log and anything seems somehow "normal", my PBX tries to REGISTER, receives 401, and then nothing

[asterisk-users] Cannot register sip on asterisk running on amazon

I have installed Freepbx successfully on the Amazon Ec2 micro instance I finally could access to the Freepbx and it show the state success. I create the extension on this instance then I wonder why when i try to register my sip client to this instance it seems like no any action. Could you please

Re: [asterisk-users] Cannot register sip on asterisk running on amazon

A few Things. 1. Nat on the Extension you are setting up. 2. Security groups on the EC2 instance to accept sip traffic from your IP 3. Your home firewall allowing SIP traffic to come back from your EC2 instance. On Mon, Sep 14, 2015 at 11:33 AM, Thyda ENG wrote: > I have

Re: [asterisk-users] Cannot register sip on asterisk running on amazon

Have you opened the required ports through the firewall? That is where I would start. On Mon, Sep 14, 2015 at 10:33 AM, Thyda ENG wrote: > I have installed Freepbx successfully on the Amazon Ec2 micro instance I > finally could access to the Freepbx and it show the state

Re: [asterisk-users] Extra REGISTER messages sent by Asterisk when subscribe for MWI is defined in zoiper

Hello, I noticed something that might be a result from the fix suggested here, so I'll continue a bit on this thread. After removing the callbackextension field from my realtime sip peer table, the following started happening: I issued command 'sip reload' on the cli and get the following

[asterisk-users] Extra REGISTER messages sent by Asterisk when subscribe for MWI is defined in zoiper

Hello all, I have an Asterisk installation with Kamailio using realtime integration. I have gotten my clients to register, but there is something odd about the sip message flow with some of my clients. My clients are Zoiper and Asterisk is 11.10.2. When I set 'Subscribe to MWI' value to 'both',

Re: [asterisk-users] Extra REGISTER messages sent by Asterisk when subscribe for MWI is defined in zoiper

Olli Heiskanen wrote: Hello all, Bonjour, I have an Asterisk installation with Kamailio using realtime integration. I have gotten my clients to register, but there is something odd about the sip message flow with some of my clients. My clients are Zoiper and Asterisk is 11.10.2. When I set

Re: [asterisk-users] Extra REGISTER messages sent by Asterisk when subscribe for MWI is defined in zoiper

Hello, Thanks for your response, I actually verified that the Zoiper setting is not the reason for Asterisk to start sending REGISTERs, it only looked like it as I checked the Kamailio output before Asterisk sent the first REGISTER to Kamailio, right after I had played with that setting. (sorry,

Re: [asterisk-users] Extra REGISTER messages sent by Asterisk when subscribe for MWI is defined in zoiper

Olli Heiskanen wrote: Hello, Thanks for your response, I actually verified that the Zoiper setting is not the reason for Asterisk to start sending REGISTERs, it only looked like it as I checked the Kamailio output before Asterisk sent the first REGISTER to Kamailio, right after I had played

Re: [asterisk-users] Extra REGISTER messages sent by Asterisk when subscribe for MWI is defined in zoiper

Thanks, there are no register lines in my sip.conf, but I have defined callbackextension fields in the realtime table, to be the same value as the extension name. In this case, extension 771 has callbackextension value 771. I tried replacing those with null values but that had no effect on the

Re: [asterisk-users] Extra REGISTER messages sent by Asterisk when subscribe for MWI is defined in zoiper

Olli Heiskanen wrote: Thanks, there are no register lines in my sip.conf, but I have defined callbackextension fields in the realtime table, to be the same value as the extension name. In this case, extension 771 has callbackextension value 771. I tried replacing those with null values but that

Re: [asterisk-users] Extra REGISTER messages sent by Asterisk when subscribe for MWI is defined in zoiper

Wow, thanks Joshua, it would've taken me forever to find the answer there. It did the trick and the registrations look much better. Merci beaucoup! - Olli 2014-07-15 16:26 GMT+03:00 Joshua Colp jc...@digium.com: Olli Heiskanen wrote: Thanks, there are no register lines in my sip.conf,

Re: [asterisk-users] Initial REGISTER Request: Contains Credentials before 401: KDDI Japan

Brian, KDDI does provide a list of supported equipment and vendors. Specific hardware or license based software products that quickly become cost prohibitive. I doubt that Asterisk will find it's way on the list any time soon. Because KDDI follows the traditional big telco method of

[asterisk-users] Initial REGISTER Request: Contains Credentials before 401

My SIP provider is not happy that credentials (in the Authorization header field) are provided in the initial REGISTER request. The SIP provider ONLY wants the credentials AFTER rejecting the message with a 401. I know it's dumb, because the RFC says that the the initial REGISTER message MAY

