Re: [OSL | CCIE_Voice] CUE integration issue
looks like you are running CUCM Lic, change that to CME and then try again. On Mon, Feb 28, 2011 at 2:08 AM, Ravindra Lakpriya lakpr...@gmail.comwrote: Hi Guys I'm trying to integrate CUE with the CUCME. in CUE gue its asking for the primary call manager details. there when im submitting the web user name it give an error about incorrect user name. Exact error massage is Web Login failed I checked that username by directly logging to CUCME gui. it works fine. I check the license as well. You guys have any idea about the debugging command for this specific scenario ? Many Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUC - CUCME - SCCP Integration - MWI issue on SIP Phones !!!!
My fren, Friderich is doing CUC integration with UCME, not CUE. Is there any reason that you mwi server to be 202.1 instead of CUC address? Shingei On Mon, Feb 28, 2011 at 11:25 AM, Romain romain.mull...@gmail.com wrote: Isn't your CUE at 10.10.202.2 ? That is the IP you need for your mwi server instead of .1 Sent from my iPhone On Feb 27, 2011, at 8:12 PM, Friderich Claude cfrider...@netcore.lu wrote: Hello, After a couple of tests and after trying to configure this feature, impossible to make it work L MWI on sccp phone is working fine MWI on SIP phone doesn’t work. Normaly in sip-ua *mwi-server ipv4:10.10.202.1 unsolicited* (CUCME IP Adress in telephony-service) In telephony-service *mwi relay*** in the voice register dn 1 *mwi*** *sip-ua * retry invite 3 timers trying 300 * mwi-server ipv4:10.10.202.1 expires 3600 port 5060 transport udp unsolicited* debug ccsip gives me the results below: Sent: NOTIFY sip:3005@10.10.202.50:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bK8F14A9 From: sip:3005@10.10.202.1;tag=137E2EC-1A2C To: sip:3005@10.10.202.50 Call-ID: 942D5193-420111E0-8216F0AE-E383DE3B@10.10.202.1 942D5193-420111E0-8216F0AE-E383DE3B@10.10.202.1 *CSeq: 101 NOTIFY*** Max-Forwards: 70 Date: Sun, 27 Feb 2011 23:38:56 GMT User-Agent: Cisco-SIPGateway/IOS-12.x Event: message-summary Subscription-State: active Contact: sip:3005@10.10.202.1:5060 Content-Type: application/message-summary Content-Length: 23 *Messages-Waiting: yes*** Feb 27 23:38:56.959: //-1//SIP/Msg/ccsipDisplayMsg: Received: *SIP/2.0 400 Bad Request* --- Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bK8F14A9 From: sip:3005@10.10.202.1;tag=137E2EC-1A2C To: sip:3005@10.10.202.50 Call-ID: 942D5193-420111E0-8216F0AE-E383DE3B@10.10.202.1 942D5193-420111E0-8216F0AE-E383DE3B@10.10.202.1 Date: Sun, 27 Feb 2011 23:38:55 GMT *Warning: 399 Bad MWI NOTIFY* - ? CSeq: 101 NOTIFY Content-Length: 0 Does anybody encountered this problem concerning MWI on sip phones only ?? Does it really work ?? Any suggestions are appreciated. Regards Claude. *Claude Friderich* *PreSales Support* *image001.gif*** *NETCORE PSF S.A.*** 49 rue du Baerendall B.P.65 L-8201 Mamer Téléphone: 31 33 80-407 Fax: 31 33 80 8-407 GSM: 621 303 616 E-mail: cfrider...@netcore.lu -- This email was Anti Virus checked. Disclaimer The information in this Internet e-mail is confidential and may be legally privileged. It is intended solely for the addressee. Access to this Internet e-mail by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. When addressed to our clients any opinions or advice contained in this e-mail are subject to the terms and conditions expressed in our governing terms of business. ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.comwww.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CallManager Trace
What is the behavior that you experience when placing the call? - Pablo Meneses. On Sat, Feb 26, 2011 at 1:18 PM, Ki Wi kiwi.vo...@gmail.com wrote: in serviceablity page, under trace select server CM services Cisco Callmanager turn on cm trace on h225 gatekeeper... For me usually if such scenario, i will go CME to turn on debug instead... debug voice dialpeer debug h225 asn1 is useful. Most of the time, it's just some digits manipulation needed. Anyway, for GK. I think ACF is more important than LCF. Because LCF simply means it knows where to route the calls? On Sun, Feb 27, 2011 at 1:21 AM, Angila Smith asangiesmit...@gmail.comwrote: Call Setup PhoneA --- UCM --- Gatekeeper A -- Remote GK --- CME -- PhoneB I am trying to make a call from phoneA to PhoneB call fails. When I run deb h225 asn1 and deb ras on GK A, I can see LRQ and LCF that mean remote GK is able route the call. UCM and CME is failing to negotiate the parameters. What trace can I run on UCM to find out why this call failed? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Calling to PSTN-WAN remote Zone
Hi Jay,
Looks like your call is going from HQ --- GK --- PSTN-WAN and it is
getting to the PSTN-WAN router.
Can you please do a debug voip dialpeer on the PSTN-WAN router?
Thanks
Kalyan
On Tue, Feb 22, 2011 at 6:57 PM, Jay Woods cisco...@live.com wrote:
Hello Everyone,
I'm having trouble calling the India number through the PSTN-WAN zone of
the Gatekeeper. I have posted the relevant debugs below.
HQ#debug gatek main 10
HQ#
HQ#
HQ#
HQ#
*Feb 22 22:26:57.538: ////GK/gk_process:
QUEUE_EVENT (minor 0) wakeup
*Feb 22 22:26:57.678: ////GK/gk_process:
QUEUE_EVENT (minor 0) wakeup
*Feb 22 22:27:01.758: ////GK/gk_process:
QUEUE_EVENT (minor 0) wakeup
*Feb 22 22:27:02.270: ////GK/gk_process:
QUEUE_EVENT (minor 0) wakeup
*Feb 22 22:27:02.270: ////GK/gk_rassrv_arq:
arqp=0x4837F484,crv=0x3, answerCall=0
*Feb 22 22:27:02.270: ////GK/gk_rassrv_sep_arq: ARQ
Didn't use GK_AAA_PROC
*Feb 22 22:27:02.270: //802EE55D0300/802EE55D0300/GK/gk_dns_query: No Name
servers
*Feb 22 22:27:02.270: //802EE55D0300/802EE55D0300/GK/rassrv_get_addrinfo:
(916745738932) Tech-prefix match failed.
