Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 134, Issue 1

[Kaloyan Kovachev]

Hi,
 can you try with Asterisk 13 instead and the same version of libss7.
 This looks as mismatch by 1 bug, but for near 4 years (with 13.x) never 
had similar problems, so it is most likely introduced at some later 
point.


На 2018-02-20 16:24, Dovid Bender написа:

Asterisk version 15.1.4. Using the latest from 
https://github.com/asterisk/libss7


On Mon, Feb 19, 2018 at 1:19 PM, Alexandr Dranchuk 
 wrote:


Any errors in logs?

And just to make sure. Can you tell versions of asterisk and ss7?

20 февр. 2018 г. 0:00 пользователь 
 написал:

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than "Re: Contents of asterisk-ss7 digest..."

Today's Topics:

1. CICS stay in pending state (Dovid Bender)

--

Message: 1
Date: Sun, 18 Feb 2018 22:56:06 -0500
From: Dovid Bender 
To: asterisk-ss7@lists.digium.com
Subject: [asterisk-ss7] CICS stay in pending state
Message-ID:

Content-Type: text/plain; charset="utf-8"

Hi,

We have an issue where some CICs stay in pending and wont go into Idle. 
If
we do a block the other side see's it. The same goes for an unblock. 
For
some reason no matter what we do certain cic's (it seems to be random) 
will

stay in pending while the remote side see's them as Idle. Some times a
reset will free it up, other times it wont. When doing a reset the 
other
side responds that the cic has been reset yet Asterisk wont free it up. 
Any

idea whats going on?

[root@ast01 asterisk]# asterisk -rx'ss7 show cics 1' | grep Pending
2   518  2  Pending
[root@ast01 asterisk]#

st01*CLI> ss7 reset cic 1 518 2
Sent RSC for linkset 1 on CIC 2 DPC 518
[1] Len = 11 [ fe ac 08 85 06 c2 98 20 02 00 12 ]
[1] FSN: 44 FIB 1
[1] BSN: 126 BIB 1
[1] >[520:0] MSU
[1] [ fe ac 08 ]
[1] Network Indicator: 2 Priority: 0 User Part: ISUP (5)
[1] [ 85 ]
[1] OPC 611 DPC 518 SLS 2
[1] [ 06 c2 98 20 ]
[1] CIC: 2
[1] [ 02 00 ]
[1] Message Type: RSC(0x12)
[1] [ 12 ]
[1]
[1] Len = 12 [ ac ff 09 85 63 82 81 20 02 00 10 00 ]
[1] FSN: 127 FIB 1
[1] BSN: 44 BIB 1
[1] <[520:0] MSU
[1] [ ac ff 09 ]
[1] Network Indicator: 2 Priority: 0 User Part: ISUP (5)
[1] [ 85 ]
[1] OPC 518 DPC 611 SLS 2
[1] [ 63 82 81 20 ]
[1] CIC: 2
[1] [ 02 00 ]
[1] Message Type: RLC(0x10)
[1] [ 10 ]
[1]
Linkset 1: Processing event: ISUP_EVENT_RLC
ast01*CLI>
ast01*CLI>
ast01*CLI> quit
Asterisk cleanly ending (0).
Executing last minute cleanups
[root@ast01 asterisk]# asterisk -rx'ss7 show cics 1' | grep Pending
2   518  2  Pending
[root@ast01 asterisk]#

TIA.

Dovid
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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 134, Issue 1

[Dovid Bender]

Asterisk version 15.1.4. Using the latest from
https://github.com/asterisk/libss7




On Mon, Feb 19, 2018 at 1:19 PM, Alexandr Dranchuk 
wrote:

> Any errors in logs?
>
> And just to make sure. Can you tell versions of asterisk and ss7?
>
> 20 февр. 2018 г. 0:00 пользователь 
> написал:
>
>> Send asterisk-ss7 mailing list submissions to
>> asterisk-ss7@lists.digium.com
>>
>> To subscribe or unsubscribe via the World Wide Web, visit
>> http://lists.digium.com/mailman/listinfo/asterisk-ss7
>> or, via email, send a message with subject or body 'help' to
>> asterisk-ss7-requ...@lists.digium.com
>>
>> You can reach the person managing the list at
>> asterisk-ss7-ow...@lists.digium.com
>>
>> When replying, please edit your Subject line so it is more specific
>> than "Re: Contents of asterisk-ss7 digest..."
>>
>>
>> Today's Topics:
>>
>>1. CICS stay in pending state (Dovid Bender)
>>
>>
>> --
>>
>> Message: 1
>> Date: Sun, 18 Feb 2018 22:56:06 -0500
>> From: Dovid Bender 
>> To: asterisk-ss7@lists.digium.com
>> Subject: [asterisk-ss7] CICS stay in pending state
>> Message-ID:
>> > ail.com>
>> Content-Type: text/plain; charset="utf-8"
>>
>> Hi,
>>
>> We have an issue where some CICs stay in pending and wont go into Idle. If
>> we do a block the other side see's it. The same goes for an unblock. For
>> some reason no matter what we do certain cic's (it seems to be random)
>> will
>> stay in pending while the remote side see's them as Idle. Some times a
>> reset will free it up, other times it wont. When doing a reset the other
>> side responds that the cic has been reset yet Asterisk wont free it up.
>> Any
>> idea whats going on?
>>
>> [root@ast01 asterisk]# asterisk -rx'ss7 show cics 1' | grep Pending
>> 2   518  2  Pending
>> [root@ast01 asterisk]#
>>
>>
>> st01*CLI> ss7 reset cic 1 518 2
>> Sent RSC for linkset 1 on CIC 2 DPC 518
>> [1] Len = 11 [ fe ac 08 85 06 c2 98 20 02 00 12 ]
>> [1] FSN: 44 FIB 1
>> [1] BSN: 126 BIB 1
>> [1] >[520:0] MSU
>> [1] [ fe ac 08 ]
>> [1] Network Indicator: 2 Priority: 0 User Part: ISUP (5)
>> [1] [ 85 ]
>> [1] OPC 611 DPC 518 SLS 2
>> [1] [ 06 c2 98 20 ]
>> [1] CIC: 2
>> [1] [ 02 00 ]
>> [1] Message Type: RSC(0x12)
>> [1] [ 12 ]
>> [1]
>> [1] Len = 12 [ ac ff 09 85 63 82 81 20 02 00 10 00 ]
>> [1] FSN: 127 FIB 1
>> [1] BSN: 44 BIB 1
>> [1] <[520:0] MSU
>> [1] [ ac ff 09 ]
>> [1] Network Indicator: 2 Priority: 0 User Part: ISUP (5)
>> [1] [ 85 ]
>> [1] OPC 518 DPC 611 SLS 2
>> [1] [ 63 82 81 20 ]
>> [1] CIC: 2
>> [1] [ 02 00 ]
>> [1] Message Type: RLC(0x10)
>> [1] [ 10 ]
>> [1]
>> Linkset 1: Processing event: ISUP_EVENT_RLC
>> ast01*CLI>
>> ast01*CLI>
>> ast01*CLI> quit
>> Asterisk cleanly ending (0).
>> Executing last minute cleanups
>> [root@ast01 asterisk]# asterisk -rx'ss7 show cics 1' | grep Pending
>> 2   518  2  Pending
>> [root@ast01 asterisk]#
>>
>>
>>
>> TIA.
>>
>> Dovid
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>> 
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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 134, Issue 1

[Alexandr Dranchuk]

Any errors in logs?

And just to make sure. Can you tell versions of asterisk and ss7?

20 февр. 2018 г. 0:00 пользователь 
написал:

> Send asterisk-ss7 mailing list submissions to
> asterisk-ss7@lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-ss7
> or, via email, send a message with subject or body 'help' to
> asterisk-ss7-requ...@lists.digium.com
>
> You can reach the person managing the list at
> asterisk-ss7-ow...@lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-ss7 digest..."
>
>
> Today's Topics:
>
>1. CICS stay in pending state (Dovid Bender)
>
>
> --
>
> Message: 1
> Date: Sun, 18 Feb 2018 22:56:06 -0500
> From: Dovid Bender 
> To: asterisk-ss7@lists.digium.com
> Subject: [asterisk-ss7] CICS stay in pending state
> Message-ID:
>  gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Hi,
>
> We have an issue where some CICs stay in pending and wont go into Idle. If
> we do a block the other side see's it. The same goes for an unblock. For
> some reason no matter what we do certain cic's (it seems to be random) will
> stay in pending while the remote side see's them as Idle. Some times a
> reset will free it up, other times it wont. When doing a reset the other
> side responds that the cic has been reset yet Asterisk wont free it up. Any
> idea whats going on?
>
> [root@ast01 asterisk]# asterisk -rx'ss7 show cics 1' | grep Pending
> 2   518  2  Pending
> [root@ast01 asterisk]#
>
>
> st01*CLI> ss7 reset cic 1 518 2
> Sent RSC for linkset 1 on CIC 2 DPC 518
> [1] Len = 11 [ fe ac 08 85 06 c2 98 20 02 00 12 ]
> [1] FSN: 44 FIB 1
> [1] BSN: 126 BIB 1
> [1] >[520:0] MSU
> [1] [ fe ac 08 ]
> [1] Network Indicator: 2 Priority: 0 User Part: ISUP (5)
> [1] [ 85 ]
> [1] OPC 611 DPC 518 SLS 2
> [1] [ 06 c2 98 20 ]
> [1] CIC: 2
> [1] [ 02 00 ]
> [1] Message Type: RSC(0x12)
> [1] [ 12 ]
> [1]
> [1] Len = 12 [ ac ff 09 85 63 82 81 20 02 00 10 00 ]
> [1] FSN: 127 FIB 1
> [1] BSN: 44 BIB 1
> [1] <[520:0] MSU
> [1] [ ac ff 09 ]
> [1] Network Indicator: 2 Priority: 0 User Part: ISUP (5)
> [1] [ 85 ]
> [1] OPC 518 DPC 611 SLS 2
> [1] [ 63 82 81 20 ]
> [1] CIC: 2
> [1] [ 02 00 ]
> [1] Message Type: RLC(0x10)
> [1] [ 10 ]
> [1]
> Linkset 1: Processing event: ISUP_EVENT_RLC
> ast01*CLI>
> ast01*CLI>
> ast01*CLI> quit
> Asterisk cleanly ending (0).
> Executing last minute cleanups
> [root@ast01 asterisk]# asterisk -rx'ss7 show cics 1' | grep Pending
> 2   518  2  Pending
> [root@ast01 asterisk]#
>
>
>
> TIA.
>
> Dovid
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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 111, Issue 2

[Vallimamod Abdullah]

Hi,

Your initial request to not send SLTM looks solved. Now, you are receiving SIOS 
(out of service indication) from your provider after the link is in service: 

> [Jun  8 23:50:04] NOTICE[7939]: mtp.c:1148 mtp2_process_lssu: Got status
> indication 'OS' while INSERVICE on link 'l16’.

> [Jun  8 23:50:04] NOTICE[7939]: mtp.c:1148 mtp2_process_lssu: Got status
> indication 'OS' while INSERVICE on link 'l1’.

The best way to debug this would be to test with just one link to start and ask 
your provider what they are seeing on their side.
There are many possible causes (hardware echo cancellation, clock 
synchronisation…) It’s impossible to guess.

Post your full hardware specs and configs on the list and maybe someone will 
spot the problem.

Please always answer to the list, not to me directly.

Best Regards,
Vallimamod
.

On 08 Jun 2014, at 20:53, shazi .  wrote:

> Dear Vallimamod,
> 
> Thanks for you kind and quick response, Links still fluctutating after
> upgrading Chan_ss7 ver 2.2.0.
> 
> PFA ss7.conf, please help.
> 
> [Jun  8 23:50:02] NOTICE[7939]: mtp.c:1296 mtp2_good_frame: Sending TRA to
> peer on link 'l16'
> [Jun  8 23:50:02] WARNING[7939]: chan_ss7.c:121 process_event: MTP is now UP
> on link 'l1'.
> [Jun  8 23:50:02] NOTICE[7939]: mtp.c:1296 mtp2_good_frame: Sending TRA to
> peer on link 'l1'
> [Jun  8 23:50:04] NOTICE[7939]: mtp.c:1148 mtp2_process_lssu: Got status
> indication 'OS' while INSERVICE on link 'l16'.
> [Jun  8 23:50:04] WARNING[7939]: chan_ss7.c:125 process_event: MTP is now
> DOWN on link 'l16'.
> [Jun  8 23:50:04] NOTICE[7939]: mtp.c:665 mtp_changeover: MTP changeover
> last_ack=0, last_sent=0, from schannel 16, no INSERVICE schannel found
> [Jun  8 23:50:04] NOTICE[7939]: mtp.c:669 mtp_changeover: Failover not
> possible, no other signalling link and no other host available.
> [Jun  8 23:50:04] WARNING[7939]: chan_ss7.c:125 process_event: MTP is now
> DOWN on link 'l16'.
> [Jun  8 23:50:04] NOTICE[7939]: mtp.c:1148 mtp2_process_lssu: Got status
> indication 'OS' while INSERVICE on link 'l1'.
> [Jun  8 23:50:04] WARNING[7939]: chan_ss7.c:125 process_event: MTP is now
> DOWN on link 'l1'.
> [Jun  8 23:50:04] NOTICE[7939]: mtp.c:665 mtp_changeover: MTP changeover
> last_ack=0, last_sent=0, from schannel 16, no INSERVICE schannel found
> [Jun  8 23:50:04] NOTICE[7939]: mtp.c:669 mtp_changeover: Failover not
> possible, no other signalling link and no other host available.
> [Jun  8 23:50:04] WARNING[7939]: chan_ss7.c:125 process_event: MTP is now
> DOWN on link 'l1'.
> MS2*CLI> ss7 version
> chan_ss7 version 2.2.0
> [Jun  8 23:50:05] WARNING[7939]: chan_ss7.c:121 process_event: MTP is now UP
> on link 'l16'.
> [Jun  8 23:50:05] NOTICE[7939]: mtp.c:1296 mtp2_good_frame: Sending TRA to
> peer on link 'l16'
> [Jun  8 23:50:06] WARNING[7939]: chan_ss7.c:121 process_event: MTP is now UP
> on link 'l1'.
> [Jun  8 23:50:06] NOTICE[7939]: mtp.c:1296 mtp2_good_frame: Sending TRA to
> peer on link 'l1'
> MS2*CLI>
> Disconnected from Asterisk server
> Executing last minute cleanups
> 
> 
> 
> 
> -Original Message-
> From: asterisk-ss7-boun...@lists.digium.com
> [mailto:asterisk-ss7-boun...@lists.digium.com] On Behalf Of
> asterisk-ss7-requ...@lists.digium.com
> Sent: Sunday, June 8, 2014 10:00 PM
> To: asterisk-ss7@lists.digium.com
> Subject: asterisk-ss7 Digest, Vol 111, Issue 2
> 
> Send asterisk-ss7 mailing list submissions to
>   asterisk-ss7@lists.digium.com
> 
> To subscribe or unsubscribe via the World Wide Web, visit
>   http://lists.digium.com/mailman/listinfo/asterisk-ss7
> or, via email, send a message with subject or body 'help' to
>   asterisk-ss7-requ...@lists.digium.com
> 
> You can reach the person managing the list at
>   asterisk-ss7-ow...@lists.digium.com
> 
> When replying, please edit your Subject line so it is more specific than
> "Re: Contents of asterisk-ss7 digest..."
> 
> 
> Today's Topics:
> 
>   1. Need to disabled SLTA responses on Chan_SS7 code level (shazi .)
>   2. Re: Need to disabled SLTA responses on Chan_SS7 code level
>  (Vallimamod Abdullah)
> 
> 
> --
> 
> Message: 1
> Date: Sun, 8 Jun 2014 20:56:42 +0500
> From: "shazi ." 
> To: 
> Subject: [asterisk-ss7] Need to disabled SLTA responses on Chan_SS7
>   codelevel
> Message-ID: 
> Content-Type: text/plain; charset="us-ascii"
> 
> Hi Experts,
> 
> Can anybody please help me out to disabled SLTA responses on Chan_SS7 code
> level.
> 
> I've come across below link but don't understand much.
> http://copilotco.com/mail-archives/asterisk-ss7.2006/msg00896.html
> 
> What I need is chan_ss7 to not to wait for SLTA after sending an SLTM
> because right now if SLTA not received it change the link-state to DOWN
> after two tries & the Q.707 timer T1 (to wait to receive an SLTA) gets
> expired.
> 
> I've set sltm => no but still SLTM is being sending from my side to telco.
> 
> I am using

Re: [asterisk-ss7] Asterisk SS7 + Audiocodes Mediant 2000

[bipin singh]

Its supported SIP protocol or else.


On Fri, Aug 16, 2013 at 9:07 PM, Nick Khamis  wrote:

> Pay 8,000 USD for the Audiocodes SS7 license and forget about
> asterisk-ss7.
>
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Re: [asterisk-ss7] Asterisk SS7 + Audiocodes Mediant 2000

[Nick Khamis]

Pay 8,000 USD for the Audiocodes SS7 license and forget about
asterisk-ss7.

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Re: [asterisk-ss7] Asterisk SS7 + Audiocodes Mediant 2000

[Marcelo - Goiás - M]

Dear Mitul Limbani,
Use the Mediant 2000 because the Brazilian Regulatory claiming to be required 
to use Media Gateway approved in Anatel (Brazilian Agency of 
Telecommunications). For this reason I am using the Audiocodes gateway because 
Asterisk is not regulated.
Therefore necessary to integrate Asterisk with SS7 Audiocodes Mediant 2000. You 
or another colleague from the list has documentation, scripts and other 
information on how to do this?
Att.: MarceloM, from Brazil.
Date: Fri, 16 Aug 2013 17:38:30 +0530
From: mi...@enterux.com
To: asterisk-ss7@lists.digium.com
Subject: Re: [asterisk-ss7] Asterisk SS7 + Audiocodes Mediant 2000

Why use it when it doesnt have the license?

Dont save cost for E1 Card. Go buy one instead.

Mitul

On Aug 16, 2013 5:32 PM, "Marcelo - Goiás - M"  wrote:




Dear Nick Khamis,
My Audiocodes Mediant 2000 is without licenses SS7. He has only CAS/SIP 
licenses, at least is what is showing on his screen. See below for licenses, 
but does not have SS7 license. Therefore accurate Asterisk with SS7.

;Board: Mediant 2000;Serial Number: ;Slot Number: 1;Software Version: 
6.00A.042.005;DSP Software Version: 620AE3 => 600.09;Board IP Address: 
xxx.168.222.180
;Board Subnet Mask: 255.255.255.0;Board Default Gateway: xxx.168.222.10;Ram 
size: 128M   Flash size: 8M ;Num of DSP Cores: 24  Num DSP Channels: 120
;Profile: NONE ;Key features:;Board Type: Mediant 
2000;E1Trunks=4;T1Trunks=4;PSTN Protocols: ISDN CAS ;DSP Voice features: 
EC128mSec IpmDetector ;Coders: G729 GSM-FR G727 ; Channel Type: RTP PCI 
DspCh=124 ;Security: IPSEC MediaEncryption StrongEncryption 
EncryptControlProtocol ;Control Protocols: SIP ;Default features:;Coders: G711 
G726;
;--
You'd think with the above licenses I can do work on SS7 Mediant 2000? I've 
tried, but does not rise in SS7 Audiocodes.

If you or someone else has some tips on how I ask kindly help me. Grateful.
Att: MarceloM
 


> Date: Fri, 16 Aug 2013 07:35:36 -0400
> From: sym...@gmail.com
> To: asterisk-ss7@lists.digium.com

> Subject: Re: [asterisk-ss7] Asterisk SS7 + Audiocodes Mediant 2000
> 
> if u have the ss7 mediant 2000, then the gateway would do the ss7-sip
> conversion. No need for asterisk ss7
> 
> On 8/15/13, Marcelo - Goiás - M

>  wrote:
> > Good morning everyone,
> > I need to connect a server with Asterisk SS7 gateway Audiocodes "Mediant

> > 2000" with 4 G703 E1 trunks. Does anyone have script for Asterisk and
> > information or configuration files for Audiocodes "Mediant 2000"?Any
> > tutorials that you can use?

> > Atencionsamente,
> > MarceloMD
> >
> 
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Re: [asterisk-ss7] Asterisk SS7 + Audiocodes Mediant 2000

[Mitul Limbani]

Why use it when it doesnt have the license?

Dont save cost for E1 Card. Go buy one instead.

Mitul
On Aug 16, 2013 5:32 PM, "Marcelo - Goiás - M" 
wrote:

> Dear Nick Khamis,
>
> My Audiocodes Mediant 2000 is without licenses SS7. He has only CAS/SIP
> licenses, at least is what is showing on his screen. See below for
> licenses, but does not have SS7 license. Therefore accurate Asterisk with
> SS7.
>
> ;Board: Mediant 2000
> ;Serial Number: 
> ;Slot Number: 1
> ;Software Version: 6.00A.042.005
> ;DSP Software Version: 620AE3 => 600.09
> ;Board IP Address: xxx.168.222.180
> ;Board Subnet Mask: 255.255.255.0
> ;Board Default Gateway: xxx.168.222.10
> ;Ram size: 128M   Flash size: 8M
> ;Num of DSP Cores: 24  Num DSP Channels: 120
> ;Profile: NONE
> ;Key features:;Board Type: Mediant 2000;E1Trunks=4;T1Trunks=4;PSTN
> Protocols: ISDN CAS ;DSP Voice features: EC128mSec IpmDetector ;Coders:
> G729 GSM-FR G727 ; Channel Type: RTP PCI DspCh=124 ;Security: IPSEC
> MediaEncryption StrongEncryption EncryptControlProtocol ;Control Protocols:
> SIP ;Default features:;Coders: G711 G726;
> ;--
>
> You'd think with the above licenses I can do work on SS7 Mediant 2000?
> I've tried, but does not rise in SS7 Audiocodes.
>
> If you or someone else has some tips on how I ask kindly help me. Grateful.
>
> Att: MarceloM
>
>
>  
> ....
> > Date: Fri, 16 Aug 2013 07:35:36 -0400
> > From: sym...@gmail.com
> > To: asterisk-ss7@lists.digium.com
> > Subject: Re: [asterisk-ss7] Asterisk SS7 + Audiocodes Mediant 2000
> >
> > if u have the ss7 mediant 2000, then the gateway would do the ss7-sip
> > conversion. No need for asterisk ss7
> >
> > On 8/15/13, Marcelo - Goiás - M
> >  wrote:
> > > Good morning everyone,
> > > I need to connect a server with Asterisk SS7 gateway Audiocodes
> "Mediant
> > > 2000" with 4 G703 E1 trunks. Does anyone have script for Asterisk and
> > > information or configuration files for Audiocodes "Mediant 2000"?Any
> > > tutorials that you can use?
> > > Atencionsamente,
> > > MarceloMD
> > >
> >
> > --
> > _
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> >
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> > To UNSUBSCRIBE or update options visit:
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Re: [asterisk-ss7] Asterisk SS7 + Audiocodes Mediant 2000

[Marcelo - Goiás - M]

Dear Nick Khamis,
My Audiocodes Mediant 2000 is without licenses SS7. He has only CAS/SIP 
licenses, at least is what is showing on his screen. See below for licenses, 
but does not have SS7 license. Therefore accurate Asterisk with SS7.
;Board: Mediant 2000;Serial Number: ;Slot Number: 1;Software Version: 
6.00A.042.005;DSP Software Version: 620AE3 => 600.09;Board IP Address: 
xxx.168.222.180;Board Subnet Mask: 255.255.255.0;Board Default Gateway: 
xxx.168.222.10;Ram size: 128M   Flash size: 8M ;Num of DSP Cores: 24  Num DSP 
Channels: 120;Profile: NONE ;Key features:;Board Type: Mediant 
2000;E1Trunks=4;T1Trunks=4;PSTN Protocols: ISDN CAS ;DSP Voice features: 
EC128mSec IpmDetector ;Coders: G729 GSM-FR G727 ; Channel Type: RTP PCI 
DspCh=124 ;Security: IPSEC MediaEncryption StrongEncryption 
EncryptControlProtocol ;Control Protocols: SIP ;Default features:;Coders: G711 
G726;;--
You'd think with the above licenses I can do work on SS7 Mediant 2000? I've 
tried, but does not rise in SS7 Audiocodes.
If you or someone else has some tips on how I ask kindly help me. Grateful.
Att: MarceloM
 

> Date: Fri, 16 Aug 2013 07:35:36 -0400
> From: sym...@gmail.com
> To: asterisk-ss7@lists.digium.com
> Subject: Re: [asterisk-ss7] Asterisk SS7 + Audiocodes Mediant 2000
> 
> if u have the ss7 mediant 2000, then the gateway would do the ss7-sip
> conversion. No need for asterisk ss7
> 
> On 8/15/13, Marcelo - Goiás - M
>  wrote:
> > Good morning everyone,
> > I need to connect a server with Asterisk SS7 gateway Audiocodes "Mediant
> > 2000" with 4 G703 E1 trunks. Does anyone have script for Asterisk and
> > information or configuration files for Audiocodes "Mediant 2000"?Any
> > tutorials that you can use?
> > Atencionsamente,
> > MarceloMD
> >
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-ss7
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Re: [asterisk-ss7] Asterisk SS7 + Audiocodes Mediant 2000

[Nick Khamis]

if u have the ss7 mediant 2000, then the gateway would do the ss7-sip
conversion. No need for asterisk ss7

On 8/15/13, Marcelo - Goiás - Moreira - Serviços de Engenharia
 wrote:
> Good morning everyone,
> I need to connect a server with Asterisk SS7 gateway Audiocodes "Mediant
> 2000" with 4 G703 E1 trunks. Does anyone have script for Asterisk and
> information or configuration files for Audiocodes "Mediant 2000"?Any
> tutorials that you can use?
> Atencionsamente,
> MarceloMD
>

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[asterisk-ss7] Asterisk SS7 + Audiocodes Mediant 2000

[Marcelo - Goiás - Moreira - Serviços de Engenharia]

Good morning everyone,
I need to connect a server with Asterisk SS7 gateway Audiocodes "Mediant 2000" 
with 4 G703 E1 trunks. Does anyone have script for Asterisk and information or 
configuration files for Audiocodes "Mediant 2000"?Any tutorials that you can 
use?
Atencionsamente,
MarceloMD
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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 99, Issue 9

[Muhammad Shahzad]

Dear,

Thanks for your reply,

I'm not using zombie channels, it's simple SS7 interconnectivity or you can
say dahdi channels,

I had already tried so many versions time by time,

With asterisk 1.8.21.0, dahdi 2.6.2, chan-ss7 2.1 and wanpipe 7 (I got MTP
receivebuf issue)
With asterisk 1.6.1.6, dahdi 2.5, chan-ss7 1.3 and wanpipe 3.5 (I got
version incompatibility issue b/w wanpipe and dahdi)
With asterisk 1.6.1.6, dahdi 2.5, chan-ss7 1.3 and wanpipe 7 (code
incompatibility issue)
Now currently installed asterisk 1.6.1.6, dahdi 2.5, chan-ss7 2.3 and
wanpipe 7 (getting MTP receivebuf issue)

The node is live now and can't install any other version just for testing,

I would really appreciate if you help me what exact versions of asterisk,
dahdi, chan_ss7 and wanpipe I can use for production.

It's very urgent,

BR,
Muhammad Shahzad
 

-Original Message-
From: asterisk-ss7-boun...@lists.digium.com
[mailto:asterisk-ss7-boun...@lists.digium.com] On Behalf Of
asterisk-ss7-requ...@lists.digium.com
Sent: Tuesday, May 28, 2013 10:00 PM
To: asterisk-ss7@lists.digium.com
Subject: asterisk-ss7 Digest, Vol 99, Issue 9

Send asterisk-ss7 mailing list submissions to
asterisk-ss7@lists.digium.com

To subscribe or unsubscribe via the World Wide Web, visit
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Today's Topics:

   1. Re: getting MTP receivebuf issue (Urgent Help)


--

Message: 1
Date: Tue, 28 May 2013 00:01:50 +0500
From: [Digital^Dude] ? 
Subject: Re: [asterisk-ss7] getting MTP receivebuf issue (Urgent Help)
To: asterisk-ss7@lists.digium.com
Message-ID:

Content-Type: text/plain; charset="iso-8859-1"

Hello,

Your ss7 config looks fine to me. Do you have zombie channels on your
asterisk?

On Sun, May 26, 2013 at 3:41 AM, Muhammad Shahzad <
engrmuhammadshah...@hotmail.com> wrote:

> Dear All,
>
> ** **
>
> This is my first post on this mailing list, So my hopes are so 
> high.
>
> ** **
>
> I am getting MTP receivebuf issue, The problem arises when there are 
> above
> 25 active channels. I am using chan_ss7 version 2.1, 
> asterisk-1.8.21.0, and dahdi-linux-complete-2.6.2. 
>
> ** **
>
> Now, at first things work well:
>
> "ss7 link status" shows that both link are INSERVICE.
>
> "ss7 linestat" shows that all CICs go into Idle status.
>
> Calls flow in both directions ok, but after 10 to 15mins. Eventually, 
> something exhausts itself and everything falls apart with the 
> following message flooding repeatedly,
>
> *mtp.c:413 mtp_put: Full MTP receivebuf, event lost, type=15.*
>
> * *
>
> My configs are attached above,
>
> ** **
>
> Please help me out its urgent..
>
> ** **
>
> Thanks,
>
> ** **
>
> Muhammad Shahzad
>
> ** **
>
> ** **
>
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>
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>
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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 81, Issue 24

[Cees Harteveld]

For those how are interested, it was indeed a configuration error.

I addedSLC=0 or 1 and set for both the linkset=1.

Now I see that both links come up with the right SLC

MTP2 link up (SLC 1)

--- SS7 Up ---

Resetting CICs 1 to 15

Resetting CICs 17 to 31

Resetting CICs 97 to 111

Resetting CICs 113 to 127

Resetting CICs 129 to 143

Resetting CICs 145 to 159

Resetting CICs 1 to 15

Resetting CICs 17 to 31

Resetting CICs 97 to 111

Resetting CICs 113 to 127

Resetting CICs 129 to 143

Resetting CICs 145 to 159

MTP2 link up (SLC 0)

(chan_dahdi.conf)

[channels]

context=from-pstn

group=0,11

echocancel=no

signaling=ss7

ss7type=itu

ss7_called_nai=dynamic

ss7_calling_nai=dynamic

ss7_internationalprefix=00

ss7_nationalprefix=0

ss7_subscriberprefix=

ss7_unknownprefix=

networkindicator=national_spare

linkset=1

SLC=0

pointcode=123

adjpointcode=321

defaultdpc=231

sigchan=16

** **

 


cicbeginswith=1

channel=1-15

** **

 


cicbeginswith=17

channel=17-31

** **

 


cicbeginswith=97

channel=63-77

** **

cicbeginswith=113

channel=79-93

** **

 


group=3

cicbeginswith=129

channel=94-108

** **

 


cicbeginswith=145

channel=110-119

** **

 


group=4

cicbeginswith=155

channel=120-124

** **

 


context=from-pstn

group=1

echochannel=no

signaling=ss7

ss7type=itu

ss7_called_nai=dynamic

ss7_calling_nai=dynamic

ss7_internationalprefix=00

ss7_nationalprefix=0

ss7_subscriberprefix=

ss7_unknownprefix=

networkindicator=national_spare

linkset=1

SLC=1

pointcode=123

adjpointcode=456

defaultdpc=564

sigchan=140

cicbeginswith=1

channel=125-139

** **

 


cicbeginswith=17

channel=141-155

** **

cicbeginswith=97

channel=187-201

** **

cicbeginswith=113

channel=203-217

** **

 


group=3

cicbeginswith=129

channel=218-232

 

 


cicbeginswith=145

channel=234-243

** **

 


group=4

cicbeginswith=155

channel=244-248




Met vriendelijke groet,

kind regards

Cees Harteveld,



Op 28-11-2011 10:08, asterisk-ss7-requ...@lists.digium.com schreef:

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Today's Topics:

1. Re: two DPC with same cic numbering (bipin singh)
2. Fwd: Re:  two DPC with same cic numbering (Cees Harteveld)


--

Message: 1
Date: Mon, 28 Nov 2011 10:09:06 +0530
From: bipin singh
Subject: Re: [asterisk-ss7] two DPC with same cic numbering
To: cees.hartev...@budgetphone.nl, asterisk-ss7@lists.digium.com
Message-ID:

Content-Type: text/plain; charset="windows-1252"

Hi,
check your conf how many time configured linkset=2 . Currect it
first then check output .

On Fri, Nov 25, 2011 at 8:43 PM, Cees Harteveld<
cees.hartev...@budgetphone.nl>  wrote:


  Hello,


Maybe some one knows the answer, 

** **

** **

I got two linkset set up with 2 different DPC?s, but both usages the same
CIC?s numbering

When I  uncomment 1 off the linksets in chan_dahdi.conf  it works fine(
it does?nt metter wich one I uncomment), am I using them both I get error?s
.

(error)

unconfigured CIC 153

** **

(chan_dahdi.conf)

[channels]

context=from-pstn

group=0,11

echocancel=no

signaling=ss7

ss7type=itu

ss7_called_nai=dynamic

ss7_calling_nai=dynamic

ss7_internationalprefix=00

ss7_nationalprefix=0

ss7_subscriberprefix=

ss7_unknownprefix=

networkindicator=national_spare

linkset=1

pointcode=123

adjpointcode=321

defaultdpc=231

sigchan=16

** **

cicbeginswith=1

channel=1-15

** **

cicbeginswith=17

channel=17-31

** **

cicbeginswith=97

channel=63-77

** **

cicbeginswith=113

channel=79-93

** **

group=3,12

linkset=1

cicbeginswith=129

channel=94-108

** **

cicbeginswith=145

channel=110-119

** **

group=4,13

linkset=1

cicbeginswith=155

channel=120-124

** **

context=from-pstn

group=1

echochannel=no

signaling=ss7

ss7type=itu

ss7_called_nai=dynamic

ss7_calling_nai=dynamic

ss7_internationalprefix=00

ss7_nationalprefix=0

ss7_subscriberprefix=

ss7_unknownprefix=

networkindicator=national_spare

linkset=2

p

Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 81, Issue 13

[bipin singh]

Hi,
First check you maximum concurrent calls happen on you server .
CRBT is maximum 30 to 45 sec process so approx you process two calls per
mint per channels .

