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  • Multiple vlans access to shared pbx system

    - by Matt
    I'm new to networking and was looking for some assistance. First off I'm using packet tracer to diagram my scenario as I will be receiving my equipment next week to deploy. Hardware to be used: 2 catalyst 3560 switches all connect to a sonic wall router I have two companies that work in the same office space. I need to keep these companies separate on their own vlan. They will however need to share the phone system. (Packet tracer file uploaded to give those who have the time to see what I put together.) http://dl.dropbox.com/u/86234623/network%20build.pkt Here is my current test scenario: on switch 0 I have: company A on vlan 2 computers 172.16.1.100 and 101 255.255.0.0 FA0/10 FA0/11 company B on vlan 3 computers 172.16.2.102, 255.255.0.0 FA0/12 PBX on a trunk port 172.16.0.5, 255.255.0.0 FA0/5 trunk port on FA0/1 to connect the switches on switch 1 I have: company A on vlan 2 computers 172.16.1.102, 255.255.0.0 company B on vlan 3 computers 172.16.2.100 and 101, 255.255.0.0 trunk port on FA0/1 to connect the switches I can ping the respective computers on the same vlan but cant ping company A to B which is what I want. However neither company can talk (ping) the PBX. Here are the commands I used to configure what I have: switch 0 en conf t vlan 2 name A vlan 3 name B int fa0/10 switchport mode access switchport access vlan 2 int fa0/11 switchport mode access switchport access vlan 2 int fa0/12 switchport mode access switchport access vlan 3 int fa0/5 switchport trunk encapsulation dot1q switchport mode trunk switchport trunk allowed vlan 1-3 int fa0/1 (to connect the switches) switchport trunk encapsulation dot1q switchport mode trunk switchport trunk allowed vlan 1-3 Switch 1 en conf t vlan 2 name A vlan 3 name B int fa0/10 switchport mode access switchport access vlan 3 int fa0/11 switchport mode access switchport access vlan 3 int fa0/12 switchport mode access switchport access vlan 2 int fa0/1 (to connect the switches) switchport trunk encapsulation dot1q switchport mode trunk switchport trunk allowed vlan 1-3

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  • Moving from Analogue PBX to digital VoIP?

    - by saint
    I don't even know if this belongs here?. If not, do let me know. So we have an analogue Alkatel PABX system in our little office. We have extensions, direct lines and PBX lines. We are trying to move to a more digital/flexible way of handling the phones and I've heard good things about FreeSwitch. I have zero knowledge about it. My biggest question is how would one handle existing phone lines with such a system. Surely there must be a way to make and receive calls from outside. Just a help in the right direction would be fine. Thanks.

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  • What service do you use for music on hold?

    - by Russ Warren
    This may not be a sysadmin question for some, but it is definitely a hurdle I have to jump as the sysadmin for my company. We recently rolled-out a company wide VoiP system (Switchvox, to be exact) that has come preloaded with some royalty-free music on hold. Our customers have been complaining that the music on hold sounds like "funeral music." This may be the case (although I wouldn't want it played at my funeral), but it is all we have and we aren't willing to be sued over using music that isn't properly licensed. So, that brings me to the question asked in the title -- what and/or how do you provide decent music on hold? I'm assuming many people here use a PBX that allows customized music, so this has to apply to many of you. We've been looking at some sites that allow you to download royalty-free music for a one-time fee, but the music seems...lame. Something like a one-year subscription from ibaudio.com seems to be the best bet so far. Have you been able to discover something a little more mainstream for a decent licensing fee? Thank you. EDIT: Our PBX allows the playback of MP3 and OGG files, but does not allow streaming of a live audio source, Internet-based or otherwise. It also does not allow the use of a "line-in" source such as a CD player or radio. Don't let this stop you from sharing your setup, though. I'm interested in hearing what everyone uses!

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  • Phone system for 55 size firm

    - by Hemal
    Hello, We are in the process of upgrading our 11 years old PBX system and looking for options like Panasonic, Avaya, Toshiba etc. We welcome any suggestions/features/product models to look for 55+ size firm... Thank you in advance for any replies...

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  • What is SIP trunking?

    - by hypnocode
    Can someone explain to me in plain English what SIP trunking is, please? I've read about it on Google, but I don't really grasp it yet. Does it allow a VoIP call to be placed outside of the LAN? So if you had Asterisk setup as the PBX, then IP calls could be made outside of the network? Am I close or am I just saying stupid words?