*Feb 22 22:27:02.270: //802EE55D0300/802EE55D0300/GK/rassrv_get_addrinfo:
(916745738932) Matched zone prefix 91 and remainder 6745738932
*Feb 22 22:27:02.270:
////GK/gk_rassrv_get_ingress_network: returning
default ingress network = 1
*Feb 22 22:27:02.270:
//802EE55D0300/802EE55D0300/GK/rassrv_arq_select_viazone: about to check the
source side, src_zonep=0x496E9808
*Feb 22 22:27:02.270:
//802EE55D0300/802EE55D0300/GK/rassrv_arq_select_viazone: matched zone is
US, and z_invianamelen=0
*Feb 22 22:27:02.270:
//802EE55D0300/802EE55D0300/GK/rassrv_arq_select_viazone: about to check the
destination side, dst_zonep=0x4A93A26C
*Feb 22 22:27:02.270:
//802EE55D0300/802EE55D0300/GK/rassrv_arq_select_viazone: matched zone is
PSTN-WAN, and z_outvianamelen=0
*Feb 22 22:27:02.270: //802EE55D0300/802EE55D0300/GK/rassrv_get_addrinfo:
No tech prefix
*Feb 22 22:27:02.270: //802EE55D0300/802EE55D0300/GK/rassrv_get_addrinfo:
Alias not found
*Feb 22 22:27:02.270:
//802EE55D0300/802EE55D0300/GK/rassrv_put_remote_zones_from_zone_list: zone
PSTN-WAN
*Feb 22 22:27:02.270: //802EE55D0300/802EE55D0300/GK/send_lrq: seq_lrq 1,
use_be 0, rzone_cnt 1
*Feb 22 22:27:02.270: //802EE55D0300/802EE55D0300/GK/send_lrq: lrq array
index 5, lap 4ABAE820
*Feb 22 22:27:02.270: //802EE55D0300/802EE55D0300/GK/send_lrq: sent lrq -
zonecount 1
*Feb 22 22:27:02.542: ////GK/gk_process: got a
TIMER event
*Feb 22 22:27:02.542: ////GK/gk_handle_timers
*Feb 22 22:27:02.542: ////GK/gk_handle_timers:
managed timer expired 0x4762D750
*Feb 22 22:27:04.878: ////GK/gk_process: got a
TIMER event
*Feb 22 22:27:04.878: ////GK/gk_handle_timers
*Feb 22 22:27:04.878: ////GK/gk_handle_timers:
managed timer expired 0x4762D8F0
*Feb 22 22:27:05.270: ////GK/gk_process: got a
TIMER event
*Feb 22 22:27:05.270: ////GK/gk_handle_timers
*Feb 22 22:27:05.270: ////GK/gk_handle_timers:
managed timer expired 0x4ABAE830
*Feb 22 22:27:08.270: ////GK/gk_process: got a
TIMER event
*Feb 22 22:27:08.270: ////GK/gk_handle_timers
*Feb 22 22:27:08.270: ////GK/gk_handle_timers:
managed timer expired 0x4ABAE830
*Feb 22 22:27:08.270: //802EE55D0300/802EE55D0300/GK/gk_rassrv_sep_arq: LRQ
suspension point failed (return code = 0x4009)
HQ#
HQ#
HQ#
HQ# debug h225 asn1
prefix dialedDigits : 1#
}
}
}
}
}
mc FALSE
undefinedNode FALSE
}
gatekeeperIdentifier {US}
endpointVendor
{
vendor
{
t35CountryCode 181
t35Extension 0
manufacturerCode 18
}
}
timeToLive 60
keepAlive TRUE
endpointIdentifier {4A5677180004}
willSupplyUUIEs FALSE
}
*Feb 22 22:27:48.530: RAS OUTGOING PDU ::=
value RasMessage ::= registrationConfirm :
{
requestSeqNum 3502
protocolIdentifier { 0 0 8 2250 0 4 }
callSignalAddress
{
}
gatekeeperIdentifier {US}
endpointIdentifier {4A5677180004}
timeToLive 60
willRespondToIRR FALSE
maintainConnection FALSE
}
*Feb 22 22:27:48.530: RAS OUTGOING ENCODE BUFFER::=
12400DAD060008914A00040002005500531E0034004100350036003700370031003800300030003000300030003000300034208A0002003B01000100
*Feb 22 22:27:48.530:
*Feb 22 22:27:48.638: RAS INCOMING ENCODE BUFFER::=
[OSL | CCIE_Voice] MVA Hairpinin Remote destionation Match Problem
Hi all, Working with MVA with hairpinin I have found and issue that I don't know how to workaround it Hqph2 2002 Remotedestination, 6178632683 When I call to hqph1 from remote the identification is OK, 2002 The problem appears with DISA service. I does not recognize the remote destination number and it asks for entering it DISA Number in HQ 2123942010 application service mva http://10.10.210.10:8080/ccmivr/pages/IVRMainpage.vxml dial-peer voice 2010 voip service mva session target ipv4:10.10.210.10 incoming called-number 2010 codec g711ulaw ! dial-peer voice 20101 voip preference 1 destination-pattern 2010 voice-class h323 1 session target ipv4:10.10.210.11 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 20102 voip preference 2 destination-pattern 2010 voice-class h323 1 session target ipv4:10.10.210.10 dtmf-relay h245-alphanumeric codec g711ulaw no vad In the 1ºstage (incoming dialpeer 2010, mva application is triggered). Here is the problem when I call from the remote destination number to DISA in the HQMGCP gateway, CCM recognizes it an sends it as 2002 to h323 incoming leg so now mva ivr fails. Is anything I am missing? Does it work this way to you? Thanks in advance! Francesc Debug voice application vxml Feb 28 17:52:23.469: //300/80C50F960300/VXML:/vxml_submit_proc: submit: caching=fast fetchhint=invalid fetchtimeout=0 maxage=-1 maxstale=-1 URI(abs):http://10.10.210.10:8080/ccmivr/IVRCalleridLookup.do scheme=http host=10.10.210.10 port=8080 path=/ccmivr/IVRCalleridLookup.do Feb 28 17:52:23.469: //300/80C50F960300/VXML:/vxml_sub_attrs_proc: method=get enctype=application/x-www-form-urlencoded Feb 28 17:52:23.469: //300/80C50F960300/VXML:/vxml_nmtokens_proc: name=remotedest name=srcdir Feb 28 17:52:23.469: //300/80C50F960300/VXML:/vxml_vapp_bgload_from_proc: urlp=http://10.10.210.10:8080/ccmivr/IVRCalleridLookup.do?remotedest=2002srcdir=en_US fetchaudio=NULL delay=0 minimum=0 Feb 28 17:52:23.469: //300/80C50F960300/VXML:/vxml_vapp_bgload: url http://10.10.210.10:8080/ccmivr/IVRCalleridLookup.do?remotedest=2002srcdir=en_US cachable 1 fetchtimeout 0 maxage=-1 maxstale=-1 Feb 28 17:52:23.469: //300//AFW_:/vapp_bgload: url=http://10.10.210.10:8080/ccmivr/IVRCalleridLookup.do?remotedest=2002srcdir=en_US Feb 28 17:52:23.469: //300//AFW_:/vxml_update_cleanup_timer: cleaning timer running 0 fetchtimeout 0 Feb 28 17:52:23.469: //300/80C50F960300/VXML:/vxml_leave_scope: scope=anonymous Feb 28 17:52:23.469: //300/80C50F960300/VXML:/vxml_load_immediate_done: ... ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Pressing Message button during SRST
