Zend PHP calls are done sequentially?
I have a page that makes 4 different calls to php functions through the zend framework built into Flex 4. If I watch in a web proxy program like Charles it would appear that the calls are happening one after another. I would think they should all fire individually and return at there own given time.
..... What???
Similar Messages
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how can I send multiple pictures to the Kodak link (under file & then order prints)? if the photos are not sequential can it be done?
Does the order matter here?Because the photos will be sent for printing. If there is any other reason, can you please cite example.
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How i can use "don't disturb", but calls are coming without sounds
i want to use the "don't disturb" function.
but when i activate it, all call are send to my Voice-Box. But i want that the call is sign at my iphon without sound. so it is when i activate the MuteButton at the right side of the iphoneTry:
- iOS: Not responding or does not turn on
- Also try DFU mode after try recovery mode
How to put iPod touch / iPhone into DFU mode « Karthik's scribblings
- If not successful and you can't fully turn the iOS device fully off, let the battery fully drain. After charging for an least an hour try the above again.
- If still not successful that usually indicates a hardware problem and an appointment at the Genius Bar of an Apple store is in order. -
I was so excited about apple pay and now setting it up and boom 8 cards and you are done is that the wallet replacement they thought they'll do? really? I mean most people have 10-15 cards at least and I am, well because of my side business I have lots so 8 is nothing to me!
Not sure what you are saying. I was able to very quickly set up my AmEx card - no problems at all. For my PNC and Macy's cards, companies still working on being ApplePay-ready.
It will happen, just not as fast as we might like, but I think it is going to be fabulous. -
Built-In PHP Functions Are Not Recognized
I have a PHP project in Flash Builder for PHP 4.5 which does not recognise any of the PHP in-built functions. I have tested this on a brand new project and still the functions I type are not recognised. So I can't basically auto-complete them or look at what the function definition is.
Usually the built-in function are listed under "PHP Language Library" > "Core API". But in my PHP projects they don't show at all. Any thoughts?
I raised this problem with Zend (Which is where I bought the product from) Support and this is what they had to say:
Unfortunately, Zend not supporting the Flash Builder for PHP.
You can use the Adobe online forums to find solutions for your technical issues:
http://forums.adobe.com/community/flash_builder/
and this:
Please use the Adobe Forums to report your problem. If the problem is related to Zend part of the product (it definitely looks like this), we will help you to solve this problem - through Adobe Forums. If you browse Adobe Forums, you may have noticed that our staff is quite responsive. Just post your issue there and you will see that routing your request through the correct channel will ensure the fastest resolution.
So I am hoping Zend/Adobe Engineers are here to help me with the problem I have. ThanksHi Roy,
Here is the contents of the project .buildpath
<?xml version="1.0" encoding="UTF-8"?>
<buildpath>
<buildpathentry kind="con" path="com.zend.php.phpunit.CONTAINER"/>
<buildpathentry kind="con" path="org.eclipse.php.core.LANGUAGE"/>
<buildpathentry kind="src" path=""/>
</buildpath>
and .project
<?xml version="1.0" encoding="UTF-8"?>
<projectDescription>
<name>NAV-Contacts-PHP</name>
<comment></comment>
<projects>
</projects>
<buildSpec>
<buildCommand>
<name>org.eclipse.wst.validation.validationbuilder</name>
<arguments>
</arguments>
</buildCommand>
<buildCommand>
<name>org.eclipse.dltk.core.scriptbuilder</name>
<arguments>
</arguments>
</buildCommand>
</buildSpec>
<natures>
<nature>org.eclipse.php.core.PHPNature</nature>
<nature>org.zend.php.framework.ZendFrameworkNature</nature>
</natures>
</projectDescription>
The OS is MacOS X 10.6.7 -
Calls are not getting thru in Cisco voice GW for a particular Number
Cisco gateway is connecte to a PBX with an Qsig interface, for a particualr destination number the calls are not gettin estabilished.