Re: [asterisk-users] Initial REGISTER Request: Contains Credentials before 401

Brian LaVallee wrote: My SIP provider is not happy that credentials (in the Authorization header field) are provided in the initial REGISTER request. The SIP provider ONLY wants the credentials AFTER rejecting the message with a 401. I know it's dumb, because the RFC says that the the

[asterisk-users] Initial REGISTER Request: Contains Credentials before 401: KDDI Japan

-users@lists.digium.com Subject: Re: [asterisk-users] Initial REGISTER Request: Contains Credentials before 401 Brian LaVallee wrote: My SIP provider is not happy that credentials (in the Authorization header field) are provided in the initial REGISTER request. The SIP provider ONLY wants

Re: [asterisk-users] Can't register to Asterisk 1.6 with old Aastra phones

am also stuck with Alcatel lucent IP Touch 4018 any one connected them to Asterisk thanks Regards. Kyeyune Bob Network IT Engineer +256 774 702 258 bob.kyey...@onesolutions.ug Integrated IT services from Plot 57B Luthuli Avenue Bugolobi, Kampala On Sun, Apr 28, 2013 at 11:56 PM, Carlos

Re: [asterisk-users] Can't register to Asterisk 1.6 with old Aastra phones

Hi Carlos, On 04/28/2013 10:56 PM, Carlos Alvarez wrote: We have a new customer with a lot of old phones like the 9133i. They won't register, and we see some very strange behavior with them. If the SIP peer exists, they simply fail silently, with no error in the CLI or the messages log.

Re: [asterisk-users] Can't register to Asterisk 1.6 with old Aastra phones

Well the solution turned out to be putting the Asterisk server name in the Proxy field as well as in the server field. Then it properly formatted the SIP registration request. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ --

[asterisk-users] Can't register to Asterisk 1.6 with old Aastra phones

We have a new customer with a lot of old phones like the 9133i. They won't register, and we see some very strange behavior with them. If the SIP peer exists, they simply fail silently, with no error in the CLI or the messages log. Nothing works, but no errors. If the peer does not exist, it's

Re: [asterisk-users] Can't register to Asterisk 1.6 with old Aastra phones

On Apr 28, 2013, at 13:56, Carlos Alvarez car...@televolve.com wrote: If the SIP peer exists, they simply fail silently, with no error in the CLI or the messages log. Nothing works, but no errors. Maybe 'sip set debug peer xxx' where 'xxx' is the peer name, and then try to see if you can

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

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

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

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

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

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

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

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

Re: [asterisk-users] sip register peer (the quest for near 100% availability)

[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

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,

[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

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

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

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

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

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

[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

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

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.

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)

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

[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)

[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

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/

[asterisk-users] Fwd: Register SIP on Asterisk 1.6.2.14 on Centos 5.5 64bit

-- Forwarded message -- From: Phuong Hoang ducphuongbk200...@gmail.com Date: Thu, Nov 18, 2010 at 9:16 AM Subject: Register SIP on Asterisk 1.6.2.14 on Centos 5.5 64bit To: asterisk-users@lists.digium.com Hi all, I registered sip phone(X-Lite) on Asterisk 1.6.2.14 on CentOS 5.5

Re: [asterisk-users] Fwd: Register SIP on Asterisk 1.6.2.14 on Centos 5.5 64bit

On 18 Nov 2010, at 10:33, Phuong Hoang wrote: I registered sip phone(X-Lite) on Asterisk 1.6.2.14 on CentOS 5.5 64 bit but not successful, Can anyone help me to do it? Given that you haven't given any error messages, any logs, or your sip.conf, or the manner in which it is not working

Re: [asterisk-users] sip.conf register in realtime DB

On 08/07/2010 01:11 AM, unsero...@aol.com wrote: Why don't you use 'real' realtime meaning to have your sip peers in your database? Then you would not have to do a reload after adding new peers to your db. And you can still have sip peers additionally in sip.conf. I have all of my sip

Re: [asterisk-users] sip.conf register in realtime DB

Please can anyone help me with this ?! I have tried renaming the sip.conf file, or tried including another file into sip.conf like sippy.conf and then add sippy.conf = mysql,AsteriskDB,ast_config to extconfig.conf but all this is not working. The only thing that changes something is my

Re: [asterisk-users] sip.conf register in realtime DB

You cannot use realtime static and the other realtime tables at the same time. You will need to use realtime and then use something like the EXEC command in sip.conf to execute a script that then pulls the register statement from your database. Or use the realtime static table for

Re: [asterisk-users] sip.conf register in realtime DB

On 08/06/2010 06:45 PM, Carlos Chavez wrote: You cannot use realtime static and the other realtime tables at the same time. You will need to use realtime and then use something like the EXEC command in sip.conf to execute a script that then pulls the register statement from your

