[asterisk-ss7] Re: asterisk-ss7 Digest, Vol 17, Issue 14
Faris
faris at wavenet.lk
Wed Jul 26 20:45:53 MST 2006
Hello All,
I am having a problem starting my asterisk box with chan_ss7 module. i have
configured 2 asterisk box connected cross cable (according to the Wiki pages
for chan_ss7) its gives me the following error.
how do i sort it out ?
[root at maldives2 ~]# asterisk -vc
Asterisk 1.2.9.1, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer <markster at digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for
details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it
under
certain conditions. Type 'show license' for details.
=========================================================================
Asterisk Event Logger Started /var/log/asterisk/event_log
Asterisk Dynamic Loader loading preload modules:
Jul 27 09:33:33 NOTICE[8585]: cdr.c:1191 do_reload: CDR simple logging
enabled.
Asterisk PBX Core Initializing
Registering builtin applications:
[AbsoluteTimeout]
[Answer]
[BackGround]
[Busy]
[Congestion]
[DigitTimeout]
[Goto]
[GotoIf]
[GotoIfTime]
[ExecIfTime]
[Hangup]
[NoOp]
[Progress]
[ResetCDR]
[ResponseTimeout]
[Ringing]
[SayNumber]
[SayDigits]
[SayAlpha]
[SayPhonetic]
[SetAccount]
[SetAMAFlags]
[SetGlobalVar]
[SetLanguage]
[Set]
[SetVar]
[ImportVar]
[Wait]
[WaitExten]
Asterisk Dynamic Loader Starting:
[res_musiconhold.so] => (Music On Hold Resource)
[res_agi.so] => (Asterisk Gateway Interface (AGI))
[res_features.so] => (Call Features Resource)
[res_indications.so] => (Indications Configuration)
[res_adsi.so] => (ADSI Resource)
[res_monitor.so] => (Call Monitoring Resource)
[res_crypto.so] => (Cryptographic Digital Signatures)
[pbx_loopback.so] => (Loopback Switch)
[pbx_dundi.so] => (Distributed Universal Number Discovery (DUNDi))
[pbx_realtime.so] => (Realtime Switch)
[pbx_spool.so] => (Outgoing Spool Support)
[pbx_functions.so] => (Builtin dialplan functions)
[pbx_config.so] => (Text Extension Configuration)
Jul 27 09:33:33 WARNING[8585]: pbx.c:3762 ast_merge_contexts_and_delete:
Requested contexts didn't get merged
[pbx_ael.so] => (Asterisk Extension Language Compiler)
[chan_agent.so] => (Agent Proxy Channel)
[chan_iax2.so] => (Inter Asterisk eXchange (Ver 2))
[chan_features.so] => (Feature Proxy Channel)
[chan_mgcp.so] => (Media Gateway Control Protocol (MGCP))
[chan_sip.so] => (Session Initiation Protocol (SIP))
[chan_phone.so] => (Linux Telephony API Support)
[chan_ss7.so] => (SS7 Protocol Support)
Jul 27 09:33:33 NOTICE[8585]: config.c:516 load_config_link: Configured link
'l1' on linkset 'siuc', firstcic=1
Jul 27 09:33:33 WARNING[8585]: config.c:675 load_config_host: Missing
interface entries for host 'maldives2'.
Jul 27 09:33:33 NOTICE[8585]: config.c:840 load_config: Configuring DPC 2
for linkset 'siuc'.
-- Starting cluster thread, pid=8585.
Jul 27 09:33:33 NOTICE[8585]: mtp.c:1938 mtp_init: Initialising 1 signalling
links
-- Starting MTP thread, pid=8585.
-- Starting continuity check thread, pid=8585.
-- SS7 channel loaded successfully.
[chan_skinny.so] => (Skinny Client Control Protocol (Skinny))
Jul 27 09:33:33 WARNING[8585]: chan_skinny.c:3206 reload_config: Failed to
bind to 0.0.0.0:2000: Address already in use
[chan_local.so] => (Local Proxy Channel)
[chan_zap.so] => (Zapata Telephony w/PRI)
-- Starting monitor thread, pid=8585.
Jul 27 09:33:33 NOTICE[8602]: mtp.c:1543 mtp_thread_main: Empty Zaptel
output buffer detected, outgoing packets may have been lost on link 'l1'.
[chan_oss.so] => (OSS Console Channel Driver)
[format_pcm_alaw.so] => (Raw aLaw 8khz PCM Audio support)
[format_gsm.so] => (Raw GSM data)
[app_mp3.so] => (Silly MP3 Application)
[app_page.so] => (Page Multiple Phones)
[app_parkandannounce.so] => (Call Parking and Announce Application)
[format_sln.so] => (Raw Signed Linear Audio support (SLN))
[app_macro.so] => (Extension Macros)
[app_settransfercapability.so] => (Set ISDN Transfer Capability)
[app_meetme.so] => (MeetMe conference bridge)
[app_sms.so] => (SMS/PSTN handler)
[app_sayunixtime.so] => (Say time)
[codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator)
Jul 27 09:33:33 NOTICE[8602]: mtp.c:1442 mtp_thread_main: Excessive poll
delay 20926!
