required VoIP Dialer?

I tried 3 different VoiP providers, and I can't get an any of them to work with Outlook 2007. The problem must be on my end.

Currently I have Vonage. Still no luck. I read somewhere that Dialer Outlook 2007 is not configured correctly for VoiP if if I go to control panel-> phone and modem-> advanced, I have the following options:

Microsoft HID phone TSP
Proxy NDIS TAPI service provider
TAPI Kernel-Mode service provider
Unimodem 5 service provider

I also have the ability to add Microsoft Windows remote service provider. AnyOf these work or I need to install/download a certain dialer?

Thank you!

Outlook has no dialer. The Dialer is part of the victory, and requires a standard telephone line.

Most voip providers have their own Dialer

for example http://www.vonage.com/features_available_options.php?feature=softphone

Tags: Windows

Similar Questions

DX 80 - dial PSTN call does not

We have setup in the video below. But in recorded in CUCM 10.5 x SX80 try calling PSTN numbers, does not.

Hit gateways voice with error, error of unavailable resource. But other devices work properly with the same number of rank. Help, please

Hello

Can you put the codec g729r8 on your existing voice class codec and affect your pointing to your (not only G711ulaw) CUCM voip dial-Exchange.

Also configure transcoding on router resources and register with CUCM.

In my view, that it is somewhat related to DSP.

Kind regards

Raclot.

CVP 4.0 (2) bootstrap.vxml

Hello

I have a CVP 3.1 working system that I'm migrating to 4.0 (2) CVP.

I upgraded my IOS 12.4 GW/GA 2811 (6) T and applied the config which was OK in 3.1 CVP. I downloaded the new .tcl and .vxml files of the distribution and reloaded the router so that they are active with the new code.

The leg of the switch worked well. If I changed my script ICM to return a label on the customer SVC routing to one of my phones, the phone rang.

If I inserted a node 'Send to URV' he fell from the call, and I could see the trace of the bridge (debug on vxml application error) he tried to call my server call with undefined://undefined: not defined/cvp/VBservlet.

I managed to make it work by hacking the bootstrap.vxml to set what must have been completed in the above URL:

PROTOCOL

HOST

PORT

make the script look a bit like the bootstrap.vxml on CVP 3.1. Here are my changes:

ISN-vxml is defined by the ip host table.

I had to set the DNIS differently as above - I could see that the error in the CallServer DNIS was a required argument.

My question is this.

I guess the CallServer is supposed to be passing values such as PORT, PROTOCOL etc. in the file bootstrap.vxml when he calls it, but mine is obviously not and I don't know what I should have done to make this work.

Things are definitely different with 4 CVP. ;-)

Kind regards

Geoff

Hi Geoff,

These get variables defined in bootstrap.tcl. It seems to me that bootstrap.vxml have called directly instead of via bootstrap.tcl. Date of arrival:

1. that the inbound voip dial-peer on the VRU leg calls the service attached to bootstrap.tcl.

2 set "debug script voip application" and "debug voip application tcl" and make sure you see bootstrap.tcl have invoked. Him debugs should show a procedure of transfer of bootstrap.vxml with all these settings.

-janet

Error 623 - system could not find the directory entry for this connection

This error box 623 often appears when I'm online and it's boring, what the hell. It is a "network connection error", and it says "cannot load the dialogue."

I had tried a "fix" that I found on the microsoft site using microsoft, but it was not useful. He said that my computer tries to dial a number for the VPN connection. But I don't have a VPN connection and none are listed under my network connections.

Help, please!

Hi Crudblue.

There is a KB detailing this problem and the workaround. http://support.Microsoft.com/default.aspx/KB/320693

To work around this problem:
1. Click Start, click Control Panel, and then click on network connections and Internet.
2. Click on network connections.
3. Right-click the VPN connection, and then click Properties.
4. Click to select the checkbox Dial another connection first and then click on the appropriate connection. If there is no dial-up connection to select, you will need to create a dial-up connection and repeat these steps.
  If your VPN connection does not require a dial-up connection, click to clear the checkbox Dial another connection first , and then click OK.
I hope this helps! If you have any questions please come back!

Shawn - Support Engineer - MCP, MCDST
Microsoft Answers Support Engineer
Visit our Microsoft answers feedback Forum and let us know what you think

help with SPA 3102 (question graphcal)

HI guys here is my situation (I draw so it would be easier for future reference):

I want to pick up A phone and dial the Ext 101 101, 102 for Ext 102 and so on.
also, I want to put routed my call for location 2 location 1 for example transferring calls to 104.

any help would be really appreciated even a starting point as for example the configuration of spa to route my call not NAT, so it can connect to the other.

castro69 wrote:
also, I want to put routed my call for location 2 location 1 for example transferring calls to 104.
any help would be really appreciated even a starting point as for example the configuration of spa to route my call not NAT, so it can connect to the other.

I guess that's an analog PBX, otherwise you wouldn't need the SPA3102 through the internet.

For communications between the SPA3102, I would use direct ip call, using the external ip address and the sip port numbers. Think that the SPA3102 is two separate cards inside the box where treat you everyone with its sip port number added to the external ip address common.

I would setup port sip distinctive numbers on each of the baths to keep things straight. You have a number of separate port for tabs of line 1 and line PSTN. You will need to send these to the SPA3102 adapter port numbers in their respective routers or firewall of the router will reject incoming packets on the internet. I would also convey the port range rtp for voice, flow packs.
On each tab of the line 1 and RTC, you define NAT Mapping Enable: YES, not record, make call without Reg Yes, years call without Reg No. I put your external ip address on the Sip tab under EXT IP. This will tell the SPA3102 to use this address in the sip signaling. I assume you are using static external ip addresses. On each tab of the line 1 you would activate IP Dial Yes.

The analog PBX is connected to the FXO port on one of the Spa. You should check the voltage level hung up and won and then set the line parameter usage on the RTC of the SPA line tab to halfway between the two readings. You can read the levels of tension on the PSTN line tab. Calls to the PBX of the PSTN line tab will go through the voip to PSTN gateway. I set up the catwalk with http authentication and configure a user name and password.

Details are starting to become quite complicated. I'd get running through steps. Get a job step before moving on to the next step.

The 1st step would be to get A phone call/receive calls to a PBX. You can configure the line 1 for FXS phone attached A to use port location PSTN 2 as the proxy using http authentication, and you can then dial the extensions you want to call. Location 1 SPA3102 will send a guest of the sip Protocol to the tab location 2 SPA3102 from pstn line and the SPA3102 will dial the number on the FXO port to the PBX.

