[cisco-voip] HC and 7945 with 9-3-1SR1-1
Good afternoon, I have a 7945 I've recovered back from a repair vendor that came to be pre-bricked (I guess they wanted to erase the firmware they used or something). I was not able to get it to join my cluster, running 9-3-1SR1-1 . It meets the minimum requirement, but it basically just did nothing, spinning around on term45.default.loads forever. I grabbed the new hotness from cisco.com and let it sit with the phone on an isolated network, and it loaded that right up. However, now it says that my load file is *term45.default*, and it keeps trying to downgrade back to 9-3-1SR1-1 and failing with HC as the reason. I don't know why this phone would fail on that load, unless it's some sort of glitch, or the requirements changed somewhere along the way? Thoughts? Regards Adam Pawlowski University at Buffalo ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] SIP call issue, call get connected after 10 sec.
Hello Raj, What dtmf preference is set on sip trunk, as oob rfc 2833 method needs most mtp. On 01-Apr-2014 11:43 am, Rajkumar Yadav rajkumarya...@y7mail.com wrote: Hi Brian, I got the traces for the call without MTP checked. The call went good but the DTMF issue came up. Please find the attched logs. 20:20:04.259 |//SIP/SIPCdpc(2,68,209650)/ci=48708179/ccbId=943721/scbId=0/StartTransition: requireInactiveSDPForMidcallMediaChange=0, isTrunkEnabledForVoiceEO=1|2,100,68,209650.1^*^* Here the EO is working due to the SIP profile for that SIP trunk having EO provisioned and MTP unchecked. 20:20:04.259 |MediaUtility::isMTPNeededForDTMFBeforeCutThru, there is DTMF MISMATCH, party1SuppDTMFMethod=1 party2DtmfConfig=3|*^*^* In the traces the MTP is allocated too. 20:20:04.261 |MRM::updateMtpCounter devName=MTP_3, countChange=1|2,100,153,1.1418009^10.128.0.2^* 20:20:04.261 |MRM::updateXcodeCounter devName=MTP_3, countChange=1|2,100,153,1.1418009^10.128.0.2^* However as per the person having issue that only Digit 1 is passed on to the IVR system (SIP phone) and rest 8 digits are not. DTMFMTPSide(1), mtpNeededForDtmf(1) firstconnDTMF(2) secondconnDTMF(1)|2,100,57,1.1030654^10.14.0.46^* 20:20:05.285 |MediaManager(710186)::needToInjectDigitsToMTP, isMTPNeededDueToDTMFCapMismatch dtmfMTPSide(1)index(0) injecttoMTP(1)|2,100,57,1.1030654^10.14.0.46^* 20:20:05.285 |MediaManager(710186)::can2833beNegotiatedForCall, --2833 can nto be negotiated for the call|2,100,57,1.1030654^10.14.0.46^* 20:20:05.285 |MediaManager(710186)::setSubscriptionInfo, subscribe(1), passthru(0), inject(1) index(0)|2,100,57,1.1030654^10.14.0.46^* 20:20:05.285 |MediaManager(710186)::allocatedMtpSupportsAnyDtmfCapability|2,100,57,1.1030654^10.14.0.46^* 20:20:05.285 |MediaManager(710186)::DTMFMTPSide(1), mtpNeededForDtmf(1) firstconnDTMF(2) secondconnDTMF(1)|2,100,57,1.1030654^10.14.0.46^* 20:20:05.285 |MediaManager(710186)::needToInjectDigitsToMTP, isMTPNeededDueToDTMFCapMismatch dtmfMTPSide(1)index(1) injecttoMTP(0)|2,100,57,1.1030654^10.14.0.46^* 20:20:05.285 |MediaManager(710186)::can2833beNegotiatedForCall, --2833 can nto be negotiated for the call|2,100,57,1.1030654^10.14.0.46^* Please suggest and have a look into the traces for more detail. Kind Regards, Raaj. -- *From:* Brian Meade bmead...@vt.edu *To:* Rajkumar Yadav rajkumarya...@y7mail.com *Cc:* Amit Kumar amit3@gmail.com; Heim, Dennis dennis.h...@wwt.com; cisco-voip@puck.nether.net cisco-voip@puck.nether.net *Sent:* Monday, 31 March 2014 7:07 PM *Subject:* Re: [cisco-voip] SIP call issue, call get connected after 10 sec. Raaj, CUCM should dynamically insert an MTP when needed for a DTMF mismatch. Would probably need to investigate what is happening when you leave MTP Required unchecked. MTP Required overrides the normal Early Offer process when turned on via the SIP Profile on the trunk. It still results in an Early Offer invite though but through a different process. Brian On Mon, Mar 31, 2014 at 12:07 AM, Rajkumar Yadav rajkumarya...