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SOLVED - SIP Trunk to Analog phone on FXS port using CMEFor the life of me, I can't seem to get what I thought to be a simple concept to work...
I have a SIP trunk with an ephone right now. That works perfectly inbound/outbound. I've added a FXS card and DSP to my 3825. When I plug in my analog phone, I can make outbound calls and I can call between extension. However, I cannot find a way to simply allow an inbound call to reach this phone.
I've look into the sccp/stcapp config (which you apparently need an analog gateway to make properly work...) and tried a creating a new dial-peer to route inbound calls from my DID to the voice port, but it never worked. Any help or suggestion would be greatly appreciated!
voice-port is 0/0/0 |
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Re: [HELP] SIP Trunk to Analog phone on FXS port using CMENot seeing your config here it's a bit of a shot in the dark. But you'll need a new pots dial-peer pointed at voice-port 0/0/0 with destination-pattern of XXXX. XXXX being the DNIS coming in from the SIP trunk that you want to ring on this analog port. |
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You know, I didn't know exactly what to add from the config... that's why I wanted to leave it open... What you suggested is the one thing I didn't try! I'll build it now, but I'll have to check it later. Thanks for the reply! |
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sk1939 Premium Member join:2010-10-23 Frederick, MD ARRIS SB8200 Ubiquiti UDM-Pro Juniper SRX320
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to DaSneaky1D
said by DaSneaky1D:For the life of me, I can't seem to get what I thought to be a simple concept to work...
I have a SIP trunk with an ephone right now. That works perfectly inbound/outbound. I've added a FXS card and DSP to my 3825. When I plug in my analog phone, I can make outbound calls and I can call between extension. However, I cannot find a way to simply allow an inbound call to reach this phone.
I've look into the sccp/stcapp config (which you apparently need an analog gateway to make properly work...) and tried a creating a new dial-peer to route inbound calls from my DID to the voice port, but it never worked. Any help or suggestion would be greatly appreciated!
voice-port is 0/0/0 When you get a chance, post you dial-patterns and/or a show tech-support sanitized. |
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Well, I tried variations of rsaturns' suggestions, but nothing is working.
"DID is 314.555.1234"
I've tried:
dial-peer voice 20 pots incoming called-number 1234 direct-inward-dial port 0/0/0
dial-peer voice 20 pots incoming called-number 3145551234 direct-inward-dial port 0/0/0
- and -
dial-peer voice 20 pots destination-pattern 3145551234 port 0/0/0
The main parts of the FXS config is as follows:
voice-port 0/0/0 timeouts ringing 20 caller-id enable ! voice-port 0/0/1 ! voice-port 0/0/2 ! voice-port 0/0/3 ! dial-peer voice 102 pots destination-pattern 102 port 0/0/0 |
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DaSneaky1D |
I've tried applying the config (suggestions) listed at both of these links, to no avail. I'm completely lost as to what I'm missing: » supportforums.cisco.com/ ··· -h323-gw» www.techexams.net/forums ··· unk.htmlAll in all, I have a perfectly working system with my ephone. I've tried removing the ephone-dn's from the config to see if that made a difference, too. Unfortunately it didn't. I don't mind posting the whole config, but everything else is just outbound dial-peers, COR lists and translation rules for outbound calls...nothing that should impact getting an inbound call to a FXS port. |
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sk1939 Premium Member join:2010-10-23 Frederick, MD |
sk1939
Premium Member
2014-Apr-23 10:23 pm
You do realize that to receive incoming calls via POTS you need an FXO card right? You cannot use an FXS card for incoming calls. Sorry I missed that earlier. |
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Correct, but I'm trying to receive the call via the SIP trunk, but have the call ring to a phone plugged into the FXS port. Currently I can only get calls to ring into my IP phone. |
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sk1939 Premium Member join:2010-10-23 Frederick, MD ARRIS SB8200 Ubiquiti UDM-Pro Juniper SRX320
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sk1939
Premium Member
2014-Apr-23 11:26 pm
Your dial pattern is off if your going from VoIP to POTs then I believe. The POTs command you have listed indicates a dial peer using basic telephone service, not VOIP
Try:
dial-peer voice 20 voip destination-pattern 3145551234 port 0/0/0 |
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Yeah, I tried that earlier... didn't work:
3825-1(config)#dial-peer voice 20 voip 3825-1(config-dial-peer)#port ? % Unrecognized command 3825-1(config-dial-peer)#port |
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sk1939 Premium Member join:2010-10-23 Frederick, MD |
sk1939
Premium Member
2014-Apr-24 1:03 am
Are you sure of your port numbers?
