[asterisk-ss7] [help]no ringback tone and deteriorating audio...

leonimar cape leo_mac_ph at yahoo.com
Thu Mar 9 00:32:47 MST 2006


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
> 
> 		
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