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  • Can i have a Asterisk IP PBX Server Behind ISA 2000

    - by garyb32234234
    Hello Is it a simple procedure to configure ISA Server 2000 to allow an Asterisk IPPBX connect to SIP provider. On asterisk forums they say the ISA has difficulties handling SIP, softphones that i have installed behind the firewall work fine with the provider when the firewall client is installed on the workstation. With asterisk being a linux based system this will not be an option. Is the config a matter setting up port forwarding, is this a more complicated task on ISA server than just selecting the ports i need and then the ip of the internal machine i want to forward them to? UPDATE: I dont think this is possible from what ive researched Regards Gary

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  • Considerations for a business looking to transition from PSTN to IP Telephony

    - by Bryce Thomas
    Full disclosure - This is related to a homework assignment question. I am not asking you to do my work for me, I am merely looking for some pointers and considerations to direct me in my further research. I have an assignment I'm working on where I've been given a scenario where a business wants to look into transitioning to using "Internet Telephone" as opposed to a traditional PSTN/PBX system and I need to write a report on it. I'm after some high level pointers from people, especially anyone that has been involved in a real life transition of this nature, on what some of the most important considerations are. These can be financial considerations, initial setup considerations, ongoing administrative considerations, quality of service considerations or anything else that is pertinent to performing such a transition.

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  • Using a PC + headset as a telephone without VOIP

    - by user76782
    I'm trying to find a way to realize a decentralized callcenter, so that the callcenter agents can talk from their home office with just a PC + headset. The big challange is that some of the agents have very low bandwith and the quality with VOIP is too bad. So my question is: What other solutions are possible when VOIP is not a option? What exactly do I need to do if I try to achieve this with for example Landline/PBX or GSM? (e.g. Which software do I need to install? Which device do I need?)

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  • Provisioning SIP Phones over the internet

    - by Jorge Fernandez
    I have a few SIP Phones that are located of site and connect to my PBX over the internet to make calls. For some reason one of these phones has become unprovisioned. In my office phones get provisioned by the server via TFTP. The ones that I have off site I pre-provisioned manually before I sent them off-site (I'm in Florida the phone is in New Jersey). Whats the best way to provision these over the internet? TFTP is very insecure. Sending the plain text profiles with the SIP Account and Password over the internet is out of the question. The phones have been off-site for about 6 months without any issues. Im using Trixbox and Cisco 7940 Phones.

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  • Call connects through Emapthy, but no voice

    - by Arthur
    I am trying to make calls using Empathy on Debian.The call rings and connects, but I can't hear voice, or answering machines. I had a similar problem a while back that was caused by my PBX not being compatible with a protocol. The protocol issue is fixed now and Linphone works fine on a different pc. I tried using Ekiga, on the problem computer, and it works fine.The problem seems to be with the settings on Empathy. I need to get this going. Any help would be greatly appreciated.

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  • How can i automatically backup Nortel PBX config into subversion?

    - by user14604
    PBX: Nortel Opt 81c Connection: Analog lines through Procomm or Reflections We are thinking about backing up the Nortel PBX config and checking them into the subversion. I am wondering if anyone has found the a way to automate the pulling of the configs off the Nortel PBX. Configs would be programming listing in plain text that we can diff in subversion to see what has been changed. An example config would be to go into ld 21 and printing the RDBs.

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  • Router gets disconnected once I terminate my SIP application

    - by TacB0sS
    Hey, Here is an interesting one, I have a SIP VoIP application which is able to register to the PBX server, and I can invite and see the user call on the callee end receiving an Invite, and on the caller end I see the Ringing response... now here is interesting part, if I close my application with out any notification to the server my router disconnects and restart, after a short while (30 - 150 sec). I could fix that if I would complete the ACK BYE process, but I'm just wondering why does my router hangs up? any ideas? My Router is TNN-Siemens SL2-141, thought this might matter Update: this is what I found: SIP ALG allows two or more simultaneous VoIP phone calls made by VoIP clients through this router. which means that if I disable it I would not be able to do the testing I'm trying so badly to do, and since I don't have access to another router, I must handle it with the bug then... I can say that this never happened to me with one user connecting, but then again I didn't have anyone to invite then, I received from the SIP UAS 503 when I tried to invite an imaginary user. This bug only occur after I connected the second SIP UAC and invited it and closed the application. Adam.

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  • What is SIP trunking?

    - by hypnocode
    Can someone explain to me in plain English what SIP trunking is, please? I've read about it on Google, but I don't really grasp it yet. Does it allow a VoIP call to be placed outside of the LAN? So if you had Asterisk setup as the PBX, then IP calls could be made outside of the network? Am I close or am I just saying stupid words?

