Sonny Rajagopalan
2015-Jan-08 16:15 UTC
[asterisk-users] Asterisk 13.1.0/PJSIP peer IP address issue
I am following the instructions in
https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality and I am
trying to make a call from extension Alice (6001) to extension for Bob
(6002). When I make the call, I can hear the ringing on Alice's phone
(caller), but Bob's phone (callee) doesn't ring, or show a call coming
in
from Alice. My setup and environment is as follows: Alice, Bob and Asterisk
all in the same 192.168.1.0/24 network, and they are able to register to
the Asterisk server running 13.1.0/PJSIP. The rest of the configuration is
the same as the aforementioned wiki page, but is shown here for clarity:
root at asterisk13FFP:/var/log/asterisk# more /etc/asterisk/extensions.conf
[from-internal]
exten=>6001,1,Dial(PJSIP/demo-alice)
exten=>6002,1,Dial(PJSIP/demo-bob)
exten=>6003,1,Answer()
same =>6003,n,Playback(hello-world)
same =>6003,n,Hangup()
What I do observe is that I when I request the output of pjsip show
endpoints, I get Contact information for the two SIP peers that have
registered different from their actual IP addresses. I suspect this has
something to do with their calls being routed elsewhere. If my assumption
is correct--how do I fix this? Alice should be at 192.168.1.50 and Bob
should be at 192.168.1.149, instead, they (both) show IP address
146.115.163.234. Any help is deeply appreciated. Thanks.
asterisk13FFP*CLI> pjsip show endpoints
Endpoint: <Endpoint/CID.....................................>
<State.....> <Channels.>
I/OAuth:
<AuthId/UserName...........................................................>
Aor: <Aor............................................>
<MaxContact>
Contact: <Aor/ContactUri...............................>
<Status....> <RTT(ms)..>
Transport: <TransportId........> <Type> <cos>
<tos>
<BindAddress..................>
Identify:
<Identify/Endpoint.........................................................>
Match: <ip/cidr.........................>
Channel: <ChannelId......................................>
<State.....> <Time(sec)>
Exten: <DialedExten...........> CLCID:
<ConnectedLineCID.......>
========================================================================================
Endpoint: demo-alice
Unavailable 0 of inf
InAuth: demo-alice/demo-alice
Aor: demo-alice 1
Contact: demo-alice/sip:demo-alice@*146.115.163.234*:38519 Unknown
nan
Endpoint: demo-bob Not in use
0 of inf
InAuth: demo-bob/demo-bob
Aor: demo-bob 1
Contact: demo-bob/sip:demo-bob@*146.115.163.234*:38321;tra Unknown
nan
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Scott Griepentrog
2015-Jan-08 16:48 UTC
[asterisk-users] Asterisk 13.1.0/PJSIP peer IP address issue
It would appear that you have the Asterisk server on a public IP address, your two endpoints are behind a NAT, and you have rewrite_contact enabled in pjsip.conf. In which case, what you are seeing is correct. In order to be able to send a call to an extension where it is behind NAT, Asterisk must update the contact to have the current IP and port that the phone registered via (i.e. the WAN IP of the NAT, and the WAN port that it is retaining state for). On Thu, Jan 8, 2015 at 10:15 AM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote:> I am following the instructions in > https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality and I > am trying to make a call from extension Alice (6001) to extension for Bob > (6002). When I make the call, I can hear the ringing on Alice's phone > (caller), but Bob's phone (callee) doesn't ring, or show a call coming in > from Alice. My setup and environment is as follows: Alice, Bob and Asterisk > all in the same 192.168.1.0/24 network, and they are able to register to > the Asterisk server running 13.1.0/PJSIP. The rest of the configuration is > the same as the aforementioned wiki page, but is shown here for clarity: > > root at asterisk13FFP:/var/log/asterisk# more /etc/asterisk/extensions.conf > [from-internal] > exten=>6001,1,Dial(PJSIP/demo-alice) > exten=>6002,1,Dial(PJSIP/demo-bob) > exten=>6003,1,Answer() > same =>6003,n,Playback(hello-world) > same =>6003,n,Hangup() > > > What I do observe is that I when I request the output of pjsip show > endpoints, I get Contact information for the two SIP peers that have > registered different from their actual IP addresses. I suspect this has > something to do with their calls being routed elsewhere. If my assumption > is correct--how do I fix this? Alice should be at 192.168.1.50 and Bob > should be at 192.168.1.149, instead, they (both) show IP address > 146.115.163.234. Any help is deeply appreciated. Thanks. > > asterisk13FFP*CLI> pjsip show endpoints > > Endpoint: <Endpoint/CID.....................................> > <State.....> <Channels.> > I/OAuth: > <AuthId/UserName...........................................................> > Aor: <Aor............................................> > <MaxContact> > Contact: <Aor/ContactUri...............................> > <Status....> <RTT(ms)..> > Transport: <TransportId........> <Type> <cos> <tos> > <BindAddress..................> > Identify: > <Identify/Endpoint.........................................................> > Match: <ip/cidr.........................> > Channel: <ChannelId......................................> > <State.....> <Time(sec)> > Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......> > > ========================================================================================> > Endpoint: demo-alice > Unavailable 0 of inf > InAuth: demo-alice/demo-alice > Aor: demo-alice 1 > Contact: demo-alice/sip:demo-alice@*146.115.163.234*:38519 > Unknown nan > > Endpoint: demo-bob Not in > use 0 of inf > InAuth: demo-bob/demo-bob > Aor: demo-bob 1 > Contact: demo-bob/sip:demo-bob@*146.115.163.234*:38321;tra > Unknown nan > > > -- > _____________________________________________________________________ > -- 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 >-- [image: Digium logo] Scott Griepentrog Digium, Inc ? Software Developer 445 Jan Davis Drive NW ? Huntsville, AL 35806 ? US direct/fax: +1 256 428 6239 ? mobile: +1 256 580 6090 Check us out at: http://digium.com ? http://asterisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150108/11475a27/attachment.html>