On Fri, Nov 11, 2011 at 12:52 PM, Peter Hyeroba  wrote:

> The flow is a call comes in from the pstn and  the switch forwards it to
> my asterisk box which checks if that person is registered for crbt if yes
> then we play the tune and as soon as the call is answered, control is
> passed back to the switch or IN.
>
> on my clients gsm network there are at least 50,000 registered users for
> crbt and these calls are also happening at any one time.
>
> I need a way for his network to pass the call to my asterisk box and me to
> pass it back when its answered.
>
> Kind Regards,
>
> HP.
>
>
> On 10 November 2011 21:00,  wrote:
>
>> Send asterisk-ss7 mailing list submissions to
>>asterisk-ss7@lists.digium.com
>>
>> To subscribe or unsubscribe via the World Wide Web, visit
>>http://lists.digium.com/mailman/listinfo/asterisk-ss7
>> or, via email, send a message with subject or body 'help' to
>>asterisk-ss7-requ...@lists.digium.com
>>
>> You can reach the person managing the list at
>>asterisk-ss7-ow...@lists.digium.com
>>
>> When replying, please edit your Subject line so it is more specific
>> than "Re: Contents of asterisk-ss7 digest..."
>>
>>
>> Today's Topics:
>>
>>   1. Re: CRBT with ss7 and asterisk (flor...@gruendler.net)
>>
>>
>> --
>>
>> Message: 1
>> Date: Thu, 10 Nov 2011 18:03:58 +0100
>> From: 
>> Subject: Re: [asterisk-ss7] CRBT with ss7 and asterisk
>> To: 
>> Message-ID: <042c01cc9fca$bd1bfd00$3753f700$@net>
>> Content-Type: text/plain;   charset="iso-8859-1"
>>
>> Our friend must indeed have made a mistake at the decimal level. 100k
>> concurrent calls is a lot. On the other hand it isn't if you were China
>> *com
>> or so but then you'd get a special discount on Huawei I guess ;-) Now,
>> before we get into hardware and brand questions, I would also like to hear
>> more about the intended call flow.
>>
>> Is this a service innovation you consider for your subscribers to give a
>> personalized music instead of a ring back tone to inbound callers or is it
>> intended to playback treatment loops on certain releases on outbound calls
>> to make that more fancy than just a reorder tone on all remote clearings?
>>
>> Generally speaking, it should be feasible to script your INAP GSM/PSTN
>> switch to forge the call to the (Asterisk-) HMP playback box when the
>> PBX/LD-switch signals progress, similar as you would to a voicemail system
>> except that the channel won't answer and just plays the tune as early
>> media.
>> And on event connect on the bearer channel to the cell tower/PBX, you'd
>> redirect the subscriber media channel. But hey, why not just playing the
>> tune directly on the big iron itself? I know an excellent Dialogic based
>> supplier for you if your switch can't do it to extend its lifecycle or
>> licensing frightens you off. But vendors tend to be flexible in pricing
>> if a
>> generation change is due anyway.
>>
>> Let us know,
>>
>> Florian
>>
>>
>>
>>
>> > -Urspr?ngliche Nachricht-
>> > Von: asterisk-ss7-boun...@lists.digium.com [mailto:asterisk-ss7-
>> > boun...@lists.digium.com] Im Auftrag von Michael Mueller
>> > Gesendet: Donnerstag, 10. November 2011 16:17
>> > An: asterisk-ss7@lists.digium.com
>> > Betreff: Re: [asterisk-ss7] CRBT with ss7 and asterisk
>> >
>> > what's CRBT?
>> >
>> > 100K calls seems like lot of connections to the PSTN - especially for
>> > Asterisk
>> >
>> > more detail please
>> >
>> > On Thu, Nov 10, 2011 at 5:59 AM, Peter Hyeroba 
>> > wrote:
>> > > Hello All,
>> > > Has anyone implemented CRBT with ss7 on asterisk, what hardware would
>> > > one recommend for over 100,000 concurrent calls.
>> > > If anyone has any ideas or pointers please let me know.
>> > >
>> > > Kind Regards,
>> > >
>> > > --
>> > > Hyeroba Wegulo Peter
>> > >
>> > > website: www.jezitech.com
>> > > Phones : +256-414-533238, +256-392-897155
>> > > Mobiles: +256-782-479192, +256-718-206135 other emails:
>> > > jezit...@yahoo.com, phyer...@jezitech.com
>> > >
>> > >
>> > > --
>> > > _
>> > > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> > >
>> > > asterisk-ss7 mailing list
>> > > To UNSUBSCRIBE or update options visit:
>> > > ? http://lists.digium.com/mailman/listinfo/asterisk-ss7
>> > >
>> >
>> > --
>> > _
>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> >
>> > asterisk-ss7 mailing list
>> > To UNSUBSCRIBE or update options visit:
>> >http://lists.digium.com/mailman/listinfo/asterisk-ss7
>>
>>
>>
>>
>> --
>>
>> ___
>> --Bandwidth and Coloca

Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 81, Issue 13

[Peter Hyeroba]

The flow is a call comes in from the pstn and  the switch forwards it to my
asterisk box which checks if that person is registered for crbt if yes then
we play the tune and as soon as the call is answered, control is passed
back to the switch or IN.

on my clients gsm network there are at least 50,000 registered users for
crbt and these calls are also happening at any one time.

I need a way for his network to pass the call to my asterisk box and me to
pass it back when its answered.

Kind Regards,

HP.


On 10 November 2011 21:00,  wrote:

> Send asterisk-ss7 mailing list submissions to
>asterisk-ss7@lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
>http://lists.digium.com/mailman/listinfo/asterisk-ss7
> or, via email, send a message with subject or body 'help' to
>asterisk-ss7-requ...@lists.digium.com
>
> You can reach the person managing the list at
>asterisk-ss7-ow...@lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-ss7 digest..."
>
>
> Today's Topics:
>
>   1. Re: CRBT with ss7 and asterisk (flor...@gruendler.net)
>
>
> --
>
> Message: 1
> Date: Thu, 10 Nov 2011 18:03:58 +0100
> From: 
> Subject: Re: [asterisk-ss7] CRBT with ss7 and asterisk
> To: 
> Message-ID: <042c01cc9fca$bd1bfd00$3753f700$@net>
> Content-Type: text/plain;   charset="iso-8859-1"
>
> Our friend must indeed have made a mistake at the decimal level. 100k
> concurrent calls is a lot. On the other hand it isn't if you were China
> *com
> or so but then you'd get a special discount on Huawei I guess ;-) Now,
> before we get into hardware and brand questions, I would also like to hear
> more about the intended call flow.
>
> Is this a service innovation you consider for your subscribers to give a
> personalized music instead of a ring back tone to inbound callers or is it
> intended to playback treatment loops on certain releases on outbound calls
> to make that more fancy than just a reorder tone on all remote clearings?
>
> Generally speaking, it should be feasible to script your INAP GSM/PSTN
> switch to forge the call to the (Asterisk-) HMP playback box when the
> PBX/LD-switch signals progress, similar as you would to a voicemail system
> except that the channel won't answer and just plays the tune as early
> media.
> And on event connect on the bearer channel to the cell tower/PBX, you'd
> redirect the subscriber media channel. But hey, why not just playing the
> tune directly on the big iron itself? I know an excellent Dialogic based
> supplier for you if your switch can't do it to extend its lifecycle or
> licensing frightens you off. But vendors tend to be flexible in pricing if
> a
> generation change is due anyway.
>
> Let us know,
>
> Florian
>
>
>
>
> > -Urspr?ngliche Nachricht-
> > Von: asterisk-ss7-boun...@lists.digium.com [mailto:asterisk-ss7-
> > boun...@lists.digium.com] Im Auftrag von Michael Mueller
> > Gesendet: Donnerstag, 10. November 2011 16:17
> > An: asterisk-ss7@lists.digium.com
> > Betreff: Re: [asterisk-ss7] CRBT with ss7 and asterisk
> >
> > what's CRBT?
> >
> > 100K calls seems like lot of connections to the PSTN - especially for
> > Asterisk
> >
> > more detail please
> >
> > On Thu, Nov 10, 2011 at 5:59 AM, Peter Hyeroba 
> > wrote:
> > > Hello All,
> > > Has anyone implemented CRBT with ss7 on asterisk, what hardware would
> > > one recommend for over 100,000 concurrent calls.
> > > If anyone has any ideas or pointers please let me know.
> > >
> > > Kind Regards,
> > >
> > > --
> > > Hyeroba Wegulo Peter
> > >
> > > website: www.jezitech.com
> > > Phones : +256-414-533238, +256-392-897155
> > > Mobiles: +256-782-479192, +256-718-206135 other emails:
> > > jezit...@yahoo.com, phyer...@jezitech.com
> > >
> > >
> > > --
> > > _
> > > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > >
> > > asterisk-ss7 mailing list
> > > To UNSUBSCRIBE or update options visit:
> > > ? http://lists.digium.com/mailman/listinfo/asterisk-ss7
> > >
> >
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-ss7 mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-ss7
>
>
>
>
> --
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-ss7
>
> End of asterisk-ss7 Digest, Vol 81, Issue 13
> 
>



-- 
Hyeroba Wegulo Peter

Managing Director
Jezitech Solutions Limited
P.O.Box 36285,
Plot 27,
Sir. A

Re: [asterisk-ss7] asterisk-ss7@lists.digium.com

[peterpet]

Hi,
try this :

Alternate release of libss7 with additional features

svn co https://observer.router.hu/repos_pub/chan_dahdi/trunk chan_dahdi
svn co https://observer.router.hu/repos_pub/libss7/trunk libss7

Peter,

On 11/09/2011 10:47 PM, Sal Aguilar wrote:


Thanks for the feedback Joseph.

But do anyone have an idea how to debug this, libss7 ignoring the 
redirection that is located on the ACM part of the ISUP dialog?


Does any other SS7 stack supports this?

Sal Aguilar

**

*Email: s...@kom-1.com*

*From:*asterisk-ss7-boun...@lists.digium.com 
[mailto:asterisk-ss7-boun...@lists.digium.com] *On Behalf Of *Joseph

*Sent:* Wednesday, November 09, 2011 2:08 PM
*To:* asterisk-ss7@lists.digium.com
*Subject:* Re: [asterisk-ss7] asterisk-ss7@lists.digium.com

Does this help or work?

http://www.voip-info.org/wiki/index.php?page=Asterisk+libss7#CallRedirection

--

regards, Joseph

On Nov 9, 2011, at 2:54 PM, Sal Aguilar wrote:



Hi,

Anyone know if there is a fork of libss7 that supports Redirecting 
according to ISUP'92? (Redirecting is in optional parameters in the 
ACM message in this ISUP flavor).


If there is a version of libss7 that supports this, then how is it 
handled?


Thanks,

Sal Aguilar

**

*Email:s...@kom-1.com <mailto:s...@kom-1.com>*

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Re: [asterisk-ss7] asterisk-ss7@lists.digium.com

[Sal Aguilar]

Thanks for the feedback Joseph.

 

But do anyone have an idea how to debug this, libss7 ignoring the
redirection that is located on the ACM part of the ISUP dialog?

 

Does any other SS7 stack supports this?

 

Sal Aguilar

 

Email: s...@kom-1.com

 

From: asterisk-ss7-boun...@lists.digium.com
[mailto:asterisk-ss7-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Wednesday, November 09, 2011 2:08 PM
To: asterisk-ss7@lists.digium.com
Subject: Re: [asterisk-ss7] asterisk-ss7@lists.digium.com

 

Does this help or work?

 

http://www.voip-info.org/wiki/index.php?page=Asterisk+libss7#CallRedirection

 

--

regards, Joseph

 

On Nov 9, 2011, at 2:54 PM, Sal Aguilar wrote:





Hi,

 

Anyone know if there is a fork of libss7 that supports Redirecting according
to ISUP'92? (Redirecting is in optional parameters in the ACM message in
this ISUP flavor).

 

If there is a version of libss7 that supports this, then how is it handled?

 

Thanks,

 

Sal Aguilar

 

Email: s...@kom-1.com

 

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  http://lists.digium.com/mailman/listinfo/asterisk-ss7

 

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Re: [asterisk-ss7] asterisk-ss7@lists.digium.com

[Joseph]

Does this help or work?

http://www.voip-info.org/wiki/index.php?page=Asterisk+libss7#CallRedirection

--

regards, Joseph

On Nov 9, 2011, at 2:54 PM, Sal Aguilar wrote:

> Hi,
>  
> Anyone know if there is a fork of libss7 that supports Redirecting according 
> to ISUP'92? (Redirecting is in optional parameters in the ACM message in this 
> ISUP flavor).
>  
> If there is a version of libss7 that supports this, then how is it handled?
>  
> Thanks,
>  
> Sal Aguilar
>  
> Email: s...@kom-1.com
>  
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-ss7

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[asterisk-ss7] asterisk-ss7@lists.digium.com

[Sal Aguilar]

Hi,

 

Anyone know if there is a fork of libss7 that supports Redirecting according
to ISUP'92? (Redirecting is in optional parameters in the ACM message in
this ISUP flavor). 

 

If there is a version of libss7 that supports this, then how is it handled?

 

Thanks,

 

Sal Aguilar

 

Email: s...@kom-1.com

 

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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 81, Issue 8

[Peter Hyeroba]

Hello Would anyone recommend good hardware for CRBT, that would work with
asterisk?

HP.

On 9 November 2011 09:25, Peter Hyeroba  wrote:

> Hello,
>
> Has anyone used digium cards for setting up a CRBT platform on asterisk?
>
> Do u guys think its a viable solution?
>
> Kind Regards,
>
> HP.
>
>
> On 9 November 2011 06:26,  wrote:
>
>> Send asterisk-ss7 mailing list submissions to
>>asterisk-ss7@lists.digium.com
>>
>> To subscribe or unsubscribe via the World Wide Web, visit
>>http://lists.digium.com/mailman/listinfo/asterisk-ss7
>> or, via email, send a message with subject or body 'help' to
>>asterisk-ss7-requ...@lists.digium.com
>>
>> You can reach the person managing the list at
>>asterisk-ss7-ow...@lists.digium.com
>>
>> When replying, please edit your Subject line so it is more specific
>> than "Re: Contents of asterisk-ss7 digest..."
>>
>>
>> Today's Topics:
>>
>>   1. Re: Help (bipin singh)
>>
>>
>> --
>>
>> Message: 1
>> Date: Wed, 9 Nov 2011 08:53:41 +0530
>> From: bipin singh 
>> Subject: Re: [asterisk-ss7] Help
>> To: asterisk-ss7@lists.digium.com
>> Message-ID:
>>> 3933pa8mlknr09j2olx85p8rsnywk2xvz_8zhkb0n2...@mail.gmail.com>
>> Content-Type: text/plain; charset="iso-8859-1"
>>
>> Hi,
>>If you are new for asterisk then use only digium card . All
>> digium pri (1 port , 2 port or 4 port ) card work with ss7 link . If you
>> required good voice quality
>> then use only echo cancellation digium card . We are using all card and
>> work fine with ss7 link .
>>
>>
>> On Tue, Nov 8, 2011 at 7:57 PM, Rafael Machado > >wrote:
>>
>> > Greetings to all the list.
>> > I make a purchase of a Digium card and the recommendation of friends
>> would
>> > like to indicate what the best card to work with SS7 signaling.
>> >
>> > Can anyone direct me from experience?
>> >
>> > ** **
>> >
>> > ** **
>> >
>> > Rafael Machado
>> >
>> > Tel.: +55 (14) 3402 9702
>> >
>> > Visite.: www.life.com.br
>> >
>> > Email.: rafaelmach...@life.com.br
>> >
>> > [image: Descri??o: cid:F315BA5D-EAA8-4573-B25B-3E3923679507]
>> >
>> > ** **
>> >
>> > --
>> > _
>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> >
>> > asterisk-ss7 mailing list
>> > To UNSUBSCRIBE or update options visit:
>> >   http://lists.digium.com/mailman/listinfo/asterisk-ss7
>> >
>>
>>
>>
>> --
>> BIPIN RAGHUVANSHI
>> OPERATION HEAD
>> ASTERISK (DEVELOPMENT AND RESEARCH)
>> WWW.EHORIZONS.IN
>> 011-32323262
>> 011-46334633
>> -- next part --
>> An HTML attachment was scrubbed...
>> URL: <
>> http://lists.digium.com/pipermail/asterisk-ss7/attachments/2009/dc56f50d/attachment.htm
>> >
>> -- next part --
>> A non-text attachment was scrubbed...
>> Name: not available
>> Type: image/png
>> Size: 20128 bytes
>> Desc: not available
>> URL: <
>> http://lists.digium.com/pipermail/asterisk-ss7/attachments/2009/dc56f50d/attachment.png
>> >
>>
>> --
>>
>> ___
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-ss7 mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-ss7
>>
>> End of asterisk-ss7 Digest, Vol 81, Issue 8
>> ***
>>
>
>
>
> --
> Hyeroba Wegulo Peter
>
> Managing Director
> Jezitech Solutions Limited
> P.O.Box 36285,
> Plot 27,
> Sir. Apollo Kagwa Road
> Kampala Uganda.
>
> website: www.jezitech.com 
> Phones : +256-414-533238, +256-392-897155
> Mobiles: +256-782-479192, +256-718-206135
> other emails: jezit...@yahoo.com, phyer...@jezitech.com
>
> This e-mail and any attachments are confidential and intended solely for
> the addressee and may also be privileged or exempt from disclosure under
> applicable law. If you are not the addressee, or have received this e-mail
> in error, please notify the sender immediately, delete it from your system
> and do not copy, disclose or otherwise act upon any part of this e-mail or
> its attachments.
>
>


-- 
Hyeroba Wegulo Peter

Managing Director
Jezitech Solutions Limited
P.O.Box 36285,
Plot 27,
Sir. Apollo Kagwa Road
Kampala Uganda.

website: www.jezitech.com 
Phones : +256-414-533238, +256-392-897155
Mobiles: +256-782-479192, +256-718-206135
other emails: jezit...@yahoo.com, phyer...@jezitech.com 

This e-mail and any attachments are confidential and intended solely for
the addressee and may also be privileged or exempt from disclosure under
applicable law. If you are not the addressee, or have received this e-mail
in error, please notify the sender immediately, delete it from your system
and do not copy, disclose or otherwise act upon any part of this e-mail or
its attachments.
--

Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 81, Issue 8

[Peter Hyeroba]

Hello,

Has anyone used digium cards for setting up a CRBT platform on asterisk?

Do u guys think its a viable solution?

Kind Regards,

HP.

On 9 November 2011 06:26,  wrote:

> Send asterisk-ss7 mailing list submissions to
>asterisk-ss7@lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
>http://lists.digium.com/mailman/listinfo/asterisk-ss7
> or, via email, send a message with subject or body 'help' to
>asterisk-ss7-requ...@lists.digium.com
>
> You can reach the person managing the list at
>asterisk-ss7-ow...@lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-ss7 digest..."
>
>
> Today's Topics:
>
>   1. Re: Help (bipin singh)
>
>
> --
>
> Message: 1
> Date: Wed, 9 Nov 2011 08:53:41 +0530
> From: bipin singh 
> Subject: Re: [asterisk-ss7] Help
> To: asterisk-ss7@lists.digium.com
> Message-ID:
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi,
>If you are new for asterisk then use only digium card . All
> digium pri (1 port , 2 port or 4 port ) card work with ss7 link . If you
> required good voice quality
> then use only echo cancellation digium card . We are using all card and
> work fine with ss7 link .
>
>
> On Tue, Nov 8, 2011 at 7:57 PM, Rafael Machado  >wrote:
>
> > Greetings to all the list.
> > I make a purchase of a Digium card and the recommendation of friends
> would
> > like to indicate what the best card to work with SS7 signaling.
> >
> > Can anyone direct me from experience?
> >
> > ** **
> >
> > ** **
> >
> > Rafael Machado
> >
> > Tel.: +55 (14) 3402 9702
> >
> > Visite.: www.life.com.br
> >
> > Email.: rafaelmach...@life.com.br
> >
> > [image: Descri??o: cid:F315BA5D-EAA8-4573-B25B-3E3923679507]
> >
> > ** **
> >
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-ss7 mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-ss7
> >
>
>
>
> --
> BIPIN RAGHUVANSHI
> OPERATION HEAD
> ASTERISK (DEVELOPMENT AND RESEARCH)
> WWW.EHORIZONS.IN
> 011-32323262
> 011-46334633
> -- next part --
> An HTML attachment was scrubbed...
> URL: <
> http://lists.digium.com/pipermail/asterisk-ss7/attachments/2009/dc56f50d/attachment.htm
> >
> -- next part --
> A non-text attachment was scrubbed...
> Name: not available
> Type: image/png
> Size: 20128 bytes
> Desc: not available
> URL: <
> http://lists.digium.com/pipermail/asterisk-ss7/attachments/2009/dc56f50d/attachment.png
> >
>
> --
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-ss7
>
> End of asterisk-ss7 Digest, Vol 81, Issue 8
> ***
>



-- 
Hyeroba Wegulo Peter

Managing Director
Jezitech Solutions Limited
P.O.Box 36285,
Plot 27,
Sir. Apollo Kagwa Road
Kampala Uganda.

website: www.jezitech.com 
Phones : +256-414-533238, +256-392-897155
Mobiles: +256-782-479192, +256-718-206135
other emails: jezit...@yahoo.com, phyer...@jezitech.com 

This e-mail and any attachments are confidential and intended solely for
the addressee and may also be privileged or exempt from disclosure under
applicable law. If you are not the addressee, or have received this e-mail
in error, please notify the sender immediately, delete it from your system
and do not copy, disclose or otherwise act upon any part of this e-mail or
its attachments.
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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 78, Issue 15

[bipin singh]

Hi,
 Sangoma A200 FXS/FXO Analog this card not work with libss7 .

On Fri, Aug 19, 2011 at 8:00 PM,  wrote:

> Emanuel,
>
> ** **
>
> I am missing your motivation? Is this just a question out of academic
> interest or what’s the point? Could you kill a mouse with a stinger
>  missile? The answer is, you probably could with some major software tweaks
> to that 38’000$ baby . But what problem would it solve and what added value
> would it provide over throwing a stone at the mouse which I don’t see at the
> moment?
>
> ** **
>
> Don’t lose the focus what an analogue line is: It’s a customer facing
> electric interface to provide a current for a phone, a dial tone, receive
> tone signals (=inband signals) and let you talk. It was never meant to do
> much more.
>
> ** **
>
> Just as BGP is not suitable for a home internet router, SS7 is not suitable
> for residential requirements. SS7 is a reliable out of band message transfer
> protocol to request, mediate features, connect and tear down circuits. One
> link is capable to handle thousands of circuits. In order to join the SS7
> network (I mean a real world one, not lab) you need an assignment of a
> globally unique point-code addressing element for your network. The mid
> range PRI ISDN signaling protocol is Q.931/DSS1, which is still a CPE
> signaling protocol but requires much less “official auditing” while still
> providing most ISUP (POTS related protocol subset) features. 
>
> ** **
>
> Regards, Florian
>
> ** **
>
> ** **
>
> *Von:* asterisk-ss7-boun...@lists.digium.com [mailto:
> asterisk-ss7-boun...@lists.digium.com] *Im Auftrag von *Emmanuel Buamah
> *Gesendet:* Freitag, 19. August 2011 12:43
> *An:* asterisk-ss7@lists.digium.com
> *Betreff:* Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 78, Issue 15
>
> ** **
>
>
> Thank you all for your replies. My next question is, I have "Sangoma A200
> FXS/FXO Analog card", will libss7 or chan_ss7 work with that? I asked
> because, I know the ss7 works with only digital cards.
>
> Thanks!
>
> --- On *Thu, 8/18/11, asterisk-ss7-requ...@lists.digium.com <
> asterisk-ss7-requ...@lists.digium.com>* wrote:
>
>
> From: asterisk-ss7-requ...@lists.digium.com <
> asterisk-ss7-requ...@lists.digium.com>
> Subject: asterisk-ss7 Digest, Vol 78, Issue 15
> To: asterisk-ss7@lists.digium.com
> Date: Thursday, August 18, 2011, 8:08 PM
>
> Send asterisk-ss7 mailing list submissions to
> 
> asterisk-ss7@lists.digium.com<http://mc/compose?to=asterisk-ss7@lists.digium.com>
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-ss7
> or, via email, send a message with subject or body 'help' to
> 
> asterisk-ss7-requ...@lists.digium.com<http://mc/compose?to=asterisk-ss7-requ...@lists.digium.com>
>
> You can reach the person managing the list at
> 
> asterisk-ss7-ow...@lists.digium.com<http://mc/compose?to=asterisk-ss7-ow...@lists.digium.com>
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-ss7 digest..."
>
>
> Today's Topics:
>
>1. Re: Hardware for libss7 (bipin singh)
>2. Re: Hardware for libss7 (Abdul Basit)
>3. ??:  Hardware for libss7 (Wayne)
>4. Re: ??:  Hardware for libss7 (James zhu)
>
>
> --
>
> Message: 1
> Date: Fri, 19 Aug 2011 10:33:30 +0530
> From: bipin singh 
> http://mc/compose?to=bipinraghuvan...@gmail.com>
> >
> Subject: Re: [asterisk-ss7] Hardware for libss7
> To: 
> asterisk-ss7@lists.digium.com<http://mc/compose?to=asterisk-ss7@lists.digium.com>
> Message-ID:
> 
> http://mc/compose?to=3_xzakfzqp68ksmwmkugvs3rn3jfyhzra8awh2yad-...@mail.gmail.com>
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi ,
> Goto digium or sangoma sites and check what type hardware you
> want to need . If you need to install libss7 then use only digium card .
>
> On Thu, Aug 18, 2011 at 5:44 PM, Emmanuel Buamah 
> http://mc/compose?to=wasa...@yahoo.com>>
> wrote:
>
> > Hi All,
> >
> > I want to setup libss7 on ubuntu. But for now, I don't know the hardware
> I
> > need to accomplish this in the absent of computer. And also, who can link
> me
> > to a very good tutorial on how to setup a very good working libss7?
> >
> > Cheers!
> >
> > --
> > _
> > -- Bandwidth and C

Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 78, Issue 15

[florian]

Emanuel,

 

I am missing your motivation? Is this just a question out of academic
interest or what's the point? Could you kill a mouse with a stinger
missile? The answer is, you probably could with some major software tweaks
to that 38'000$ baby . But what problem would it solve and what added value
would it provide over throwing a stone at the mouse which I don't see at the
moment?

 

Don't lose the focus what an analogue line is: It's a customer facing
electric interface to provide a current for a phone, a dial tone, receive
tone signals (=inband signals) and let you talk. It was never meant to do
much more.

 

Just as BGP is not suitable for a home internet router, SS7 is not suitable
for residential requirements. SS7 is a reliable out of band message transfer
protocol to request, mediate features, connect and tear down circuits. One
link is capable to handle thousands of circuits. In order to join the SS7
network (I mean a real world one, not lab) you need an assignment of a
globally unique point-code addressing element for your network. The mid
range PRI ISDN signaling protocol is Q.931/DSS1, which is still a CPE
signaling protocol but requires much less "official auditing" while still
providing most ISUP (POTS related protocol subset) features. 

 

Regards, Florian

 

 

Von: asterisk-ss7-boun...@lists.digium.com
[mailto:asterisk-ss7-boun...@lists.digium.com] Im Auftrag von Emmanuel
Buamah
Gesendet: Freitag, 19. August 2011 12:43
An: asterisk-ss7@lists.digium.com
Betreff: Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 78, Issue 15

 



Thank you all for your replies. My next question is, I have "Sangoma A200
FXS/FXO Analog card", will libss7 or chan_ss7 work with that? I asked
because, I know the ss7 works with only digital cards.

Thanks!

--- On Thu, 8/18/11, asterisk-ss7-requ...@lists.digium.com
 wrote:


From: asterisk-ss7-requ...@lists.digium.com

Subject: asterisk-ss7 Digest, Vol 78, Issue 15
To: asterisk-ss7@lists.digium.com
Date: Thursday, August 18, 2011, 8:08 PM

Send asterisk-ss7 mailing list submissions to
asterisk-ss7@lists.digium.com

To subscribe or unsubscribe via the World Wide Web, visit
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asterisk-ss7-ow...@lists.digium.com

When replying, please edit your Subject line so it is more specific
than "Re: Contents of asterisk-ss7 digest..."


Today's Topics:

   1. Re: Hardware for libss7 (bipin singh)
   2. Re: Hardware for libss7 (Abdul Basit)
   3. ??:  Hardware for libss7 (Wayne)
   4. Re: ??:  Hardware for libss7 (James zhu)


--

Message: 1
Date: Fri, 19 Aug 2011 10:33:30 +0530
From: bipin singh 
Subject: Re: [asterisk-ss7] Hardware for libss7
To: asterisk-ss7@lists.digium.com
Message-ID:

Content-Type: text/plain; charset="iso-8859-1"

Hi ,
Goto digium or sangoma sites and check what type hardware you
want to need . If you need to install libss7 then use only digium card .

On Thu, Aug 18, 2011 at 5:44 PM, Emmanuel Buamah  wrote:

> Hi All,
>
> I want to setup libss7 on ubuntu. But for now, I don't know the hardware I
> need to accomplish this in the absent of computer. And also, who can link
me
> to a very good tutorial on how to setup a very good working libss7?
>
> Cheers!
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-ss7
>



-- 
BIPIN RAGHUVANSHI
OPERATION HEAD
ASTERISK (DEVELOPMENT AND RESEARCH)
WWW.EHORIZONS.IN
011-32323262
011-46334633
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Message: 2
Date: Fri, 19 Aug 2011 12:54:01 +0500
From: Abdul Basit 
Subject: Re: [asterisk-ss7] Hardware for libss7
To: asterisk-ss7@lists.digium.com
Message-ID:

Content-Type: text/plain; charset="iso-8859-1"

libss7 and chan_ss7 both work excellent with sangoma hardware.

-- 
Regards,

Abdul Basit

On Fri, Aug 19, 2011 at 10:03 AM, bipin singh
wrote:

> Hi ,
> Goto digium or sangoma sites and check what type hardware you
> want to need . If you need to install libss7 then use only digium card .
>
> On Thu, Aug 18, 2011 at 5:44 PM, Emmanuel Buamah wrote:
>
>>  Hi All,
>>
>> I want to setup libss7 on ubuntu. But for now, I don&

Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 78, Issue 15

[Emmanuel Buamah]

Thank you all for your replies. My next question is, I have "Sangoma A200 
FXS/FXO Analog card", will libss7 or chan_ss7 work with that? I asked because, 
I know the ss7 works with only digital cards.

Thanks!

--- On Thu, 8/18/11, asterisk-ss7-requ...@lists.digium.com 
 wrote:

From: asterisk-ss7-requ...@lists.digium.com 

Subject: asterisk-ss7 Digest, Vol 78, Issue 15
To: asterisk-ss7@lists.digium.com
Date: Thursday, August 18, 2011, 8:08 PM

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Today's Topics:

   1. Re: Hardware for libss7 (bipin singh)
   2. Re: Hardware for libss7 (Abdul Basit)
   3. ??:  Hardware for libss7 (Wayne)
   4. Re: ??:  Hardware for libss7 (James zhu)


--

Message: 1
Date: Fri, 19 Aug 2011 10:33:30 +0530
From: bipin singh 
Subject: Re: [asterisk-ss7] Hardware for libss7
To: asterisk-ss7@lists.digium.com
Message-ID:
    
Content-Type: text/plain; charset="iso-8859-1"

Hi ,
            Goto digium or sangoma sites and check what type hardware you
want to need . If you need to install libss7 then use only digium card .

On Thu, Aug 18, 2011 at 5:44 PM, Emmanuel Buamah  wrote:

> Hi All,
>
> I want to setup libss7 on ubuntu. But for now, I don't know the hardware I
> need to accomplish this in the absent of computer. And also, who can link me
> to a very good tutorial on how to setup a very good working libss7?
>
> Cheers!
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-ss7
>



-- 
BIPIN RAGHUVANSHI
OPERATION HEAD
ASTERISK (DEVELOPMENT AND RESEARCH)
WWW.EHORIZONS.IN
011-32323262
011-46334633
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--

Message: 2
Date: Fri, 19 Aug 2011 12:54:01 +0500
From: Abdul Basit 
Subject: Re: [asterisk-ss7] Hardware for libss7
To: asterisk-ss7@lists.digium.com
Message-ID:
    
Content-Type: text/plain; charset="iso-8859-1"

libss7 and chan_ss7 both work excellent with sangoma hardware.

-- 
Regards,

Abdul Basit

On Fri, Aug 19, 2011 at 10:03 AM, bipin singh wrote:

> Hi ,
>             Goto digium or sangoma sites and check what type hardware you
> want to need . If you need to install libss7 then use only digium card .
>
> On Thu, Aug 18, 2011 at 5:44 PM, Emmanuel Buamah wrote:
>
>>  Hi All,
>>
>> I want to setup libss7 on ubuntu. But for now, I don't know the hardware I
>> need to accomplish this in the absent of computer. And also, who can link me
>> to a very good tutorial on how to setup a very good working libss7?
>>
>> Cheers!
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-ss7 mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-ss7
>>
>
>
>
> --
> BIPIN RAGHUVANSHI
> OPERATION HEAD
> ASTERISK (DEVELOPMENT AND RESEARCH)
> WWW.EHORIZONS.IN
> 011-32323262
> 011-46334633
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-ss7
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Message: 3
Date: Fri, 19 Aug 2011 15:59:54 +0800
From: "Wayne" 
Subject: [asterisk-ss7] ??:  Hardware for libss7
To: 
Message-ID: <000f01cc5e45$fea45a40$fbed0ec0$@cn>
Content-Type: text/plain; charset="gb2312"

Libss7 and chan_ss7 also work excellent with OpenVox E1/T1 cards. J

 

--Regards.

