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Gamma SIP Lines

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DaveAvaya123

DaveAvaya123

Programmer
May 22, 2012
91
GB
Hello

I am having trouble with Gamma sip lines and need some help. The customer has assured sip and I have connected there router to LAN2. I have outgoing calls with no Audio and no incoming. The Avaya IP office is the latest version. Has anyone done anything with assured sip and do I need a firewall after there router?

Cheers
 
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Gamma don't use proper/any SBCs AFAIK, so you will need to do some port forwarding to allow SIP to work usually. Start with 5060 and 5061 and seer what you get, they also quote a large RTP range to be forwarded but I would try and avoid doing that. Often STUN is required though Gamma claim to "hate" it :)

 
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You need to call gamma and tell them the LAN2 IP address, they can then log into the router and do the port forwarding for you.

They tell you it needs to be done via a form to process properly but you can usually sweet talk them into doing it whilst you're on the phone if you tell them you need to get the service up ASAP.

APSS/ACIS/ACSS-SME
not arrogant, just succinct.
 
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Hi

I have just done a assured 10. As they provide a cisco router I got them to assigned a static internal IP to match the IP range of the IP office. also got them to turn Off DHCP and also port forward the relevant ports to the IP office. They will want to lock down source address etc.So the IPDC from Gamma and your stun server. All worked no problem.
 
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I have spoken to gamma and now have external calls working for outbound but nothing inbound.

INVITE sip:01142479125@192.168.1.240;user=phone SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: <sip:01142479125@88.215.61.201:5060;user=phone>
From: <sip:07876474762@88.215.61.201>;tag=3620632998-543669
P-Asserted-Identity: <sip:07876474762@88.215.61.201;user=phone>
Call-ID: 219424-3620632998-543661@MSX24.mydomain.com
CSeq: 1 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Via: SIP/2.0/UDP 88.215.61.201:5060;branch=z9hG4bK23a91751052c59b37a674bf2232c3515
Contact: <sip:07876474762@88.215.61.201:5060>
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 215

v=0
o=MSX24 8050197444223917890 1 IN IP4 88.215.61.201
s=sip call
c=IN IP4 88.215.61.202
t=0 0
m=audio 35076 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
935215mS SIP Rx: UDP 88.215.61.201:5060 -> 192.168.1.240:5060
INVITE sip:01142479125@192.168.1.240;user=phone SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: <sip:01142479125@88.215.61.201:5060;user=phone>
From: <sip:07876474762@88.215.61.201>;tag=3620632998-543669
P-Asserted-Identity: <sip:07876474762@88.215.61.201;user=phone>
Call-ID: 219424-3620632998-543661@MSX24.mydomain.com
CSeq: 1 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Via: SIP/2.0/UDP 88.215.61.201:5060;branch=z9hG4bK23a91751052c59b37a674bf2232c3515
Contact: <sip:07876474762@88.215.61.201:5060>
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 215

v=0
o=MSX24 8050197444223917890 1 IN IP4 88.215.61.201
s=sip call
c=IN IP4 88.215.61.202
t=0 0
m=audio 35076 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
936224mS SIP Rx: UDP 88.215.61.201:5060 -> 192.168.1.240:5060
INVITE sip:01142479125@192.168.1.240;user=phone SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: <sip:01142479125@88.215.61.201:5060;user=phone>
From: <sip:07876474762@88.215.61.201>;tag=3620632998-543669
P-Asserted-Identity: <sip:07876474762@88.215.61.201;user=phone>
Call-ID: 219424-3620632998-543661@MSX24.mydomain.com
CSeq: 1 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Via: SIP/2.0/UDP 88.215.61.201:5060;branch=z9hG4bK23a91751052c59b37a674bf2232c3515
Contact: <sip:07876474762@88.215.61.201:5060>
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 215

v=0
o=MSX24 8050197444223917890 1 IN IP4 88.215.61.201
s=sip call
c=IN IP4 88.215.61.202
t=0 0
m=audio 35076 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
936815mS SIP Rx: UDP 88.215.61.201:5060 -> 192.168.1.240:5060
INVITE sip:01142479125@192.168.1.240;user=phone SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: <sip:01142479125@88.215.61.201:5060;user=phone>
From: <sip:07876474762@88.215.61.201>;tag=3620633000-640734
P-Asserted-Identity: <sip:07876474762@88.215.61.201;user=phone>
Call-ID: 219458-3620633000-640726@MSX24.mydomain.com
CSeq: 1 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Via: SIP/2.0/UDP 88.215.61.201:5060;branch=z9hG4bK5b79eafa8c9daa724533bb8aeeef5f98
Contact: <sip:07876474762@88.215.61.201:5060>
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 215

