Can register Sip Trunk but no voice??

Discussion in 'General' started by lawsu, Aug 28, 2009.

  1. lawsu

    Joined:
    Aug 28, 2009
    Messages:
    9
    Likes Received:
    0
    Hi,

    I am having real difficulty to get this working.

    My provider force me a 2Wire device for Sip call. I manage to get away and have success with Linksys PAP2T and softphone like X-lite. I just cannot get Asterisk to work.

    I manage to register the trunk.

    I can make outgoing call but there is no voice on both ends.

    I check log status and it return with
    "rtp.c: RTP Transmission error of packet to 0.0.0.0:0: Invalid argument"
    I am using a public IP address. I end up puting "nat and externip" to get rid of the error.

    Having getting rid of the error, I still have no voice when connected. I tried alot of combination of command still dun work.

    username=6xxxxxx
    type=peer
    secret=xxxxxxx
    qualify=yes
    nat=yes
    localhost=127.0.0.1
    insecure=very
    host=voip.xxxx.com
    fromdomain=voip.xxxx.com
    externip=6x.x.x.x
    dtmfmode=rfc2283
    disallow=all
    context=default
    auth=md5
    allow=ulaw&alaw
    relaxdtmf=yes

    Log files....

    I appreciate if any guru here can advice me further to next stage. thank you.
     
  2. Megabyte

    Joined:
    Mar 28, 2009
    Messages:
    327
    Likes Received:
    0
    Why do you have auth=md5 in your string?

    try like this maybe it could help you



    register => bootcamp3:bootcamp@172.16.79.63



    [to_sipprovider]
    type=peer
    username=bootcamp3
    fromuser=bootcamp3
    fromdomain=example.com
    secret=bootcamp
    canreinvite=no
    insecure=invite,port
    host=172.16.79.63
    deny=0.0.0.0/0
    permit=172.16.79.63/255.255.255.255
    disallow=all
    allow=gsm
    allow=ulaw
    allow=alaw
    qualify=yes
    nat=no


    And remember the register string
     
  3. lawsu

    Joined:
    Aug 28, 2009
    Messages:
    9
    Likes Received:
    0
    With "nat=no", give me this error again....
    "rtp.c: RTP Transmission error of packet to 0.0.0.0:0: Invalid argument"
    Not too sure why it is pointing to 0.0.0.0 ?????........ I turn on nat, the error is gone.

    I tried your setting, same thing, can register and connect, still no voice.....
     
  4. dicko

    Joined:
    Oct 24, 2008
    Messages:
    4,099
    Likes Received:
    0
    I don't believe you are clear about your network here,

    You say you have a public IP address,

    but you don't state your firewall/router configuration .

    If a public "network" (not an IP address, but a real internet network <ip network address>/<net mask> ) than all your devices must be within that network, , the interface must be configured correctly as to gateway/address/netmask and DNS server and there is no need for NAT, if you have a "private network" behind your router than , then yes you need

    externip=
    localnet=

    etc. in your sip*.conf file.

    It looks to me like you have a NAT solution behind a ISP provided NAT box

    further the attempts to connect rtp to 0.0.0.0 shows that it incorrectly configured. 0.0.0.0 is a network, and a very large one at that. rtp needs a host, which is atomic by nature.


    I suggests you start at the first page of "Elastix Without Tears" and continue until you understand now to set up Linux/Asterisk?Elastix behind a router.

    It has worked for most folks here. Unfortunately there is no substitute for comprehension :)
     
  5. lawsu

    Joined:
    Aug 28, 2009
    Messages:
    9
    Likes Received:
    0
    hi dicko......

    I am rather sure I have configured the network portion correctly.

    I tried configure another trunk with provider "voipuser.org" with nat=off in the same asterisk server and it did not give me problems nor any error as mention above. Strange right?

    By the way, I have updated my Elastix to 1.5.2-2.3.... still no go.....
     
  6. dicko

    Joined:
    Oct 24, 2008
    Messages:
    4,099
    Likes Received:
    0
    I'm sure you are sure, but just for shits and grins, please feel free to document your network from the ISP, the interface on your elastix box and any router/NAT settings between these two

    ( please obfuscate any public ip's)
     
  7. lawsu

    Joined:
    Aug 28, 2009
    Messages:
    9
    Likes Received:
    0
    No problem.

