It occured to me that I never wrote up how I managed to get Asterisk working with my Cisco 1760 as an FXO gateway. So, here it is:

I have a Cisco 1760 that began it’s life as a CallManager Express (CME). After a while I transitioned it to an MGCP gateway for CM 4.1 and then just a SIP gateway for the same CM4.1 See, CallerID didn’t work for my VIC with MGCP, but it worked with SIP. Now the router is a SIP gateway for Asterisk, which is my primary system. Verizon comes into the FXO on the router and then is trunked via SIP to Asterisk. So…on to the configs. Here’s the important part of the router config:

!
voice rtp send-recv
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
!
!
voice-port 0/0
timing hookflash-out 50
connection plar 401
caller-id enable
!
dial-peer voice 8 pots
destination-pattern 8.T
port 0/0
!
dial-peer voice 400 voip
destination-pattern 4..
session protocol sipv2
session target ipv4:192.168.x.x:5060
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
sip-server ipv4:192.168.x.x
!
!
telephony-service
fxo hook-flash
max-conferences 4 gain -6
transfer-system full-blind
transfer-pattern ….

I’m quite certain all of that is not necessary but it works, so I’ll leave it alone.

Here’s the config in AMP. I got this working through the raw config files but wanted to figure it out in AMP. If you’re going to use A@H and AMP it’s best to get it working through AMP. AMP has a nice habit of writing over certain files because it know’s it correct! Anyway, here’s the config for the SIP trunk:

Outbound caller-id: Set to your PSTN number (although I don’t think it matters)
Max Channels: Set to 2 but I think it should probably be 1
Dial Rules: Whatever’s appropriate for you situation
Outgoing Settings: trunk name should be something meaningful
allow=ulaw
context=from-pstn
disallow=all
dtmfmode=rfc2833
host=192.168.x.x
insecure=very
ipaddr=192.168.x.x
type=peer

Replace x.x with the address of the router. Make sure to add the insecure=very! I can’t tell you how important that is. I went round and round before stumbling across that. The partial tutorials on the intarweb don’t specify that. From what I can understand you need this because there’s no means of authenticating the SIP session with the router, aside from the source IP address.

Leave everything else blank. Don’t forget to add your outbound routing dial patterns.

That’s it. If someone has a comment please email me or leave it in the comments here.

Comments (55)

  1. B.

    Reply

    Dude, you are a genius! Seriously, I racked my brain trying to get things working, and apparently my only issue was I was missing this command:
    voice rtp send-recv

    Needless to say I need to do some more reading up on it. Do you do this as a hobby for fun or is this some how related to work?

    I’m just playing around with it for now, and just got voicemail working (although through AMP). I’m noticing some odd behavior from AMP, but reading your post that’s probably the norm.

  2. Scott

    Reply

    It’s related to work. Although, I guess it’s a bit of a hobby too. I’m a CCNP with some Voice certs so I try to keep up with this stuff. Running it at home is sort of a personal requirement.

  3. B.

    Reply

    Hate to be a pest, I’ve been working with the integration of CCME and Asterisk for a while now. I’m pretty much where I need to be, except for one annoyance. I’m using several Cisco 7910 IP Phones (and a few 7960s – but 7910s are the majority). I can not get the MWI light to light once a message is left. I’ve read several peoples posts that got it working. And it seems to be centered around using the unsolite option for the mwi sip-server command (either under Telephony-service or sip-ua). I’ve tried multiple ways to do this now. I am still developing on an Asterisk server running inside a virutal machine, but I don’t see why that would matter.

    Voicemail works great, there’s just no way know a message is left without checking all the time. I’m running IOS 12.3.11.T6. I’m thinking that may be the issue, but all the options I need are there (according to the two and a half guides there are on the internet).

    Just thought I’d check to see if you had tried and if so had any luck.

