IP TelephonySponsored by Jamf Now

IP telephony (Internet Protocol telephony) is a general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). Using the Internet, calls travel as packets of data on shared lines, avoiding the tolls of the PSTN. The challenge in IP telephony is to deliver the voice, fax, or video packets in a dependable flow to the user.

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Hello There,

I have a a2billing installaed on centos , to witch I want to send the number base64 encoded, and to transalte it from a2billing,

I can do this from asterisk to asterisk , from sipphone to asterisk , but not from sipphone to a2billing,

I tried modifing the a2billing extension :

[a2billing]
exten => _123.,1,SET(EXTEN=${BASE64_DECODE(${EXTEN))})
exten => _123.,2,AGI(a2billing.php,1)
exten => _123.,n,Hangup()

But i dosent work , any idea would be appreciated
0
Hi
we have configured h323 Gatway.

when we call landline number 3175 IP Phone to 61871111 and 6187100 are working fine. but other all landline number eg 61871324 and 62300192 etc unable to connect getting busy tone.
0
Hi
we have configured h323 Gatway.

when we call landline number 3175 IP Phone to 61871111 and 6187100 are working fine. but other all landline number eg 61871324 and 62300192 etc unable to connect getting busy tone.
0
I have a 14 office MPLS link that was working with a Mitel phone system.   We had an emergency need to move the offices so I came up with a solution to use an Cisco ASA to create an IPSec VPN tunnel from the new location to the firewall.

Currently, I only get one-way audio when making calls.  
Additionally, my Mitel Controller is showing a SIP LINK Failure alarm.
I got the tunnel up and running and everything can be pinged.

I am so confused.

Maybe this is a problem with my Access-Control List?  

Here is the top part of my Cisco Config:

ASA Version 8.4(2)8
!
hostname dartmouth-asa
domain-name test.com
enable password OzKLyBY8hcexbQv8 encrypted
passwd 2KFQnbNIdI.2KYOU encrypted
names
!
interface Ethernet0/0
 description INTERNET/OUTSIDE
 switchport access vlan 2
!
interface Ethernet0/1
 description VOICE
!
interface Ethernet0/2
 description DATA
 switchport access vlan 3
!
interface Ethernet0/3
 description DATA
 switchport access vlan 3
!
interface Ethernet0/4
!
interface Ethernet0/5
!
interface Ethernet0/6
!
interface Ethernet0/7
!
interface Vlan1
 nameif voice
 security-level 100
 ip address 192.168.171.1 255.255.255.0
!
interface Vlan2
 nameif outside
 security-level 0
 ip address dhcp setroute
!
interface Vlan3
 nameif data
 security-level 100
 ip address 192.168.172.1 255.255.255.0
!
boot system disk0:/asa842-8-k8.bin
ftp mode passive
dns server-group DefaultDNS
 domain-name test.com
same-security-traffic…
0
I have configured asterisk by ./configure.
It has been completed successfully. But "./configure –with-crypto –with-ssl –with-srtp=/usr/local/lib" command not working.

please have a look.

[root@localhost asterisk-13.6.0]# make menuselect.makeopts
make: `menuselect.makeopts' is up to date.

[root@localhost asterisk-13.6.0]# menuselect/menuselect enable format_mp3 enable res_config_mysql enable app_mysql enable app_saycountpl enable cdr_mysql enable EXTRA-SOUNDS-EN-GSM
**************************************************
*** Install ncurses to use the menu interface! ***
**************************************************

but i have seen ncurses has been installed.
[root@localhost asterisk-13.6.0]# rpm -qa | grep ncurses
ncurses-libs-5.9-14.20130511.el7_4.x86_64
ncurses-5.9-14.20130511.el7_4.x86_64
ncurses-devel-5.9-14.20130511.el7_4.x86_64
ncurses-base-5.9-14.20130511.el7_4.noarch


how can be solve. please help.
0
Hello all,

I need to configure the Sonus gateway to route inbound calls to Toll Free number (main number) based on their area code to their particular response group on my Skype for Business server.