the output of the Q931 debug :
Aug 16 16:17:46.145: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8 callref = 0x7E05
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98396
Exclusive, Channel 22
Facility i = 0x9FAA068001008201008B0100A16E0202070102011530650201010A010
1800101A111A00FA50D0A010212083530303035393938A211A00FA50D0A010212083530303035393
938A312801054454C45434F4D20574F524B524F4F4DA412801054454C45434F4D20574F524B524F4
F4DA50C06062B0C02FF373730020500
Facility i = 0x9FAA068001008201008B0100A11D0202010002010080144E455453202
F204C4F4E472044495354414E4345
Calling Party Number i = 0x2183, '8168911010'
Plan:ISDN, Type:National
Called Party Number i = 0x89, '18553808521'
Plan:Private, Type:Unknown
Sending Complete
Aug 16 16:17:46.149: ISDN Se0/0/0:23 Q931: TX -> CALL_PROC pd = 8 callref = 0xF
E05
Channel ID i = 0xA98396
Exclusive, Channel 22
Aug 16 16:17:55.709: ISDN Se0/0/0:23 Q931: TX -> DISCONNECT pd = 8 callref = 0x
FE05
Cause i = 0x80BF - Service/option not available, unspecified
Aug 16 16:17:55.741: ISDN Se0/0/0:23 Q931: RX <- RELEASE pd = 8 callref = 0x7E0
5
Aug 16 16:17:55.741: ISDN Se0/0/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref =
0xFE05
The Qsig and dial-peer configration :
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-qsig
isdn overlap-receiving
isdn incoming-voice voice
isdn send-alerting
no cdp enable
dial-peer voice 1 voip
description To CBTS GK
destination-pattern +1T
signaling forward rawmsg
session protocol sipv2
session target ipv4:10.9.5.10
session transport tcp
voice-class codec 1
dtmf-relay rtp-nte
no vad
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-qsig
isdn overlap-receiving
isdn incoming-voice voice
isdn send-alerting
no cdp enable
dial-peer voice 1 voip
description To CBTS GK
destination-pattern +1T
signaling forward rawmsg
session protocol sipv2
session target ipv4:10.9.5.10
session transport tcp
voice-class codec 1
dtmf-relay rtp-nte
no vadHi Raj,
My name is Edson Pineiro, I understand that your problem description is in regards to failed incoming calls from a qsig trunk.
According to the received q931 setup message I can see the called party number is 18553808521 and as so the gateway should route the dnis based on the best match in destination-pattern. My first suggestion would be to ensure your outgoing dial-peers has a matching destination-pattern that matches the dialed number, for example:
dial-peer voice 1 voip
destination-pattern 1T
The T is a wild card for any digit any length
Or you can be very specific.
dial-peer voice 1 voip
destinaton-pattern 18553808521
The next suggestion would be to ensure that your incoming pots dial-peers contains 'direct-inward dial'. This is so that you don't receive secondary dial tone when dialing in, which I don't think is happening here.
Another suggestion would be to remove 'isdn overlap-receiving' from interface serial 0/0/0:23. Reason being is that the DNIS received is enbloc and not overlapping. You can clearly see that the complete e164 number is received within the setup and no further digits are needed.
But overall the disconnect cause code is 0x80BF the 80 portion is related to the source of the disconnect which is the router and BF "Service/option not available, unspecified" which is described as:
The network or remote equipment cannot provide the service option that the user requests, due to an unspecified reason. A subscription problem can cause this issue.
Any ways seems like the router does not support the protocol or type of message included in the Setup. After decoding one of the facility message:
Facility i = 0x9FAA068001008201008B0100A16E0202070102011530650201010A010
1800101A111A00FA50D0A010212083530303035393938A211A00FA50D0A010212083530303035393
938A312801054454C45434F4D20574F524B524F4F4DA412801054454C45434F4D20574F524B524F4
F4DA50C06062B0C02FF373730020500
decode -->
Facility IE first byte (protocol profile): 0x9f(Network Extentions), depends on Network Protocol Profile
**Note:
**0x91/0x9f both be used by older qsig spec, including:
**ISO 11582:1995, ETSI 300 239:1993/1995
**newer qsig spec use 0x9f only, including:
**ISO 11582:1995/Cor.1:1999, ECMA 165(4th), ETSI 300 239:2003
**see CSCeb58118 for CCM compatibility issue
NetworkFacilityExtension ::= {
sourceEntity: 0
destinationEntity: 0
NetworkProtocolProfile not present
APDU is a ROSE
0
DivertingLegInformation2Invoke ::= {
invokeID: 1793
operationValue: 21
argument: DivertingLegInformation2Arg ::= {
diversionCounter: 1
diversionReason: 1
originalDiversionReason: 1
divertingNr: PrivatePartyNumber ::= {
privateTypeOfNumber: 2
privateNumberDigits: 50005998
originalCalledNr: PrivatePartyNumber ::= {
privateTypeOfNumber: 2
privateNumberDigits: 50005998
redirectingName: 54 45 4C 45 43 4F 4D 20 57 4F 52 4B 52 4F 4F 4D
originalCalledName: 54 45 4C 45 43 4F 4D 20 57 4F 52 4B 52 4F 4F 4D
Looks like this is a redirected call (call forward or transfer), the redireted number is "50005998" and the other end of the PRI maybe attempting to do either a 2 B channel transfer or B channel optimization, which is not supported certain gateways or needs the use of a tcl scripts. Any ways is it possible to confirm if such features are enabled on the other end of the qsig trunk? and what the number 50005998 is assigned too. This may warrant a TAC case.
However please ensure your carry through the first three configuration changes before looking at the possible bad facility message.