WIC 0 ports: 0/0/0, 0/0/1 WIC 1 ports: 0/1/0, 0/1/1 WIC 2 ports: 0/2/0, 0/2/1 WIC 3 ports: 0/3/0, 0/3/1 |
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They're 0/0/0 thru 0/0/3 |
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sk1939 Premium Member join:2010-10-23 Frederick, MD ARRIS SB8200 Ubiquiti UDM-Pro Juniper SRX320
3 edits |
sk1939
Premium Member
2014-Apr-24 2:30 am
If Direct Inward Dialing (DID) is configured in the inbound POTS dial peer, the router uses one-stage dialing, which means that the full dialed string is used to match outbound dial peers.
You also need to have your destination pattern, port, VoIP, and session-target in the command. You can also try the answer-address command.
dial-peer voice 1 pots incoming called-number (phone number) voice-port (port) direct-inward-dial dial-peer voice 2 pots destination-pattern (phone number) session target (VoIP Provider)
dial-peer voice 4 pots answer-address (Phone Number) session target (VoIP provider)
(Session Target for IP clients and Port X/X/X for POTS.)
Edit: Please post your config, it is very hard to troubleshoot with only partial pictures. |
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Thanks for your continued assistance! I'll try those tonight when I get home.
voice service voip ip address trusted list ipv4 208.100.39.53 ipv4 208.100.39.54 ip address trusted call-block cause not-in-cug gcid clid substitute name allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip no supplementary-service sip moved-temporarily no supplementary-service sip refer sip e911 asserted-id ppi midcall-signaling passthru no call service stop ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g711alaw ! voice translation-rule 1 rule 5 /^10[1-5]$/ /[ACTUAL DID]/ ! voice translation-rule 2 rule 5 /^\([2-9]......\)$/ /314\1/ ! ! ! voice translation-profile 314-7-to-10-Digit translate calling 1 translate called 2 ! voice translation-profile Caller-ID translate calling 1 ! voice-port 0/0/0 timeouts ringing 20 description Cordless Phone station-id name Cordless caller-id enable ! voice-port 0/0/1 ! voice-port 0/0/2 ! voice-port 0/0/3 ! dial-peer cor custom name 911 name Local name LD ! ! dial-peer cor list call-Local member Local ! dial-peer cor list call-LD member LD ! dial-peer cor list Ext-Allowed-Calls member Local member LD ! dial-peer cor list All-Allowed-Calls member 911 member Local member LD ! dial-peer cor list EMERGENCY member 911 ! ! dial-peer voice 5 voip corlist outgoing EMERGENCY description Dialing Plan for 911 preference 1 destination-pattern 911 session protocol sipv2 session target dns:chicago3.voip.ms session transport udp voice-class codec 1 no voice-class sip block 180 no voice-class sip block 183 no voice-class sip block 181 dtmf-relay rtp-nte no vad ! dial-peer voice 10 voip corlist outgoing call-LD description Dialing Plan for LD translation-profile outgoing Caller-ID preference 1 destination-pattern 1[2-9]..[2-9]...... session protocol sipv2 session target dns:chicago3.voip.ms session transport udp voice-class codec 1 no voice-class sip block 180 no voice-class sip block 183 no voice-class sip block 181 dtmf-relay rtp-nte no vad ! dial-peer voice 15 voip corlist outgoing call-Local description 7-digit Local for 314 translation-profile outgoing 314-7-to-10-Digit preference 1 destination-pattern [2-9]...... session protocol sipv2 session target dns:chicago3.voip.ms session transport udp voice-class codec 1 no voice-class sip block 180 no voice-class sip block 183 no voice-class sip block 181 dtmf-relay rtp-nte no vad ! dial-peer voice 102 pots description All Analog Phones destination-pattern 102 port 0/0/0 ! telephony-service no auto-reg-ephone max-ephones 5 max-dn 6 ip source-address 10.11.1.1 port 2000 timeouts interdigit 3 load 7960-7940 P00307020200 max-conferences 3 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Mar 31 2014 23:58:13 ! ! ephone-dn 1 dual-line number [ACTUAL DID] no-reg primary label Home description Bat Cave corlist incoming All-Allowed-Calls ! ! ephone-dn 2 dual-line number 101 no-reg primary label Bat Cave - 101 corlist incoming Ext-Allowed-Calls ! ! ephone 1 device-security-mode none mac-address 0017.5976.86D4 type 7960 button 1:1 2:2 |
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DaSneaky1D |
to sk1939
This isn't going to work, as the pots dial-peer cannot have a session target, let alone one to an IP address:
dial-peer voice 2 pots destination-pattern (phone number) session target (VoIP Provider)
**tried and the router really didn't like that! |
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sk1939 Premium Member join:2010-10-23 Frederick, MD ARRIS SB8200 Ubiquiti UDM-Pro Juniper SRX320
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sk1939
Premium Member
2014-Apr-24 9:42 pm
said by DaSneaky1D:This isn't going to work, as the pots dial-peer cannot have a session target, let alone one to an IP address:
dial-peer voice 2 pots destination-pattern (phone number) session target (VoIP Provider)
**tried and the router really didn't like that! I had mentioned below that pots = port and VoIP = session target (might have forgotten that). |
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Tried applying the below, but the call didn't route to the voice port. Only ephone-dn 1 rang.
