Thanks Omar. I figured this out in the end. It was indeed the watchdog interval. I had to make that smaller than the wait node time before the chat was re-queued ... View more

Thank you guys. Much appreciated ... View more

Hi Omar, Based on your suggestions above. I am trying to use the wait node to reroute the call to another skill group, however ICM just aborts the call after the wait node timesout. It does not reroute to another skill group. What could be wrong? Thanks as always for the help ... View more

Hi Mike, Thank you for this. Would you mind sharing the toastr concept with me. I assume this is a custom script you wrote. Would you mind PM me the code? ... View more

Hi Apologies if I was not clear earlier. You dont need to create dial-peers for phones with extensions because there dial-peers will be automatically created. But you need dial-peers for any other number that is not registered automatically. Your hunt pilot number will not be registered automatically in SRST. You will need to define it for it to be available in SRST and hence you need to create dial-peers to route to it.   ... View more

In your configuration, I don't see the second dial peer to be used during SRST.. ... View more

Please provide the full call flow. Where does the gateway send the call to? CUCM? or is this a CCME deployment.. What is the protocol used in the call flow. You need to see the full picture to accurately diagnose what the issue is. When we need the full flow and protocols used we can request additional logs to help point us in the right direction ... View more

Typically when a REGISTER message is sent with Expires = 0, this is usually indicative of a keep-alive message sent to the backup registrar. It is possible that when the gateway performed DNS lookup on your registrar more than one hosts were returned and its trying to register to primary and send keep-alive to the secondary. To confirm this, I will need the following:   1. Turn on debug ccsip info, debug ccsip message, debug ccsip feature registration 2. initiate a new register request ( you can do this by doing the following: sip-ua no  registrar dns:XXX.XXX.XXX expires 60 registrar dns:XXX.XXX.XXX expires 60   Collect the logs and attach here   ... View more

Is the hunt pilot number registered in SRST? I doubt it, hence why you will need the dial-peer. In SRST the gateway will automatically create dial-peers for your phones with extension on them but for any other number like a hunt pilot you have to tell the gateway where to send the calls to. ... View more

Cofi, Based on your logs here is what I see:   leg 1: PSTN------>CUBE: isbco4ktp4wd9drsqtko4tpd4vwsqitvtotv@SoftX3000 leg 2: CUBE--------->.CUCM: 801BB756-764211E9-93549002-D9FECD1@10.151.1.5 This leg advertises G711Alaw only leg 3: CUCM-------->.CUBE ( forwardall):a9425980-cdc103c8-713397-43640c0a@10.12.100.67 This leg advertises only G711ulaw leg 4: CUBE------->PSTN ( for forwarded call):8037A35A-764211E9-935C9002-D9FECD1@10.151.10.240   As you can see leg 2 and leg 3 advertises different codecs and this is likely because of the following: 1. Dial-peer to CUCM has only G711alaw configured 2. The codec preference list for the SIP trunk to the gateway is not setup properly   To address this, try and enable G711ulaw on the dial-peer to CUCM. Ie add multiple codec to the dial-peer doing the following: voice class codec 20 codec preference 1 G711ulaw codec preference 2 G711alaw   2. ensure that the audio bit rate for the region assigned to the SIP trunk is set to 64Kbs   If the call still fails collect another set of logs and then attach the CUCM SDL traces ... View more

Since this is a h323 dial-peer, you will need to configure the following, so that when CUCM is down, calls will route to SRST gateway and when CUCM is up, calls will always go to CUCM first.   voice class h323 1 h225 timeout tcp establish 2 h225 timeout setup 2   ++ dial-peer to CUCM ++ dial-peer voice 2 voip preference 1 voice-class h323 1 destination-pattern 5000 session target ipv4:192.168.1.100 dtmf-relay h245-alphanumeric codec g711ulaw   ++ dial-peer to SRST gateway IP ++ dial-peer voice 2 voip preference 2 voice-class h323 1 destination-pattern 5000 session target ipv4:X.X.X.X ( where X.X.X.X is your SRST gateway ip ( you can use loopback address if you have one on the gateway) dtmf-relay h245-alphanumeric codec g711ulaw ... View more

It does not work like that. What you need to do is to configure two translation rules and then apply one for normal call routing and apply the second one during SRST. For SRST configure and  apply the translation profile as follows:   voice translation-rule 10 rule 1 /^0944994433$/ /5000/   voice translation-profile abbr-dial  translate called 10   ++ SCCP phones +++ call-manager-fallback  translation-profile incoming abbr-dial ! ++ SIP Phones +++ voice register pool  2  translation-profile incoming abbr-dial ... View more