Here are some good documents on ISDN, IOS dial-peers and call legs:
Understanding debug isdn q931 Disconnect Cause Codes
http://www.cisco.com/en/US/tech/tk801/tk379/technologies_tech_note09186a008012e95f.shtml
Configuring Telephony Call-Redirect Features
Two B-Channel Transfer
http://www.cisco.com/en/US/docs/ios/voice/ivr/pre12.3_14_t/configuration/guide/ivrapp.pdf
Understanding Dial Peers and Call Legs on Cisco IOS Platforms
http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a008010ae1c.shtml
Understanding Direct-Inward-Dial (DID) on IOS Voice Digital (T1/E1) Interfaces
http://www.cisco.com/en/US/partner/tech/tk652/tk653/technologies_tech_note09186a00801142f8.shtml
Understanding Inbound and Outbound Dial Peers Matching on IOS Platforms
http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml#prereq
Voice Translation Rules
http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a0080325e8e.shtml
Let me know how you go.
Thanks again for asking the tuff questions.
Cheers
Edson -
Incoming calls are going directly to Voicemail, Iphone 5S
I switched phones from an Iphone 4 to this Iphone 5S and my incoming calls are going directly to voice-mail and not ringing the phone. The only app I downloaded that is different from my Iphone 4 is the GroupMe app, but I don't know if that is necessarily the cause. Anyone have a suggestion? thanks
Hi, I don't know if this info will help many, but for a few it may. I had this problem over the weekend. Called Tech Support 2x, they tried everything, resetting,, *73, *228, *900 , there were no "do not disturb" settings, etc. They prepared a work ticket for the next level of help.
Meanwhile, I couldn't figure out what changed all of a sudden. I knew I had no calls coming in after accessing my voicemail. I looked at my recent calls and noticed i had dialed a *83 prior to the voicemail *86. Fat thumbs. I remember years ago, having the features of *83 for do not disturb and *84 to stop the feature. Just for giggles, i dialed *84, tried from another phone calling my number and WHOOLA, it came thru.
If anyone has these issues, try the *73 first. But just for the heck of it, do *84 as well. Takes half a second. If those dont' work, then call Tech Support. I wanted to share to help at least some of you as I understand the frustration of not having incoming calls. I even called Tech Support and shared the info. As this was not an option given to me from Tech Support, but then again, i don't believe *83 or *84 is offered anymore. I just dated myself, but hope it helps others ! -
Issue with SPA525g registation and FXO port call calls are not disconnecting properly
Hi,
I have a UC540 and updated it to the latest IOS version with the latest firmware to my phones and i am having registration problems with SPA525g IP Phones. I updated the firmware of the phones as well and create manual tftp bindings with but still it is not registering. I run a couple of debugs (debug tftp events and debug ephone registration) I can see from the logs and in the phone that it is taking the proper VLAN and being discovered via CDP and being pointed to the TFTP server and still wont register. I can see that it is also taking its own .cnf file properly then the output sccp token regected invalid devices error is shown I have a SPA502G and it is working fine. Also there is a previous issue that all the voice port are shown as engage or offhook even the calls are disconnected thus make the main PSTN number busy am based in UAE and our service provider is etisalat I have check with them about the proper disconnection values but still it the same. That's why I have arrived in the conclusion to just update everything including the IOS and the phones firmware. I have put my config in this post, I am also trying to take the CCNA Voice exam on the 2nd week of april and I think that if i don't know how fix this issue for our customer then I would probably fail that exam. any suggestion and help is greatly appreciated cisco experts.
