[asterisk-ss7] Version 0.8.2 of chan_ss7 for asterisk
atik khan
atik.khan at gmail.com
Thu Mar 9 20:54:56 MST 2006
hello,
i have downloaded chan_ss7 by changing the DNS server on my LAN.
I use US Based DNS server, it works for me.
If some one still have Problem with downloading then just send me a
mail, i will attach the file.
Thanks
atik
On 3/10/06, asterisk-ss7-request at lists.digium.com
<asterisk-ss7-request at lists.digium.com> wrote:
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> Today's Topics:
>
> 1. Re: Version 0.8.2 of chan_ss7 for asterisk released
> (leonimar cape)
> 2. Re: Version 0.8.2 of chan_ss7 for asterisk released
> (Matt Riddell [NZ])
> 3. Re: [help]no ringback tone and deteriorating audio...
> (leonimar cape)
> 4. Re: Version 0.8.2 of chan_ss7 for asterisk released (Atif Rasheed)
> 5. Re: Version 0.8.2 of chan_ss7 for asterisk released (Soren Rathje)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Wed, 8 Mar 2006 16:42:51 -0800 (PST)
> From: leonimar cape <leo_mac_ph at yahoo.com>
> Subject: Re: [asterisk-ss7] Version 0.8.2 of chan_ss7 for asterisk
> released
> To: asterisk-ss7 at lists.digium.com
> Message-ID: <20060309004251.86388.qmail at web60013.mail.yahoo.com>
> Content-Type: text/plain; charset=iso-8859-1
>
> Hi,
>
> Is there any problem in the download link? I cannot
> access it.
>
> Thanks,
>
> --- Anders Baekgaard <ab at sifira.com> wrote:
>
> > An updated version of chan_ss7 for asterisk has been
> > released as version
> > 0.8.2.
> >
> > New in version 0.8.2
> > - Handling of iSUP suspend/resume
> >
> > New in version 0.8.1
> > - Introduced subservice configuration option for
> > linksets.
> > - Fixed bug that causes crash when received CGU and
> > other circuit group
> > messages
> > - Fixed bug that causes repeat warning when doing
> > continuity check
> > - Fixed a problem with the initial alignment
> > procedure
> > - Tested with asterisk version 1.2.4.
> >
> > Additional information and the source of chan_ss7
> > can be found at
> > http://www.sifira.dk/chan-ss7.
> >
> > Best regards
> > Anders Baekgaard
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com
> > --
> >
> > asterisk-ss7 mailing list
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> >
> >
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> ------------------------------
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> Message: 2
> Date: Thu, 09 Mar 2006 20:10:31 +1300
> From: "Matt Riddell [NZ]" <matt.riddell at sineapps.com>
> Subject: Re: [asterisk-ss7] Version 0.8.2 of chan_ss7 for asterisk
> released
> To: asterisk-ss7 at lists.digium.com
> Message-ID: <440FD4E7.8070301 at sineapps.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> leonimar cape wrote:
> > Hi,
> >
> > Is there any problem in the download link? I cannot
> > access it.
>
> Down from here too.
>
> --
> Cheers,
>
> Matt Riddell
> _______________________________________________
>
> http://www.sineapps.com/news.php (Daily Asterisk News - html)
> http://freevoip.gedameurope.com (Free Asterisk Voip Community)
> http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
>
>
> ------------------------------
>
> Message: 3
> Date: Wed, 8 Mar 2006 23:32:47 -0800 (PST)
> From: leonimar cape <leo_mac_ph at yahoo.com>
> Subject: Re: [asterisk-ss7] [help]no ringback tone and deteriorating
> audio...
> To: asterisk-ss7 at lists.digium.com
> Message-ID: <20060309073247.97599.qmail at web60024.mail.yahoo.com>
> Content-Type: text/plain; charset=iso-8859-1
>
> Hi,
>
> I am now testing the chan_ss7 v 0.8.2 and the SUS and
> RES works great, Thanks to the Sifira guys!!!! :)
>
> But I still have a problem with the ring back tone
> when the caller is originating from the pstn side. I
> can see that asterisk is sending ALERT_CALL in
> progress but it seems that the Nortel switch which I
> am connected cannot see it. I even try to apply patch
> for the RBT posted on the voip-info but to no avail.
> Any about this scenario? Also I get Buffer warnings,
> is a way to omit this things.
>
> Here is the tarce I got.