[app_cut.so] => (Cut out information from a string)
[app_groupcount.so] => (Group Management Routines)
[app_enumlookup.so] => (ENUM Lookup)
[app_url.so] => (Send URL Applications)
[func_enum.so] => (ENUM Related Functions)
[cdr_csv.so] => (Comma Separated Values CDR Backend)
[app_setrdnis.so] => (Set RDNIS Number)
[app_stack.so] => (Stack Routines)
[app_chanspy.so] => (Listen to the audio of an active channel
)
[app_zapras.so] => (Zap RAS Application)
[app_waitforring.so] => (Waits until first ring after time)
[app_softhangup.so] => (Hangs up the requested channel)
[app_nbscat.so] => (Silly NBS Stream Application)
[app_while.so] => (While Loops and Conditional Execution)
[app_alarmreceiver.so] => (Alarm Receiver for Asterisk)
[app_setcidnum.so] => (Set CallerID Number)
[app_disa.so] => (DISA (Direct Inward System Access) Application)
[app_cdr.so] => (Tell Asterisk to not maintain a CDR for the current call)
[app_mixmonitor.so] => (Mixed Audio Monitoring Application)
[app_voicemail.so] => (Comedian Mail (Voicemail System))
[app_hasnewvoicemail.so] => (Indicator for whether a voice mailbox has
messages in a given folder.
[codec_alaw.so] => (A-law Coder/Decoder)
[format_jpeg.so] => (JPEG (Joint Picture Experts Group) Image Format)
[app_read.so] => (Read Variable Application)
[app_zapscan.so] => (Scan Zap channels application)
[app_dumpchan.so] => (Dump Info About The Calling Channel)
[format_ilbc.so] => (Raw iLBC data)
[app_queue.so] => (True Call Queueing)
[format_wav_gsm.so] => (Microsoft WAV format (Proprietary GSM))
[format_g726.so] => (Raw G.726 (16/24/32/40kbps) data)
[app_externalivr.so] => (External IVR Interface Application)
[app_setcallerid.so] => (Set CallerID Application)
[cdr_custom.so] => (Customizable Comma Separated Values CDR Backend)
[format_g723.so] => (G.723.1 Simple Timestamp File Format)
[app_system.so] => (Generic System() application)
[codec_a_mu.so] => (A-law and Mulaw direct Coder/Decoder)
[app_echo.so] => (Simple Echo Application)
[app_userevent.so] => (Custom User Event Application)
[app_privacy.so] => (Require phone number to be entered, if no CallerID
sent)
[codec_lpc10.so] => (LPC10 2.4kbps (signed linear) Voice Coder)
Jul 27 09:33:33 NOTICE[8602]: mtp.c:1442 mtp_thread_main: Excessive poll
delay 20983!
[app_lookupcidname.so] => (Look up CallerID Name from local database)
[app_milliwatt.so] => (Digital Milliwatt (mu-law) Test Application)
[app_dictate.so] => (Virtual Dictation Machine)
[app_lookupblacklist.so] => (Look up Caller*ID name/number from blacklist
database)
[app_transfer.so] => (Transfer)
[app_setcdruserfield.so] => (CDR user field apps)
[app_db.so] => (Database Access Functions)
[app_random.so] => (Random goto)
[app_realtime.so] => (Realtime Data Lookup/Rewrite)
[cdr_manager.so] => (Asterisk Call Manager CDR Backend)
[codec_gsm.so] => (GSM/PCM16 (signed linear) Codec Translator)
[format_g729.so] => (Raw G729 data)
[app_waitforsilence.so] => (Wait For Silence)
[app_festival.so] => (Simple Festival Interface)
[app_directory.so] => (Extension Directory)
[app_flash.so] => (Flash zap trunk application)
[app_directed_pickup.so] => (Directed Call Pickup Application)
[codec_adpcm.so] => (Adaptive Differential PCM Coder/Decoder)
[app_test.so] => (Interface Test Application)
[app_senddtmf.so] => (Send DTMF digits Application)
[app_talkdetect.so] => (Playback with Talk Detection)
[app_controlplayback.so] => (Control Playback Application)
[app_getcpeid.so] => (Get ADSI CPE ID)
[app_chanisavail.so]Jul 27 09:33:33 NOTICE[8602]: mtp.c:1442
mtp_thread_main: Excessive poll delay 20985!