@y7mail.com wrote: we do had unchecked the MTP and tested the call, all calls were properly treated. However the DTMF issue came up, since the Genesys side SIP softphone is supporting only inband (RFC 2833) and MGCP gateway would be configured with OOB DTMF method. (since SCCP phone support both DTMF method can we change the dtmf method in MGCP gateway itself) Also from the traces it fails to do EO due to configuration issue. 16:15:17.424 |//SIP/SIPCdpc(2,68,205069)/ci=48665604/ccbId=918923/scbId=0/isTrunkConfiguredforVoiceEO: Trunk configured for EO but EO will not be effective due to other configuration - MTPRequired=1 IPAddrMode=0 MediaPreference=0|2,100,68,205069.1^*^* Kind Regards, Raaj -- *From:* Brian Meade bmead...@vt.edu *To:* Amit Kumar amit3@gmail.com *Cc:* Heim, Dennis dennis.h...@wwt.com; cisco-voip@puck.nether.net cisco-voip@puck.nether.net; Rajkumar Yadav rajkumarya...@y7mail.com *Sent:* Monday, 31 March 2014 9:12 AM *Subject:* Re: [cisco-voip] SIP call issue, call get connected after 10 sec. The only difference I was able to find between the working and nonworking is that the Create Connection Response is received before the outgoing invite in the working scenario: Create Connection: 16:15:17.423 |MGCPHandler send msg SUCCESSFULLY to: 10.128.0.2 CRCX 1822574 S0/SU0/DS1-0/11@10.128.0.2 MGCP 0.1 C: D2e6940300F5851d X: b L: p:20, a:PCMU, s:off, t:b8 M: recvonly R: D/[0-9ABCD*#] Q: process,loop |2,100,153,1.1390558^10.128.0.2^* Create Connection Response: 16:15:17.433 |MGCPHandler received msg from: 10.128.0.2 200 1822574 OK I: D2D7 v=0 c=IN IP4 10.128.0.2 m=audio 19388 RTP/AVP 0 100 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194,200-202 a=X-sqn:0 a=X-cap: 1 audio RTP/AVP 100 a=X-cpar: a=rtpmap:100 X-NSE/8000
Re: [cisco-voip] HC and 7945 with 9-3-1SR1-1
Adam, The HC load rejected means it either cant downgrade or upgrade. 7945 phones with hardware revision 13 and higher must run 9.3.1SR3 or later firmware. So even though you have SR1S it will not downgrade. Check the hardware version on the phone. Get at least SR3 and upload to to CUCM. Restart TFTP. You can change the defaults and hardcode only this phone if you want to test. On the flip side if you have any thing older then 8.5.2 it wont upgrade either. You have to first upgrade to 8.5.2 and then to 9.3.1SR1s. Regards Mehtab Shinwari | CCNP RS/V Senior Support Engineer Original message From: Pawlowski, Adam aj...@buffalo.edu Date: To: cisco-voip@puck.nether.net Subject: [cisco-voip] HC and 7945 with 9-3-1SR1-1 Good afternoon, I have a 7945 I've recovered back from a repair vendor that came to be pre-bricked (I guess they wanted to erase the firmware they used or something). I was not able to get it to join my cluster, running 9-3-1SR1-1 . It meets the minimum requirement, but it basically just did nothing, spinning around on term45.default.loads forever. I grabbed the new hotness from cisco.com and let it sit with the phone on an isolated network, and it loaded that right up. However, now it says that my load file is *term45.default*, and it keeps trying to downgrade back to 9-3-1SR1-1 and failing with HC as the reason. I don't know why this phone would fail on that load, unless it's some sort of glitch, or the requirements changed somewhere along the way? Thoughts? Regards Adam Pawlowski University at Buffalo ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] SIP call issue, call get connected after 10 sec.
