I moved our company to Freeswitch keeping Asterisk only as a Digium PSTN gateway (didn't want to mess with OpenZAP at this point) Dave wrote:
I used to install the asterisk on centos 5 box, and has little experience with sipX. In this weekend, I installed the freeswitch on the same box, and have a test drive. Here it is couple of my questions.
1. it seems that both asterisk and freeswitch will use the same port (5060, I guess), so if I like to start the freeswitch, I must kill the asterisk, can I change the default port to another port.
I did the reverse, changed asterisk to run on port 7000. But you can- check conf/sip_profiles/internal.xml (or external.xml)
2. follow the first question, can I use freeswitch along with asterisk, if I can, what may need to be taken care.
Nothing really. Make a SIP trunk from Asterisk to FS and vice versa.
3. I saw the integration guide of "sipX and freeswitch", is this one work with the latest FS release.
4. If I like to use the FS library as the telephony layer for some other application, do you have the demo or samples, the closet doc I can found from your web site is the "embedded application guide".
5. currently, I am using the vbuzzer as the voip service provider, like to know if there is anyway to keep the same VSP, but switch from asterisk to freeswitch, if can, is there a "how to" guide, if can not with vbuzzer, is there any VSP who claim the support of the FS.
Any SIP provider would work- most likely better than Asterisk as FS implements SIP completely (including CNG, etc)
Thanks in advance.
Dave
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