! Last configuration change at 13:36:42 ZP4 Thu Sep 13 2012 by Nick
! NVRAM config last updated at 13:45:41 ZP4 Thu Sep 13 2012 by Nick
version 15.1
parser config cache interface
no service pad
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
service internal
service compress-config
service sequence-numbers
hostname UC540
boot-start-marker
boot system flash:uc500-advipservicesk9-mz.151-2.T4
boot-end-marker
logging buffered 64000
enable secret 5 $1$3CIf$.rXyHeJQrwd97X/f2dS0M1
no aaa new-model
clock timezone ZP4 4 0
crypto pki token default removal timeout 0
crypto pki trustpoint TP-self-signed-3558175224
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-3558175224
revocation-check none
crypto pki certificate chain TP-self-signed-3558175224
certificate self-signed 01 nvram:IOS-Self-Sig#3.cer
dot11 syslog
dot11 ssid cisco-data
vlan 1
authentication open
dot11 ssid cisco-voice
vlan 100
authentication open
ip source-route
ip cef
ip dhcp relay information trust-all
ip dhcp excluded-address 10.1.3.1 10.1.3.10
ip dhcp pool phone
network 10.1.3.0 255.255.255.0
default-router 10.1.3.1
option 150 ip 10.1.3.1
ip name-server 213.42.20.20
ip name-server 195.229.241.222
ip inspect WAAS flush-timeout 10
ip inspect name SDM_LOW cuseeme
ip inspect name SDM_LOW dns
ip inspect name SDM_LOW ftp
ip inspect name SDM_LOW h323
ip inspect name SDM_LOW https
ip inspect name SDM_LOW icmp
ip inspect name SDM_LOW imap
ip inspect name SDM_LOW pop3
ip inspect name SDM_LOW netshow
ip inspect name SDM_LOW rcmd
ip inspect name SDM_LOW realaudio
ip inspect name SDM_LOW rtsp
ip inspect name SDM_LOW esmtp
ip inspect name SDM_LOW sqlnet
ip inspect name SDM_LOW streamworks
ip inspect name SDM_LOW tftp
ip inspect name SDM_LOW tcp router-traffic
ip inspect name SDM_LOW udp router-traffic
ip inspect name SDM_LOW vdolive
no ipv6 cef
multilink bundle-name authenticated
stcapp ccm-group 1
stcapp
stcapp supplementary-services
port 0/0/0
fallback-dn 301
port 0/0/1
fallback-dn 302
port 0/0/2
fallback-dn 303
port 0/0/3
fallback-dn 304
trunk group ALL_FXO
max-retry 5
voice-class cause-code 1
hunt-scheme longest-idle
translation-profile outgoing PROFILE_ALL_FXO
trunk group ALL_FX0
voice call send-alert
voice rtp send-recv
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
sip
no update-callerid
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
voice class dualtone-detect-params 1
freq-max-deviation 50
freq-max-power 0
freq-min-power 13
freq-power-twist 4
cadence-variation 6
voice class custom-cptone UAE-CUSTOM
dualtone disconnect
frequency 406
cadence 398 344 237 527 400
voice class custom-cptone CCAjointone
dualtone conference
frequency 600 900
cadence 300 150 300 100 300 50
voice class custom-cptone CCAleavetone
dualtone conference
frequency 400 800
cadence 400 50 200 50 200 50
voice class cause-code 1
no-circuit
voice register global
voice hunt-group 1 parallel
list 301,302,303
timeout 24
pilot 511
voice translation-rule 4
rule 15 // //
voice translation-rule 1000
rule 1 /.*/ //
voice translation-rule 1111
voice translation-rule 1112
rule 1 /^9/ //
rule 3 /^0/ //
voice translation-rule 2222
voice translation-rule 3265
rule 1 /\(^..........$\)/ /9\1/
rule 2 /\(^.........$\)/ /9\1/
rule 15 /\(^ABCD$\)/ /ABCD\1/
voice translation-profile CALLER_ID_TRANSLATION_PROFILE
translate calling 1111
voice translation-profile CallBlocking
translate called 2222
voice translation-profile INCOMING_CallerID_PROFILE
translate calling 3265
voice translation-profile OUTGOING_TRANSLATION_PROFILE
translate called 1112
voice translation-profile PROFILE_ALL_FXO
translate calling 4
voice translation-profile nondialable
translate called 1000
voice-card 0
dspfarm
dsp services dspfarm
license udi pid UC540W-FXO-K9 sn FHK143074G6
archive
log config
logging enable
logging size 600
hidekeys
username cisco privilege 15 secret 5 $1$vjNa$OFKLhupqR8al6x2b8Xmcj/
username adminac privilege 15 secret 5 $1$NDC.$PtD0y4YGIj5SqI1gghxWE1
username Nick privilege 15 secret 5 $1$iAmL$tsg7Jf2TEND1NN.h8z2dy/
ip tftp source-interface Loopback0
bridge irb
interface Loopback0
description $FW_INSIDE$
ip address 10.1.10.2 255.255.255.252
ip access-group 101 in
ip nat inside
ip virtual-reassembly in
interface FastEthernet0/0
description $FW_OUTSIDE$
ip address 192.168.101.2 255.255.255.252
ip nat outside
ip virtual-reassembly in
duplex auto
speed auto
interface Integrated-Service-Engine0/0
description cue is initialized with default IMAP group
ip unnumbered Loopback0
ip nat inside
ip virtual-reassembly in
service-module ip address 10.1.10.1 255.255.255.252
service-module ip default-gateway 10.1.10.2
interface FastEthernet0/1/0
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/1