>
> Mar 9 02:15:31 WARNING[10460] mtp.c: Excessive poll
> delay 10226!
> Mar 9 02:15:31 VERBOSE[10902] logger.c: --
> Executing AGI("SS7/31", "fixlocalprefix") in new stack
> Mar 9 02:15:31 VERBOSE[10902] logger.c: --
> Launched AGI Script
> /var/lib/asterisk/agi-bin/fixlocalprefix
> Mar 9 02:15:31 WARNING[10460] mtp.c: Full Zaptel
> input buffer detected, incoming packets may have been
> lost on link 'l1'.
> Mar 9 02:15:31 WARNING[10460] mtp.c: Empty Zaptel
> output buffer detected, outgoing packets may have been
> lost on link 'l1'.
> Mar 9 02:15:31 WARNING[10460] mtp.c: Excessive poll
> delay 9520!
> Mar 9 02:15:31 NOTICE[10460] mtp.c: MTP2 bitstream
> frame format error, entering octet counting mode on
> link 'l1'.
> Mar 9 02:15:31 DEBUG[10460] mtp.c: Last raw bits on
> link 'l1': f0 bf 2a ba 80 38 f8 5f 95 5d 40 1c 7c 2f
> ca ae a0 0e 3e 17 e5 57 50 07 1f 0b f2 ab 5d 40 1c 7c
> 2f ca ae a0
> Mar 9 02:15:31 WARNING[10460] mtp.c: Full Zaptel
> input buffer detected, incoming packets may have been
> lost on link 'l1'.
> Mar 9 02:15:31 WARNING[10460] mtp.c: Empty Zaptel
> output buffer detected, outgoing packets may have been
> lost on link 'l1'.
> Mar 9 02:15:31 WARNING[10460] mtp.c: Excessive poll
> delay 9453!
> Mar 9 02:15:31 NOTICE[10460] mtp.c: MTP2 bitstream
> frame format error, entering octet counting mode on
> link 'l1'.
> Mar 9 02:15:31 DEBUG[10460] mtp.c: Last raw bits on
> link 'l1': 2f ca ae a0 0e 3e 17 e5 57 50 07 1f 0b f2
> ab a8 03 8f 85 f9 55 d4 01 c7 c2 fc aa ea 50 07 1f 0b
> f2 ab a8 03
> Mar 9 02:15:31 WARNING[10460] mtp.c: Full Zaptel
> input buffer detected, incoming packets may have been
> lost on link 'l1'.
> Mar 9 02:15:31 WARNING[10460] mtp.c: Empty Zaptel
> output buffer detected, outgoing packets may have been
> lost on link 'l1'.
> Mar 9 02:15:31 NOTICE[10460] mtp.c: MTP2 bitstream
> frame format error, entering octet counting mode on
> link 'l1'.
> Mar 9 02:15:31 DEBUG[10460] mtp.c: Last raw bits on
> link 'l1': 1f 0b f2 ab a8 03 8f 85 f9 55 d4 01 c7 c2
> fc aa ea 00 e3 e1 7e 55 75 00 71 f0 bf 2a ba 80 c7 c2
> fc aa ea 00
> Mar 9 02:15:31 VERBOSE[10902] logger.c: -- AGI
> Script fixlocalprefix completed, returning 0
> Mar 9 02:15:31 VERBOSE[10902] logger.c: --
> Executing SetVar("SS7/31", "OUTNUM=6326945113") in new
> stack
> Mar 9 02:15:31 VERBOSE[10902] logger.c: --
> Executing Cut("SS7/31", "custom=OUT_5|:|1") in new
> stack
> Mar 9 02:15:31 WARNING[10902] ast_expr2.y:
> non-numeric argument
> Mar 9 02:15:31 DEBUG[10902] pbx.c: Expression result
> is '0'
> Mar 9 02:15:31 VERBOSE[10902] logger.c: --
> Executing GotoIf("SS7/31", "0?16") in new stack
> Mar 9 02:15:31 DEBUG[10902] pbx.c: Not taking any
> branch
> Mar 9 02:15:31 VERBOSE[10902] logger.c: --
> Executing Dial("SS7/31", "IAX2/mg2prod/6326945113") in
> new stack
> Mar 9 02:15:31 VERBOSE[10902] logger.c: -- Called
> mg2prod/6326945113
> Mar 9 02:15:31 WARNING[10460] mtp.c: Excessive poll
> delay 9336!