For calls coming from the other direction of a PBX to slot 2 SPA3102 the only place where you can connect a voip call is in the SPA3102 numbering plan. If you want to call only phone that is easy, install you just dialers-messengers automatic telephone in the pstn-to-voip dial plan.

I'm not clear about what you want to do with phone B I take is Extension 104.

I like your designs. Can save a lot of words.

Problem with GCE international calls

Hi all

I have my router CME with two different link:

the first is the VOIP and the second is POTS

I need to allow international outgoing calls be routed via the link of POTS

So I made these commands:

voice pots Dial-peer 5
corlist outgoing international call
International call through TT description
preference 1
destination-model 000 t
port 0/0/0:15

I also configured the international calls through SIP TRunk by using these commands

Dial-peer voice voip 2200
corlist outgoing international call
Description * International Outgoing Call to SIP Trunk *.
translation-profile outgoing PSTN_Outgoing
destination-model 000 t
session protocol sipv2
session target ipv4:192.168.2.51
codec voice-class 1
voice-class sip dtmf-relay rtp - nte force
DTMF-relay rtp - nte
No vad

The problem is that calls always goes in the SIP trunk while I've specified the upper value of the preference for POTS

Y at - it all missconfiguration

Please help me it is urgent!

Add "preference 2" under the VoIP dial peer. Please note that when you set the order of preference, the lower part, number preferably, more priority. Absolute priority is given to the counterpart of dial with preference order 0 and it is the default value.

Manish

-Does the rate of useful messages

Keypad to make a call

Hello

I'm working on voip Dialer app. In the present, I have to design a dial pad.

I want to ask questions on the keypad of the phone.

Is it possible to use only digital keyboard in my application. I try with invoke phone. But I want to use only the dial pad.

Is this possible?

Or any other means for keypad?

Thanks for the help...

In advance...

If I understand your question, you ask if it is possible to use the same keyboard Dialer in your application, as is used by the phone application?

The answer is that this dialer control is not, as far as I'm aware, available through the API, so is not available for third-party developers.

Sorry, you will need to create one yourself. I see you have another Thread talking are contiguous to try to do that, so I guess you reached that conclusion yourself.

Need serious help. New 8831 phones problems.

Thank you in advance for helping with this.

I pulled out recently of new 8831 conference phones to replace my old phones 7937 spider, unfortunately phones know terrible incoming audio. Phones are on a branch site, connected via MPLS and registered at 8.6 CUCM to Headquarters. We have a total of 900 7940 phones that have no problems in any of the different location and band bandwidth/congestion is not a problem. The audio input signal poor that happens to every call. Audio output is fine. 8831 phones are having the same issues at several sites of the branch, but seems more predominant in some. The question has been described as 'audio static, choppy, distorted, with a few breaks.

I tested with G.729 and G.711 without some luck, after using the web interface to enter stats here, this is what they look like:

Remote address
Local address
10:33:26 departure time has
Flow in Active status
Host name
Sender packages 7180
Sender 143600 bytes
The sender Codec G.729
Reports of the sender sent 0
Sender report time sent to 00:00:00
RCVR 2215 of lost packets
Jitter AVG 20
RCVR Codec G.729
RCVR sent reports 0
RCVR report time sent to 00:00:00
RCVR packages 7178
RCVR 143560 bytes
MOS LQK 2.8657
AVG MOS LQK 2.3054
Min MOS LQK 2.0000
Max MOS LQK 3.0295
MOS LQK Version 0.95
Cumulative conceal 0.1356 Ratio
Interval to conceal Ratio 0,0560
Max conceal 0.2766 Ratio
Hide the dry 140
Severely hide dry 120
Latency 40
Jitter of Max 160
Sender, size 20 ms
The sender of the received reports 0
Report to sender time received at 00:00:00
RCVR size 20 ms
1266 thrown RCVR
RCVR reports received 0
RCVR report time to 00:00:00

As you can see that the MOS is pretty bad, jitter and ignored packets seem to be bad too. Intrasite tests have resulted in better looking for numbers. Initially, I immediately thought "QoS" but all other phones are fine and the problem is with audio INCOMING do not outgoing.

All comments on this are appreciated.

Thank you.

Hello

As you can see receiver lost packets number is very high, which will cause audio choppy. Looks like you face below default

https://Tools.Cisco.com/bugsearch/bug/CSCuq03746

Also can ensure you codec used is g729r8 not g729br8. g729br8 has integrated VAD have so it will drop some packets. Also disable VAD in any voip dial peer by using the command "no vad.

Kind regards

Mohit Singh

Call Cisco UBE is not received endpoint

Hello

I am tryng to install what UBE between two GUY, the idea is that R1 will send the call to the CUBE that will send the call to R2 (following a pattern). Earlier today, when the call is sent to R1 call is received for CUBE but resemble the call is not to send to R2, even when leaving to R2 dial-peer is matched. I installed a phone in CUBE and I can can appeal to R2 using the same outgoing dial peer match R2, so I don't know why CUBE isn't a R1 to R2 call diversion

Schema:

H323 H323

R1 ----------------> CUBE -----------------> R2

DN = 2... DN = 3...

Debugging

======

Call is received of R2 and outgoing dial-Exchange R2 files is put in correspondence, but call for R2

GK #debug voip dialpeer

VoIP dialpeer default debug is on

* Jun 29 12:56:00.927: //-1/C7785C1D8183/DPM/dpAssociateIncomingPeerCore:

Number = 2003, called number = 3001, Voice-Interface = 0 x 0.

Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,

Peer Type Info = DIALPEER_INFO_SPEECH

* Jun 29 12:56:00.927: //-1/C7785C1D8183/DPM/dpAssociateIncomingPeerCore:

Result = Success (0) after DP_MATCH_INCOMING_DNIS; Incoming dial-peer = 1999

* Jun 29 12:56:00.927: //-1/C7785C1D8183/DPM/dpMatchSafModulePlugin:

dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0

* Jun 29 12:56:00.931: //-1/C7785C1D8183/DPM/dpAssociateIncomingPeerCore:

Number = 2003, called number = 3001, Voice-Interface = 0 x 0.

Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,

Peer Type Info = DIALPEER_INFO_SPEECH

* Jun 29 12:56:00.931: //-1/C7785C1D8183/DPM/dpAssociateIncomingPeerCore:

Result = Success (0) after DP_MATCH_INCOMING_DNIS; Incoming dial-peer = 1999

* Jun 29 12:56:00.931: //-1/C7785C1D8183/DPM/dpMatchSafModulePlugin:

dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0

* Jun 29 12:56:00.931: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Number =, called number = 3001, Peer Info Type = DIALPEER_INFO_SPEECH

* Jun 29 12:56:00.935: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Associate the rule of = DP_MATCH_DEST; Called number = 3001

* Jun 29 12:56:00.935: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Result = Success (0) after DP_MATCH_DEST

* Jun 29 12:56:00.935: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

dialstring = 3001, saf_enabled = 0, saf_dndb_lookup = 1, dp_result = 0

* Jun 29 12:56:00.935: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

Result = Success (0)

The outgoing dial matched host list:

1: Tag dial-peer = 2000

CUBE see the race

============

GK #show run
Building configuration...

Current configuration: 2628 bytes
!
! Last modification of the configuration at 12:31:19 UTC Friday, June 29, 2012
version 15.1
horodateurs service debug datetime msec
Log service timestamps datetime msec
no password encryption service
!
hostname GK
!
boot-start-marker
start the flash c2800nm-adventerprisek9_ivs_li - mz.151 - 4.M3.bin system
boot-end-marker
!
!
enable secret 5 $1$ szeU$ enWyM69bTodk0Aiwyxw5R.
!
No aaa new-model
!
!
dot11 syslog
IP source-route
!
!
IP cef
!
!
!
no ip domain search
No ipv6 cef
!
Authenticated MultiLink bundle-name Panel
!
!
!
!
!
!
!
voip phone service
h323 connections allow h323
allow connections h323 to SIP
allow connections sip h323
allow sip to sip connections
H323
SIP
!
!
!
!
!
voice-card 0
dspfarm
DSP services dspfarm
!
Crypto pki token removal timeout default 0
!
!
!
!
license udi pid CISCO2811 sn FHK1225F3KT
!
redundancy
!
!
controller LAN 0/0/0
!
!
!
!
!
!
!
!
!
interface FastEthernet0/0
192.168.4.1 IP address 255.255.255.0
automatic duplex
automatic speed
!
interface FastEthernet0/1
no ip address
Shutdown
automatic duplex
automatic speed
!
interface Serial0/2/0
no ip address
Shutdown
no fair queue
2000000 clock frequency
!
interface Serial0/2/1
no ip address
Shutdown
2000000 clock frequency
!
!
Router eigrp 1
192.168.4.0 network
!
IP forward-Protocol ND
no ip address of the http server
no ip http secure server
!
!
!
NLS RESP-timeout 1
CPD cr id 1
!
!
!
!
!
!
control plan
!
!
Voice-port 1/0/0
!
Voice-port 1/0/1
!
!
!
profile MGCP default
!
SCCP local FastEthernet0/0
SCCP ccm 192.168.4.1 ID 1 version4.0
SCCP
!
SCCP ccm Group 1
associate the ccm 1 priority 1
the associated profile 1 registry GK-VOUGEOT
!
transcode dspfarm profile 1
Codec g729r8
Codec g729br8
Codec g729ar8
Codec g729abr8
Codec g711alaw
Codec g711ulaw
maximum sessions 3
associate the PCRS application
!
Dial-peer voice 2000 voip
destination-model 3...
session target ipv4:192.168.3.2
Transparent codec
!
voice pots Dial-peer 5001
destination-model 5001
port 1/0/0
!
Dial-peer voice voip 1999
session target ipv4:192.168.3.1
incoming called-number 3...
!
!
!
!
access controller
area local zone1 micasa.com 192.168.4.1 intrazone enable
area local micasa.com enable-intrazone zone2
zone prefix zone1 2... GW-priority 10 CME - Lab
zone prefix zone2 3... GW-priority 10 CME-Legacy
GW-type-prefix 1 #* by default-technology
no downtime
!
!
phone service
5 units of sdspfarm
sdspfarm tag 1 GK-VOUGEOT
Max-joined 4
Max - dn 4
IP source-address 192.168.4.1 port 2000
MAX conferences 8-6 win
transfer full-consult system
!
!
!
Line con 0
password leoleo
opening of session
line to 0
line vty 0 4
password leoleo
opening of session
transport of entry all
!
Scheduler allocate 20000 1000
end

GK #.

R1 show race

=========

CME - Lab #show run
Building configuration...