Re: [asterisk-users] sip.conf register in realtime DB

On 08/06/2010 06:45 PM, Carlos Chavez wrote: Or use the realtime static table for everything. What do you mean by everything ?! What is this everything ?! You mean all the sip options in a database and so no sip.conf file ?! Kind regards, Jonas. --

Re: [asterisk-users] sip.conf register in realtime DB

You cannot use realtime static and the other realtime tables at the same time. You will need to use realtime and then use something like the EXEC command in sip.conf to execute a script that then pulls the register statement from your database. Or use the realtime static table for

[asterisk-users] sip.conf register in realtime DB

Hello list, scrambling different pieces of info together I've come with the following : I want to have my register = statements in a MySQL-database, so I've made the following table. table ast_config : id 1 cat_metric 0 var_metric 0 commented 0 filename sip.conf category general

[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:

Re: [asterisk-users] res_cepstral, register existing Cepstral licenses.

Kevin P. Fleming wrote: There is no need; your existing Cepstral-supplied licenses will continue to operate, and will be added to any Digium-supplied licenses you purchase and activate. Hi Kevin. That didn't work. If I use 'swift -n Allison-8kHz -o test.wav Hello, test and play the

[asterisk-users] res_cepstral, register existing Cepstral licenses.

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I have a license for Allison-8kHz and two concurrent port licenses that I purchased from Cepstral at the end of last year. I just got around to installing to my * 1.6.0.10 machine. I've decided that the best way for me to integrate the two would be

Re: [asterisk-users] res_cepstral, register existing Cepstral licenses.

Barry L. Kline wrote: I'm going to end up buying more ports from Digium but I'd like to also use the existing voice/port licenses that I currently have. Is this possible? Is there anyway to migrate the licenses to the Digium implementation of Cepstral? There is no need; your existing

Re: [asterisk-users] res_cepstral, register existing Cepstral licenses.

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Kevin P. Fleming wrote: There is no need; your existing Cepstral-supplied licenses will continue to operate, and will be added to any Digium-supplied licenses you purchase and activate. Thanks Kevin. So I shouldn't worry about this?

[asterisk-users] Relay Register

Good morning everybody. My question is simple. Is there a way to perform relay register with Asterisk ? More precisely, I want my clients regiter to a Proxy Registrar (OpenSIPS/Kamailio) through my Asterisk : REGISTER REGISTER Client Asterisk

Re: [asterisk-users] Relay Register

Not sure about this. It seems you are trying to find a solution to a problem which you do not actually describe. I.E, you have problem X, you think that doing Y might be the solution, but you don't know how to do Y (and in this case, neither do I). How about exposing underlying problem X to the

Re: [asterisk-users] Relay Register

: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Jean-Michel Hiver Envoyé : mardi 24 mars 2009 12:19 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Relay Register Not sure about this. It seems you

Re: [asterisk-users] Relay Register

: Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Relay Register Not sure about this. It seems you are trying to find a solution to a problem which you do not actually describe. I.E, you have problem X, you think that doing Y might be the solution

[asterisk-users] Manipulating REGISTER messages

Hello, I would like to add SIP headers to the REGISTER messages Asterisk (1.6) sends to an external proxy. Also, I want to be able to reorder the lines. Is it possible? If yes, how? Thanks. ___ -- Bandwidth and Colocation Provided by

[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

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

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

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.

Re: [asterisk-users] RealTime register for iax DID

Hello list, Is there a way to dynamically register a DID when the iax config is all RealTime? Something the equivalent of the register=... statement? Something like a register field in the iax mysql table or any other way outside of a flat file? I've been googling for that without

[asterisk-users] RealTime register for iax DID

Hello list, Is there a way to dynamically register a DID when the iax config is all RealTime? Something the equivalent of the register=... statement? Something like a register field in the iax mysql table or any other way outside of a flat file? I've been googling for that without finding

[asterisk-users] iax register gets facility rejected

I'm trying to setup zoiper ( formerly idefisk ) to use my asterisk server at work from home. I've setup zoiper for iax, set the ip address to work's fixed ip address, user: home, password: password but the zoiper log shows: 11:02:35 Rejected registration for '[EMAIL PROTECTED]' with cause

Re: [asterisk-users] Using register = to let Asterisk register to another softswitch via SIP

The same way you do it with IAX2, pretty much. http://www.voip-info.org/wiki-Asterisk+config+sip.conf On Fri, 19 Oct 2007, bilal ghayyad wrote: Hi All; Alot of softswitches or PBX's does not accept to manipulate any SIP call without being registered firstly. So that means, I need asterisk

[asterisk-users] Using register = to let Asterisk register to another softswitch via SIP

