| 03:13.56 | *** join/#asterisk-doc snuff-home (n=snuffy@C-59-101-204-236.bur.connect.net.au) |
| 08:12.56 | *** join/#asterisk-doc DiscoKing (n=123@hlfxns0161w-142068043110.pppoe-dynamic.ns.aliant.net) |
| 08:13.10 | *** part/#asterisk-doc DiscoKing (n=123@hlfxns0161w-142068043110.pppoe-dynamic.ns.aliant.net) |
| 17:23.40 | lmadsen | file: ping! |
| 17:23.46 | file | no. |
| 17:23.51 | lmadsen | yes! |
| 17:25.35 | lmadsen | SIP options: replace replaces -- I get that on one of my boxes, but not another, and they have the same config. I think that is causing my 481 issues.... because the REFER packet I'm getting on the box that works does not have the 'replaces' stuff in 'sip show settings', and the REFER has the username@otherbox.tld of the SIP username connection between the two boxes. But the boxes with the 'replaces' changes the REFER |
| 17:25.36 | lmadsen | packet it seems to be phonenumber@otherbox.tld....... |
| 17:25.43 | lmadsen | so I'm wondering how the heck to turn off the replaces.... |
| 17:27.21 | file | wha? you can't... |
| 17:27.38 | lmadsen | ya... I didn't see an option... but I'm confused why one has it setup... and the other not so much |
| 17:28.14 | lmadsen | when it works... I get this.... |
| 17:28.16 | lmadsen | [Feb 23 11:17:22] DEBUG[13622]: chan_sip.c:11039 sip_dump_history: * SIP Call |
| 17:28.16 | lmadsen | [Feb 23 11:17:22] DEBUG[13622]: chan_sip.c:11042 sip_dump_history: 001. Rx INVITE / 102 INVITE / sip:PHONE-4001@10.19.18.235 |
| 17:28.16 | lmadsen | [Feb 23 11:17:22] DEBUG[13622]: chan_sip.c:11042 sip_dump_history: 002. NewChan Channel SIP/amsvoip03-00acd140 - from 13605cbb6c99226e7b85e89f0 |
| 17:28.16 | lmadsen | [Feb 23 11:17:22] DEBUG[13622]: chan_sip.c:11042 sip_dump_history: 003. TxResp SIP/2.0 / 102 INVITE - 100 Trying |
| 17:28.18 | lmadsen | [Feb 23 11:17:22] DEBUG[13622]: chan_sip.c:11042 sip_dump_history: 004. TxResp SIP/2.0 / 102 INVITE - 180 Ringing |
| 17:28.25 | lmadsen | [Feb 23 11:17:22] DEBUG[13622]: chan_sip.c:11042 sip_dump_history: 005. TxRespRel SIP/2.0 / 102 INVITE - 200 OK |
| 17:28.29 | lmadsen | [Feb 23 11:17:22] DEBUG[13622]: chan_sip.c:11042 sip_dump_history: 006. Rx ACK / 102 ACK / sip:PHONE-4001@10.19.18.235 |
| 17:28.31 | lmadsen | [Feb 23 11:17:22] DEBUG[13622]: chan_sip.c:11042 sip_dump_history: 007. Masq Old channel: SIP/0004F213EC3F-00b8f620<ZOMBIE> |
| 17:28.34 | lmadsen | [Feb 23 11:17:22] DEBUG[13622]: chan_sip.c:11042 sip_dump_history: 008. Masq (cont) ...new owner: SIP/amsvoip03-00acd140 |
| 17:28.39 | lmadsen | [Feb 23 11:17:22] DEBUG[13622]: chan_sip.c:11042 sip_dump_history: 009. Hangup Cause Normal Clearing |
| 17:28.42 | lmadsen | [Feb 23 11:17:22] DEBUG[13622]: chan_sip.c:11042 sip_dump_history: 010. SchedDestroy 32000 ms |
| 17:28.44 | lmadsen | [Feb 23 11:17:22] DEBUG[13622]: chan_sip.c:11042 sip_dump_history: 011. TxReqRel BYE / 102 BYE - -UNKNOWN- |
| 17:28.47 | lmadsen | [Feb 23 11:17:22] DEBUG[13622]: chan_sip.c:11042 sip_dump_history: 012. RTCPaudio Quality:ssrc=1082186115;themssrc=657654148;lp=215;rxjitter=0.00 |
| 17:28.50 | lmadsen | [Feb 23 11:17:22] DEBUG[13622]: chan_sip.c:11042 sip_dump_history: 013. Rx SIP/2.0 / 102 BYE / 200 OK |
| 17:28.57 | lmadsen | <PROTECTED> |