By default, it will end up on opening greeting when SRST user press the message button. Is there a way to make the users enter their own voicemail account directly(attempt sign in page) ? I'm aware of a way currently which is to set the calling number to in the hunt pilot but the method is not so graceful. There's chances that someone else last 4 digits number is the same or the system will recording the original calling number as 4 digits instead of maybe 10 or 11 digits long. Any interesting workaround for this? [?] 330.gif___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] 5 day exam course
Anyone going this month to this in Columbus and would like to share a cab, or can give me a lift from the airport ? If so pls contact me off list Cheers Leslie ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Pressing Message button during SRST
There's one more way which is to translate the calling number from original number into 4 digits extension. This way works as well. I suppose this is the best solution? Is that vm-integration going to do some magic? I tried it just now but seems like it's not working for the message button. ! voice translation-rule 1 rule 1 /^617863\(\)/ /\1/ ! voice translation-rule 5 rule 1 // // type any national plan any isdn ! voice translation-profile voicemail translate calling 1 translate called 5 ! dial-peer voice 15 pots translation-profile outgoing voicemail destination-pattern 912123945888 port 0/1/0:23 forward-digits 0 prefix 12123945888 On Tue, Mar 1, 2011 at 3:00 AM, Ki Wi kiwi.vo...@gmail.com wrote: By default, it will end up on opening greeting when SRST user press the message button. Is there a way to make the users enter their own voicemail account directly(attempt sign in page) ? I'm aware of a way currently which is to set the calling number to in the hunt pilot but the method is not so graceful. There's chances that someone else last 4 digits number is the same or the system will recording the original calling number as 4 digits instead of maybe 10 or 11 digits long. Any interesting workaround for this? [?] 330.gif___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUC - CUCME - SCCP Integration - MWI issue on SIP Phones !!!!
Hi Friderich, The SIP server has to be Unity Connection. Try it out, it should work. Regards, On Mon, Feb 28, 2011 at 1:08 PM, Friderich Claude cfrider...@netcore.luwrote: Hi all, Thanks for reply and your suggestions Effectively I’m doing a Cisco Unity Connection Integration not CUE. Imagine CUE isn’t present. Good Question Shingei J why CUCME IP address instead of CUC ??? Normally you have to put the CUC IP Address in mwi server when you integrate CUC with CUCME in a SIP configuration. But with SCCP integration and with SIP phones configured on CUCME, I think it’s correct, we have to put the CUCME ip address in mwi-server. Best Regards, Claude. *Claude Friderich* *PreSales Support* *[image: ccvp_voice_sm]*** *NETCORE PSF S.A.*** 49 rue du Baerendall B.P.65 L-8201 Mamer Téléphone: 31 33 80-407 Fax: 31 33 80 8-407 GSM: 621 303 616 E-mail: cfrider...@netcore.lu *From:* ShinGei Yong [mailto:shingei.y...@gmail.com] *Sent:* lundi 28 février 2011 08:42 *To:* Romain; Friderich Claude; OSL Questions *Subject:* Re: [OSL | CCIE_Voice] CUC - CUCME - SCCP Integration - MWI issue on SIP Phones My fren, Friderich is doing CUC integration with UCME, not CUE. Is there any reason that you mwi server to be 202.1 instead of CUC address? Shingei On Mon, Feb 28, 2011 at 11:25 AM, Romain romain.mull...@gmail.com wrote: Isn't your CUE at 10.10.202.2 ? That is the IP you need for your mwi server instead of .1 Sent from my iPhone On Feb 27, 2011, at 8:12 PM, Friderich Claude cfrider...@netcore.lu wrote: Hello, After a couple of tests and after trying to configure this feature, impossible to make it work L MWI on sccp phone is working fine MWI on SIP phone doesn’t work. Normaly in sip-ua *mwi-server ipv4:10.10.202.1 unsolicited* (CUCME IP Adress in telephony-service) In telephony-service *mwi relay* in the voice register dn 1 *mwi* *sip-ua * retry invite 3 timers trying 300 * mwi-server ipv4:10.10.202.1 expires 3600 port 5060 transport udp unsolicited* debug ccsip gives me the results below: Sent: NOTIFY sip:3005@10.10.202.50:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bK8F14A9 From: sip:3005@10.10.202.1;tag=137E2EC-1A2C To: sip:3005@10.10.202.50 Call-ID: 942D5193-420111E0-8216F0AE-E383DE3B@10.10.202.1 *CSeq: 101 NOTIFY* Max-Forwards: 70 Date: Sun, 27 Feb 2011 23:38:56 GMT User-Agent: Cisco-SIPGateway/IOS-12.x Event: message-summary Subscription-State: active Contact: sip:3005@10.10.202.1:5060 Content-Type: application/message-summary Content-Length: 23 *Messages-Waiting: yes* Feb 27 23:38:56.959: //-1//SIP/Msg/ccsipDisplayMsg: Received: *SIP/2.0 400 Bad Request* --- Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bK8F14A9 From: sip:3005@10.10.202.1;tag=137E2EC-1A2C To: sip:3005@10.10.202.50 Call-ID: 942D5193-420111E0-8216F0AE-E383DE3B@10.10.202.1 Date: Sun, 27 Feb 2011 23:38:55 GMT *Warning: 399 Bad MWI NOTIFY* - ? CSeq: 101 NOTIFY Content-Length: 0 Does anybody encountered this problem concerning MWI on sip phones only ?? Does it really work ?? Any suggestions are appreciated. Regards Claude. *Claude Friderich* *PreSales Support* *image001.gif* *NETCORE PSF S.A.* 49 rue du Baerendall B.P.65 L-8201 Mamer Téléphone: 31 33 80-407 Fax: 31 33 80 8-407 GSM: 621 303 616 E-mail: cfrider...@netcore.lu -- This email was Anti Virus checked. Disclaimer The information in this Internet e-mail is confidential and may be legally privileged. It is intended solely for the addressee. Access to this Internet e-mail by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. When addressed to our clients any opinions or advice contained in this e-mail are subject to the terms and conditions expressed in our governing terms of business. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com image001.gif___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUC - CUCME - SCCP Integration - MWI issue on SIP Phones !!!!
