Hello Everyone,
I apologize in advance for my lack of knowledge in this arena. I am trying to learn but for some reason I must be missing something, or a few things. I’ll explain what I have to work with and what my end goal is. If any config files are needed, I’ll be happy to post.
I have 2 PSTN lines that come in via a cable ISP and terminate in the cable modem. I run a line from each RJ-11 in the cable modem to a Cisco FXO card installed in a Cisco 3745 router. The router IOS version is 12.4-4(T). The router is acting as my Voice Gateway. 2 Cisco IP Phones, a 7960G and a 7940G, both loaded with SIP Firmware 8-12-00. The phones both plug into a Catalyst 3550 (Layer 3 Switch) on their own VLAN. The phones are assigned static IP addresses. Basic Network info:
Subnet 172.16.1.0 / 255.255.255.240
Router 172.16.1.1
Trixbox (2.6.2.3) = 172.16.1.3, this is also my TFTP server. I am not using DHCP anywhere in the mix.
What I want to do is to have both PSTN lines going into Trixbox, so that Trixbox will “manage” both PSTN numbers. I’d like to assign private extensions to the IP phones and have it so that either phone can receive a call coming in from either PSTN line, as well as be able to make calls outbound on either PSTN line. Once this is configured, I would also like to add the capability for softphones to connect via SIP. I’ve been going at this for over a month now with varying levels of success. Each time I think something is going right, I find out it isn’t really. Is what I want to do possible? I would assume it is with an IP PBX. I would like each phone to be able to dial 7 digit dialing for local numbers (as my PSTN doesn’t support 10 digit dialing) as well as be able to dial long distance (1+10 digits). Dialing a prefix for an outside line would be preferable, 9 for instance. This way Ideally I can dial a private extension directly, whether it be local in the building or eventually a softphone from another site using SIP. The ability to transfer calls from one extension to another. I believe is pretty basic stuff. I have googled many things, as well as perused much information on this site.
To put it bluntly, I am confused more than anything else at this point, and now starting to wonder if what I am trying to accomplish is even possible? I can’t help feel that I am missing something. Perhaps something small, or maybe I am not grasping a key concept somewhere. I am obviousely using a SIP trunk to connect from the FreePBX (Trixbox) to the Cisco 3745 Router (Voice Gateway). I’m comfortable working with editing config files as well as working with IOS from the CLI. Any guidance would be very much appreciated. Thank you everybody in advance.
I have a system that does similar to what you describe:
I have two PSTN lines (that are actually from our main PBX) and then I have four SIP phones. Mine are Grandstream, but I don’t think that matters here. When we call the PBX, you get to the IVR and then you can either punch in the extension if you know it or select it by number from the IVR.
My SIP phone users can dial out, they have two do two 9’s, due to being inside another PBX and my dial pattern in the outbound route and dial rules in the trunk (which are confusing) had to take this into consideration.
I also installed a couple of 3CX SIP clients on a couple of PCs.
Works pretty good. What problems are you having? Error messages? Where are you seeing them? What versions of stuff do you have?
I now have Trixbox dialing out as well as receiving inbound calls. However, it is bittersweet. Call quality is ok, but lots of echo, especially when using conferencing functionality. My ultimate goal is to configure a Cisco CallManager system. Due to the massive complexity involved with a CallManager (not CallManager Express) system, I wanted to get basic functionality going on a system like Trixbox first. I referenced some data about configuring a Cisco VG200 router and applied it in a similar fashion to my 3745. Hey, some success is better than beating my head into a brick wall and tearing more of my hair out, lol. How is 3CX by the way? I looked into it briefly, but the free version only allows for 3 extensions. I didn’t want to go through the setting up of a Windows Server 2003 system for only 3 max extensions. I appreciate your quick response however. Chances are as I run into more issues, which I am sure will happen, I will be posting here again. When you ask “What versions of stuff I have”, would you please be a little more specific? Are you referring to IOS? Phones? Phone firmware? OS? Clarification would be helpful. Once again, thank you for your q
First of all, I am rookie at this. The system I am configuring is intended for a home environment and we use it to connect to our small Metal Shop next door.
The kind of version info that is needed is things like the status command results. You should read this post: http://www.freepbx.org/forum/freepbx/installation/so-you-have-a-problem-and-want-help
If you have versions that are way off what I have, my experience may not help, but some of the gurus might know the answer, because they have seen each upgrade evolve.
This is what I post whenever I ask a question:
Operating system CentOS Linux 5.2
FreePBX Base Version: 2.6.0.0
Asterisk (Ver. 1.4.21.2):
- Asterisk Source Version : 1.4.21.2
- Zaptel Source Version : 1.4.12.1
- Libpri Source Version : 1.4.7
- Addons Source Version : 1.4.7
MySQL client version: 5.0.45
phpMyAdmin - 2.11.0
Server version: 5.0.45
Protocol version: 10
Webmin version 1.420
Apache version 2.2.3
Flash Operator Panel op_server.pl version 0.30
Hardware:
CPU & Companion Chips VIA C7 1.5GHz + CN700
Memory 1GB DDRII (SO-DIMM)
Digium Wildcard 4-port TDM PCI TDM410P
Grandstream phones: Product Model: GXP2000 (HW1.2B)
Software Version: Program-- 1.1.6.16 Bootloader-- 1.1.6.5
So, can you be a little more specific?
I am not having echo problems, but I have read about them.
If the people you are talking to sometimes complain that they hear their own voices echoed back to them, try changing the FXS Port Output Gain to a slightly lower value.
For example, if it is currently set to -3, try changing it to -6 or -9. This will lower the volume that you hear in your telephone’s receiver so if you set this too low, you’ll have difficulty hearing people, which is why we suggest making only small changes. Note that if you have a telephone with a receiver volume control and you have this set on “high”, try turning it down to the “normal” setting and see if that fixes the problem first, before you change this value on the adapter.
• If you sometimes hear your own voice echoed back to you, try changing the FXS Port Input Gain to a slightly lower value. For example, if it is currently set to -3, try changing it to -6 or -9. This will lower the volume of your speech going out, so if you set this too low, those you call will have difficulty hearing you, and/or touch tones you enter on your phone’s keypad will not be recognized. Unless you are having severe problems with your voice echoing back to you, we don’t suggest setting this below about -6.
I was pretty impressed with 3CX’s free client. It looks nice. Works great. Can’t speak to quality,because I have a very cheap ear piece.
I also played with their free Windows PBX server software. I was likewise impressed. Can’t beat free. We would have had to purchase a different piece of gear to do what we needed, and I got the FreePBX system hardware problems resolved first, so we are still using the FreePBX.