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Thread: RE: nat=no
baz

Replies: 8
Views: 15881

RE: nat=no 30.01.2013 23:54 Forum: Feature Requests

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 smile

edit: but i can't make calls unglücklich

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?

Thread: RE: Google Voice Trunks not working, Encryption Failure
baz

Replies: 3
Views: 10195

RE: Google Voice Trunks not working, Encryption Failure 30.01.2013 18:56 Forum: Bugs

try now Augenzwinkern

Thread: RE: nat=no
baz

Replies: 8
Views: 15881

RE: nat=no 30.01.2013 11:34 Forum: Feature Requests

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

Thread: RE: nat=no
baz

Replies: 8
Views: 15881

RE: nat=no 30.01.2013 02:18 Forum: Feature Requests

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!

Thread: RE: nat=no
baz

Replies: 8
Views: 15881

RE: nat=no 28.01.2013 00:12 Forum: Feature Requests

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"

Thread: RE: nat=no
baz

Replies: 8
Views: 15881

nat=no 26.01.2013 18:01 Forum: Feature Requests

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" smile

Thread: RE: cisco 7945 / milkfish / proxy
baz

Replies: 1
Views: 7693

cisco 7945 / milkfish / proxy 26.01.2013 10:42 Forum: Terminal Equipment

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?

Thread: RE: Cisco 7941G "Registering" and Symetric NAT
baz

Replies: 7
Views: 31443

RE: Cisco 7941G "Registering" and Symetric NAT 25.01.2013 00:45 Forum: Terminal Equipment

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.

Thread: RE: Queue Statistics
baz

Replies: 1
Views: 3671

Queue Statistics 18.01.2013 18:46 Forum: Bugs

Hi

My queue statistics dont apepar to be working. It's just a blank screen. What should show there?

Thread: RE: Dutch voice in queue doesn't work
baz

Replies: 1
Views: 7020

RE: Dutch voice in queue doesn't work 16.01.2013 14:43 Forum: Queues, Digital Receptionist, Faxmail, Voicemail and Ring Groups

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 Augenzwinkern

Thread: RE: how do I tell which DID is calling?
baz

Replies: 4
Views: 11426

RE: how do I tell which DID is calling? 20.12.2012 17:13 Forum: Providers

So you have an inbound route called semick0110-200 that goes to extension 200? I can't see why the AA would answer that. Strange.

Thread: RE: Multiple Phones/Devices on one single extension?
baz

Replies: 7
Views: 11963

RE: Multiple Phones/Devices on one single extension? 20.12.2012 17:07 Forum: Miscellaneous

Hmm, I'm pretty sure it registers both of my phones to one extension... Obvisouly I can only answer one of them. I'll try later tonight (if I remember).

Thread: RE: Two Things (Multiple Phone Numbers and G729)
baz

Replies: 2
Views: 4759

RE: Two Things (Multiple Phone Numbers and G729) 20.12.2012 17:05 Forum: Miscellaneous

You'll need two trunks

Thread: RE: how do I tell which DID is calling?
baz

Replies: 4
Views: 11426

RE: how do I tell which DID is calling? 07.12.2012 18:03 Forum: Providers

I tihnk you'd have to have a seperate trunk for the individual numbers to do that.

Thread: RE: Trunk override with Custom dial pattern - doesnt work? "0|." becomes "0", an
baz

Replies: 4
Views: 9082

RE: Trunk override with Custom dial pattern - doesnt work? "0|." becomes "0", an 29.11.2012 00:07 Forum: Bugs

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?

Thread: RE: SIP URI calling
baz

Replies: 16
Views: 66069

RE: SIP URI calling 29.08.2012 09:02 Forum: Feature Requests

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?

Thread: RE: I can't forward call with SIP URI
baz

Replies: 2
Views: 8177

RE: I can't forward call with SIP URI 25.08.2012 12:16 Forum: Bugs

Any luck with this? Whats in the logs?

Thread: RE: SIP URI calling
baz

Replies: 16
Views: 66069

RE: SIP URI calling 25.08.2012 12:07 Forum: Feature Requests

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...?

Thread: RE: Outbound routing - dial pattern to identify based on lenght
baz

Replies: 1
Views: 11125

RE: Outbound routing - dial pattern to identify based on lenght 15.08.2012 12:13 Forum: PBXes PRO

can you cliarify? A dialing rul of xxxxx should activate on any 5 digit number. Are you saying 5xxxx (4x) doens't work as expected?

Thread: RE: G729 Supported Sip Client for Android
baz

Replies: 1
Views: 11268

RE: G729 Supported Sip Client for Android 01.02.2012 12:02 Forum: Terminal Equipment

csipsimple
though it's a bit buggy with TCP and pbxes, which they blame on pbxes not supporting tcp correctly.

Showing posts 1 to 20 of 136 results Pages (7): [1] 2 3 next » ... last »

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