you also will need to disable two sip supplementary services: refer and moved temporarily. On Mon, Feb 28, 2011 at 2:31 PM, George Goglidze gogli...@gmail.com wrote: Hi Friderich, The SIP server has to be Unity Connection. Try it out, it should work. Regards, On Mon, Feb 28, 2011 at 1:08 PM, Friderich Claude cfrider...@netcore.luwrote: Hi all, Thanks for reply and your suggestions Effectively I’m doing a Cisco Unity Connection Integration not CUE. Imagine CUE isn’t present. Good Question Shingei J why CUCME IP address instead of CUC ??? Normally you have to put the CUC IP Address in mwi server when you integrate CUC with CUCME in a SIP configuration. But with SCCP integration and with SIP phones configured on CUCME, I think it’s correct, we have to put the CUCME ip address in mwi-server. Best Regards, Claude. *Claude Friderich* *PreSales Support* *[image: ccvp_voice_sm]*** *NETCORE PSF S.A.*** 49 rue du Baerendall B.P.65 L-8201 Mamer Téléphone: 31 33 80-407 Fax: 31 33 80 8-407 GSM: 621 303 616 E-mail: cfrider...@netcore.lu *From:* ShinGei Yong [mailto:shingei.y...@gmail.com] *Sent:* lundi 28 février 2011 08:42 *To:* Romain; Friderich Claude; OSL Questions *Subject:* Re: [OSL | CCIE_Voice] CUC - CUCME - SCCP Integration - MWI issue on SIP Phones My fren, Friderich is doing CUC integration with UCME, not CUE. Is there any reason that you mwi server to be 202.1 instead of CUC address? Shingei On Mon, Feb 28, 2011 at 11:25 AM, Romain romain.mull...@gmail.com wrote: Isn't your CUE at 10.10.202.2 ? That is the IP you need for your mwi server instead of .1 Sent from my iPhone On Feb 27, 2011, at 8:12 PM, Friderich Claude cfrider...@netcore.lu wrote: Hello, After a couple of tests and after trying to configure this feature, impossible to make it work L MWI on sccp phone is working fine MWI on SIP phone doesn’t work. Normaly in sip-ua *mwi-server ipv4:10.10.202.1 unsolicited* (CUCME IP Adress in telephony-service) In telephony-service *mwi relay* in the voice register dn 1 *mwi* *sip-ua * retry invite 3 timers trying 300 * mwi-server ipv4:10.10.202.1 expires 3600 port 5060 transport udp unsolicited* debug ccsip gives me the results below: Sent: NOTIFY sip:3005@10.10.202.50:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bK8F14A9 From: sip:3005@10.10.202.1;tag=137E2EC-1A2C To: sip:3005@10.10.202.50 Call-ID: 942D5193-420111E0-8216F0AE-E383DE3B@10.10.202.1 *CSeq: 101 NOTIFY* Max-Forwards: 70 Date: Sun, 27 Feb 2011 23:38:56 GMT User-Agent: Cisco-SIPGateway/IOS-12.x Event: message-summary Subscription-State: active Contact: sip:3005@10.10.202.1:5060 Content-Type: application/message-summary Content-Length: 23 *Messages-Waiting: yes* Feb 27 23:38:56.959: //-1//SIP/Msg/ccsipDisplayMsg: Received: *SIP/2.0 400 Bad Request* --- Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bK8F14A9 From: sip:3005@10.10.202.1;tag=137E2EC-1A2C To: sip:3005@10.10.202.50 Call-ID: 942D5193-420111E0-8216F0AE-E383DE3B@10.10.202.1 Date: Sun, 27 Feb 2011 23:38:55 GMT *Warning: 399 Bad MWI NOTIFY* - ? CSeq: 101 NOTIFY Content-Length: 0 Does anybody encountered this problem concerning MWI on sip phones only ?? Does it really work ?? Any suggestions are appreciated. Regards Claude. *Claude Friderich* *PreSales Support* *image001.gif* *NETCORE PSF S.A.* 49 rue du Baerendall B.P.65 L-8201 Mamer Téléphone: 31 33 80-407 Fax: 31 33 80 8-407 GSM: 621 303 616 E-mail: cfrider...@netcore.lu -- This email was Anti Virus checked. Disclaimer The information in this Internet e-mail is confidential and may be legally privileged. It is intended solely for the addressee. Access to this Internet e-mail by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. When addressed to our clients any opinions or advice contained in this e-mail are subject to the terms and conditions expressed in our governing terms of business. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com image001.gif___ For more information regarding industry leading CCIE Lab training,
Re: [OSL | CCIE_Voice] HQ GK to PSTN-WAN(Backbone) GK use g711foradmission request , a new bug?
It's already changed else the initial ARQ from CM won't be bandwidth 160.
My issue occurs only when HQ GK relay the message to a remote zone (Backbone
GK) , the ARQ becomes bandwidth 1280. I'm just puzzle by this. Is it a
normal behavior?
On Tue, Mar 1, 2011 at 3:49 AM, Friderich Claude cfrider...@netcore.luwrote:
Hi,
Change in the service parameters the default intraregion codec to G.729.
Should reolve the problem ….
Regards
Claude
*Claude Friderich*
*PreSales Support*
*[image: ccvp_voice_sm]***
*NETCORE PSF S.A.***
49 rue du Baerendall
B.P.65 L-8201 Mamer
Téléphone: 31 33 80-407
Fax: 31 33 80 8-407
GSM: 621 303 616
E-mail: cfrider...@netcore.lu
*From:* Ki Wi [mailto:kiwi.vo...@gmail.com]
*Sent:* lundi 28 février 2011 04:59
*To:* Friderich Claude
*Cc:* OSL Questions
*Subject:* Re: [OSL | CCIE_Voice] HQ GK to PSTN-WAN(Backbone) GK use
g711foradmission request , a new bug?
Looks like the behaviour is still the same. The ARQ message looks longer
now instead after enabling BRQ and of course there's this new BRQ message.
I always thought that from UCM -- HQ GK --- PSTN-WAN GK, between the 2 GK
the ARQ should be the same, however it proof me wrong? Maybe it's a bug?
This BRQ message from callmanager seems useless for call establishment. It
only comes in after ACF. I think it's useful during call transfer/conference
kind of things when media type might changed.
I have attached the logs for reference.
