In order to test our SRST, we took down the call managers in Boston (HQ).
No office can receive ANY incoming phone calls. This includes London, Menlo Park and Boston. Callers either get no answer, a recording from ATT, or a dial tone.
We cannot dial another office. For example Boston cannot call a number in Menlo Park either with 4 digit dial or dialing the whole 10 digits.
Voicemail doesn?t pickup for the main number in any office.
The only things that work in SRST are we can make outgoing calls from the phones that registered, and we can call within the office to any phones that registered. Could you please advise where do I have to look for this issue?
i have very small branch implementations 2-10 phones. i am using MGCP with the specifc configurations for this. also ensured that mgcp profile is default.
created secondary dial-peer pots patterns as you would do for h323.
created as well, plar connections on all my voice ports to go to specific extensions.
then finally, created alias commands to ensure hunting.
do you have your IOS/SRST dialPlan configured properly? this is what will allow incoming calls as well as the ability to dial the other branches via 4 digit dialing. (inbound & outbound dialPeers are a big part of this in multi branch environments)
see the following link for SRST configuration:
(find for your specific SRST version)
When you fall back to SRST do you see your phones registering on the correct SRST gateways? Also are your PRI's passing the correct number of digts as what you have programmed for extensions? Meaning if your extensions are 4 digts the PRI's should send only 4 digits unles you are using some sort of translation pattern within your config. Other thing to check is that if you are only running SRST on one gateway per site make sure that it has the first( primary) PRI in that is coming from TELCO or you will get disconnect messages....
I would agree I dont see in your config an incoming dial peer statement. Like I said before make sure that the PRI is sending the correct amount of digits or you will have to implement a translation pattern. Also how many gateways do you have at your remote sites and do all of your phones register to just one of them?
We have three SRST gateways in three different sites. Do you see anything wrong with the config?
Unity also fails to respond. Is it because I don't have an dial-peer voice for the voicemail?
Our extensions are in the range of 66XX and 67XX, do I need to add these destination in the command you just sent me? Like:
dial-peer voice 1 pots
incoming called-number 6...
We also fail to dial 4 digits dialing between offieces. The extensions are as follow:
Menlo Park 2XXX
London (UK): 02XX
Could you please let me know what is the config that I have to do in order to resolve this issue?
dont know if im missing something here, but if youre testing SRST then the wan link is down. therefore, by default you would not bve able to dial extension to remote brach or access voice mail. to do this you need to configure translation parameters on the srst router to enable users dialing extension to route using this translation pattern when link is down. if this is the case check this out and see if it helps
what i was saying before as well is that you said that the srst router cannot receive calls but only make. incoming-number should take care of it, however, i use a plar connection to the operator extension and then configure aliases for call forward no-answer an busy etc.
therefore, to direct a call when in srst mode to an extension i use something like this
(config#) voice-port 1/0:0
connection plar 6900 (operator phone)
then configure aliases to suite
I need to configure a dialpeer for our voicemail in SRST mode. Here's what I am going to configure as dialpeer:
dial-peer voice 12 voip
description ** cue voicemail pilot number **
session target ipv4:188.8.131.52
The above config is when the WAN link is up but the CallManager is down. Am I right? Could you please let me know how do I have to configure the same dialpeer in the case when the CallManager is up but the WAN is down?
If you CCM is up and WAN is down then you have to get to voice mail through PSTN.
therefore to get to pstn when in srst,
1. you need to tell the rtr that when i press voice-mail go to a number via my PSTN instead of through my WAN. that number should then be dialed to the location of you active voice mail and should prompt as if it were transparent to user.
therefore if i understand your question correctly you need to perform some SRST configuration for voice mail
that line can be linked to the IPT network, maybe a simple plar connection again to the pilot.
if you need further information on digit manipulation again you can check this out
finally, im sure that someone will give you a bit more detail with this type of configuration.
i hope this helps
The Unity is a full blond (real unity) server. My question is how do I have to configure my dialpeer when the WAN link is down? As the WAN is down I can't use the voip dialpeer and have to use pots. Am I right? Could you please give me the example of configuring a dialpeer for the voicemail with the pilot number 5500 when the WAN is down?
When the WAN is down and Unity resides across the broken link, you have to dial into Unity across the PSTN. On the unity side of things I setup up a DID that dialed directly into Unity.
Where 555-555-5555 is the DID setup for my Unity Voicemail.
No dial-peer would be required for this.