dial-peer voice 1 pots incoming called-number [DID NUMBER] port 0/0/0 direct-inward-dial
dial-peer voice 2 pots destination-pattern [DID NUMBER] port 0/0/0
dial-peer voice 3 voip preference 1 destination-pattern [DID NUMBER] session protocol sipv2 session target dns:chicago3.voip.ms session transport udp voice-class codec 1 no voice-class sip block 180 no voice-class sip block 183 no voice-class sip block 181 dtmf-relay rtp-nte no vad |
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sk1939 Premium Member join:2010-10-23 Frederick, MD ARRIS SB8200 Ubiquiti UDM-Pro Juniper SRX320
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sk1939
Premium Member
2014-Apr-24 11:43 pm
It seems like your missing some stuff. Note VoIP.ms's sample configuration: » wiki.voip.ms/article/PBX ··· isco_IOSYour doing something that is increasingly rarely done (going from VoIP to POTS) instead of the other way around. It is frankly easier to use an ATA and configure CME as a strict gateway/PBX imo, and most Cisco-suggest configurations reflect that. However, your router should be able to do what your trying to do. |
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Well.... if I could have just paid attention to that page 5 days ago, I could have got it ringing!
It's working! Thanks for your help and suggestions. Both got me to where I needed to be!
### The Fix ###
I've trimmed down the dial-peers that I strictly needed to get this working:
dial-peer voice 1 pots destination-pattern [FULL DID] port 0/0/0 ! dial-peer voice 2 voip description Main Line Inbound session protocol sipv2 session target dns:chicago3.voip.ms session transport udp incoming called-number [FULL DID] voice-class codec 1 no voice-class sip block 180 no voice-class sip block 183 no voice-class sip block 181 dtmf-relay rtp-nte no vad
The problem arose because I had an ephone-dn that had my DID directly assigned as the "number" on it. That ephone-dn was taking precedent and accepting all the calls. Once I added the above dial-peers and removed it, calls ran perfectly.
Can't thank you enough for your assistance! |
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sk1939 Premium Member join:2010-10-23 Frederick, MD ARRIS SB8200 Ubiquiti UDM-Pro Juniper SRX320
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sk1939
Premium Member
2014-Apr-25 12:27 am
said by DaSneaky1D:Well.... if I could have just paid attention to that page 5 days ago, I could have got it ringing!
It's working! Thanks for your help and suggestions. Both got me to where I needed to be!
### The Fix ###
I've trimmed down the dial-peers that I strictly needed to get this working:
dial-peer voice 1 pots destination-pattern [FULL DID] port 0/0/0 ! dial-peer voice 2 voip description Main Line Inbound session protocol sipv2 session target dns:chicago3.voip.ms session transport udp incoming called-number [FULL DID] voice-class codec 1 no voice-class sip block 180 no voice-class sip block 183 no voice-class sip block 181 dtmf-relay rtp-nte no vad
The problem arose because I had an ephone-dn that had my DID directly assigned as the "number" on it. That ephone-dn was taking precedent and accepting all the calls. Once I added the above dial-peers and removed it, calls ran perfectly.
Can't thank you enough for your assistance! Your welcome, glad that it is finally working. I assume that your VoIP line continues to work, correct? |
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I can make inbound/outbound calls just fine. The ephone is actually the least of my concern. I really wanted to get the analog phones working. If I could get both ringing inbound and answering successfully, then that's a plus, but not a major concern. |
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