*HQ-PSTN*
*
*value RasMessage ::= admissionRequest :
{
requestSeqNum 1951
callType pointToPoint : NULL
endpointIdentifier {4A9F11A40002}
destinationInfo
{
dialedDigits : 916745738932
}
srcInfo
{
dialedDigits : 5001
}
srcCallSignalAddress ipAddress :
{
ip '0A0AD20B'H
port 60610
}
*bandWidth 160*
callReferenceValue 8
conferenceID '80909B2FAE18B1D608005503C0A8020A'H
activeMC FALSE*
PSTN-WAN *
Feb 28 03:38:22.069: RAS OUTGOING PDU ::=
value RasMessage ::= admissionRequest :
{
requestSeqNum 34029
callType pointToPoint : NULL
callModel direct : NULL
endpointIdentifier {48722121}
destinationInfo
{
dialedDigits : 916745738932
}
srcInfo
{
dialedDigits : 5001,
h323-ID : {Site A Home...}
}
srcCallSignalAddress ipAddress :
{
ip '0A0AD20B'H
port 60610
}
* bandWidth 1280*
callReferenceValue 55
nonStandardData
{
nonStandardIdentifier h221NonStandard :
{
t35CountryCode 181
t35Extension 0
manufacturerCode 18
}
data '8010C001810C1453697465204120486F6D65'
On Mon, Feb 28, 2011 at 8:59 AM, Friderich Claude cfrider...@netcore.lu
wrote:
Just to give a precision as I said manage your BW in your gk-trunk. I
wanted to say put the right device pool giving you g711 or g729 codec
(G711=128k and G729=16K)
Sorry for this lack of precision
Regards,
Claude
*Claude Friderich*
*PreSales Support*
Error! Filename not specified.
*NETCORE PSF S.A.*
49 rue du Baerendall
B.P.65 L-8201 Mamer
Téléphone: 31 33 80-407
Fax: 31 33 80 8-407
GSM: 621 303 616
E-mail: cfrider...@netcore.lu
*From:* ccie_voice-boun...@onlinestudylist.com [mailto:
ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Friderich Claude
*Sent:* dimanche 27 février 2011 21:14
*To:* Ki Wi; OSL Questions
*Subject:* Re: [OSL | CCIE_Voice] HQ GK to PSTN-WAN(Backbone) GK use
g711foradmission request , a new bug?
Go to Service parameters in CUCM and in Advanced services, put the BRQ
option to TRUE.
It lets you take into account the bw of the originating device (CUCM) in
the ARQ instead of GK
After that, manage your bw in your gk-trunk depending of what they ask in
the exam . It should work !!!
Regards,
Claude
*Claude Friderich*
*PreSales Support*
Error! Filename not specified.
*NETCORE PSF S.A.*
49 rue du Baerendall
B.P.65 L-8201 Mamer
Téléphone: 31 33 80-407
Fax: 31 33 80 8-407
GSM: 621 303 616
E-mail: cfrider...@netcore.lu
*From:* ccie_voice-boun...@onlinestudylist.com [mailto:
ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Ki Wi
*Sent:* dimanche 27 février 2011 08:30
*To:* OSL Questions
*Subject:* [OSL | CCIE_Voice] HQ GK to PSTN-WAN(Backbone) GK use g711
foradmission request , a new bug?
There's a famous bug around which the CM7.0.1 will use g711 for admission
request to GK , even it's set to g729
So i applied the workaround which is to set intracluster calling signalling
to g729, blah blah blah
Now if i call from UCM to PSTN-WAN GK , it will request for bandwidth
1280 even in HQ's GK the ARQ bandwidth is 160.
Anyone seem such behavior before? To verify it, i send the zone remote
bandwidth to
Re: [OSL | CCIE_Voice] Pressing Message button during SRST
Why did you not put for this user in CUC an alternate number with the complete calling number like 6178631XXX ?? Regards Claude Friderich PreSales Support NETCORE PSF S.A. 49 rue du Baerendall B.P.65 L-8201 Mamer Téléphone: 31 33 80-407 Fax: 31 33 80 8-407 GSM: 621 303 616 E-mail: cfrider...@netcore.lu mailto:cfrider...@netcore.lu From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ki Wi Sent: lundi 28 février 2011 20:00 To: OSL Questions Subject: [OSL | CCIE_Voice] Pressing Message button during SRST By default, it will end up on opening greeting when SRST user press the message button. Is there a way to make the users enter their own voicemail account directly(attempt sign in page) ? I'm aware of a way currently which is to set the calling number to in the hunt pilot but the method is not so graceful. There's chances that someone else last 4 digits number is the same or the system will recording the original calling number as 4 digits instead of maybe 10 or 11 digits long. Any interesting workaround for this? image001.gifimage002.gif___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Pressing Message button during SRST
There's always situation whereby you are restricted to do so. I have seem someone asking this before so I tried it out myself. Just wanted to test all possible scenario. On Tue, Mar 1, 2011 at 4:19 AM, Friderich Claude cfrider...@netcore.luwrote: Why did you not put for this user in CUC an alternate number with the complete calling number like 6178631XXX ?? Regards *Claude Friderich* *PreSales Support* *[image: ccvp_voice_sm]*** *NETCORE PSF S.A.*** 49 rue du Baerendall B.P.65 L-8201 Mamer Téléphone: 31 33 80-407 Fax: 31 33 80 8-407 GSM: 621 303 616 E-mail: cfrider...@netcore.lu *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Ki Wi *Sent:* lundi 28 février 2011 20:00 *To:* OSL Questions *Subject:* [OSL | CCIE_Voice] Pressing Message button during SRST By default, it will end up on opening greeting when SRST user press the message button. Is there a way to make the users enter their own voicemail account directly(attempt sign in page) ? I'm aware of a way currently which is to set the calling number to in the hunt pilot but the method is not so graceful. There's chances that someone else last 4 digits number is the same or the system will recording the original calling number as 4 digits instead of maybe 10 or 11 digits long. Any interesting workaround for this? image001.gifimage002.gif___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUC - CUCME - SCCP Integration - MWI issue on SIP Phones !!!!