Raaj, I looked at these traces and see the MTP is being pulled in due to the early offer configuration but would also be needed for the DTMF mismatch: 20:20:05.285 |MediaManager(710186)::wait_AuConnectRequest, CI(48708178,48708179), capCount(11,1), mcNodeId(0,0), xferMode(12,16), reConnectType(0), mrid (0, 0) IFCreated(0 0) proIns(0 0), AC(0,0) party1DTMF(1 1 0 1 0) party2DTMF(3 2 101 1 0),reConnFlag=0, connType(3,3), IFHand(0,0),MTP(0,0),MRGL(aaf6462d-d96d-8d78-e98b-f96a899cd470,aaf6462d-d96d-8d78-e98b-f96a899cd470) videoCap(0 0), mmCallType(0),FS(0,0) IpAddrMode(0 0) aPid(2, 135, 1366), bPid(2, 68, 209650) EOType(0 2)|2,100,57,1.1030654^10.14.0.46^* 20:20:05.285 |MediaManager(710186)::isMTPNeededForDTMF, isMTPNeeded(1)|2,100,57,1.1030654^10.14.0.46^* party1 only supports out of band and party 2 only supports in-band. It looks like the SIP Trunk has a DTMF Preference hard set as well. As far as only getting 1 digit, I only see one digit actually being sent by the MGCP gateway: 20:20:11.021 |MGCPHandler received msg from: 10.128.0.2 NTFY 200530244 S0/SU0/DS1-0/11@10.128.0.2 MGCP 0.1 N: ca@10.128.4.10:2427 X: b O: D/1 |2,100,152,1.5777096^10.128.0.2^* CUCM sends a request notification to continue receiving any future digits: 20:20:11.022 |MGCPHandler send msg SUCCESSFULLY to: 10.128.0.2 RQNT 1863107 S0/SU0/DS1-0/11@10.128.0.2 MGCP 0.1 X: b R: D/[0-9ABCD*#] Q: process,loop But the gateway never seems to send any further digits. Brian On Tue, Apr 1, 2014 at 2:13 AM, Rajkumar Yadav rajkumarya...@y7mail.comwrote: Hi Brian, I got the traces for the call without MTP checked. The call went good but the DTMF issue came up. Please find the attched logs. 20:20:04.259 |//SIP/SIPCdpc(2,68,209650)/ci=48708179/ccbId=943721/scbId=0/StartTransition: requireInactiveSDPForMidcallMediaChange=0, isTrunkEnabledForVoiceEO=1|2,100,68,209650.1^*^* Here the EO is working due to the SIP profile for that SIP trunk having EO provisioned and MTP unchecked. 20:20:04.259 |MediaUtility::isMTPNeededForDTMFBeforeCutThru, there is DTMF MISMATCH, party1SuppDTMFMethod=1 party2DtmfConfig=3|*^*^* In the traces the MTP is allocated too. 20:20:04.261 |MRM::updateMtpCounter devName=MTP_3, countChange=1|2,100,153,1.1418009^10.128.0.2^* 20:20:04.261 |MRM::updateXcodeCounter devName=MTP_3, countChange=1|2,100,153,1.1418009^10.128.0.2^* However as per the person having issue that only Digit 1 is passed on to the IVR system (SIP phone) and rest 8 digits are not. DTMFMTPSide(1), mtpNeededForDtmf(1) firstconnDTMF(2) secondconnDTMF(1)|2,100,57,1.1030654^10.14.0.46^* 20:20:05.285 |MediaManager(710186)::needToInjectDigitsToMTP, isMTPNeededDueToDTMFCapMismatch dtmfMTPSide(1)index(0) injecttoMTP(1)|2,100,57,1.1030654^10.14.0.46^* 20:20:05.285 |MediaManager(710186)::can2833beNegotiatedForCall, --2833 can nto be negotiated for the call|2,100,57,1.1030654^10.14.0.46^* 20:20:05.285 |MediaManager(710186)::setSubscriptionInfo, subscribe(1), passthru(0), inject(1) index(0)|2,100,57,1.1030654^10.14.0.46^* 20:20:05.285 |MediaManager(710186)::allocatedMtpSupportsAnyDtmfCapability|2,100,57,1.1030654^10.14.0.46^* 20:20:05.285 |MediaManager(710186)::DTMFMTPSide(1), mtpNeededForDtmf(1) firstconnDTMF(2) secondconnDTMF(1)|2,100,57,1.1030654^10.14.0.46^* 20:20:05.285 |MediaManager(710186)::needToInjectDigitsToMTP, isMTPNeededDueToDTMFCapMismatch dtmfMTPSide(1)index(1) injecttoMTP(0)|2,100,57,1.1030654^10.14.0.46^* 20:20:05.285 |MediaManager(710186)::can2833beNegotiatedForCall, --2833 can nto be negotiated for the call|2,100,57,1.1030654^10.14.0.46^* Please suggest and have a look into the traces for more detail. Kind Regards, Raaj. -- *From:* Brian Meade bmead...