  4. Mike

    Reply

    Hi There
    I need some help, I am hosting Asterisk and constantly having issues with ATA cards. I gat Cisco 2400 router with FXS module. How to configure it to connect to Asterisk SIP PBX via Ethernet and use FXS as phone termination
    Thanks

  5. Reply

    Mike, the configuration will be similar to what I have in this post. I imagine the big difference will be adding additional dial plan configuration to the Asterisk to send calls to the FXS ports.

    You also said ATA though, which is different from an FXS module (VIC) on the router. If you have an ATA-18X you’ll need to set that up just like a SIP or SCCP phone. The ATA will need to register with something, either Asterisk or CME running on the router.

    In other words, the ATA is treated like a VoIP phone where the FXS VIC is treated like a dial pattern destination on the router.

  6. Mike

    Reply

    Hi Scott

    When I sad ATA cards I meant, let’s say for example, I have a client that has an existing analog system I have to hand over a dial-tone, that means terminate via fxs card but I had a bad experience with ATA cards in the past, they drop connection time to time. I terminate my Asterisk with my ISP-Voip guys via cloud and don’t have a problem config Asterisk as far as trunking. My problem is haw to config Cisco router: I have 2400 with Eight FXS Analog Voice Module and fa interfaces. Please if you can give me an idea. I would be extremely grateful.
    My e-mail marslove@usa.com

    Thanks
    Mike

  7. Reply

    Mike, if you’re referring to the IAD2400 I really don’t have any familiarity with that. I can tell you though that there a couple of common causes of problems with back to back analog lines.

    Impedance, echo-cancellation, output attenuation, threshold and the signaling type can all effect performance. If your setup isn’t working with the default settings then you might need to tweak any or all of these. It’s really just trial and error. I’d start with Impedance, Output attenuation and signaling type.

    Good luck!

  8. Peter

    Reply

    I was wandering if this solution makes any use of a PVDM module in the Cisco router? Any sought on this?

  9. Chris

    Reply

    Scott,

    Did you ever find out why caller-id did not work with CME4.1 on the FXO VIP on the 1760?

    Thanks

  10. Reply

    Chris,

    The problem with caller-id was strictly related to MGCP. Turns out MGCP doesn’t support CID on FXO interfaces in Cisco routers. There are specific references to this on CCO although they are buried pretty deep and there’s apparently no desire to fix this issue. SIP trunks don’t have the same problem but you lose remote control of the gateway. So, CID or remote control.

    We have the same issue in the office but we’ve managed to have a mixed environment where we have some MGCP controlled ports for an overhead pager we use and h.323 for all of the inbound FXO trunks. We’ll be converting to CM6.1 very soon and may go to SIP but I suspect the solution will look similar.

  11. Luis

    Reply

    I followed step by step manual but can not send calls to cisco router

    This is the message at the Console asterisk

    PLS Help Me !!!

  12. Luis

    Reply

    I followed step by step manual but can not send calls to cisco router

    This is the message at the Console asterisk

    Channels Unavailables!!!

    PLS Help Me !!!

  13. Reply

    Luis, Channels probably refers to some channel, SIP perhaps, not working properly on Asterisk. Not sure how you built Asterisk but I’d start with making sure it’s set up right. Try configuring a softphone in Asterisk and make sure you can hear some of the test scripts.

  14. Jimmy

    Reply

    Scott,
    Thanks for the post this is really great. I am tring to do the same thing with my 1760 but I do not have the option for allow-connections sip to h323. If I do:
    allow-connections ?
    I get
    h323 from-type h323 (I do not see a sip option)
    show version says I am running:
    version 12.3(8)T, Release Software (fc2)
    Do you have any help?

    By the way I have a blog for colleting stuff like this at http://www.qvlweb.blogspot.com and I will link to your site. Thanks again.