Hi Friderich, Yep, as soon as I posted my last answer, I realised you had SCCP integration of CUC with CUCME not the SIP. You're right, you should have CUCME IP Address as mwi-server defined under sip-ua. it should be solicited!!! not unsolicited! let me know if it works please. Regards, On Mon, Feb 28, 2011 at 8:09 PM, Friderich Claude cfrider...@netcore.luwrote: Still not working …. Put mwi-server ipv4:ip address of CUC In voice service voip sip supplementary services was disabled But why should I put the ip address of CUC in mwi-server ?? CUCME with CUC integration is SCCP. I think the big issue is to pass the mwi updates through a SCCP connection. And with mwi-server configured as cucme ip address I gonna pass the mwi update for sip phones through sccp. It’s what I understand …. And currently, I can see the updates on CUC when the sip phone receives a voicemail. See below. It’s very strange …. Best Regards, * * *Claude Friderich* *PreSales Support* *[image: ccvp_voice_sm]*** *NETCORE PSF S.A.*** 49 rue du Baerendall B.P.65 L-8201 Mamer Téléphone: 31 33 80-407 Fax: 31 33 80 8-407 GSM: 621 303 616 E-mail: cfrider...@netcore.lu *From:* romain mullier [mailto:romain.mull...@gmail.com] *Sent:* lundi 28 février 2011 20:34 *To:* George Goglidze *Cc:* Friderich Claude; ShinGei Yong; OSL Questions *Subject:* Re: [OSL | CCIE_Voice] CUC - CUCME - SCCP Integration - MWI issue on SIP Phones you also will need to disable two sip supplementary services: refer and moved temporarily. On Mon, Feb 28, 2011 at 2:31 PM, George Goglidze gogli...@gmail.com wrote: Hi Friderich, The SIP server has to be Unity Connection. Try it out, it should work. Regards, On Mon, Feb 28, 2011 at 1:08 PM, Friderich Claude cfrider...@netcore.lu wrote: Hi all, Thanks for reply and your suggestions Effectively I’m doing a Cisco Unity Connection Integration not CUE. Imagine CUE isn’t present. Good Question Shingei J why CUCME IP address instead of CUC ??? Normally you have to put the CUC IP Address in mwi server when you integrate CUC with CUCME in a SIP configuration. But with SCCP integration and with SIP phones configured on CUCME, I think it’s correct, we have to put the CUCME ip address in mwi-server. Best Regards, Claude. *Claude Friderich* *PreSales Support* *[image: ccvp_voice_sm]* *NETCORE PSF S.A.* 49 rue du Baerendall B.P.65 L-8201 Mamer Téléphone: 31 33 80-407 Fax: 31 33 80 8-407 GSM: 621 303 616 E-mail: cfrider...@netcore.lu *From:* ShinGei Yong [mailto:shingei.y...@gmail.com] *Sent:* lundi 28 février 2011 08:42 *To:* Romain; Friderich Claude; OSL Questions *Subject:* Re: [OSL | CCIE_Voice] CUC - CUCME - SCCP Integration - MWI issue on SIP Phones My fren, Friderich is doing CUC integration with UCME, not CUE. Is there any reason that you mwi server to be 202.1 instead of CUC address? Shingei On Mon, Feb 28, 2011 at 11:25 AM, Romain romain.mull...@gmail.com wrote: Isn't your CUE at 10.10.202.2 ? That is the IP you need for your mwi server instead of .1 Sent from my iPhone On Feb 27, 2011, at 8:12 PM, Friderich Claude cfrider...@netcore.lu wrote: Hello, After a couple of tests and after trying to configure this feature, impossible to make it work L MWI on sccp phone is working fine MWI on SIP phone doesn’t work. Normaly in sip-ua *mwi-server ipv4:10.10.202.1 unsolicited* (CUCME IP Adress in telephony-service) In telephony-service *mwi relay* in the voice register dn 1 *mwi* *sip-ua * retry invite 3 timers trying 300 * mwi-server ipv4:10.10.202.1 expires 3600 port 5060 transport udp unsolicited* debug ccsip gives me the results below: Sent: NOTIFY sip:3005@10.10.202.50:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bK8F14A9 From: sip:3005@10.10.202.1;tag=137E2EC-1A2C To: sip:3005@10.10.202.50 Call-ID: 942D5193-420111E0-8216F0AE-E383DE3B@10.10.202.1 *CSeq: 101 NOTIFY* Max-Forwards: 70 Date: Sun, 27 Feb 2011 23:38:56 GMT User-Agent: Cisco-SIPGateway/IOS-12.x Event: message-summary Subscription-State: active Contact: sip:3005@10.10.202.1:5060 Content-Type: application/message-summary Content-Length: 23 *Messages-Waiting: yes* Feb 27 23:38:56.959: //-1//SIP/Msg/ccsipDisplayMsg: Received: *SIP/2.0 400 Bad Request* --- Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bK8F14A9 From: sip:3005@10.10.202.1;tag=137E2EC-1A2C To: sip:3005@10.10.202.50 Call-ID: 942D5193-420111E0-8216F0AE-E383DE3B@10.10.202.1 Date: Sun, 27 Feb 2011 23:38:55 GMT *Warning: 399 Bad MWI NOTIFY* - ? CSeq: 101 NOTIFY Content-Length: 0 Does anybody encountered this problem concerning MWI on sip phones only ?? Does it
Re: [OSL | CCIE_Voice] Pressing Message button during SRST
You should have first in telephony-service or call-manager fallback the command voicemail 5888 And a dial-peer : dial-peer voice 5600 pots translation-profile outgoing voicemail destination-pattern 5888 no digit-strip port 0/1/0:23 prefix 1212394 Claude Friderich PreSales Support NETCORE PSF S.A. 49 rue du Baerendall B.P.65 L-8201 Mamer Téléphone: 31 33 80-407 Fax: 31 33 80 8-407 GSM: 621 303 616 E-mail: cfrider...@netcore.lu mailto:cfrider...@netcore.lu From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ki Wi Sent: lundi 28 février 2011 20:13 To: OSL Questions Subject: Re: [OSL | CCIE_Voice] Pressing Message button during SRST There's one more way which is to translate the calling number from original number into 4 digits extension. This way works as well. I suppose this is the best solution? Is that vm-integration going to do some magic? I tried it just now but seems like it's not working for the message button. ! voice translation-rule 1 rule 1 /^617863\(\)/ /\1/ ! voice translation-rule 5 rule 1 // // type any national plan any isdn ! voice translation-profile voicemail translate calling 1 translate called 5 ! dial-peer voice 15 pots translation-profile outgoing voicemail destination-pattern 912123945888 port 0/1/0:23 forward-digits 0 prefix 12123945888 On Tue, Mar 1, 2011 at 3:00 AM, Ki Wi kiwi.vo...@gmail.com wrote: By default, it will end up on opening greeting when SRST user press the message button. Is there a way to make the users enter their own voicemail account directly(attempt sign in page) ? I'm aware of a way currently which is to set the calling number to in the hunt pilot but the method is not so graceful. There's chances that someone else last 4 digits number is the same or the system will recording the original calling number as 4 digits instead of maybe 10 or 11 digits long. Any interesting workaround for this? image001.gifimage002.gif___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MVA Hairpinin Remote destionation Match Problem
Yes! It was right under my nose…. And It was a very easy workaround ☺ Thank you very Claude! Regards, Francesc De: Friderich Claude [mailto:cfrider...@netcore.lu] Enviado el: lunes, 28 de febrero de 2011 21:45 Para: Roig Borrell, Francesc Xavier; ccie_voice@onlinestudylist.com Asunto: RE: [OSL | CCIE_Voice] MVA Hairpinin Remote destionation Match Problem I think that enabling an incoming voice translation-rule on your h323 dial-peer from your route pattern to this dial-peer is gonna resolve your problem. Of course just put the right calling number you want to have for DISA in your voice translation rule Regards Claude Friderich PreSales Support [cid:image001.gif@01CBD79F.2C85F6E0] NETCORE PSF S.A. 49 rue du Baerendall B.P.65 L-8201 Mamer Téléphone: 31 33 80-407 Fax: 31 33 80 8-407 GSM: 621 303 616 E-mail: cfrider...@netcore.lumailto:cfrider...