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/2
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/3
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/4
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/5
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/6
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/7
switchport access vlan 20
spanning-tree portfast
interface FastEthernet0/1/8
switchport access vlan 100
macro description cisco-switch
interface Dot11Radio0/5/0
no ip address
shutdown
ssid cisco-data
ssid cisco-voice
speed basic-1.0 basic-2.0 basic-5.5 6.0 9.0 basic-11.0 12.0 18.0 24.0 36.0 48.0 54.0
station-role root
interface Dot11Radio0/5/0.1
encapsulation dot1Q 1 native
bridge-group 1
bridge-group 1 subscriber-loop-control
bridge-group 1 spanning-disabled
bridge-group 1 block-unknown-source
no bridge-group 1 source-learning
no bridge-group 1 unicast-flooding
interface Dot11Radio0/5/0.100
encapsulation dot1Q 100
bridge-group 100
bridge-group 100 subscriber-loop-control
bridge-group 100 spanning-disabled
bridge-group 100 block-unknown-source
no bridge-group 100 source-learning
no bridge-group 100 unicast-flooding
interface Vlan1
no ip address
bridge-group 1
bridge-group 1 spanning-disabled
interface Vlan20
ip address 10.10.10.1 255.255.255.0
interface Vlan100
no ip address
bridge-group 100
bridge-group 100 spanning-disabled
interface BVI1
description $FW_INSIDE$
no ip address
ip nat inside
ip virtual-reassembly in
shutdown
interface BVI100
description $FW_INSIDE$
ip address 10.1.3.1 255.255.255.0
ip nat inside
ip virtual-reassembly in
ip forward-protocol nd
ip http server
ip http authentication local
ip http secure-server
ip http path flash:/gui
ip dns server
ip nat inside source list 1 interface FastEthernet0/0 overload
ip route 0.0.0.0 0.0.0.0 192.168.101.1
ip route 10.1.10.1 255.255.255.255 Integrated-Service-Engine0/0
logging esm config
access-list 1 remark SDM_ACL Category=2
access-list 1 permit 192.168.10.0 0.0.0.255
access-list 1 permit 10.1.3.0 0.0.0.255
access-list 1 permit 10.1.10.0 0.0.0.3
access-list 100 remark auto generated by SDM firewall configuration
access-list 100 remark SDM_ACL Category=1
access-list 100 deny ip 192.168.10.0 0.0.0.255 any
access-list 100 deny ip host 255.255.255.255 any
access-list 100 deny ip 127.0.0.0 0.255.255.255 any
access-list 100 permit ip any any
access-list 101 remark auto generated by SDM firewall configuration##NO_ACES_8##
access-list 101 remark SDM_ACL Category=1
access-list 101 permit tcp 10.1.3.0 0.0.0.255 eq 2000 any
access-list 101 permit udp 10.1.3.0 0.0.0.255 eq 2000 any
access-list 101 deny ip 10.1.3.0 0.0.0.255 any
access-list 101 deny ip 192.168.10.0 0.0.0.255 any
access-list 101 deny ip 192.168.101.0 0.0.0.3 any
access-list 101 deny ip host 255.255.255.255 any
access-list 101 deny ip 127.0.0.0 0.255.255.255 any
access-list 101 permit ip any any
access-list 102 remark auto generated by SDM firewall configuration##NO_ACES_6##
access-list 102 remark SDM_ACL Category=1
access-list 102 deny ip 10.1.10.0 0.0.0.3 any
access-list 102 deny ip 10.1.3.0 0.0.0.255 any
access-list 102 deny ip 192.168.101.0 0.0.0.3 any
access-list 102 deny ip host 255.255.255.255 any
access-list 102 deny ip 127.0.0.0 0.255.255.255 any
access-list 102 permit ip any any
access-list 102 permit ip 192.168.101.0 0.0.0.3 any
access-list 103 remark auto generated by SDM firewall configuration##NO_ACES_8##
access-list 103 remark SDM_ACL Category=1
access-list 103 permit tcp 10.1.10.0 0.0.0.3 any eq 2000
access-list 103 permit udp 10.1.10.0 0.0.0.3 any eq 2000
access-list 103 deny ip 10.1.10.0 0.0.0.3 any
access-list 103 deny ip 192.168.10.0 0.0.0.255 any
access-list 103 deny ip 192.168.101.0 0.0.0.3 any
access-list 103 deny ip host 255.255.255.255 any
access-list 103 deny ip 127.0.0.0 0.255.255.255 any
access-list 103 permit ip any any
access-list 105 permit ip any any
snmp-server community public RO
tftp-server flash:/phones/521_524/cp524g-8-1-17.bin alias cp524g-8-1-17.bin
tftp-server flash:/phones/5x5/spa5x5-7-1-3c.bin alias spa5x5-7-1-3c.bin
tftp-server flash:/phones/525/spa525g-7-4-8.bin alias spa525g-7-4-8.bin
control-plane
bridge 1 route ip
bridge 100 route ip
voice-port 0/0/0
cptone GB
station-id name Cordless
station-id number 329
caller-id enable
voice-port 0/0/1
cptone AE
caller-id enable
voice-port 0/0/2
cptone AE
caller-id enable
voice-port 0/0/3
cptone AE
caller-id enable
voice-port 0/1/0
trunk-group ALL_FX0 64
translation-profile incoming INCOMING_CallerID_PROFILE
supervisory disconnect dualtone mid-call
supervisory custom-cptone UAE-CUSTOM
input gain 14
cptone GB
connection plar opx 511
impedance 600c
description Configured by CCA 4FXO-0/1/0-Custom-BG
