Calls to long distance get disconnected before answer.
Telco: Alestra
Country: Mexico
System: Elastix 2.2
Digital Card: Digium TE122

Log:

[Dec 28 14:37:44] VERBOSE[4586] pbx.c: — Executing [+525552622900@default:1] Set(“SIP/OCS_TRUNK-000001bf”, “EXT=015552622900″) in new stack
[Dec 28 14:37:44] VERBOSE[4586] pbx.c: — Executing [+525552622900@default:2] Dial(“SIP/OCS_TRUNK-000001bf”, “DAHDI/g1/015552622900,60″) in new stack
[Dec 28 14:37:44] VERBOSE[4586] app_dial.c: — Called DAHDI/g1/015552622900
[Dec 28 14:37:44] DEBUG[4586] chan_dahdi.c: bits changed in chan 1
[Dec 28 14:37:53] DEBUG[4586] chan_dahdi.c: disconnecting MFC/R2 call on chan 1
[Dec 28 14:37:53] DEBUG[4586] chan_dahdi.c: ast cause 0 resulted in openr2 cause 6/Normal Clearing
[Dec 28 14:37:53] VERBOSE[4586] chan_dahdi.c: — Hungup ‘DAHDI/1-1′
[Dec 28 14:37:53] VERBOSE[4586] pbx.c: == Spawn extension (default, +525552622900, 2) exited non-zero on ‘SIP/OCS_TRUNK-000001bf’
[Dec 28 14:37:53] VERBOSE[9190] chan_dahdi.c: MFC/R2 call end on channel 1

Found this email list, but I think is too old.

http://www.mail-archive.com/asterisk-users@lists.digium.com/msg205765.html

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Hi list,

I have installed SIPp into my server. But not able to used it properly.
how to configure with my server ? how to see logs on webpage ?
how to start call testing ….

when i start SIPp then found verious hits on myserver.

*CLI:- *
[Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
Call from ” to extension ‘service’ rejected because extension not found in
context ‘default’.
== Using SIP RTP CoS mark 5
[Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
Call from ” to extension ‘service’ rejected because extension not found in
context ‘default’.
== Using SIP RTP CoS mark 5
[Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
Call from ” to extension ‘service’ rejected because extension not found in
context ‘default’.
== Using SIP RTP CoS mark 5
[Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
Call from ” to extension ‘service’ rejected because extension not found in
context ‘default’.
== Using SIP RTP CoS mark 5
[Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
Call from ” to extension ‘service’ rejected because extension not found in
context ‘default’.
== Using SIP RTP CoS mark 5
[Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
Call from ” to extension ‘service’ rejected because extension not found in
context ‘default’.
== Using SIP RTP CoS mark 5
[Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
Call from ” to extension ‘service’ rejected because extension not found in
context ‘default’.
== Using SIP RTP CoS mark 5
[Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
Call from ” to extension ‘service’ rejected because extension not found in
context ‘default’.
== Using SIP RTP CoS mark 5
[Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
Call from ” to extension ‘service’ rejected because extension not found in
context ‘default’.
haddock8-astrx*CLI>

After upgrading one of my server to asterisk 1.8.7.2 (the older is running 1.4.39)

When I try to dialin on asterisk-1.4.39 I get an error:
NOTICE[2414]: chan_iax2.c:9541 socket_process: Rejected connect attempt from 192.168.141.8, requested/capability 0×2/0×703 incompatible with our
capability 0xc.
NOTICE[2417]: chan_iax2.c:9541 socket_process: Rejected connect attempt from 192.168.141.8, requested/capability 0×2/0×703
incompatible with our capability 0xc.

On asterisk-1.8.7 I get:
WARNING[4277]: chan_iax2.c:10666 socket_process: Call rejected by 192.168.141.1: Unable to negotiate codec

I’m using ulaw / alaw code; why don’t they communicate?

iax.conf (1.4.39)
[home_server]
disallow=all
allow=ulaw
allow=alaw

iax.conf (1.8.7)
[clinic_server]
disallow=all
allow=ulaw
allow=alaw
requirecalltoken=auto

Hello list,

An Asterisk installation that was doing fine suddenly stared segfaulting
a couple of times per day. I enabled all the logging and debugging to
try to find a pattern but there was too much information to see exactly
where it broke. So I enabled core dump and did backtraces and all of
them seem to break on ast_setstate, setting the state to AST_STATE_DOWN.
That’s pretty much the only thing I can make of it, don’t even know if
that’s correct.

Does anyone have any ideas on why this is happening? The backtrace is
attached.

P.S.: I’ve switched the whole hardware already, including the BRI card
(B400P, OpenVox). Also tried different versions of Asterisk, Dahdi and
mISDN. I’m stuck with 1.4 Asterisk branch and mISDN v1.

Best regards,
Paulo Santos

Core was generated by `/usr/sbin/asterisk’.
Program terminated with signal 11, Segmentation fault.
[New process 21726]
[New process 24376]
[New process 24375]
[New process 24374]
[New process 24371]
[New process 24344]
[New process 23560]
[New process 22868]
[New process 22329]
[New process 22327]
[New process 22325]
[New process 22324]
[New process 22323]
[New process 22322]
[New process 22321]
[New process 22320]
[New process 22319]
[New process 22318]
[New process 22317]
[New process 22316]
[New process 22315]
[New process 22259]
[New process 22208]
[New process 22203]
[New process 22185]
[New process 22184]
[New process 22160]
[New process 21515]
[New process 21725]
[New process 21687]
[New process 21686]
[New process 21685]
[New process 21681]
[New process 21659]
[New process 21658]
[New process 21648]
[New process 21647]
[New process 21609]
[New process 21594]
[New process 21542]
[New process 21540]
[New process 21516]
#0 0x080851ee in ast_setstate (chan=0xb3401c00, state=AST_STATE_DOWN) at /usr/src/asterisk-1.4.42/include/asterisk/strings.h:37
37 return (!s || (*s == ‘’));
#0 0x080851ee in ast_setstate (chan=0xb3401c00, state=AST_STATE_DOWN) at /usr/src/asterisk-1.4.42/include/asterisk/strings.h:37
name = “mISDN/400u111100ݴ

Hi All,
I have installed centos 5.6 32 bit on xeon server and i have also installed
latest version of asterisk 1.6 and dahdi as well.
I want to install chan_ss7 for this server and I want to know about the
following device.
Digium TE420B
I dont know much about the configuration files for Digium TE420B.
Can anybody provide me required ss7.conf file and also provide dahdi
configuration which is needed for this device.
Thanks you so much in advance!!

Thanks,
Max Alex