Matt Scott
2005-May-17 09:59 UTC
[Asterisk-Users] multiple sip accounts from same sip registrar
Dear all, I have an asterisk sip issue which I don't believe is unique. I use a registrar (sipgate.co.uk) where I have 3 different accounts. These accounts provide me with three seperate local phone numbers which allow me to allocate them to seperate users. By using just one of these accounts I can set asterisk up to send and receive calls no problem. However, when I start to introduce an additional account I start to run into problems. if I do a 'sip show peers' with a good config I think it may outline the problem sip show peers Name/username Host Dyn Nat ACL Mask Port Status 1005/1005 (Unspecified) D 255.255.255.255 0 Unmonitored 1004/1004 (Unspecified) D 255.255.255.255 0 Unmonitored 1003/1003 (Unspecified) D 255.255.255.255 0 Unmonitored 1002/1002 10.0.0.52 D 255.255.255.255 5060 Unmonitored 1001/1001 10.0.0.51 D 255.255.255.255 5060 Unmonitored sipgate1/321**** 217.10.79.219 N 255.255.255.255 5060 OK (52 ms) I think it maybe a host specific ip address which must be in a table somewhere in asterisk. I have tried setting it up as a peer and dynamic but still no joy. Is there a limitation to this within asterisk. I have provided a sip.conf below (adjusted), will I need to implement a SER box (more things to learn which is all good provided it sorts my problem) [general] port = 5060 bindaddr = 0.0.0.0 disallow=all allow=ulaw allow=alaw allow=gsm ; register => *********.******@sipgate.co.uk/******* register => ***:***********@sipgate.co.uk/****** [sipgate1] type=friend context=from-sipgate1 fromuser=****** username=**** authuser=***** secret=****** host=sipgate.co.uk fromdomain=sipgate.co.uk nat=yes dtmfmode=info qualify=yes insecure=very canreinvite=no ; [sipgate2] type=friend context=from-sipgate2 fromuser=********* username=****** authuser=******* secret=********* host=sipgate.co.uk fromdomain=sipgate.co.uk nat=yes dtmfmode=info qualify=yes insecure=very canreinvite=no ; [1001] type=friend username=1001 secret=***** host=dynamic dtmfmode=rfc2833 context=from-sipphones ;mailbox=1001 allow=alaw allow=ulaw kindest regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050517/a06513f9/attachment.htm
Matt Scott
2005-May-17 10:01 UTC
[Asterisk-Users] Re: multiple sip accounts from same sip registrar
----- Original Message ----- From: Matt Scott To: asterisk-users@lists.digium.com Sent: Tuesday, May 17, 2005 5:59 PM Subject: multiple sip accounts from same sip registrar Dear all, I have an asterisk sip issue which I don't believe is unique. I use a registrar (sipgate.co.uk) where I have 3 different accounts. These accounts provide me with three seperate local phone numbers which allow me to allocate them to seperate users. By using just one of these accounts I can set asterisk up to send and receive calls no problem. However, when I start to introduce an additional account I start to run into problems. if I do a 'sip show peers' with a good config I think it may outline the problem sip show peers Name/username Host Dyn Nat ACL Mask Port Status 1005/1005 (Unspecified) D 255.255.255.255 0 Unmonitored 1004/1004 (Unspecified) D 255.255.255.255 0 Unmonitored 1003/1003 (Unspecified) D 255.255.255.255 0 Unmonitored 1002/1002 10.0.0.52 D 255.255.255.255 5060 Unmonitored 1001/1001 10.0.0.51 D 255.255.255.255 5060 Unmonitored sipgate1/321**** 217.10.79.219 N 255.255.255.255 5060 OK (52 ms) I think it maybe a host specific ip address which must be in a table somewhere in asterisk. I have tried setting it up as a peer and dynamic but still no joy. Is there a limitation to this within asterisk. I have provided a sip.conf below (adjusted), will I need to implement a SER box (more things to learn which is all good provided it sorts my problem) [general] port = 5060 bindaddr = 0.0.0.0 disallow=all allow=ulaw allow=alaw allow=gsm ; register => *********.******@sipgate.co.uk/******* register => ***:***********@sipgate.co.uk/****** [sipgate1] type=friend context=from-sipgate1 fromuser=****** username=**** authuser=***** secret=****** host=sipgate.co.uk fromdomain=sipgate.co.uk nat=yes dtmfmode=info qualify=yes insecure=very canreinvite=no ; [sipgate2] type=friend context=from-sipgate2 fromuser=********* username=****** authuser=******* secret=********* host=sipgate.co.uk fromdomain=sipgate.co.uk nat=yes dtmfmode=info qualify=yes insecure=very canreinvite=no ; [1001] type=friend username=1001 secret=***** host=dynamic dtmfmode=rfc2833 context=from-sipphones ;mailbox=1001 allow=alaw allow=ulaw kindest regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050517/f162d4c8/attachment.htm
trixter http://www.0xdecafbad.com
2005-May-17 11:13 UTC
[Asterisk-Users] multiple sip accounts from same sip registrar
