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TEL ALIAS not working in version 3.0.0.2998


kelvin
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hi Support,

 

i am make use of tel alias as global alias for inter branch call, so that branch extension able contact each others using this global alias but after upgrade to version 3.0.0.2998. It stop working, everytime call global alias pbx return not found message. kindly advice the solution. thx

 

regards,

kelvin

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Did you confirm in the documentation that that is how one is supposed to use the Global?

 

We had to change a couple of things with the alias. The calls must now be routed through a trunk to the other domain. The old way is not suitable for server farms where you have no idea on which server a domain (or tel:-alias) is physically located. As we catch up with the documentation we'll elaborate that on the Wiki in further detail.

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We had to change a couple of things with the alias. The calls must now be routed through a trunk to the other domain. The old way is not suitable for server farms where you have no idea on which server a domain (or tel:-alias) is physically located. As we catch up with the documentation we'll elaborate that on the Wiki in further detail.

 

for this case, what alternate way i can use to route inter branch calls. thx

 

regards,

kelvin

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hi support,

 

any update on above request? thx

 

regards,

kelvin

 

Treat calls to other domains just like calls to other companies. As a trunk you can use the outbound proxy "127.0.0.1" - which loops the request back to itself. Make sure that this trunk is a "global" trunk, then you need only one.

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Treat calls to other domains just like calls to other companies. As a trunk you can use the outbound proxy "127.0.0.1" - which loops the request back to itself. Make sure that this trunk is a "global" trunk, then you need only one.

 

hi support,

 

i using above method, and my snom return Authentication required with busy tone when try call interbranch extension. below is the log. thx

 

SIP/2.0 183 Ringing

Via: SIP/2.0/TLS 192.168.1.176:3101;branch=z9hG4bK-2v9nief449ot;rport=3101

From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=dkm152o0i5

To: <sip:365215@pbx.pgcomms.com.my;user=phone>;tag=539287bafe

Call-ID: 3c3a1fd5d921-1d8euqo6pq07

CSeq: 1 INVITE

Contact: <sip:102@192.168.1.10:5081;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pgcomms-PBX/3.0.0.2998

Require: 100rel

RSeq: 1

Content-Type: application/sdp

Content-Length: 433

 

v=0

o=- 18443 18443 IN IP4 192.168.1.10

s=-

c=IN IP4 192.168.1.10

t=0 0

m=audio 54216 RTP/AVP 18 3 2 0 8 9 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:iIE4YWtoEmL5VNDncRdIeA2SKR2GKZly23Qc+k83

a=rtpmap:18 g729/8000

a=fmtp:18 annexb=no

a=rtpmap:3 gsm/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

[7] 2008/09/04 17:26:40: SIP Rx udp:127.0.0.1:5080:

INVITE sip:65215@127.0.0.1:5080;user=phone SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-adf90a1753c88364b9213f3cfa2f4e48;rport

From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=15046

To: <sip:65215@127.0.0.1:5080;user=phone>

Call-ID: 7b24e046@pbx

CSeq: 18780 INVITE

Max-Forwards: 70

Contact: <sip:102@127.0.0.1:5080;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pgcomms-PBX/3.0.0.2998

P-Asserted-Identity: "Normala" <sip:102@pbx.pgcomms.com.my>

Content-Type: application/sdp

Content-Length: 331

 

v=0

o=- 20980 20980 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 58324 RTP/AVP 18 3 2 0 8 9 101

a=rtpmap:18 g729/8000

a=fmtp:18 annexb=no

a=rtpmap:3 gsm/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

[5] 2008/09/04 17:26:40: Received loopback request without tag

[7] 2008/09/04 17:26:40: UDP: Opening socket on port 57660

[7] 2008/09/04 17:26:40: UDP: Opening socket on port 57661

[5] 2008/09/04 17:26:40: Identify trunk (IP address/port match) 23

[9] 2008/09/04 17:26:40: Resolve 17492: aaaa udp 127.0.0.1 5080

[9] 2008/09/04 17:26:40: Resolve 17492: a udp 127.0.0.1 5080

[9] 2008/09/04 17:26:40: Resolve 17492: udp 127.0.0.1 5080

[7] 2008/09/04 17:26:40: SIP Tx udp:127.0.0.1:5080:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-adf90a1753c88364b9213f3cfa2f4e48;rport=5080