> Mar 9 02:15:31 WARNING[10460] mtp.c: Full Zaptel
> input buffer detected, incoming packets may have been
> lost on link 'l1'.
> Mar 9 02:15:31 WARNING[10460] mtp.c: Empty Zaptel
> output buffer detected, outgoing packets may have been
> lost on link 'l1'.
> Mar 9 02:15:31 NOTICE[10460] mtp.c: MTP2 bitstream
> frame format error, entering octet counting mode on
> link 'l1'.
> Mar 9 02:15:31 DEBUG[10460] mtp.c: Last raw bits on
> link 'l1': 5d 40 1c 7c 2f ca ae a0 0e 3e 17 e5 57 50
> 07 1f 0b f2 ab a8 03 8f 85 f9 55 d4 01 c7 c2 fc 8f 85
> f9 55 d4 01
> Mar 9 02:15:31 VERBOSE[10448] logger.c: -- Call
> accepted by 202.58.255.130 (format alaw)
> Mar 9 02:15:31 VERBOSE[10448] logger.c: -- Format
> for call is alaw
> Mar 9 02:15:31 VERBOSE[10902] logger.c: --
> IAX2/mg2prod-1 is ringing
> Mar 9 02:15:31 DEBUG[10902] chan_ss7.c: SS7 indicate
> CIC=31.
> Mar 9 02:15:31 DEBUG[10902] chan_ss7.c: Sending
> ALERTING call progress for CIC=31..
> Mar 9 02:15:31 DEBUG[10460] mtp.c: Queue MSU, lsi=0,
> last_send_ix=0, linkset=siuc, m->link=l1
> Mar 9 02:15:31 DEBUG[10460] mtp.c: Queue MSU, lsi=0,
> last_send_ix=0, linkset=siuc, m->link=l1
> Mar 9 02:15:31 DEBUG[10460] mtp.c: Sending buffer to
> zaptel len=14, on link 'l1'.
> Mar 9 02:15:31 DEBUG[10460] mtp.c: Sending buffer to
> zaptel len=21, on link 'l1'.
> Mar 9 02:15:32 VERBOSE[10902] logger.c: --
> IAX2/mg2prod-1 is ringing
> Mar 9 02:15:34 DEBUG[10906] manager.c: Manager
> received command 'login'
> Mar 9 02:15:34 VERBOSE[10906] logger.c: == Parsing
> '/etc/asterisk/manager.conf': Mar 9 02:15:34
> VERBOSE[10906] logger.c: == Parsing
> '/etc/asterisk/manager.conf': Found
>
> Thanks in advance! :)
>
> Leonimar Cape
>
>
>
> --- ryan nalupa <ryanalupa at yahoo.com.ph> wrote:
>
> > march 8, 2006
> >
> > hi all, i'm ryan and i've just joined this mailing
> > list. i've tried installing chan_ss7 version 0.8.1
> > on my two servers. followed the step-by-step setup
> > to test e1 cards with ss7 signalling that i found at
> > voip-info.org. i was hoping somebody can help me
> > with my problem here.
> > i've setup a master server with te411p in it and a
> > slave with te110p inside it. used centos 4.2 x64 and
> > asterisk 1.2.5 and zaptel 1.2.4. my problem is when
> > i dial i can't hear a ringing tone from my handset
> > but the dialed fone rings. at first i thought of the
> > codec, btw i'm using alaw as my primary codec, tried
> > using ilbc but i'm having echo at the callee side.
> > also that when the call is going on for already
> > about past 5 minutes, the audio deteriorates, it
> > becomes choppy and i'm having these debug messages
> > coming out from my master server.
> >
> > Mar 8 13:15:06 NOTICE[10692] chan_ss7.c: Write
> > buffer full on CIC=30 (wrote only 0 of 160), audio
> > lost.
> > Mar 8 13:15:06 NOTICE[10692] chan_ss7.c: Write
> > buffer full on CIC=30 (wrote only 0 of 160), audio
> > lost.
> > Mar 8 13:15:07 NOTICE[10692] chan_ss7.c: Write
> > buffer full on CIC=30 (wrote only 0 of 160), audio
> > lost.
> > Mar 8 13:15:07 NOTICE[10692] chan_ss7.c: Write
> > buffer full on CIC=30 (wrote only 0 of 160), audio
> > lost.
> > Mar 8 13:15:07 NOTICE[10692] chan_ss7.c: Write
> > buffer full on CIC=30 (wrote only 0 of 160), audio
> > lost.