Wayne

 

???: asterisk-ss7-boun...@lists.digium.com
[mailto:asterisk-ss7-boun...@lists.digium.com] ?? Abdul Basit
: 2011?8?19 15:54
???: asterisk-ss7@lists.digium.com
??: Re: [asterisk-ss7] Hardware for libss7

 

libss7 and chan_ss7 both work excellent with sangoma hardware.

-- 

Regards,


Abdul Basit

 

On Fri, Aug 19, 2011 at 10:03 AM, bipin singh 
wrote:

Hi ,
            Goto digium or sangoma sites and check what 

Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 71, Issue 18

[Torrey Searle]

You need to have libpcap0.8-dev installed.

Kindest regards,
Torrey

2011/1/24 José Pablo Méndez Soto 

> Thanks Torrey,
>
> That one worked, and after doing a ./configure, make threw some errors,
> same as make install. Can you please advise?:
>
> dahdi_pcap.c:9:18: error: pcap.h: No such file or directory
> dahdi_pcap.c:74: error: expected declaration specifiers or ‘...’ before
> ‘pcap_dumper_t’
> dahdi_pcap.c: In function ‘log_packet’:
> dahdi_pcap.c:79: error: storage size of ‘hdr’ isn’t known
> dahdi_pcap.c:86: error: ‘DLT_LINUX_LAPD’ undeclared (first use in this
> function)
> dahdi_pcap.c:86: error: (Each undeclared identifier is reported only once
> dahdi_pcap.c:86: error: for each function it appears in.)
> dahdi_pcap.c:160: warning: implicit declaration of function ‘pcap_dump’
> dahdi_pcap.c:160: error: ‘dump’ undeclared (first use in this function)
> dahdi_pcap.c:161: warning: implicit declaration of function
> ‘pcap_dump_flush’
> dahdi_pcap.c:79: warning: unused variable ‘hdr’
> dahdi_pcap.c: In function ‘main’:
> dahdi_pcap.c:183: error: ‘DLT_MTP2_WITH_PHDR’ undeclared (first use in this
> function)
> dahdi_pcap.c:209: error: ‘DLT_LINUX_LAPD’ undeclared (first use in this
> function)
> dahdi_pcap.c:271: error: ‘pcap_t’ undeclared (first use in this function)
> dahdi_pcap.c:271: error: ‘pcap’ undeclared (first use in this function)
> dahdi_pcap.c:271: warning: implicit declaration of function
> ‘pcap_open_dead’
> dahdi_pcap.c:272: error: ‘pcap_dumper_t’ undeclared (first use in this
> function)
> dahdi_pcap.c:272: error: ‘dump’ undeclared (first use in this function)
> dahdi_pcap.c:272: warning: implicit declaration of function
> ‘pcap_dump_open’
> dahdi_pcap.c:291: error: too many arguments to function ‘log_packet’
> dahdi_pcap.c:295: error: too many arguments to function ‘log_packet’
> make[1]: *** [dahdi_pcap.o] Error 1
> make[1]: Leaving directory `/usr/src/dahdi-tools'
> make: *** [all] Error 2
>
>  *José Pablo Méndez
>  *
>
>
>> Date: Sun, 23 Jan 2011 15:02:22 +0100
>> From: Torrey Searle 
>> Subject: Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 71, Issue 15
>> To: asterisk-ss7@lists.digium.com
>> Message-ID:
>>
>> Content-Type: text/plain; charset="iso-8859-1"
>>
>> yes, dahdi_pcap is a new file that the patch adds.
>>
>> I think you should appy the patch with -p1 from within the dahdi-tools
>> directory
>>
>> Kindest regards,
>> Torrey
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-ss7
>
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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 71, Issue 18

[mitul]

You dont have libpcap-devel package. 

Mitul 

José Pablo Méndez Soto  wrote:

>Thanks Torrey,
>
>That one worked, and after doing a ./configure, make threw some errors, same
>as make install. Can you please advise?:
>
>dahdi_pcap.c:9:18: error: pcap.h: No such file or directory
>dahdi_pcap.c:74: error: expected declaration specifiers or ‘...’ before
>‘pcap_dumper_t’
>dahdi_pcap.c: In function ‘log_packet’:
>dahdi_pcap.c:79: error: storage size of ‘hdr’ isn’t known
>dahdi_pcap.c:86: error: ‘DLT_LINUX_LAPD’ undeclared (first use in this
>function)
>dahdi_pcap.c:86: error: (Each undeclared identifier is reported only once
>dahdi_pcap.c:86: error: for each function it appears in.)
>dahdi_pcap.c:160: warning: implicit declaration of function ‘pcap_dump’
>dahdi_pcap.c:160: error: ‘dump’ undeclared (first use in this function)
>dahdi_pcap.c:161: warning: implicit declaration of function
>‘pcap_dump_flush’
>dahdi_pcap.c:79: warning: unused variable ‘hdr’
>dahdi_pcap.c: In function ‘main’:
>dahdi_pcap.c:183: error: ‘DLT_MTP2_WITH_PHDR’ undeclared (first use in this
>function)
>dahdi_pcap.c:209: error: ‘DLT_LINUX_LAPD’ undeclared (first use in this
>function)
>dahdi_pcap.c:271: error: ‘pcap_t’ undeclared (first use in this function)
>dahdi_pcap.c:271: error: ‘pcap’ undeclared (first use in this function)
>dahdi_pcap.c:271: warning: implicit declaration of function ‘pcap_open_dead’
>dahdi_pcap.c:272: error: ‘pcap_dumper_t’ undeclared (first use in this
>function)
>dahdi_pcap.c:272: error: ‘dump’ undeclared (first use in this function)
>dahdi_pcap.c:272: warning: implicit declaration of function ‘pcap_dump_open’
>dahdi_pcap.c:291: error: too many arguments to function ‘log_packet’
>dahdi_pcap.c:295: error: too many arguments to function ‘log_packet’
>make[1]: *** [dahdi_pcap.o] Error 1
>make[1]: Leaving directory `/usr/src/dahdi-tools'
>make: *** [all] Error 2
>
> *José Pablo Méndez
>     *
>
>
>> Date: Sun, 23 Jan 2011 15:02:22 +0100
>> From: Torrey Searle 
>> Subject: Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 71, Issue 15
>> To: asterisk-ss7@lists.digium.com
>> Message-ID:
>>
>> Content-Type: text/plain; charset="iso-8859-1"
>>
>> yes, dahdi_pcap is a new file that the patch adds.
>>
>> I think you should appy the patch with -p1 from within the dahdi-tools
>> directory
>>
>> Kindest regards,
>> Torrey
>
>--
>_
>-- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
>asterisk-ss7 mailing list
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-ss7
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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 71, Issue 18

[José Pablo Méndez Soto]

Thanks Torrey,

That one worked, and after doing a ./configure, make threw some errors, same
as make install. Can you please advise?:

dahdi_pcap.c:9:18: error: pcap.h: No such file or directory
dahdi_pcap.c:74: error: expected declaration specifiers or ‘...’ before
‘pcap_dumper_t’
dahdi_pcap.c: In function ‘log_packet’:
dahdi_pcap.c:79: error: storage size of ‘hdr’ isn’t known
dahdi_pcap.c:86: error: ‘DLT_LINUX_LAPD’ undeclared (first use in this
function)
dahdi_pcap.c:86: error: (Each undeclared identifier is reported only once
dahdi_pcap.c:86: error: for each function it appears in.)
dahdi_pcap.c:160: warning: implicit declaration of function ‘pcap_dump’
dahdi_pcap.c:160: error: ‘dump’ undeclared (first use in this function)
dahdi_pcap.c:161: warning: implicit declaration of function
‘pcap_dump_flush’
dahdi_pcap.c:79: warning: unused variable ‘hdr’
dahdi_pcap.c: In function ‘main’:
dahdi_pcap.c:183: error: ‘DLT_MTP2_WITH_PHDR’ undeclared (first use in this
function)
dahdi_pcap.c:209: error: ‘DLT_LINUX_LAPD’ undeclared (first use in this
function)
dahdi_pcap.c:271: error: ‘pcap_t’ undeclared (first use in this function)
dahdi_pcap.c:271: error: ‘pcap’ undeclared (first use in this function)
dahdi_pcap.c:271: warning: implicit declaration of function ‘pcap_open_dead’
dahdi_pcap.c:272: error: ‘pcap_dumper_t’ undeclared (first use in this
function)
dahdi_pcap.c:272: error: ‘dump’ undeclared (first use in this function)
dahdi_pcap.c:272: warning: implicit declaration of function ‘pcap_dump_open’
dahdi_pcap.c:291: error: too many arguments to function ‘log_packet’
dahdi_pcap.c:295: error: too many arguments to function ‘log_packet’
make[1]: *** [dahdi_pcap.o] Error 1
make[1]: Leaving directory `/usr/src/dahdi-tools'
make: *** [all] Error 2

 *José Pablo Méndez
 *


> Date: Sun, 23 Jan 2011 15:02:22 +0100
> From: Torrey Searle 
> Subject: Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 71, Issue 15
> To: asterisk-ss7@lists.digium.com
> Message-ID:
>
> Content-Type: text/plain; charset="iso-8859-1"
>
> yes, dahdi_pcap is a new file that the patch adds.
>
> I think you should appy the patch with -p1 from within the dahdi-tools
> directory
>
> Kindest regards,
> Torrey
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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 71, Issue 15

[Torrey Searle]

yes, dahdi_pcap is a new file that the patch adds.

I think you should appy the patch with -p1 from within the dahdi-tools
directory

Kindest regards,
Torrey

2011/1/23 José Pablo Méndez Soto 

> Excuse me, gmail shortcuts misbehaving and sending emails before time
>
>
>
> Hi Torrey,
>
> Last time I got the SS7 digest version of the mail so I got wrong who
> answered my first question about dahdi_pcap installation.
>
> I tried your instructions as follows:
>
>1.
>
>svn checkout http://svn.asterisk.org/svn/dahdi/linux/trunk dahdi-linux
>
>2. Uncommented the define directive to look like
>
>#define CONFIG_DAHDI_MIRROR
>
>3.
>
>svn checkout http://svn.asterisk.org/svn/dahdi/tools/trunk dahdi-tools
>
>4.
>
>svn checkout http://svn.asterisk.org/svn/libsst/trunk libss7
>
>5. Apply patch as in issue 18362
>
> wget 'https://issues.asterisk.org/file_download.php?file_id=27784&type=bug'
> -O - | patch -p0
>
>
> The problem Im having is that patch can't find a file to actually patch.
> dahdi_pcap.c is not inside dahdi_tools package:
>
> root@ubuntu:/usr/src/dahdi-tools# wget '
> https://issues.asterisk.org/file_download.php?file_id=27784&type=bug' -O -
> | patch -p0
> --2011-01-22 22:19:52--
> https://issues.asterisk.org/file_download.php?file_id=27784&type=bug
> Resolving issues.asterisk.org... 76.164.171.231
> Connecting to issues.asterisk.org|76.164.171.231|:443... connected.
> HTTP request sent, awaiting response... 200 OK
> Length: 7720 (7.5K) [text/plain]
> Saving to: `STDOUT'
>
> 100%[=>]
> 7,720   24.0K/s   in 0.3s
>
> 2011-01-22 22:19:54 (24.0 KB/s) - `-' saved [7720/7720]
>
> patching file tools/dahdi_pcap.c
> can't find file to patch at input line 313
> Perhaps you used the wrong -p or --strip option?
> The text leading up to this was:
> --
> |Index: tools/Makefile
> |===
> |--- tools/Makefile(revision 9495)
> |+++ tools/Makefile(working copy)
> --
> File to patch:
>
>
> What am I doing wrong?
>
> Thanks,
>
>  *José Pablo Méndez
>  *
>
>
> 2011/1/13 José Pablo Méndez Soto 
>
>> Thanks Krzysztof,
>>
>>
>> Will try this as soon as possible. I see this is probably a forward from
>> Torrey's email already in the mailing lists. Can you tell me where you found
>> this post? I tried to find something in the issue tracker and in the ss7
>> mailing list to no avail.
>>
>>
>> Thanks
>>
>>>
>>>
>>> --
>>>
>>> Message: 2
>>> Date: Thu, 13 Jan 2011 18:20:36 +0100
>>> From: Krzysztof Drewicz 
>>> Subject: Re: [asterisk-ss7] adding ss7 pcap support to dahdi
>>> To: asterisk-ss7@lists.digium.com
>>> Message-ID:
>>>
>>> Content-Type: text/plain; charset="iso-8859-1"
>>>
>>> 2011/1/13 Torrey Searle 
>>>
>>> > First you take the latest svn version of the dahdi driver.
>>> >
>>> > modify
>>> >
>>> > include/dahdi/dahdi_config.h
>>> >
>>> > and uncomment the following line
>>> >
>>> > /* #define CONFIG_DAHDI_MIRROR */
>>> >
>>> >
>>> > compile and install this version of dahdi
>>> >
>>> >
>>> Only Digium's cards are supported, or every dahdi-compatible cards like
>>> Sangoma will work?
>>> -- next part --
>>> An HTML attachment was scrubbed...
>>> URL: <
>>> http://lists.digium.com/pipermail/asterisk-ss7/attachments/20110113/c8826b7a/attachment-0001.html
>>>
>>
>>
>>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
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> To UNSUBSCRIBE or update options visit:
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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 71, Issue 15

[José Pablo Méndez Soto]

Excuse me, gmail shortcuts misbehaving and sending emails before time


Hi Torrey,

Last time I got the SS7 digest version of the mail so I got wrong who
answered my first question about dahdi_pcap installation.

I tried your instructions as follows:

   1.

   svn checkout http://svn.asterisk.org/svn/dahdi/linux/trunk dahdi-linux

   2. Uncommented the define directive to look like

   #define CONFIG_DAHDI_MIRROR

   3.

   svn checkout http://svn.asterisk.org/svn/dahdi/tools/trunk dahdi-tools

   4.

   svn checkout http://svn.asterisk.org/svn/libsst/trunk libss7

   5. Apply patch as in issue 18362

wget 'https://issues.asterisk.org/file_download.php?file_id=27784&type=bug'
-O - | patch -p0


The problem Im having is that patch can't find a file to actually patch.
dahdi_pcap.c is not inside dahdi_tools package:

root@ubuntu:/usr/src/dahdi-tools# wget '
https://issues.asterisk.org/file_download.php?file_id=27784&type=bug' -O - |
patch -p0
--2011-01-22 22:19:52--
https://issues.asterisk.org/file_download.php?file_id=27784&type=bug
Resolving issues.asterisk.org... 76.164.171.231
Connecting to issues.asterisk.org|76.164.171.231|:443... connected.
HTTP request sent, awaiting response... 200 OK
Length: 7720 (7.5K) [text/plain]
Saving to: `STDOUT'

100%[=>]
7,720   24.0K/s   in 0.3s

2011-01-22 22:19:54 (24.0 KB/s) - `-' saved [7720/7720]

patching file tools/dahdi_pcap.c
can't find file to patch at input line 313
Perhaps you used the wrong -p or --strip option?
The text leading up to this was:
--
|Index: tools/Makefile
|===
|--- tools/Makefile(revision 9495)
|+++ tools/Makefile(working copy)
--
File to patch:


What am I doing wrong?

Thanks,

 *José Pablo Méndez
 *


2011/1/13 José Pablo Méndez Soto 

> Thanks Krzysztof,
>
> Will try this as soon as possible. I see this is probably a forward from
> Torrey's email already in the mailing lists. Can you tell me where you found
> this post? I tried to find something in the issue tracker and in the ss7
> mailing list to no avail.
>
>
> Thanks
>
>>
>>
>> --
>>
>> Message: 2
>> Date: Thu, 13 Jan 2011 18:20:36 +0100
>> From: Krzysztof Drewicz 
>> Subject: Re: [asterisk-ss7] adding ss7 pcap support to dahdi
>> To: asterisk-ss7@lists.digium.com
>> Message-ID:
>>
>> Content-Type: text/plain; charset="iso-8859-1"
>>
>> 2011/1/13 Torrey Searle 
>>
>> > First you take the latest svn version of the dahdi driver.
>> >
>> > modify
>> >
>> > include/dahdi/dahdi_config.h
>> >
>> > and uncomment the following line
>> >
>> > /* #define CONFIG_DAHDI_MIRROR */
>> >
>> >
>> > compile and install this version of dahdi
>> >
>> >
>> Only Digium's cards are supported, or every dahdi-compatible cards like
>> Sangoma will work?
>> -- next part --
>> An HTML attachment was scrubbed...
>> URL: <
>> http://lists.digium.com/pipermail/asterisk-ss7/attachments/20110113/c8826b7a/attachment-0001.html
>>
>
>
>
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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 71, Issue 15

[José Pablo Méndez Soto]

Hi Torrey,

Last time I got the SS7 digest version of the mail so I got wrong who
answered my first question about dahdi_pcap installation.

I tried your instructions as follows:

   1.

   svn checkout http://svn.asterisk.org/svn/dahdi/linux/trunk dahdi-linux

   2. Uncommented the define directive to look like

   #define CONFIG_DAHDI_MIRROR

   3.

   svn checkout http://svn.asterisk.org/svn/dahdi/tools/trunk dahdi-tools

   4.

   svn checkout http://svn.asterisk.org/svn/libsst/trunk libss7

   5. Apply patch as in issue 18362

wget 'https://issues.asterisk.org/file_download.php?file_id=27784&type=bug'
-O - | patch -p0


 *José Pablo Méndez
 *


2011/1/13 José Pablo Méndez Soto 

> Thanks Krzysztof,
>
> Will try this as soon as possible. I see this is probably a forward from
> Torrey's email already in the mailing lists. Can you tell me where you found
> this post? I tried to find something in the issue tracker and in the ss7
> mailing list to no avail.
>
>
> Thanks
>
>>
>>
>> --
>>
>> Message: 2
>> Date: Thu, 13 Jan 2011 18:20:36 +0100
>> From: Krzysztof Drewicz 
>> Subject: Re: [asterisk-ss7] adding ss7 pcap support to dahdi
>> To: asterisk-ss7@lists.digium.com
>> Message-ID:
>>
>> Content-Type: text/plain; charset="iso-8859-1"
>>
>> 2011/1/13 Torrey Searle 
>>
>> > First you take the latest svn version of the dahdi driver.
>> >
>> > modify
>> >
>> > include/dahdi/dahdi_config.h
>> >
>> > and uncomment the following line
>> >
>> > /* #define CONFIG_DAHDI_MIRROR */
>> >
>> >
>> > compile and install this version of dahdi
>> >
>> >
>> Only Digium's cards are supported, or every dahdi-compatible cards like
>> Sangoma will work?
>> -- next part --
>> An HTML attachment was scrubbed...
>> URL: <
>> http://lists.digium.com/pipermail/asterisk-ss7/attachments/20110113/c8826b7a/attachment-0001.html
>>
>
>
>
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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 71, Issue 15

[José Pablo Méndez Soto]

Thanks Krzysztof,

Will try this as soon as possible. I see this is probably a forward from
Torrey's email already in the mailing lists. Can you tell me where you found
this post? I tried to find something in the issue tracker and in the ss7
mailing list to no avail.


Thanks

>
>
> --
>
> Message: 2
> Date: Thu, 13 Jan 2011 18:20:36 +0100
> From: Krzysztof Drewicz 
> Subject: Re: [asterisk-ss7] adding ss7 pcap support to dahdi
> To: asterisk-ss7@lists.digium.com
> Message-ID:
>
> Content-Type: text/plain; charset="iso-8859-1"
>
> 2011/1/13 Torrey Searle 
>
> > First you take the latest svn version of the dahdi driver.
> >
> > modify
> >
> > include/dahdi/dahdi_config.h
> >
> > and uncomment the following line
> >
> > /* #define CONFIG_DAHDI_MIRROR */
> >
> >
> > compile and install this version of dahdi
> >
> >
> Only Digium's cards are supported, or every dahdi-compatible cards like
> Sangoma will work?
> -- next part --
> An HTML attachment was scrubbed...
> URL: <
> http://lists.digium.com/pipermail/asterisk-ss7/attachments/20110113/c8826b7a/attachment-0001.html
>
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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 71, Issue 3

[Anita Hall]

Hi

The update is that chan_ss7 worked flawlessly on my link with mostly default
settings.

So that means the link is not making error. Perhaps there is a bug in libss7
or some config problem. I will update.

Thanks for looking at this.

Here is my working chan_ss7 1.4.3, wanpipe 3.5.18 and asterisk 1.6.2 config
files that work with Sangoma A108 card.

# cat /etc/asterisk/ss7.conf
[linkset-siuc]
enabled => yes
enable_st => no
use_connect => yes
hunting_policy => even_mru
context => ss7
language => da
t35 => 15000,timeout
subservice => auto

[link-l1]
linkset => siuc
channels => 1-15,17-31
schannel => 16

; The first CIC
firstcic => 1

enabled => yes
echocancel => no
echocan_train => 350
echocan_taps => 128

[host-debian]
; The host is enabled
enabled => yes
opc => X
dpc => siuc:

; Syntax: links => link-name:digium-connector-no
; The links on the host is 'l1', connected to span/connector #1
links => l1:1


[devices]
wanpipe1 = WAN_AFT_TE1, Comment

[interfaces]
w1g1 = wanpipe1, , TDM_VOICE, Comment



# cat /etc/wanpipe/wanpipe1.conf
[wanpipe1]
CARD_TYPE = AFT
S514CPU = A
CommPort = PRI
AUTO_PCISLOT = NO
PCISLOT = 4
PCIBUS = 5
FE_MEDIA = E1
FE_LCODE = HDB3
FE_FRAME = NCRC4
FE_LINE = 1
TE_CLOCK = NORMAL
TE_REF_CLOCK = 0
TE_SIG_MODE = CCS
TE_HIGHIMPEDANCE = NO
TE_RX_SLEVEL = 430
LBO = 120OH
FE_TXTRISTATE = NO
MTU = 1500
UDPPORT = 9000
TTL = 255
IGNORE_FRONT_END = NO
TDMV_SPAN = 1
TDMV_DCHAN = 0
TE_AIS_MAINTENANCE = NO


TDMV_HW_DTMF = NO
TDMV_HW_FAX_DETECT = NO
HWEC_OPERATION_MODE = OCT_NORMAL


HWEC_DTMF_REMOVAL = NO
HWEC_NOISE_REDUCTION = NO
HWEC_ACUSTIC_ECHO = NO
HWEC_NLP_DISABLE = NO
HWEC_TX_AUTO_GAIN = 0
HWEC_RX_AUTO_GAIN = 0
HWEC_TX_GAIN = 0
HWEC_RX_GAIN = 0

[w1g1]
ACTIVE_CH = ALL
TDMV_HWEC = NO
MTU = 8

It may be noted that wanpipe1.conf for libss7 and chan_ss7 is the same with
both having TDMV_DCHAN = 0

# cat /etc/dahdi/system.conf
loadzone=us
defaultzone=us

#Sangoma A108 port 1 [slot:4 bus:5 span:1] 
span=1,1,0,ccs,hdb3
bchan=1-31
echocanceller=mg2,1-15,17-31


Note that in dahdi/system.conf all channels are defined as bearer channels.
chan_ss7 deals with the "D Channel" or signchan in its internal space as
defined in ss7.conf
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[asterisk-ss7] Asterisk SS7 to Audiocodes Mediant 2000 SS7

[dave george]

Anyone have any experience getting asterisk SS7 ANSI (libss7) to talk to
Mediant 2000 SS7?

 

Thanks,

Dave George

1 561 674 3838

 

 

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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 66, Issue 3

[Jorge Antillon]

Hi,

http://www.netfors.com/products

Jorge,

On Wed, 2010-08-11 at 10:32 +0530, CBC CBS wrote:

> Hi All,
> I need some information about SS7 Card.
> 
> My requirement to use this card is
> 
> We want to develop one CBC its cell broadcast system like SMSC.
> 
> The card or product can support SS7/SIGTRAN/x.25 links or protocols.
> 
> Please suggest some information about your product which can suite to
> above requirements.
> 
> THanks,
> ~ASR


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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 66, Issue 3

[CBC CBS]

Hi All,
I need some information about SS7 Card.

My requirement to use this card is

We want to develop one CBC its cell broadcast system like SMSC.

The card or product can support SS7/SIGTRAN/x.25 links or protocols.

Please suggest some information about your product which can suite to
above requirements.

THanks,
~ASR
On 8/4/10, asterisk-ss7-requ...@lists.digium.com
 wrote:
> Send asterisk-ss7 mailing list submissions to
>   asterisk-ss7@lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
>   http://lists.digium.com/mailman/listinfo/asterisk-ss7
> or, via email, send a message with subject or body 'help' to
>   asterisk-ss7-requ...@lists.digium.com
>
> You can reach the person managing the list at
>   asterisk-ss7-ow...@lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-ss7 digest..."
>
>
> Today's Topics:
>
>1. Blocking and unblocking CICs. (Jorge Antillon)
>2. Re: libss7-ss7cluster asterisk problem (Jorge Antillon)
>
>
> --
>
> Message: 1
> Date: Tue, 03 Aug 2010 13:14:04 -0600
> From: Jorge Antillon 
> Subject: [asterisk-ss7] Blocking and unblocking CICs.
> To: asterisk-ss7@lists.digium.com
> Message-ID: <1280862844.1664.5.ca...@doughnut>
> Content-Type: text/plain; charset="utf-8"
>
> Hi again,
>
> I have a 20 E1 cluster on chan_ss7 1.3, for some reason, when I try to
> block a CIC by issuing "ss7 block XX 31", the CICs do appear as blocked,
> however they do not really block the channels as the machine keeps on
> trying to send (and actually does send) calls on those cics that appear
> as Blocked Local Maintenance.
>
>
> Any clues?
>
> Thanks,
>
> -jorge
>
>
> -- next part --
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>
> --
>
> Message: 2
> Date: Tue, 03 Aug 2010 13:19:32 -0600
> From: Jorge Antillon 
> Subject: Re: [asterisk-ss7] libss7-ss7cluster asterisk problem
> To: asterisk-ss7@lists.digium.com
> Message-ID: <1280863172.1664.11.ca...@doughnut>
> Content-Type: text/plain; charset="utf-8"
>
> Hi there!,
>
> If I get chan_ss7 cluster right, that is only for sharing signaling
> channels on a single set of systems, do not think it has the ability to
> reroute/overflow incoming SIP calls , I am not sure though...
> You will then have the task to distribute in an even fashion from your
> other endpoint into the target machines.
>
> best regards,
>
> Jorge.
>
>
>
> On Fri, 2010-07-30 at 16:05 +, Nacho FX Cirijillo Najazamapeletinan
> wrote:
>
>> Estimado Jorge:
>>
>>  Como funciona tu sistema de cluster?
>>
>>  Mi problema es este:
>>
>>  Mediante un modulo que creamos para asterisk (que incluimos
>> en /asterisk/res y compilamos junto con el asterisk) generamos dos
>> llamadas y creamos un bridge entre ellas. Todo esto funciona sin
>> problemas. La duda es por ejemplo en la maquina A (que ser?a el master
>> en el cluster por as? decirlo al que le llega la se?alizaci?n) se me
>> acaban los cics libres, la pregunta es :
>>
>> ?la funcionalidad del cluster chan_ss7 cursar?a las llamadas en los
>> cics de la maquina B (que ser?a el slave)?.
>>
>> ?o el cluster tiene otro objetivo?
>>
>> Espero me ayudes a resolver esta duda.
>>
>> Muchas gracias y saludos.
>>
>> Ignacio Zamora
>>
>>
>>
>> __
>> From: jantil...@ticom.co.cr
>> To: asterisk-ss7@lists.digium.com
>> Date: Mon, 26 Jul 2010 13:01:47 -0600
>> Subject: Re: [asterisk-ss7] libss7-ss7cluster asterisk problem
>>
>> Hi,
>>
>> [linkset-ls1]
>> enabled => yes
>> enable_st => no
>> use_connect => no
>> hunting_policy => seq_lth
>> context => congestion
>> language => en
>> t35 => 15000,timeout
>> subservice => auto
>> noa => 1
>> variant => ITU
>>
>> [link-l1]
>> linkset => ls1
>> channels => 1-30
>> schannel => 3...@sjo_p0bx_01:12000
>> firstcic => 1
>> enabled => yes
>> echocancel => 31speech
>> echocan_train => 350
>> echocan_taps => 128
>>
>> [link-l2]
>> linkset => ls1
>> channels => 1-31
>> schannel =>
>> firstcic => 32
>> enabled => no
>>
>> down to the other links.
>> ...
>>
>> [host-SJO_P0BX_01]
>> enabled => yes
>> opc => 0x601
>> dpc => ls1:0x103
>> default_linkset => ls1
>> links => l1:1,l2:2,l3:3,l4:4
>> ssn => 7
>> if-1 => 192.168.10.24
>>
>> ...
>>
>> down to the other hosts.
>>
>>
>> Regards,
>>
>> jorge.
>>
>> On Mon, 2010-07-26 at 15:11 +, Nacho FX Cirijillo
>> Najazamapeletinan wrote:
>>
>> Jorge:
>>
>>  Podr?as mostrarme tu ss7.conf por favor
>>
>>  Please show me your ss7.conf.
>>
>> Thanks
>>
>>
>>
>>
>> __
>>
>> From: nachofxciriji...@hotmail.com
>> To: asterisk-ss7@lists.digium.com
>> Dat

Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 61, Issue 35

[Ziad Salameh]

Dear All,

I am not sure if I can post this here or I should start a new thread but
here goes...
I setup Libss7 with asterisk 1.6 , linkset is UP, I can make successful
outbound calls , I can hear the other party well.
Now the odd thing is that when someone tries an inbound call , they hear
nothing I tried setting up MOH, I also tried to record the incoming call but
I hear nothing.
Is anyone facing such an issue, will you please help me shed some light on
this issue.

Thank you,
Ziad

-Original Message-
From: asterisk-ss7-boun...@lists.digium.com
[mailto:asterisk-ss7-boun...@lists.digium.com] On Behalf Of
asterisk-ss7-requ...@lists.digium.com
Sent: Friday, March 26, 2010 9:09 AM
To: asterisk-ss7@lists.digium.com
Subject: asterisk-ss7 Digest, Vol 61, Issue 35

Send asterisk-ss7 mailing list submissions to
asterisk-ss7@lists.digium.com

To subscribe or unsubscribe via the World Wide Web, visit
http://lists.digium.com/mailman/listinfo/asterisk-ss7
or, via email, send a message with subject or body 'help' to
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You can reach the person managing the list at
asterisk-ss7-ow...@lists.digium.com

When replying, please edit your Subject line so it is more specific
than "Re: Contents of asterisk-ss7 digest..."


Today's Topics:

   1. help (paul marcovici)


--

Message: 1
Date: Fri, 26 Mar 2010 00:08:30 -0700 (PDT)
From: paul marcovici 
Subject: [asterisk-ss7] help
To: asterisk-ss7@lists.digium.com
Message-ID: <966245.47373...@web113601.mail.gq1.yahoo.com>
Content-Type: text/plain; charset="us-ascii"







From: "asterisk-ss7-requ...@lists.digium.com"

To: asterisk-ss7@lists.digium.com
Sent: Fri, March 26, 2010 8:27:57 AM
Subject: asterisk-ss7 Digest, Vol 61, Issue 34

Send asterisk-ss7 mailing list submissions to
asterisk-ss7@lists.digium.com

To subscribe or unsubscribe via the World Wide Web, visit
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When replying, please edit your Subject line so it is more specific
than "Re: Contents of asterisk-ss7 digest..."


Today's Topics:

   1. Re: IAM-REL instead of IAM-ACM-REL (Bruno Rodrigues de Mello)
   2. Re: IAM-REL instead of IAM-ACM-REL (Domjan Attila)
   3. Where's Matt been?  Well, here's the explanation
  (Matthew Fredrickson)
   4. Re: IAM-REL instead of IAM-ACM-REL (Anil Gupta)
   5. Re: IAM-REL instead of IAM-ACM-REL (Anil Gupta)


--

Message: 1
Date: Thu, 25 Mar 2010 15:07:30 -0300
From: "Bruno Rodrigues de Mello" 
Subject: Re: [asterisk-ss7] IAM-REL instead of IAM-ACM-REL
To: 
Message-ID: 
Content-Type: text/plain; format=flowed; charset="utf-8";
reply-type=original

Hi Attila, your patch more one time work without problems thank you again.

Regarding this email let me check other problem with you. I have a asterisk 
box between a TDM Switch and a Cisco Gateway, using SS7 to TDM and ISDN to 
CISCO

TDMASTERISK<---ISDN---> CISCO

The call come from SS7 side and asterisk forward this call to ISDN. The 
cisco gateway plays a message in PROCEEDING. This message ask the user to 
put some digits. The calling side listen the message but when he put the 
digits the ISDN side don't receive the DTMF.

I make a dahdi_monitor in SS7 channel and in ISDN channel. In the SS7 I can 
listen the audio and the dtmf and in ISDN side I don't listen the DTMF only 
the audio

I don't know why asterisk don't foward the audio received on SS7 side before

the ANM  during the early media.

Do you know anything about this ?


Regards,
Bruno Rodrigues

--
From: "Attila Domjan" 
Sent: Wednesday, March 17, 2010 9:17 AM
To: 
Subject: Re: [asterisk-ss7] IAM-REL instead of IAM-ACM-REL

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Message: 2
Date: Thu, 25 Mar 2010 23:47:30 +0100
From: Domjan Attila 
Subject: Re: [asterisk-ss7] IAM-REL instead of IAM-ACM-REL
To: asterisk-ss7@lists.digium.com
Message-ID: <1269557250.2046.1.ca...@localhost>
Content-Type: text/plain; charset="utf-8"

Hi,
I'm not sure it should work... it is not ss7 it is dahdi issue

On Thu, 2010-03-25 at 15:07 -0300, Bruno Rodrigues de Mello wrote:
> Hi Attila, your patch more one time work without problems thank you again.
> 
> Regarding this email let me check other problem with you. I have a
asterisk 
> box between a TDM Switch and a Cisco Gateway

[asterisk-ss7] Asterisk+ss7 to NEC SV8300

[Richard Kenner]

I'm new to Asterisk and SS7 and I'm trying to see if I can use it to
connect a NEC SV8300 to Asterisk.  I'm using a Digium card and was able to
make a PRI connection to a CD-PRTA card on the SV8300.  I now added a
CD-CCTA card and using ss7 instead of PRI, but this time am having
problems.  The linkset doesn't come up and it appears that no messages are
being recieved from the SV8300.  Does anybody have any suggestions for a
next step?  Has this ever been tried?  I'm using 

span=2,2,0,esf,b8zs
# termtype: te
bchan=25-47
mtp2=48
echocanceller=mg2,25-47

in system.conf.