For instance, I have a response group for Chicago with DID 312XXXXXX and I have a main toll free number in which case all the calls come to this particular DID (800XXXXXXX) so I need to route all incoming calls from Chicago to the Chicago response group and the same thing for every other state.

I made the area codes already so all calls coming from Chicago state is directly forwarded to the Chicago response groups but the only thing I couldn't do is route all inbound calls coming to the Toll Free number to the response groups.

I would appreciate any advise on this.
ThanksAreaCode.jpg
0
I have a Cisco UCM cluster ver 10.5.2. We recently migrated all of our outbound calling to an enterprise SIP trunk from a major telecom.
The issue I have is that my inbound calling is on PRI's from a Different major telecom. 3 PRI's in an NFAS group. The calls only come in as 4 digit DNIS. (Previous to UCM the system was on a Nortel PBX) Inbound to my users works fine. However, when a user attempts to forward their line it will not go through.

Trace from RTMT shows 2 call legs. The first as inbound from external number to DNIS with the appropriate routing. The second, rerouting leg show that the call is being sent to the extension again.

"Start Time","Stop Time","Initial Speaker","From","To","Protocol","Duration","Packets","State","Comments"
"59.901472","64.552939","10.136.XX.XX [UMC SUB]",""Internal User x6900" <sip:6900@10.136.XX.XX","<sip:Forwardednumber@10.220.xx.xx"[internalsignalonSBC],"SIP","00:00:04","7","COMPLETED","INVITE 200"
"59.905243","64.606663","10.220.xx.xx"[ExternalSignalOnSBC],""InternalUser x6900" <sip:6900@10.220.xx.xx"[ExternalSignalOnSBC],"<sip:6900@10.138.0.11"[MajorCarrierSignalInterface],"SIP","00:00:04","10","COMPLETED","INVITE 200 200"


I believe that the issue is in the re-routing calling search space. But I am unable to get the correct syntax. Any help?
0
The Samsung OfficeServ 7200 is behind our Spectrum EVPL (Ethernet Virtual Private Line). We have one Cisco Firewall at the main site. All of our other locations (except for three of them) are also connected to the Spectrum EVPL so no Firewall are needed at these sites. We are able to communicate with our other Samsung OfficeServ systems (100, 500, 7100, 7200) without any issues. But for the three sites that are not on the Spectrum EVPL network, we are using AT&T and Frontier at these sites and using the Cisco ASA 5505 to establish a Site-to-Site IPsec VPN. So when I try to make a call, the phone will ring but when I pick up we have no audio on either side. But here is the strange part. When I created the Site-to-Site IP Sec VPNs at all three locations, I set it up to where they could communicate not only to the main site but also to the remote sites. When we dial the extensions at the remote site, we have audio both ways. My phone vendor keeps telling him that it is a Firewall issue but if that was true, then why am I am able to call the other phone systems. They are going through the same Site-to-Site IPsec VPN tunnel. I think it has to do with the updated firmware that is installed on the Samsung OfficeServ 7200 at the main site. He updated it to a more recent firmware because we are using the Samsung CMS software for reporting. Please help.
0
Have been on a Cisco/Linksys E3000 router for a few years.  My Mitel 5360 IP phone has worked flawlessly connecting to my office the entire time.

Replaced my router with a Netgear Nighthawk X4 R7500V2.  The IP phone gets an IP address (DHCP) but hangs at "Contacting Server."  Cannot connect to my office.

Reconnected the old router and the IP phone works fine so it's not the phone.  

Suspect I have to open up a port on the new router but have no idea where to start.