  15. Reply

    Jimmy, according to the Cisco Feature Navigator (http://www.cisco.com/go/fn) it is supported in 12.3(8)T but only on certain feature sets.  You must have one of the following:

    Adv Ent Serv
    Adv Ip Serv
    Ent Serv
    IP Voice
    SP Serv
    And there are some complicated ones like IP/ADSL/IPX/AT/IBM/VOX/FW/IDS PLUS IPSEC 3DES.

    Good grief they used to get complicated.

    If you’re running something lower, such as IP Base or maybe even just IP Voice it’s possible there’s limited support for h.323 but not for SIP.  The 1760 is getting pretty long in the tooth and the SIP support in the early days was definitely a “high-end” feature.

  16. Jimmy

    Reply

    I contacted cisco support, it looks like I have IP VOICE and the following IOS should have the SIP options:
    12.4.15.T8
    Thanks for the help.

  17. Reply

    Hello!

    This page ended up helping me a lot when I was setting up a Cisco 3825 with a four-port FXO VIC and a two-port FXS VIC.

    It gave me one idea to try because I was having a bit of an issue with receiving calls on one of  the FXO ports.  It would make a SIP connection to my Asterisk setup, then drop, but keep ringing on the calling side (I used my cell phone to do the testing).  Then the phones on my setup would start ringing again.

    From my config:

    voice-port 0/2/0
    […]
    connection plar opx 1001
    […]

    And of course, an accompanying dial-peer for this config on the Cisco, and an extension 1001 to ring on the Asterisk machine, which I could then hand to the actual extensions I wanted to ring. 

    I got this elsewhere on the ‘net, but this was the only vexing issue.  It did the exact same thing whether I was using a separate Sipura or the Cisco’s FXS port to handle my telephone.

    Changed it to:

    voice-port 0/2/0
    […]
    connection 1001
    […]

    (pulling the “opx” statement out), and it works just fine.

    Also, for CallerID to work, you need “caller-id enable” on both the FXO (to collect the information) andFXS (to pass it on to the phone connected to it) and everything works.

    I dunno what the details about CCME on this router are, but I’m running IOS Version 12.3(11)T8 on this device.  Other devices on this router include two T1 VWICs, one a two-port, the other a one-port.  Since I’m using this in a residental environment, there’s simply no way I can use this, of course.

    Anyway, thanks for these tips and I hope what I’ve written here helps others.

    –Ian.

  18. Reply

    I continue to be amazed by how often this post gets hit!  I should probably do an update but not much is different.

    Ian, the OPX thing seems to be a problem for a lot of people.  From the Command Reference:

    Specifies a PLAR off-premises extension (OPX) connection. Using this option, the local voice port provides a local response before the remote voice port receives an answer. On Foreign Exchange Office (FXO) interfaces, the voice port will not answer until the remote side has answered.

    I’m not sure but I think this creates a problem with how the Asterisk server answers the call.  Without the OPX the call is simply forwarded through.

    The other thing that seems to be a common problem is the context.  I have to admit I’m still a little daunted by contexts.  I understand how they work but the actually implementation still confuses me and, as far as I can tell, there’s not GUI way to manage them.  That appears to be a piece still embedded in obscurity in the files.

  19. Jim

    Reply

    This question is a little off topic, but when doing a google search I hit this site. Anyway, I have a Cisco 1760-V which is running CME. I think it is version 12.4(15)T voice load with CME 4.1. I have a VIC-FXO2 card. I can make calls internal with no issue. When I call external, the party can hear me (using a Cisco 7940), but there is NO voice from them. I don’t even hear the ring when calling out. I have tried 2 different VIC cards.. DO you have any idea what might be causing this issue?

  20. Reply

    Your internal phones are connected to a switch which is connected directly to the 1760-V?  It’s not unusual for something like this to be caused by VAD being enabled.  I always turn off VAD in situations where I don’t need to save bandwidth.  It seems to cause more problems than it’s worth.  “no vad” on the dial peer.  Let me know if that works!