Current configuration: 7806 bytes
!
! Last modification of the configuration at 08:23:06 LAPAZ Friday, June 29, 2012
! NVRAM config update at 13:13:31 LAPAZ Thursday, June 28, 2012
!
version 15.1
horodateurs service debug datetime msec
Log service timestamps datetime msec
encryption password service
!
CME - Lab host name
!
boot-start-marker
start the flash c2800nm-adventerprisek9 - mz.151 - 3.T3.bin system
boot-end-marker
!
!
type 0 2 t1 card
enable secret 5 $1$jJ.7$b/Yaq0M36fgZSrYpCK5LD/
!
No aaa new-model
!
clock timezone LAPAZ-4 0
network-clock-participate wic 2
!
dot11 syslog
IP source-route
!
!
IP cef
!
DHCP excluded-address 192.168.1.1 IP 192.168.1.20
DHCP excluded-address IP 192.168.2.1 192.168.2.20
DHCP excluded-address IP 192.168.3.1 192.168.3.20
DHCP excluded-address IP 192.168.6.1 192.168.6.20
!
IP dhcp DATA pool
network 192.168.1.0 255.255.255.0
option 150 ip 192.168.2.1
default router 192.168.1.1
!
pool IP dhcp VOZ - CME
network 192.168.2.0 255.255.255.0
option 150 ip 192.168.2.1
default router 192.168.2.1
!
VOZ-CUCM dhcp IP pool
network 192.168.6.0 255.255.255.0
option 150 ip 192.168.6.10
router by default - 192.168.6.1
!
!
no ip domain search
No ipv6 cef
!
Authenticated MultiLink bundle-name Panel
!
!
!
!
primary-qsig ISDN switch type
!
!
!
voip phone service
list of approved IP addresses
IPv4 192.168.3.2
SIP
!
voice class codec 1
g711ulaw codec preference 1
codec preference 2 g729r8
g729br8 preferably 3 codec
g711alaw preferably 4 codec
!
!
!
!
translation of the voice-rule 1
rule 1 3... / / 1809544\0 /.
!
voice translation-rule 100
rule 1 /2/ /1002/
!
voice translation-rule 101
rule 1 /1002/ /2/
!
!
voice translation-profile SITE-CODE-CallerID
definition of 100 calls
!
SITE-CODE-DNIS voice translation-profile
translate called 101
!
!
voice-card 0
dspfarm
DSP services dspfarm
!
Crypto pki token removal timeout default 0
!
!
!
!
license udi pid CISCO2811 sn FHK1225F3GX
Archives
The config log
hidekeys
!
redundancy
!
!
controller T1 2/0/0
the clock source internal
long CableLength 0dB
time intervals PRI - Group 1-2: 24
!
!
class-map correspondence-any WEB browsing
http protocol game
secure http protocol game
class-map matching VOICE
match Protocol rtp
!
!
OUTBOUND-POLICY policy-map
!
!
!
!
!
!
!
!
!
interface FastEthernet0/0
no ip address
automatic duplex
automatic speed
!
interface FastEthernet0/0.10
encapsulation dot1Q 10
IP 192.168.2.1 255.255.255.0
H323-gateway voip interface
H323-gateway voip id zone1 ipaddr 192.168.4.1 1719
H323-gateway voip h323-id CME - Lab
port of link voip H323-gateway 192.168.2.1
!
interface FastEthernet0/0.11
encapsulation dot1Q 11
192.168.6.1 IP address 255.255.255.0
!
interface FastEthernet0/0.50
encapsulation dot1Q 50
IP 192.168.1.1 255.255.255.0
!
interface FastEthernet0/1
no ip address
Shutdown
automatic duplex
automatic speed
!
interface Serial0/0/0
address 192.168.3.1 IP 255.255.255.0
encapsulation ppp
64000 clock frequency
!
interface Serial0/0/1
no ip address
Shutdown
2000000 clock frequency
!
interface Serial0/2/0:23
no ip address
encapsulation hdlc
primary-qsig ISDN switch type
ISDN timer T310 120000
ISDN-emulate network protocol
ISDN incoming-voice
No cdp enable
!
!
Router eigrp 1
network 192.168.1.0
network 192.168.2.0
192.168.3.0 network
network 192.168.6.0
Auto-resume
!
IP forward-Protocol ND
no ip address of the http server
no ip http secure server
!
!
!
exploitation forest esm config
!
!
!
!
!
flash:/PHONE/6921_6941_6961/Sccp/cmterm-69xx-sccp.9-2-1-0.tar cmterm-69xx-sccp.9-2-1-0.tar alias TFTP server
alias of server TFTP flash:/PHONE/6921_6941_6961/Sccp/BOOT69xx.0-0-0-14.zz.sgn BOOT69xx.0 - 0-0 - 14.zz.sgn
alias of server TFTP flash:/PHONE/6921_6941_6961/Sccp/SP69xx.0-0-0-8.zz.sgn SP69xx.0 - 0-0 - 8.zz.sgn
alias of flash:/PHONE/6921_6941_6961/Sccp/SCCP69xx.9-2-1-0.loads SCCP69xx.9 - 2-1 TFTP server - 0.loads
alias of flash:/PHONE/6921_6941_6961/Sccp/SCCP69xx.9-2-1-0.zz.sgn SCCP69xx.9 - 2-1 TFTP server - 0.zz.sgn
Server TFTP flash:/PHONE/7942_7962/Sccp/apps42.9-2-1TH1-13.sbn alias apps42.9 - 2-1TH1 - 13.sbn
Server TFTP flash:/PHONE/7942_7962/Sccp/cnu42.9-2-1TH1-13.sbn alias cnu42.9 - 2-1TH1 - 13.sbn
Server TFTP flash:/PHONE/7942_7962/Sccp/cvm42sccp.9-2-1TH1-13.sbn alias cvm42sccp.9 - 2-1TH1 - 13.sbn
Server TFTP flash:/PHONE/7942_7962/Sccp/dsp42.9-2-1TH1-13.sbn alias dsp42.9 - 2-1TH1 - 13.sbn
Server TFTP flash:/PHONE/7942_7962/Sccp/jar42sccp.9-2-1TH1-13.sbn alias jar42sccp.9 - 2-1TH1 - 13.sbn
alias server TFTP flash:/PHONE/7942_7962/Sccp/SCCP42.9-2-1S.loads SCCP42.9 - 2 - 1 S .loads
flash:/PHONE/7942_7962/Sccp/term42.default.loads alias term42.default.loads TFTP server
flash:/PHONE/7942_7962/Sccp/term62.default.loads alias term62.default.loads TFTP server
Server TFTP flash:/PHONE/7945_7965/Sccp/cmterm-7945_7965-sccp.9-2-1.tar alias cmterm-7945_7965 - sccp.9 - 2 - 1.tar
Server TFTP flash:/PHONE/7945_7965/Sccp/apps45.9-2-1TH1-13.sbn alias apps45.9 - 2-1TH1 - 13.sbn
Server TFTP flash:/PHONE/7945_7965/Sccp/cnu45.9-2-1TH1-13.sbn alias cnu45.9 - 2-1TH1 - 13.sbn
Server TFTP flash:/PHONE/7945_7965/Sccp/cvm45sccp.9-2-1TH1-13.sbn alias cvm45sccp.9 - 2-1TH1 - 13.sbn
Server TFTP flash:/PHONE/7945_7965/Sccp/dsp45.9-2-1TH1-13.sbn alias dsp45.9 - 2-1TH1 - 13.sbn
Server TFTP flash:/PHONE/7945_7965/Sccp/jar45sccp.9-2-1TH1-13.sbn alias jar45sccp.9 - 2-1TH1 - 13.sbn
alias server TFTP flash:/PHONE/7945_7965/Sccp/SCCP45.9-2-1S.loads SCCP45.9 - 2 - 1 S .loads
flash:/PHONE/7945_7965/Sccp/term45.default.loads alias term45.default.loads TFTP server
flash:/PHONE/7945_7965/Sccp/term65.default.loads alias term65.default.loads TFTP server
alias of server TFTP flash:/PHONE/6921_6941_6961/Sccp/DSP69xx.0-0-0-8.zz.sgn DSP69xx.0 - 0-0 - 8.zz.sgn
!
control plan
!
!
voice-port 0/2/0:23
!
Voice-port 1/0/0
signal groundStart
name of station id Leo Salcie
activation of the caller ID
!
Voice-port 1/0/1
!
voice-port 3/0/0
!
voice-port 1/0/3
!
voice-port 0/3/2
!
voice-port 0/3/3
!
CCM-Manager mgcp
CCM-Manager config server 192.168.6.10
!
MGCP
type of service mgcp MGCP call-agent 192.168.6.10 version 0.1
MGCP ecm t38 fax
!
profile MGCP default
!
SCCP local FastEthernet0/0.10
SCCP ccm 192.168.2.1 ID 1 priority 1 version 4.1
SCCP
!
SCCP ccm Group 1
associate the ccm 1 priority 1
the associated profile 1 save CME-VOUGEOT
!
transcode dspfarm profile 1
Codec g711ulaw
Codec g711alaw
Codec g729ar8
Codec g729abr8
Codec g729br8
Codec g723r53
maximum sessions 4
associate the PCRS application
!
Dial-peer cor custom
name 911call
name localcall
name ldcall
!
!
Dial-peer cor list LD
Member 911call
Member localcall
Member ldcall
!
Dial-peer cor list localcall
Member localcall
!
Dial-peer cor list ldcall
Member ldcall
!
!
Dial-peer voice 1000 voip
destination-model 3...
session target ipv4:192.168.4.1
!
!
!
!
phone service
5 units of sdspfarm
sessions to transcode sdspfarm 4
sdspfarm tag 1 GUY-VOUGEOT
Logout em 0:0 0:0 0:0
Max-joined 12
Max - dn 12
Source-address 192.168.2.1 IP port 2000
Auto assign 1 to 6
load SCCP45.9 - 2-1 7965 s
MAX conferences 8-6 win
transfer full-consult system
create the file cnf version-stamp 7960 21 June 2012 09:37:02
!
!
ePhone-dn 1
number 2001
!
!
ePhone-dn 2
number 2002
!
!
ePhone-dn 3
2003 Edition
!
!
ePhone-dn 4
issue 2004
!
!
ePhone-dn 5
issue 2005
!
!
ePhone-dn 6
Number of 2006
!
!
ePhone-dn 9 double line
number 9999
!
!
ePhone 1
security-mode device no
503D Mac address. E5E9.2001
Max-calls-by button-2
type of 6921
button 1:1
!
!
!
ePhone 2
security-mode device no
A 64-0 Mac address. E715. F5B2
type of 7965
key 1:2
!
!
!
ePhone 5
security-mode device no
503D Mac address. E52F. C3E4
Max-calls-by button-2
key 1:3
!
!
!
ePhone 9
security-mode device no
00F0.1D00.0081 Mac address
!
!
!
ePhone 99
security-mode device no
address Mac AABB. CC11.2233
key 1:9
!
!
!
!
Line con 0
line to 0
line vty 0 4
password 7 151E0E03082F24
opening of session
transport of entry all
!
Scheduler allocate 20000 1000
NTP master 3
Server NTP 192.168.2.1
end