| 17:28.59 | lmadsen | when it fails... I get this |
| 17:29.01 | file | what you are referring to comes from the options of the remote SIP peer |
| 17:29.01 | lmadsen | <PROTECTED> |
| 17:29.03 | lmadsen | <PROTECTED> |
| 17:29.05 | lmadsen | [18:10] <lmadsen> [Feb 22 16:51:20] DEBUG[25722]: chan_sip.c:11042 sip_dump_history: 019. Rx REFER / 3 REFER / sip:8646169308@192.168.1.17 |
| 17:29.10 | lmadsen | [18:10] <lmadsen> [Feb 22 16:51:20] DEBUG[25722]: chan_sip.c:11042 sip_dump_history: 020. TxResp SIP/2.0 / 3 REFER - 202 Accepted |
| 17:29.13 | lmadsen | [18:10] <lmadsen> [Feb 22 16:51:20] DEBUG[25722]: chan_sip.c:11042 sip_dump_history: 021. TxReqRel NOTIFY / 103 NOTIFY - -UNKNOWN- |
| 17:29.16 | lmadsen | [18:10] <lmadsen> [Feb 22 16:51:20] DEBUG[25722]: chan_sip.c:11042 sip_dump_history: 022. Xfer Refer failed |
| 17:29.19 | lmadsen | so in the working box, it seems to masquerade the channels properly |
| 17:29.23 | lmadsen | like... the polycom phone? |
| 17:29.30 | file | yes. |
| 17:29.33 | lmadsen | hrmmmmmmmmmmmmmmm |
| 17:29.47 | lmadsen | wonder what option I have different on my phones here than what I have on the phones at the clients location |
| 17:30.53 | lmadsen | any suggestions of what I might be looking for? |
| 17:31.12 | file | I'm pretty sure your first post is an attended transfer btw, and the second is blind |
| 17:31.31 | file | unless it's a REFER with Replaces... |
| 17:31.40 | lmadsen | I use the conference button on the polycom's in both cases |
| 17:31.57 | file | O.o |
| 17:32.02 | lmadsen | incoming call, answer with polycom, hit conference button, call end point, answer, conference button, 3 way call.... then hangup |
| 17:32.18 | lmadsen | in the working scenario, the polycom hangs up, and the two other callers remain connected |
| 17:33.04 | file | I can't answer you without full SIP traces of working/non-working |
| 17:33.05 | lmadsen | so I'm kinda confused why the SIP options: replace replaces shows up on the asterisk box when I do 'sip show settings' |
| 17:33.18 | lmadsen | I have full sip traces of non-working |
| 17:33.23 | lmadsen | I could get you working traces too... |
| 17:35.05 | lmadsen | I sent the non-working traces to your digium account.. I'll get some working traces in case you actually decide to look at them ;) |
| 17:35.09 | file | and like, sip show settings doesn't show those options... |
| 17:35.15 | file | sip show channel and sip show peer does |
| 17:35.44 | lmadsen | I'm sorry, you're right |
| 17:36.22 | lmadsen | but the connection between the two asterisk boxes is what I'm looking at, so I'm not sure why it is different on one set of boxes from the other when I'm using the same config (pulled from svn, and verified with diff -u) |
| 17:37.43 | file | does it send back a 481 after the Xfer Refer failed? |
| 17:37.59 | lmadsen | I don't actually see the 481 packet -- it just shows up in the Asterisk sip debug |
| 17:38.05 | lmadsen | let me send you another file |
| 17:38.17 | file | so is that a yes? |
| 17:40.21 | lmadsen | it's kinda a yes.... 'sip debug' shows it, but not the ethereal trace |
| 17:40.46 | file | is pedantic turned on? |
| 17:41.09 | lmadsen | yes it is |
| 17:41.28 | lmadsen | on both sets of boxes (verified) |