=> (Check channel availability)
[codec_ulaw.so] => (Mu-law Coder/Decoder)
[app_verbose.so] => (Send verbose output)
[format_pcm.so] => (Raw uLaw 8khz Audio support (PCM))
[format_ogg_vorbis.so] => (OGG/Vorbis audio)
[app_txtcidname.so] => (TXTCIDName)
[app_image.so] => (Image Transmission Application)
[format_h263.so] => (Raw h263 data)
[app_zapateller.so] => (Block Telemarketers with Special Information Tone)
[app_record.so] => (Trivial Record Application)
[app_forkcdr.so] => (Fork The CDR into 2 separate entities.)
[app_dial.so] => (Dialing Application)
[app_zapbarge.so] => (Barge in on Zap channel application)
[app_exec.so] => (Executes applications)
[app_setcidname.so] => (Set CallerID Name)
[app_md5.so] => (MD5 checksum applications)
[app_sendtext.so] => (Send Text Applications)
[app_authenticate.so] => (Authentication Application)
[app_ices.so] => (Encode and Stream via icecast and ices)
[app_adsiprog.so] => (Asterisk ADSI Programming Application)
[app_curl.so] => (Load external URL)
[func_uri.so] => (URI encode/decode functions)
[func_callerid.so] => (Caller ID related dialplan function)
[app_readfile.so] => (Stores output of file into a variable)
[app_playback.so] => (Sound File Playback Application)
[codec_g726.so] => (ITU G.726-32kbps G726 Transcoder)
[format_au.so] => (Sun Microsystems AU format (signed linear))
[format_vox.so] => (Dialogic VOX (ADPCM) File Format)
[format_wav.so] => (Microsoft WAV format (8000hz Signed Linear))
[app_math.so] => (Basic Math Functions)
[app_eval.so] => (Reevaluates strings)
Asterisk Ready.
*CLI> Jul 27 09:33:33 NOTICE[8602]: mtp.c:1442 mtp_thread_main: Excessive
poll delay 20968!
Jul 27 09:33:33 NOTICE[8602]: mtp.c:1442 mtp_thread_main: Excessive poll
delay 20976!
Jul 27 09:33:33 NOTICE[8602]: mtp.c:1442 mtp_thread_main: Excessive poll
delay 20986!
Jul 27 09:33:33 NOTICE[8602]: mtp.c:1442 mtp_thread_main: Excessive poll
delay 20989!
Jul 27 09:33:33 NOTICE[8602]: mtp.c:1442 mtp_thread_main: Excessive poll
delay 20989!
Jul 27 09:33:33 NOTICE[8602]: mtp.c:1442 mtp_thread_main: Excessive poll
delay 20988!
Jul 27 09:33:33 NOTICE[8602]: mtp.c:1442 mtp_thread_main: Excessive poll
delay 20989!
Jul 27 09:33:33 NOTICE[8602]: mtp.c:1442 mtp_thread_main: Excessive poll
delay 20990!
.....
....
....
Please help me if anybody knows this issue..
Thanks !
Faris.
----- Original Message -----
From: <asterisk-ss7-request at lists.digium.com>
To: <asterisk-ss7 at lists.digium.com>
Sent: Wednesday, July 26, 2006 1:00 AM
Subject: asterisk-ss7 Digest, Vol 17, Issue 14
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> Today's Topics:
>
> 1. Re: RE : Re: [asterisk-ss7] chan_ss7 connected to a telco (Anton)
> 2. Double Ring on Asterisk 1.2.x (asterisk at nicox.org)
> 3. RE : Re: RE : Re: [asterisk-ss7] chan_ss7 connected to a
> telco (harry gaillac)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Tue, 25 Jul 2006 01:01:42 +0500
> From: Anton <anton.vazir at gmail.com>
> Subject: Re: RE : Re: [asterisk-ss7] chan_ss7 connected to a telco
> To: asterisk-ss7 at lists.digium.com
> Message-ID: <200607250101.42093.anton.vazir at gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Have to mention the audio lost issue, binded with network
> jitter. For some situations it makes the channel driver
> hardly usable...
>
> On 24 July 2006 18:01, Kai Militzer wrote:
> > Hello Harry,
> >
> > My experiences are as followed:
> >
> > harry gaillac wrote:
> > > SS7 protocols:
> > > MTP layers 1-3
> > > ISUP
> >
> > The MTP-Layer seems to be implented fully. At least to
> > Layer 2, not sure about Layer 3, but I don't use things
> > as failover, so I cannot tell for sure.
> >
> > ISUP is not completely implemented. The most message
> > types/features are there and working, but AFAIK some
> > things are not there, but also nothing that I would need.
> > When I first tried chan_ss7 in late 2005, there wasn't an
> > implemantation of CCR test, but we let that implement by
> > a developer and since then all features that we need are
> > there.
> >
> > > Basic calls :
> > > inbound-outbound
> >
> > Work as they should. Nothing much to say about that.