@vt.edu *To:* Rajkumar Yadav rajkumarya...@y7mail.com *Cc:* Amit Kumar amit3@gmail.com; Heim, Dennis dennis.h...@wwt.com; cisco-voip@puck.nether.net cisco-voip@puck.nether.net *Sent:* Monday, 31 March 2014 7:07 PM *Subject:* Re: [cisco-voip] SIP call issue, call get connected after 10 sec. Raaj, CUCM should dynamically insert an MTP when needed for a DTMF mismatch. Would probably need to investigate what is happening when you leave MTP Required unchecked. MTP Required overrides the normal Early Offer process when turned on via the SIP Profile on the trunk. It still results in an Early Offer invite though but through a different process. Brian On Mon, Mar 31, 2014 at 12:07 AM, Rajkumar Yadav rajkumarya...@y7mail.com wrote: we do had unchecked the MTP and tested the call, all calls were properly treated. However the DTMF issue came up, since the Genesys side SIP softphone is supporting only inband (RFC 2833) and MGCP gateway would be configured with OOB DTMF method. (since SCCP phone support both DTMF method can we change the dtmf method in MGCP gateway itself) Also from the traces it fails to do EO due to configuration issue. 16:15:17.424
Re: [cisco-voip] HC and 7945 with 9-3-1SR1-1
Mehtab, I must have missed that in a release note somewhere. On the back it looks like it says V14. In the log I have: 3898: DBG 00:39:05.810548 image: phone hw-compat value 12 3899: NOT 00:39:05.811249 image: loadhwc=7 myhwc=12 3900: WRN 00:39:05.811835 image: Load rejected - hardware compatibility mismatch 3901: WRN 00:39:05.812395 image: Must use 9.3(1) SR or later release on this phone Which doesn't really correspond to anything helpful, and it's stickered to 9.3(1)SR1 on the back. It was on our system and working previously, but I wonder if perhaps it had been upgraded in the repair lab or such, and now cannot downgrade further? I can probably get that firmware onto the cluster for that phone, but without having tested it to see if there are any new bugs or issues I wouldn't want to just toss the set back out. Oh well. Thank you for your reply, I do appreciate it. Adam Pawlowski University at Buffalo ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] CIACO VG224 issue
Hi Ryan, Thank you for the suggestion Will try with a different image file. Also, my TFTP server is remote location to the VG224 Regards Siva -Original Message- From: Ryan Huff [mailto:rthconsulta...@gmail.com] Sent: Monday, March 31, 2014 3:59 PM To: s...@dateksystems.com Cc: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] CIACO VG224 issue Have you tried downloading a new image from Cisco.com? Checksum error may indicate a corrupted image (among other things). What does the topology between your tftp server and gateway look like? layer 2? Layer 3? Try and get your tftp server on the the same segment as the gateway if it isn't already. Sent from my iPhone On Mar 31, 2014, at 3:04 PM, Sivakumar Donthamchetty s...@dateksystems.com wrote: Hi, I have a VG224 and not able to boot it. When I boot it comes to ROMMON prompt. I tried to install new image file vi TFTP, it downloads the file and at the end it says checksum error Please need help how I can rectify this unit? Regards Siva ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] CURRI