@netcore.lu From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roig Borrell, Francesc Xavier Sent: lundi 28 février 2011 19:11 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MVA Hairpinin Remote destionation Match Problem Hi all, Working with MVA with hairpinin I have found and issue that I don’t know how to workaround it Hqph2 2002 Remotedestination, 6178632683 When I call to hqph1 from remote the identification is OK, 2002 The problem appears with DISA service. I does not recognize the remote destination number and it asks for entering it DISA Number in HQ 2123942010 application service mva http://10.10.210.10:8080/ccmivr/pages/IVRMainpage.vxml dial-peer voice 2010 voip service mva session target ipv4:10.10.210.10 incoming called-number 2010 codec g711ulaw ! dial-peer voice 20101 voip preference 1 destination-pattern 2010 voice-class h323 1 session target ipv4:10.10.210.11 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 20102 voip preference 2 destination-pattern 2010 voice-class h323 1 session target ipv4:10.10.210.10 dtmf-relay h245-alphanumeric codec g711ulaw no vad In the 1ºstage (incoming dialpeer 2010, mva application is triggered). Here is the problem when I call from the remote destination number to DISA in the HQMGCP gateway, CCM recognizes it an sends it as 2002 to h323 incoming leg so now mva ivr fails. Is anything I am missing? Does it work this way to you? Thanks in advance! Francesc Debug voice application vxml Feb 28 17:52:23.469: //300/80C50F960300/VXML:/vxml_submit_proc: submit: caching=fast fetchhint=invalid fetchtimeout=0 maxage=-1 maxstale=-1 URI(abs):http://10.10.210.10:8080/ccmivr/IVRCalleridLookup.do scheme=http host=10.10.210.10 port=8080 path=/ccmivr/IVRCalleridLookup.do Feb 28 17:52:23.469: //300/80C50F960300/VXML:/vxml_sub_attrs_proc: method=get enctype=application/x-www-form-urlencoded Feb 28 17:52:23.469: //300/80C50F960300/VXML:/vxml_nmtokens_proc: name=remotedest name=srcdir Feb 28 17:52:23.469: //300/80C50F960300/VXML:/vxml_vapp_bgload_from_proc: urlp=http://10.10.210.10:8080/ccmivr/IVRCalleridLookup.do?remotedest=2002srcdir=en_US fetchaudio=NULL delay=0 minimum=0 Feb 28 17:52:23.469: //300/80C50F960300/VXML:/vxml_vapp_bgload: url http://10.10.210.10:8080/ccmivr/IVRCalleridLookup.do?remotedest=2002srcdir=en_US cachable 1 fetchtimeout 0 maxage=-1 maxstale=-1 Feb 28 17:52:23.469: //300//AFW_:/vapp_bgload: url=http://10.10.210.10:8080/ccmivr/IVRCalleridLookup.do?remotedest=2002srcdir=en_US Feb 28 17:52:23.469: //300//AFW_:/vxml_update_cleanup_timer: cleaning timer running 0 fetchtimeout 0 Feb 28 17:52:23.469: //300/80C50F960300/VXML:/vxml_leave_scope: scope=anonymous Feb 28 17:52:23.469: //300/80C50F960300/VXML:/vxml_load_immediate_done: ... -- This email was Anti Virus checked. inline: image001.gif___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Pressing Message button during SRST
Hi Kiwi, What is the ANI when the call arrive HQ GW? Shingei On Tue, Mar 1, 2011 at 6:16 AM, Friderich Claude cfrider...@netcore.luwrote: You should have first in telephony-service or call-manager fallback the command voicemail 5888 And a dial-peer : dial-peer voice 5600 pots translation-profile outgoing voicemail destination-pattern 5888 no digit-strip port 0/1/0:23 prefix 1212394 *Claude Friderich* *PreSales Support* *[image: ccvp_voice_sm]*** *NETCORE PSF S.A.*** 49 rue du Baerendall B.P.65 L-8201 Mamer Téléphone: 31 33 80-407 Fax: 31 33 80 8-407 GSM: 621 303 616 E-mail: cfrider...@netcore.lu *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Ki Wi *Sent:* lundi 28 février 2011 20:13 *To:* OSL Questions *Subject:* Re: [OSL | CCIE_Voice] Pressing Message button during SRST There's one more way which is to translate the calling number from original number into 4 digits extension. This way works as well. I suppose this is the best solution? Is that vm-integration going to do some magic? I tried it just now but seems like it's not working for the message button. ! voice translation-rule 1 rule 1 /^617863\(\)/ /\1/ ! voice translation-rule 5 rule 1 // // type any national plan any isdn ! voice translation-profile voicemail translate calling 1 translate called 5 ! dial-peer voice 15 pots translation-profile outgoing voicemail destination-pattern 912123945888 port 0/1/0:23 forward-digits 0 prefix 12123945888 On Tue, Mar 1, 2011 at 3:00 AM, Ki Wi kiwi.vo...@gmail.com wrote: By default, it will end up on opening greeting when SRST user press the message button. Is there a way to make the users enter their own voicemail account directly(attempt sign in page) ? I'm aware of a way currently which is to set the calling number to in the hunt pilot but the method is not so graceful. There's chances that someone else last 4 digits number is the same or the system will recording the original calling number as 4 digits instead of maybe 10 or 11 digits long. Any interesting workaround for this? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com image001.gifimage002.gif___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Pressing Message button during SRST
Depends on what I set, if I used the translation rule it will be 4 digits. I'm asking this because there's some special key sequence for unity family such as ##2 will bring you directly into the target's voicemail greeting and let you leave a voicemail. I'm wondering is there any other such special key sequence? It is useful for a lot of deployment as well especially a lot of countries are still using analog lines but what if those are hunting lines ? The user will never get into their own voicemail even with alternate extension set. I have a upcoming deployment which uses IP for signaling but actual voice will still go thru PSTN. It will be good to know extra options! If not, the voicemail access will have to stay on the VoIP network while leaving a voicemail and voice call will be on PSTN. Sent from my iPhone Pls pardon my fat fingers. On Mar 1, 2011, at 10:22 AM, ShinGei Yong shingei.y...@gmail.com wrote: Hi Kiwi, What is the ANI when the call arrive HQ GW? Shingei On Tue, Mar 1, 2011 at 6:16 AM, Friderich Claude cfrider...@netcore.lu wrote: You should have first in telephony-service or call-manager fallback the command voicemail 5888 And a dial-peer : dial-peer voice 5600 pots translation-profile outgoing voicemail destination-pattern 5888 no digit-strip port 0/1/0:23 prefix 1212394 Claude Friderich PreSales Support image001.gif NETCORE PSF S.A. 49 rue du Baerendall B.P.65 L-8201 Mamer Téléphone: 31 33 80-407 Fax: 31 33 80 8-407 GSM: 621 303 616 E-mail: cfrider...@netcore.lu From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ki Wi Sent: lundi 28 février 2011 20:13 To: OSL Questions Subject: Re: [OSL | CCIE_Voice] Pressing Message button during SRST There's one more way which is to translate the calling number from original number into 4 digits extension. This way works as well. I suppose this is the best solution? Is that vm-integration going to do some magic? I tried it just now but seems like it's not working for the message button. ! voice translation-rule 1 rule 1 /^617863\(\)/ /\1/ ! voice translation-rule 5 rule 1 // // type any national plan any isdn ! voice translation-profile voicemail translate calling 1 translate called 5 ! dial-peer voice 15 pots translation-profile outgoing voicemail destination-pattern 912123945888 port 0/1/0:23 forward-digits 0 prefix 12123945888 On Tue, Mar 1, 2011 at 3:00 AM, Ki Wi kiwi.vo...@gmail.com wrote: By default, it will end up on opening greeting when SRST user press the message button. Is there a way to make the users enter their own voicemail account directly(attempt sign in page) ? I'm aware of a way currently which is to set the calling number to in the hunt pilot but the method is not so graceful. There's chances that someone else last 4 digits number is the same or the system will recording the original calling number as 4 digits instead of maybe 10 or 11 digits long. Any interesting workaround for this? image002.gif ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] HQ GK to PSTN-WAN(Backbone) GK use g711foradmission request , a new bug?