Any suggestions appreciated.
0
I'm setting up an application that integrates with asterisk. From my webapp, a user registers and connects to the asterisk. I have successfully setup the billing of user A calling user B with the a2billing. Now there is another feature of my app that requires B-Party(the callee) to be billed for receiving calls. I've been searching online for any clue but couldn't find any.
Please can someone with the a2billing knowledge help out with how to get this done?
0
Hi,

We have VOIP service through intermedia, they do not offer call recording. Is there some third party service i can use to record calls?
0
I have a small housing community which has 7 IP cameras which currently feed back to a NVR which records streaming in real time. Currently the Enginus wifi antanas. The management company of the property wants to have the NVR removed from the property and placed at their corp office which is about 20 miles away. I can get a good (fast) interent connection to the property and I am pretty sure I will need to setup a router at the current location to hand out IP addreses.

The question is, how can I get a connection from the housing community property all the way to the management property so that the NVR being placed there, will be able to see and record the IP cameras from the HOA property ?

I hope this makes sense.
0
Hello, I work with asterisk servers and mainly soft SIP clients. Customers would like to have the possibility to hang up an incoming call. That's the first step...no big deal. But what if the call is a "call group" call i.e. a call to multiple devices at the same time and you want the command from the client not only to hang up his device while the others keep ringing, but to hang up all legs of the call - caller and all callees. Is that possible?
0
I don't much about the Avaya IP-Office 500.  I've attached some pics that may help.  The first pic is the blue cable that connects to the phone but is no longer providing service.  
During our rack equipment move, we may have re plugged in one cable on the IP Office in the wrong port.  I'm not sure if that matters as I'm not proficient in PBX.  I had one of the staff members mark cables based on what phone goes out if we unplug it.  I'm not sure if that's done yet.  I left for another meeting.  
Anyway, are the patch cables placement very fickle for the IP-Office?  We may have accidentally plugged in one cable to the wrong port when moving things and that caused the phone to no longer light up.   Sorry, I'm walking blind here.  I would greatly appreciate some feedback.
blue_cable_from_face_plate_to_phone.jpg
a_and_b_drops.jpg
IP-Office_Ports.jpg
0
we create a sip trunk, cisco phones can call to avaya, when a try to call a cisco show "INCOMPATIBLE" im not a expert on avaya, ideas  ?

log avaya -> cisco

274138126mS PRN: ++++ END OF TCP MONITOR CLIENT DUMP ++++
 274160750mS Sip: SIP Line (17): License, Valid 1, Available 15, Consumed 0
 274160750mS Sip: SIP Line (17): sip_trunk_config_items 0002c10c, voip.flags 00040949
 274160750mS Sip: SIPDialog f172d9f4 created, dialogs 1
 274160756mS Sip: 0a3c1e8c0003346c 17.210028.0 33506 SIPTrunk Endpoint(f172f28c) received CMSetup
 274160757mS Sip: 0a3c1e8c0003346c 17.210028.0 33506 SIPTrunk Endpoint(f172d9f4) SetLocalRTPAddress to 10.60.30.140:46754
 274160759mS SIP Call Tx: 17
                    INVITE sip:50528337@10.120.200.20 SIP/2.0
                    Via: SIP/2.0/TCP 10.60.30.140:5060;rport;branch=z9hG4bKfdfadfc4705f6d30aeaf1ab49c1310a3
                    From: "Karina Bolado" <sip:SIPDefault@10.120.200.20>;tag=b25a03e400a8ebb0
                    To: <sip:50528337@10.120.200.20>
                    Call-ID: 554091e18ddc18667746c51e49f9a926
                    CSeq: 740975927 INVITE
                    Contact: "Karina Bolado" <sip:SIPDefault@10.60.30.140:5060;transport=tcp>
                    Max-Forwards: 70
                    Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
                    Supported: timer,100rel
                    P-Early-Media: supported
                    Min-SE: 90
                    Session-Expires: …
0
Hi

I've been wrapping my head around this for quite some time - still accomplishing nothing but headache. Here's the setup

Lync 2013 mediation server
Lync PSTN Trunk points to Avaya Aura RED
Avaya Aura RED routes PSTN traffic to Avaya phone central

We have landline numbers in Lync, and cell phone number with a PSTN provider

Lync <-> avaya RED gateway <-> Avaya phone system

Also - we've setup so that calls to cell phone in PSTN, will also simultaneous call Lync number.

so;
1. user call my cell phone (tel: 99 99 99) from his IP-phone/cell phone
2. this will call on my cell phone, but will also trigger a call to landline on lync (tel: 22 22 22). Since we've  set this up with PSTN provider
3. this will also make Lync call (at 22 22 22)
4. so the user called cell phone at 99 99 99 and both cell phone will ring, together with Lync client.
5. if I answer the call with either cell phone or lync - the ringing will stop on the other device

so far so good.

