Welcome to the Cisco Networking Professionals Ask the Expert conversation. This is an opportunity to discuss Voice H323/SIP Gateways and Gatekeepers, ITS- Call Manager Express and SRST with Cisco expert Taimoor Husain. Taimoor is a Senior Engineer of the Multiservice group at the Technical Assistance Center (TAC) at Cisco Systems, Inc. He supports Voice related technologies, including session protocols such as SIP, H323, MGCP and troubleshoots networks running VOIP/VOFR/VOATM. He also works with ITS (Telephony service) and SRST. Feel free to post any questions relating to Voice H323/SIP Gateways and Gatekeepers, ITS- Call Manager Express and SRST. Remember to use the rating system to let Taimoor know if youve received an adequate response.
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I was wondering on an H323 gateway will we ever be able to specify a secondary Call Manager in the same dial-peer? As it stands now it is my understanding that in order to specify a secondary Call Manager you have to create 2 dial-peers, one for the primamry and one for the secondary. This gets a bit hairy with a lot of dial-peers on teh config. I think an MGCP gateway has a set it globally. When (if ever) H323 have the same?
Unfortunately you cannot do that. THe only way to do that in h323 is using multiple dialpeers and useing their preferences.
Hello Again Taimoor,
The 3600 series routers cannot have the * symbol as a lead character on a dial-peer. Other routers can though. Is this something that is going to change with a later IOS revision or is my only option going to be upgrading to a different router? Thanks in advance!
That sounds a bit odd and not right.
When you mean a * as a lead character you mean, in the destination pattern, or on the tag. Please be a little more specific, it sounds like a potential bug, either way, and i can advise you what you might need to do.
Sample dial peer config would help.. :)
I am told that a destination patternt like belowwill not work on a 3600 series router. I am using a TCL script on a 3600. The router is acting as an Auto-Attendant and routing based on dial-peers. Here is what I want to do
dial-peer voice 11 voip
session target ipv4:10.0.0.50
I want to have my users sign in to Unity by pressing * when the AA picks up. I am trying to approximate their experience with the legacy key system I am phasing out. I would send that * over to Call Mananger and do a translation from * to 5000 (the first VM port) Your thoughts?
The destination pattern of * will indeed work, however the interpretation by the tcl script, may or may not. The tcl script should recognize the star as a digit and look for a dial peer match and send the call over. This is indeed interesting and should be quite possible. I will try and test it out and see if I come up with anything, but please let me know if it works for you.
My feeling... it should... :)
I am trying to find a VoIP solution for some employees working from home. The ideal scenario is having both an FXS and a FXO port at everybodys house/apartement. This way, you would dial via IP when possible and PSTN when the other end is non-ip. Dialing to IP and nonIP phones would be completely transparent, that is the goal.
Since I need to place one unit at every employee larger gateways such as the VG200 is out of question. We need some small SOHO router which can do it or possibly the ATA series. The ATA is a FXS port only and so I would still need some other device with a FXO. I've seen the ubr924 router which has two voice ports (FXS) and one "line". Now what is that line port? Is it an FXO and would I thereby be able to use a ubr router as a complete voice gateway??
The line port is like a FXO port and is very limited in functionality. This line port will only function when the UBR924 loses power and VoIP calls cannot be made. In that case, you can use this line port to make outgoing calls to the PSTN. Upon resume of power, if a pstn call is connected, ubr will wait for that call to clear and then reboot, and get ready for VoIP. So, as it seems it doesn't help you in your scenerio. Moreover this ubr924 is EOL.
Unfortunately there is no small solution that has an FXO AND FXS port together. The smallest you could possible go would be a 1750 with two vic cards, which would probably amount to bein costly.
As a side note, you could investigate into a hosted call manager or sip solution, which would allow your users to have ip phones at their homes registered with call manager. Doesnt solve the PSTN availability though.
Sorry about that.
How can I reject in a pots or voip dialpeer all phone numbers that have less or more than 7 digits ?(or other number of digits).
You can use the enhanced translation rules to reject patterns of your choosing. However in your situation it is going to be a little more tricky since you want to do it by the number of digits.
THe best design for this would be to do the call rejection on the destination patterns. Essentially if there is no destination pattern match in the gateway the clal will not get through.
There fore you should have very specific destination patterns in order to get this to work.
Here is some documentation on the enhanced translation rules :
I have more than 1400 IP phones, you can realize that this not practicall at all because I will need a dialpeer for every IP Phone.
You can maybe isolate the phone and create a few dial peers with the appropriate ranges for your ip phones. They are all 7 digit numbers im assuming so even leaving a wild card 7 digit destination pattern out there would be ok, but in your case it may make more sense to use the enhanced translation rules.
The problem is:
When I use a 4 digit wilcard (for example 252....) , it also match with a number of 3 to n digits (for example 2524,25243,252431,2524312,2524351, etc)
The enhanced Translation rules (using the reject rule) does not verify the numbers of digit. If the first n digits are matched the entire number is rejected.
I hope ure ready for some translation rule trickery :
voice translation-rule 1
rule 1 reject /\(........\)/
rule 2 /\(.......\)/ /\1/
rule 3 reject //
This will only allow 7 digits and nothing more and nothing less...
640-1#test voice translation-rule 1 123456578
123456578 blocked on rule 1
3640-1#test voice translation-rule 1 123456
123456 blocked on rule 3
3640-1#test voice translation-rule 1 1234567
Matched with rule 2
Original number: 1234567 Translated number: 1234567
Original number type: none Translated number type: none
Original number plan: none Translated number plan: none
Although it is a very tricky way to do things, it will accomplsh what you are trying to do.