> >
> > > Supplementary services:
> > > Redirecting number
> >
> > I think it got transmited, but I am not sure, if this
> > gets somehow into other channel types like SIP.
> >
> > > CLIP/CLIR/COLP/COLR
> >
> > The calling number is always in ISUP messages. Only if
> > one bit is set, you must not show it to the end point
> > (the user). Asterisk with chan_ss7 can set this bit. I
> > don't know if COLP works. Never testes this.
> >
> > > Network configuration:
> > > SEP functionality
> > > Connection to other SEP or STP
> >
> > Never done anything into that direction, so I cannot
> > tell.
> >
> > In overall I can say, that it works stable since about
> > four month in a friendly-user test.
> >
> > Regards,
> > Kai
>
>
> ------------------------------
>
> Message: 2
> Date: Mon, 24 Jul 2006 23:15:08 +0200 (CEST)
> From: asterisk at nicox.org
> Subject: [asterisk-ss7] Double Ring on Asterisk 1.2.x
> To: asterisk-ss7 at lists.digium.com
> Message-ID: <Pine.LNX.4.64.0607242308580.6500 at localhost.localdomain>
> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
>
>
> Hello,
> I have a problem with the ringing on a Asterisk 1.2.x and a Digium TE410
> and TE411P.
>
> if i Dial without any dial parameter through a Zaptel channel i hear the
> ringing from the telco and the ringing generated from asterisk.
>
> If i Dial through the Zaptel with Parameter "r" i get the ringing from
> asterisk, but its always the same, and not the correct ringing of india
> (in example), also i get no messages like, the number you dialed is not
> existing.
>
> If i Answer the call before i dial out the ringing is okay and the
> messages are okay, but the cdr's aren't.
>
> did anybody know this problem?
>
> i hope for a fast and good solution.
>
> thanks a lot
>
>
> nico
>
>
>
> ------------------------------
>
> Message: 3
> Date: Mon, 24 Jul 2006 23:33:36 +0200 (CEST)
> From: harry gaillac <gaillacharry at yahoo.fr>
> Subject: RE : Re: RE : Re: [asterisk-ss7] chan_ss7 connected to a
> telco
> To: asterisk-ss7 at lists.digium.com
> Message-ID: <20060724213336.77525.qmail at web25510.mail.ukl.yahoo.com>
> Content-Type: text/plain; charset=iso-8859-1
>
> Which is the best digium card for building a SIP/SS7
> gateway
> what about the latest digium cards with a DSP .
>
> --- Anton <anton.vazir at gmail.com> a écrit :
>
> > Have to mention the audio lost issue, binded with
> > network
> > jitter. For some situations it makes the channel
> > driver
> > hardly usable...
> >
> > On 24 July 2006 18:01, Kai Militzer wrote:
> > > Hello Harry,
> > >
> > > My experiences are as followed:
> > >
> > > harry gaillac wrote:
> > > > SS7 protocols:
> > > > MTP layers 1-3
> > > > ISUP
> > >
> > > The MTP-Layer seems to be implented fully. At
> > least to
> > > Layer 2, not sure about Layer 3, but I don't use
> > things
> > > as failover, so I cannot tell for sure.
> > >
> > > ISUP is not completely implemented. The most
> > message
> > > types/features are there and working, but AFAIK
> > some
> > > things are not there, but also nothing that I
> > would need.
> > > When I first tried chan_ss7 in late 2005, there
> > wasn't an
> > > implemantation of CCR test, but we let that
> > implement by
> > > a developer and since then all features that we
> > need are
> > > there.
> > >
> > > > Basic calls :
> > > > inbound-outbound
> > >
> > > Work as they should. Nothing much to say about
> > that.
> > >
> > > > Supplementary services:
> > > > Redirecting number
> > >
> > > I think it got transmited, but I am not sure, if
> > this
> > > gets somehow into other channel types like SIP.
> > >
> > > > CLIP/CLIR/COLP/COLR
> > >
> > > The calling number is always in ISUP messages.
> > Only if
> > > one bit is set, you must not show it to the end
> > point
> > > (the user). Asterisk with chan_ss7 can set this
> > bit. I
> > > don't know if COLP works. Never testes this.
> > >
> > > > Network configuration:
> > > > SEP functionality
> > > > Connection to other SEP or STP
> > >
> > > Never done anything into that direction, so I
> > cannot
> > > tell.
> > >
> > > In overall I can say, that it works stable since
> > about
> > > four month in a friendly-user test.
> > >
> > > Regards,
> > > Kai
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com
> > --
> >
> > asterisk-ss7 mailing list
> > To UNSUBSCRIBE or update options visit:
> >
> >
> http://lists.digium.com/mailman/listinfo/asterisk-ss7
> >
>
>
>
>
>
>
>
>
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