R2, see the race

==========

CME-Legacy #show run
Building configuration...

Current configuration: 3189 bytes
!
! Last modification of the configuration at 22:44:24 UTC Thursday, June 28, 2012
!
version 15.1
horodateurs service debug datetime msec
Log service timestamps datetime msec
encryption password service
!
CME-Legacy host name
!
boot-start-marker
boot-end-marker
!
!
type 0 3 t1 card
enable secret 5 $1$ $0iFd Um7r8JhKS6fMrEpFH64rB0
!
No aaa new-model
!
network-clock-participate wic 3
!
dot11 syslog
IP source-route
!
!
IP cef
!
!
!
no ip domain search
No ipv6 cef
!
Authenticated MultiLink bundle-name Panel
!
!
!
!
primary-qsig ISDN switch type
!
!
!
voip phone service
list of approved IP addresses
IPv4 192.168.2.1
IPv4 192.168.3.1
SIP
!
voice class codec 1
g711ulaw codec preference 1
codec preference 2 g729r8
g729br8 preferably 3 codec
g711alaw preferably 4 codec
!
!
!
!
!
voice-card 0
dspfarm
DSP services dspfarm
!
Crypto pki token removal timeout default 0
!
!
!
!
license udi pid CISCO2811 sn FHK1246F3KE
Archives
The config log
hidekeys
!
redundancy
!
!
controller LAN 0/0/0
!
controller T1 3/0/0
the clock source internal
long CableLength 0dB
time intervals PRI - Group 1-2: 24
!
!
!
!
!
!
!
!
!
interface Loopback0
IP 192.168.100.1 address 255.255.255.0
!
interface FastEthernet0/0
IP 192.168.4.2 255.255.255.0
automatic duplex
automatic speed
!
interface FastEthernet0/1
no ip address
Shutdown
automatic duplex
automatic speed
!
interface Serial0/2/0
IP 192.168.3.2 255.255.255.0
encapsulation ppp
!
interface Serial0/2/1
no ip address
Shutdown
2000000 clock frequency
!
interface Serial0/3/0:23
no ip address
encapsulation hdlc
primary-qsig ISDN switch type
ISDN incoming-voice
No cdp enable
!
!
Router eigrp 1
192.168.3.0 network
192.168.4.0 network
network 192.168.100.0
Auto-resume
!
IP forward-Protocol ND
no ip address of the http server
no ip http secure server
!
!
!
exploitation forest esm config
!
!
!
!
!
!
control plan
!
!
voice-port 3/0/0:23
!
Voice-port 1/0/0
!
Voice-port 1/0/1
!
CCM-Manager mgcp
CCM-Manager config server 192.168.6.10
!
MGCP
type of service mgcp MGCP call-agent 192.168.6.10 version 0.1
MGCP ecm t38 fax
!
profile MGCP default
!
SCCP local Loopback0
SCCP ccm 192.168.100.1 ID 1 priority 1 version 7.0
SCCP
!
SCCP ccm Group 1
bind the interface Loopback0
associate the ccm 1 priority 1
the associated profile 1 save CME-VOUGEOT
!
transcode dspfarm profile 1
Codec g711ulaw
Codec g711alaw
Codec g729ar8
Codec g729abr8
Codec g729r8
Codec g723r53
maximum sessions 2
associate the PCRS application
!
voice pots Dial-peer 3001
destination-model 3001
port 1/0/0
!
Dial-peer voice 1000 voip
destination-model 2...
session target ipv4:192.168.3.1
!
voice pots Dial-peer 3002
destination-model 3002
port 0/1/1
!
Dial-peer voice voip 5000
session target ipv4:192.168.4.1
incoming called-number 3...
!
!
!
!
phone service
5 units of sdspfarm
sessions to transcode sdspfarm 2
sdspfarm tag 1 GUY-VOUGEOT
Logout em 0:0 0:0 0:0
Max-joined 4
Max - dn 4
IP source-address 192.168.100.1 port 2000
MAX conferences 8-6 win
transfer full-consult system
create a cnf-files version-stamp 1 January 2002 00:00:00
!
!
!
Line con 0
password 7 09404B 06150018
opening of session
line to 0
line vty 0 4
password 7 11051C0A1B1704
opening of session
transport of entry all
!
Scheduler allocate 20000 1000
end

CME-Legacy #.

