The Trunk Provider says it is impossible. How ever , the complex solution you talk about :
It is the last part of this Discussion ?
Edit : OK it was the right Thinking. I will Try it next Week. yes @ the moment it is a test but in two weeks it goes life :-)
The really critical parts working with the frist sip provider. But i have to understand SIP Profiles to quick react with Problems of the Trunk of the German Telekom. This provider is not testable for me until it goes life.
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Thanks for the hint,
SIP is with the Devil ;-)
I Testet the sip Profile on inbound. Nothing happend. NO Change in the Caller ID Name.
" But pay attention to dial-peer, for my experience you can apply a sip profile to outgoing dial-peer. But you need the translation on incoming traffic." That hit that nothing happen.
What Feature we are talking about ? XML Scripting?
When i change the mode from CME to Boarder Element will the Rest of my config work ? The CME Part i mean.
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Can this part to Modify so that i got set there a String in it ?
Like "Call for Company X".
I would place this Header Manipulation per Dial-Peer not on Global Level , in the End i will have 3 SIP Trunks Inbound.
request INVITE sip-header From modify ".*(<sip:.*@.*>)" "For Company 1 " ?
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it works but in the Phonedisplay stands +49myHandynumber 70myhandynumber. I Can dial back but the Display Missed calls info is strange.
It seems that are to Points are Displayed. From what info get the Phone the +49myhandynumber.
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i am using Cisco 2911 with IOS 15.7
I have the Problem that i need the "From" Part of an incoming SIP Trunk. The Provider is providing the number with international Prefixes. I have to strip "+49" and Make from it "70" . SO the calling number is displayed as 70XXXXXXXX instead of +49xxxxxx for example.
First i tried to using Translation Profiles to Strip +49 but is not working. So i think i need to change it in the sip header. But as Cisco Docs say is the is only usable on outbound Dial peers. I Test request from invite header to manipulate SIP header but i think i have a general understanding problem of the headers.
Provider--> SIP-Trunk -->SIP-Profile --> Dial-peer -- Translation Profile -- > Ephone-dn Is that correct how it works ?
when i know make
request any sip-header from modify "" "" in the sip profile i have 2 . Problems:
1. Match Pattern
2. Is this the right way ?
! ! No configuration change since last restart ! version 15.7 service timestamps debug datetime msec service timestamps log datetime msec service password-encryption ! hostname PT2911 ! boot-start-marker boot-end-marker ! ! ! no aaa new-model clock timezone utc 1 0 clock summer-time utc recurring last Sun Mar 2:00 last Sun Oct 3:00 clock calendar-valid ! ! ! ! ! ! ! ! ! ip dhcp excluded-address 192.168.3.1 192.168.3.10 ! ip dhcp pool Phone network 192.168.3.0 255.255.255.0 default-router 192.168.3.1 option 150 ip 192.168.3.1 option 66 ip 192.168.3.1 option 42 ip 192.168.3.1 option 120 ip 192.168.3.1 ! ! ! ip domain name "my Domain" ip name-server "my Name Server" ip cef no ipv6 cef multilink bundle-name authenticated ! ! ! ! ! ! ! voice-card 0 ! ! ! voice service voip ip address trusted list ipv4 220.127.116.11 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip no supplementary-service sip handle-replaces sip subscription maximum accept 100 subscription maximum originate 100 registrar server expires max 600 min 60 ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g711alaw codec preference 3 g729br8 codec preference 4 g729r8 codec preference 5 g723r63 codec preference 6 g723ar63 ! ! voice class sip-profiles 1 ! voice class sip-profiles 2 request INVITE sip-header From modify "firstname.lastname@example.org" "Username@fpbx.de" ! ! #Tenant Only for Test after SIp-Profiles Working voice class tenant 1 registrar 1 dns:sip-trunk.telekom.de expires 3600 credentials username Test1 password 7 073B245F5A58 realm sip-trunk.telekom.de authentication username test1 password 7 051F031C351D realm testrealm1 timers buffer-invite 10000 sip-server dns:reg.sip-trunk.telekom.de no pass-thru content custom-sdp
