Short calls with voip.co.uk

Discussion in 'UK VOIP' started by Tony Arnold, Jan 16, 2006.

  1. Tony Arnold

    Tony Arnold Guest

    I've recently signed to use voip.co.uk as a bit of an experiment. I'm
    using Ubuntu Linux and the Linphone package. I can call PSTN numbers but I
    get cut off after about 100 seconds or so. Below is my configuration for
    Linphone. Does this look right? It seems to work, but I had to play around
    to get it to do so. Oh, I've replaced my id with nnnnnn and my password with ******

    Regards,
    Tony.

    [net]
    con_type=3
    use_nat=0
    nat_address=81.86.172.58

    [sip]
    sip_port=5060
    guess_hostname=0
    contact=sip:
    use_info=0
    use_ipv6=0
    default_proxy=0

    [rtp]
    audio_rtp_port=7078
    video_rtp_port=9078
    audio_jitt_comp=60
    video_jitt_comp=60

    [sound]
    dev_id=1
    rec_lev=80
    play_lev=80
    source=m
    local_ring=/usr/share/sounds/linphone/rings/oldphone.wav
    remote_ring=/usr/share/sounds/linphone/ringback.wav

    [video]
    enabled=0
    show_local=1

    [audio_codec_0]
    mime=PCMU
    rate=8000
    enabled=1

    [audio_codec_1]
    mime=GSM
    rate=8000
    enabled=1

    [audio_codec_2]
    mime=PCMA
    rate=8000
    enabled=1

    [audio_codec_3]
    mime=speex
    rate=8000
    enabled=1

    [audio_codec_4]
    mime=speex
    rate=16000
    enabled=1

    [audio_codec_5]
    mime=1015
    rate=8000
    enabled=1

    [proxy_0]
    reg_proxy=sip:proxy.voip.co.uk
    reg_identity=sip:
    reg_expires=120
    reg_sendregister=1
    publish=1

    [auth_info_0]
    username=nnnnn
    passwd=*******
    realm="voip.co.uk"

    [auth_info_1]
    username=nnnnnn
    passwd=******
    realm="80.249.108.5"
     
    Tony Arnold, Jan 16, 2006
    #1
    1. Advertising

  2. Tony Arnold

    NutCracker Guest

    On Mon, 16 Jan 2006 22:51:05 +0000, Tony Arnold <> wrote:

    >I've recently signed to use voip.co.uk as a bit of an experiment. I'm
    >using Ubuntu Linux and the Linphone package. I can call PSTN numbers but I
    >get cut off after about 100 seconds or so. Below is my configuration for
    >Linphone. Does this look right? It seems to work, but I had to play around
    >to get it to do so. Oh, I've replaced my id with nnnnnn and my password with ******
    >
    >Regards,
    >Tony.
    >
    >[net]
    >con_type=3
    >use_nat=0
    >nat_address=81.86.172.58
    >
    >[sip]
    >sip_port=5060
    >guess_hostname=0
    >contact=sip:
    >use_info=0
    >use_ipv6=0
    >default_proxy=0
    >
    >[rtp]
    >audio_rtp_port=7078
    >video_rtp_port=9078
    >audio_jitt_comp=60
    >video_jitt_comp=60
    >
    >[sound]
    >dev_id=1
    >rec_lev=80
    >play_lev=80
    >source=m
    >local_ring=/usr/share/sounds/linphone/rings/oldphone.wav
    >remote_ring=/usr/share/sounds/linphone/ringback.wav
    >
    >[video]
    >enabled=0
    >show_local=1
    >
    >[audio_codec_0]
    >mime=PCMU
    >rate=8000
    >enabled=1
    >
    >[audio_codec_1]
    >mime=GSM
    >rate=8000
    >enabled=1
    >
    >[audio_codec_2]
    >mime=PCMA
    >rate=8000
    >enabled=1
    >
    >[audio_codec_3]
    >mime=speex
    >rate=8000
    >enabled=1
    >
    >[audio_codec_4]
    >mime=speex
    >rate=16000
    >enabled=1
    >
    >[audio_codec_5]
    >mime=1015
    >rate=8000
    >enabled=1
    >
    >[proxy_0]
    >reg_proxy=sip:proxy.voip.co.uk
    >reg_identity=sip:
    >reg_expires=120
    >reg_sendregister=1
    >publish=1
    >
    >[auth_info_0]
    >username=nnnnn
    >passwd=*******
    >realm="voip.co.uk"
    >
    >[auth_info_1]
    >username=nnnnnn
    >passwd=******
    >realm="80.249.108.5"
    >



