Hi all,
Next week I’ll be installing my first Asterisk based system at one of the companies I work for. I have all of my phones, extensions, features, voicemail and all of that working well.
I have one final hurdle to jump over to get this installation done- connect it to a Comcast SIP Trunk.
At first I thought I had it under control when I saw they sent me a config guide, but unfortunately it was for Free PBX, and I’m just using Asterisk alone. I’ve tried to piece together what I think the proper config for the trunk would be by looking at the Free PBX instructions but I don’t have much confidence.
I’m curious if anyone else has recently configured an Asterisk system with a Comcast SIP Trunk and would be willing to share the basic outline of the trunk config they used, omitting specific IP addresses and stuff.
Hopefully someone can help.
Thanks in advance.
If you have the FreePBX instructions you can easly implement it on plain Asterisk it is pretty much the same thing if you what you are doing anyway you can post the instructions here and we can help you. Just hide any sensitive information
Thanks for the reply. I figured that the config is almost the same but I wanted to make sure I understand everything. I’ve been learning Asterisk for about the last year, but this is my first actual implementation. Here is a link to the guide they provided me with:
I believe most of the pertinent info is on pages 8-12.
Thank you again. Please let me know if you need me to provide anything else.
I havent read the whole document but start creating a SIP peer using this configuration , codec selection should be one by line instead of ulaw&alaw
The one line format for codecs is perfectly OK. However, you must disallow all first, otherwise you will be trying to add codecs when all are a already added, whichever syntax you use.
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Thanks. Based on what I saw in the guide and what you both have said, this is what I would start with:
[Comcast]
host=10.64.4.71
type=peer
disallow=all
allow=g711&ulaw
session-minse=930
session-expires=1920
session-timers=originate
dtmfmode=rfc2833
They do talk about trustrpid and sendrpid as well as context on page 12. Will those need to be defined in the above config as well?
context needs to defined unless you want incoming calls to be handled in the default one.
trustrpid may be needed, if they don’t use the From header for caller ID, or they do connected line updates.
sendrpid may be needed if they allow you to specify the caller ID and don’t use the From header for this.
They are using a private network. This may be incompatible with directmedia, which I believe is now the default.
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Thanks David. I think I have the info I need.
I’ll let you guys know how it goes and if I need any further help.
Thanks again.
I said I’d update so I will. Overall this went pretty well, but I ended up having to define a couple more things; an outbound proxy in sip.conf, as well as set the caller id number for my outbound route in extensions.conf. I was sure I’d need something like the caller ID parameter prior to the cutover, I just wasn’t sure what would need to be added. A quick google search and I had the info I needed. Adding the line to set the caller ID would ensure all outbound calls showed as coming from the building’s main number. As for the ouboundproxy line in sip.conf, I set that to the inside/LAN IP of Comcast’s sip gateway. I believe this is required due to the sip gateway having that inside interface on our network.
It has been working for over a month now; a ton of calls have been made/received and the users seem happy with it.
*edit/last minute thought- it was also necessary to separate the two codec lines in sip.conf. It would not work with them on the same line.
Thanks to all who helped!