http://www.ciscosystems.com/en/US/tech/tk1077/technologies_white_paper09186a00800c5f67.shtml
From troubleshooting gatekeeper bandwidth management, seems like it's
normal?
On Tue, Mar 1, 2011 at 3:52 AM, Ki Wi kiwi.vo...@gmail.com wrote:
It's already changed else the initial ARQ from CM won't be bandwidth
160.
My issue occurs only when HQ GK relay the message to a remote zone
(Backbone GK) , the ARQ becomes bandwidth 1280. I'm just puzzle by this.
Is it a normal behavior?
On Tue, Mar 1, 2011 at 3:49 AM, Friderich Claude cfrider...@netcore.luwrote:
Hi,
Change in the service parameters the default intraregion codec to G.729.
Should reolve the problem ….
Regards
Claude
*Claude Friderich*
*PreSales Support*
*[image: ccvp_voice_sm]***
*NETCORE PSF S.A.***
49 rue du Baerendall
B.P.65 L-8201 Mamer
Téléphone: 31 33 80-407
Fax: 31 33 80 8-407
GSM: 621 303 616
E-mail: cfrider...@netcore.lu
*From:* Ki Wi [mailto:kiwi.vo...@gmail.com]
*Sent:* lundi 28 février 2011 04:59
*To:* Friderich Claude
*Cc:* OSL Questions
*Subject:* Re: [OSL | CCIE_Voice] HQ GK to PSTN-WAN(Backbone) GK use
g711foradmission request , a new bug?
Looks like the behaviour is still the same. The ARQ message looks longer
now instead after enabling BRQ and of course there's this new BRQ message.
I always thought that from UCM -- HQ GK --- PSTN-WAN GK, between the 2
GK the ARQ should be the same, however it proof me wrong? Maybe it's a bug?
This BRQ message from callmanager seems useless for call establishment. It
only comes in after ACF. I think it's useful during call transfer/conference
kind of things when media type might changed.
I have attached the logs for reference.
*HQ-PSTN*
*
*value RasMessage ::= admissionRequest :
{
requestSeqNum 1951
callType pointToPoint : NULL
endpointIdentifier {4A9F11A40002}
destinationInfo
{
dialedDigits : 916745738932
}
srcInfo
{
dialedDigits : 5001
}
srcCallSignalAddress ipAddress :
{
ip '0A0AD20B'H
port 60610
}
*bandWidth 160*
callReferenceValue 8
conferenceID '80909B2FAE18B1D608005503C0A8020A'H
activeMC FALSE*
PSTN-WAN *
Feb 28 03:38:22.069: RAS OUTGOING PDU ::=
value RasMessage ::= admissionRequest :
{
requestSeqNum 34029
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On Mon, Feb 28, 2011 at 8:59 AM, Friderich Claude cfrider...@netcore.lu
wrote:
Just to give a precision as I said manage your BW in your gk-trunk. I
wanted to say put the right device pool giving you g711 or g729 codec
(G711=128k and G729=16K)
Sorry for this lack of precision
Regards,
Claude
*Claude Friderich*
*PreSales Support*
Error! Filename not specified.
*NETCORE PSF S.A.*
49 rue du Baerendall
B.P.65 L-8201 Mamer
Téléphone: 31 33 80-407
Fax: 31 33 80 8-407
GSM: 621 303 616
E-mail: cfrider...@netcore.lu
*From:* ccie_voice-boun...@onlinestudylist.com [mailto:
ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Friderich Claude
*Sent:* dimanche 27 février 2011 21:14
*To:* Ki Wi; OSL Questions
*Subject:* Re: [OSL | CCIE_Voice] HQ GK to PSTN-WAN(Backbone) GK use
g711foradmission request , a new bug?
Go to Service parameters in CUCM and in Advanced services, put the BRQ
option to TRUE.
It lets you take into account the bw of the originating device (CUCM) in
the ARQ instead of GK
After that, manage your bw in your gk-trunk depending of what they ask in
the exam . It should work !!!
Regards,
Claude
*Claude Friderich*
*PreSales Support*
Error! Filename not specified.
*NETCORE PSF S.A.*
49 rue du Baerendall
B.P.65 L-8201 Mamer
Téléphone: 31 33 80-407
Fax: 31 33 80 8-407
GSM: 621 303 616
E-mail: cfrider...@netcore.lu
*From:* ccie_voice-boun...@onlinestudylist.com [mailto:
ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Ki Wi
*Sent:* dimanche 27 février 2011 08:30
*To:* OSL Questions
*Subject:* [OSL | CCIE_Voice] HQ GK to PSTN-WAN(Backbone) GK use g711
foradmission request , a new bug?
There's a famous bug around which the CM7.0.1 will use g711 for admission
request to GK , even it's set to g729
So i applied the workaround which is to set intracluster