#active Testing in Progress Tenant 2 ! voice class tenant 2 registrar 1 dns:fpbx.de expires 3600 credentials username Username password 7 Password realm fpbx.de authentication username Username password 7 Password realm fpbx.de remote-party-id timers buffer-invite 10000 sip-server dns:fpbx.de host-registrar no pass-thru content custom-sdp ! ! voice register global mode cme source-address 172.20.0.20 port 5060 max-dn 30 max-pool 30 authenticate register timezone 28 time-format 24 date-format D/M/Y auto-register ! ! ! voice translation-rule 2 rule 1 /hidden/ /993/ ! voice translation-rule 5 rule 1 /993/ /hidden/ ! voice translation-rule 154 rule 1 /^7/ // ! ! voice translation-profile Test translate called 2 ! voice translation-profile Test2 translate called 154 ! ! ! vxml logging-tag license udi pid CISCO2911/K9 sn FCZ164961HD license boot suite AdvUCSuiteK9 ! ! file privilege 0 username admin privilege 15 secret 5 $1$/gBO$WmNKEpDpk//ue0tAgJY1x0 ! redundancy ! ! ! translation-rule 2 ! ! ! ! ! interface Embedded-Service-Engine0/0 no ip address shutdown ! interface GigabitEthernet0/0 no ip address duplex auto speed auto ! interface GigabitEthernet0/0.3 encapsulation dot1Q 3 ip address 192.168.3.1 255.255.255.0 ! interface GigabitEthernet0/0.20 encapsulation dot1Q 20 ip address 172.20.0.15 255.255.0.0 ! interface GigabitEthernet0/1 no ip address shutdown duplex auto speed auto ! interface GigabitEthernet0/2 no ip address shutdown duplex auto speed auto ! ip forward-protocol nd ! ip http server ip http authentication local no ip http secure-server ip http path flash0:/CME-GUI-12.2 ! ip route 0.0.0.0 0.0.0.0 172.20.0.1 ! ! ! ! control-plane ! ! ! ! ! ! mgcp behavior rsip-range tgcp-only mgcp behavior comedia-role none mgcp behavior comedia-check-media-src disable mgcp behavior comedia-sdp-force disable ! mgcp profile default ! ! ! ! dial-peer voice 1 voip description Telekom inbound voice-class sip tenant 1 ! dial-peer voice 2 voip description Inbound-Placetel Test translation-profile incoming Test session protocol sipv2 session target dns:fpbx.de incoming called-number hidden voice-class sip profiles 1 inbound voice-class sip tenant 2 dtmf-relay rtp-nte codec g711ulaw ! dial-peer voice 4 voip description Outbound-Placetel Test translation-profile outgoing Test2 destination-pattern 7.T session protocol sipv2 session target dns:fpbx.de voice-class sip profiles 2 voice-class sip tenant 2 codec g711alaw ! ! sip-ua no remote-party-id presence enable ! ! ! gatekeeper shutdown ! ! telephony-service max-ephones 58 max-dn 300 ip source-address 192.168.3.1 port 2000 max-redirect 20 caller-id block code *# calling-number initiator service phone ehookEnable 1 service phone webAccess 0 service dnis overlay timeouts interdigit 2 system message Test cnf-file location flash: cnf-file perphone network-locale DE time-zone 28 time-format 24 date-format dd-mm-yy max-conferences 8 gain 0 web admin system name admin secret 5 $1$2bp3$GzxOzTCdM5JtzFK1eBvD.. dn-webedit transfer-system full-consult transfer-pattern .T transfer-pattern 0.T transfer-pattern 9T log table retain-timer 30 log table max-size 100 secondary-dialtone 0 directory last-name-first ! ! ephone-dn 3 number 993 description test ! ! ! ephone 13 description TEST mac-address 001D.A21A.4635 type 7960 button 1:3 ! ! ! vstack ! line con 0 logging synchronous login local line aux 0 line 2 no activation-character no exec transport preferred none transport output pad telnet rlogin lapb-ta mop udptn v120 ssh stopbits 1 line vty 0 4 login local transport input ssh ! scheduler allocate 20000 1000 ntp server de.net.pool.org ntp server de.pool.ntp.org ! end
With voice-translation rule 2 i change the called number to match the ephone-dn 3 Number. It works, calls can go out and come in but the displayed calling number can´t be called because of wrong format. To select the correct outbound dial-peer there is an 7 need to select the right Sip Provider.
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System : CISCO2911
IOS Version: c2900-universalk9-mz.SPA.157-3.M3.bin
USE : CME
I configured a sip Inbound Dial peer with Multi tenant.
voice class tenant 2 registrar 1 dns:fpbx.de expires 3600 credentials username xxxxx password 7 xxxxx realm fpbx.de authentication username xxxxx password 7 xxxx realm fpbx.de no remote-party-id timers buffer-invite 10000 sip-server dns:fpbx.de host-registrar no pass-thru content custom-sdp
dial-peer voice 2 voip translation-profile incoming Test session protocol sipv2 session target dns:fpbx.de incoming called-number xxxxxxxxxxxx voice-class sip profiles 1 voice-class sip tenant 2 dtmf-relay rtp-nte codec g711ulaw
i use an Voice Translation Profile to change the called number from the Trunknumber to an internal Number of an ephone-dn.
