Thread: RE: nat=no |
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Hm, strange. I'm seeing this:
22:47:30.260577 IP 192.168.1.108.51791 > www1.pbxes.com.5060: SIP, length: 981
22:47:30.379100 IP www3.pbxes.com.27570 > 192.168.1.108.5060: SIP, length: 457
22:47:30.393256 IP www3.pbxes.com.27570 > 192.168.1.108.5060: SIP, length: 572
22:47:30.932572 IP www1.pbxes.com.5060 > 192.168.1.108.52082: SIP, length: 4
22:47:30.932663 IP www1.pbxes.com.5060 > 192.168.1.108.52146: SIP, length: 4
22:47:30.932705 IP www1.pbxes.com.5060 > 192.168.1.108.52061: SIP, length: 4
So I thought "Why does www1 not reply to the right port?" So, I specified www3.pbxes.com and not pbxes.com as the proxy. Are they configured the same? Is my nat=no not set on www1? Either way, we have registration with the impossible Cisco 7945 using www3
edit: but i can't make calls
edit 2:
More curious, you only reply on 506 for the initial registration request. Then you seem to revert to the (incorrect for UDP) behavior of replying symetrically.
registration stuff
23:13:28.127162 IP SEP0024C442AFCB.52736 > www3.pbxes.com.5060: SIP, length: 981
23:13:28.230148 IP www3.pbxes.com.27570 > SEP0024C442AFCB.5060: SIP, length: 457
23:13:28.244327 IP www3.pbxes.com.27570 > SEP0024C442AFCB.5060: SIP, length: 572
23:13:28.256626 IP SEP0024C442AFCB.51218 > www3.pbxes.com.5060: SIP, length: 1178
23:13:28.346176 IP www3.pbxes.com.27570 > SEP0024C442AFCB.5060: SIP, length: 457
23:13:28.361915 IP www3.pbxes.com.27570 > SEP0024C442AFCB.5060SIP, length: 544
23:13:28.715275 IP SEP0024C442AFCB.52832 > www3.pbxes.com.5060: SIP, length: 640
23:13:28.743531 IP SEP0024C442AFCB.51669 > www3.pbxes.com.5060: SIP, length: 1352
making a call (part of)
23:15:55.162948 IP (tos 0x78, ttl 45, id 0, offset 0, flags [DF], proto UDP (17), length 906)
www3.pbxes.com.5060 > SEP0024C442AFCB.51738: SIP, length: 878
SIP/2.0 404 Not Found
this isn't right. https://issues.asterisk.org/jira/browse/ASTERISK-17535 maybe?
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Thread: RE: nat=no |
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Anything obvious other than the 401? Would the nat change have made it 401 for any reason? I'll double check the config when I'm back at the phone but it looked ok
Jan 30 09:25:14 VERBOSE[33711] logger.c:
<-- SIP read
REGISTER sip:188.40.65.170 SIP/2.0
Via: SIP/2.0/UDP 86.152.69.250:5060;branch=z9hG4bKf8c5e7ff
From: <sip:xxxx-103@188.40.65.170>;tag=0024c442afcb006690b72be5-f3af97a3
To: <sip:xxx-103@188.40.65.170>
Call-ID: 0024c442-afcb0064-89fc9141-baab6417@86.152.69.250
Max-Forwards: 70
Date: Wed, 30 Jan 2013 09:25:01 GMT
CSeq: 199 REGISTER
User-Agent: Cisco-CP7945G/9.3.1
Contact: <sip:xxx-103@86.152.69.250:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0024c442afcb>";+u.sip!devicename.ccm.cisco.com="SEP0024C442AFCB";+u.sip!model.ccm.cisco.com="435"
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.0,X-cisco-xsi-8.5.1
Content-Length: 0
Reason: SIP;cause=200;text="cisco-alarm:20 Name=SEP0024C442AFCB Load=SIP45.9-3-1SR1-1S Last=phone-keypad"
Expires: 600
Jan 30 09:25:14 VERBOSE[33711] logger.c: --- (14 headers 0 lines)Jan 30 09:25:14 VERBOSE[33711] logger.c: --- (14 headers 0 lines)---
Jan 30 09:25:14 VERBOSE[33711] logger.c: Using latest REGISTER request as basis request
Jan 30 09:25:14 VERBOSE[33711] logger.c: Transmitting (no NAT)
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 86.152.69.250:5060;branch=z9hG4bKf8c5e7ff;received=88.198.69.250
From: <sip:xxx-103@188.40.65.170>;tag=0024c442afcb006690b72be5-f3af97a3
To: <sip:xxx-103@188.40.65.170>
Call-ID: 0024c442-afcb0064-89fc9141-baab6417@86.152.69.250
CSeq: 199 REGISTER
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:xxx-103@88.198.69.250:27570>
Content-Length: 0
---
Jan 30 09:25:14 VERBOSE[33711] logger.c: Transmitting (no NAT)
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 86.152.69.250:5060;branch=z9hG4bKf8c5e7ff;received=88.198.69.250
From: <sip:xxx-103@188.40.65.170>;tag=0024c442afcb006690b72be5-f3af97a3
To: <sip:xxx-103@188.40.65.170>;tag=as09a399ac
Call-ID: 0024c442-afcb0064-89fc9141-baab6417@86.152.69.250
CSeq: 199 REGISTER
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:xxx-103@88.198.69.250:27570>
WWW-Authenticate: Digest realm="pbxes.org", nonce="4da247670d1077b622e66a7262e748292e4bf08d"
Content-Length: 0
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Thread: RE: nat=no |
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Great thanks. It almost works, I'm getting responses from you on the right port now, but I'm getting a 401 when i try to register.
