Sangoma A102 SS7 setup

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Setup, which is currently in operation:

below you can find my working eamples of wanpipe.cfg, ss7.conf, and extensions.conf from the box, which is currently interconnected with mobile operators switch.

wanpipe1.conf



[devices]
wanpipe1 = WAN_AFT, Comment

[interfaces]
w1g1 = wanpipe1, , TDM_VOICE, Comment

[wanpipe1]
CARD_TYPE = AFT
S514CPU = A
CommPort = PRI
AUTO_PCISLOT = NO
PCISLOT = 9
PCIBUS = 2
FE_MEDIA = E1
FE_LCODE = HDB3
FE_FRAME = NCRC4
FE_LINE = 1
TE_CLOCK = NORMAL
TE_REF_CLOCK = 0
ACTIVE_CH = ALL
TE_HIGHIMPEDANCE = NO
FE_TXTRISTATE = NO
MTU = 1500
UDPPORT = 9000
TTL = 255
IGNORE_FRONT_END = NO
TDMV_SPAN = 1
TDMV_DCHAN = 0

[w1g1]
ACTIVE_CH = ALL
TDMV_ECHO_OFF = NO


ss7.conf



[linkset-siuc]
enabled => yes
enable_st => no
use_connect => yes
hunting_policy => even_mru

context => ss7
language => da

[link-l1]
linkset => siuc

channels => 1-15,17-31
schannel => 16
firstcic => 1
enabled => yes

[link-l2]
linkset => siuc
channels => 1-15,17-31
schannel => 16
firstcic => 1
enabled => yes

[host-ss7-1]
enabled => yes
opc => 9000
dpc => siuc:2155
links => l1:1

extensions.conf



[ss7]
exten => _00555.,1,Dial(SIP/${EXTEN:5})
exten => _00556.,1,Dial(OOH323/${EXTEN:5})

[local]
include => ss7



If, while interconnecting two asterisk boxes, you will get framing errors with backlog from chan_ss7 that it switches to byte-counting mode - try to setup an option
"True Encoding Type" in "Interface setup->Advanced" to "Enabled" state.

Note, that host-ss7-1 have to match the host name of the box, where asterisk have been installed, since chan_ss7 configures itself by finding host-matching config this way.

Found issues:


1. While calling from mobile phone through chan_ss7, the customer does not receive an RBT (Ring Back Tone), so there is a complete silenece, until called party picks-up the telephone. This issue is most likely an issue of ITU Erricson ISUP dialect. I did not sound any info how to tune chan_ss7 to use different dialects, but seems there is a support for a different dialects, but anyway i did hardcoded the difference in the code. See chan_ss7 quick patch to enable RBT
2. While using free g723.1 codec there was output buffer overflow errors, reported bu chan_ss7, that does not happen while using free g729A codec from Intel.
3. Having a segmentation fault, when issuing "stop now" asterisk command, but can't say now, is is chan_ss7 issue or chan_h323.


Asterisk ss7 channels

Created: Sat 25 of Feb, 2006 (13:46 UTC)
HITS: 21413

Anton V. Gnitko
http://www.eastera.net

Where to Buy

  • INDIA - Alliance Infotech | National distributors in INDIA.Phn:9811334456, abhishek.baid@alliance-infotech.com
  • USA - sooho-voip-phone - Best Prices and fast shipping on Sangoma A102 card
  • UK & Worldwide - VoIPon - Best Prices and Support on Sangoma PRI Cards - Call for reseller pricing or International Shipping. VoIPon.
  • Romania - www.modulo.ro - Sangoma Reseller for the Romanian market
Setup, which is currently in operation:

below you can find my working eamples of wanpipe.cfg, ss7.conf, and extensions.conf from the box, which is currently interconnected with mobile operators switch.

wanpipe1.conf



[devices]
wanpipe1 = WAN_AFT, Comment

[interfaces]
w1g1 = wanpipe1, , TDM_VOICE, Comment

[wanpipe1]
CARD_TYPE = AFT
S514CPU = A
CommPort = PRI
AUTO_PCISLOT = NO
PCISLOT = 9
PCIBUS = 2
FE_MEDIA = E1
FE_LCODE = HDB3
FE_FRAME = NCRC4
FE_LINE = 1
TE_CLOCK = NORMAL
TE_REF_CLOCK = 0
ACTIVE_CH = ALL
TE_HIGHIMPEDANCE = NO
FE_TXTRISTATE = NO
MTU = 1500
UDPPORT = 9000
TTL = 255
IGNORE_FRONT_END = NO
TDMV_SPAN = 1
TDMV_DCHAN = 0

[w1g1]
ACTIVE_CH = ALL
TDMV_ECHO_OFF = NO


ss7.conf



[linkset-siuc]
enabled => yes
enable_st => no
use_connect => yes
hunting_policy => even_mru

context => ss7
language => da

[link-l1]
linkset => siuc

channels => 1-15,17-31
schannel => 16
firstcic => 1
enabled => yes

[link-l2]
linkset => siuc
channels => 1-15,17-31
schannel => 16
firstcic => 1
enabled => yes

[host-ss7-1]
enabled => yes
opc => 9000
dpc => siuc:2155
links => l1:1

extensions.conf



[ss7]
exten => _00555.,1,Dial(SIP/${EXTEN:5})
exten => _00556.,1,Dial(OOH323/${EXTEN:5})

[local]
include => ss7



If, while interconnecting two asterisk boxes, you will get framing errors with backlog from chan_ss7 that it switches to byte-counting mode - try to setup an option
"True Encoding Type" in "Interface setup->Advanced" to "Enabled" state.

Note, that host-ss7-1 have to match the host name of the box, where asterisk have been installed, since chan_ss7 configures itself by finding host-matching config this way.

Found issues:


1. While calling from mobile phone through chan_ss7, the customer does not receive an RBT (Ring Back Tone), so there is a complete silenece, until called party picks-up the telephone. This issue is most likely an issue of ITU Erricson ISUP dialect. I did not sound any info how to tune chan_ss7 to use different dialects, but seems there is a support for a different dialects, but anyway i did hardcoded the difference in the code. See chan_ss7 quick patch to enable RBT
2. While using free g723.1 codec there was output buffer overflow errors, reported bu chan_ss7, that does not happen while using free g729A codec from Intel.
3. Having a segmentation fault, when issuing "stop now" asterisk command, but can't say now, is is chan_ss7 issue or chan_h323.


Asterisk ss7 channels

Created: Sat 25 of Feb, 2006 (13:46 UTC)
HITS: 21413

Anton V. Gnitko
http://www.eastera.net

Where to Buy

  • INDIA - Alliance Infotech | National distributors in INDIA.Phn:9811334456, abhishek.baid@alliance-infotech.com
  • USA - sooho-voip-phone - Best Prices and fast shipping on Sangoma A102 card
  • UK & Worldwide - VoIPon - Best Prices and Support on Sangoma PRI Cards - Call for reseller pricing or International Shipping. VoIPon.
  • Romania - www.modulo.ro - Sangoma Reseller for the Romanian market
Created by: vazir, Last modification: Thu 26 of Jul, 2012 (10:33 UTC) by motorcicle
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