  21. Jim

    Reply

    Yes, I have a Cisco 3650 switch with VLAN 1oo for all the phones. Data is on VLAN 200. The 1760-V is attached directly to the switch. In the 1760-V is the VIC-2FXO card. Here is the voice port is configured as follows;

    voice-port 2/0
      input gain 6
      connection plar opx 7001
    voice-port 2/1
      input gain 6
      connection plar opx 7001

    I only have dial-peers to these POTS interfaces (except for the phone-dn of course). Here is a sample of the dial-peers;

    dial-peer voice 100 pots
      destination-pattern 9[2-9]……
      port 2/0
      forward-digits 7

    On my dial-peers I do not have the option of “No VAD”. I guess this is for “VoIP” dial-peers? Do you think the Cisco 1760 hardware is bad. I can call internally ok. I can even call from a outside line to an internal extension and have that forward to another outside number and the two outside phones can talk together fine.  However, when I place an outside call I don’t hear the ring tones or the person voice…it is just totally dead..

    Thanks for your response..

  22. Jim

    Reply

    My issue was the fact I had ” No IP route” on the 1760.  Thanks so much for your help anyway..

  23. Jimmy

    Reply

    Scott,
    I got our Cisco 1760 calling into the asterisk box and I get the voice tree. But when I dial an extension the sip phone rings but when I pick it up (asterisk console says Native bridging), there is no sound in or out. Any Idea what that could be?

    Also I have not been able to call out yet. I get fast busy any suggestions would be appreciated. Not sure if it is Asterisk or Cisco.

    Jimmy

  24. Reply

    Jimmy, it could be a codec mismatch.  Make sure you’re the same on both sides.  Probably G711 ulaw.

  25. Jimmy

    Reply

    I got the outbound dialing working. I was missing the
    dial-peer voice 8 pots
    destination-pattern 8.T
    port 0/0

    calls ring in and out but still no voice. I put in a call with cisco. codecs are the same I think.
    On the router: codec g711ulaw
    On asterisk allow=ulaw

  26. Jimmy

    Reply

    Oddly enough if I go to voice mail I am able to leave a message and hear it when I play back voicemail.

  27. Reply

    Are your phones on a different subnet?  In other words, is there a way for traffic to route between the router and the phones?  The call setup could be going through the Asterisk but then the call path is failing.  Potentially 2 different things.

  28. Jimmy

    Reply

    I think you are on the right track. I can ping from my router to asterisk but not to the phone. Asterisk can ping to both router and phone. Looks like Asterisk establishes the bridge and expects the phone and router to talk to each other. All 3 devices are on the same subnet so I suspect a vlan issue on the switch.

  29. Jimmy

    Reply

    Got this working and the problem was vlans. The router and phone were talking to asterisk but not to each other.

    I started a series of posts to document my progress step-by-step, and so far I am up to post 5. http://qvlweb.blogspot.com/2009/04/asterisk-pbx-install-index.html

    Now I am trying to get multiple lines working (two right now). Inbound everything is fine, and I can call in just fine on both lines at the same time. Outbound the first call out connects but the second call to outside line fails Asterisks gets a busy message from the the 1760.

    Do I need to configure Asterisk to tell the 1760 which channel to use?

    Any thoughts Scott. This is my config info for both pots lines:
    Cisco-

    dial-peer voice 100 pots
    huntstop
    preference 2
    destination-pattern .T
    port 0/0
    forward-digits all
    !
    dial-peer voice 2212 pots
    huntstop
    preference 1
    destination-pattern .T
    port 0/1
    forward-digits all

    sip.conf –
    [Cisco1760]
    context=Outside
    type=friend
    host=172.x.x.x (ip of my 1760)
    dtmfmode=rfc2833
    disallow=all
    allow=ulaw
    insecure=very

    extensions.conf
    [outbound]
    exten => _9NXXXXXX,1,Dial(SIP/Cisco1760/${EXTEN:1})
    exten => _9NXXXXXX,n,Congestion()
    exten => _9NXXXXXX,n,Hangup()

  30. Jimmy

    Reply

    The problem was solved after removing the “huntstop” from the dial-peers.

