[asterisk-users] Allison Smith AMA
For anyone interested, Allison Smith's AMA (not sure she's still around): http://www.reddit.com/r/IAmA/comments/2rrb7m/iama_professional_telephone_voice_ama/ -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] Asterisk 13.1.0/PJSIP peer IP address issue
Well, I thought it worked, but it actually doesn't--I am able to get the caller pick up the phone, but for some reason, I cannot hear anything on either side no matter who does the calling. Again, my two SIP phones are on the local 192.168.1.0/24 network (do not go over the Internet) and the Asterisk server is located in the same network (not accessed over the Internet). Any help is appreciated. Does the fact that Asterisk is running on a VirtualBox VM on the same machine as one of the SIP phones matter? I am able to access the ARI REST interface of the Asterisk server quite fine on the host machine. I suspect it has to do with RTP not being set up, but all the codec support is there. Here's a log for the SIP request from 192.168.1.50: --- Received SIP request (1229 bytes) from UDP:192.168.1.50:64009 --- INVITE sip:6002@192.168.1.139;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 146.115.163.234:64009 ;branch=z9hG4bK-d8754z-5803642ad92cbd00-1---d8754z- Max-Forwards: 70 Contact: sip:demo-alice@146.115.163.234:64009;transport=UDP To: sip:6002@192.168.1.139;transport=UDP From: sip:demo-alice@192.168.1.139;transport=UDP;tag=b661670b Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.21933 r21903 Authorization: Digest username=demo-alice,realm=asterisk,nonce=[removed],uri= sip:6002@192.168.1.139 ;transport=UDP,response=[removed],cnonce=[removed],nc=0001,qop=auth,algorithm=md5,opaque=[removed] Allow-Events: presence, kpml Content-Length: 245 v=0 o=Z 0 0 IN IP4 146.115.163.234 s=Z c=IN IP4 146.115.163.234 t=0 0 m=audio 8000 RTP/AVP 0 3 110 8 98 101 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- Transmitting SIP response (319 bytes) to UDP:192.168.1.50:64009 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 146.115.163.234:64009 ;rport=64009;received=192.168.1.50;branch=z9hG4bK-d8754z-5803642ad92cbd00-1---d8754z- Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE. From: sip:demo-alice@192.168.1.139;tag=b661670b To: sip:6002@192.168.1.139 CSeq: 2 INVITE Content-Length: 0 Any help is appreciated. A topology is shown below in ASCII. ( Big bad Internet ) GW/NAPT/Router | -- / \ || Host A Host B - - | Alice | | Bob | | 192.168.1.50 | | 192.168.1.149 | |---| |---| | Asterisk sr | |(VM) | | 192.168.1.239 | |---| On Thu, Jan 8, 2015 at 2:32 PM, Sonny Rajagopalan sonny.rajagopa...@gmail.com wrote: 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 sgriepent...@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
Re: [asterisk-users] Asterisk executable suddenly about 40KB larger - modules (Andres)
I would also start by putting an audit rule on the binary. Something like this: auditctl -w /usr/sbin/asterisk -p war -k asterisk-bin then you can get a report on who modified it and when by using: ausearch -f /usr/sbin/asterisk Its a start, but eventually you might need to monitor even keystrokes with pam_tty_audit.so to understand who is doing this: http://poorlydocumented.com/2014/05/enabling-pam_tty_audit-on-rhel-centos-o r-scientific-linux/ Thanks I'll keep that in mind. Just to report back, stopping pre-linking as detailed yesterday and setting immutable with chattr on the Asterisk executable on the Head Office box here appears to have solved the problem. The box did not crash this morning as it did the previous two days and is working fine... strange, but good. Previous to the problem starting on Tuesday, the box had been running fine for about three years 24/7 - so I might still have some kind of compromise going on. Anyway thanks for the assistance everyone Regards Stefan -- _ -- 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] SEMI OFF-TOPIC - Fail2ban
Hi list , someone on the list has seen this type of connection attempts in asterisk, fail2ban does not stop 2015-01-08 14:59:47] SECURITY[21515] res_security_log.c: SecurityEvent=ChallengeSent,EventTV=1420750787-386840,Severity=Informational,Service=SIP,EventVersion=1,AccountID=sip:100@173.230.133.20,SessionID=0x169f528,LocalAddress=IPV4/UDP/173.230.133.20/5060,RemoteAddress=IPV4/UDP/63.141.229.58/5078,Challenge=770e84a3 [2015-01-08 15:20:20] SECURITY[21515] res_security_log.c: SecurityEvent=ChallengeSent,EventTV=1420752020-854997,Severity=Informational,Service=SIP,EventVersion=1,AccountID=sip:102@173.230.133.20,SessionID=0x169f528,LocalAddress=IPV4/UDP/173.230.133.20/5060,RemoteAddress=IPV4/UDP/198.204.241.58/5074,Challenge=23965594 I modified the fail2ban with the filter, but still not detected asterisk.conf log_prefix= \[\]\s*(?:NOTICE|SECURITY)%(__pid_re)s:?(?:\[\S+\d*\])? \S+:\d* failregex = ^%(log_prefix)s Registration from '[^']*' failed for 'HOST(:\d+)?' - Wrong password$ ^%(log_prefix)s Registration from '[^']*' failed for 'HOST(:\d+)?' - No matching peer found$ ^%(log_prefix)s Registration from '[^']*' failed for 'HOST(:\d+)?' - Username/auth name mismatch$ ^%(log_prefix)s Registration from '[^']*' failed for 'HOST(:\d+)?' - Device does not match ACL$ ^%(log_prefix)s Registration from '[^']*' failed for 'HOST(:\d+)?' - Peer is not supposed to register$ ^%(log_prefix)s Registration from '[^']*' failed for 'HOST(:\d+)?' - ACL error \(permit/deny\)$ ^%(log_prefix)s Registration from '[^']*' failed for 'HOST(:\d+)?' - Not a local domain$ ^%(log_prefix)s Call from '[^']*' \(HOST:\d+\) to extension '\d+' rejected because extension not found in context 'default' \.