But if this process is repeated, but this time the call is initiated from a Lync client at the internal office

1. user call my cell phone (tel: 99 99 99) from enterprise voice enabled lync client
2. this will call on my cell phone, but will also trigger a call to landline on lync (tel: 22 22 22). Since we've  set this up with PSTN provider
3. this will also make Lync call (at 22 22 22)
4. so the user called cell phone at 99 99 99 and both cell phone will ring, together with Lync client.
5. if I …
0
Hello Everyone

I'm currently in the process of migrating our current PBX system away from asterisk to Freeswitch. I am using FusionPBX on Debian 8. I am using the freeswitch webapi to originate calls. I am at the stage where when I execute the command, it rings the call centre agents phone and the customer automatically without the agent manually dialling the number. I would like the ability to manually specify a caller id number for the outbound leg of the call. At the moment it is not sending any caller ID. I can manually specify a caller ID number in the extensions page, and it works statically, however we have a need for the caller ID to be dynamic.

http://X.X.X.X:8080/webapi/originate?{click_to_call=true,origination_caller_id_name='Click to Call',origination_caller_id_number=1000,instant_ringback=true,ringback=\'%(400,200,400,450);%(400,2200,400,450)\',presence_id=630@X.X.X.X,call_direction=outbound,sip_auto_answer=true,domain_uuid=52b92yy9-7fb7-52ae-9e9e0595058bcdaa,domain_name=X.X.X.X}user/630@X.X.X.X &transfer('SOME EXTERNAL NUMBER XML X.X.X.X')

What do i need to add to this web address to get it to send a custom caller ID number to the customer outbound?

Many thanks in advance.
0
We are on Avaya IP Office 8.x (?) ....   yes we are planning on an upgrade to 10.? later this summer.  Everything had been fine until about 3 weeks ago when all of a sudden our voicemails stopped going to email.  We are not having any luck with our vendor figuring out the issue.  Have stopped and restarted voicemail services.  I don't understand a lot about Avaya IP Office as I was more familiar with Cisco Call Manager.    We have many other services that send emails through our SMTP server so I don't think that is the issue.  Can anyone give me some insight or point me in the right direction?

Thanks
0
Hi

We have a 3CX voip server that is hosted in the cloud and we have a T49G phone we need to configure to work with the 3CX server.

Now, 3CX server doesn't officially support T49G phone but they say it should work as a normal SIP phone without any provisioning.

I have tried simple SIP config by putting in the extension number and its password but the SIP registration keeps failing.

The packet capture on the 3CX server is showing 407 proxy authentication error.

Can someone help me configure Yealink T49G phone on a 3CX Voip server?

Happy to provide packet capture or any other logs you may need.
0
the system was setup with an auto attendant and 3 call groups for voice mail and after the auto attendant the caller would choose 1 , 2 or 3  and that would send you to the proper mail box and then you would hear that message. i can not figure out how to rerecord the auto attendant message or change the message for the mail boxes that auto attendant send the caller to. i can do intercom 500 from every extension and that takes me to voice mail management, but I can not figure out how to manage another mail box other than the extension I am on or how to change the recording on auto attendant. I can access both systems with my laptop.
0
Hi,

We currently have a Cisco CCM system which runs Vlan 10 as the Voice lan and Vlan 2 as the Data, the hosted platform we are going to can have vlan which is set via portal etc, it gets a normal IP and then boots into the correct lan ..... phone will connect ok but will no allow data pass through .... the Cisco phones are working fine but not sure why the hosted wont work ...... any ideas ?