The first rule blocks numbers greater than 7. The seoncd one allows 7 digits, and does not translate them. The third rule blocks anything else.
You will need to make a translation profile and then apply the profile to either an inbound or outbound dial peer.
What could be the reasons for a call dropping in between conversation between an IP phone user and a PSTN phone? Callmanager with MGCP and H323 gateways.
As far as an ideal world goes.. nothing... but something has to be happening signalling-wise to make the call drop. Either of the party has to be initiating a disconnect, or the tcp session between the gateway and the call manager could be timing out, therefore dropping the mgcp registration.
I would troubleshoot on the gateway for tcp transactions, mgcp process and signalling to see why the call is dropping.
I have to set up a trunk between two pabx's connecting to E&M modules in 2610's running IOS 12.3 WAN access is FR into an MPLS network. The MPLS Network has been set up to support VoIP COS & DSCP. Can I set up the dial-peer's as voip and route them over the MPLS via the FR access or will I have to set up the dial-peer's as vofr and work with the provider to switch the calls over the FR network ?
PS if you have any URL's on this then I would be very grateful!
You can setup dial peers as voip and give them session targets for ip addresses. This is called Voice over IP over Frame and is very common.
Please take a look at these links :
I want to ask about a H.323 command. My gateway is connected between callmanager and PSTN. I have config the gateway in H.323. I want to ask about the dial-peer. The destination-pattern is set out to POTS. The pattern is "9T". How many types of pattern can use in the command?
If you mean what are the valid destination patterns you can use, there are many.
9T means that the number begins with a 9 and can have any number of digits trailing it.
5..... means that the number begins with 5 and is a six digit number
[2-9]... means the number begins with anything from two to nine and is a four digit number.
There are many combinations that you can use. Remember the more speciofic the dialplan the better.
I'm in the final stages of Designing a IP Telephony Solution where we shall be deploying more than 20 Call Manager (all at different location) each representing a Single CCM Cluster, additionally we shall be integrating Voice Gateways for calling EPABX users almost at all locations. Also 3 Gatekeeper and one Directory Gatekeeper shall be used for CAC and Call Routing.
There will be few location without any Call Manager and IP Phone deployed at that these location shall register to nearest CCM Cluster.
Location with Call Manager are connected using 2MB link and non-CCM location have 64k Link.
I have following issues which are yet to be resolved.
1. For QoS can I use Compression at 64k Link and no compression at 2MB Link, which mean call travelling from end to end may have compression enable at extreme ends but not enable in centre. Similarly for Fragmentation also.........
2. Also voice call end to end has to cross 2-3 hops and to the best of my undestanding GK doesn't have complete topology, so how can i restrict or manage the number of calls from one to other.
Request you to suggest something on this.
I'm not sure if we are supposed to reply to the "ask the expert" posts, so please forgive me if that is the case.
I'm assuming Frame Relay, so check if you can enable cRTP on an individual PVC basis without affecting the need to turn it on the physical interface (at the head-end) thereby alleviating the necessity for all PVC to be affected. Other things to consider will be LFI and VoiceAdaptiveFRTS.
Addressing your second question, you're right! H.323 does not, nor does any other signaling protocol for that matter, offer an effective call admission control [CAC] methodology over multiple hops. Further, to my knowledge SIP and MGCP don't even try!
The only true end-to-end CAC mechanism is RSVP. RSVP implicitly performs CAC by refusing a call attempt if the resources (including bandwidth) are not available to guarantee the desired quality parameters set forth. To date, RSVP has had a bad rap when it comes to scalability, post-dial-delay, etc. If you choose to, we can 'talk' off-line about ways to get around this. This is a pro-Cisco solution, I just don't want to use this forum as a place for marketing.
For your first question, the only kind of compression you can do is compression achieved by using differnt codec. You cannot do rtp header compression for calls going from ip phones to pstn or between the two sites. Depending on how you configure regions and locations you can configure some sites to use different codecs than others, however in your scenario, I would almost always suggest you to use g729 codec between sites even on the 2MB link between call managers.
Secondly to restrict number of calls going from site to site, you can still use the GK, im not sure what you mean by GK does not have complete topology, but in a gatkeeper there is a concept of zones, and you can identify differnt zones for differnt sites, and you can restrict the bandwidth for each zone or interzone calls, so when a call admission is requested, the gatekeeper checks against the bandwidth and either allows it or denies it.
Similarly if you dont want to use a gatekeeper for this then you will have to configure intelligent queuing, maybe rsvp flows or policing to limit the amount of voice calls traversing the network, a much more complicated solution.
I would also encourage you to get youre account team involved in helping yo make these decisions.
Hope thie helps.
SRST H.323 & MGCP -
Will SRST only work with H.323 gateways?
If so, can you configure an MGCP gateway to also use H.323? Any disadvantages to a dual configuration with MGCP for CCM operation and H.323 for SRST?
SRST can work on mgcp and H323 gateways.
The way this works is when you fall into SRST mode, the gateway falls out of MGCP and into h323, this is acheived by using :
call applicaition alternate defaul
ON a side note, you can configure a gatewya to be both h323 and mgcp but not recomeended, how would you tell the gateway which aspects the call manager controls and which are controlled by the gateway. It can still process h323 calls in and out, but gets extremely complicated configuration wise.
Assuming an MGCP gateway also running SRST. When phones lose connectivity to CCM, MGCP falls back to H323. So what about Dial-peers required to make inbound/outbound calls ? Dont we have to configure these or is it automatic (which i dont think will work).
The simple answer is you can configure dial peers if you want to, but dont need them because you can use the access-code command under the fallback to create an automatic dial peer to go out of the voice ports. Inbound will work fine as all phones registering with the gateway will create dial peers for themselves automatically.