bearer-cap Speech
caller-id enable
voice-port 0/1/1
trunk-group ALL_FX0 64
translation-profile incoming INCOMING_CallerID_PROFILE
supervisory disconnect dualtone mid-call
supervisory custom-cptone UAE-CUSTOM
input gain 14
cptone GB
connection plar opx 511
impedance 600c
description Configured by CCA 4 FXO-0/1/1-Custom-BG
bearer-cap Speech
caller-id enable
voice-port 0/1/2
trunk-group ALL_FX0 64
translation-profile incoming INCOMING_CallerID_PROFILE
supervisory disconnect dualtone mid-call
supervisory custom-cptone UAE-CUSTOM
supervisory dualtone-detect-params 1
input gain 14
cptone GB
connection plar opx 511
impedance 600c
description Configured by CCA 4 FXO-0/1/2-Custom-BG
bearer-cap Speech
caller-id enable
voice-port 0/1/3
trunk-group ALL_FX0 64
translation-profile incoming INCOMING_CallerID_PROFILE
supervisory disconnect dualtone mid-call
supervisory custom-cptone UAE-CUSTOM
input gain 14
cptone GB
connection plar opx 511
impedance 600c
description Configured by CCA 4 FXO-0/1/3-Custom-BG
bearer-cap Speech
caller-id enable
voice-port 0/4/0
auto-cut-through
signal immediate
input gain auto-control -15
description Music On Hold Port
sccp local Loopback0
sccp ccm 10.1.3.1 identifier 1 version 4.0
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register confprof1
dspfarm profile 1 conference
description DO NOT MODIFY, active CCA conference profile - CCA2.0 codec729
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
associate application SCCP
dial-peer cor custom
name internal
name local
name local-plus
name international
name national
name national-plus
name emergency
name toll-free
dial-peer cor list call-internal
member internal
dial-peer cor list call-local
member local
dial-peer cor list call-local-plus
member local-plus
dial-peer cor list call-national
member national
dial-peer cor list call-national-plus
member national-plus
dial-peer cor list call-international
member international
dial-peer cor list call-emergency
member emergency
dial-peer cor list call-toll-free
member toll-free
dial-peer cor list user-internal
member internal
member emergency
dial-peer cor list user-local
member internal
member local
member emergency
member toll-free
dial-peer cor list user-local-plus
member internal
member local
member local-plus
member emergency
member toll-free
dial-peer cor list user-national
member internal
member local
member local-plus
member national
member emergency
member toll-free
dial-peer cor list user-national-plus
member internal
member local
member local-plus
member national
member national-plus
member emergency
member toll-free
dial-peer cor list user-international
member internal
member local
member local-plus
member international
member national
member national-plus
member emergency
member toll-free
dial-peer voice 1 pots
port 0/0/0
no sip-register
dial-peer voice 2 pots
port 0/0/1
no sip-register
dial-peer voice 3 pots
port 0/0/2
no sip-register
dial-peer voice 4 pots
port 0/0/3
no sip-register
dial-peer voice 5 pots
description ** MOH Port **
destination-pattern ABC
port 0/4/0
no sip-register
dial-peer voice 50 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/1/0
dial-peer voice 51 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/1/1
dial-peer voice 52 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/1/2
dial-peer voice 53 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/1/3
dial-peer voice 54 pots
description ** FXO pots dial-peer **
destination-pattern A0
port 0/1/0
no sip-register
dial-peer voice 55 pots
description ** FXO pots dial-peer **
destination-pattern A1
port 0/1/1
no sip-register
dial-peer voice 56 pots
description ** FXO pots dial-peer **
destination-pattern A2
port 0/1/2
no sip-register
dial-peer voice 2000 voip
description ** cue voicemail pilot number **
destination-pattern 388
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 6 pots
description "catch all dial peer for BRI/PRI"
translation-profile incoming nondialable
incoming called-number .%
direct-inward-dial
dial-peer voice 57 pots
description ** FXO pots dial-peer **
destination-pattern A3
port 0/1/3
no sip-register
dial-peer voice 69 pots
destination-pattern 329
port 0/0/0
dial-peer voice 300 pots
trunkgroup ALL_FX0
description Local Numbers
destination-pattern 9T
forward-digits 9
dial-peer voice 301 voip
destination-pattern 2..
session target ipv4:192.168.201.2
dial-peer voice 303 pots
trunkgroup ALL_FXO
trunkgroup ALL_FX0
description **InternationalCall**
destination-pattern 88T
dial-peer voice 304 pots
trunkgroup ALL_FX0
description *EM1*
destination-pattern 9[1-9]T
forward-digits 3
dial-peer voice 302 pots
trunkgroup ALL_FX0
description **Mobiles**
destination-pattern 9.[0-9].[0-9]......