Any help would be appreciated

Concerning

Technically, this isn't a CUBE configuration. The calls come in via a VOIP dial-peer and they are sent to a pots dial-peer.

Can you get debug voip ccapi inout of R2. We have to see what happens when the call comes in on this router and why the call disconnects.

If you can put the debugging in a text file and join here it will be better to post directly here

Please note the useful messages

"For the love of God is broader than the extent of the human spirit and the heart of the eternal is most wonderfully kind."

Configure incoming calls to route to the internal unit

I have a Cisco 2921 router which has a 4 FXO inside card. I would like to configure so that ALL incoming calls on all 4 ports to be forwarded to a post internal (1001), it is a test environment and I can't seem to understand what Miss me. The config is below:

Building configuration...

Current configuration: 8500 bytes

!

! Last configuration change at 08:19:46 EST Friday, March 1, 2013 by sjones

!

version 15.1

horodateurs service debug datetime msec localtime

Log service timestamps datetime msec localtime

no password encryption service

sequence numbers service

!

hostname WH-VOIP-2900

!

boot-start-marker

boot-end-marker

!

!

logging buffered 10000000

!

AAA new-model

!

!

AAA authentication login default group Ganymede + local line

/NOAUTH AAA authentication login no

default AAA authorization exec group Ganymede + local no

/NOAUTH AAA authorization exec no

orders accounting AAA 15 by default start-stop Ganymede group.

Default connection accounting AAA power Ganymede group.

!

!

!

!

!

AAA - the id of the joint session

!

clock timezone IS - 5 0

summer time clock IS recurring

!

No ipv6 cef

IP source-route

IP cef

!

!

!

!

!

no ip domain search

IP domain name mgsd.edu

!

Authenticated MultiLink bundle-name Panel

!

!

!

!

!

!

FXO trunk group

!

Crypto pki token removal timeout default 0

!

Crypto pki trustpoint TP-self-signed-3979560690

enrollment selfsigned

name of the object cn = IOS - Self - signed - certificate - 3979560690

revocation checking no

!

!

TP-self-signed-3979560690 crypto pki certificate chain

certificate self-signed 01

308201B 6 A0030201 02020101 3082024D 300 D 0609 2A 864886 F70D0101 04050030

2 060355 04031326 494F532D 53656 C 66 2 AND 536967 6E65642D 43657274 31312F30

69666963 33393739 35363036 6174652D 3930301E 170 3130 31323232 31333533

30375A 17 0D 323030 31303130 30303030 305A 3031 06035504 03132649 312F302D

4F532D53 5369676E 656C662D 43 65727469 66696361 74652 33 39373935 65642D

36303639 3030819F 300 D 0609 2A 864886 01050003, 818, 0030, 81890281 F70D0101

8100DD47 9227149F 2D084CE5 3 D 7DBF4FCA 227595 C3519000 3F468821 D56F653A

E74FCBAD B4936598 F0C26B2B 6132ADE7 1B1BDC89 44D3C53F 63DDAF78 8E08FCA7

7044095A DBE38889 7CD 48871 94ED1CF9 F2ECC50A 8BD21AFC 5BC3B3FC B322E161

F3CE339A 88AA803B E3705349 03A7D918 C11E5844 ECF039EB FEC44CDF 52A59AE5

0C 430203 010001A 3 75307330 1 130101 FF040530 030101FF 30200603 0F060355

551 1104 19301782 1557482D 564F4950 2 D 302E6D67 323930 73642E65 6475301F

23041830 16801463 9BA90049 2F6005DC F2A35FC3 0EDB2530 0603551D 4138 329D

1 D 060355 1D0E0416 0414639B A900492F 6005DCF2 A35FC332 9D41380E DB25300D

06092A 86 01010405 00038181 005C2C45 9F687AEF 3219F567 337E55CD 4886F70D

9E524A1B 7879B3B1 F3C872F9 DFF7F014 FFE0D84B 67252EFE 3DFF8959 9565ADE2

79857E34 FFF2C3DE 667D5D62 8A4E4690 D874CF4A 8B 180832 7748D1E8 BB71543B

BC404126 02DABACB DDF24EE6 6F63F8CE F7F8494C 66115C B768BC77 DA2D5C2C 77

984DC376 A16F2B81 D1CBD44F F23B8605 D4

quit smoking

voice-card 0

DSP services dspfarm

!

!

!

voip phone service

h323 connections allow h323

allow connections h323 to SIP

allow connections sip h323

allow sip to sip connections

redirect ip2ip

Fax protocol t38 ls-redundancy version 0 0 hs-redundancy 0 help none

H323

!

voice class codec 1

g711ulaw codec preference 1

codec preference 2 g729r8

!

vocal h323 class 1

H225 timeout tcp establish 3

Call slow start

prerogative of the call

!

!

!

!

!

license udi pid CISCO2921/K9 sn FTX1448AJ6B

HW-module pvdm 0/0

!

!

!

username admin privilege 15 secret 5 $1$ iKc / $uQJli0iQG9VAu4PiFeYC8 /.

!

redundancy

!

!

!

!

!

!

!

!

!

!

!

interface GigabitEthernet0/0

Description inside LAN

IP 10.40.0.51 255.255.0.0

automatic duplex

automatic speed

H323-gateway voip interface

H323-gateway voip bind port 10.40.0.51

!

interface GigabitEthernet0/1

no ip address

Shutdown

automatic duplex

automatic speed

!

interface GigabitEthernet0/2

no ip address

Shutdown

automatic duplex

automatic speed

!

IP forward-Protocol ND

!

IP http server

local IP http authentication

IP http secure server

IP http timeout policy slowed down 60 life 86400 request 10000

!

IP route 0.0.0.0 0.0.0.0 10.40.0.1

!

!

!

!

!

!

SNMP-Server RO community mgsdvoip

SNMP-Server RO community mhsswitch

location of Server SNMP "Mooresville High School"

Server enable SNMP traps snmp authentication linkdown, linkup warmstart cold start

Enable SNMP-Server intercepts ATS

Server enable SNMP traps eigrp

Enable SNMP traps envmon fan supply temperature State of the server stop

Server enable SNMP traps insertion withdrawal flash

SNMP-Server enable traps energywise

Server enable SNMP traps cef resources-failure-change of State peer peer-fib-state-change inconsistency

Server enable SNMP traps config-copy

config SNMP-server enable traps

Server enable SNMP traps config-ctid

entity of traps activate SNMP Server

Server enable SNMP traps hsrp

Enable SNMP-server holds the CPU threshold

Server enable SNMP traps syslog

Server enable SNMP traps vtp

Server enable SNMP traps srst

SNMP-Server enable traps voice

SNMP-server host 10.65.0.252 version 2 c mgsdvoip

SNMP-server host 10.10.0.252 version 2 c mhsswitch

RADIUS-server host 10.60.253.10 key Pa$ $word

RADIUS-server application made

!