I'm not 100% clear on th eapplication server for CURRI. This is a generic web server, correct? ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] CIACO VG224 issue
Hi Amit, Thank you I did it on LAN and it worked. Not sure if it was NETWORK issue or the FLASH as I changed the Flash card as well Regards Siva From: Amit Kumar [mailto:amit3@gmail.com] Sent: Tuesday, April 01, 2014 12:26 PM To: s...@dateksystems.com Subject: Re: [cisco-voip] CIACO VG224 issue It would be worth to have a local tftp, instead of WAN. On Tue, Apr 1, 2014 at 11:24 PM, Sivakumar Donthamchetty s...@dateksystems.com wrote: Hi Ryan, Thank you for the suggestion Will try with a different image file. Also, my TFTP server is remote location to the VG224 Regards Siva -Original Message- From: Ryan Huff [mailto:rthconsulta...@gmail.com] Sent: Monday, March 31, 2014 3:59 PM To: s...@dateksystems.com Cc: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] CIACO VG224 issue Have you tried downloading a new image from Cisco.com? Checksum error may indicate a corrupted image (among other things). What does the topology between your tftp server and gateway look like? layer 2? Layer 3? Try and get your tftp server on the the same segment as the gateway if it isn't already. Sent from my iPhone On Mar 31, 2014, at 3:04 PM, Sivakumar Donthamchetty s...@dateksystems.com wrote: Hi, I have a VG224 and not able to boot it. When I boot it comes to ROMMON prompt. I tried to install new image file vi TFTP, it downloads the file and at the end it says checksum error Please need help how I can rectify this unit? Regards Siva ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Requested circuit/channel not available
Hi, Started to get this issue this morning with one of our MGCP gateways. Incoming calls are working correctly on an ISDN30, but outgoing calls are being denied, and re-routed via a backup route. Outgoing calls are hitting the gateway, but is getting a 'Requested circuit/channel not available' See trace below. I've tried to change the channel selection order, but still the same. I've checked everything that I can think of, and I'm beginning to think that this is a Telco issue, but thought that I'd ask the group to see if there is anything else to check before I take it up with BT. Thanks, Hefin 2014-04-01 22:07:55 local3/7 Apr 1 21:07:54.602: ISDN Se0/0/0:15 Q931: TX - SETUP pd = 8 callref = 0x0004 2014-04-01 22:07:55 local3/7Sending Complete 2014-04-01 22:07:55 local3/7Bearer Capability i = 0x8090A3 2014-04-01 22:07:55 local3/7Standard = CCITT 2014-04-01 22:07:55 local3/7Transfer Capability = Speech 2014-04-01 22:07:55 local3/7Transfer Mode = Circuit 2014-04-01 22:07:55 local3/7Transfer Rate = 64 kbit/s 2014-04-01 22:07:55 local3/7Channel ID i = 0xA9839F 2014-04-01 22:07:55 local3/7Exclusive, Channel 31 2014-04-01 22:07:55 local3/7Calling Party Number i = 0x0081, '2456' 2014-04-01 22:07:55 local3/7Plan:Unknown, Type:Unknown 2014-04-01 22:07:55 local3/7Called Party Number i = 0x80, '622456' 2014-04-01 22:07:55 local3/7Plan:Unknown, Type:Unknown 2014-04-01 22:07:55 local3/7 Apr 1 21:07:54.682: ISDN Se0/0/0:15 Q931: RX - RELEASE_COMP pd = 8 callref = 0x8004 2014-04-01 22:07:55 local3/7Cause i = 0x82AC - Requested circuit/channel not available ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