Hi All; Alot of softswitches or PBX's does not accept to manipulate any SIP call without being registered firstly. So that means, I need asterisk to register firstly then I can route my calls to that SIP trunk. In IAX2, we use the register = , so what shall we do in Asterisk? And how its format

[asterisk-users] ekiga register problems

returning newbie. Trying to register ekiga for the first time to my asterisk server only. [204] user=204 context=internal type=friend secret=xxx insecure=very canreinvite=no host=dynamic disallow=all allow=ulaw allow=alaw nat=no Can anyone tell me what I am missing? I am not

Re: [asterisk-users] ekiga register problems

On Mon, May 28, 2007 at 10:45:25PM -0400, Brad Sumrall wrote: returning newbie. Trying to register ekiga for the first time to my asterisk server only. [204] user=204 context=internal type=friend secret=xxx insecure=very canreinvite=no host=dynamic disallow=all

RE: [asterisk-users] ekiga register problems

:17 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ekiga register problems On Mon, May 28, 2007 at 10:45:25PM -0400, Brad Sumrall wrote: returning newbie. Trying to register ekiga for the first time to my asterisk server only. [204] user=204 context=internal type

[asterisk-users] Can't Register Client - Multiple Subnets

I am unable to get any softphone to register to my asterisk server when I am connected via VPN. I have tried Ekiga, LinPhone, and Twinkle... on multiple machines. It works fine when locally connected (same subnet). The VPN is not NAT'ing anything... and all other connections work fine across

[asterisk-users] SPA3000 register in asterisk

I have like clients several spa 3000, problem is that spa3000 is not registered or something by the east style problem must to be by bandwidth? spa3000 verifies bandwidth qeu can use and that is registered or no?very I am intrigued with this problemilla. Thanks your help.

[asterisk-users] IAX2 register refuse but Dial cmd works!

Hi, I just set two asterisk connect with iax2 trunk. B server [user1] type=user trunk=yes context=from-trunk username=user1 auth=plaintext secret=passwd notransfer=yes A server register = user1:[EMAIL PROTECTED] I notice on A's CLI, it shows Registration of 'user1' rejected: 'Registration

[asterisk-users] Repost: Register message received from realtime peer crashes Asterisk

When Asterisk (1.2.12.1) receives a SIP register message for a realtimepeer, the CLI reports "Disconnected from Asterisk server". Asterisk hasdisappeared:asterisk -rUnable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctlexist?)A look at the full log doesn't reveal much:Sep 17

[asterisk-users] Extensions Register but don't ring when called, can call others though

Not sure where to even start with this. I am running asterisk 1.2.1 and freepbx as the frontend. I have a remote user with soft phones (tried 3 all register and call out) that registers fine (see it in asterisk and see it in phone) and can call extensions on the asterisk box with no problem. Other

Re: [asterisk-users] Extensions Register but don't ring when called, can call others though

Try to active callwaiting in those unreachable extensions. You just need to dial *70 from every SIP extension. Be aware that *70 (call waiting ) may be disabled in your freepbx. Hope it helps, Marco Mouta Please give me some feedback On 7/17/06, Tim P [EMAIL PROTECTED] wrote: Not sure where

Re: [Asterisk-Users] show register users

21 jun 2006 kl. 03.29 skrev unplug: Hi all, As I know, I can show registered users in CLI using sip show users/sip show registry. There is no such thing as registered users. Only peers register with Asterisk and you will see their status with sip show peers In case of using ARA (realtime

[Asterisk-Users] show register users

Hi all, As I know, I can show registered users in CLI using sip show users/sip show registry. In case of using ARA (realtime mode), there is no record shown after issuing the above command in CLI. How can I know the register users in the system if realtime mode is using? Thanks, unplug

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

[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:

[Asterisk-Users] Sipura register with FWD every 60sec

Despite the fact that Sipura has registration set to Register Expire: 3600 Line1 register with FWD every 60sec. How to change it?(it seems to me Line1 has incorrect time as well). -- #Joseph ___ --Bandwidth and Colocation provided by

[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

[Asterisk-Users] IAX2 register problem

Hi guys, I am trying to register IP IAX2 phone to our Asterisk server. this is what I see on traffic debug between the asterisk server and IP phone. I do not see anything in asterisk console. Can somebody give me hints what could be the reason that phone is not registering? Thank you in

Re: [Asterisk-Users] IAX2 register problem

On Mar 3, 2006, at 9:57 AM, Bartosz Jozwiak wrote: Hi guys,   I am trying to register IP IAX2 phone to our Asterisk server. this is what I see on traffic debug between the asterisk server and IP phone. I do not see anything in asterisk console.   Can somebody give me hints what could be the

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