Sonny Rajagopalan
2015-Jan-08 19:32 UTC
[asterisk-users] Asterisk 13.1.0/PJSIP peer IP address issue
Thank you for your note, Scott. I set rewrite_contact=yes for both contacts, and I also had to do remove_existing=yes because I had to remove the existing contact information (max_contacts = 1 was preventing new contact information) using pjsip qualify demo-alice etc., after which the right IP addresses showed in pjsip show endpoints. Anyway, it works as expected now, I think. My pjsip.conf is now [transport-udp] type=transport protocol=udp bind=0.0.0.0 local_net=192.168.1.0/24 ;Templates for the necessary config sections [endpoint_internal](!) type=endpoint context=from-internal disallow=all allow=ulaw [auth_userpass](!) type=auth auth_type=userpass [aor_dynamic](!) type=aor max_contacts=1 remove_existing=yes ;Definitions for our phones, using the templates above [demo-alice](endpoint_internal) auth=demo-alice aors=demo-alice mailboxes=box_a rewrite_contact=yes [demo-alice](auth_userpass) password=demo-alice ; put a strong, unique password here instead username=demo-alice [demo-alice](aor_dynamic) [demo-bob](endpoint_internal) auth=demo-bob aors=demo-bob mailboxes=box_b rewrite_contact=yes [demo-bob](auth_userpass) password=demo-bob ; put a strong, unique password here instead username=demo-bob [demo-bob](aor_dynamic) Thank you for your help! On Thu, Jan 8, 2015 at 11:48 AM, Scott Griepentrog <sgriepentrog at digium.com> wrote:> It would appear that you have the Asterisk server on a public IP address, > your two endpoints are behind a NAT, and you have rewrite_contact enabled > in pjsip.conf. > > In which case, what you are seeing is correct. In order to be able to > send a call to an extension where it is behind NAT, Asterisk must update > the contact to have the current IP and port that the phone registered via > (i.e. the WAN IP of the NAT, and the WAN port that it is retaining state > for). > > On Thu, Jan 8, 2015 at 10:15 AM, Sonny Rajagopalan < > sonny.rajagopalan at gmail.com> wrote: > >> I am following the instructions in >> https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality and I >> am trying to make a call from extension Alice (6001) to extension for Bob >> (6002). When I make the call, I can hear the ringing on Alice's phone >> (caller), but Bob's phone (callee) doesn't ring, or show a call coming in >> from Alice. My setup and environment is as follows: Alice, Bob and Asterisk >> all in the same 192.168.1.0/24 network, and they are able to register to >> the Asterisk server running 13.1.0/PJSIP. The rest of the configuration is >> the same as the aforementioned wiki page, but is shown here for clarity: >> >> root at asterisk13FFP:/var/log/asterisk# more /etc/asterisk/extensions.conf >> [from-internal] >> exten=>6001,1,Dial(PJSIP/demo-alice) >> exten=>6002,1,Dial(PJSIP/demo-bob) >> exten=>6003,1,Answer() >> same =>6003,n,Playback(hello-world) >> same =>6003,n,Hangup() >> >> >> What I do observe is that I when I request the output of pjsip show >> endpoints, I get Contact information for the two SIP peers that have >> registered different from their actual IP addresses. I suspect this has >> something to do with their calls being routed elsewhere. If my assumption >> is correct--how do I fix this? Alice should be at 192.168.1.50 and Bob >> should be at 192.168.1.149, instead, they (both) show IP address >> 146.115.163.234. Any help is deeply appreciated. Thanks. >> >> asterisk13FFP*CLI> pjsip show endpoints >> >> Endpoint: <Endpoint/CID.....................................> >> <State.....> <Channels.> >> I/OAuth: >> <AuthId/UserName...........................................................> >> Aor: <Aor............................................> >> <MaxContact> >> Contact: <Aor/ContactUri...............................> >> <Status....> <RTT(ms)..> >> Transport: <TransportId........> <Type> <cos> <tos> >> <BindAddress..................> >> Identify: >> <Identify/Endpoint.........................................................> >> Match: <ip/cidr.........................> >> Channel: <ChannelId......................................> >> <State.....> <Time(sec)> >> Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......> >> >> ========================================================================================>> >> Endpoint: demo-alice >> Unavailable 0 of inf >> InAuth: demo-alice/demo-alice >> Aor: demo-alice 1 >> Contact: demo-alice/sip:demo-alice@*146.115.163.234*:38519 >> Unknown nan >> >> Endpoint: demo-bob Not in >> use 0 of inf >> InAuth: demo-bob/demo-bob >> Aor: demo-bob 1 >> Contact: demo-bob/sip:demo-bob@*146.115.163.234*:38321;tra >> Unknown nan >> >> >> -- >> _____________________________________________________________________ >> -- 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 >> > > > > -- > [image: Digium logo] > Scott Griepentrog > Digium, Inc ? Software Developer > 445 Jan Davis Drive NW ? Huntsville, AL 35806 ? US > direct/fax: +1 256 428 6239 ? mobile: +1 256 580 6090 > 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 >-------------- next part -------------- An HTML attachment was scrubbed... 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