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  • Commercial SIP Trunking in mainland China [closed]

    - by Patrick
    Is there any regulation preventing the use/sale of SIP trunks in mainland China? I've set up and used commercial-grade SIP trunks in places where previously we would have used ISDN T1/E1 connections. Here in Shanghai I'm looking for a similar service, and while E1 30B+D services are readily available, every telecoms company we speak with says that SIP trunking is not available in China with re-sellers of both China Telecom and China Unicom. But no one seems to know why. It seems logical to me that SIP trunks are cheaper to operate than ISDN services given that the first mile transit can be run over already-existing Internet infrastructure, and SIP signaling reduces the amount of configuration required by subscribers which is why it appeals to me. As such I've come to expect SIP services to be available in modern markets, and I've used them in quite a few countries. For example, one place I know it's not possible is in India. Government regulations in India make it illegal to provide PSTN service using VoIP. (Citations: 1, 2). However it seems this may be changing. Perhaps China has something similar.

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  • Touch tone/DTMF "translator"?

    - by IVR Avenger
    Hi, all. Does anyone know of a device that can be plugged into a telephone handset that will display the value of any telephone keys pressed by someone on the other end? So, if I'm on the phone and the other party dials "1234", I'd look down at the device plugged into my handset, and see "1234." Thanks! IVR Avenger

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  • IP telephony open source systems

    - by danke
    I'm trying to pick an IP telephony technology to learn. I heard of Asterisk, trixbox, freePBX, and my head was already spinning being not sure what to learn. Then I came across this article listing some more like Kamailio, Yate, CallWeaver, FreeSWITCH, SipXecs and now my head REALLY is spinning http://www.cio.com.au/article/323016/five_open_source_ip_telephony_projects_watch . Can someone give me a run down of how all these technologies tie together? What is the trend now, because I'd like to start learning. Note: Anyone please re-tag this question if you know better, because I'm new to this field and not sure about the best tags.

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  • Is there a SIP provider in the UK which provides the P-Asserted-Identity header?

    - by nbolton
    In the US, Flowroute (low cost SIP trunking provider) provides P-Asserted-Identity in the SIP invite request header (example screenshots). It also allows you to set the caller ID for outgoing calls, for example by using the follow in extensions.conf for Asterisk: exten => id,n,Set(CALLERID(all)=123) However, in the UK, I've tried a couple of SIP providers and none of them let me do either of those things (see P-Asserted-Identity or set the caller-ID). Is this because of some sort of restriction in the UK phone networks, or is it only available to really expensive SIP trunking providers?

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  • How to display incoming trunk name at called device end with Asterisk?

    - by netmano
    We have an Asterisk with FreePBX, and using Grandstream and Panasonic VoIP phones. Now when an incoming call rings on the phones only displays "Line 1" (as it is configured at account 1 on phone) and the caller number. We would need to see the trunk name where the call comes. Please, suggest how can be achieved. I was wondering about rewriting the callerID to extend by the trunk name (like "LP - +12345").

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  • Shoretel Upgrade Path

    - by Brian
    I currently have a Shoretel Server running Server 2003 x32 as a virtual machine paired with a ShoreGear 90 switch and another unused switch of the same model being reserved for manual failover. I am getting the software mailed to me from my partner, but with limited support since the server is in a relatively remote area. I am tempted to upgrade the OS at the same time as performing the upgrade, but want to know if there are any horror stories or advice I should know about before diving in. I'm upgrading from Shoretel 9.2 Build. I will be upgrading first to version 10.1 then finally to 11.1. The system has been bullet proof since it was installed and we are upgrading mainly to get a client that is a little more modern. My question boils down to: Should I even bother with an OS upgrade or even possibly a fresh install of Windows with an install of Shoretel 11.1 and just transfer the configuration? Should I just stay with Server 2003 since it is supported in my target version of Shoretel and the upgrade will be more than enough to keep me busy as a novice?

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  • Unregister SIP UAC message

    - by TacB0sS
    Hi, I've looked so much on the internet, but I could not find a any SIP Unregister example, and when I search RFC 3261,3665 the word does not even appear, perhaps I'm searching for the wrong phrase. I manage to understand the part of setting the expires to zero, but it still does not work and I could not find documentation about how a formal unregister should be. Does anyone knows how to compose an Unregister SIP Request? or what should I search for it? Thanks in advance, Adam Zehavi.

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  • router gets disconnected once I terminate my SIP application

    - by TacB0sS
    Hey, Here is an interesting one, I have a SIP VoIP application which is able to register to the PBX server, and I can invite and see the user call on the callee end receiving an Invite, and on the caller end I see the Ringing response... now here is interesting part, if I close my application with out any notification to the server my router disconnects and restart, after a short while (30 - 150 sec). I could fix that if I would complete the ACK BYE process, but I'm just wondering why does my router hangs up? any ideas?

    Read the article

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