> > Mar 8 13:15:08 NOTICE[10692] chan_ss7.c: Write
> > buffer full on CIC=30 (wrote only 0 of 160), audio
> > lost.
> > Mar 8 13:15:08 NOTICE[10692] chan_ss7.c: Write
> > buffer full on CIC=30 (wrote only 0 of 160), audio
> > lost.
> > Mar 8 13:15:09 NOTICE[10692] chan_ss7.c: Write
> > buffer full on CIC=30 (wrote only 0 of 160), audio
> > lost.
> > Mar 8 13:15:09 NOTICE[10692] chan_ss7.c: Write
> > buffer full on CIC=30 (wrote only 0 of 160), audio
> > lost.
> > Mar 8 13:15:09 NOTICE[10692] chan_ss7.c: Write
> > buffer full on CIC=30 (wrote only 0 of 160), audio
> > lost.
> > Mar 8 13:15:09 NOTICE[10692] chan_ss7.c: Write
> > buffer full on CIC=30 (wrote only 0 of 160), audio
> > lost.
> > Mar 8 13:15:09 NOTICE[10692] chan_ss7.c: Write
> > buffer full on CIC=30 (wrote only 0 of 160), audio
> > lost.
> > Mar 8 13:15:09 NOTICE[10692] chan_ss7.c: Write
> > buffer full on CIC=30 (wrote only 0 of 160), audio
> > lost.
> > Mar 8 13:15:09 NOTICE[10692] chan_ss7.c: Write
> > buffer full on CIC=30 (wrote only 0 of 160), audio
> > lost.
> > Mar 8 13:15:09 NOTICE[10692] chan_ss7.c: Write
> > buffer full on CIC=30 (wrote only 0 of 160), audio
> > lost.
> > Mar 8 13:15:09 NOTICE[10692] chan_ss7.c: Write
> > buffer full on CIC=30 (wrote only 0 of 160), audio
> > lost.
> > Mar 8 13:15:09 NOTICE[10692] chan_ss7.c: Write
> > buffer full on CIC=30 (wrote only 0 of 160), audio
> > lost.
> > Mar 8 13:15:09 NOTICE[10692] chan_ss7.c: Write
> > buffer full on CIC=30 (wrote only 0 of 160), audio
> > lost.
> > Mar 8 13:15:09 NOTICE[10692] chan_ss7.c: Write
> > buffer full on CIC=30 (wrote only 0 of 160), audio
> > lost.
> >
> > can anyone help me work this out or just direct me
> > to something i can read on? thanks in advance!
> >
> > regards,
> >
> > ryan
> >
> >
> > ---------------------------------
> > Do you Yahoo!?
> > Try the new Yahoo! Philippines Front Page!>
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> > --
> >
> > asterisk-ss7 mailing list
> > To UNSUBSCRIBE or update options visit:
> >
> >
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> ------------------------------
>
> Message: 4
> Date: Thu, 09 Mar 2006 13:13:10 +0500
> From: Atif Rasheed <atif at iphonica.com>
> Subject: Re: [asterisk-ss7] Version 0.8.2 of chan_ss7 for asterisk
> released
> To: asterisk-ss7 at lists.digium.com
> Message-ID: <440FE396.9090306 at iphonica.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> well, me never found it up as well.
>
> Matt Riddell [NZ] wrote:
>
> >leonimar cape wrote:
> >
> >
> >>Hi,
> >>
> >>Is there any problem in the download link? I cannot
> >>access it.
> >>
> >>
> >
> >Down from here too.
> >
> >
> >
>
>
>
> ------------------------------
>
> Message: 5
> Date: Thu, 9 Mar 2006 09:50:56 +0100
> From: "Soren Rathje" <asterisk at lolle.org>
> Subject: Re: [asterisk-ss7] Version 0.8.2 of chan_ss7 for asterisk
> released
> To: <asterisk-ss7 at lists.digium.com>
> Message-ID: <006401c64356$975c87f0$baf6d7c3 at soren>
> Content-Type: text/plain; charset="Windows-1252"
>
> Matt Riddell [NZ] wrote:
> > leonimar cape wrote:
> >> Hi,
> >>
> >> Is there any problem in the download link? I cannot
> >> access it.
> >
> > Down from here too.
>
> Works for me... (Denmark)
>
> /Soren
>
>
>
> ------------------------------
>
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