  31. Mahmoud

    Reply

    Scott,

    Thanks for sharing this article it has helped me heaps with understanding cisco voice gateway integration with asterisk.

    MZ

  32. Frank_Bauer

    Reply

    I have followed this guide , using a cisco 2600xm , vwic 2FXO/FXS. I am able to ring from the FXS port using a telephone to Asterisk voip extensions, however I am not able to ring from the Asterisk extensions to the FXS, help would be appreciated.

    service timestamps debug datetime msec
    service timestamps log datetime msec
    no service password-encryption
    no service dhcp
    !
    hostname ffvr1
    !
    boot-start-marker
    boot-end-marker
    !
    enable secret 5 $1$frQ.$EQU9xcPniWJLjaSuRT.D1/
    !
    no aaa new-model
    memory-size iomem 10
    clock timezone GMT 0
    no network-clock-participate slot 1
    no network-clock-participate wic 0
    ip cef
    !
    !
    !
    !
    ip auth-proxy max-nodata-conns 3
    ip admission max-nodata-conns 3
    !
    !
    !
    !
    !
    voice rtp send-recv
    !
    voice service voip
    allow-connections h323 to h323
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    h323
    h245 tunnel disable
    h245 caps mode restricted
    sip
    header-passing
    !
    !
    !
    voice class codec 1
    codec preference 1 g711ulaw
    codec preference 2 g711alaw
    !
    !
    !
    !
    !
    !
    !
    !
    !
    !
    !
    !
    !
    !
    !
    !
    !
    !
    interface FastEthernet0/0
    ip address 172.30.0.12 255.255.0.0
    duplex auto
    speed auto
    !
    ip forward-protocol nd
    ip route 0.0.0.0 0.0.0.0 172.30.0.2
    !
    !
    ip http server
    no ip http secure-server
    !
    !
    !
    !
    control-plane
    !
    !
    !
    voice-port 1/0/0
    disconnect-ack
    disc_pi_off
    description FXS Port 1/0/0
    station-id name test
    station-id number 707
    !
    voice-port 1/0/1
    !
    voice-port 1/1/0
    cptone GB
    timing hookflash-out 50
    connection plar opx 704
    !
    voice-port 1/1/1
    !
    !
    !
    !
    !
    dial-peer voice 600 pots
    destination-pattern 706
    port 1/1/0
    !
    dial-peer voice 402 voip
    destination-pattern 2…
    session protocol sipv2
    session target ipv4:172.30.0.32:5060
    codec g711ulaw
    no vad
    !
    dial-peer voice 401 voip
    destination-pattern 7..
    session protocol sipv2
    session target ipv4:172.30.0.32:5060
    codec g711ulaw
    no vad
    !
    dial-peer voice 8 pots
    destination-pattern 9[0-9]T
    port 1/1/0
    !
    dial-peer voice 700 pots
    destination-pattern 707
    port 1/0/0
    forward-digits 7
    !
    sip-ua
    registrar ipv4:172.30.0.12 expires 3600
    !
    !
    !
    !
    gatekeeper
    no shutdown
    !
    !
    telephony-service
    fxo hook-flash
    max-conferences 4 gain -6
    transfer-system full-blind
    transfer-pattern ..
    !
    !
    line con 0
    line aux 0
    line vty 0 4
    password xxxx login
    !
    !
    end

    ffvr1#

  33. Reply

    @Frank_Bauer

    You don’t say how you have Asterisk configured for outdialing so I need to make some guesses here.

    Just to be clear, you’re having trouble dialing the 707 extension pointing to 1/0/0? I’m not sure if it’s the problem but why are you forwarding 7 digits when you only have 3 in the dial pattern?