SEP0024C442AFCB.52859 > www1.pbxes.com.5060: SIP, length: 981 REGISTER sip:188.40.65.170 SIP/2.0
www3.pbxes.com.27570 > SEP0024C442AFCB.5060: SIP, length: 457
SIP/2.0 100 Trying
www3.pbxes.com.27570 > SEP0024C442AFCB.5060: SIP, length: 572
SIP/2.0 401 Unauthorized
I've checked the config and it looks ok... but it's pretty late so who knows!
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Thread: RE: nat=no |
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I think it'll disable the symmetric NAT won't it? So that asterisk will then respect the contact header instead of replying to the same port the packet was sent on. Trying to get my cisco phone to work...
edit:
well, asper the rfcs it should disable it. I've set up a freepbx on amazons cloud and have successfully got my cisco to register and make and receive calls. both devices are behind natted, extensions is "nat=no"
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Thread: RE: nat=no |
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any chance of being able to configure the nat option for some extensions?
Or, if you're feeling generous, just set my extension 103 to be "no"
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Thread: RE: cisco 7945 / milkfish / proxy |
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Hi all,
I'm hoping someone more knowledgeable than me can help me out with configuring my cisc0 7945. I'm running ddwrt with milkfish on which is supposed to act like a sip proxy and mangle the data that the cisco sends so that it might actually work.
However, I can't get it to work. I cant see it showing in the milkfish at all. <outboundProxy>192.168.1.1</outboundProxy>
<outboundProxyPort>5060</outboundProxyPort>
Should work, shouldn't it?
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Thread: RE: Cisco 7941G "Registering" and Symetric NAT |
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uh, these are a pain. I'd suggest that cisco made them annoyingly deliberately to make you buy their software for a lan environment.
anyway, has anyone managed to get one to register with pbxes? I'm having all sorts of trouble.
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Thread: RE: Queue Statistics |
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Hi
My queue statistics dont apepar to be working. It's just a blank screen. What should show there?
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Thread: RE: Dutch voice in queue doesn't work |
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Did you ever fix this? I'm set to "British English" which works fine other than in queues where I get an American woman instead.
edit: check the language in the appropriate trunk. I still has it as "english" and not "british". English now appears to mean American
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Thread: RE: Trunk override with Custom dial pattern - doesnt work? "0|." becomes "0", an |
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Hi
Is that the only rule you have in that particular route? As far as I can tell it scans the list of rules and if any overlap it deletes the second one. "0|." is a valid rule, so I'm not sure why it wouldn't like it.
The thing is, this would send any number beginning with a 0 over that trunk. Is that what you want?
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Thread: RE: SIP URI calling |
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Hi,
Yeah I thought that was the problem. I can dial it directly from my softphone though. I can't exclude it from any dial patterns with any ease as it's a regular landline number here in the uk. Any other ideas?
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Thread: RE: SIP URI calling |
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Just to bump this really old thread but I'm having trouble too. I have a trunk, that goes to a ring group, that has one extension in which calls the sip uri. it hangs up immediately. I can call the SIP uri from my softphone fine.
Aug 25 11:05:33 VERBOSE[76241] logger.c: -- Called numberxxx@sip.gradwell.net
Aug 25 11:05:34 VERBOSE[66817] chan_sip.c: SIP response 100 to standard invite
Aug 25 11:05:34 VERBOSE[66817] chan_sip.c: SIP response 407 to standard invite
Aug 25 11:05:34 NOTICE[66817] chan_sip.c: Failed to authenticate on INVITE to '"My CID" <sip:mycidxxx@88.198.69.250:27504>;tag=as1d6c5d5a'
Aug 25 11:05:34 VERBOSE[76241] logger.c: -- SIP/sip.gradwell.net-71c5 is circuit-busy
edit:
to be clear numberxx is just a number. I think that pbxes is trying to dial the number using sip.gradwell.net as a proxy, rather than assuming it's on that system. If that makes sense...?
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