From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=15046

To: <sip:65215@127.0.0.1:5080;user=phone>;tag=c054a4dbeb

Call-ID: 7b24e046@pbx

CSeq: 18780 INVITE

Content-Length: 0

 

[7] 2008/09/04 17:26:40: SIP Rx udp:127.0.0.1:5080:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-adf90a1753c88364b9213f3cfa2f4e48;rport=5080

From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=15046

To: <sip:65215@127.0.0.1:5080;user=phone>;tag=c054a4dbeb

Call-ID: 7b24e046@pbx

CSeq: 18780 INVITE

Content-Length: 0

 

[9] 2008/09/04 17:26:40: Resolve 17493: aaaa udp 127.0.0.1 5080

[9] 2008/09/04 17:26:40: Resolve 17493: a udp 127.0.0.1 5080

[9] 2008/09/04 17:26:40: Resolve 17493: udp 127.0.0.1 5080

[7] 2008/09/04 17:26:40: SIP Tx udp:127.0.0.1:5080:

SIP/2.0 401 Authentication Required

Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-adf90a1753c88364b9213f3cfa2f4e48;rport=5080

From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=15046

To: <sip:65215@127.0.0.1:5080;user=phone>;tag=c054a4dbeb

Call-ID: 7b24e046@pbx

CSeq: 18780 INVITE

User-Agent: pgcomms-PBX/3.0.0.2998

WWW-Authenticate: Digest realm="pbx.pgcomms.com.my",nonce="4cf3b7a72fbee5868e7e410c0e4ee2e4",domain="sip:65215@127.0.0.1:5080;user=phone",algorithm=MD5

Content-Length: 0

 

[7] 2008/09/04 17:26:40: SIP Rx udp:127.0.0.1:5080:

SIP/2.0 401 Authentication Required

Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-adf90a1753c88364b9213f3cfa2f4e48;rport=5080

From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=15046

To: <sip:65215@127.0.0.1:5080;user=phone>;tag=c054a4dbeb

Call-ID: 7b24e046@pbx

CSeq: 18780 INVITE

User-Agent: pgcomms-PBX/3.0.0.2998

WWW-Authenticate: Digest realm="pbx.pgcomms.com.my",nonce="4cf3b7a72fbee5868e7e410c0e4ee2e4",domain="sip:65215@127.0.0.1:5080;user=phone",algorithm=MD5

Content-Length: 0

 

[7] 2008/09/04 17:26:40: Call 7b24e046@pbx#15046: Clear last INVITE

[9] 2008/09/04 17:26:40: Resolve 17494: url sip:127.0.0.1:5080

[9] 2008/09/04 17:26:40: Resolve 17494: udp 127.0.0.1 5080

[7] 2008/09/04 17:26:40: SIP Tx udp:127.0.0.1:5080:

ACK sip:65215@127.0.0.1:5080;user=phone SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-adf90a1753c88364b9213f3cfa2f4e48;rport

From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=15046

To: <sip:65215@127.0.0.1:5080;user=phone>;tag=c054a4dbeb

Call-ID: 7b24e046@pbx

CSeq: 18780 ACK

Max-Forwards: 70

Contact: <sip:102@127.0.0.1:5080;transport=udp>

P-Asserted-Identity: "Normala" <sip:102@pbx.pgcomms.com.my>

Content-Length: 0

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hi support,

 

i using above method, and my snom return Authentication required with busy tone when try call interbranch extension. below is the log. thx

 

.... try to use a ANI for the extension that wants to call the other name or a Prefix on the trunk. The PBX probably thinks the call comes from an extension, not from a trunk.

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.... try to use a ANI for the extension that wants to call the other name or a Prefix on the trunk. The PBX probably thinks the call comes from an extension, not from a trunk.

 

by default the ANI = caller extension no right? i want to display Caller extension so that we can identify the calls. by the way what is this message mean? "[5] 2008/09/05 14:06:22: Received loopback request without tag"

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