$ ^%(log_prefix)s Host HOST failed to authenticate as '[^']*'$ ^%(log_prefix)s No registration for peer '[^']*' \(from HOST\)$ ^%(log_prefix)s Host HOST failed MD5 authentication for '[^']*' \([^)]+\)$ ^%(log_prefix)s Failed to authenticate (user|device) [^@]+@HOST\S*$ ^%(log_prefix)s (?:handle_request_subscribe: )?Sending fake auth rejection for (device|user) \d*sip:[^@]+@HOST;tag=\w+\S* $ ^%(log_prefix)s SecurityEvent=(FailedACL|InvalidAccountID|ChallengeResponseFailed|InvalidPassword),EventTV=[\d-]+,Severit y=[\w]+,Service=[\w]+,EventVersion=\d+,AccountID=\d+,SessionID=0x[\da-f]+,LocalAddress=IPV[46]/(UD|TC)P/[\da-fA-F:.]+/\d+,Rem oteAddress=IPV[46]/(UD|TC)P/HOST/\d+(,Challenge=\w+,ReceivedChallenge=\w+)?(,ReceivedHash=[\da-f]+)?$ ignoreregex = -- rickygm http://gnuforever.homelinux.com -- _ -- 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] queue reload command
Hi
I'm using asterisk 1.8
Does anyone know how to use the queue reload command. The built in help
doesn't really help.
queue reload {parameters|membe Reload queues, members, queue rules, or
parameters
Regards
Ish
--
Ishfaq Malik
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, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] queue reload command
That's what I would have guessed but it's not working:
[ish@??? ~]$ asterisk -rx 'queue show axon-all'
axon-all has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime, 0s
talktime), W:0, C:0, A:2, SL:0.0% within 20s
Members:
AXON200 (realtime) (Not in use) has taken no calls yet
AXON201 (realtime) (Not in use) has taken no calls yet
AXON202 (realtime) (Not in use) has taken no calls yet
AXON203 (realtime) (Not in use) has taken no calls yet
AXON204 (realtime) (In use) has taken no calls yet
AXON205 (realtime) (Not in use) has taken no calls yet
AXON206 (realtime) (Not in use) has taken no calls yet
AXON207 (realtime) (Not in use) has taken no calls yet
AXON208 (realtime) (Unavailable) has taken no calls yet
AXON209 (realtime) (Not in use) has taken no calls yet
AXON210 (realtime) (Unavailable) has taken no calls yet
AXON211 (realtime) (Unavailable) has taken no calls yet
AXON214 (realtime) (Not in use) has taken no calls yet
AXON221 (realtime) (Not in use) has taken no calls yet
AXON222 (realtime) (Not in use) has taken no calls yet
AXON223 (realtime) (Unavailable) has taken no calls yet
AXON225 (realtime) (Not in use) has taken no calls yet
AXON231 (realtime) (Unavailable) has taken no calls yet
AXON232 (realtime) (Not in use) has taken no calls yet
AXON233 (realtime) (Not in use) has taken no calls yet
No Callers
[ish@??? ~]$ asterisk -rx 'queue reload axon-all'
No such command 'queue reload axon-all' (type 'core show help queue reload
axon-all' for other possible commands)
On 8 January 2015 at 14:23, Andrew Colin and...@convergedgroup.net wrote:
Hi
queue reload(queue name) or queue reload all
for example
queue reload reception
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ishfaq Malik
*Sent:* Thursday, January 8, 2015 2:10 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] queue reload command
Hi
I'm using asterisk 1.8
Does anyone know how to use the queue reload command. The built in help
doesn't really help.
queue reload {parameters|membe Reload queues, members, queue rules, or
parameters
Regards
Ish
--
Ishfaq Malik
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, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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--
Ishfaq Malik
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, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
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:
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Re: [asterisk-users] queue reload command
Hi
queue reload(queue name) or queue reload all
for example
queue reload reception
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Thursday, January 8, 2015 2:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] queue reload command
Hi
I'm using asterisk 1.8
Does anyone know how to use the queue reload command. The built in help
doesn't really help.
queue reload {parameters|membe Reload queues, members, queue rules, or
parameters
Regards
Ish
--
Ishfaq Malik
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, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
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:
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[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@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
Re: [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.rajagopa...@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@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 use0 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
Re: [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 sgriepent...@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.rajagopa...@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@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 use0 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