    Have you run some of the dial-peer debugs? Or perhaps a SIP debug? Make sure the destination of 707 is actually what’s being sent and that it’s properly matching against the dial-peer.

  34. Frank_Bauer

    Reply

    Thanks for the speedy reply. yes 707 extension points to 1/0/0

    Here is my asterisk config:
    host=172.30.0.12
    context=from-trunk
    type=friend
    allow=all
    disallow=none
    port=5060
    dtmfmode=rfc2833
    insecure=very

    I’m doing the debugs and will let you know what I see

  35. Frank_Bauer

    Reply

    Ok now we are getting somwehre, from the debug sip, I noticed that sip=ua was trying to peer with the routers own address, so I changed the regisrar address to the Asterisk, and dial peer 600 (FXO) came up as registered, but still no luck with dial peer 700.

    Tell, me do I also have to configure the extension Asterisk for it to register ?

  36. Reply

    Are you trying to register the FXS port as a SIP extension in Asterisk? Not sure how that would work. You should be able to establish a dial pattern in asterisk that points 707 to the SIP trunk with the router. But you will need to have a dial (destination?) pattern configured in Asterisk for that.

  37. Frank_Bauer

    Reply

    Yes, that’s exactly trying exactly this , I’m using the Freebox GUI version, any idea where to start for this?…

  38. Frank_Bauer

    Reply

    Done it , worked ! 🙂 You probably hear this all the time.. but.. LOVE YOUR WORK…. thanks man, really fast response, problem diagnosed and solved with a total VOIP novice….

  39. Reply

    Glad to hear it. It’s amazing how useful this post has been even though it’s over 3 years old now! I’m just astounded by the age of it.

  40. Frank_Bauer

    Reply

    Damnit.. spoke too soon, ok now the registration on the Gateway is timing out randomly.. any ideas ?

  41. Reply

    That’s a tough one without more info. Could be related to the timers in the sip-ua. Debug is your friend! Don’t forget to open a debug console on Asterisk to see what it’s trying to do.

  42. Frank_Bauer

    Reply

    .. thanks will look into it and get back to you , leaving to go home from work now, talk to you soon .

  43. Frank_Bauer

    Reply

    O.. I have narrowed it down to the “qualify=yes” settings on the asterisk trunk and extension settings,asterisk is sending sip options to the gateway every 10 seconds by default, if no response is recieved in 2 seconds by default, then asterisk considers the peers / extensions unreachable. I have played arround with the default 10 seconds, for example “qualify=3000” which makes the sip connection last for 3000 seconds, but then it goes down again. I’m looking into etending the time out, so I’m guessing the next logical step is to extend the hold default hold timer of 2 seconds.

  44. Matt Miller

    Reply

    Hi. I came across your page when trying to find a solution to my problem. I’ve got a 1760 (latest T-train IOS, max memory) with 4 FXO ports and 4 FXS ports. I’m currently using the 1760 to provide toll bypass to remote offices and I’d like to use it to provide voicemail for an analog system whose voicemail component died (I’d prefer to use Asterisk anyway, but the old one going is forcing me to push this to the top of my priority list).

    The FXO ports are configured with:

    connection plar opx 18005551212

    There are voice dial peers configured with 18005551212 as the destination pattern, so when a call comes in, it rings the first available FXS port (and remote offices can place a sip call to the 1760’s IP with the office’s main number as the destination). There are also dial peers configured for outbound calls to the analog lines as well as our remote locations.

    Everything to this point has been working flawlessly. I’ve setup a dialpeer to asterisk, so when I dial 1191 from one of the analog ports, it routes to asterisk. Again, this works. The problem I have is that I’m not sure how to configure call forward no answer for the FXO ports on the router. I know I could do it if I had 2 ATAs to provide the FXO ports, but I don’t want to invest in any more hardware.