dial-peer voice 305 pots
trunkgroup ALL_FX0
description **800-**
destination-pattern 9[0-9][0-9][0-9]T
no dial-peer outbound status-check pots
telephony-service
sdspfarm conference mute-on 111 mute-off 222
sdspfarm units 5
sdspfarm tag 1 confprof1
conference hardware
video
fxo hook-flash
max-ephones 40
max-dn 300
ip source-address 10.1.3.1 port 2000
max-redirect 20
auto assign 1 to 1 type bri
calling-number initiator
service phone videoCapability 1
service phone webAccess 0
service dnis overlay
service dnis dir-lookup
timeouts interdigit 5
system message American Center
url services http://10.1.10.1/voiceview/common/login.do
url authentication http://10.1.10.2/CCMCIP/authenticate.asp
load 521G-524G cp524g-8-1-17
load 525G spa525g-7-4-8
load 501G spa5x5-7-1-3c
load 502G spa5x5-7-1-3c
load 504G spa5x5-7-1-3c
load 508G spa5x5-7-1-3c
load 509G spa5x5-7-1-3c
time-zone 35
date-format dd-mm-yy
voicemail 388
max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
hunt-group logout HLog
moh MOH2.wav
multicast moh 239.10.16.16 port 2000
web admin system name cisco secret 5 $1$iDgA$MKNi2RWfsO0KjuC82kgLJ1
dn-webedit
time-webedit
transfer-system full-consult dss
transfer-pattern 9.T
transfer-pattern .T
secondary-dialtone 9
fac standard
create cnf-files version-stamp 7960 Aug 29 2012 12:00:04
line con 0
privilege level 15
logging synchronous
no modem enable
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport input all
line vty 0 4
exec-timeout 0 0
logging synchronous
login local
transport input all
line vty 5 100
login local
transport input all
ntp master
end
Some of the output are not shown becaus it is to long I have attach the whole config for reference and any advice on how could I optimize and resolve my issues is greatly appreciated. ThanksNicolo - First off this stuff gets crazy sometimes. No worries about the exam. Sometimes when FXO ports go crazy it is due to battery reversal. If you go to the FXO port settings try turning battery reversal on and or off... depending on its current setting. See if that helps.
As for the 525s not registering.. These are inside the network correct? Are you connecting one directly to the UC500 with a Cat5E or Cat6 patch cable and the same thing happens? Does the MAC address on the phone match a MAC address under the EPHONE settings?
If you telnet into the UC500 can you execute a "dir" command at the CLI prompt and "CD" (change directory) into the phones folder and then the spa525g folder? Do files exist in there?
Also I only see an IP address under BVI100? This is the voice side of things what happened to the IP address under BVI1 (Data VLAN). Can you give us some information about the internal network? Cna you PING this phone system from the network? What IP address does it have? -
Can i call multiple actions sequentially in a separate action/script?
i execute multiple actions on a single file. usually these actions number more than 20. is there a way to automate invoking of these actions so that i run something only once and all smaller actions are applied sequentially and automatically on the same file?
You can combine the commands in all of these Actions to a single Action,
but you can't call them automatically one after another if they're separate.
On Thu, Dec 11, 2014 at 10:22 AM, navpreet87 <[email protected]> -
My incoming calls are very quiet
Just got an iPhone and noticed that my incoming/outgoing calls are really quiet. My 'ringer' is turned up to the max, but don't where to look or what to change. Need help please.
Please correct me if I'm wrong, did you mean the analog phone connected on your SPA is not ringing when there's an incoming call ? If your answer is yes , try changing the analog phone connected to it to isolate the problem properly. In some cases , you will be needing to change the ring wave form on the SPA just like the one stated here
-
SPA122 FW upgraded to 1.2.1 (004) now all incoming calls are droped
I upgraded my SPA122 to Version 1.2.1 (004) and Now all my incoming (inbound) calls are dropped "Status: 480 Temporarily Not available"
Outbound calls are working Ok.
With the same config and with FW 1.1.0 (011) all was working Fine.
Is there any way to donwgrade to the previous "working" FW version ?I'm sorry, don't want be rude, but i can't post my config on a public forum.
I just want a "Working" way (Not the ones listed on admin guide) to Downgrade the FW to the previous "Working Good" version.
Thanks for your interest in my Problem.
Attached the "debug Log" from the SPA122 -
8330 Call Forwarding...How do I set the number of rings before calls are forwarded? Right now it instantly forwards my calls but I don't want that. I want to change it so that let's say that if after 5 rings I don't answer the call is forwarded. I've looked everywhere for an answer but have had no luck. Any help would be appreciated. I am with Virgin Mobile if this makes a difference.
