| Polycom VoIP Support Forum Technical support, how-to guides, troubleshooting, and general assistance for the Polycom line of VoIP products. |  | 
August 1st, 2006, 03:56 PM
| | Member | | Join Date: Jun 2006
Posts: 31
| | BYOD on Polycom Line2 Hi,
I am using openSER, asterisk connected to one PSTN line, and am considering adding a BYOD sip provider and was wondering if anyone has done this where i can relay calls thru their services, especially w.r.t. long distance. I plan on putting their sip server url/uri data in line 2, however, all other lines use my sip server domain. I am a newbie still to the voip industry, so if I am way off, please let me know. I have not purchased it yet, and need to know if this can be done.
Thanks in advance. | 
August 2nd, 2006, 10:37 PM
| | Member | | Join Date: Sep 2003
Posts: 56
| | RE: BYOD on Polycom Line2 You can set up the Polycom with two separate lines; one would register with your Asterisk, the other with your BYOD provider. I have not used an OpenSER server with Polycom, but if this configuration works with your existing hardware, it will work with the Polycom.
Vick | 
August 9th, 2006, 07:27 PM
| | Junior Member | | Join Date: May 2006
Posts: 15
| | Re: RE: BYOD on Polycom Line2 Quote: |
Originally Posted by Vick You can set up the Polycom with two separate lines; one would register with your Asterisk, the other with your BYOD provider. I have not used an OpenSER server with Polycom, but if this configuration works with your existing hardware, it will work with the Polycom.
Vick |
Hmmm, I haven't yet been able to get this working with an IP430. I'm also running openser and I wanted the second line to register with
an external Asterix server. I have this working with a Linksys SPA942,works in/out-bound behind many different NATs for both lines.
But the Polycom tries to register with my openser box for both accounts,
sending my openser box REGISTER sip:asterixIP:5060 instead of sending that to the asterix box.
I'm still poking at this, as time allows...but my current assessment is that individual phone users should avoid Polycom.
-mark | 
August 15th, 2006, 02:27 PM
| | Member | | Join Date: Jun 2006
Posts: 31
| | Thanks for all of the replies,
I, too, have tried repeatedly in the past to register to my asterisk server but have had no luck. I was only able to add other subscriber lines to openSER. However, I now have free DIDs coming into this Polycom, so i am somewhat satisfied that openSER happily accepted. I am glad to learn that all SIP phones do not behave like this and that others do have the ability to register different lines to other providers or your own particular SIP router or B2BUA . I guess I should have done my research more thoroughly before purchasing the Polycom.
Another thing that I learned, but not until after the purchase, was that the Polycom 500 does not utilize the *86 to call into my voicemail messages, while the Polycom 500SIP does. Does anyone know if there is anyway to get the Polycom 500 to utilize this feature?
I also have one Polycom 500SIP that makes calls outbound but stopped accepting calls in, such as calls inbound to it only accept voicemail now. Even when attempting to call it by dialing the extension, my * server prints:
Executing Dial("Local/22222@default-39a5,2", "SIP/22222|10|r") in new stack
-- Called 22222
-- Local/22222@default-39a5,1 is ringing [ONLY RINGS ONCE BUT ON PICKUP NOTHING THERE - GOES RIGHT TO VOICEMAIL]
-- Got SIP response 482 "Loop Detected" back from xx.xx.xx.xx
-- Now forwarding Local/22222@default-39a5,2 to 'Local/22222@default' (thanks to SIP/22222-09453d58)
== Spawn extension (default, 22222, 1) exited non-zero on 'Local/22222@default-90fc,2'
== Spawn extension (default, 22222, 1) exited non-zero on 'Local/22222@default-e9a5,2'
== Spawn extension (default, 22222, 1) exited non-zero on 'Local/22222@default-3e99,2'
==
... over and over again. So, basically I plan to go to the site where the phone is located and reset it hoping that will help.
TIA | 
August 15th, 2006, 03:06 PM
| | Member | | Join Date: Jun 2006
Posts: 31
| | Hi all,
Well i did get the *86 function to retrieve voicemail from asterisk to work by putting this in my phoneXXXXX.cfg file:
msg.mwi.1.callBackMode="contact" msg.mwi.1.callBack="*86"
just in case someone else may run into something simliiar. But I still do not know why one of the Polycom 500SIP's is acting like i stated in a prior post. Still would like help on that one, nor have I went to the site to reset it yet.
TIA | 
August 17th, 2006, 01:04 AM
| | Member | | Join Date: Jun 2006
Posts: 31
| | Hi all,
no need to continue this thread, it has already been taken care of.
TIA. Cheers! |  | | Thread Tools | | | | Display Modes | Rate This Thread | Linear Mode | |
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