!

control plan

!

!

voice-port 0/0/0

1 FXO-group of circuits

connection ÉRA 1001

Description 704-799-0516

!

voice-port 1/0/0

2 FXO-group of circuits

connection ÉRA 1001

!

voice-port 0/0/2

3 FXO trunk-group

connection ÉRA 1001

!

voice-port 0/0/3

4 FXO-group of circuits

connection ÉRA 1001

Description ==> 911

!

!

!

SCCP local GigabitEthernet0/0

SCCP ccm 10.65.0.63 identifier 1 version7.0

SCCP

!

SCCP ccm Group 1

link interface GigabitEthernet0/0

associate the profile 1 WH-2900_CFB register

the associated profile 2 registry WH-2900_MTP

!

dspfarm profile Conference 1

Codec g711ulaw

Codec g711alaw

Codec g729ar8

Codec g729abr8

Codec g729r8

Codec g729br8

maximum sessions 4

associate the PCRS application

!

dspfarm profile 2 PSG

Codec g711ulaw

maximum sessions 2 material

associate the PCRS application

!

voice POTS dial-peer 1

trunkgroup FXO

incoming called-number.

!

Dial-peer voice 2 pots

destination-model 9 [2-9] 11

Setup progress_ind allow 3

alert progress_ind activate 8

progress_ind enable progress 8

port 0/0/3

Forward-digits 3

!

Dial-peer voice 3 pots

destination-model $ 911

Setup progress_ind allow 3

alert progress_ind activate 8

progress_ind enable progress 8

port 0/0/3

Forward-digits all the

!

Dial-peer voice 4 pots

trunkgroup FXO

destination-model 9 [2-9]... [2-9]......

Setup progress_ind allow 3

alert progress_ind activate 8

progress_ind enable progress 8

Forward-digits 10

!

voice pots Dial-peer 5

trunkgroup FXO

destination-model 91 [2-9]... [2-9]......

Setup progress_ind allow 3

alert progress_ind activate 8

progress_ind enable progress 8

Forward-digit 11

!

Dial-peer voice 6 pots

trunkgroup FXO

destination-style 9011T

Setup progress_ind allow 3

alert progress_ind activate 8

progress_ind enable progress 8

prefix 011

!

Dial-peer voice 32 pots

trunkgroup FXO

composition of 4-digit SRST Description to other sites

destination-model 2...

Forward-digits all the

prefix 704658

!

Dial-peer voice 100 voip

preference 1

destination-model [2]...

Setup progress_ind allow 3

progress_ind connect enable 8

progress_ind disconnect switch 8

session target ipv4:10.65.0.23

codec voice-class 1

h323 voice-class 1

DTMF-relay h245 alphanumeric

rate of 14400 Fax

IP qos dscp cs5 signaling

No vad

!

Dial-peer voice voip 101

preference 2

destination-model [2]...

Setup progress_ind allow 3

progress_ind connect enable 8

progress_ind disconnect switch 8

session target ipv4:10.65.0.63

codec voice-class 1

h323 voice-class 1

DTMF-relay h245 alphanumeric

rate of 14400 Fax

IP qos dscp cs5 signaling

No vad

!

!

!

!

access controller

Shutdown

!

!

Call-Manager-emergency

secondary-tone 9

MAX conferences 4-6 win

transfer full-consult system

3 timeouts interdigit

IP source address 10.40.0.51 port 2000

Max-joined 50

Max - dn 100 double line

primary phone message system is offline

secondary system message standalone

1 7046582 model numbering plan... extension-length 4

transfer-model. T

KeepAlive 10

voicemail 2525

call-Park select non-auto-match

ground of appeal forwards. T

call forward availability 97046582525

timeout before call 97046582525 16 noan

aa-mm-dd date format

!

!

VM integration

direct model * GNC

peer-to-peer of nonresponse 5 FDN of mires * GNC *.

peer-to-peer busy 7 FDN of mires * GNC *.

safe-to-post non-response 4 FDN of mires * GNC *.

safe-to-position 6 FDN of mires * GNC *.

!

!

Line con 0

password V01pG8te

line to 0

line vty 0 4

access-class 23 in

privilege level 15

password V01pG8te

transport input telnet ssh

line vty 5 15

access-class 23 in

privilege level 15

password V01pG8te

transport input telnet ssh

line vty 16 1114

transport of entry all

!

Scheduler allocate 20000 1000

NTP 129.6.15.29 Server

end

Jeff,

I guess that 100 & 101 voip dial peers point to a CuCM?

The destination model on the voip dial peer does not 1001 on the ERA and they must change to something like: -.

Dial-peer voice voip 101

voice mail Dial 100

destination-model [12]...

voice mail Dial 101

destination-model [12]...

destination-model [2]...

Hope this helps,

Craig

PLEASE EVALUATE THE MESSAGES USEFUL

CUCM SIP-trunk multiple Ports

I would like to know if it is possible to create a Script of SIP standardization in CUCM which change the Destination SIP port based on the phone which makes the phone.

Current issue:

The ITSP provider asked that I send calls from different regions to the same address IP SBC but a different port.

Example:

Phone (A) makes a call

RP - RL - RG---> SIP Trunk (5060)---> CUBE (5060<-->5001)---> ITSP1

Telephone (B) makes a call

RP - RL - RG---> SIP Trunk (5060)--->---> CUBE ITSP1 (5060<-->5002)

I can do it on the CUBE, but which require several dial-Exchange and SIP profiles, is it possible in the CUCM to change the sip by device-pool port?.

Or is there another I can match and send calls via the same destination with different dial-peer sip ports.?

Thank you.

Zakiab,

You can do this, its all just impossible. Your signage ports are defined not on the endpoints, but on the B2BUA (CUCM or CUBE). The CUBE, you have the most flexibility, because you can change the port of signs based on your dial-peers. However, as you pointed out quite rightly, need you an insurmountable amount of dial-peers to have this for each endpoint. So I suggest tell you your ITSP to change their design or moving to a new.

Fax Server help

We have our PRI come to our 2950 some of the DID attached to the PRI are to our server, GFI Faxmaker. This server is using a Brooketrout TR114 + P4C card 2 ports boot loop 0-1 and 2 ports DID 2-3 ports. On the 2950 router I have a port 0 of the brooktrout plugged into port 0/1/2 of the VIC-4FXS/DID card. Server Fax Send and receive all calls. Questions, this is the router does not send data to the fax server so that he knows who is supposed to get the fax. I'm debugging and I see the called number is correct, but after that, I'm lost. So two things here is the card we have our old building he was connected to a line ISDN supporting our supplier is no longer correct. If the card is good what Miss me?