Thank you
Message Edited by 8WDDdotcom on 08-01-2009 02:39 PM
Message Edited by 8WDDdotcom on 08-01-2009 02:44 PMWell, call forwarding is a feature provided by your carrier -- it is not a function of the device. However it works for your carrier is how it works. Actually, from what I understand about call forwarding, it normally is an all-or-nothing circumstance...calls forward to the number you've asked them to be forwarded to, keeping you in a DND status. Regardless, though, if your carrier tells you how it works on their system, there's nothing that the device can do to override that. Unless there is some kind of 3rd party app that would do it, bypassing (somehow) the carrier network CF feature...but I've never heard of such a thing.
Occam's Razor nearly always applies when troubleshooting technology issues!
If anyone has been helpful to you, please show your appreciation by clicking the button inside of their post. Please click here and read, along with the threads to which it links, for helpful information to guide you as you proceed. I always recommend that you treat your BlackBerry like any other computing device, including using a regular backup schedule...click here for an article with instructions.
Join our BBM Channels
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PIN: C0001B7B4 Display/Scan Bar Code
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PIN: C0005A9AA Display/Scan Bar Code -
RFC calls are taking too many work process -- system knock out
Hi,
I have an interface remote function, which is going to be called by a java program.
Data is going to be sent to the system, processed and sent back syncronous to the caller.
The calls are working on SAP tables, so I needed to lock and unlock the tables. The java program
is opening many parallell threads and starts them the same time.
That means, that the sap system is getting attacked by many rfc calls on the same function, with the
same user at the same time.
The system has in development 15 dialog work processes set up. If I run that with up to 15 parallell rfc calls, all is working ok!
Let me now explain the problem:
I have programmed the table lock, that it´s trying to lock the table at the beginning of the function.
If it´s not possible it tries it again. At first I have just inserted a one second wait and let that run maximum 3 times. If the table is still locked, it breaks down with an error.
But because the calls come nearly to exactly the same time to the system, the first call locks all others out, and the rest ran out of the 3 seconds and ended up with an error.
So I tried to build an endless loop, which is just ending, when the table could be locked again, or when the runtime of 5 seconds has reached. That means, it´s not waiting for a second, if the table is locked, but it´s trying again and again up to 5 seconds.
That is all ok, because sometime the table is free again and all calls are then processed alright.
BUT if I now run that with MORE than 15 rfc calls from the java program (15 dialog processes are set up), the system got knocked out! It is using all free work processes for the RFC calls and get stucked. I think, the first call is locking the table, and the got stopped by the system. And then all the others are running in the endless loop....
So, shouldn´t be there a limit for dialog processes taken by RFC calls? For example here for this user? In this case it should f.e. be set to 10 free processes with 5, which are reserved for other users...
Any advices?
The second is, how can I change the lock mechanism, that I don´t use an endless loop? Waiting for one second and doing that 3 times, is not good enough... Because all the rfc calls wait then for one second, up to 3 times, and end with an error....
Is it possible to wait for half seconds, or defined milli seconds?
Thank you for all advices!Hi,
What I have understood is you are trying to lock the table in a loop and just checking the duration by explicitly putting wait statement.
Hope you are NOT locking the entire table rather a particular record in each lock.
No need to use explicit wait statement rather its better to let SAP thinks when to lock the table T1 when its already locked by another work process. So just set the wait parameter for the Enqueue function module as 'X' and check how it behaves. Until the table is not unlocked the process will wait.
If you see the wait parameter for the Enqueue function module is not available then create your own lock object for the table and use that. Make sure its getting dequeued else you can understand better
Cheers
Somnath -
My previous calls are not displaying in the recent call section. How can I fix this?
There are some recovery programs out there if you google them - many with free trials. I haven't tried them so I have no opinion.Read any professional tech and user reviews you can find before you try one. I have read here in the past that some have worked for people who have posted here.
To help in the future, have you considered using Jump Desktop - Remote Desktop? I know it won't solve the problem you're dealing with now (unless notes were taken on your computer.) It's in the app store. It's the #1 paid app in the business category and the reviews are great. It allows you to see anything that's on your computer on your ipad. If you take your notes on your ipad then you might want to email them to yourself, put them on Word in your computer, for example, and then if you need them in class, you remotely connect your ipad to your computer and you will see your notes. This way you won't have to rely so much on the cloud. The explanation would be too long here, but you might want to take a look at it. You can use the free trial on jumpdesk.com and then if it works for you, buy it in the app store.
Meanwhile, it's hours since you sent this post. Hopefully others who can help solve your problem who are on this forum now can help you. I hope you get your notes back. Don't give up too quickly.
Hope this helps. -
What are the process are done in compile time?
what are the process are done in compile time?
if this process occur in runtime that's called
runtime polymorphismExactly. That's usually the case, especially with interfaces. I'm not entirely sure, but I think if you're calling final methods, those are statically linked because they can't be overridden anyway. I guess it saves a method table lookup at runtime.
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