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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 55, Issue 13

[Rajesh Mahajan]

Hi

We don't have sccp protocol we are looking for same.

We want to integrate INAP/MAP based applications like smsc

We have look into Openss7 but how we integrate the same with asterisk ss7 stack



On Wed, Sep 23, 2009 at 10:30 PM,
 wrote:
> Send asterisk-ss7 mailing list submissions to
>        asterisk-...@lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
>        http://lists.digium.com/mailman/listinfo/asterisk-ss7
> or, via email, send a message with subject or body 'help' to
>        asterisk-ss7-requ...@lists.digium.com
>
> You can reach the person managing the list at
>        asterisk-ss7-ow...@lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-ss7 digest..."
>
>
> Today's Topics:
>
>   1. How to integrate SCCP over MTP3 (Rajesh Mahajan)
>   2. Re: How to integrate SCCP over MTP3 (Jan Berger)
>
>
> --
>
> Message: 1
> Date: Wed, 23 Sep 2009 16:05:57 +0530
> From: Rajesh Mahajan 
> Subject: [asterisk-ss7] How to integrate SCCP over MTP3
> To: asterisk-ss7@lists.digium.com
> Message-ID:
>        
> Content-Type: text/plain; charset=ISO-8859-1
>
> Dear All
>
> What is possible ways to integrate SCCP Protocol Over MTP3 Layer in SS7
>
>
>
> --
>
> Message: 2
> Date: Wed, 23 Sep 2009 17:42:24 +0200
> From: Jan Berger 
> Subject: Re: [asterisk-ss7] How to integrate SCCP over MTP3
> To: 
> Message-ID: 
> Content-Type: text/plain; charset="iso-8859-1"
>
>
> hi Rajesh,
>
>
>
> Do you have a SCCP protocol or are you looking for one?
>
>
>
> Also what is your application/usage?
>
>
>
> You might want to look at OpenSS7 and see what they have on the SCCP/TCAP 
> side. Other than that its not much out in open source on the higher layers 
> yet.
>
>
>
> Jan
>
>> Date: Wed, 23 Sep 2009 16:05:57 +0530
>> From: rajeshmahaja...@gmail.com
>> To: asterisk-ss7@lists.digium.com
>> Subject: [asterisk-ss7] How to integrate SCCP over MTP3
>>
>> Dear All
>>
>> What is possible ways to integrate SCCP Protocol Over MTP3 Layer in SS7
>>
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Re: [asterisk-ss7] Asterisk/SS7/M3UA

[Russ Meyerriecks]

Just FYI if you don't want to bother learning about and 
setting up SS7, Verisign offers a ss7-sip gateway that
might be an easier option.

Russ Meyerriecks
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
direct: +1 256-428-6025
Check us out at: www.digium.com & www.asterisk.org

- Original Message -
From: "James Wiegand" 
To: asterisk-ss7@lists.digium.com
Sent: Friday, September 18, 2009 10:10:12 AM GMT -06:00 US/Canada Central
Subject: [asterisk-ss7] Asterisk/SS7/M3UA

Hi,

I'm new to all this SS7 stuff and we need to get Verisign working on
Asterisk.  What is the general cookbook for getting this going,
assuming Asterisk/SS7/M3UA is a workable option?

Thanks in advance,
-jim

--
--
Jim Wiegand
---
Home:  originaljimda...@gmail.com
AIM: originaljimdandy

-- 
-- 
Jim Wiegand
---
Home:  originaljimda...@gmail.com
AIM: originaljimdandy

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[asterisk-ss7] Asterisk/SS7/M3UA

[James Wiegand]

Hi,

I'm new to all this SS7 stuff and we need to get Verisign working on
Asterisk.  What is the general cookbook for getting this going,
assuming Asterisk/SS7/M3UA is a workable option?

Thanks in advance,
-jim

--
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---
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AIM: originaljimdandy

-- 
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---
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AIM: originaljimdandy

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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 55, Issue 4

[Rajesh Mahajan]

Thx, Now in asterisk ss7 is up.

Is there specific configuration of ISUP layer.


On Tue, Sep 15, 2009 at 10:30 PM,
 wrote:
> Send asterisk-ss7 mailing list submissions to
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>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-ss7 digest..."
>
>
> Today's Topics:
>
>   1. Asterisk SS7 Linkset Configurations (Rajesh Mahajan)
>   2. Re: Asterisk SS7 Linkset Configurations (Attila Domjan)
>
>
> --
>
> Message: 1
> Date: Tue, 15 Sep 2009 13:28:23 +0530
> From: Rajesh Mahajan 
> Subject: [asterisk-ss7] Asterisk SS7 Linkset Configurations
> To: asterisk-ss7@lists.digium.com
> Message-ID:
>        
> Content-Type: text/plain; charset=ISO-8859-1
>
> Dear All
>
> We are using following Software/Hardware.
> 1.asterisk-1.6.1.4
> 2.dahdi-linux-2.2.0.2
> 3.dahdi-tools-2.2.0
> 4.libss7-1.0.2
> 5.wanpipe-3.5.6
>
> Hardware:
> Sangoma Technologies Corp. A104u Quad T1/E1 AFT
>
> Configurations:
>
> /etc/asterisk/chan_dahdi.conf
>
> [channels]
> ;switchtype=euroisdn
> usecallerid=yes
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> group=1
> callgroup=1
> pickupgroup=1
>
>
> signalling = ss7
> ss7type = itu
> ss7_called_nai=dynamic
> ss7_calling_nai=dynamic
> networkindicator=international
>
> ; port 1
> linkset = 1
> group = 1
> signalling=ss7
> ss7type = itu
> context = default
> pointcode = 8002
> adjpointcode = 9146
> defaultdpc = 9146
> networkindicator = international
> cicbeginswith = 1
> channel => 1-15
> cicbeginswith = 17
> channel => 17-31
> sigchan = 16
> ;
>
> /etc/dahdi/system.conf
>
> loadzone=us
> defaultzone=us
>
> #Sangoma A104 port 1 [slot:1 bus:12 span:1] 
> span=1,0,0,ccs,hdb3
> bchan=1-15,17-31
> #echocanceller=mg2,1-15,17-31
> #hardhdlc=16
> dchan=16
>
>
> On Asterisk it is showing ss7 stack is up :(
>
> 1. asterisk -rx "ss7 show linkset 1"
> No SS7 running on linkset 1
>
> 2.asterisk -rx  "dahdi show channels"
>   Chan Extension  Context         Language   MOH Interpret
> Blocked    State
>  pseudo            default                    default
>       In Service
>
>
>
> Pls suggest what's wrong in the configurations
>
> While it is coming up by
>
> ./ss7linktest 16 itu 8002 9146
> Starting link 1
> Link state change: IDLE -> NOTALIGNED
> Len = 4 [ ff ff 01 00 ]
> FSN: 127 FIB 1
> BSN: 127 BIB 1
>>[0] LSSU SIO
>
> Len = 4 [ ff ff 01 00 ]
> FSN: 127 FIB 1
> BSN: 127 BIB 1
> <[0] LSSU SIO
>
> Link state change: NOTALIGNED -> ALIGNED
> Len = 4 [ ff ff 01 02 ]
> FSN: 127 FIB 1
> BSN: 127 BIB 1
>>[0] LSSU SIE
>
> Len = 4 [ ff ff 01 02 ]
> FSN: 127 FIB 1
> BSN: 127 BIB 1
> <[0] LSSU SIE
>
> Link state change: ALIGNED -> PROVING
> T4 expired!
> Link state change: PROVING -> ALIGNEDREADY
> Len = 3 [ ff ff 00 ]
> FSN: 127 FIB 1
> BSN: 127 BIB 1
>>[0] FISU
>
> Len = 3 [ ff ff 00 ]
> FSN: 127 FIB 1
> BSN: 127 BIB 1
> <[0] FISU
>
> Link state change: ALIGNEDREADY -> INSERVICE
> [0] MTP2 link up
> Len = 9 [ ff 80 06 80 ba a3 d0 07 17 ]
> FSN: 0 FIB 1
> BSN: 127 BIB 1
>>[0] MSU
> [ ff 80 06 ]
> ??? Network Indicator: 2 Priority: 0 User Part: NET_MNG (0)
> ??? [ 80 ]
> ??? OPC 8002 DPC 9146 SLS 0
> ??? [ ba a3 d0 07 ]
> ??? H0: 7 H1: 1
> ??? Message type: TRA
> ??? [ 17 ]
>
> Len = 20 [ ff 81 11 81 ba a3 d0 07 11 a0 32 35 36 34 32 38 36 32 38 38 ]
> FSN: 1 FIB 1
> BSN: 127 BIB 1
>>[0] MSU
> [ ff 81 11 ]
> ??? Network Indicator: 2 Priority: 0 User Part: STD_TEST (1)
> ??? [ 81 ]
> ??? OPC 8002 DPC 9146 SLS 0
> ??? [ ba a3 d0 07 ]
> ??? H0: 1 H1: 1
> ??? [ 11 ]
>
> Len = 14 [ ff 80 0b 81 42 9f ee 08 11 40 01 02 03 04 ]
> FSN: 0 FIB 1
> BSN: 127 BIB 1
> <[0] MSU
> [ ff 80 0b ]
> ??? Network Indicator: 2 Priority: 0 User Part: STD_TEST (1)
> ??? [ 81 ]
> ??? OPC 9146 DP

Re: [asterisk-ss7] Asterisk SS7 Linkset Configurations

[Attila Domjan]

Hi,
try move the sigchan line after the networkindicator line.
A

On Tue, 2009-09-15 at 13:28 +0530, Rajesh Mahajan wrote:
> Dear All
> 
> We are using following Software/Hardware.
> 1.asterisk-1.6.1.4
> 2.dahdi-linux-2.2.0.2
> 3.dahdi-tools-2.2.0
> 4.libss7-1.0.2
> 5.wanpipe-3.5.6
> 
> Hardware:
> Sangoma Technologies Corp. A104u Quad T1/E1 AFT
> 
> Configurations:
> 
> /etc/asterisk/chan_dahdi.conf
> 
> [channels]
> ;switchtype=euroisdn
> usecallerid=yes
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> group=1
> callgroup=1
> pickupgroup=1
> 
> 
> signalling = ss7
> ss7type = itu
> ss7_called_nai=dynamic
> ss7_calling_nai=dynamic
> networkindicator=international
> 
> ; port 1
> linkset = 1
> group = 1
> signalling=ss7
> ss7type = itu
> context = default
> pointcode = 8002
> adjpointcode = 9146
> defaultdpc = 9146
> networkindicator = international
> cicbeginswith = 1
> channel => 1-15
> cicbeginswith = 17
> channel => 17-31
> sigchan = 16
> ;
> 
> /etc/dahdi/system.conf
> 
> loadzone=us
> defaultzone=us
> 
> #Sangoma A104 port 1 [slot:1 bus:12 span:1] 
> span=1,0,0,ccs,hdb3
> bchan=1-15,17-31
> #echocanceller=mg2,1-15,17-31
> #hardhdlc=16
> dchan=16
> 
> 
> On Asterisk it is showing ss7 stack is up :(
> 
> 1. asterisk -rx "ss7 show linkset 1"
> No SS7 running on linkset 1
> 
> 2.asterisk -rx  "dahdi show channels"
>Chan Extension  Context Language   MOH Interpret
> BlockedState
>  pseudodefaultdefault
>In Service
> 
> 
> 
> Pls suggest what's wrong in the configurations
> 
> While it is coming up by
> 
> ./ss7linktest 16 itu 8002 9146
> Starting link 1
> Link state change: IDLE -> NOTALIGNED
> Len = 4 [ ff ff 01 00 ]
> FSN: 127 FIB 1
> BSN: 127 BIB 1
> >[0] LSSU SIO
> 
> Len = 4 [ ff ff 01 00 ]
> FSN: 127 FIB 1
> BSN: 127 BIB 1
> <[0] LSSU SIO
> 
> Link state change: NOTALIGNED -> ALIGNED
> Len = 4 [ ff ff 01 02 ]
> FSN: 127 FIB 1
> BSN: 127 BIB 1
> >[0] LSSU SIE
> 
> Len = 4 [ ff ff 01 02 ]
> FSN: 127 FIB 1
> BSN: 127 BIB 1
> <[0] LSSU SIE
> 
> Link state change: ALIGNED -> PROVING
> T4 expired!
> Link state change: PROVING -> ALIGNEDREADY
> Len = 3 [ ff ff 00 ]
> FSN: 127 FIB 1
> BSN: 127 BIB 1
> >[0] FISU
> 
> Len = 3 [ ff ff 00 ]
> FSN: 127 FIB 1
> BSN: 127 BIB 1
> <[0] FISU
> 
> Link state change: ALIGNEDREADY -> INSERVICE
> [0] MTP2 link up
> Len = 9 [ ff 80 06 80 ba a3 d0 07 17 ]
> FSN: 0 FIB 1
> BSN: 127 BIB 1
> >[0] MSU
> [ ff 80 06 ]
> Network Indicator: 2 Priority: 0 User Part: NET_MNG (0)
> [ 80 ]
> OPC 8002 DPC 9146 SLS 0
> [ ba a3 d0 07 ]
> H0: 7 H1: 1
> Message type: TRA
> [ 17 ]
> 
> Len = 20 [ ff 81 11 81 ba a3 d0 07 11 a0 32 35 36 34 32 38 36 32 38 38 ]
> FSN: 1 FIB 1
> BSN: 127 BIB 1
> >[0] MSU
> [ ff 81 11 ]
> Network Indicator: 2 Priority: 0 User Part: STD_TEST (1)
> [ 81 ]
> OPC 8002 DPC 9146 SLS 0
> [ ba a3 d0 07 ]
> H0: 1 H1: 1
> [ 11 ]
> 
> Len = 14 [ ff 80 0b 81 42 9f ee 08 11 40 01 02 03 04 ]
> FSN: 0 FIB 1
> BSN: 127 BIB 1
> <[0] MSU
> [ ff 80 0b ]
> Network Indicator: 2 Priority: 0 User Part: STD_TEST (1)
> [ 81 ]
> OPC 9146 DPC 8002 SLS 0
> [ 42 9f ee 08 ]
> H0: 1 H1: 1
> [ 11 ]
> 
> Len = 14 [ 80 82 0b 81 ba a3 d0 07 21 40 01 02 03 04 ]
> FSN: 2 FIB 1
> BSN: 0 BIB 1
> >[0] MSU
> [ 80 82 0b ]
> Network Indicator: 2 Priority: 0 User Part: STD_TEST (1)
> [ 81 ]
> OPC 8002 DPC 9146 SLS 0
> [ ba a3 d0 07 ]
> H0: 1 H1: 2
> [ 21 ]
> 
> [0] --- SS7 Up ---
> Len = 14 [ 80 83 0b 85 ba a3 d0 07 01 00 17 01 01 17 ]
> FSN: 3 FIB 1
> BSN: 0 BIB 1
> >[0] MSU
> [ 80 83 0b ]
> Network Indicator: 2 Priority: 0 User Part: ISUP (5)
> [ 85 ]
> OPC 8002 DPC 9146 SLS 0
> [ ba a3 d0 07 ]
> CIC: 1
> [ 01 00 ]
> Message Type: GRS
> [ 17 ]
> --VARIABLE LENGTH PARMS[1]--
> Range and status:
> Range: 23
> [ 01 17 ]
> 
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[asterisk-ss7] Asterisk SS7 Linkset Configurations

[Rajesh Mahajan]

Dear All

We are using following Software/Hardware.
1.asterisk-1.6.1.4
2.dahdi-linux-2.2.0.2
3.dahdi-tools-2.2.0
4.libss7-1.0.2
5.wanpipe-3.5.6

Hardware:
Sangoma Technologies Corp. A104u Quad T1/E1 AFT

Configurations:

/etc/asterisk/chan_dahdi.conf

[channels]
;switchtype=euroisdn
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
group=1
callgroup=1
pickupgroup=1


signalling = ss7
ss7type = itu
ss7_called_nai=dynamic
ss7_calling_nai=dynamic
networkindicator=international

; port 1
linkset = 1
group = 1
signalling=ss7
ss7type = itu
context = default
pointcode = 8002
adjpointcode = 9146
defaultdpc = 9146
networkindicator = international
cicbeginswith = 1
channel => 1-15
cicbeginswith = 17
channel => 17-31
sigchan = 16
;

/etc/dahdi/system.conf

loadzone=us
defaultzone=us

#Sangoma A104 port 1 [slot:1 bus:12 span:1] 
span=1,0,0,ccs,hdb3
bchan=1-15,17-31
#echocanceller=mg2,1-15,17-31
#hardhdlc=16
dchan=16


On Asterisk it is showing ss7 stack is up :(

1. asterisk -rx "ss7 show linkset 1"
No SS7 running on linkset 1

2.asterisk -rx  "dahdi show channels"
   Chan Extension  Context Language   MOH Interpret
BlockedState
 pseudodefaultdefault
   In Service



Pls suggest what's wrong in the configurations

While it is coming up by

./ss7linktest 16 itu 8002 9146
Starting link 1
Link state change: IDLE -> NOTALIGNED
Len = 4 [ ff ff 01 00 ]
FSN: 127 FIB 1
BSN: 127 BIB 1
>[0] LSSU SIO

Len = 4 [ ff ff 01 00 ]
FSN: 127 FIB 1
BSN: 127 BIB 1
<[0] LSSU SIO

Link state change: NOTALIGNED -> ALIGNED
Len = 4 [ ff ff 01 02 ]
FSN: 127 FIB 1
BSN: 127 BIB 1
>[0] LSSU SIE

Len = 4 [ ff ff 01 02 ]
FSN: 127 FIB 1
BSN: 127 BIB 1
<[0] LSSU SIE

Link state change: ALIGNED -> PROVING
T4 expired!
Link state change: PROVING -> ALIGNEDREADY
Len = 3 [ ff ff 00 ]
FSN: 127 FIB 1
BSN: 127 BIB 1
>[0] FISU

Len = 3 [ ff ff 00 ]
FSN: 127 FIB 1
BSN: 127 BIB 1
<[0] FISU

Link state change: ALIGNEDREADY -> INSERVICE
[0] MTP2 link up
Len = 9 [ ff 80 06 80 ba a3 d0 07 17 ]
FSN: 0 FIB 1
BSN: 127 BIB 1
>[0] MSU
[ ff 80 06 ]
    Network Indicator: 2 Priority: 0 User Part: NET_MNG (0)
    [ 80 ]
    OPC 8002 DPC 9146 SLS 0
    [ ba a3 d0 07 ]
    H0: 7 H1: 1
    Message type: TRA
    [ 17 ]

Len = 20 [ ff 81 11 81 ba a3 d0 07 11 a0 32 35 36 34 32 38 36 32 38 38 ]
FSN: 1 FIB 1
BSN: 127 BIB 1
>[0] MSU
[ ff 81 11 ]
    Network Indicator: 2 Priority: 0 User Part: STD_TEST (1)
    [ 81 ]
    OPC 8002 DPC 9146 SLS 0
    [ ba a3 d0 07 ]
    H0: 1 H1: 1
    [ 11 ]

Len = 14 [ ff 80 0b 81 42 9f ee 08 11 40 01 02 03 04 ]
FSN: 0 FIB 1
BSN: 127 BIB 1
<[0] MSU
[ ff 80 0b ]
    Network Indicator: 2 Priority: 0 User Part: STD_TEST (1)
    [ 81 ]
    OPC 9146 DPC 8002 SLS 0
    [ 42 9f ee 08 ]
    H0: 1 H1: 1
    [ 11 ]

Len = 14 [ 80 82 0b 81 ba a3 d0 07 21 40 01 02 03 04 ]
FSN: 2 FIB 1
BSN: 0 BIB 1
>[0] MSU
[ 80 82 0b ]
    Network Indicator: 2 Priority: 0 User Part: STD_TEST (1)
    [ 81 ]
    OPC 8002 DPC 9146 SLS 0
    [ ba a3 d0 07 ]
    H0: 1 H1: 2
    [ 21 ]

[0] --- SS7 Up ---
Len = 14 [ 80 83 0b 85 ba a3 d0 07 01 00 17 01 01 17 ]
FSN: 3 FIB 1
BSN: 0 BIB 1
>[0] MSU
[ 80 83 0b ]
    Network Indicator: 2 Priority: 0 User Part: ISUP (5)
    [ 85 ]
    OPC 8002 DPC 9146 SLS 0
    [ ba a3 d0 07 ]
    CIC: 1
    [ 01 00 ]
    Message Type: GRS
    [ 17 ]
    --VARIABLE LENGTH PARMS[1]--
    Range and status:
    Range: 23
    [ 01 17 ]

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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 54, Issue 3

[Domjan Attila]

Hi, 
the sangome released new firmware for aft cards:

Firmware bug fix for Parity errors
This bug can cause some systems to have a low-level parity
error on the PCI/PCI-X/PCIe bus resulting in a kernel panic or
machine check exception panic.  If you experience a kernel
panic or machine check exception after this update please
contact techd...@sangoma.com so we can resolve the issue

next window I will test it

A

On Mon, 2009-08-03 at 20:47 +0200, Domjan Attila wrote:
> On Mon, 2009-08-03 at 21:17 +0300, RESEARCH wrote:
> > Hi Attila
> > 
> > I had the mtp2 setup and not hardhdlc. Im using sangoma 3.5.4.17. I think it
> > also crush the system as I tried to put the crossover cable and the system
> > constantly reboot. I will try 3.4.1 shortly and revert
> > 
> 
> Yes it's an sangoma bug. Please capture the kernel crash if it is
> possible. I wrote this bug to the Sangoma, but I had to revert
> immedietly because I had a short maintenance window.
> 
> A
> > Sam
> > --
> > Hi,
> > the problem is:
> > 
> > hardhdlc=16
> > replace to
> > mtp2=16
> > 
> > hint: don't try wanpipe 3.4.4 this cause kernel panic with mtp2
> > wanpipe-3.4.1 + dahdi 2.1.0.4 works fine here
> > 
> > Regards,
> > Attila
> > 
> > On Mon, 2009-08-03 at 00:38 -0500, resea...@businesstz.com wrote:
> > > Hi Alejendro
> > > 
> > > Have you managed to find out the solution to your problem!! Im facing the
> > > same problem. The same link connects on another digium card running in
> > > parallel with exactly the same configuration.
> > > 
> > > Please share with me if you have a clue as I want to try sangoma..i'm
> > > using A104DE instead
> > > 
> > > Kind regards
> > > Sam
> > > 
> > > 
> > > --
> > > 
> > > Message: 1
> > > Date: Thu, 4 Jun 2009 11:29:15 -0600
> > > From: Alejandro Mej?a Evertsz 
> > > Subject: [asterisk-ss7] Sangoma A102 + Dahdi + libss7 + Asterisk -
> > >   Linkset DOWN
> > > To: 
> > > Message-ID: <006501c9e539$fd815d20$f88417...@com>
> > > Content-Type: text/plain; charset="iso-8859-1"
> > > 
> > > Hi,
> > > 
> > > 
> > > 
> > > I have problems, as linkset shows as DOWN all the time with the following
> > > scenario:
> > > 
> > > 
> > > 
> > > -Dell server with Sangoma A102 (2 E1/T1 Card)
> > > 
> > > -Ubuntu Server 8.10
> > > 
> > > -Dahdi Linux 2.1.0.4
> > > 
> > > -libss7 1.0.2
> > > 
> > > -wanpipe 3.4.1
> > > 
> > > -Asterisk 1.6.1.0
> > > 
> > > 
> > > 
> > > ===/etc/dahdi/system.conf===
> > > 
> > > loadzone=us
> > > 
> > > defaultzone=us
> > > 
> > > 
> > > 
> > > #Sangoma A102 port 1 [slot:2 bus:10 span:1] 
> > > 
> > > span=1,0,0,ccs,hdb3
> > > 
> > > bchan=1-15,17-31
> > > 
> > > echocanceller=mg2,1-15,17-31
> > > 
> > > hardhdlc=16
> > > 
> > > 
> > > 
> > > #Sangoma A102 port 2 [slot:2 bus:10 span:2] 
> > > 
> > > span=2,1,0,ccs,hdb3
> > > 
> > > bchan=32-46,48-62
> > > 
> > > echocanceller=mg2,32-46,48-62
> > > 
> > > hardhdlc=47
> > > 
> > > ===END==
> > > 
> > > 
> > > 
> > > ===/etc/asterisk/chan_dahdi.conf===
> > > 
> > > [trunkgroups]
> > > 
> > > 
> > > 
> > > [channels]
> > > 
> > > context=default
> > > 
> > > usecallerid=yes
> > > 
> > > hidecallerid=no
> > > 
> > > callwaiting=yes
> > > 
> > > usecallingpres=yes
> > > 
> > > callwaitingcallerid=yes
> > > 
> > > threewaycalling=yes
> > > 
> > > transfer=yes
> > > 
> > > canpark=yes
> > > 
> > > cancallforward=yes
> > > 
> > > callreturn=yes
> > > 
> > > echocancel=yes
> > > 
> > > echocancelwhenbridged=yes
> > > 
> > > relaxdtmf=yes
> > > 
> > > rxgain=0.0
> > > 
> > > txgain=0.0
> > > 
> > > group=1
> > > 
> > > callgroup=1
> > > 
> > > pickupgroup=1
> > > 
> > > immediate=no
> > > 
> > > 
> > > 
> > > ;Sangoma A102 port 1 [slot:2 bus:10 span:1] 
> > > 
> > > context=from-pstn
> > > 
> > > group=0
> > > 
> > > echocancel=yes
> > > 
> > > ;switchtype=euroisdn
> > > 
> > > signalling=ss7
> > > 
> > > ss7type=itu
> > > 
> > > linkset = 1
> > > 
> > > pointcode = 1234
> > > 
> > > adjpointcode = 4321
> > > 
> > > defaultdpc = 4132
> > > 
> > > cicbeginswith = 1
> > > 
> > > networkindicator=national
> > > 
> > > sigchan = 16
> > > 
> > > ;sigchannel = 16
> > > 
> > > channel => 1-15,17-31
> > > 
> > > 
> > > 
> > > ;Sangoma A102 port 2 [slot:2 bus:10 span:2] 
> > > 
> > > context=from-pstn
> > > 
> > > group=1
> > > 
> > > echocancel=yes
> > > 
> > > ;switchtype=euroisdn
> > > 
> > > signalling=ss7
> > > 
> > > ss7type=itu
> > > 
> > > linkset = 2
> > > 
> > > pointcode = 4321
> > > 
> > > adjpointcode = 1234
> > > 
> > > defaultdpc = 4132
> > > 
> > > cicbeginswith = 32
> > > 
> > > networkindicator=national
> > > 
> > > sigchan = 47
> > > 
> > > ;sigchannel = 47
> > > 
> > > channel => 32-46,48-62
> > > 
> > > =END=
> > > 
> > > 
> > > 
> > > As I don?t have an SS7 equipment/link to test, I decided to make a
> > crossover
> > > cable (1<--

Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 54, Issue 3

[Domjan Attila]

On Mon, 2009-08-03 at 21:17 +0300, RESEARCH wrote:
> Hi Attila
> 
> I had the mtp2 setup and not hardhdlc. Im using sangoma 3.5.4.17. I think it
> also crush the system as I tried to put the crossover cable and the system
> constantly reboot. I will try 3.4.1 shortly and revert
> 

Yes it's an sangoma bug. Please capture the kernel crash if it is
possible. I wrote this bug to the Sangoma, but I had to revert
immedietly because I had a short maintenance window.

A
> Sam
> --
> Hi,
> the problem is:
> 
> hardhdlc=16
> replace to
> mtp2=16
> 
> hint: don't try wanpipe 3.4.4 this cause kernel panic with mtp2
> wanpipe-3.4.1 + dahdi 2.1.0.4 works fine here
> 
> Regards,
> Attila
> 
> On Mon, 2009-08-03 at 00:38 -0500, resea...@businesstz.com wrote:
> > Hi Alejendro
> > 
> > Have you managed to find out the solution to your problem!! Im facing the
> > same problem. The same link connects on another digium card running in
> > parallel with exactly the same configuration.
> > 
> > Please share with me if you have a clue as I want to try sangoma..i'm
> > using A104DE instead
> > 
> > Kind regards
> > Sam
> > 
> > 
> > --
> > 
> > Message: 1
> > Date: Thu, 4 Jun 2009 11:29:15 -0600
> > From: Alejandro Mej?a Evertsz 
> > Subject: [asterisk-ss7] Sangoma A102 + Dahdi + libss7 + Asterisk -
> > Linkset DOWN
> > To: 
> > Message-ID: <006501c9e539$fd815d20$f88417...@com>
> > Content-Type: text/plain; charset="iso-8859-1"
> > 
> > Hi,
> > 
> > 
> > 
> > I have problems, as linkset shows as DOWN all the time with the following
> > scenario:
> > 
> > 
> > 
> > -Dell server with Sangoma A102 (2 E1/T1 Card)
> > 
> > -Ubuntu Server 8.10
> > 
> > -Dahdi Linux 2.1.0.4
> > 
> > -libss7 1.0.2
> > 
> > -wanpipe 3.4.1
> > 
> > -Asterisk 1.6.1.0
> > 
> > 
> > 
> > ===/etc/dahdi/system.conf===
> > 
> > loadzone=us
> > 
> > defaultzone=us
> > 
> > 
> > 
> > #Sangoma A102 port 1 [slot:2 bus:10 span:1] 
> > 
> > span=1,0,0,ccs,hdb3
> > 
> > bchan=1-15,17-31
> > 
> > echocanceller=mg2,1-15,17-31
> > 
> > hardhdlc=16
> > 
> > 
> > 
> > #Sangoma A102 port 2 [slot:2 bus:10 span:2] 
> > 
> > span=2,1,0,ccs,hdb3
> > 
> > bchan=32-46,48-62
> > 
> > echocanceller=mg2,32-46,48-62
> > 
> > hardhdlc=47
> > 
> > ===END==
> > 
> > 
> > 
> > ===/etc/asterisk/chan_dahdi.conf===
> > 
> > [trunkgroups]
> > 
> > 
> > 
> > [channels]
> > 
> > context=default
> > 
> > usecallerid=yes
> > 
> > hidecallerid=no
> > 
> > callwaiting=yes
> > 
> > usecallingpres=yes
> > 
> > callwaitingcallerid=yes
> > 
> > threewaycalling=yes
> > 
> > transfer=yes
> > 
> > canpark=yes
> > 
> > cancallforward=yes
> > 
> > callreturn=yes
> > 
> > echocancel=yes
> > 
> > echocancelwhenbridged=yes
> > 
> > relaxdtmf=yes
> > 
> > rxgain=0.0
> > 
> > txgain=0.0
> > 
> > group=1
> > 
> > callgroup=1
> > 
> > pickupgroup=1
> > 
> > immediate=no
> > 
> > 
> > 
> > ;Sangoma A102 port 1 [slot:2 bus:10 span:1] 
> > 
> > context=from-pstn
> > 
> > group=0
> > 
> > echocancel=yes
> > 
> > ;switchtype=euroisdn
> > 
> > signalling=ss7
> > 
> > ss7type=itu
> > 
> > linkset = 1
> > 
> > pointcode = 1234
> > 
> > adjpointcode = 4321
> > 
> > defaultdpc = 4132
> > 
> > cicbeginswith = 1
> > 
> > networkindicator=national
> > 
> > sigchan = 16
> > 
> > ;sigchannel = 16
> > 
> > channel => 1-15,17-31
> > 
> > 
> > 
> > ;Sangoma A102 port 2 [slot:2 bus:10 span:2] 
> > 
> > context=from-pstn
> > 
> > group=1
> > 
> > echocancel=yes
> > 
> > ;switchtype=euroisdn
> > 
> > signalling=ss7
> > 
> > ss7type=itu
> > 
> > linkset = 2
> > 
> > pointcode = 4321
> > 
> > adjpointcode = 1234
> > 
> > defaultdpc = 4132
> > 
> > cicbeginswith = 32
> > 
> > networkindicator=national
> > 
> > sigchan = 47
> > 
> > ;sigchannel = 47
> > 
> > channel => 32-46,48-62
> > 
> > =END=
> > 
> > 
> > 
> > As I don?t have an SS7 equipment/link to test, I decided to make a
> crossover
> > cable (1<-->4, 2<-->5) and plugged port 1 to port 2 on the same card.
> > 
> > Everything seems to come up correctly, but when I do: ?ss7 show linkset 1?
> > it shows as down
> > 
> > 
> > 
> > *CLI> ss7 show linkset 1
> > 
> > SS7 linkset 1 status: Down
> > 
> > 
> > 
> > Even if links on the card, and dahdi status seems ok:
> > 
> > 
> > 
> > *CLI> dahdi show status
> > 
> > Description  Alarms  IRQbpviol CRC4   Fra
> > Codi Options  LBO
> > 
> > wanpipe1 card 0  OK  0  0  0  CCS
> > HDB3 YEL  0 db (CSU)/0-133 feet (DSX-1)
> > 
> > wanpipe2 card 1  OK  0  0  0  CCS
> > HDB3 YEL  0 db (CSU)/0-133 feet (DSX-1)
> > 
> > 
> > 
> > When using ss7linktest I get this:
> > 
> > 
> > 
> > # /usr/src/libss7-1.0.2/ss7linktest 16 itu 1234 4321
> > 
> > Starting link 1
> > 
> > Link state change: IDLE -> NOTALIGNED
> > 
> > Len = 4 [ ff ff 01 00 ]
> > 
> > FSN: 127 FIB 1
> > 
> > BSN: 127 BIB 1
> > 
> > >[0

Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 54, Issue 3

[RESEARCH]