| 17:42.05 | file | soooooooo that probably means the tags aren't matching |
| 17:42.08 | file | so it isn't finding the dialog |
| 17:42.09 | file | and freaks out |
| 17:42.13 | lmadsen | makes sense |
| 17:42.31 | lmadsen | I see REFER's on both sets of boxes, but they are slightly different from each other |
| 17:43.17 | lmadsen | on the working set.... I get this: |
| 17:43.17 | lmadsen | <PROTECTED> |
| 17:43.18 | lmadsen | REFER sip:amsvoip03@10.19.18.235 SIP/2.0 |
| 17:43.18 | lmadsen | Via: SIP/2.0/UDP 192.168.128.112:5501;branch=z9hG4bK8ef15e72D9DB8685 |
| 17:43.19 | lmadsen | From: <sip:0004F213EC3F@192.168.128.112:5501>;tag=986429CD-74AB26DA |
| 17:43.19 | lmadsen | To: "Maria Lang" <sip:amsvoip03@10.19.18.235>;tag=as1b856074 |
| 17:43.20 | lmadsen | CSeq: 3 REFER |
| 17:43.24 | lmadsen | Call-ID: 026f19216d838c8f4a807ded6ac12edd@10.19.18.235 |
| 17:43.28 | lmadsen | Contact: <sip:0004F213EC3F@192.168.128.112:5501> |
| 17:43.30 | lmadsen | User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.0.0047 |
| 17:43.32 | lmadsen | Refer-To: <sip:4004@amsvoip4.digium.internal;user=phone?Replaces=53ddf862-17e7202c-35313bb5%40192.168.128.112%3Bto-tag%3Das3e638d43%3Bfrom-tag%3D82EE4B86-11231277> |
| 17:43.35 | lmadsen | Referred-By: <sip:0004F213EC3F@amsvoip4.digium.internal> |
| 17:43.39 | lmadsen | Max-Forwards: 70 |
| 17:43.41 | lmadsen | Content-Length: 0 |
| 17:43.43 | lmadsen | <PROTECTED> |
| 17:43.45 | lmadsen | and you can see the non-working one in the traces I sent you |
| 17:43.49 | lmadsen | notice the REFER line is different between them |
| 17:44.01 | lmadsen | one has the username of the other asterisk box, but the non-working one has the extension number |
| 17:44.07 | lmadsen | or phone number, depending on how it gets setup |
| 17:45.14 | lmadsen | Refer-To: <sip:4044@192.168.1.4;user=phone?Replaces=6fe6358e-9a0661-9eeb7b94%40192.168.1.93%3Bto-tag%3Das6cca869a%3Bfrom-tag%3DE48C6127-3A90031A> |
| 17:45.33 | lmadsen | that's on the non-working refer packet... it tries to refer to the phone directly on the other box |
| 17:45.44 | file | that's not the reason. |
| 17:45.45 | lmadsen | err... nevermind, that is the same in both |
| 17:45.59 | lmadsen | Request-Line: REFER sip:8646169308@192.168.1.17 SIP/2.0 |
| 17:46.01 | lmadsen | that's what I meant |
| 17:46.10 | lmadsen | but probably still not the reason eh |
| 17:46.19 | file | the ONLY reason that a 481 would creep up is if it couldn't find the dialog because of the given callid, to-tag, and from-tag |
| 17:46.29 | file | which is happening because you have pedantic turned on... |
| 17:46.42 | file | so something is amuck there. |
| 17:46.56 | lmadsen | aye.... which is so strange because both sets of boxes have the same configuration |
| 17:47.09 | lmadsen | one is the test environment (working), and the other is production (non-working) |
| 17:47.47 | lmadsen | but the configuration of the polycom phones is probably slightly different because they are not in a repo (I'm using my home set of configs, and Matt setup the other configs for the production phones) |
| 17:48.18 | lmadsen | both running ABE C.1.4 |
| 17:51.39 | file | that is the extent of what you get from me at this period of time, bbl! |
| 17:51.44 | lmadsen | :) |