    ISP Router (Public IP)
    IP : X.X.X.161
    Sub : 255.255.255.224

    NAT Router ( I am not going thru NAT router )
    Wan : X.X.X.162 Sub : 255.255.255.224
    Lan : 192.168.1.1 Sub : 255.255.255.0


    Asterisk (Direct connection to Main ISP Router)
    IP : X.X.X.167
    Sub : 255.255.255.224
    Gateway : X.X.X.161
    DNS : ISP Provided. (No problem as I register sip as hostname and I manage to do yum update as well :) )

    I am actually OK with network setup but comes to Asterisk troubleshooting, I am bit lost.

    Thank you.......
     
  8. dicko

    Joined:
    Oct 24, 2008
    Messages:
    4,099
    Likes Received:
    0
    You've lost me here

    What is the

    Lan : 192.168.1.1 Sub : 255.255.255.0

    about, surely the router is acting either as a bridge to the identical local network or a NAT,

    If a bridge

    Then no NAT
    all your clients must be within the /255.255.255.224 network

    if NATting into 192.168.1.1, and your clients are there then the Elastix ifconfig doesn't compute if it is in the same physical network, if it's not then I suggest your are doin it the hard way.
     
  9. lawsu

    Joined:
    Aug 28, 2009
    Messages:
    9
    Likes Received:
    0
    Hi, please refer picture for my setup.

    My asterisk + client are not in NAT (192.168.1.X).

    I purposely put the Asterisk + Test Client on Public to eliminate any NAT related issues. [​IMG]
     
  10. dicko

    Joined:
    Oct 24, 2008
    Messages:
    4,099
    Likes Received:
    0
    Then from your original post


    assuming no NAT in sip*.conf (because it shopuldn't be there)

    username=6xxxxxx
    type=peer
    secret=xxxxxxx
    qualify=yes
    nat=yes ; WHY DO YOU HAVE THAT
    localhost=127.0.0.1 ; WHY DO YOU HAVE THAT
    insecure=very
    host=voip.xxxx.com
    fromdomain=voip.xxxx.com
    externip=6x.x.x.x ; SHOULD NOT BE NECESSARY
    dtmfmode=rfc2283
    disallow=all
    context=default ; ARE YOU SURE ABOUT THAT
    auth=md5 ; ARE YOU SURE ABOUT THAT
    allow=ulaw&alaw
    relaxdtmf=yes ; ARE YOU SURE ABOUT THAT


    I suggest a leisurely visit to http://voip-info.org will elucidate Asterisk for you

    I further suggest a hardware phone/ata for testing preferably two, then you can at least talk to yourself(less than 30 bucks each on e-bay) especially if your host machine with xlite is in any way compromised (i.e. by OS?)
     
  11. lawsu

    Joined:
    Aug 28, 2009
    Messages:
    9
    Likes Received:
    0
    actually, when I first started off, it was like few key statements.

    To be frank, I am not sure. I done some reading on working setup includiing Elastix Without Tears few months ago and other asterisk website. Trying to play around with lots of different combination.

    dicko, do you think you can test on my problem if I send you sip details?? No obligations tho.... :)
     
  12. dicko

    Joined:
    Oct 24, 2008
    Messages:
    4,099
    Likes Received:
    0
    To be quite honest, I suggest you just try "Elastix Without Tears" one more time, this time with feeling, it has been brought up to 1.5 date and it has worked for the vast majority of others here. I don't feel there is any point doing it over another way.
     
  13. lawsu

    Joined:
    Aug 28, 2009
    Messages:
    9
    Likes Received:
    0
    I just read the Trunk portion 8.4.

    I remove my existing trunk and create again.

    username=xxxxXXXX
    type=peer
    secret= Password
    insecure=very
    host=voip.provider.com
    dtmfmode=rfc2833
    disallow=all
    allow=alaw
    canreinvite=no

    When I use "ulaw" it throws me "rtp.c: RTP Transmission error of packet to 0.0.0.0:0: Invalid argument"

    There is no error when I use "alaw".

    With both tries, there was no nat entry in trunk setting. (therefore, its off)

    However, both attempts still can connect but no voice.
     