Thank you

I'm not sure to understand what you're trying to say. "The fax jobs, but they are sent to the default account" as far as I can see the trace that you sent me, said the call are connected to 0442. port 0/1/2.

But since you say that you are using the GFI-fax server, I think that a should be a voip connection. The call does not connect to the voip dial-peer. Where did the GFI fax server connected to?

The trace when the call was dial-position 22 road, he rejected...

UN 2012 29 MDT 13:46:24.968: / / 154577/F1A9F8699426/CCAPI/ccCallSetAAA_Accounting:

Accounting = 1, Call Id = 154577

June 29, 2012 MDT 13:46:24.968: / / 154577/F1A9F8699426/CCAPI/ccCallDisconnect:

Value = 21, Tag = 0x0, entry calls (previous disconnection Cause = 0, remove the Cause = 21)

June 29, 2012 MDT 13:46:24.968: / / 154577/F1A9F8699426/CCAPI/ccCallDisconnect:

Value = 21, entered calls (Responsed = TRUE, Cause value = 21)

What is this ip address? 172.16.17.22 is the one where the call should be sent to the GFI Server?

Please note all useful posts

"There is a width in the mercy of God as the width of the sea. There is a kindness in his justice, which is more freedom"

SAI 243 x as SIP <>modem router ISDN with iLBC

Hi all

I have a couple of SAI 243 x (specifically IAD2431 and IAD2432) running IOS 15.1 (4) M5 that I use as SIP for ISDN PRI E1 gateways. They are configured with POTS by a dial peer for ISDN and a couple of VoIP dial-peers according to the new number called ISDN. He works very well with G.711 A-law-to-end, but now I'm trying to work with iLBC and having problems.

My dial-peer configuration looks like this:

voice POTS dial-peer 1

Description peer ISDN30

translation-profile incoming incoming bt

service logon

destination-model. T

alert progress_ind-band 8

No Strip number

direct line to inside

port 1/0:15

Forward-digits all the

!

Dial-peer voice 3 voip

your reminder alert-non-PI

Description peer to number of bearer

huntstop

service logon

destination-model 01234567890

RTP payload type nte your 102

RTP payload type comfort-noise 13

session protocol sipv2

session target ipv4:1.2.3.4:5060

codec voice-class 1

numbers-fall of DTMF-relay rtp - nte

IP qos dscp cs3 signaling

CLID network provided

replacement CLID name

!

Dial-office of communications telephone voip 65008

your reminder alert-non-PI

Description peer for the main range

huntstop

service logon

destination-model 01234567 [1-8].

RTP payload type nte your 102

RTP payload type comfort-noise 13

session protocol sipv2

session target ipv4:1.2.3.4:5060

codec voice-class 1

numbers-fall of DTMF-relay rtp - nte

IP qos dscp cs3 signaling

CLID network provided

replacement CLID name

Another relevant config:

voice, send rtp-received

!

voice service pots

keepalive RTCP

!

voip phone service

list of approved IP addresses

IPv4 1.2.3.4

keepalive RTCP

Standard DTMF-interworking

Fax protocol t38 ls-redundancy version 0 0 hs-redundancy 0 relief pass through g711alaw

SIP

udp tcp transport switch

No anat

block 183 present sdp

!

voice class codec 1

mode of ilbc codec preference 1 30

codec preference 2 g729r8

g711alaw preferably 3 codec

!

Now, when I place a call from the ISDN through the gateway, the SDP, he offers to my SBC is:

v = 0

o = CiscoSystemsSIP-GW-UserAgent 7963 1810 IN IP4 6,7,8,9

s = call SIP

c = IN IP4 6,7,8,9

t = 0 0

m = audio RTP/AVP 116 18 8 101 13 18710

c = IN IP4 6,7,8,9

a = rtpmap:116 iLBC/8000

a = fmtp:116 mode = 30

a G729/8000 rtpmap:18 =

a = fmtp:18 annex b = No.

a = rtpmap:8 PCMA/8000

a rtpmap:101 telephone-event/8000 =

a = fmtp:101 0-16

a = rtpmap:13 CN/8000

It is very good - both ends of iLBC agreement and the call comes. However, if I make a call on the side SIP through ISDN, I offer you this:

v = 0

o =-78207127932020 78207127932020 IN IP4 1.2.3.4

s = -.

c = IN IP4 1.2.3.4

t = 0 0

m = 24132 audio RTP/AVP 110 8 127

a = rtpmap:110 iLBC/8000

a rtpmap:127 telephone-event/8000 =

a = fmtp:110 mode = 30

a = silenceSupp: suite.

a = ptime:20

Then I'm a 200 OK with that:

v = 0

o = CiscoSystemsSIP-GW-UserAgent 9977 8349 IN IP4 6,7,8,9

s = call SIP

c = IN IP4 6,7,8,9

t = 0 0

m = 17650 audio RTP/AVP 8 127

c = IN IP4 6,7,8,9

a = rtpmap:8 PCMA/8000

a rtpmap:127 telephone-event/8000 =

a = fmtp:127 0-16

a = ptime:20

This means that we use G.711 A - Law. If I've got this remove the offer and propose simply iLBC, the IAD rejects the appeal.

Clearly, it takes care of calls using G.711 on the ISDN and iLBC side SIP - how to persuade to allow calls made from the side SIP at work and calls made from the side of ISDN?

Thank you very much in advance for your suggestions!

Sean

You should have "incoming called-number." under voip RFP.

Copy rotobrush different video layer mask

Hello!
I am currently working on some work in AE title. The text moves slowly away from the camera and I must add a few splashes of blood surrounding only the text box. I followed the movement and the rotobrushed text. However the rotobrush masks only the bottom of the layer with the title. Is there anyway to get this mask on the layer with the splash of blood?
Thank you!

Use as a matte. May be required before dialing.

Mylenium

SPA3000 dial of Voip to PSTN problem

Do it with the dial plan. Set the PSTN Caller default DP a number available and configure the dial plan:

(S0<:[email protected]>) where [email protected] is the sip uri when you want to transfer the call.

If it's a PSTN number you want to transfer the call to you must configure tab PSTN line with the credentials of voip to voip account, you want to use to transfer the call.

Of course, the line 1 tab you set enable component IP: Yes, if you send to a sip uri.

required VoIP Dialer?

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