Have you run into the problem where calls inbound from any of the numbers will all goto the same context even if each as a seperate context defined in sip.conf ? I have that problem with a couple different providers (which makes me think its not the providers) and may even be related to this issue, or maybe not (I have multiple providers with the same proxy and all show in sip show peers and sip show registry as unique entries). On Tue, 2005-05-17 at 17:59 +0100, Matt Scott wrote:> Dear all, > > I have an asterisk sip issue which I don't believe is unique. > I use a registrar (sipgate.co.uk) where I have 3 different accounts. > These accounts provide me with three seperate local phone numbers > which allow me to allocate them to seperate users. > By using just one of these accounts I can set asterisk up to send and > receive calls no problem. > However, when I start to introduce an additional account I start to > run into problems. > > if I do a 'sip show peers' with a good config I think it may outline > the problem > > sip show peers > Name/username Host Dyn Nat ACL Mask > Port Status > 1005/1005 (Unspecified) D 255.255.255.255 > 0 Unmonitored > 1004/1004 (Unspecified) D 255.255.255.255 > 0 Unmonitored > 1003/1003 (Unspecified) D 255.255.255.255 > 0 Unmonitored > 1002/1002 10.0.0.52 D 255.255.255.255 > 5060 Unmonitored > 1001/1001 10.0.0.51 D 255.255.255.255 > 5060 Unmonitored > sipgate1/321**** 217.10.79.219 N 255.255.255.255 > 5060 OK (52 ms) > > I think it maybe a host specific ip address which must be in a table > somewhere in asterisk. > I have tried setting it up as a peer and dynamic but still no joy. > > Is there a limitation to this within asterisk. I have provided a > sip.conf below (adjusted), will I need to implement a SER box (more > things to learn which is all good provided it sorts my problem) > > [general] > port = 5060 > bindaddr = 0.0.0.0 > disallow=all > allow=ulaw > allow=alaw > allow=gsm > ; > register => *********.******@sipgate.co.uk/******* > register => ***:***********@sipgate.co.uk/****** > [sipgate1] > type=friend > context=from-sipgate1 > fromuser=****** > username=**** > authuser=***** > secret=****** > host=sipgate.co.uk > fromdomain=sipgate.co.uk > nat=yes > dtmfmode=info > qualify=yes > insecure=very > canreinvite=no > ; > [sipgate2] > type=friend > context=from-sipgate2 > fromuser=********* > username=****** > authuser=******* > secret=********* > host=sipgate.co.uk > fromdomain=sipgate.co.uk > nat=yes > dtmfmode=info > qualify=yes > insecure=very > canreinvite=no > ; > [1001] > type=friend > username=1001 > secret=***** > host=dynamic > dtmfmode=rfc2833 > context=from-sipphones > ;mailbox=1001 > allow=alaw > allow=ulaw > > kindest regards > _______________________________________________ > 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-- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050517/cfbf9dfb/attachment.pgp
I have an * 1.0.7 box that keeps loosing its registration to both of the other servers it registers with. When I start * it connects and registers fine and I can make calls. but after a few hours it shows the status as Auth. Sent but I can no longer make calls to the other servers. The other servers are not mine they are Providers servers. I am behind a nat with port forwarding, and am running on a xercom install that I upgraded to a full debian install. This has happened to me from day one. Any help would be great Thanks Joel -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.12 - Release Date: 5/17/2005
Peter Bowyer
2005-May-18 00:25 UTC
[Asterisk-Users] multiple sip accounts from same sip registrar
On 17/05/05, Matt Scott <matt@mgsnet.co.uk> wrote:> Dear all, > > I have an asterisk sip issue which I don't believe is unique. > I use a registrar (sipgate.co.uk) where I have 3 different accounts. > These accounts provide me with three seperate local phone numbers which > allow me to allocate them to seperate users. > By using just one of these accounts I can set asterisk up to send and > receive calls no problem. > However, when I start to introduce an additional account I start to run into > problems. > > if I do a 'sip show peers' with a good config I think it may outline the > problem > > sip show peers > Name/username Host Dyn Nat ACL Mask Port > Status > 1005/1005 (Unspecified) D 255.255.255.255 0 > Unmonitored > 1004/1004 (Unspecified) D 255.255.255.255 0 > Unmonitored > 1003/1003 (Unspecified) D 255.255.255.255 0 > Unmonitored > 1002/1002 10.0.0.52 D 255.255.255.255 5060 > Unmonitored > 1001/1001 10.0.0.51 D 255.255.255.255 5060 > Unmonitored > sipgate1/321**** 217.10.79.219 N 255.255.255.255 5060 > OK (52 ms)I'm not sure what you think the problem is, you haven't told us... but anyway, I haven't succeeded in sending sipgate inbound calls through separate contexts, but I deal with them all in a single context - the calls will arrive at an extension matching the individual sipgate username in the register command. Works for me and several others.... Peter -- Peter Bowyer Email: peter@bowyer.org Tel: +44 1296 768003 VoIP: sip:peter@bowyer.org