    I’ve got a fix, but I’m looking for a better solution. Right now, I can use connection plar opx to send all incoming calls to 1191 (the sip connection to asterisk), then have asterisk place a sip call back to the 1760 to ring the FXS ports. Asterisk would have the timeout built in to the dial command to divert the call to voicemail. I’d like to avoid this scenario because the Asterisk server is on older hardware and I do not want all incoming calls to be dependant on the Asterisk server.
    If the server were to fail, there is nobody at that location that could troubleshoot/repair the server and if it goes out, it would prevent all incoming calls from ringing. I’m hoping that someone on here might have an idea on how I can get this to work. Maybe a hunt group, or something… I’m not sure how to do this since the dial-peer doesn’t have the CFNA options that ephones have.

  45. Matt Miller

    Reply

    @Matt Miller
    I should also add that the 1760 is not part of a larger call manager configuration. All of the toll bypass is done with static SIP connections between phone systems. I’ve seen mention to STCAPP online, but only in reference to call manager, not call manager express (which is what’s on the 1760). It sounds like what I’m trying to do, but I don’t know that it’s supported.

  46. Reply

    Matt, I’m not real clear on what your setup is. Sounds like you have PSTN lines on the FXO’s and other SIP trunks from offices. You have analog phones at that office on the FXS ports (not IP phones). You want the Asterisk to only provide VM and get out of the way for regular calls if it’s down.

    Best bet is probably the hunt group like you suggested. I don’t have time to build that right now but basically the first hop in the hunt group is the FXS extension and the second is the Asterisk trunk to the VM. If it gets to Asterisk it immediately goes to VM. Just be sure to work through the hunt group logic to be sure you aren’t black-holing the call if Asterisk is down.

  47. Matt Miller

    Reply

    You’re correct on my setup. I had played with the ephone-hunt command, but didn’t think it would work given that the fxs ports aren’t ephones. I did discover the voice hunt-group command that in theory will do what I need. I haven’t had any luck confugiring it from home. I’ll have to drive over to that office and play with it when I get some time. From what I could see from a debug voice dialpeer detail, it’s trying to match an incoming and outgoing dialpeer and I never get transferred to Asterisk. Asterisk never even shows an inbound SIP connection. I tried a sequential hunt group, but there are several other types available now, so I’ll have to mess with those. If I get something to work, I’ll post the configs in case someone else is looking to do something similar.

  48. Reply

    Hey i know this thread is old and i believe it got these all these hits because it really was ahead of its time…… I am currently in the process of implementing a Cisco 3825 as a gateway for 3CX……. has anybody attempted this and if so could you shed any light on the matter……… Thanks a Million Adam……..
    akusta@itlynk.com

  49. Jim

    Reply

    Hy all, I have made the interconnection wint 1760 and Asterisk, and everything is OK (For about 15 Minutes). After that I get in the CLI:
    ===============================
    set_destination: Parsing for address/port to send to
    set_destination: set destination to 192.168.0.56, port 5060
    Audio is at 172.16.12.59 port 18248
    Adding codec 0x8 (alaw) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    Reliably Transmitting (NAT) to 192.168.0.56:55350:
    INVITE sip:2310688184@192.168.0.56:5060 SIP/2.0
    Via: SIP/2.0/UDP 172.16.12.59:5060;branch=z9hG4bK69033545;rport
    Max-Forwards: 70
    From: ;tag=as1fc2060e
    To: ;tag=E8764B79-16C6
    Contact:
    Call-ID: FF2BA038-319711E0-9C32BD05-1C07ED8A@192.168.0.56
    CSeq: 102 INVITE
    User-Agent: FPBX-2.8.1(1.6.2.13)
    Require: timer
    Session-Expires: 1800;refresher=uas
    Min-SE: 90
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    X-asterisk-Info: SIP re-invite (Session-Timers)
    Content-Type: application/sdp
    Content-Length: 237

    v=0
    o=root 1270771100 1270771100 IN IP4 172.16.12.59
    s=Asterisk PBX 1.6.2.13
    c=IN IP4 172.16.12.59
    t=0 0
    m=audio 18248 RTP/AVP 8 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv

    SIP/2.0 420 Bad Extension
    Via: SIP/2.0/UDP 172.16.12.59:5060;branch=z9hG4bK69033545;rport
    From: ;tag=as1fc2060e
    To: ;tag=E8764B79-16C6
    Call-ID: FF2BA038-319711E0-9C32BD05-1C07ED8A@192.168.0.56
    CSeq: 102 INVITE
    Unsupported: timer
    Content-Length: 0

    — (8 headers 0 lines) —
    — Got SIP response 420 “Bad Extension” back from 192.168.0.56
    set_destination: Parsing for address/port to send to
    set_destination: set destination to 192.168.0.56, port 5060
    Transmitting (NAT) to 192.168.0.56:5060:
    ACK sip:2310688184@192.168.0.56:5060 SIP/2.0
    Via: SIP/2.0/UDP 172.16.12.59:5060;branch=z9hG4bK69033545;rport
    Max-Forwards: 70
    From: ;tag=as1fc2060e
    To: ;tag=E8764B79-16C6
    Contact:
    Call-ID: FF2BA038-319711E0-9C32BD05-1C07ED8A@192.168.0.56
    CSeq: 102 ACK
    User-Agent: FPBX-2.8.1(1.6.2.13)
    Content-Length: 0


    — Executing [h@macro-dial:1] Macro(“SIP/Cisco-0000009f”, “hangupcall”) in new stack
    — Executing [s@macro-hangupcall:1] GotoIf(“SIP/Cisco-0000009f”, “1?skiprg”) in new stack
    — Goto (macro-hangupcall,s,4)
    — Executing [s@macro-hangupcall:4] GotoIf(“SIP/Cisco-0000009f”, “0?skipblkvm”) in new stack
    — Executing [s@macro-hangupcall:5] NoOp(“SIP/Cisco-0000009f”, “Cleaning Up Block VM Flag: BLKVM/600/SIP/Cisco-0000009f”) in new stack
    — Executing [s@macro-hangupcall:6] NoOp(“SIP/Cisco-0000009f”, “Deleting: BLKVM/600/SIP/Cisco-0000009f “) in new stack
    — Executing [s@macro-hangupcall:7] GotoIf(“SIP/Cisco-0000009f”, “1?theend”) in new stack
    — Goto (macro-hangupcall,s,9)
    — Executing [s@macro-hangupcall:9] Hangup(“SIP/Cisco-0000009f”, “”) in new stack
    == Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/Cisco-0000009f’ in macro ‘hangupcall’
    == Spawn extension (macro-dial, s, 7) exited non-zero on ‘SIP/Cisco-0000009f’ in macro ‘dial’
    == Spawn extension (ext-group, 600, 14) exited non-zero on ‘SIP/Cisco-0000009f’
    == MixMonitor close filestream
    == End MixMonitor Recording SIP/Cisco-0000009f
    Elastix203*CLI>
    ===============================

    Have you ever met the same eroor anyone?

  50. Joe

    Reply

    Hi Scott, I’m unable to found “SIP” in this command “allow-connections sip to h323”. Device Cisco 1760, IOS version is 12.3(7)XR3. any clue? regards.

    • scott

      Reply

      Joe, it’s probably because of the feature set you’re running. I don’t recall the specific feature sets, but you’ll need something like IP Advanced. The basic feature set probably doesn’t include SIP.

  51. Ashish

    Reply

    Dear Scott,

    You must be feeling delighted when you see that your post from 2006 is still helping people till 2018.
    Thanks for this…

    • scott

      Reply

      Hopefully this isn’t a sneaky spam tactic. Regardless, it is still one of my most popular posts. 12 years! I guess Cisco hasn’t changed much in their code. Thanks!

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