Hi Attila

I had the mtp2 setup and not hardhdlc. Im using sangoma 3.5.4.17. I think it
also crush the system as I tried to put the crossover cable and the system
constantly reboot. I will try 3.4.1 shortly and revert

Sam
--
Hi,
the problem is:

hardhdlc=16
replace to
mtp2=16

hint: don't try wanpipe 3.4.4 this cause kernel panic with mtp2
wanpipe-3.4.1 + dahdi 2.1.0.4 works fine here

Regards,
Attila

On Mon, 2009-08-03 at 00:38 -0500, resea...@businesstz.com wrote:
> Hi Alejendro
> 
> Have you managed to find out the solution to your problem!! Im facing the
> same problem. The same link connects on another digium card running in
> parallel with exactly the same configuration.
> 
> Please share with me if you have a clue as I want to try sangoma..i'm
> using A104DE instead
> 
> Kind regards
> Sam
> 
> 
> --
> 
> Message: 1
> Date: Thu, 4 Jun 2009 11:29:15 -0600
> From: Alejandro Mej?a Evertsz 
> Subject: [asterisk-ss7] Sangoma A102 + Dahdi + libss7 + Asterisk -
>   Linkset DOWN
> To: 
> Message-ID: <006501c9e539$fd815d20$f88417...@com>
> Content-Type: text/plain; charset="iso-8859-1"
> 
> Hi,
> 
> 
> 
> I have problems, as linkset shows as DOWN all the time with the following
> scenario:
> 
> 
> 
> -Dell server with Sangoma A102 (2 E1/T1 Card)
> 
> -Ubuntu Server 8.10
> 
> -Dahdi Linux 2.1.0.4
> 
> -libss7 1.0.2
> 
> -wanpipe 3.4.1
> 
> -Asterisk 1.6.1.0
> 
> 
> 
> ===/etc/dahdi/system.conf===
> 
> loadzone=us
> 
> defaultzone=us
> 
> 
> 
> #Sangoma A102 port 1 [slot:2 bus:10 span:1] 
> 
> span=1,0,0,ccs,hdb3
> 
> bchan=1-15,17-31
> 
> echocanceller=mg2,1-15,17-31
> 
> hardhdlc=16
> 
> 
> 
> #Sangoma A102 port 2 [slot:2 bus:10 span:2] 
> 
> span=2,1,0,ccs,hdb3
> 
> bchan=32-46,48-62
> 
> echocanceller=mg2,32-46,48-62
> 
> hardhdlc=47
> 
> ===END==
> 
> 
> 
> ===/etc/asterisk/chan_dahdi.conf===
> 
> [trunkgroups]
> 
> 
> 
> [channels]
> 
> context=default
> 
> usecallerid=yes
> 
> hidecallerid=no
> 
> callwaiting=yes
> 
> usecallingpres=yes
> 
> callwaitingcallerid=yes
> 
> threewaycalling=yes
> 
> transfer=yes
> 
> canpark=yes
> 
> cancallforward=yes
> 
> callreturn=yes
> 
> echocancel=yes
> 
> echocancelwhenbridged=yes
> 
> relaxdtmf=yes
> 
> rxgain=0.0
> 
> txgain=0.0
> 
> group=1
> 
> callgroup=1
> 
> pickupgroup=1
> 
> immediate=no
> 
> 
> 
> ;Sangoma A102 port 1 [slot:2 bus:10 span:1] 
> 
> context=from-pstn
> 
> group=0
> 
> echocancel=yes
> 
> ;switchtype=euroisdn
> 
> signalling=ss7
> 
> ss7type=itu
> 
> linkset = 1
> 
> pointcode = 1234
> 
> adjpointcode = 4321
> 
> defaultdpc = 4132
> 
> cicbeginswith = 1
> 
> networkindicator=national
> 
> sigchan = 16
> 
> ;sigchannel = 16
> 
> channel => 1-15,17-31
> 
> 
> 
> ;Sangoma A102 port 2 [slot:2 bus:10 span:2] 
> 
> context=from-pstn
> 
> group=1
> 
> echocancel=yes
> 
> ;switchtype=euroisdn
> 
> signalling=ss7
> 
> ss7type=itu
> 
> linkset = 2
> 
> pointcode = 4321
> 
> adjpointcode = 1234
> 
> defaultdpc = 4132
> 
> cicbeginswith = 32
> 
> networkindicator=national
> 
> sigchan = 47
> 
> ;sigchannel = 47
> 
> channel => 32-46,48-62
> 
> =END=
> 
> 
> 
> As I don?t have an SS7 equipment/link to test, I decided to make a
crossover
> cable (1<-->4, 2<-->5) and plugged port 1 to port 2 on the same card.
> 
> Everything seems to come up correctly, but when I do: ?ss7 show linkset 1?
> it shows as down
> 
> 
> 
> *CLI> ss7 show linkset 1
> 
> SS7 linkset 1 status: Down
> 
> 
> 
> Even if links on the card, and dahdi status seems ok:
> 
> 
> 
> *CLI> dahdi show status
> 
> Description  Alarms  IRQbpviol CRC4   Fra
> Codi Options  LBO
> 
> wanpipe1 card 0  OK  0  0  0  CCS
> HDB3 YEL  0 db (CSU)/0-133 feet (DSX-1)
> 
> wanpipe2 card 1  OK  0  0  0  CCS
> HDB3 YEL  0 db (CSU)/0-133 feet (DSX-1)
> 
> 
> 
> When using ss7linktest I get this:
> 
> 
> 
> # /usr/src/libss7-1.0.2/ss7linktest 16 itu 1234 4321
> 
> Starting link 1
> 
> Link state change: IDLE -> NOTALIGNED
> 
> Len = 4 [ ff ff 01 00 ]
> 
> FSN: 127 FIB 1
> 
> BSN: 127 BIB 1
> 
> >[0] LSSU SIO
> 
> 
> 
> I have no experience at all using ss7 (I only have used ISDN PRI), so
please
> be patient if you find I?m doing something totally wrong hehehe.
> 
> 
> 
> Thanks in advance for your help!
> 
> 
> 
> Regards,
> 
> 
> 
> Alejandro Mej?a
> 
> 
> 
> 
> 
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-ss7
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[asterisk-ss7] Asterisk SS7 voice problem

[shoieb_arshad]

Hello,

I have configured asterisk with ss7 signaling between two asterisk  
servers using Sangoma A101 cards.
I have used the following versions.
asterisk- trunk
dahdi-trunk and dahdi tools
libss7 branch 1.0.

Now the problem is that signaling channel is working fine and the call  
setup is working properly, but once the call has been established no  
voice is detected on both ends, after that by sending DTMF from either  
end bearer channel start working properly.
Can you please tell me where the problem is?and how can i solve it?
->All  the echocancel are off (hardware and software both).
->echotraining is also set to "no" in chan_dahdi.conf.
->COT check is set to none in IAM message.

Muhammad Shoieb Arshad
Lecturer
Department of Electrical Engineering,
Comsats Institute of Information Technology,
Islamabad, Pakistan.
PH # 03006805270



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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 51, Issue 21

[aka]

>
> Hi


Thankx for reply.i am really new fro libss7.i am using asterisk 1.4.25 and
freepbx 2.4. currently using zaptel 1.4.is this pre request match for
install libss7.if there has any document please send me.

thanx
akalanka

>
>
> 
> From: aka 
> To: asterisk-ss7@lists.digium.com
> Sent: Tuesday, May 26, 2009 12:57:30 PM
> Subject: [asterisk-ss7] send call via define cic range
>
> Hi
>
> i am using chan_ss7 1.1 to connect ss7 & asterisk.in my configuration
> there has a two links(E1 s) to connect asterisk ss7.is there has any way
> to handle outsides calls in link wise.i try to configure asterisk this way
> but no luck for this.(SS7/link1/$OUTNUM)
>
>
>
>
> -- next part --
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> http://lists.digium.com/pipermail/asterisk-ss7/attachments/20090526/b8dc4b40/attachment-0001.htm
>
> --
>
> Message: 4
> Date: Tue, 26 May 2009 09:27:42 -0300
> From: Marcelo Pacheco 
> Subject: Re: [asterisk-ss7] send call via define cic range
> To: asterisk-ss7@lists.digium.com
> Message-ID: <4a1be03e.5060...@m2j.com.br>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Dahdi groups is the feature you want.
> It's well documented in chan_dahdi.conf
>
> Antoine Megalla wrote:
> > Hi,
> >
> > I did not find a way to do that in chan_ss7.
> > However you can use the hunting_policy in the linkset definition to
> > try achieve that in a way
> >
> > you can use
> > hunting_policy => seq_lth
> > to use the first E1 to send calls
> >
> > if you want to use the second E1 then use
> > hunting_policy => seq_htl
> >
> > This will only ensure that the first 30 outbound calls go to the E1 of
> > your choice.
> > Of course you can set a global counter that you increment as the calls
> > are coming in and decrement on hangup and then check to make sure that
> > only 30 calls are sent. This will ensure that only 1 E1 is used for
> > oubound.
> >
> > Exmaple to use the first E1:
> >
> > in ss7.conf use
> > hunting_policy => seq_lth
> > in the linkset definition
> >
> > exten => _X.,1,SetGlobalVar(CALLCOUNTER=${MATH(${CALLCOUNTER}+1,int)})
> > exten => _X.,2,GoToIf($[${CALLCOUNTER} >  30] ? 3:4)
> > exten => _X.,3,Dial(SS7/MSC1/00${EXTEN}|60)
> > exten => _X.,4,Hangup()
> >
> > exten => h,1,SetGlobalVar(CALLCOUNTER=${MATH(${CALLCOUNTER}-1,int)})
> >
> > Regrads,
> >
> > Antoine Megalla.
> >
> > 
> > *From:* aka 
> > *To:* asterisk-ss7@lists.digium.com
> > *Sent:* Tuesday, May 26, 2009 12:57:30 PM
> > *Subject:* [asterisk-ss7] send call via define cic range
> >
> > Hi
> >
> > i am using chan_ss7 1.1 to connect ss7 & asterisk.in
> >  my configuration there has a two links(E1 s) to
> > connect asterisk ss7.is  there has any way to handle
> > outsides calls in link wise.i try to configure asterisk this way but
> > no luck for this.(SS7/link1/$OUTNUM)
> >
> > 
> >
> > ___
> > --Bandwidth and Colocation Provided by http://www.api-digital.com--
> >
> > asterisk-ss7 mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-ss7
>
>
>
>
> --
>
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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 51, Issue 17

[Jorge F. Churio]

Redfone has their own download site for its tools, but requires having a
FoneBRIDGE device, see http://support.red-fone.com/downloads


The SS7 stack is in dicea´s site, there are an open version with some ISUP
features partially implemented and a full-featured one named "premium"


On Thu, May 21, 2009 at 2:00 PM, wrote:

> Send asterisk-ss7 mailing list submissions to
>asterisk-ss7@lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
>http://lists.digium.com/mailman/listinfo/asterisk-ss7
> or, via email, send a message with subject or body 'help' to
>asterisk-ss7-requ...@lists.digium.com
>
> You can reach the person managing the list at
>asterisk-ss7-ow...@lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-ss7 digest..."
>
>
> Today's Topics:
>
>   1. Re: Improved SS7 for Asterisk-Fonebridge (Kasun Daminda)
>   2. Re: Improved SS7 for Asterisk-Fonebridge (Sam Deller - Airnet NZ)
>
>
> --
>
> Message: 1
> Date: Thu, 21 May 2009 12:06:54 +0530
> From: Kasun Daminda 
> Subject: Re: [asterisk-ss7] Improved SS7 for Asterisk-Fonebridge
> To: asterisk-ss7@lists.digium.com
> Message-ID:
>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Dear ,,
>
> How can I download this ?
>
> Thanks for ur early response.
>
> Rgds
>
> Daminda
>
> On Tue, May 19, 2009 at 10:53 PM, Jorge F. Churio  >wrote:
>
> > Redfone and Dicea had announced yesterday they released a new and
> improved
> > chan_SS7 stack, suited to work seamlessly with redfone devices. Among
> > other benefits, it overcomes one of the biggest problems that is the
> > possibility to have multiple linksets spread among multiple boxes and
> voce
> > versa: create clusters of Asterisk sharing only one linkset in one box.
> >
> >
> > ___
> > --Bandwidth and Colocation Provided by http://www.api-digital.com--
> >
> > asterisk-ss7 mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-ss7
> >
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>
> Message: 2
> Date: Thu, 21 May 2009 20:04:39 +1200
> From: "Sam Deller - Airnet NZ" 
> Subject: Re: [asterisk-ss7] Improved SS7 for Asterisk-Fonebridge
> To: 
> Message-ID:
>
>  <069df3f895842f4ab31e2fa2ea30ee5da8a...@airnetsbs.airnetsbs.airnet.local>
>
> Content-Type: text/plain; charset="us-ascii"
>
> www.justfuckinggoogleit.com
>
> 
>
> From: asterisk-ss7-boun...@lists.digium.com
> [mailto:asterisk-ss7-boun...@lists.digium.com] On Behalf Of Kasun
> Daminda
> Sent: Thursday, 21 May 2009 6:37 p.m.
> To: asterisk-ss7@lists.digium.com
> Subject: Re: [asterisk-ss7] Improved SS7 for Asterisk-Fonebridge
>
>
> Dear ,,
>
> How can I download this ?
>
> Thanks for ur early response.
>
> Rgds
>
> Daminda
>
>
> On Tue, May 19, 2009 at 10:53 PM, Jorge F. Churio
>  wrote:
>
>
>Redfone and Dicea had announced yesterday they released a new
> and improved chan_SS7 stack, suited to work seamlessly with redfone
> devices.
>Among other benefits, it overcomes one of the biggest problems
> that is the possibility to have multiple linksets spread among multiple
> boxes and voce versa: create clusters of Asterisk sharing only one
> linkset in one box.
>
>
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>
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Re: [asterisk-ss7] Asterisk SS7 configuration Problem asterisk-1.6.0.9

[Gobinda Paul]

Thanks Vashkar, For Your Reply , That Problem Is Already Being Solved.

_
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Re: [asterisk-ss7] Asterisk SS7 configuration Problem asterisk-1.6.0.9

[Vashkar]

Hi!,

Are you connecting to any Ericsson MSC/GMSC? check for subservice.There is
an issue with Ericsson switches. And also check that the SS7 is set to ITU-T
(WHITE for BD)

the error number indicates ISDN error 34, no circuit/channel available

Check that the operator has CRC4 enabled or not. normally it sould be
NO-CRC4 in most of the operators in BD

The reason the asterisk can not create channel DAHDI is the SS7 BChans are
not in idle state and this also means that the SS7 link MTP3 is not
operational. try block and unblock on the linkset channels. turn on the ISUP
debug. Hope you get the reason.

Also check if your operator accesspets the leading '880' to be dialed or its
without them. May be '01720039748' you can try.

If they want to strip digits, set the extentions.conf to

exten => _X.,3,Dial(DAHDI/g1/${EXTEN}:2)

.

-Vashkar

On Thu, May 14, 2009 at 6:28 AM, Gobinda Paul  wrote:

> Hi,
>
> Any Idia?
>
> *if i dial any number , i got 34 error. can you please tell what's the
> wrong done by me?
> *
> -- Executing [8801720039...@default:1]
> Answer("SIP/XXX.XXX.XXX.XXX-116b9aa0", "") in new stack
> -- Executing [8801720039...@default:2]
> Set("SIP/XXX.XXX.XXX.XXX-116b9aa0", "CALLERID(all)=14159928422") in new
> stack
> -- Executing [8801720039...@default:3]
> Dial("SIP/XXX.XXX.XXX.XXX-116b9aa0", "DAHDI/g1/920172XX") in new stack
> [Apr 19 02:21:19] WARNING[11603]: app_dial.c:1468 dial_exec_full: Unable to
> create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
>   == Everyone is busy/congested at this time (1:0/1/0)
> -- Executing [8801720039...@default:4]
> Hangup("SIP/XXX.XXX.XXX.XXX-116b9aa0", "") in new stack
>
> Thanks in Advance .
>
>
> Gobinda Paul
>
>
>
> --
> From: gobi...@live.com
> To: asterisk-ss7@lists.digium.com
> Subject: RE: Asterisk SS7 configuration Problem asterisk-1.6.0.9
> Date: Sun, 19 Apr 2009 03:19:15 +1100
>
> hi,
>
>
> Yes, i am sill facing the problem.
>
>
> Thanks,
> Gobinda
>
>
>
>
> --
> From: gobi...@live.com
> To: asterisk-ss7@lists.digium.com
> Subject: Re: Asterisk SS7 configuration Problem asterisk-1.6.0.9
> Date: Sun, 19 Apr 2009 01:29:33 +1100
>
> Hi mesbah,
>
> Thanks for your reply, i have made some changes ,
> -
> [r...@localhost ~]# cat /etc/dahdi/system.conf
> span=1,1,0,ccs,hdb3,crc4
> bchan=2-31
> mtp2=1
> echocanceller=mg2,2-31
> span=2,2,0,ccs,hdb3,crc4
> bchan=32-62
> echocanceller=mg2,32-62
> span=3,3,0,ccs,hdb3,crc4
> bchan=63-93
> echocanceller=mg2,63-93
> loadzone= us
> defaultzone = us
> -
> [r...@localhost ~]# cat /etc/asterisk/chan_dahdi.conf
>  [trunkgroups]
> [channels]
> language=en
> context=ss7
> switchtype=euroisdn
> signalling=ss7
> toneduration=100
> usecallerid=yes
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> group=1
> callgroup=1
> pickupgroup=1
> ss7type = itu
> ss7_called_nai=dynamic
> ss7_calling_nai=dynamic
> linkset = 1
> pointcode = 3924
> adjpointcode = 9322
> defaultdpc = 9322
> networkindicator=international
> cicbeginswith = 2
> mtp2=1
> sigchan = 1
> channel = 2-31
> cicbeginswith = 32
> channel =32-62
> cicbeginswith = 63
> channel =63-93
> 
> [r...@localhost ~]# cat /etc/asterisk/dahdi-channels.conf
> group=1
> context=ss7
> switchtype = euroisdn
> signalling = ss7
> channel => 2-31
> channel => 32-62
> channel => 63-93
> ---
> [r...@localhost ~]# cat /etc/asterisk//extensions.conf
>
> [default]
> exten => s,1,Answer()
> exten => s,2,Playback(hello-world)
> exten => s,3,hangup()
> include => ss7
> [ss7]
> exten => s,1,Answer()
> exten => s,2,hangup()
> exten => _X.,1,Answer()
> exten => _X.,2,Set(CALLERID(all)=14159928422)
> exten => _X.,3,Dial(DAHDI/g1/${EXTEN})
> exten => _X.,4,hangup()
>
> -
> **CLI> dahdi restart*
> *-*
> **
>   == Starting SS7 linkset on span 1
> Huh?! Got FISU in link state 1
> --- SS7 Down ---
> MTP2 link up (SLC 0)
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnack

Re: [asterisk-ss7] Asterisk SS7 configuration Problem asterisk-1.6.0.9

[bipin singh]

Hi
  ur ss7 links not working properly .reset  ur link

On Thu, May 14, 2009 at 5:58 AM, Gobinda Paul  wrote:

>  Hi,
>
> Any Idia?
>
> *if i dial any number , i got 34 error. can you please tell what's the
> wrong done by me?
> *
> -- Executing [8801720039...@default:1]
> Answer("SIP/XXX.XXX.XXX.XXX-116b9aa0", "") in new stack
> -- Executing [8801720039...@default:2]
> Set("SIP/XXX.XXX.XXX.XXX-116b9aa0", "CALLERID(all)=14159928422") in new
> stack
> -- Executing [8801720039...@default:3]
> Dial("SIP/XXX.XXX.XXX.XXX-116b9aa0", "DAHDI/g1/920172XX") in new stack
> [Apr 19 02:21:19] WARNING[11603]: app_dial.c:1468 dial_exec_full: Unable to
> create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
>   == Everyone is busy/congested at this time (1:0/1/0)
> -- Executing [8801720039...@default:4]
> Hangup("SIP/XXX.XXX.XXX.XXX-116b9aa0", "") in new stack
>
> Thanks in Advance .
>
>
> Gobinda Paul
>
>
>
> --
> From: gobi...@live.com
> To: asterisk-ss7@lists.digium.com
> Subject: RE: Asterisk SS7 configuration Problem asterisk-1.6.0.9
> Date: Sun, 19 Apr 2009 03:19:15 +1100
>
> hi,
>
>
> Yes, i am sill facing the problem.
>
>
> Thanks,
> Gobinda
>
>
>
>
> --
> From: gobi...@live.com
> To: asterisk-ss7@lists.digium.com
> Subject: Re: Asterisk SS7 configuration Problem asterisk-1.6.0.9
> Date: Sun, 19 Apr 2009 01:29:33 +1100
>
> Hi mesbah,
>
> Thanks for your reply, i have made some changes ,
> -
> [r...@localhost ~]# cat /etc/dahdi/system.conf
> span=1,1,0,ccs,hdb3,crc4
> bchan=2-31
> mtp2=1
> echocanceller=mg2,2-31
> span=2,2,0,ccs,hdb3,crc4
> bchan=32-62
> echocanceller=mg2,32-62
> span=3,3,0,ccs,hdb3,crc4
> bchan=63-93
> echocanceller=mg2,63-93
> loadzone= us
> defaultzone = us
> -
> [r...@localhost ~]# cat /etc/asterisk/chan_dahdi.conf
> [trunkgroups]
> [channels]
> language=en
> context=ss7
> switchtype=euroisdn
> signalling=ss7
> toneduration=100
> usecallerid=yes
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> group=1
> callgroup=1
> pickupgroup=1
> ss7type = itu
> ss7_called_nai=dynamic
> ss7_calling_nai=dynamic
> linkset = 1
> pointcode = 3924
> adjpointcode = 9322
> defaultdpc = 9322
> networkindicator=international
> cicbeginswith = 2
> mtp2=1
> sigchan = 1
> channel = 2-31
> cicbeginswith = 32
> channel =32-62
> cicbeginswith = 63
> channel =63-93
> 
> [r...@localhost ~]# cat /etc/asterisk/dahdi-channels.conf
> group=1
> context=ss7
> switchtype = euroisdn
> signalling = ss7
> channel => 2-31
> channel => 32-62
> channel => 63-93
> ---
> [r...@localhost ~]# cat /etc/asterisk//extensions.conf
>
> [default]
> exten => s,1,Answer()
> exten => s,2,Playback(hello-world)
> exten => s,3,hangup()
> include => ss7
> [ss7]
> exten => s,1,Answer()
> exten => s,2,hangup()
> exten => _X.,1,Answer()
> exten => _X.,2,Set(CALLERID(all)=14159928422)
> exten => _X.,3,Dial(DAHDI/g1/${EXTEN})
> exten => _X.,4,hangup()
>
> -
> **CLI> dahdi restart*
> *-*
> **
>   == Starting SS7 linkset on span 1
> Huh?! Got FISU in link state 1
> --- SS7 Down ---
> MTP2 link up (SLC 0)
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> [Apr 19 02:26:28] WARNING[11777]: chan_dahdi.c:9836 ss7_linkset: MTP2 link
> down (SLC 0)
> MTP2 link up (SLC 0)
> --- SS7 Up ---
> Resetting CICs 2 to 33
> Resetting CICs 34 to 65
> Resetting CICs 66 to 93
> [Apr 19 02:26:30] WARNING[11777]: chan_dahdi.c:9836 ss7_linkset: MTP2 link
> down (SLC 0)
> MTP2 link up (SLC 0)
> Received out of sequence MSU w/ fsn of 2, lastfsnacked = 0, requesting
> retransmission
> MSU received, though still waiting for retransmission 

Re: [asterisk-ss7] Asterisk SS7 configuration Problem asterisk-1.6.0.9

[Gobinda Paul]

Hi, 

 

Any Idia?

if i dial any number , i got 34 error. can you please tell what's the wrong 
done by me?
 
-- Executing [8801720039...@default:1] 
Answer("SIP/XXX.XXX.XXX.XXX-116b9aa0", "") in new stack
-- Executing [8801720039...@default:2] Set("SIP/XXX.XXX.XXX.XXX-116b9aa0", 
"CALLERID(all)=14159928422") in new stack
-- Executing [8801720039...@default:3] Dial("SIP/XXX.XXX.XXX.XXX-116b9aa0", 
"DAHDI/g1/920172XX") in new stack
[Apr 19 02:21:19] WARNING[11603]: app_dial.c:1468 dial_exec_full: Unable to 
create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
  == Everyone is busy/congested at this time (1:0/1/0)
-- Executing [8801720039...@default:4] 
Hangup("SIP/XXX.XXX.XXX.XXX-116b9aa0", "") in new stack

Thanks in Advance .

 

 

Gobinda Paul


 


From: gobi...@live.com
To: asterisk-ss7@lists.digium.com
Subject: RE: Asterisk SS7 configuration Problem asterisk-1.6.0.9
Date: Sun, 19 Apr 2009 03:19:15 +1100



hi,
 
 
Yes, i am sill facing the problem.
 
 
Thanks,
Gobinda



 


From: gobi...@live.com
To: asterisk-ss7@lists.digium.com
Subject: Re: Asterisk SS7 configuration Problem asterisk-1.6.0.9
Date: Sun, 19 Apr 2009 01:29:33 +1100



Hi mesbah,
 
Thanks for your reply, i have made some changes ,
-
[r...@localhost ~]# cat /etc/dahdi/system.conf
span=1,1,0,ccs,hdb3,crc4
bchan=2-31
mtp2=1
echocanceller=mg2,2-31
span=2,2,0,ccs,hdb3,crc4
bchan=32-62
echocanceller=mg2,32-62
span=3,3,0,ccs,hdb3,crc4
bchan=63-93
echocanceller=mg2,63-93
loadzone= us
defaultzone = us
-
[r...@localhost ~]# cat /etc/asterisk/chan_dahdi.conf 
[trunkgroups]
[channels]
language=en
context=ss7
switchtype=euroisdn
signalling=ss7
toneduration=100
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
group=1
callgroup=1
pickupgroup=1
ss7type = itu
ss7_called_nai=dynamic
ss7_calling_nai=dynamic
linkset = 1
pointcode = 3924
adjpointcode = 9322
defaultdpc = 9322
networkindicator=international
cicbeginswith = 2
mtp2=1
sigchan = 1
channel = 2-31
cicbeginswith = 32
channel =32-62
cicbeginswith = 63
channel =63-93

[r...@localhost ~]# cat /etc/asterisk/dahdi-channels.conf
group=1
context=ss7
switchtype = euroisdn
signalling = ss7
channel => 2-31
channel => 32-62
channel => 63-93
---
[r...@localhost ~]# cat /etc/asterisk//extensions.conf
 
[default]
exten => s,1,Answer()
exten => s,2,Playback(hello-world)
exten => s,3,hangup()
include => ss7
[ss7]
exten => s,1,Answer()
exten => s,2,hangup()
exten => _X.,1,Answer()
exten => _X.,2,Set(CALLERID(all)=14159928422)
exten => _X.,3,Dial(DAHDI/g1/${EXTEN})
exten => _X.,4,hangup()

-
*CLI> dahdi restart
-

  == Starting SS7 linkset on span 1
Huh?! Got FISU in link state 1
--- SS7 Down ---
MTP2 link up (SLC 0)
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
[Apr 19 02:26:28] WARNING[11777]: chan_dahdi.c:9836 ss7_linkset: MTP2 link down 
(SLC 0)
MTP2 link up (SLC 0)
--- SS7 Up ---
Resetting CICs 2 to 33
Resetting CICs 34 to 65
Resetting CICs 66 to 93
[Apr 19 02:26:30] WARNING[11777]: chan_dahdi.c:9836 ss7_linkset: MTP2 link down 
(SLC 0)
MTP2 link up (SLC 0)
Received out of sequence MSU w/ fsn of 2, lastfsnacked = 0, requesting 
retransmission
MSU received, though still waiting for retransmission start.  Dropping.
Received out of sequence MSU w/ fsn of 2, lastfsnacked = 0, requesting 
retransmission
MSU received, though still waiting for retransmission start.  Dropping.
Received out of sequence MSU w/ fsn of 3, lastfsnacked = 0, requesting 
retransmission
Received out of sequence MSU w/ fsn of 3, lastfsnacked = 1, requesting 
retransmission
-
 
if i dial any number , i got 34 error. can you please tell w

Re: [asterisk-ss7] Asterisk SS7 configuration Problem asterisk-1.6.0.9

[bipin singh]

hardware irq missing try another server
or use lspci -bv

On 4/18/09, Gobinda Paul  wrote:
>
>  Hi,
>
> I am facing some problem to configure ss7 with asterisk.Here is my
> configuration bellow.
>
> I am using digium 410p card, According to my telco,
> Signal Chanel: 1st Channel Of First E1,
> My pointcode: 3924
> Telco pointcode: 9322
>
>
> --
> [r...@localhost project]# dahdi_hardware
> pci::06:01.0 wct4xxp+ d161:0410 Wildcard TE410P (4th Gen)
>
> --
>
> [r...@localhost project]# cat /etc/dahdi/system.conf
> span=1,1,0,ccs,hdb3,crc4
> bchan=2-31
> mtp2=1
> echocanceller=mg2,2-31
> span=2,2,0,ccs,hdb3,crc4
> bchan=32-62
> echocanceller=mg2,32-62
> span=3,3,0,ccs,hdb3,crc4
> bchan=63-93
> echocanceller=mg2,63-93
> span=4,4,0,ccs,hdb3,crc4
> bchan=94-124
> echocanceller=mg2,94-124
> loadzone= us
> defaultzone = us
>
>
> --
> [r...@localhost project]# cat /etc/asterisk/chan_dahdi.conf
> [trunkgroups]
> [channels]
> language=en
> context=ss7
> switchtype=euroisdn
> signalling=ss7
> toneduration=100
> usecallerid=yes
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> group=1
> callgroup=1
> pickupgroup=1
> ss7type = itu
> ss7_called_nai=dynamic
> ss7_calling_nai=dynamic
> linkset = 1
> pointcode = 3924
> adjpointcode = 9322
> defaultdpc = 9322
> networkindicator=international
> cicbeginswith = 2
> mtp2=1
> sigchan = 1
> channel = 2-31
> cicbeginswith = 32
> channel =33-62
> cicbeginswith = 63
> channel =63-93
> cicbeginswith = 94
> channel = 94-124
>
> --
>
> [r...@localhost project]# cat /etc/asterisk/dahdi-channels.conf
> group=0,11
> context=from-pstn
> switchtype = euroisdn
> signalling = ss7
> channel => 2-31
> context = ss7
> group = 63
> group=0,12
> context=from-pstn
> switchtype = euroisdn
> signalling = ss7
> channel => 32-62
> context = ss7
> group = 63
> group=0,13
> context=from-pstn
> switchtype = euroisdn
> signalling = ss7
> channel => 63-93
> context = ss7
> group = 63
> group=0,14
> context=from-pstn
> switchtype = euroisdn
> signalling = ss7
> channel => 94-124
> context = ss7
> group = 63
>
>
> --
> [r...@localhost project]# cat /etc/asterisk/extensions.conf
> [general]
> static=yes
> writeprotect=no
> [globals]
> [default]
> exten => s,1,Answer()
> exten => s,2,Playback(hello-world)
> exten => s,3,hangup()
> include =>ss7
> [ss7]
> exten => s,1,Answer()
> exten => s,2,hangup()
> exten => _X.,1,Answer()
> exten => _X.,2,Dial(DAHDI/r1/${EXTEN})
> exten => _X.,3,hangup()
>
>
> 
>
>
> *CLI> ss7 show linkset 1
> SS7 linkset 1 status: Up
>
> *CLI> dahdi show status
> Description  Alarms  IRQbpviol CRC4   Fra
> Codi Options  LBO
> T4XXP (PCI) Card 0 Span 1REC 0  0  0  CCS
> HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) [ 1st Channel Used For Signal ]
> T4XXP (PCI) Card 0 Span 2YEL/REC 0  0  0  CCS
> HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)
> T4XXP (PCI) Card 0 Span 3REC 0  0  0  CCS
> HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)
> T4XXP (PCI) Card 0 Span 4RED 0  0  0  CCS
> HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) [ Red Because, I used 3 cable ]
>
>
> *CLI> dahdi restart
> -
> -
> -- Reconfigured channel 124, SS7 signalling
>   == Starting SS7 linkset on span 1
> Huh?! Got FISU in link state 1
> --- SS7 Down ---
> MTP2 link up (SLC 0)
> [Apr 18 19:02:09] WARNING[7672]: chan_dahdi.c:9836 ss7_linkset: MTP2 link
> down (SLC 0)
> Received out of sequence FISU w/ fsn of 0, lastfsnacked = 127, requesting
> retransmission
> MTP2 link up (SLC 0)
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> --- SS7 Up ---
> Resetting CICs 2 to 33
> Resetting CICs 34 to 61
> Resetting CICs 63 to 94
> Resetting CICs 95 to 124
>
> *CLI> dahdi show channel 2
> Channel: 2
> File Descriptor: 16
> Span: 1
> Extension:
> Dialing: no
> Context: ss7
> Caller ID:
> Calling TON: 0
> Caller ID name:
> Mailbox: none
> Destroy: 0
> InAlarm: 1
> Signalling Type: SS7
> Radio: 0
> Owner: 
> Real: 
> Callwait: 
> Threeway: 
> Confno: -1
> Propagated Conference: -1
> Real in conference: 0
> DSP: no
> Busy Detection: no
> TDD: no
> Relax DTMF: no
> Dialing/CallwaitCAS: 0/0
> Default law: alaw
> Fax Handled: no
> Pulse phone: no
> DND: no
> Echo Cancellation:
> 128 taps
> currently OFF

Re: [asterisk-ss7] Asterisk SS7 configuration Problem asterisk-1.6.0.9

[Gobinda Paul]

hi,

 

 

Yes, i am sill facing the problem.

