I have a setup of two sites as follows:
|||| SITE A |||| |||| SITE B ||||
legacy pabx(ericcson --------cisco 2911 ---------------------------------WAN-------------------------------cisco 2911-----------legacy Pabx (ericcson) -----------PSTN.
The links between the 2911 are all SIP dial-peers, and both cisco 2911 are connected to the legacy pabx via an ISDN E1. The cisco 2911s act also CMEs and host some cisco SCCP together with some SIP clients. Only Site B has a link to the PSTN>
I have a problem when a call is forwarded between sites. For example, a PSTN user dial in our PSTN number and someone on the legacy pabx answers the call. He then decides to forward the call to site B. The call is transferred successfully between sites and the end user in site A can take the call normally. However the end user in site A does not see the call coming from the orginal caller (PSTN user) but see the call coming from the user in site B.
I have enabled ISDN support on both gateways. Basically:
voice service voip
signaling forward unconditional
interface serial 0/0/0:15
isdn supp-service name calling
The same problem appears also when a user (user1) from the legacy pabx at site B contacts another legacy pabx user (user2) from site B as well, who then forwards the call to User3 on Site A. User 3 does not see that User1 is calling but sees User2.
Does anyone have an insite on this problem? Another point i would like to mention is the fact that this problem was not there when the dial-peers between to two link were H323. I know that H323 is based on qsig and therefore more interoperable with ISDN, which might explain why the problem did not occur then.
Thank you in advance for your help.
could you send a copy of
debug ccsip messagers
debug isdn q931
along with a copy of the gateways running config?
Thank you for your reply and sorry for the delay.
Configs and debugs attached.
From the sip messages i noticed that the Site A gateway is sending an INFO sip message to SITE B Gateway after the call transfer is performed.
From the debug:
Caller 6852 ----> Calling 7192 -----> diverts to 7092. in this case user with extension 6852 continues to see 7192 and the calling number and is not updated to 7092. in this case caller 7092 sees correctly 6852 as the caller.
PS: you will see 6852 as 25646852 and 7192 as 287192
Could uyou upload site A debugs? In the site B debugs the called number in the SIP messaging never changes from 7192
I have added the debug on site A. There is reference to 7092 in the debugs in the SIP info message ( extract ** CORTIS MAUR I 7092 ***). The call was oringinated from a traditional pabx on site B, and both destinations 7192, and 7092 reside in another traditional pabx on site A.
Another point to note is that this problem did not occur when then links between the cisco VG were using an h323 trunk, since h323 is based on qsig.
Its been a while since i heard back from you. After the call is being transferred from extension 7192, it should send an update also to extension 6853 that now the calling number has changed from 7192 to 7092, but this is not happening and extension 6853 still sees 7192 as the called number.
Any help would be greatly appreciated.
I am unable to decode your debugs due to some strange characters (I believe file encoding problem).
anyways, what I noticed that no REFER SIP messages are present in your debugs which should be there as we are talking about call transfer over SIP trunk. Please make sure that SIP Refer supplementery services is enabled in your router.
Once we get the refer message, we need to look at remote-id header to see whats the displayed name.
I have expalined mapping of display between ISDN and different protocols in my blog. I am sharing the link here and hope it can help.
Hi Mohammed, Long time. I sent you a PM a while ago. Did you see it?
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"There is a wideness in God's mercy Like the wideness of the sea.There's a kindness in His justice Which is more than liberty"
Thanks for asking ... In fact I went out of forum for last two weeks since I was preparing for my VMware VCP exam ... Yesterday only I completed it and now back to track but in two forums (Cisco and VMware)
I will check my private messages and will respond soon.
Thank you for your post and link.
As you can see in my first post the extensions I used are 6852-->7192-->7092.
Now extension 6852 is on an Ericcson Pabx at Site B, whilst the other extensions are both on the ericcson pabx at Site A. The problem is that when a user in site A transfers a call no update / sip refer messages are sent back to pabx at site B. However this problem only popped up when the trunk between the two 2921s was changed from h323 to SIP.
SIP supplementary services are enabled on the router.