    Your lucky! I'm using a Sipura SPA-3000 and I get disconnected after about
    45-50 seconds! Have reported it to voip.co.uk.
     
    NutCracker, Jan 17, 2006
    #2
    1. Advertising

  3. Tony Arnold

    alexd Guest

    Tony Arnold wrote:

    > I've recently signed to use voip.co.uk as a bit of an experiment. I'm
    > using Ubuntu Linux and the Linphone package.

    <snip>

    Whatever you do with Linphone, make sure you use it with ALSA instead of
    OSS, as I've found it to be a complete and utter pig with OSS.

    Might be worth watching your network traffic with gkrellm or iptraf to see
    which side of the call gets cut off. Also, drop into an Asterisk console
    with 'asterisk -rc' and watch for error messages. Whilst in the console,
    'sip show peers', 'sip debug' and 'sip no debug' can be helpful.
    And one more thing, I found creating an exten that lets one record a
    message and play it back, helpful for isolating issues with particual SIP
    peers.

    alexd
    --
    <http://ale.cx/> (AIM:troffasky) ()
    19:23:22 up 1 day, 23:38, 2 users, load average: 0.00, 0.05, 0.07
    This is my BOOOOOOOOOOOOOOOOOOOOOMSTICK
     
    alexd, Jan 17, 2006
    #3
  4. Tony Arnold

    Tony Arnold Guest

    Thanks. I've had a response from the support people at voip.co.uk saying
    they are aware of this problem with some devices. They are planning a
    software upgrade to address the problem this week-end, so hopefully on
    Monday all will be well again!

    Regards,
    Tony.

    On Tue, 17 Jan 2006 19:33:04 +0000, alexd
    wrote:

    > Tony Arnold wrote:
    >
    >> I've recently signed to use voip.co.uk as a bit of an experiment. I'm
    >> using Ubuntu Linux and the Linphone package.

    > <snip>
    >
    > Whatever you do with Linphone, make sure you use it with ALSA instead of
    > OSS, as I've found it to be a complete and utter pig with OSS.
    >
    > Might be worth watching your network traffic with gkrellm or iptraf to see
    > which side of the call gets cut off. Also, drop into an Asterisk console
    > with 'asterisk -rc' and watch for error messages. Whilst in the console,
    > 'sip show peers', 'sip debug' and 'sip no debug' can be helpful.
    > And one more thing, I found creating an exten that lets one record a
    > message and play it back, helpful for isolating issues with particual SIP
    > peers.
    >
    > alexd
     
    Tony Arnold, Jan 20, 2006
    #4
    1. Advertising

Want to reply to this thread or ask your own question?

It takes just 2 minutes to sign up (and it's free!). Just click the sign up button to choose a username and then you can ask your own questions on the forum.
Similar Threads
  1. Brian McCrary
    Replies:
    2
    Views:
    1,207
    Anthony
    Jul 1, 2005
  2. Vox Humana

    voip to voip calls

    Vox Humana, Mar 11, 2005, in forum: VOIP
    Replies:
    2
    Views:
    565
    Vox Humana
    Mar 11, 2005
  3. ubifone
    Replies:
    0
    Views:
    4,545
    ubifone
    Jul 29, 2005
  4. joseph
    Replies:
    3
    Views:
    1,349
  5. Giganews
    Replies:
    27
    Views:
    2,439
    Brian
    Oct 10, 2006
Loading...

Share This Page