 

 

Thanks,

Gobinda



 


From: gobi...@live.com
To: asterisk-ss7@lists.digium.com
Subject: Re: Asterisk SS7 configuration Problem asterisk-1.6.0.9
Date: Sun, 19 Apr 2009 01:29:33 +1100



Hi mesbah,
 
Thanks for your reply, i have made some changes ,
-
[r...@localhost ~]# cat /etc/dahdi/system.conf
span=1,1,0,ccs,hdb3,crc4
bchan=2-31
mtp2=1
echocanceller=mg2,2-31
span=2,2,0,ccs,hdb3,crc4
bchan=32-62
echocanceller=mg2,32-62
span=3,3,0,ccs,hdb3,crc4
bchan=63-93
echocanceller=mg2,63-93
loadzone= us
defaultzone = us
-
[r...@localhost ~]# cat /etc/asterisk/chan_dahdi.conf 
[trunkgroups]
[channels]
language=en
context=ss7
switchtype=euroisdn
signalling=ss7
toneduration=100
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
group=1
callgroup=1
pickupgroup=1
ss7type = itu
ss7_called_nai=dynamic
ss7_calling_nai=dynamic
linkset = 1
pointcode = 3924
adjpointcode = 9322
defaultdpc = 9322
networkindicator=international
cicbeginswith = 2
mtp2=1
sigchan = 1
channel = 2-31
cicbeginswith = 32
channel =32-62
cicbeginswith = 63
channel =63-93

[r...@localhost ~]# cat /etc/asterisk/dahdi-channels.conf
group=1
context=ss7
switchtype = euroisdn
signalling = ss7
channel => 2-31
channel => 32-62
channel => 63-93
---
[r...@localhost ~]# cat /etc/asterisk//extensions.conf
 
[default]
exten => s,1,Answer()
exten => s,2,Playback(hello-world)
exten => s,3,hangup()
include => ss7
[ss7]
exten => s,1,Answer()
exten => s,2,hangup()
exten => _X.,1,Answer()
exten => _X.,2,Set(CALLERID(all)=14159928422)
exten => _X.,3,Dial(DAHDI/g1/${EXTEN})
exten => _X.,4,hangup()

-
*CLI> dahdi restart
-

  == Starting SS7 linkset on span 1
Huh?! Got FISU in link state 1
--- SS7 Down ---
MTP2 link up (SLC 0)
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
[Apr 19 02:26:28] WARNING[11777]: chan_dahdi.c:9836 ss7_linkset: MTP2 link down 
(SLC 0)
MTP2 link up (SLC 0)
--- SS7 Up ---
Resetting CICs 2 to 33
Resetting CICs 34 to 65
Resetting CICs 66 to 93
[Apr 19 02:26:30] WARNING[11777]: chan_dahdi.c:9836 ss7_linkset: MTP2 link down 
(SLC 0)
MTP2 link up (SLC 0)
Received out of sequence MSU w/ fsn of 2, lastfsnacked = 0, requesting 
retransmission
MSU received, though still waiting for retransmission start.  Dropping.
Received out of sequence MSU w/ fsn of 2, lastfsnacked = 0, requesting 
retransmission
MSU received, though still waiting for retransmission start.  Dropping.
Received out of sequence MSU w/ fsn of 3, lastfsnacked = 0, requesting 
retransmission
Received out of sequence MSU w/ fsn of 3, lastfsnacked = 1, requesting 
retransmission
-
 
if i dial any number , i got 34 error. can you please tell what's the wrong 
done by me?
 
-- Executing [8801720039...@default:1] 
Answer("SIP/XXX.XXX.XXX.XXX-116b9aa0", "") in new stack
-- Executing [8801720039...@default:2] Set("SIP/XXX.XXX.XXX.XXX-116b9aa0", 
"CALLERID(all)=14159928422") in new stack
-- Executing [8801720039...@default:3] Dial("SIP/XXX.XXX.XXX.XXX-116b9aa0", 
"DAHDI/g1/920172XX") in new stack
[Apr 19 02:21:19] WARNING[11603]: app_dial.c:1468 dial_exec_full: Unable to 
create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
  == Everyone is busy/congested at this time (1:0/1/0)
-- Executing [8801720039...@default:4] 
Hangup("SIP/XXX.XXX.XXX.XXX-116b9aa0", "") in new stack

 
--
 
Thanks ,
Gobinda

 



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Re: [asterisk-ss7] Asterisk SS7 configuration Problem asterisk-1.6.0.9

[Mesbahuddin Malik]

Dear Gobinda,


Still have the problem. I am little bit busy  now .  Tomorrow I may check
with you .

BR
Mesbah

On Sat, Apr 18, 2009 at 8:29 PM, Gobinda Paul  wrote:

> Hi mesbah,
>
> Thanks for your reply, i have made some changes ,
> -
> [r...@localhost ~]# cat /etc/dahdi/system.conf
> span=1,1,0,ccs,hdb3,crc4
> bchan=2-31
> mtp2=1
> echocanceller=mg2,2-31
> span=2,2,0,ccs,hdb3,crc4
> bchan=32-62
> echocanceller=mg2,32-62
> span=3,3,0,ccs,hdb3,crc4
> bchan=63-93
> echocanceller=mg2,63-93
> loadzone= us
> defaultzone = us
> -
> [r...@localhost ~]# cat /etc/asterisk/chan_dahdi.conf
>  [trunkgroups]
> [channels]
> language=en
> context=ss7
> switchtype=euroisdn
> signalling=ss7
> toneduration=100
> usecallerid=yes
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> group=1
> callgroup=1
> pickupgroup=1
> ss7type = itu
> ss7_called_nai=dynamic
> ss7_calling_nai=dynamic
> linkset = 1
> pointcode = 3924
> adjpointcode = 9322
> defaultdpc = 9322
> networkindicator=international
> cicbeginswith = 2
> mtp2=1
> sigchan = 1
> channel = 2-31
> cicbeginswith = 32
> channel =32-62
> cicbeginswith = 63
> channel =63-93
> 
> [r...@localhost ~]# cat /etc/asterisk/dahdi-channels.conf
> group=1
> context=ss7
> switchtype = euroisdn
> signalling = ss7
> channel => 2-31
> channel => 32-62
> channel => 63-93
> ---
> [r...@localhost ~]# cat /etc/asterisk//extensions.conf
>
> [default]
> exten => s,1,Answer()
> exten => s,2,Playback(hello-world)
> exten => s,3,hangup()
> include => ss7
> [ss7]
> exten => s,1,Answer()
> exten => s,2,hangup()
> exten => _X.,1,Answer()
> exten => _X.,2,Set(CALLERID(all)=14159928422)
> exten => _X.,3,Dial(DAHDI/g1/${EXTEN})
> exten => _X.,4,hangup()
>
> -
> **CLI> dahdi restart*
> *-*
> **
>   == Starting SS7 linkset on span 1
> Huh?! Got FISU in link state 1
> --- SS7 Down ---
> MTP2 link up (SLC 0)
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> [Apr 19 02:26:28] WARNING[11777]: chan_dahdi.c:9836 ss7_linkset: MTP2 link
> down (SLC 0)
> MTP2 link up (SLC 0)
> --- SS7 Up ---
> Resetting CICs 2 to 33
> Resetting CICs 34 to 65
> Resetting CICs 66 to 93
> [Apr 19 02:26:30] WARNING[11777]: chan_dahdi.c:9836 ss7_linkset: MTP2 link
> down (SLC 0)
> MTP2 link up (SLC 0)
> Received out of sequence MSU w/ fsn of 2, lastfsnacked = 0, requesting
> retransmission
> MSU received, though still waiting for retransmission start.  Dropping.
> Received out of sequence MSU w/ fsn of 2, lastfsnacked = 0, requesting
> retransmission
> MSU received, though still waiting for retransmission start.  Dropping.
> Received out of sequence MSU w/ fsn of 3, lastfsnacked = 0, requesting
> retransmission
> Received out of sequence MSU w/ fsn of 3, lastfsnacked = 1, requesting
> retransmission
> -
>
> *if i dial any number , i got 34 error. can you please tell what's the
> wrong done by me?*
>
> -- Executing [8801720039...@default:1]
> Answer("SIP/XXX.XXX.XXX.XXX-116b9aa0", "") in new stack
> -- Executing [8801720039...@default:2]
> Set("SIP/XXX.XXX.XXX.XXX-116b9aa0", "CALLERID(all)=14159928422") in new
> stack
> -- Executing [8801720039...@default:3]
> Dial("SIP/XXX.XXX.XXX.XXX-116b9aa0", "DAHDI/g1/920172XX") in new stack
> [Apr 19 02:21:19] WARNING[11603]: app_dial.c:1468 dial_exec_full: Unable to
> create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
>   == Everyone is busy/congested at this time (1:0/1/0)
> -- Executing [8801720039...@default:4]
> Hangup("SIP/XXX.XXX.XXX.XXX-116b9aa0", "") in new stack
>
>
> --
>
> Thanks ,
> Gobinda
>
>
>
> 

Re: [asterisk-ss7] Asterisk SS7 configuration Problem asterisk-1.6.0.9

[Gobinda Paul]

Hi mesbah,

 

Thanks for your reply, i have made some changes ,

-

[r...@localhost ~]# cat /etc/dahdi/system.conf
span=1,1,0,ccs,hdb3,crc4
bchan=2-31
mtp2=1
echocanceller=mg2,2-31
span=2,2,0,ccs,hdb3,crc4
bchan=32-62
echocanceller=mg2,32-62
span=3,3,0,ccs,hdb3,crc4
bchan=63-93
echocanceller=mg2,63-93
loadzone= us
defaultzone = us

-

[r...@localhost ~]# cat /etc/asterisk/chan_dahdi.conf 
[trunkgroups]
[channels]
language=en
context=ss7
switchtype=euroisdn
signalling=ss7
toneduration=100
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
group=1
callgroup=1
pickupgroup=1
ss7type = itu
ss7_called_nai=dynamic
ss7_calling_nai=dynamic
linkset = 1
pointcode = 3924
adjpointcode = 9322
defaultdpc = 9322
networkindicator=international
cicbeginswith = 2
mtp2=1
sigchan = 1
channel = 2-31
cicbeginswith = 32
channel =32-62
cicbeginswith = 63
channel =63-93


[r...@localhost ~]# cat /etc/asterisk/dahdi-channels.conf
group=1
context=ss7
switchtype = euroisdn
signalling = ss7
channel => 2-31
channel => 32-62
channel => 63-93

---

[r...@localhost ~]# cat /etc/asterisk//extensions.conf

 

[default]
exten => s,1,Answer()
exten => s,2,Playback(hello-world)
exten => s,3,hangup()
include => ss7

[ss7]
exten => s,1,Answer()
exten => s,2,hangup()

exten => _X.,1,Answer()
exten => _X.,2,Set(CALLERID(all)=14159928422)
exten => _X.,3,Dial(DAHDI/g1/${EXTEN})
exten => _X.,4,hangup()


-

*CLI> dahdi restart

-



  == Starting SS7 linkset on span 1
Huh?! Got FISU in link state 1
--- SS7 Down ---
MTP2 link up (SLC 0)
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
[Apr 19 02:26:28] WARNING[11777]: chan_dahdi.c:9836 ss7_linkset: MTP2 link down 
(SLC 0)
MTP2 link up (SLC 0)
--- SS7 Up ---
Resetting CICs 2 to 33
Resetting CICs 34 to 65
Resetting CICs 66 to 93
[Apr 19 02:26:30] WARNING[11777]: chan_dahdi.c:9836 ss7_linkset: MTP2 link down 
(SLC 0)
MTP2 link up (SLC 0)
Received out of sequence MSU w/ fsn of 2, lastfsnacked = 0, requesting 
retransmission
MSU received, though still waiting for retransmission start.  Dropping.
Received out of sequence MSU w/ fsn of 2, lastfsnacked = 0, requesting 
retransmission
MSU received, though still waiting for retransmission start.  Dropping.
Received out of sequence MSU w/ fsn of 3, lastfsnacked = 0, requesting 
retransmission
Received out of sequence MSU w/ fsn of 3, lastfsnacked = 1, requesting 
retransmission
-

 

if i dial any number , i got 34 error. can you please tell what's the wrong 
done by me?

 

-- Executing [8801720039...@default:1] 
Answer("SIP/XXX.XXX.XXX.XXX-116b9aa0", "") in new stack
-- Executing [8801720039...@default:2] Set("SIP/XXX.XXX.XXX.XXX-116b9aa0", 
"CALLERID(all)=14159928422") in new stack
-- Executing [8801720039...@default:3] Dial("SIP/XXX.XXX.XXX.XXX-116b9aa0", 
"DAHDI/g1/920172XX") in new stack
[Apr 19 02:21:19] WARNING[11603]: app_dial.c:1468 dial_exec_full: Unable to 
create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
  == Everyone is busy/congested at this time (1:0/1/0)
-- Executing [8801720039...@default:4] 
Hangup("SIP/XXX.XXX.XXX.XXX-116b9aa0", "") in new stack


 

--

 

Thanks ,

Gobinda


 

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Re: [asterisk-ss7] Asterisk SS7 configuration Problem asterisk-1.6.0.9

[Mesbahuddin Malik]

Hi,

What  problem you face ?

1. You have some mistaken at  2nd -4th E1

2. There is some problem at your chan_dahdi.conf

Fix all those

Thanks
Mesbah






On Sat, Apr 18, 2009 at 1:23 PM, Gobinda Paul  wrote:

> Hi,
>
> I am facing some problem to configure ss7 with asterisk.Here is my
> configuration bellow.
>
> I am using digium 410p card, According to my telco,
> Signal Chanel: 1st Channel Of First E1,
> My pointcode: 3924
> Telco pointcode: 9322
>
>
> --
> [r...@localhost project]# dahdi_hardware
> pci::06:01.0 wct4xxp+ d161:0410 Wildcard TE410P (4th Gen)
>
> --
>
> [r...@localhost project]# cat /etc/dahdi/system.conf
> span=1,1,0,ccs,hdb3,crc4
> bchan=2-31
> mtp2=1
> echocanceller=mg2,2-31
> span=2,2,0,ccs,hdb3,crc4
> bchan=32-62
> echocanceller=mg2,32-62
> span=3,3,0,ccs,hdb3,crc4
> bchan=63-93
> echocanceller=mg2,63-93
> span=4,4,0,ccs,hdb3,crc4
> bchan=94-124
> echocanceller=mg2,94-124
> loadzone= us
> defaultzone = us
>
>
> --
> [r...@localhost project]# cat /etc/asterisk/chan_dahdi.conf
> [trunkgroups]
> [channels]
> language=en
> context=ss7
> switchtype=euroisdn
> signalling=ss7
> toneduration=100
> usecallerid=yes
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> group=1
> callgroup=1
> pickupgroup=1
> ss7type = itu
> ss7_called_nai=dynamic
> ss7_calling_nai=dynamic
> linkset = 1
> pointcode = 3924
> adjpointcode = 9322
> defaultdpc = 9322
> networkindicator=international
> cicbeginswith = 2
> mtp2=1
> sigchan = 1
> channel = 2-31
> cicbeginswith = 32
> channel =33-62
> cicbeginswith = 63
> channel =63-93
> cicbeginswith = 94
> channel = 94-124
>
> --
>
> [r...@localhost project]# cat /etc/asterisk/dahdi-channels.conf
> group=0,11
> context=from-pstn
> switchtype = euroisdn
> signalling = ss7
> channel => 2-31
> context = ss7
> group = 63
> group=0,12
> context=from-pstn
> switchtype = euroisdn
> signalling = ss7
> channel => 32-62
> context = ss7
> group = 63
> group=0,13
> context=from-pstn
> switchtype = euroisdn
> signalling = ss7
> channel => 63-93
> context = ss7
> group = 63
> group=0,14
> context=from-pstn
> switchtype = euroisdn
> signalling = ss7
> channel => 94-124
> context = ss7
> group = 63
>
>
> --
> [r...@localhost project]# cat /etc/asterisk/extensions.conf
> [general]
> static=yes
> writeprotect=no
> [globals]
> [default]
> exten => s,1,Answer()
> exten => s,2,Playback(hello-world)
> exten => s,3,hangup()
> include =>ss7
> [ss7]
> exten => s,1,Answer()
> exten => s,2,hangup()
> exten => _X.,1,Answer()
> exten => _X.,2,Dial(DAHDI/r1/${EXTEN})
> exten => _X.,3,hangup()
>
>
> 
>
>
> *CLI> ss7 show linkset 1
> SS7 linkset 1 status: Up
>
> *CLI> dahdi show status
> Description  Alarms  IRQbpviol CRC4   Fra
> Codi Options  LBO
> T4XXP (PCI) Card 0 Span 1REC 0  0  0  CCS
> HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) [ 1st Channel Used For Signal ]
> T4XXP (PCI) Card 0 Span 2YEL/REC 0  0  0  CCS
> HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)
> T4XXP (PCI) Card 0 Span 3REC 0  0  0  CCS
> HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)
> T4XXP (PCI) Card 0 Span 4RED 0  0  0  CCS
> HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) [ Red Because, I used 3 cable ]
>
>
> *CLI> dahdi restart
> -
> -
> -- Reconfigured channel 124, SS7 signalling
>   == Starting SS7 linkset on span 1
> Huh?! Got FISU in link state 1
> --- SS7 Down ---
> MTP2 link up (SLC 0)
> [Apr 18 19:02:09] WARNING[7672]: chan_dahdi.c:9836 ss7_linkset: MTP2 link
> down (SLC 0)
> Received out of sequence FISU w/ fsn of 0, lastfsnacked = 127, requesting
> retransmission
> MTP2 link up (SLC 0)
> Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting
> retransmission
> --- SS7 Up ---
> Resetting CICs 2 to 33
> Resetting CICs 34 to 61
> Resetting CICs 63 to 94
> Resetting CICs 95 to 124
>
> *CLI> dahdi show channel 2
> Channel: 2
> File Descriptor: 16
> Span: 1
> Extension:
> Dialing: no
> Context: ss7
> Caller ID:
> Calling TON: 0
> Caller ID name:
> Mailbox: none
> Destroy: 0
> InAlarm: 1
> Signalling Type: SS7
> Radio: 0
> Owner: 
> Real: 
> Callwait: 
> Threeway: 
> Confno: -1
> Propagated Conference: -1
> Real in conference: 0
> DSP: no
> Busy Detection: no
> TDD: no
> Relax DTMF: no
> Dialing/CallwaitCAS: 0/0
> Default 

[asterisk-ss7] Asterisk SS7 configuration Problem asterisk-1.6.0.9

[Gobinda Paul]

Hi,
 
I am facing some problem to configure ss7 with asterisk.Here is my 
configuration bellow.
 
I am using digium 410p card, According to my telco, 
Signal Chanel: 1st Channel Of First E1,
My pointcode: 3924
Telco pointcode: 9322
 
--
[r...@localhost project]# dahdi_hardware 
pci::06:01.0 wct4xxp+ d161:0410 Wildcard TE410P (4th Gen)
--
 
[r...@localhost project]# cat /etc/dahdi/system.conf
span=1,1,0,ccs,hdb3,crc4
bchan=2-31
mtp2=1
echocanceller=mg2,2-31
span=2,2,0,ccs,hdb3,crc4
bchan=32-62
echocanceller=mg2,32-62
span=3,3,0,ccs,hdb3,crc4
bchan=63-93
echocanceller=mg2,63-93
span=4,4,0,ccs,hdb3,crc4
bchan=94-124
echocanceller=mg2,94-124
loadzone= us
defaultzone = us
 
--
[r...@localhost project]# cat /etc/asterisk/chan_dahdi.conf 
[trunkgroups]
[channels]
language=en
context=ss7
switchtype=euroisdn
signalling=ss7
toneduration=100
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
group=1
callgroup=1
pickupgroup=1
ss7type = itu
ss7_called_nai=dynamic
ss7_calling_nai=dynamic
linkset = 1
pointcode = 3924
adjpointcode = 9322
defaultdpc = 9322
networkindicator=international
cicbeginswith = 2
mtp2=1
sigchan = 1
channel = 2-31
cicbeginswith = 32
channel =33-62
cicbeginswith = 63
channel =63-93
cicbeginswith = 94
channel = 94-124

--
 
[r...@localhost project]# cat /etc/asterisk/dahdi-channels.conf
group=0,11
context=from-pstn
switchtype = euroisdn
signalling = ss7
channel => 2-31
context = ss7
group = 63
group=0,12
context=from-pstn
switchtype = euroisdn
signalling = ss7
channel => 32-62
context = ss7
group = 63
group=0,13
context=from-pstn
switchtype = euroisdn
signalling = ss7
channel => 63-93
context = ss7
group = 63
group=0,14
context=from-pstn
switchtype = euroisdn
signalling = ss7
channel => 94-124
context = ss7
group = 63

--
[r...@localhost project]# cat /etc/asterisk/extensions.conf 
[general]
static=yes
writeprotect=no
[globals]
[default]
exten => s,1,Answer()
exten => s,2,Playback(hello-world)
exten => s,3,hangup()
include =>ss7
[ss7]
exten => s,1,Answer()
exten => s,2,hangup()
exten => _X.,1,Answer()
exten => _X.,2,Dial(DAHDI/r1/${EXTEN})
exten => _X.,3,hangup()
 

 
 
*CLI> ss7 show linkset 1
SS7 linkset 1 status: Up

*CLI> dahdi show status 
Description  Alarms  IRQbpviol CRC4   Fra Codi 
Options  LBO
T4XXP (PCI) Card 0 Span 1REC 0  0  0  CCS HDB3 
CRC4 0 db (CSU)/0-133 feet (DSX-1) [ 1st Channel Used For Signal ]
T4XXP (PCI) Card 0 Span 2YEL/REC 0  0  0  CCS HDB3 
CRC4 0 db (CSU)/0-133 feet (DSX-1) 
T4XXP (PCI) Card 0 Span 3REC 0  0  0  CCS HDB3 
CRC4 0 db (CSU)/0-133 feet (DSX-1)
T4XXP (PCI) Card 0 Span 4RED 0  0  0  CCS HDB3 
CRC4 0 db (CSU)/0-133 feet (DSX-1) [ Red Because, I used 3 cable ]

 
*CLI> dahdi restart 
-

-

-- Reconfigured channel 124, SS7 signalling
  == Starting SS7 linkset on span 1
Huh?! Got FISU in link state 1
--- SS7 Down ---
MTP2 link up (SLC 0)
[Apr 18 19:02:09] WARNING[7672]: chan_dahdi.c:9836 ss7_linkset: MTP2 link down 
(SLC 0)
Received out of sequence FISU w/ fsn of 0, lastfsnacked = 127, requesting 
retransmission
MTP2 link up (SLC 0)
Received out of sequence MSU w/ fsn of 1, lastfsnacked = 127, requesting 
retransmission
--- SS7 Up ---
Resetting CICs 2 to 33
Resetting CICs 34 to 61
Resetting CICs 63 to 94
Resetting CICs 95 to 124
 
*CLI> dahdi show channel 2
Channel: 2
File Descriptor: 16
Span: 1
Extension: 
Dialing: no
Context: ss7
Caller ID: 
Calling TON: 0
Caller ID name: 
Mailbox: none
Destroy: 0
InAlarm: 1
Signalling Type: SS7
Radio: 0
Owner: 
Real: 
Callwait: 
Threeway: 
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Busy Detection: no
TDD: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: alaw
Fax Handled: no
Pulse phone: no
DND: no
Echo Cancellation:
128 taps
currently OFF
CIC: 2
Hookstate (FXS only): Onhook


 
I talked with telco they said there configuration ok . What's the wrong is done 
me on above configuration?
 
With Regards,
 
Gobinda Paul

Email & Msn: gobi...@live.com
Yahoo: gp00...@yahoo.com
Gtalk:   gobind...@gmail.com


_
Drag n’ drop—Get easy photo sharing with Windows Live™ Photos.

ht

Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 46, Issue 25

[Arix koffi]

Hi Everybody,

Am facing this current problem. I got this message on the console after that 
the server accept DTMF input from the mobile phone. I use Chan_ss7 beta 95.
Short read on linkset "green" CIC=44 (read only 0 of 160) errno=11 (Resource 
temporarily unavailable) (supressed 0)
there is a way to ignore this notice and allow the server to get input anyway ?


Br

Aristide



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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 44, Issue 20

[包海全]

ss7.conf

[linkset-1]
enabled => yes
use_connect => no
enable_st => yes
hunting_policy => even_mru
subservice => auto
context => default

[link-l1]
linkset => 1
channels => 1-15,17-31
schannel => 16
firstcic => 1
enabled => yes

[host-localhostname]
enabled => yes
opc=>0x8e0
dpc=>1:0x3fff
links => l1:1

wanpip1.con
[devices]
wanpipe1 = WAN_AFT_TE1, Comment

[interfaces]
w1g1 = wanpipe1, , TDM_VOICE, Comment

[wanpipe1]
CARD_TYPE   = AFT
S514CPU = A
CommPort= PRI
AUTO_PCISLOT= NO
PCISLOT = 1
PCIBUS  = 3
FE_MEDIA= E1
FE_LCODE= HDB3
FE_FRAME= NCRC4
FE_LINE = 1
TE_CLOCK= NORMAL
TE_REF_CLOCK= 0
TE_SIG_MODE = CCS
TE_HIGHIMPEDANCE= NO
LBO = 120OH
FE_TXTRISTATE   = NO
MTU = 1500
UDPPORT = 9000
TTL = 255
IGNORE_FRONT_END = NO
TDMV_SPAN   = 1
TDMV_DCHAN  = NO
TDMV_HW_DTMF= NO

[w1g1]
ACTIVE_CH   = ALL
TDMV_ECHO_OFF   = NO
TDMV_HWEC   = NO


On Tue, Oct 28, 2008 at 1:00 AM, <[EMAIL PROTECTED]>wrote:

> Send asterisk-ss7 mailing list submissions to
>asterisk-ss7@lists.digium.com
>
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> or, via email, send a message with subject or body 'help' to
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>
> You can reach the person managing the list at
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>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-ss7 digest..."
>
>
> Today's Topics:
>
>   1. Initial Alignment Failure on asterisk+ss7 setup (Ifeoluwa Oyeneye)
>
>
> --
>
> Message: 1
> Date: Mon, 27 Oct 2008 14:11:34 + (GMT)
> From: Ifeoluwa Oyeneye <[EMAIL PROTECTED]>
> Subject: [asterisk-ss7] Initial Alignment Failure on asterisk+ss7
>setup
> To: asterisk-ss7@lists.digium.com
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="utf-8"
>
> Hi folks,
> I'm new to asterisk-ss7.I'm having problem connecting an asterisk box to a
> ss7 link. I'm making use of chan_ss7-1.0.0 with? a sangoma A104 card.
> Here is my configuration:
> ???
> /etc/asterisk/ss7.conf:
> [linkset-suic]
> enabled => yes
> use_connect => no
> enable_st => yes
> hunting_policy => even_mru
> subservice => auto
> context => default
>
> [link-l1]
> linkset => suic
> channels => 1-15,17-31
> schannel => 16
> firstcic => 1
> enabled => yes
>
> [link-l2]
> linkset => suic
> channels => 1-31
> schannel =>
> firstcic => 33
> enabled => no
>
> [link-l3]
> linkset => suic
> channels => 1-31
> schannel =>
> firstcic => 65
> enabled => no
>
> [link-l4]
> linkset => suic
> channels => 1-31
> schannel =>
> firstcic => 97
> enabled => no
>
> [host-ast]
> enabled => yes
> opc => 123
> dpc => suic:98
> links => l1:1
>
> /etc/wanpipe/wanpipe1.conf:
> [devices]
> wanpipe1 = WAN_AFT_TE1, Comment
>
> [interfaces]
> w1g1 = wanpipe1, , TDM_VOICE, Comment
>
> [wanpipe1]
> CARD_TYPE?? = AFT
> S514CPU = A
> CommPort??? = PRI
> AUTO_PCISLOT??? = NO
> PCISLOT = 4
> PCIBUS? = 12
> FE_MEDIA??? = E1
> FE_LCODE??? = HDB3
> FE_FRAME??? = NCRC4
> FE_LINE = 1
> TE_CLOCK??? = NORMAL
> TE_REF_CLOCK??? = 0
> TE_HIGHIMPEDANCE??? = NO
> TE_RX_SLEVEL??? = 120
> LBO = 120OH
> TE_SIG_MODE = CCS
> FE_TXTRISTATE?? = NO
> MTU = 1500
> UDPPORT = 9000
> TTL = 255
> IGNORE_FRONT_END = NO
> TDMV_SPAN?? = 1
> TDMV_DCHAN? = 0
> TDMV_HW_DTMF??? = NO
>
> [w1g1]
> ACTIVE_CH?? = ALL
> TDMV_ECHO_OFF?? = NO
> TDMV_HWEC?? = NO
>
> /etc/zaptel.conf:
> loadzone=uk
> defaultzone=uk
>
> #Sangoma A104 port 1 [slot:4 bus:12 span:1] 
> span=1,0,0,ccs,hdb3
> bchan=1-31
>
> ztcfg - shows no error
>
> this is the error it outputs in the asterisk console
> [Oct 27 14:30:35] WARNING[4139]: mtp.c:457 t2_timeout: MTP2 timer T2
> timeout (failed to receive 'O', 'N', or 'E' after sending 'O'), initial
> alignment failed on link 'l1'.
>
> A similar problem was posted on 15th January, 2008 by Pawel Ratajewski
> (Forweb) rataj at 4web.pl
> but i could not see any posted solution in the archives.
>
> Please help!
>
> Best Regards
> Ife
> [EMAIL PROTECTED]
>
>
>
>
>
> -- next part --
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> URL:
> http://lists.digium.com/pipermail/asterisk-ss7/attachments/20081027/2bcebe4c/attachment-0001.htm
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> End of asterisk-ss7 Digest, Vol 44, Issue 20
> 
>
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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 44, Issue 7

[Adriel Cerrud]

So, i will send a sigchan  question to my SS7 Provider. Well i have a
asterisk with sangoma running ss7boost
ok with this S77 but i want to move to digium + asterisk 1.6 + libss7, i
have search for the configuration files but i have not found it, where
may i search? i think if i find this configuration files  with all the
variables configuration i  could install the ss7 in the new server 




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Re: [asterisk-ss7] Asterisk SS7 as a STP for Number Portability GW

[Joseph]

On 09/02/08, Rony Ron wrote:
> Hi Joseph,
> your solution is very elegant,
> what are those parameters:
> 
> _SS7_LSPI_IDENT=ON
> _SS7_RLT_ON=YES
> 

This tells * to drop out of the call path and let the ss7 provider take
it back.

-- 
respectfully,
Joseph



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Re: [asterisk-ss7] Asterisk SS7 as a STP for Number Portability GW

[Rony Ron]

Hi Joseph,
your solution is very elegant,
what are those parameters:

_SS7_LSPI_IDENT=ON
_SS7_RLT_ON=YES

?

regards



Joseph a écrit :
> On 09/02/08, Rony Ron wrote:
>   
>> Hi,
>> imho you can do it with call forward,
>> you receive the number
>> you check the database if the number is there
>> then forward to the new number (prefixing it with what ever you want)
>> BR,
>> 
>
> There is a way to redirect your call back to the central(Ericsson AXE)
> instead of keeping the media in your path.
>
> Here is a sample:
>
> exten => _X.,1,Set(_SS7_LSPI_IDENT=ON)
> exten => _X.,n,Set(_SS7_RLT_ON=YES)
> exten => _X.,n,Answer()
> exten => _X.,n,Playback(demo-congrats)
>
> back to the ss7 switch based on your lookup results
> and drop out of the media path>
>
> exten => _X.,n,Dial(ZAP/r2/8005551212,30)
> exten => _X.,n,Hangup()
>
>
>
>   
>> Virmones Pereira a écrit :
>> 
>>> Hi,
>>>
>>> I would like to use asterisk with SS7 as a STP for Number Portability 
>>> GW, the idea of the system is follow:
>>>
>>> When the SS7 central(Ericsson AXE) receive the call this should be 
>>> route to the Asterisk to trigger the number portability database by 
>>> SS7/ISUP method if the asterisk found this destination number in the 
>>> number portability database asterisk will insert the Routing Number in 
>>> the begin of the called number and then route back this call to the 
>>> SS7 central.
>>>
>>> Ex:
>>>
>>> user dial 551132323232 this call go the asterisk and asterisk turn 
>>> back with 55112551132323232.
>>>
>>> I wanna do this operation using asterisk as a STP where the SS7 use 
>>> only the signaling channel, the media should go directly to the SSP
>>>
>>> somebody knows how to do it?
>>>
>>>   
>>>   
>
>   
> 
>
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Re: [asterisk-ss7] Asterisk SS7 as a STP for Number Portability GW

[Joseph]

On 09/02/08, Virmones Pereira wrote:
> Regards for everybody
> 
> Joseph your solution is very interesting, but I think I will have another
> two problems the first trouble is with billing, Because if I answer the call
> it will be charged right?
> And the second trouble is when I send the Hangup the asterisk will drop the
> call right?


If you answer the call, that will create a CDR. 

But you do need to answer the call.

If the redirect works like expected, than the call is gone, so you can't
drop it anymore :)


-- 
respectfully,
Joseph



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Re: [asterisk-ss7] Asterisk SS7 as a STP for Number Portability GW

[Wasim Baig]

On Tue, Sep 2, 2008 at 8:26 PM, Virmones Pereira <[EMAIL PROTECTED]>wrote:

> Regards for everybody
>
> Joseph your solution is very interesting, but I think I will have another
> two problems the first trouble is with billing, Because if I answer the call
> it will be charged right?
> And the second trouble is when I send the Hangup the asterisk will drop the
> call right?
>

playback has an option |noanswer that won't answer the call



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Re: [asterisk-ss7] Asterisk SS7 as a STP for Number Portability GW

[Virmones Pereira]

Regards for everybody

Joseph your solution is very interesting, but I think I will have another
two problems the first trouble is with billing, Because if I answer the call
it will be charged right?
And the second trouble is when I send the Hangup the asterisk will drop the
call right?