Thanks
0
Hi Experts,
I installed CME (ISR 4321 IOS version 15.5), with 40 Phones 7821, 1 IP Phone 8841, another 8851, and 2 IP Conference 8831 using SIP Protocol
all of these SIP Phones work fine, but the call transfer doesn't work, I don't know if some config missing, you find below the sh run,
thank you for your help.
CME-EHM#sh run
Building configuration...

Current configuration : 10874 bytes
!
! No configuration change since last restart
!
version 15.5
service timestamps debug datetime msec
service timestamps log datetime msec
no platform punt-keepalive disable-kernel-core
!
hostname CME-EHM
!
boot-start-marker
boot-end-marker
!
vrf definition Mgmt-intf
 !
 address-family ipv4
 exit-address-family
 !
 address-family ipv6
 exit-address-family
!
card type e1 0 1
!
no aaa new-model
!
subscriber templating
multilink bundle-name authenticated
!
isdn switch-type primary-net5
!
crypto pki trustpoint TP-self-signed-246793832
 enrollment selfsigned
 subject-name cn=IOS-Self-Signed-Certificate-246793832
 revocation-check none
 rsakeypair TP-self-signed-246793832
!
crypto pki certificate chain TP-self-signed-246793832
 certificate self-signed 01
  ////////////////////////****////
!
voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 h323
  call start slow
 sip
  bind …
0
In RTMT you can view "Gateway Activity" and choose say MGCP PRI. But the thing that's weird is that it shows PRI channels per CUCM server. Given the description I would think that it would make out how many active calls are happening at the router/gateway not the UCM. Is there a way to get this broken out to you see activity per actual gateway?
0
Hello Everyone,

My name is George and I was hoping to get some assistance in configuring a Cisco 7914 expansion module.  I have been thrown into the fire with this Cisco phone system.  I have very experience with the CallManager system but I am making some headway.  The biggest problem is that my company is running very outdated versions of the software.  We are running CallManager Administration 4.1 (0.11).  I know, I know...it's no way to run a business but that decision is above my pay grade!  I am told to make it work, as I am sure many of you can understand and appreciate!

So here is what happened.  We had to do a complete network reboot and I was configuring a new phone when they took everything down.  When the network was brought back up, it took a few minutes and all but one phone came back up.  Of course, the one that did not come back up was the receptionist's.  After some troubleshooting I found that somehow, her phone was no longer showing in CallManager.  I went through the process of adding the phone again and defining it and configured.  Her phone works fine now.  The problem is with the 7914 expansion module.

The module is showing all red lights on the buttons and the display is blank.  I went to Add/Update speed dials, entered all the name and extension information and clicked on Update and Close.  I thought this was all I needed to do but I was wrong.  In the Configure Speed Dial Settings for SEPXXXXXXXXX window, it says 'Speed Dial Settings not …
0
How can you monitor DSP usage on a Cisco ISR used for MTP, transcoding, conferencing? Is there an OID which would let  us graph this?

mtp01-sc#sho sccp conn
sess_id    conn_id      stype mode     codec   sport rport ripaddr conn_id_tx

50630987   251788306    mtp   sendrecv g711u   24918 24576 172.28.72.145                                                                
50630987   251788304    mtp   sendrecv g711u   31384 18902 172.28.16.33                                                                  
50631252   50636167     mtp   sendrecv g711u   19210 17128 172.17.254.1                                                                  
50631252   50636166     mtp   sendrecv g711u   18332 50004 172.16.24.142    
Total number of active session(s) 11, and connection(s) 22
0

IP TelephonySponsored by Jamf Now

IP telephony (Internet Protocol telephony) is a general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). Using the Internet, calls travel as packets of data on shared lines, avoiding the tolls of the PSTN. The challenge in IP telephony is to deliver the voice, fax, or video packets in a dependable flow to the user.