v=0
o=MSX24 8020119524244023630 1 IN IP4 88.215.61.201
s=sip call
c=IN IP4 88.215.61.202
t=0 0
m=audio 35144 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
937331mS SIP Rx: UDP 88.215.61.201:5060 -> 192.168.1.240:5060
INVITE sip:01142479125@192.168.1.240;user=phone SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: <sip:01142479125@88.215.61.201:5060;user=phone>
From: <sip:07876474762@88.215.61.201>;tag=3620633000-640734
P-Asserted-Identity: <sip:07876474762@88.215.61.201;user=phone>
Call-ID: 219458-3620633000-640726@MSX24.mydomain.com
CSeq: 1 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Via: SIP/2.0/UDP 88.215.61.201:5060;branch=z9hG4bK5b79eafa8c9daa724533bb8aeeef5f98
Contact: <sip:07876474762@88.215.61.201:5060>
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 215

v=0
o=MSX24 8020119524244023630 1 IN IP4 88.215.61.201
s=sip call
c=IN IP4 88.215.61.202
t=0 0
m=audio 35144 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
938318mS SIP Rx: UDP 88.215.61.201:5060 -> 192.168.1.240:5060
INVITE sip:01142479125@192.168.1.240;user=phone SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: <sip:01142479125@88.215.61.201:5060;user=phone>
From: <sip:07876474762@88.215.61.201>;tag=3620633000-640734
P-Asserted-Identity: <sip:07876474762@88.215.61.201;user=phone>
Call-ID: 219458-3620633000-640726@MSX24.mydomain.com
CSeq: 1 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Via: SIP/2.0/UDP 88.215.61.201:5060;branch=z9hG4bK5b79eafa8c9daa724533bb8aeeef5f98
Contact: <sip:07876474762@88.215.61.201:5060>
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 215

v=0
o=MSX24 8020119524244023630 1 IN IP4 88.215.61.201
s=sip call
c=IN IP4 88.215.61.202
t=0 0
m=audio 35144 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
938903mS SIP Rx: UDP 88.215.61.201:5060 -> 192.168.1.240:5060
INVITE sip:01142479125@192.168.1.240;user=phone SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: <sip:01142479125@88.215.61.201:5060;user=phone>
From: <sip:07876474762@88.215.61.201>;tag=3620633002-720377
P-Asserted-Identity: <sip:07876474762@88.215.61.201;user=phone>
Call-ID: 219491-3620633002-720368@MSX24.mydomain.com
CSeq: 1 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Via: SIP/2.0/UDP 88.215.61.201:5060;branch=z9hG4bK90578a4055e28b848904e54217d02a46
Contact: <sip:07876474762@88.215.61.201:5060>
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 215

v=0
o=MSX24 8052106803637712007 1 IN IP4 88.215.61.201
s=sip call
c=IN IP4 88.215.61.202
t=0 0
m=audio 35210 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
939394mS SIP Rx: UDP 88.215.61.201:5060 -> 192.168.1.240:5060
INVITE sip:01142479125@192.168.1.240;user=phone SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: <sip:01142479125@88.215.61.201:5060;user=phone>
From: <sip:07876474762@88.215.61.201>;tag=3620633002-720377
P-Asserted-Identity: <sip:07876474762@88.215.61.201;user=phone>
Call-ID: 219491-3620633002-720368@MSX24.mydomain.com
CSeq: 1 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Via: SIP/2.0/UDP 88.215.61.201:5060;branch=z9hG4bK90578a4055e28b848904e54217d02a46
Contact: <sip:07876474762@88.215.61.201:5060>
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 215

v=0
o=MSX24 8052106803637712007 1 IN IP4 88.215.61.201
s=sip call
c=IN IP4 88.215.61.202
t=0 0
m=audio 35210 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
940394mS SIP Rx: UDP 88.215.61.201:5060 -> 192.168.1.240:5060
INVITE sip:01142479125@192.168.1.240;user=phone SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: <sip:01142479125@88.215.61.201:5060;user=phone>
From: <sip:07876474762@88.215.61.201>;tag=3620633002-720377
P-Asserted-Identity: <sip:07876474762@88.215.61.201;user=phone>
Call-ID: 219491-3620633002-720368@MSX24.mydomain.com
CSeq: 1 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Via: SIP/2.0/UDP 88.215.61.201:5060;branch=z9hG4bK90578a4055e28b848904e54217d02a46
Contact: <sip:07876474762@88.215.61.201:5060>
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 215

v=0
o=MSX24 8052106803637712007 1 IN IP4 88.215.61.201
s=sip call
c=IN IP4 88.215.61.202
t=0 0
m=audio 35210 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

Any help would be amazing.
 
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Have you done an IP route to the Gamma IP address, you normally need to go one for the signalling address & the rtp address (.169 & .170 at the moment iirc)
 
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i have an ip route of 0.0.0.0 to the gateway to allow all traffic.
 
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use internal data for all 4 drop downs. the incoming and outgoing are 100 which match my sip line.
 
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I have tried it with * instead of internal data for a second line and then in the setup guide by avaya it does not have it so removed it and still the same issue.
 
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You need to post some screenshots of the config really, we will be toing and froing with replies for days at this rate :)

 
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all sorted. All I can say is a day wasted due to useless sip line provider. Thank you for your help.
 
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I spoke to soon. I have added the * in for local, contact and display name.

and still the same result. There was a divert in place for the main number but i was not informed this would stop all incomming. I have now have the divert removed and still the same.

I hate SIP!!! ISDN all the way. [mad]
 
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