2008/9/2 Joseph <[EMAIL PROTECTED]>

> On 09/02/08, Rony Ron wrote:
> > Hi,
> > imho you can do it with call forward,
> > you receive the number
> > you check the database if the number is there
> > then forward to the new number (prefixing it with what ever you want)
> > BR,
>
> There is a way to redirect your call back to the central(Ericsson AXE)
> instead of keeping the media in your path.
>
> Here is a sample:
>
> exten => _X.,1,Set(_SS7_LSPI_IDENT=ON)
> exten => _X.,n,Set(_SS7_RLT_ON=YES)
> exten => _X.,n,Answer()
> exten => _X.,n,Playback(demo-congrats)
>
>   back to the ss7 switch based on your lookup results
>and drop out of the media path>
>
> exten => _X.,n,Dial(ZAP/r2/8005551212,30)
> exten => _X.,n,Hangup()
>
>
>
> >
> > Virmones Pereira a écrit :
> > > Hi,
> > >
> > > I would like to use asterisk with SS7 as a STP for Number Portability
> > > GW, the idea of the system is follow:
> > >
> > > When the SS7 central(Ericsson AXE) receive the call this should be
> > > route to the Asterisk to trigger the number portability database by
> > > SS7/ISUP method if the asterisk found this destination number in the
> > > number portability database asterisk will insert the Routing Number in
> > > the begin of the called number and then route back this call to the
> > > SS7 central.
> > >
> > > Ex:
> > >
> > > user dial 551132323232 this call go the asterisk and asterisk turn
> > > back with 55112551132323232.
> > >
> > > I wanna do this operation using asterisk as a STP where the SS7 use
> > > only the signaling channel, the media should go directly to the SSP
> > >
> > > somebody knows how to do it?
> > >
> > >
>
> --
> respectfully,
> Joseph
>
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.9 (GNU/Linux)
>
> iEYEARECAAYFAki9GN4ACgkQ5CyZqOno04xQ5ACfdnzkePZP4Ubt/A20ZuK6o9E9
> iS0AniIN8HWmn3iASu7VU3RA8zNqJQOl
> =rRpr
> -END PGP SIGNATURE-
>
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Re: [asterisk-ss7] Asterisk SS7 as a STP for Number Portability GW

[Joseph]

On 09/02/08, Rony Ron wrote:
> Hi,
> imho you can do it with call forward,
> you receive the number
> you check the database if the number is there
> then forward to the new number (prefixing it with what ever you want)
> BR,

There is a way to redirect your call back to the central(Ericsson AXE)
instead of keeping the media in your path.

Here is a sample:

exten => _X.,1,Set(_SS7_LSPI_IDENT=ON)
exten => _X.,n,Set(_SS7_RLT_ON=YES)
exten => _X.,n,Answer()
exten => _X.,n,Playback(demo-congrats)

   

exten => _X.,n,Dial(ZAP/r2/8005551212,30)
exten => _X.,n,Hangup()



> 
> Virmones Pereira a écrit :
> > Hi,
> >
> > I would like to use asterisk with SS7 as a STP for Number Portability 
> > GW, the idea of the system is follow:
> >
> > When the SS7 central(Ericsson AXE) receive the call this should be 
> > route to the Asterisk to trigger the number portability database by 
> > SS7/ISUP method if the asterisk found this destination number in the 
> > number portability database asterisk will insert the Routing Number in 
> > the begin of the called number and then route back this call to the 
> > SS7 central.
> >
> > Ex:
> >
> > user dial 551132323232 this call go the asterisk and asterisk turn 
> > back with 55112551132323232.
> >
> > I wanna do this operation using asterisk as a STP where the SS7 use 
> > only the signaling channel, the media should go directly to the SSP
> >
> > somebody knows how to do it?
> >
> >   

-- 
respectfully,
Joseph


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Re: [asterisk-ss7] Asterisk SS7 as a STP for Number Portability GW

[Rony Ron]

Yep this is another way of doing it but * does not support yet INAP;
we could then setup * as an scp and * would become OpenIN :)
regards,

Charl Barnard a écrit :
>
> Hi,
>
> You’d need INAP to implement this, configuring the exchange 
> (SSP/”central”) to at a trigger detection point (TDP) perform an IN 
> query to your node, which would act as an SCP, do the lookup, and 
> return with a “Connect” instruction with the new number to the SSP.
>
> Asterisk only supports ISUP, not INAP, and therefore you cannot do 
> this. You would be able to do it if you left Asterisk in the media 
> path, in other words making the call to the translated number, and 
> bridging the two legs, but I understand you don’t want to do this. For 
> small networks it might be viable, though.
>
> Cheers,
>
> Charl
>
> *From:* [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] *On Behalf Of *Virmones 
> Pereira
> *Sent:* 02 September 2008 08:58
> *To:* asterisk-ss7@lists.digium.com
> *Subject:* [asterisk-ss7] Asterisk SS7 as a STP for Number Portability GW
>
> Hi,
>
> I would like to use asterisk with SS7 as a STP for Number Portability 
> GW, the idea of the system is follow:
>
> When the SS7 central(Ericsson AXE) receive the call this should be 
> route to the Asterisk to trigger the number portability database by 
> SS7/ISUP method if the asterisk found this destination number in the 
> number portability database asterisk will insert the Routing Number in 
> the begin of the called number and then route back this call to the 
> SS7 central.
>
> Ex:
>
> user dial 551132323232 this call go the asterisk and asterisk turn 
> back with 55112551132323232.
>
> I wanna do this operation using asterisk as a STP where the SS7 use 
> only the signaling channel, the media should go directly to the SSP
>
> somebody knows how to do it?
>
>
> _
> ( )
> ( DB )
> ( )
> (_)
> /|\
> |
> ___\|/_
> | ASTERISK |
> |___| /|\ | | | 55121 1132323232
> 551132323232 | |
> ___|___\|/___ _
> ___ 551132323232 | | 55121 1132323232 | |
> /_\>| SS7 Central |>| 
> SSP |
> | | | |
> |_| |_|
>
>
>
>
>
>
>
>
>
>
>
>
> 
>
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Re: [asterisk-ss7] Asterisk SS7 as a STP for Number Portability GW

[Rony Ron]

i was thinking that he doesn't need to use ss7 routing (i don't think * 
can do that)
but he could try to handle it in the dial plan
regards,

Attila Domjan a écrit :
> But, have to implement the network routing number nature of address
> (=8), like ss7_internationalprefix it could be ss7_networkroutingprefix.
>
> On Tue, 2008-09-02 at 07:38 +, Rony Ron wrote:
>   
>> Hi,
>> imho you can do it with call forward,
>> you receive the number
>> you check the database if the number is there
>> then forward to the new number (prefixing it with what ever you want)
>> BR,
>>
>> Virmones Pereira a écrit :
>> 
>>> Hi,
>>>
>>> I would like to use asterisk with SS7 as a STP for Number Portability 
>>> GW, the idea of the system is follow:
>>>
>>> When the SS7 central(Ericsson AXE) receive the call this should be 
>>> route to the Asterisk to trigger the number portability database by 
>>> SS7/ISUP method if the asterisk found this destination number in the 
>>> number portability database asterisk will insert the Routing Number in 
>>> the begin of the called number and then route back this call to the 
>>> SS7 central.
>>>
>>> Ex:
>>>
>>> user dial 551132323232 this call go the asterisk and asterisk turn 
>>> back with 55112551132323232.
>>>
>>> I wanna do this operation using asterisk as a STP where the SS7 use 
>>> only the signaling channel, the media should go directly to the SSP
>>>
>>> somebody knows how to do it?
>>>
>>>   
>>> 
>>>  _
>>> ( 
>>> )   
>>> ( DB  
>>> )   
>>> ( 
>>> )   
>>> 
>>> (_)   
>>>   
>>> /|\ 
>>>
>>> |  
>>>
>>> ___\|/_
>>>   | ASTERISK  
>>> |   
>>>   |___|
>>>  /|\   
>>> |   
>>> ||  55121 1132323232
>>>551132323232   ||   
>>>___|___\|/___  
>>>  _  
>>>   ___  551132323232   | |  55121 
>>> 1132323232   | |
>>>   /_\>| SS7 Central 
>>> |>|   SSP   |
>>>   | | 
>>> | |
>>>   |_|
>>>  |_|
>>>
>>>
>>> 
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> 
>>>
>>> ___
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>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-ss7
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Re: [asterisk-ss7] Asterisk SS7 as a STP for Number Portability GW

[Charl Barnard]

Hi,

You'd need INAP to implement this, configuring the exchange (SSP/"central")
to at a trigger detection point (TDP) perform an IN query to your node,
which would act as an SCP, do the lookup, and return with a "Connect"
instruction with  the new number to the SSP.

Asterisk only supports ISUP, not INAP, and therefore you cannot do this. You
would be able to do it if you left Asterisk in the media path, in other
words making the call to the translated number, and bridging the two legs,
but I understand you don't want to do this. For small networks it might be
viable, though.

Cheers,

Charl

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Virmones Pereira
Sent: 02 September 2008 08:58
To: asterisk-ss7@lists.digium.com
Subject: [asterisk-ss7] Asterisk SS7 as a STP for Number Portability GW

 

Hi,

I would like to use asterisk with SS7 as a STP for Number Portability GW,
the idea of the system is follow:

When the SS7 central(Ericsson AXE) receive the call this should be route to
the Asterisk to trigger the number portability database by SS7/ISUP method
if the asterisk found this destination number in the number portability
database asterisk will insert the Routing Number in the begin of the called
number and then route back this call to the SS7 central.

Ex:

user dial 551132323232 this call go the asterisk and asterisk turn back with
55112551132323232.

I wanna do this operation using asterisk as a STP where the SS7 use only the
signaling channel, the media should go directly to the SSP

somebody knows how to do it?

   
 
_
( )

( DB  )

( )

(_)

  /|\

   |

   ___\|/_

  | ASTERISK  |

  |___|
/|\   |
||  55121 1132323232
   551132323232   ||
   ___|___\|/___
_   
  ___  551132323232   | |  55121 1132323232   |
|
  /_\>| SS7 Central |>|
SSP   |
  | | |
| 
  |_|
|_| 














 

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Re: [asterisk-ss7] Asterisk SS7 as a STP for Number Portability GW

[Attila Domjan]

But, have to implement the network routing number nature of address
(=8), like ss7_internationalprefix it could be ss7_networkroutingprefix.

On Tue, 2008-09-02 at 07:38 +, Rony Ron wrote:
> Hi,
> imho you can do it with call forward,
> you receive the number
> you check the database if the number is there
> then forward to the new number (prefixing it with what ever you want)
> BR,
> 
> Virmones Pereira a écrit :
> > Hi,
> >
> > I would like to use asterisk with SS7 as a STP for Number Portability 
> > GW, the idea of the system is follow:
> >
> > When the SS7 central(Ericsson AXE) receive the call this should be 
> > route to the Asterisk to trigger the number portability database by 
> > SS7/ISUP method if the asterisk found this destination number in the 
> > number portability database asterisk will insert the Routing Number in 
> > the begin of the called number and then route back this call to the 
> > SS7 central.
> >
> > Ex:
> >
> > user dial 551132323232 this call go the asterisk and asterisk turn 
> > back with 55112551132323232.
> >
> > I wanna do this operation using asterisk as a STP where the SS7 use 
> > only the signaling channel, the media should go directly to the SSP
> >
> > somebody knows how to do it?
> >
> >   
> > 
> >  _
> > ( 
> > )   
> > ( DB  
> > )   
> > ( 
> > )   
> > 
> > (_)   
> >   
> > /|\ 
> >
> > |  
> >
> > ___\|/_
> >   | ASTERISK  
> > |   
> >   |___|
> >  /|\   
> > |   
> > ||  55121 1132323232
> >551132323232   ||   
> >___|___\|/___  
> >  _  
> >   ___  551132323232   | |  55121 
> > 1132323232   | |
> >   /_\>| SS7 Central 
> > |>|   SSP   |
> >   | | 
> > | |
> >   |_|
> >  |_|
> >
> >
> > 
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > 
> >
> > ___
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> >http://lists.digium.com/mailman/listinfo/asterisk-ss7
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Re: [asterisk-ss7] Asterisk SS7 as a STP for Number Portability GW

[Attila Domjan]

Hi,
asterisk can't be an STP.

On Tue, 2008-09-02 at 03:57 -0300, Virmones Pereira wrote:
> Hi,
> 
> I would like to use asterisk with SS7 as a STP for Number Portability
> GW, the idea of the system is follow:
> 
> When the SS7 central(Ericsson AXE) receive the call this should be
> route to the Asterisk to trigger the number portability database by
> SS7/ISUP method if the asterisk found this destination number in the
> number portability database asterisk will insert the Routing Number in
> the begin of the called number and then route back this call to the
> SS7 central.
> 
> Ex:
> 
> user dial 551132323232 this call go the asterisk and asterisk turn
> back with 55112551132323232.
> 
> I wanna do this operation using asterisk as a STP where the SS7 use
> only the signaling channel, the media should go directly to the SSP
> 
> somebody knows how to do it?
> 
>
> 
> _
> 
> ( )
> 
> ( DB  )
> 
> ( )
> 
> (_)
>   /|
> \  
> 
> |   
>___
> \|/_ 
>   | ASTERISK
> |
>   |___|
>  /|\   |
> ||  55121 1132323232
>551132323232   ||
>___|___\|/___
> _   
>   ___  551132323232   | |  55121
> 1132323232   | |
>   /_\>| SS7 Central
> |>|   SSP   |
>   | |
> | | 
>   |_|
> |_| 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> ___
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Re: [asterisk-ss7] Asterisk SS7 as a STP for Number Portability GW

[Rony Ron]

Hi,
imho you can do it with call forward,
you receive the number
you check the database if the number is there
then forward to the new number (prefixing it with what ever you want)
BR,

Virmones Pereira a écrit :
> Hi,
>
> I would like to use asterisk with SS7 as a STP for Number Portability 
> GW, the idea of the system is follow:
>
> When the SS7 central(Ericsson AXE) receive the call this should be 
> route to the Asterisk to trigger the number portability database by 
> SS7/ISUP method if the asterisk found this destination number in the 
> number portability database asterisk will insert the Routing Number in 
> the begin of the called number and then route back this call to the 
> SS7 central.
>
> Ex:
>
> user dial 551132323232 this call go the asterisk and asterisk turn 
> back with 55112551132323232.
>
> I wanna do this operation using asterisk as a STP where the SS7 use 
> only the signaling channel, the media should go directly to the SSP
>
> somebody knows how to do it?
>
>   
> 
>  _
> ( 
> )   
> ( DB  
> )   
> ( 
> )   
> 
> (_)   
>   
> /|\ 
>
> |  
>
> ___\|/_
>   | ASTERISK  
> |   
>   |___|
>  /|\   
> |   
> ||  55121 1132323232
>551132323232   ||   
>___|___\|/___  
>  _  
>   ___  551132323232   | |  55121 
> 1132323232   | |
>   /_\>| SS7 Central 
> |>|   SSP   |
>   | | 
> | |
>   |_|
>  |_|
>
>
> 
>
>
>
>
>
>
>
>
>
>
>
> 
>
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[asterisk-ss7] Asterisk SS7 as a STP for Number Portability GW

[Virmones Pereira]

Hi,

I would like to use asterisk with SS7 as a STP for Number Portability GW,
the idea of the system is follow:

When the SS7 central(Ericsson AXE) receive the call this should be route to
the Asterisk to trigger the number portability database by SS7/ISUP method
if the asterisk found this destination number in the number portability
database asterisk will insert the Routing Number in the begin of the called
number and then route back this call to the SS7 central.

Ex:

user dial 551132323232 this call go the asterisk and asterisk turn back with
55112551132323232.

I wanna do this operation using asterisk as a STP where the SS7 use only the
signaling channel, the media should go directly to the SSP

somebody knows how to do it?



 _
(
)
( DB
)
(
)

(_)

/|\

|

___\|/_
  | ASTERISK
|
  |___|
 /|\
|   |
|  55121 1132323232
   551132323232   ||
   ___|___\|/___
_
  ___  551132323232   | |  55121 1132323232
| |
  /_\>| SS7 Central
|>|   SSP   |
  | | |
|
  |_|
|_|
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Re: [asterisk-ss7] Asterisk SS7, SG, MG and MGCP

[Matthew Fredrickson]

Anton wrote:
> Hey Matthew, good to hear that!
> If you are willing to write the driver compatible for HUAWEI 
> MGCP/MEGACO (telco grade) gateways, I have 1 port gateway 
> here, and can donate it for this task.

Let me think about it.  I would like to do both components (media 
gateway and media gateway controller), so it's just a matter of which to 
do first.  I started on the media gateway earlier this summer, but I'm 
thinking about scrapping what I have and modifying Asterisk so that it 
could be a Media Gateway instead of doing a standalone application.  If 
I decide to stop, I may just start working on the media gateway 
controller instead.  The only reason I started with the media gateway 
was the lack of a true media gateway to test against when writing the 
new channel driver.

Matthew Fredrickson

> 
> On Thursday 14 August 2008 20:25, Matthew Fredrickson wrote:
>> Anton wrote:
>>> As SS7 MGCP you probably mean MEGACO protocol?
>> I'm thinking I'd start with MGCP, since it is a text
>> based protocol instead.
>>
>>> Unlikely this could be seen in asterisk anytime soon...
>> That's likely.  It's going to take a while to do this. 
>> I've started playing with this, and the first thing I've
>> started doing is writing my own MGCP media gateway, so I
>> have something to test against (I don't have any MGCP
>> equipment here).  Then I'll start the channel driver.
>>
>> Matthew Fredrickson
>>
>>> On Thursday 14 August 2008 11:11, voip me wrote:
 Dear Matthew,

 What do you think , how could one improve it then, you
 think chan_dahdi should speak MGCP or any request show
 come up to channel then bridged (or in some other way)
 to an MGCP channel (not the current but a changed MGCP
 channel) and the rest.
 I have another question Matthiew, are you familiar
 with MGCP, how could one address a channel on gateway?

 Regards.
 --
 MSH

 On 13/08/2008, Matthew Fredrickson
 <[EMAIL PROTECTED]>
>>> wrote:
> voip me wrote:
>> Dear Folks,
>>
>> Sorry for my ambigouse subject. I want to set the
>> following solution and need your idea's and
>> experinces. I want to setup an asterisk box with one
>> E1 link containing 4 or 5 SS7 signalling links and
>> have two or three asterisk boxes each with 8 E1
>> (voice only) as media gateway. What is your idea's,
>> is it possible, it seems MGCP is for this case but
>> aa more i look in asterisk MGCP channel less I get
>> what's its usage. Any one could shed some lights on
>> this topic, im somehow mixed up. How could I control
>> the media gateway asterisks to tell them which
>> channel has a call and where it should go?
> Asterisk's chan_mgcp does not support being a media
> gateway right now. You can only control media
> gateways. Also, Asterisk doesn't currently support
> signalling reception on anything other than a DS0
> directly connected to the machine.
>
> You're not going to be able to control media gateways
> either that are signalled via SS7, since chan_dahdi
> does not speak MGCP.
>
> These are both things that I would like to improve
> though...
>
> Matthew Fredrickson
> Digium, Inc.
>
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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 42, Issue 14

[Matthew Fredrickson]

anonymus anonymus wrote:
> Hi all:
> 
> I have a questions,  the strategic plain of Asterisk-SS7 not considered 
> evolution to sigtran? Remenber sigtran is a very cost effcient evolution 
> of traditional links SS7, more and more telco's STP today handle sigtran 
> signaling, and I think is more cheap cards ethernet than cards E1's.

I don't think it's a matter of not considering it, I think we definitely 
need to be able to support sigtran.  However, the importance for doing 
it has not been great, since I have mostly been concerned with making 
sure that the configuration we have right now (MTP2/MTP3/ISUP over 
T1/E1) is solid.

I would think that in the future there will probably be some level of 
sigtran support.  I have looked into it a few times, but again, I do not 
have any equipment that supports it.  Before all else though, it boils 
down to this: there are a lot of things I "could" spend my time on, but 
I've been trying to identify the most effective things to spend my time on.

As of late, I have been thinking the channel driver with MGCP and SS7 
support would be the most effective.  However, that project will 
probably include signalling gateway support as well (which will probably 
include sigtran) so I wouldn't get terribly worried that it will not be 
worked on at some point.

Matthew Fredrickson

> 
> Thanks and regards.
> 
> --- El *jue, 14/8/08, [EMAIL PROTECTED] 
> /<[EMAIL PROTECTED]>/* escribió:
> 
> De: [EMAIL PROTECTED]
> <[EMAIL PROTECTED]>
> Asunto: asterisk-ss7 Digest, Vol 42, Issue 14
> Para: asterisk-ss7@lists.digium.com
> Fecha: jueves, 14 agosto, 2008 2:00
> 
> Send asterisk-ss7 mailing list submissions to
> 
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> 
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> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-ss7 digest..."
> 
> 
> Today's Topics:
> 
>1. Re: ss7 newbie (Christopher (JVM))
>2. Re: ss7 newbie (Matthew Fredrickson)
>3. Re: ss7 newbie (olivier taylor)
> 
> 
> --
> 
> Message: 1
> Date: Thu, 14 Aug 2008 11:43:47 -0400
> From: "Christopher (JVM)" <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-ss7] ss7 newbie
> To: asterisk-ss7@lists.digium.com
> Message-ID:
>  <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset=iso-8859-1
> 
> As a newbie myself I was wondering if someone could tell me the 
> difference 
> between the pointcode the adjacent point code and the defaultdpc. I am 
> trying to obtain these infos from my carrier but he is only giving a 
> single 
> point code.
> 
> On Wed, 13 Aug 2008 10:49:26 -0500, Matthew Fredrickson wrote
> > Sriram wrote:
> > > Hi
> > > Thnks for the reply...So say if i want 120 channels over ss7 
> > > signalling...will the telco give me 4 E1s and 1 separate E1 for ss7 
> > > signalling or will the ss7 be in one of the 4 E1s itself..?
> > 
> > Sometimes.  It depends on what you negotiate with the telco you're 
> > interconnecting with.  Some of them like to waste an E1 on a single 
> > signalling link though.  And even stupider ones waste a single E1 on 
> > two signalling links.  It seems
>  ridiculous to mandate redundant 
> > signalling links and have them running over the same E1 (Don't ask,
> >  I had somebody come to me having to deal with that).
> > 
> > But in any case, you can negotiate with them how you would like to 
> > set it up typically.
> > 
> > > Do i have to just plug in those 4 E1s directly into the TE420P and 
> > > install libss7 ? thats it ?
> > 
> > Yes :-)  And configure it.
> > 
> > > Also please tell me what are the advantages of ss7 over PRI in terms
> of 
> > > 1.call handling capacity 2. connectivity 3. reliability ..
> > 
> > If you need answers to that question, you probably need to research 
> > more into what SS7 is.
> > 
> > Typically, you have a more reliable link since you can have 
> > redundant signalling channels.  The protocol also supports a lot 
> > more bearers per signalling link than you can do in PRI (or
>  even 
> > NFAS PRI).
> > 
> > Matthew Fredrickson
> > Digium, Inc.
> > 
> > ___
> > --Bandwidth and Colocation Provided by http://www.api-digital.com--
> > 
> > asterisk-ss7 mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-ss7
> 
> 
> --
> Christopher Srinivasa
> VoIP 

Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 42, Issue 14

[anonymus anonymus]

Hi all:

I have a questions,  the strategic plain of Asterisk-SS7 not considered 
evolution to sigtran? Remenber sigtran is a very cost effcient evolution of 
traditional links SS7, more and more telco's STP today handle sigtran 
signaling, and I think is more cheap cards ethernet than cards E1's.

Thanks and regards.

--- El jue, 14/8/08, [EMAIL PROTECTED] <[EMAIL PROTECTED]> escribió:
De: [EMAIL PROTECTED] <[EMAIL PROTECTED]>
Asunto: asterisk-ss7 Digest, Vol 42, Issue 14
Para: asterisk-ss7@lists.digium.com
Fecha: jueves, 14 agosto, 2008 2:00

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Today's Topics:

   1. Re: ss7 newbie (Christopher (JVM))
   2. Re: ss7 newbie (Matthew Fredrickson)
   3. Re: ss7 newbie (olivier taylor)


--

Message: 1
Date: Thu, 14 Aug 2008 11:43:47 -0400
From: "Christopher (JVM)" <[EMAIL PROTECTED]>
Subject: Re: [asterisk-ss7] ss7 newbie
To: asterisk-ss7@lists.digium.com
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain;   charset=iso-8859-1

As a newbie myself I was wondering if someone could tell me the difference 
between the pointcode the adjacent point code and the defaultdpc. I am 
trying to obtain these infos from my carrier but he is only giving a single 
point code.

On Wed, 13 Aug 2008 10:49:26 -0500, Matthew Fredrickson wrote
> Sriram wrote:
> > Hi
> > Thnks for the reply...So say if i want 120 channels over ss7 
> > signalling...will the telco give me 4 E1s and 1 separate E1 for ss7 
> > signalling or will the ss7 be in one of the 4 E1s itself..?
> 
> Sometimes.  It depends on what you negotiate with the telco you're 
> interconnecting with.  Some of them like to waste an E1 on a single 
> signalling link though.  And even stupider ones waste a single E1 on 
> two signalling links.  It seems ridiculous to mandate redundant 
> signalling links and have them running over the same E1 (Don't ask,
>  I had somebody come to me having to deal with that).
> 
> But in any case, you can negotiate with them how you would like to 
> set it up typically.
> 
> > Do i have to just plug in those 4 E1s directly into the TE420P and 
> > install libss7 ? thats it ?
> 
> Yes :-)  And configure it.
> 
> > Also please tell me what are the advantages of ss7 over PRI in terms
of 
> > 1.call handling capacity 2. connectivity 3. reliability ..
> 
> If you need answers to that question, you probably need to research 
> more into what SS7 is.
> 
> Typically, you have a more reliable link since you can have 
> redundant signalling channels.  The protocol also supports a lot 
> more bearers per signalling link than you can do in PRI (or even 
> NFAS PRI).
> 
> Matthew Fredrickson
> Digium, Inc.
> 
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-ss7


--
Christopher Srinivasa
VoIP Researcher and Developer / Chercheur et Developpeur VoIP
[EMAIL PROTECTED]
JVM INFORMATIQUE CANADA INC.
9655-E rue Ignace
Brossard, Quebec, Canada J4Y 2P3
Tel: 514-951-2981 x2034
Alt. Tel: 450-444-1884 x2034
Fax: 450-444-1884




--

Message: 2
Date: Thu, 14 Aug 2008 10:52:31 -0500
From: Matthew Fredrickson <[EMAIL PROTECTED]>
Subject: Re: [asterisk-ss7] ss7 newbie
To: asterisk-ss7@lists.digium.com
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Christopher (JVM) wrote:
> As a newbie myself I was wondering if someone could tell me the difference

> between the pointcode the adjacent point code and the defaultdpc. I am 
> trying to obtain these infos from my carrier but he is only giving a
single 
> point code.

The adjpointcode is the point code of the entity on the remote end of 
the sigchan you are using.

The defaultdpc is like the default route.  So if your telco has STPs 
between you and the entity that you are supposed to be sending your ISUP 
traffic to (basically an additional hop), you set the defaultdpc to the 
node that your ISUP traffic should go to.  If there are no STPs (or 
additional hops) between you and the point code that you are sending 
your traffic to, you should just set the adjpointcode and the defaultdpc 
to be the same thing.

Matthew Fredrickson

> 
> On Wed, 13 Aug 2008 10:49:26 -0500, Matthew Fredrickson wrote
>> Sriram wrote:
>>> Hi
>>> Thnks for the reply...So say if i want 120 channels over ss7 
>>> signalling...will t

Re: [asterisk-ss7] Asterisk SS7, SG, MG and MGCP

[Anton]

Hey Matthew, good to hear that!
If you are willing to write the driver compatible for HUAWEI 
MGCP/MEGACO (telco grade) gateways, I have 1 port gateway 
here, and can donate it for this task.

On Thursday 14 August 2008 20:25, Matthew Fredrickson wrote:
> Anton wrote:
> > As SS7 MGCP you probably mean MEGACO protocol?
>
> I'm thinking I'd start with MGCP, since it is a text
> based protocol instead.
>
> > Unlikely this could be seen in asterisk anytime soon...
>
> That's likely.  It's going to take a while to do this. 
> I've started playing with this, and the first thing I've
> started doing is writing my own MGCP media gateway, so I
> have something to test against (I don't have any MGCP
> equipment here).  Then I'll start the channel driver.
>
> Matthew Fredrickson
>
> > On Thursday 14 August 2008 11:11, voip me wrote:
> >> Dear Matthew,
> >>
> >> What do you think , how could one improve it then, you
> >> think chan_dahdi should speak MGCP or any request show
> >> come up to channel then bridged (or in some other way)
> >> to an MGCP channel (not the current but a changed MGCP
> >> channel) and the rest.
> >> I have another question Matthiew, are you familiar
> >> with MGCP, how could one address a channel on gateway?
> >>
> >> Regards.
> >> --
> >> MSH
> >>
> >> On 13/08/2008, Matthew Fredrickson
> >> <[EMAIL PROTECTED]>
> >
> > wrote:
> >>> voip me wrote:
>  Dear Folks,
> 
>  Sorry for my ambigouse subject. I want to set the
>  following solution and need your idea's and
>  experinces. I want to setup an asterisk box with one
>  E1 link containing 4 or 5 SS7 signalling links and
>  have two or three asterisk boxes each with 8 E1
>  (voice only) as media gateway. What is your idea's,
>  is it possible, it seems MGCP is for this case but
>  aa more i look in asterisk MGCP channel less I get
>  what's its usage. Any one could shed some lights on
>  this topic, im somehow mixed up. How could I control
>  the media gateway asterisks to tell them which
>  channel has a call and where it should go?
> >>>
> >>> Asterisk's chan_mgcp does not support being a media
> >>> gateway right now. You can only control media
> >>> gateways. Also, Asterisk doesn't currently support
> >>> signalling reception on anything other than a DS0
> >>> directly connected to the machine.
> >>>
> >>> You're not going to be able to control media gateways
> >>> either that are signalled via SS7, since chan_dahdi
> >>> does not speak MGCP.
> >>>
> >>> These are both things that I would like to improve
> >>> though...
> >>>
> >>> Matthew Fredrickson
> >>> Digium, Inc.
> >>>
> >>> ___
> >>> --Bandwidth and Colocation Provided by
> >>> http://www.api-digital.com--
> >>>
> >>> asterisk-ss7 mailing list
> >>> To UNSUBSCRIBE or update options visit:
> >>>  
> >>> http://lists.digium.com/mailman/listinfo/asterisk-ss7
> >
> > ___
> > --Bandwidth and Colocation Provided by
> > http://www.api-digital.com--
> >
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> > To UNSUBSCRIBE or update options visit:
> >   
> > http://lists.digium.com/mailman/listinfo/asterisk-ss7
>
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Re: [asterisk-ss7] Asterisk SS7, SG, MG and MGCP

[Matthew Fredrickson]

Anton wrote:
> As SS7 MGCP you probably mean MEGACO protocol? 

I'm thinking I'd start with MGCP, since it is a text based protocol instead.

> Unlikely this could be seen in asterisk anytime soon...

That's likely.  It's going to take a while to do this.  I've started 
playing with this, and the first thing I've started doing is writing my 
own MGCP media gateway, so I have something to test against (I don't 
have any MGCP equipment here).  Then I'll start the channel driver.

Matthew Fredrickson

> 
> On Thursday 14 August 2008 11:11, voip me wrote:
>> Dear Matthew,
>>
>> What do you think , how could one improve it then, you
>> think chan_dahdi should speak MGCP or any request show
>> come up to channel then bridged (or in some other way) to
>> an MGCP channel (not the current but a changed MGCP
>> channel) and the rest.
>> I have another question Matthiew, are you familiar with
>> MGCP, how could one address a channel on gateway?
>>
>> Regards.
>> --
>> MSH
>>
>> On 13/08/2008, Matthew Fredrickson <[EMAIL PROTECTED]> 
> wrote:
>>> voip me wrote:
 Dear Folks,

 Sorry for my ambigouse subject. I want to set the
 following solution and need your idea's and
 experinces. I want to setup an asterisk box with one
 E1 link containing 4 or 5 SS7 signalling links and
 have two or three asterisk boxes each with 8 E1
 (voice only) as media gateway. What is your idea's,
 is it possible, it seems MGCP is for this case but aa
 more i look in asterisk MGCP channel less I get
 what's its usage. Any one could shed some lights on
 this topic, im somehow mixed up. How could I control
 the media gateway asterisks to tell them which
 channel has a call and where it should go?
>>> Asterisk's chan_mgcp does not support being a media
>>> gateway right now. You can only control media gateways.
>>>  Also, Asterisk doesn't currently support signalling
>>> reception on anything other than a DS0 directly
>>> connected to the machine.
>>>
>>> You're not going to be able to control media gateways
>>> either that are signalled via SS7, since chan_dahdi
>>> does not speak MGCP.
>>>
>>> These are both things that I would like to improve
>>> though...
>>>
>>> Matthew Fredrickson
>>> Digium, Inc.
>>>
>>> ___
>>> --Bandwidth and Colocation Provided by
>>> http://www.api-digital.com--
>>>
>>> asterisk-ss7 mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-ss7
> 
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
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Re: [asterisk-ss7] Asterisk SS7, SG, MG and MGCP

[Matthew Fredrickson]

voip me wrote:
> Dear Matthew,
>  
> What do you think , how could one improve it then, you think chan_dahdi 
> should speak MGCP or any request show come up to channel then bridged 
> (or in some other way) to an MGCP channel (not the current but a changed 
> MGCP channel) and the rest.

That's basically what I have had in mind.  Either use chan_dahdi or a 
new channel driver which can talk SS7 and can also talk MGCP or 
Zap/DAHDI (and maybe SIP) for bearer control.

> I have another question Matthiew, are you familiar with MGCP, how could 
> one address a channel on gateway?