Like : xxxx to xx so the System can direct select the right user.
The Call works in and outbound.
But when i call from external i see in the Phone display the calling Number twice.
From "External Number" to (External Number).
With a translation Profile for the calling Number i can remove the "To" Part of the Display. That seems very Strange because i manipulating the calling Number not the called Number.
My Traget : the Number is displayed "+49" but i need "00049" oder "00"
What i am doing wrong ?
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sorry for my late respond. The 7.82 ASDM no log Bug hits me and i have to wait for a Downtime to get logs :-(.
I am using Objekt Nat.
i have now defined rules for every port. But now i have the Problem that a one to one port mapping not works.
The First inside smtp server in the the dmz goes out the to outside ip one
The Secound the same. I see the Problem that the match criteria not work.
The Source Ports are randomized so the first nat rule for smtp hits becaus of the Destination Port. i need a better match criteria.
The Problem is that I have to nat on the same outside ip an FTP Server.That the quickoption Public Server in the asdm is no option.
All DMZ server are configured through Objekt Nat. I need to nail two Server on this outside ip with @ least min 5 Ports.
Outside IP ONE
SMTP Server one with only 25 for inter server connection.
Outside IP Two
SMTP 2 (DMZ IP 2): 143,587, 25
The 587 and 143 is always triggerd from the outide so there is no Problem but when the client need to send a Mail the mail goes out @ IP One.
FTP (DMZ IP 3): (ftp, ftp-data)
Inbound work well but outbound not.So i think i missed something.
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i know this Problems are not new but somehow i don´t get it.
I have an ASA5510 with One Physical Interface that hold 5 IP´s. I have until 5 Server in the DMZ.
Now i need another one but it is the secound smtp Server so i need on the same Physical Interface 2 Forwards for smtp. Two different inside hosts( Mailserver).
The first one is mapped to the outside interface with ip x.x.x.x1
Now i want to map the ip x.x.x2 to the second one.
The ASDM is complaining of in use.
I read in this forum about uniqie Identifier that the asa could match the translation for Port 25.
I have two unique the ip´s or is that impossible to match for the asa ?
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So the imbound works fine, but now my sip provider is rejecting the Call outbound.
The From Header of the sip message is wrong.
The Provider need : email@example.com
but the system is using : username@cmeipaddress
Please Help a secound time.
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2 x PVDM2-16
1 x VIC2-2BRI-NT/TE
IOS Software: c2800nm-ipvoicek9-mz.151-4.M10.bin
We have a TSF-DIALOG 201 from Auerswald. An Analog a/b Doorphone.
Info: Technical Data
The DTMF Tones work fine so all features that can be programmed with no onhook after programming working.
But the elemetray feature of the called Number has to be programmed with onhook after programming.
So the Doorphone dials to fast and make no break after onhook and dials the preprogrammed number "31". But the CME makes the Discission to redirect it to another number.
But i cannot programm the System because the busytone when the remote Phone goes onhook is to fast and the System do not store the programming Information.
voice-port 0/3/0 cptone DE
timeouts ringing inifinity caller-id enable
port 0/3/0 fallback-dn 88
dial-peer voice 999 pots no tone dialtone remote-onhook tone busytone remote-onhook service stcapp port 0/3/0
number 88 label Eingang name Eingang
ephone 1 mac-address BD42.A1D1.1180 max-calls-per-button 2 type anl button 1:88
The Target Number is a Huntgroup:
ephone-hunt 88 sequential pilot 89 list 22, 13 timeout 6, 6 description Tuersprechanlage.
Can someone Help me ?
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HI all, we have two CGS-2520-16S-8PC with links in different directions. All SFP Ports are UP but only the half of it forwarding Traffic. We need all Ports . In one of the Cisco documents I found this hind : "The100BASE-FX SFP ports and the 10/100 PoE ports are grouped in pairs. The first member of the pair (port 1) is above the second member (port 2) on the left. Port 3 is above port 4, and so on. The dual-purpose ports are numbered 1 and 2. " But more about it is not found in the documemnts. So i have no clue how to bring all Ports to forward traffic. Have some one an idea ? Philipp
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