  14. dicko

    Joined:
    Oct 24, 2008
    Messages:
    4,099
    Likes Received:
    0
    Then I would investigate how x.x.x.161 handles rtp connections (the audio payload)

    ulaw and alaw are not arbitrary, you MUST use codecs that are compatible with your VSP or definityely will have no audio.

    You state:
    With both tries, there was no nat entry in trunk setting. (therefore, its off)

    this is not necessarily so, the trunk will inherit any settings in the /etc/asterisk/sip*.conf family of files.

    http://voip-info.org will help you understand this inheritance behavior in Asterisk

    Maybe it's now time for the network section of EWT ?
     
  15. lawsu

    Joined:
    Aug 28, 2009
    Messages:
    9
    Likes Received:
    0
    I tried ulaw and alaw using x-lite to connect to my sip provider. It works without any problem. Which should mean that both codecs should be supported.

    I have verify that there is no NAT entry in sip*.conf as well.

    Sorry.. what is EWT and where can I find it??

    Do you happen to know of anyone who can solve my problem?? I do not mind for a small token if required as this problem been dragging me for quite sometime.
     
  16. dicko

    Joined:
    Oct 24, 2008
    Messages:
    4,099
    Likes Received:
    0
    EWT="Elastix Without Tears"

    Palosanto our, benevolent sponsors,provide paid support, the link is available on the home page.

    Good Luck
     
  17. lawsu

    Joined:
    Aug 28, 2009
    Messages:
    9
    Likes Received:
    0
    I tried to drop them an email, till now no response.

    I read the EWT but still going nowhere.

    Any1??? Else will have to shelve this idea and at mercy of my ISP. Haiz.....
     
  18. e90

    e90

    Joined:
    Sep 14, 2009
    Messages:
    2
    Likes Received:
    0
    Lawsu,

    I have the same problem with this voip provider (looking at your error log). It seems to be related to how they are handling the RTP stream. I've opened UDP 10000-35000 and adjusted rtp.conf with no luck. Registration is fine, but no audio on outbound calls. Inbound calls are crystal clear though.

    Seems the combination of Asterisk, 2Wire NAT, and MIO voip is problematic. If anyone has gotten this to work, please share. My guess is that this provider is intentionally banning Asterisk calls.

    So for now, this trunk serves as my DISA until I can get outbound calls working.
     
  19. e90

    e90

    Joined:
    Sep 14, 2009
    Messages:
    2
    Likes Received:
    0
    Lawsu's log shows MOH being triggered for outbound calls. Anyone know how to prevent this from happening? Could be what's causing no audio on outbound calls.

    [Aug 29 00:19:46] VERBOSE[2817] logger.c: -- Called MIO/9xxxxxxx
    [Aug 29 00:19:49] VERBOSE[2817] logger.c: -- Call on SIP/MIO-0950de50 placed on hold
    [Aug 29 00:19:49] VERBOSE[2817] logger.c: -- Started music on hold, class 'default', on SIP/100-09573268
    [Aug 29 00:19:49] VERBOSE[2817] logger.c: -- SIP/MIO-0950de50 is making progress passing it to SIP/100-09573268
    [Aug 29 00:19:49] VERBOSE[2817] logger.c: -- SIP/MIO-0950de50 is ringing
    [Aug 29 00:19:49] VERBOSE[2817] logger.c: -- Call on SIP/MIO-0950de50 placed on hold
    [Aug 29 00:19:49] VERBOSE[2817] logger.c: -- Stopped music on hold on SIP/100-09573268
    [Aug 29 00:19:49] VERBOSE[2817] logger.c: -- Started music on hold, class 'default', on SIP/100-09573268
    [Aug 29 00:19:49] VERBOSE[2817] logger.c: -- SIP/MIO-0950de50 is making progress passing it to SIP/100-09573268
    [Aug 29 00:19:53] VERBOSE[2817] logger.c: -- Call on SIP/MIO-0950de50 placed on hold
     
  20. numantis

    Joined:
    Apr 14, 2011
    Messages:
    6
    Likes Received:
    0
    here is the answer to this aged question:

    Go into unembedded freepbx > tools > sip settings
    and configure the subnet for your network, then all is well.
     

Share This Page