For analog lines, it is:
aaln/[EMAIL PROTECTED]

For DS0, IIRC, it is:
ds/ds1-1/[EMAIL PROTECTED]

See rfc3435 Appendix E.1

Matthew Fredrickson


>  
> Regards.
> --
> MSH
> 
>  
> On 13/08/2008, *Matthew Fredrickson* <[EMAIL PROTECTED] 
> > wrote:
> 
> voip me wrote:
>  > Dear Folks,
>  >
>  > Sorry for my ambigouse subject. I want to set the following
> solution and
>  > need your idea's and experinces. I want to setup an asterisk box with
>  > one E1 link containing 4 or 5 SS7 signalling links and have two
> or three
>  > asterisk boxes each with 8 E1 (voice only) as media gateway. What is
>  > your idea's, is it possible, it seems MGCP is for this case but
> aa more
>  > i look in asterisk MGCP channel less I get what's its usage. Any one
>  > could shed some lights on this topic, im somehow mixed up. How
> could I
>  > control the media gateway asterisks to tell them which channel has a
>  > call and where it should go?
> 
> Asterisk's chan_mgcp does not support being a media gateway right now.
> You can only control media gateways.  Also, Asterisk doesn't currently
> support signalling reception on anything other than a DS0 directly
> connected to the machine.
> 
> You're not going to be able to control media gateways either that are
> signalled via SS7, since chan_dahdi does not speak MGCP.
> 
> These are both things that I would like to improve though...
> 
> Matthew Fredrickson
> Digium, Inc.
> 
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-ss7
> 
> 
> 
> 
> 
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> 
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Re: [asterisk-ss7] Asterisk SS7, SG, MG and MGCP

[Anton]

As SS7 MGCP you probably mean MEGACO protocol? 
Unlikely this could be seen in asterisk anytime soon...

On Thursday 14 August 2008 11:11, voip me wrote:
> Dear Matthew,
>
> What do you think , how could one improve it then, you
> think chan_dahdi should speak MGCP or any request show
> come up to channel then bridged (or in some other way) to
> an MGCP channel (not the current but a changed MGCP
> channel) and the rest.
> I have another question Matthiew, are you familiar with
> MGCP, how could one address a channel on gateway?
>
> Regards.
> --
> MSH
>
> On 13/08/2008, Matthew Fredrickson <[EMAIL PROTECTED]> 
wrote:
> > voip me wrote:
> > > Dear Folks,
> > >
> > > Sorry for my ambigouse subject. I want to set the
> > > following solution and need your idea's and
> > > experinces. I want to setup an asterisk box with one
> > > E1 link containing 4 or 5 SS7 signalling links and
> > > have two or three asterisk boxes each with 8 E1
> > > (voice only) as media gateway. What is your idea's,
> > > is it possible, it seems MGCP is for this case but aa
> > > more i look in asterisk MGCP channel less I get
> > > what's its usage. Any one could shed some lights on
> > > this topic, im somehow mixed up. How could I control
> > > the media gateway asterisks to tell them which
> > > channel has a call and where it should go?
> >
> > Asterisk's chan_mgcp does not support being a media
> > gateway right now. You can only control media gateways.
> >  Also, Asterisk doesn't currently support signalling
> > reception on anything other than a DS0 directly
> > connected to the machine.
> >
> > You're not going to be able to control media gateways
> > either that are signalled via SS7, since chan_dahdi
> > does not speak MGCP.
> >
> > These are both things that I would like to improve
> > though...
> >
> > Matthew Fredrickson
> > Digium, Inc.
> >
> > ___
> > --Bandwidth and Colocation Provided by
> > http://www.api-digital.com--
> >
> > asterisk-ss7 mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-ss7

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Re: [asterisk-ss7] Asterisk SS7, SG, MG and MGCP

[voip me]

Dear Kristian,

Ofcourse your idea is a good solution too but not possible in my case. I
might have just 2 signalling links, master and alternate. :)

Regards.
--
MSH


On 13/08/2008, Kristian Nielsen <[EMAIL PROTECTED]> wrote:
>
> "voip me" <[EMAIL PROTECTED]> writes:
>
> > Sorry for my ambigouse subject. I want to set the following solution and
> need
> > your idea's and experinces. I want to setup an asterisk box with one E1
> link
> > containing 4 or 5 SS7 signalling links and have two or three asterisk
> boxes
> > each with 8 E1 (voice only) as media gateway. What is your idea's, is it
> > possible, it seems MGCP is for this case but aa more i look in asterisk
> MGCP
> > channel less I get what's its usage. Any one could shed some lights on
> this
> > topic, im somehow mixed up. How could I control the media gateway
> asterisks to
> > tell them which channel has a call and where it should go?
>
> Wouldn't it make more sense to have half the signalling links on one box,
> and
> half on the other? Otherwise whenever you need to do maintenance on the
> single
> box with the signalling links you will loose all of your 30*16 lines. You
> could instead set up 2 clustered chan_ss7 boxes each with 8 E1 and 2 or 3
> signalling links, and then when one box is down the other can take over.
>
> But maybe I misundestood what you are trying to achieve, as I am not
> familiar
> with MGCP...
>
> - Kristian.
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-ss7
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Re: [asterisk-ss7] Asterisk SS7, SG, MG and MGCP

[voip me]

Dear Matthew,

What do you think , how could one improve it then, you think chan_dahdi
should speak MGCP or any request show come up to channel then bridged (or in
some other way) to an MGCP channel (not the current but a changed MGCP
channel) and the rest.
I have another question Matthiew, are you familiar with MGCP, how could one
address a channel on gateway?

Regards.
--
MSH


On 13/08/2008, Matthew Fredrickson <[EMAIL PROTECTED]> wrote:
>
> voip me wrote:
> > Dear Folks,
> >
> > Sorry for my ambigouse subject. I want to set the following solution and
> > need your idea's and experinces. I want to setup an asterisk box with
> > one E1 link containing 4 or 5 SS7 signalling links and have two or three
> > asterisk boxes each with 8 E1 (voice only) as media gateway. What is
> > your idea's, is it possible, it seems MGCP is for this case but aa more
> > i look in asterisk MGCP channel less I get what's its usage. Any one
> > could shed some lights on this topic, im somehow mixed up. How could I
> > control the media gateway asterisks to tell them which channel has a
> > call and where it should go?
>
> Asterisk's chan_mgcp does not support being a media gateway right now.
> You can only control media gateways.  Also, Asterisk doesn't currently
> support signalling reception on anything other than a DS0 directly
> connected to the machine.
>
> You're not going to be able to control media gateways either that are
> signalled via SS7, since chan_dahdi does not speak MGCP.
>
> These are both things that I would like to improve though...
>
> Matthew Fredrickson
> Digium, Inc.
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-ss7
>
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Re: [asterisk-ss7] Asterisk SS7, SG, MG and MGCP

[Matthew Fredrickson]

voip me wrote:
> Dear Folks,
>  
> Sorry for my ambigouse subject. I want to set the following solution and 
> need your idea's and experinces. I want to setup an asterisk box with 
> one E1 link containing 4 or 5 SS7 signalling links and have two or three 
> asterisk boxes each with 8 E1 (voice only) as media gateway. What is 
> your idea's, is it possible, it seems MGCP is for this case but aa more 
> i look in asterisk MGCP channel less I get what's its usage. Any one 
> could shed some lights on this topic, im somehow mixed up. How could I 
> control the media gateway asterisks to tell them which channel has a 
> call and where it should go?

Asterisk's chan_mgcp does not support being a media gateway right now. 
You can only control media gateways.  Also, Asterisk doesn't currently 
support signalling reception on anything other than a DS0 directly 
connected to the machine.

You're not going to be able to control media gateways either that are 
signalled via SS7, since chan_dahdi does not speak MGCP.

These are both things that I would like to improve though...

Matthew Fredrickson
Digium, Inc.

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Re: [asterisk-ss7] Asterisk SS7, SG, MG and MGCP

[Kristian Nielsen]

"voip me" <[EMAIL PROTECTED]> writes:

> Sorry for my ambigouse subject. I want to set the following solution and need
> your idea's and experinces. I want to setup an asterisk box with one E1 link
> containing 4 or 5 SS7 signalling links and have two or three asterisk boxes
> each with 8 E1 (voice only) as media gateway. What is your idea's, is it
> possible, it seems MGCP is for this case but aa more i look in asterisk MGCP
> channel less I get what's its usage. Any one could shed some lights on this
> topic, im somehow mixed up. How could I control the media gateway asterisks to
> tell them which channel has a call and where it should go?

Wouldn't it make more sense to have half the signalling links on one box, and
half on the other? Otherwise whenever you need to do maintenance on the single
box with the signalling links you will loose all of your 30*16 lines. You
could instead set up 2 clustered chan_ss7 boxes each with 8 E1 and 2 or 3
signalling links, and then when one box is down the other can take over.

But maybe I misundestood what you are trying to achieve, as I am not familiar
with MGCP...

 - Kristian.

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[asterisk-ss7] Asterisk SS7, SG, MG and MGCP

[voip me]

Dear Folks,

Sorry for my ambigouse subject. I want to set the following solution and
need your idea's and experinces. I want to setup an asterisk box with one E1
link containing 4 or 5 SS7 signalling links and have two or three asterisk
boxes each with 8 E1 (voice only) as media gateway. What is your idea's, is
it possible, it seems MGCP is for this case but aa more i look in asterisk
MGCP channel less I get what's its usage. Any one could shed some lights on
this topic, im somehow mixed up. How could I control the media gateway
asterisks to tell them which channel has a call and where it should go?

Regards.
--
MSH
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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 37, Issue 14

[Dawid Kerad]

Hi Antoine,
> Message: 1
> Date: Mon, 17 Mar 2008 13:22:11 -0700 (PDT)
> From: Antoine Megalla <[EMAIL PROTECTED]>
> Subject: [asterisk-ss7] chan_ss7 1.0.9 "combined" example
> To: asterisk-ss7@lists.digium.com
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset=iso-8859-1
>
> I have a server running chan_ss7 1.0.9 and it has been solid for the last 3 
> weeks, I have to admit that version 1.0.9 is the most stable chan_ss7 
> release I have tried so far.
> [...]
>   
I'm also testing chan_ss7 and searching for most stable configuration,
so could You please send more informations about Your system,
like asterisk, zaptel, operating system (kernel) versions ...
thanks,
Dawid


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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 33, Issue 17

[Dmitri Sologoubenko]

[EMAIL PROTECTED] ha scritto:
> Send asterisk-ss7 mailing list submissions to
>   asterisk-ss7@lists.digium.com
> 
> To subscribe or unsubscribe via the World Wide Web, visit
>   http://lists.digium.com/mailman/listinfo/asterisk-ss7
> or, via email, send a message with subject or body 'help' to
>   [EMAIL PROTECTED]
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>   [EMAIL PROTECTED]
> 
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-ss7 digest..."
> 
> 
> Today's Topics:
> 
>1. Re: Strange problem with TE120P (asterisk & Zap-SS7)
>   (Matthew Fredrickson)
> 
> 
> --
> 
> Message: 1
> Date: Tue, 27 Nov 2007 13:24:43 -0600
> From: Matthew Fredrickson <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-ss7] Strange problem with TE120P (asterisk &
>   Zap-SS7)
> To: asterisk-ss7@lists.digium.com
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
> 
> Dmitri Sologoubenko wrote:
>> Hello, guys!
>>
>> I have recently installed trunk versions of asterisk-1.4, zaptel, libpri
>> and libss7. I Have also plugged in and configured a Digium's TE120P card.
>> ztcfg, zttool and other zaptel utilities seems to work ok, zaptel driver
>> modules seems to be correctly loaded and both zaptel and asterisk start
>> with no error messages.
>> But Zap channel type is not listed in asterisk CLI, no "zap ..." CLI
>> commands are available, and no Dial(Zap/...) works.
> 
> Typically when no zap commands are available, it means that chan_zap.so 
> didn't load.  You may check /var/log/asterisk/messages and figure out 
> what the error is when asterisk starts and tries to load chan_zap.
> 
> Matthew Fredrickson
> 
>> Can anybody help me?
>>
>> Thanks in advance,
>> D.S.
>>
>> Here's the output:
>>
>> *CLI> module show like zap
>> Module Description   Use
>> Count
>> chan_zap.soZapata Telephony
>> 0
>> codec_zap.so   Generic Zaptel Transcoder Codec Translat
>> 0
>> app_zapras.so  Zaptel ISDN Remote Access Server
>> 0
>> app_zapbarge.soBarge in on Zap channel application
>> 0
>> app_zapateller.so  Block Telemarketers with Special Informa
>> 0
>> app_zapscan.so Scan Zap channels application
>> 0
>> 6 modules loaded
>>
>> 
>>
>> *CLI> core show channeltypes
>> TypeDescription  Devicestate
>> Indications  Transfer
>> --  ---  ---
>> ---  
>> Agent   Call Agent Proxy Channel yes  yes
>>no
>> Phone   Standard Linux Telephony API Driver  no   yes
>>no
>> USTMUNISTIM Channel Driver   no   yes
>>no
>> Skinny  Skinny Client Control Protocol (Skinny)  yes  yes
>>no
>> IAX2Inter Asterisk eXchange Driver (Ver 2)   yes  yes
>>yes
>> Local   Local Proxy Channel Driver   yes  yes
>>no
>> Console OSS Console Channel Driver   no   yes
>>no
>> SIP Session Initiation Protocol (SIP)yes  yes
>>yes
>> MGCPMedia Gateway Control Protocol (MGCP)yes  yes
>>no
>> --y*CLI>
>> 9 channel drivers registered.
>>
>> 
>>
>> *CLI> core show channeltype zap
>> portability*CLI>
>> zap is not a registered channel driver.
>> Command 'core show channeltype zap' failed.
>>
>> 
>>
>>
>> # lsmod
>> Module  Size  Used by
>> zttranscode11784  0
>> wcte12xp   37696  1
>> zaptel187428  6 zttranscode,wcte12xp
>> ipv6  250369  14
>> dm_mirror  29713  0
>> dm_mod 56665  1 dm_mirror
>> video  19269  0
>> sbs18533  0
>> i2c_ec  9025  1 sbs
>> button 10705  0
>> battery13637  0
>> asus_acpi  19289  0
>> ac  9157  0
>> parport_pc 29157  0
>> lp 15849  0
>> parport37513  2 parport_pc,lp
>> sg 35933  0
>> 8139too28737  0
>> r8169  31561  0
>> mii 9409  1 8139too
>> i2c_i801   11469  0
>> pcspkr  7105  0
>> i2c_core   23745  2 i2c_ec,i2c_i801
>> serio_raw  10693  0
>> crc_ccitt   6337  1 zaptel
>> usb_storage76193  0
>> ata_piix   17609  14
>> libata 96857  1 ata_piix
>> sd_mod 22977  16
>> scsi_mod  130637  4 sg,usb_storage,libata,sd_mod
>> raid1  25153  6
>> ext3  123081  6
>> jbd  

Re: [asterisk-ss7] asterisk ss7

[Charl Barnard]

I agree-isn't it in the spirit of open source, given the great work done by
the developers up to this point, to re-submit your updates to the community
so everyone may benefit from improvements you may have made?
Ciao
Charl 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Anton
> Sent: 14 November 2007 07:18
> To: asterisk-ss7@lists.digium.com
> Subject: Re: [asterisk-ss7] asterisk ss7
> 
> Hi Hoai-Anh Ngo-Vi!
> 
> What did you port? Any code changes? improvements?
> 
> On Tuesday 13 November 2007, Hoai-Anh Ngo-Vi wrote:
> > Hi,
> >
> > I've ported chan_ss7 to use in production environment.
> >
> > With 8 E1s in Germany
> >
> > cheers
> > -Ursprüngliche Nachricht-
> > Von: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] Im Auftrag von marek 
> > cervenka Gesendet: Dienstag, 13. November 2007
> > 12:02
> > An: asterisk-ss7@lists.digium.com
> > Betreff: [asterisk-ss7] asterisk ss7
> >
> > hi,
> >
> > do you have someone:
> > - libSS7 in production use? how many concurrent channels?
> > - ported chan_ss7 to 1.4.13 and in production use? how many 
> concurrent 
> > channels?
> >
> > thanks
> >
> > ---
> > Marek Cervenka
> > ===
> >
> >
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Re: [asterisk-ss7] asterisk ss7

[Anton]

Hi Hoai-Anh Ngo-Vi!

What did you port? Any code changes? improvements?

On Tuesday 13 November 2007, Hoai-Anh Ngo-Vi wrote:
> Hi,
>
> I've ported chan_ss7 to use in production environment.
>
> With 8 E1s in Germany
>
> cheers
> -Ursprüngliche Nachricht-
> Von: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Im Auftrag
> von marek cervenka Gesendet: Dienstag, 13. November 2007
> 12:02
> An: asterisk-ss7@lists.digium.com
> Betreff: [asterisk-ss7] asterisk ss7
>
> hi,
>
> do you have someone:
> - libSS7 in production use? how many concurrent channels?
> - ported chan_ss7 to 1.4.13 and in production use? how
> many concurrent channels?
>
> thanks
>
> ---
> Marek Cervenka
> ===
>
>
> ___
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> http://www.api-digital.com--
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Re: [asterisk-ss7] asterisk ss7

[Hoai-Anh Ngo-Vi]

Hi,

I've ported chan_ss7 to use in production environment. 

With 8 E1s in Germany

cheers
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von marek cervenka
Gesendet: Dienstag, 13. November 2007 12:02
An: asterisk-ss7@lists.digium.com
Betreff: [asterisk-ss7] asterisk ss7

hi,

do you have someone:
- libSS7 in production use? how many concurrent channels?
- ported chan_ss7 to 1.4.13 and in production use? how many concurrent 
channels?

thanks

---
Marek Cervenka
===


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Re: [asterisk-ss7] asterisk ss7

[Darren O'Donohoe]

I am out of the office Wednesday 14th November

If you have a support query please call +353-1-4877000 and choose option 2 or 
e-mail [EMAIL PROTECTED]

If you have a sales or account query please contact Declan Murphy on 
+353-1-4877020 or [EMAIL PROTECTED]

Thanks & Regards
Darren

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[asterisk-ss7] asterisk ss7

[marek cervenka]

hi,

do you have someone:
- libSS7 in production use? how many concurrent channels?
- ported chan_ss7 to 1.4.13 and in production use? how many concurrent 
channels?

thanks

---
Marek Cervenka
===


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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 32, Issue 15

[Mostafa Ibrahim]

Thank You. But how to handle the signaling "How can I signal among boxes".
The signaling is coming on one channel.

Could You provide me with a sample configuration ?



On 10/22/07, Jorge Churio <[EMAIL PROTECTED]> wrote:
>
>  As far as I know, the only stack that supports bearer trunks spread on
> several boxes using one common link is chanSS7.
> Regards
>
> [EMAIL PROTECTED] wrote:
>
> Send asterisk-ss7 mailing list submissions to
>   asterisk-ss7@lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
>   http://lists.digium.com/mailman/listinfo/asterisk-ss7
> or, via email, send a message with subject or body 'help' to
>   [EMAIL PROTECTED]
>
> You can reach the person managing the list at
>   [EMAIL PROTECTED]
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-ss7 digest..."
>
>
> Today's Topics:
>
>1. didicated link for signaling (Mostafa Ibrahim)
>2. didicated channel for signaling (Mostafa Ibrahim)
>
>
> --
>
> Message: 1
> Date: Sun, 21 Oct 2007 17:05:38 +0200
> From: Mostafa Ibrahim <[EMAIL PROTECTED]> <[EMAIL PROTECTED]>
> Subject: [asterisk-ss7] didicated link for signaling
> To: asterisk-ss7@lists.digium.com
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="us-ascii"
>
> Dear All,
>
> I am using libss7. We have passed the initial tests with the ss7
> providers successfully with one ss7 link. we are having a much more
> bigger deployment.
>
> If I have ss7 on 30 E1 link distributed on more than one machine and the
> provider is sending the ss7 signaling on another dedicated E1 link. Can
> we have a solution for such deployment with asterisk and any ss7
> implementation ?
>
> Any help will be appreciated
> =
> Mostafa Ibrahim
> Security Department Manager
> ValueSyS
> website: http://www.valuesys.net
> Tel: +202 22682552 +202 2682887
> Fax: +202 22674346
> Mobile:+2 0181008194
> Email: [EMAIL PROTECTED]
> =
> -- next part --
> An HTML attachment was scrubbed...
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> http://lists.digium.com/pipermail/asterisk-ss7/attachments/20071021/1660bf9e/attachment-0001.htm
>
> --
>
> Message: 2
> Date: Sun, 21 Oct 2007 18:10:38 +0200
> From: Mostafa Ibrahim <[EMAIL PROTECTED]> <[EMAIL PROTECTED]>
> Subject: [asterisk-ss7] didicated channel for signaling
> To: asterisk-ss7@lists.digium.com
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="us-ascii"
>
> Dear All,
>
> Sorry. The previous question was not accurate.
>
> I am using libss7. We have passed the initial tests with the ss7
> providers successfully with one ss7 link. we are having a much more
> bigger deployment.
>
> If I have ss7 on 30 E1 link distributed on more than one machine and the
> provider is sending the ss7 signaling only one of these E1 links. Can we
> have a solution for such deployment with asterisk and any ss7
> implementation ?
>
> Any help will be appreciated
> =
> Mostafa Ibrahim
> Security Department Manager
> ValueSyS
> website: http://www.valuesys.net
> Tel: +202 22682552 +202 2682887
> Fax: +202 22674346
> Mobile:+2 0181008194
> Email: [EMAIL PROTECTED]
> =
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> End of asterisk-ss7 Digest, Vol 32, Issue 15
> 
>
>
> _ This
> information is private and confidential and intended for the recipient only.
> If you are not the intended recipient of this message you are hereby
> notified that any review, dissemination, distribution or copying of this
> message is strictly prohibited. This communication is for information
> purposes only and shall not be regarded neither as a proposal, acceptance
> nor as a statement of will or official statement from Globant. Email
> transmission cannot be guaranteed to be secure or error-free. Therefore, we
> do not represent that this information is complete or accurate and it should
> not be relied upon as such. All information is subject to change without
> notice.
>
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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 32, Issue 15

[Jorge Churio]

As far as I know, the only stack that supports bearer trunks spread on 
several boxes using one common link is chanSS7.

Regards

[EMAIL PROTECTED] wrote:

Send asterisk-ss7 mailing list submissions to
asterisk-ss7@lists.digium.com

To subscribe or unsubscribe via the World Wide Web, visit
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or, via email, send a message with subject or body 'help' to
[EMAIL PROTECTED]

You can reach the person managing the list at
[EMAIL PROTECTED]

When replying, please edit your Subject line so it is more specific
than "Re: Contents of asterisk-ss7 digest..."


Today's Topics:

   1. didicated link for signaling (Mostafa Ibrahim)
   2. didicated channel for signaling (Mostafa Ibrahim)


--

Message: 1
Date: Sun, 21 Oct 2007 17:05:38 +0200
From: Mostafa Ibrahim <[EMAIL PROTECTED]>
Subject: [asterisk-ss7] didicated link for signaling
To: asterisk-ss7@lists.digium.com
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"

Dear All,

I am using libss7. We have passed the initial tests with the ss7
providers successfully with one ss7 link. we are having a much more
bigger deployment.

If I have ss7 on 30 E1 link distributed on more than one machine and the
provider is sending the ss7 signaling on another dedicated E1 link. Can
we have a solution for such deployment with asterisk and any ss7
implementation ?

Any help will be appreciated
=
Mostafa Ibrahim 
Security Department Manager

ValueSyS
website: http://www.valuesys.net 
Tel: +202 22682552 +202 2682887

Fax: +202 22674346
Mobile:+2 0181008194
Email: [EMAIL PROTECTED]
=
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Message: 2
Date: Sun, 21 Oct 2007 18:10:38 +0200
From: Mostafa Ibrahim <[EMAIL PROTECTED]>
Subject: [asterisk-ss7] didicated channel for signaling
To: asterisk-ss7@lists.digium.com
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"

Dear All,

Sorry. The previous question was not accurate.

I am using libss7. We have passed the initial tests with the ss7
providers successfully with one ss7 link. we are having a much more
bigger deployment.

If I have ss7 on 30 E1 link distributed on more than one machine and the
provider is sending the ss7 signaling only one of these E1 links. Can we
have a solution for such deployment with asterisk and any ss7
implementation ?

Any help will be appreciated
=
Mostafa Ibrahim 
Security Department Manager

ValueSyS
website: http://www.valuesys.net 
Tel: +202 22682552 +202 2682887

Fax: +202 22674346
Mobile:+2 0181008194
Email: [EMAIL PROTECTED]
=
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End of asterisk-ss7 Digest, Vol 32, Issue 15

  


_
This information is private and confidential and intended for the recipient
only. If you are not the intended recipient of this message you are hereby
notified that any review, dissemination, distribution or copying of this
message is strictly prohibited. This communication is for information
purposes only and shall not be regarded neither as a proposal, acceptance
nor as a statement of will or official statement from Globant. Email
transmission cannot be guaranteed to be secure or error-free. Therefore, we
do not represent that this information is complete or accurate and it should
not be relied upon as such. All information is subject to change without
notice.
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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 31, Issue 13

[Jorge Churio]

Dear TT,

Telco bypass means a way traffic is transported avoiding telco networks 
using instead IP over broadband links.
It is not directly related to the protocol you use to interface with 
PSTN but with the fact that making interconnections with telcos in each 
end of the road, you avoid passing thru Telco's TDM infraestrucure and 
hence paying them the "toll" for long distance charges.
Actually, you can use SS7, ISDN, or other protocols to interact with Telcos.

About feasibility, it depends of how the telecom market is regulated in 
each country, in some places Government regulators forbide  any bypass, 
in others (like most of America´s countries, for instance) if you make 
the bypass for your internal usage (among branches of the company, not 
involving usage of telephone networks in any side) you can do it, in 
thos elast cases, if you owns the bypass network and wants to resell 
this service for customers (as calling card oerator does, for instance) 
you need a Telecom Licence, which is easy or difficult to get depending 
each country.

By the way, I also have Russel´s book and I really recommend it for a 
deeper understanding of SS7,  we use in our Company to make an 
ITU-compliant SS7 stack for Asterisk, but if you are studying the how 
the bypass works, I would recommend you some more general reading such 
as Voip Fundamentals from Cisco Press (specially the first chapters) or 
many others around.

Regards.

Jorge Churio
CEO
Fonalix by BGH

[EMAIL PROTECTED] wrote:
> Send asterisk-ss7 mailing list submissions to
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>   [EMAIL PROTECTED]
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> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-ss7 digest..."
>
>
> Today's Topics:
>
>1. telco bypass feasibility ([EMAIL PROTECTED])
>
>
> --
>
> Message: 1
> Date: Sat, 29 Sep 2007 05:34:32 -0700 (PDT)
> From: [EMAIL PROTECTED]
> Subject: [asterisk-ss7] telco bypass feasibility
> To: asterisk-ss7@lists.digium.com
> Message-ID:
>   <[EMAIL PROTECTED]>
> Content-Type: text/plain;charset=iso-8859-1
>
> greetings ss7 users,
>
> question(s): can anyone on this list refer me to articles or feasibility
> studies on the use of SS7 products that enable organizations
> (medium-to-large companies, colleges, small villages and other
> municipalities) to connect their voip networks directly to the PSTN
> without having to be telco customer? I've heard that telco bypass is done
> in countries outside the US. Is there a technical or legal reason for
> that? Can it be done in the US? Which products from companies like digium
> and sangoma enable telco bypass, and what general architectures are these
> products found in?
>
> background: i'm preparing a feasibility study on telco bypass. i'm
> studying two books: travis russell's "Signaling System #7" and Frank
> Ohrtman's "Softswitch: Architecture for VoIP". besides being a solution to
> insomnia (the reading is so intense it tends to make one fall asleep!),
> these books have opened my eyes to a world of possibilities. but what is
> possible is not always feasible, so i'm posting to this list.
>
> -- TIA, TT
>
>
>
>
> --
>
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> End of asterisk-ss7 Digest, Vol 31, Issue 13
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[asterisk-ss7] Asterisk - SS7 Forum

[Mitul Limbani]

Hello fella members,

I have created a forum on : http://asterisk.pbx.in/asterisk-ss7 which 
can be used for discussion purpose.

If anyone has any comment on the same, please contact me directly.

Thanks & Regards,
Mitul Limbani,
Founder & CEO,
Enterux Solutions,
The Enterprise Linux Company (TM),
www.enterux.com

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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 30, Issue 3

[Zurab Rukhadze]

Must be:
cicbeginswith = 2
channel = 2-31
channel = 32-62

Zurab Rukhadze

- Original Message - 
From: <[EMAIL PROTECTED]>
To: 
Sent: Friday, August 03, 2007 10:00 AM
Subject: asterisk-ss7 Digest, Vol 30, Issue 3


> Send asterisk-ss7 mailing list submissions to
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>
> Today's Topics:
>
>1. Re: Errors on a SS7-Link with libss7 (Matthew Fredrickson)
>2. Re: Errors on a SS7-Link with libss7 (Darren O'Donohoe)
>
>
> --
>
> Message: 1
> Date: Fri, 03 Aug 2007 10:59:49 -0500
> From: Matthew Fredrickson <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-ss7] Errors on a SS7-Link with libss7
> To: asterisk-ss7@lists.digium.com
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Marc Storck wrote:
> >>> I have
> >>>
> >>> cicbeginswith = 1
> >>> channel = 2-31
> >>> channel = 32-62
> >>>
> >>> anything wrong with that?
> >> It really depends on how the other end has it set up.  Make sure your
> >> CICs map to the same DS1s and DS0s as the other end's CICs.  If you
> >> don't hear audio, it means that you have it wrong :-)
> >
> > I have checked this back and forth, but it did not help and there is
> > just no debug output. How can I get some debug output like in chan_ss7
> > (which is complaining instantly about CIC numbering issues e.g. received
> > CIC # 4 expecting #3)?
> >
>
> One way to do it is to bring up a call, then ztmonitor the zap channels
> until you find the one that the audio is on.  Then you know that
> whatever cic that came in on, that's the zap channel it corresponds to.
>
> -- 
> Matthew Fredrickson
> Software/Firmware Engineer
> Digium, Inc.
>
>
>
> --
>
> Message: 2
> Date: Fri, 03 Aug 2007 17:02:09 +0100
> From: Darren O'Donohoe <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-ss7] Errors on a SS7-Link with libss7
> To: 
> Message-ID: <[EMAIL PROTECTED]>
>
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Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 29, Issue 16

[sai jayram AKV]

Hi Aravind.

If you are running libss7/chanss7 on 8260 proc with embedded linux, you
might need to check on endianness. Also, you might try disabling fisu
transmission under idealcondition from userspace as ppc can transmit fisu
after msu.

I had similar issue and got rectified by a header file change.

Rgds
sai




On 7/31/07, [EMAIL PROTECTED] <
[EMAIL PROTECTED]> wrote:
>
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>[EMAIL PROTECTED]
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> than "Re: Contents of asterisk-ss7 digest..."
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>
> Today's Topics:
>
>   1. Re: asterisk-ss7 Digest, Vol 29, Issue 15 (Arvind Kumar)
>   2. Re: asterisk-ss7 Digest, Vol 29, Issue 15 (kiran)
>   3. Re: Errors on a SS7-Link with libss7 (Marc Storck)
>
>
> --
>
> Message: 1
> Date: Tue, 31 Jul 2007 14:17:44 +0530
> From: "Arvind Kumar" <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-ss7] asterisk-ss7 Digest, Vol 29, Issue 15
> To: asterisk-ss7@lists.digium.com
> Message-ID:
><[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi Matthew!
> I am running my SG  (Signalling Gateway ) Stack on Power PC 8260 Processor
> (Embedded Linux).
> OS is embedded Linux.
> Signalling link is up between SG Statck and another SS7 Stack.
> When I run some load through SS7 Stack, after some time the Signalling
> Link
> goes down.
> I found the problem, the problem is happening at MTP2 level.
> The link is failing due to receiving of incorrect FIB (Forward Indicator
> Bit
> ) and / or BSN (Backward Sequence Number) at MTP2 Level.
> According to recommendation Link can go down.
> Now my query is, what can be the problem, is it on my end or remote end,
> and
> how can it be resolved if possible.
>
> Thanks in Advance
> Regards,
> Arvind
>
>
>
>
>
>
>
>
>
>
>
> On 7/30/07, [EMAIL PROTECTED] <
> [EMAIL PROTECTED]> wrote:
> >
> > Send asterisk-ss7 mailing list submissions to
> > asterisk-ss7@lists.digium.com
> >
> > To subscribe or unsubscribe via the World Wide Web, visit
> > http://lists.digium.com/mailman/listinfo/asterisk-ss7
> > or, via email, send a message with subject or body 'help' to
> > [EMAIL PROTECTED]
> >
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> > [EMAIL PROTECTED]
> >
> > When replying, please edit your Subject line so it is more specific
> > than "Re: Contents of asterisk-ss7 digest..."
> >
> >
> > Today's Topics:
> >
> >1. SS7 Link Failure. (Arvind Kumar)
> >2. Re: SS7 Link Failure. (Matthew Fredrickson)
> >
> >
> > --
> >
> > Message: 1
> > Date: Mon, 30 Jul 2007 19:22:16 +0530
> > From: "Arvind Kumar" <[EMAIL PROTECTED]>
> > Subject: [asterisk-ss7] SS7 Link Failure.
> > To: asterisk-ss7@lists.digium.com
> > Message-ID:
> > <[EMAIL PROTECTED]>
> > Content-Type: text/plain; charset="iso-8859-1"
> >
> > Hi!
> > Could you please help me.
> > I running load on SS7 Signalling Link. After some time the link is going
> > down,
> > I identified it is going due to abnormal FIB, 2 out of 3 FIB are not
> > proper.
> > It is ok according to
> > ITU recommendation,
> > I want to resolve this problem. I want to know whether this is problem
> > with
> > my stack or another side stack or something else like clock etc. Kindly
> > help.
> >
> > Thanks &
> > With Best Regards,
> > Arvind
> > -- next part --
> > An HTML attachment was scrubbed...
> > URL:
> >
> http://lists.digium.com/pipermail/asterisk-ss7/attachments/20070730/fb275c72/attachment-0001.htm
> >
> > --
> >
> > Message: 2
> > Date: Mon, 30 Jul 2007 09:49:36 -0500
> > From: Matthew Fredrickson <[EMAIL PROTECTED]>
> > Subject: Re: [asterisk-ss7] SS7 Link Failure.
> > To: asterisk-ss7@lists.digium