Call Manager with H323 gateway and SIP Trunk

Unanswered Question
May 27th, 2010
User Badges:



/* Style Definitions */ table.MsoNormalTable {mso-style-name:"Table Normal"; mso-tstyle-rowband-size:0; mso-tstyle-colband-size:0; mso-style-noshow:yes; mso-style-priority:99; mso-style-qformat:yes; mso-style-parent:""; mso-padding-alt:0cm 5.4pt 0cm 5.4pt; mso-para-margin-top:0cm; mso-para-margin-right:0cm; mso-para-margin-bottom:10.0pt; mso-para-margin-left:0cm; line-height:115%; mso-pagination:widow-orphan; font-size:11.0pt; font-family:"Calibri","sans-serif"; mso-ascii-font-family:Calibri; mso-ascii-theme-font:minor-latin; mso-fareast-font-family:"Times New Roman"; mso-fareast-theme-font:minor-fareast; mso-hansi-font-family:Calibri; mso-hansi-theme-font:minor-latin; mso-bidi-font-family:"Times New Roman"; mso-bidi-theme-font:minor-bidi;}

Hi Guys,


As the title suggests, I have CallManager 5.1.2 with a h323 gateway on a remote site.  This site has recently moved and they have gone with an SIP tunk provider terminated on their SRST gateway.

When the site moved, the connectivity back to the HQ was severed so the phones operated in SRST with the SIP trunk and all worked great.  Since then, they have built a VPN back to HQ.  When the phones register with CM, calls in have stopped working.

I call in on the SIP trunk, the phone rings, when I answer nothing happens and the caller continues to hear ringback, then both disconnect.

Logically, the way I see it is that the SIP call comes in on the IOS gateway and hits the incoming SIP dial-peer, it gets bumped over to the CCM via another dial-peer and CCM rings the phone in question.  This is all signalling.  What should then happen is CCM connects the phone and the SIP call and then drops out of the loop so the phone is talking directly with it’s local gateway that terminates the sip trunk.


SIP trunks are not my area of expertise so I'm really lost on this one so any help would be appreciated.  I've seen some mention of MTP's but I don't see how it fits in this case.  Or maybe that's what my problem is!!

Thanks in advance,


Neil

  • 1
  • 2
  • 3
  • 4
  • 5
Overall Rating: 0 (0 ratings)
Loading.
Rick Arps Thu, 05/27/2010 - 05:54
User Badges:
  • Bronze, 100 points or more

What are the codecs involved?  Sounds like you might need a transcoder in there.


An MTP is required for SIP as well, but your gateway should act as your MTP in this situation.


Rick

neilobrien Thu, 05/27/2010 - 06:26
User Badges:

Hi Rick, thanks for the reply.


The SIP trunk will negotiate either G711 or G729 and there's a voice class applied to the dialpeers with both codes.  It's not hardcoded to one or the other.  But keep in mind that it works fine when Call Manager is out of the loop (in SRST).  My understanding is that Call Maanger should only be signaling when the phones are not in SRST.


Thanks,


Neil

neilobrien Fri, 05/28/2010 - 05:26
User Badges:

Sorry to bump this guys.


I'm wondering if someone would know if I need CUBE in this case.  It keeps coming up in my searching, but I'm not really sure if my setup in this case requires it or if this is actually what my problem is.


Maybe somone could throw thir opinion into the mix.


thanks,


Neil

Actions

This Discussion