[Asterisk-Users] dont write me again
Smith
laure2004 at direcway.com
Wed Dec 1 11:12:23 MST 2004
----- Original Message -----
From: <asterisk-users-request at lists.digium.com>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, December 01, 2004 7:07 AM
Subject: Asterisk-Users Digest, Vol 5, Issue 6
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> Today's Topics:
>
> 1. Re: software phones for Asterisk - is there a list? (Roger Hanson)
> 2. Re: software phones for Asterisk - is there a list?
> (Girish Gopinath)
> 3. Re: broadvoice and gsm codec (Sean Cook)
> 4. RE: Avoided deadlock (mattf)
> 5. Re: kernel: Out of storage space while 900 MB free?
> (Ronan Mullally)
> 6. SIP expiry time (HengWee Chin)
> 7. CallerID on X100P in South Africa (Thorsten Neumann)
> 8. Re: Asterisk Process Stop After few hours (Michael Manousos)
> 9. Re: software phones for Asterisk - is there a list?
> (Tomasz Chmielewski)
> 10. Re: Advantage of IAX2 to SIP? (Rich Adamson)
> 11. Asterisk Call Monitor and soxmix error (Craig Waddington)
> 12. RE: Avoided deadlock (Brian West)
> 13. RE: CallerID on X100P in South Africa (Doug Reid - Stormcorp)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Wed, 1 Dec 2004 08:23:49 -0600
> From: "Roger Hanson" <roger at makarios.us>
> Subject: Re: [Asterisk-Users] software phones for Asterisk - is there
> a list?
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID: <004201c4d7b1$6101fd00$6400000a at work.makarios.us>
> Content-Type: text/plain; format=flowed; charset="ISO-8859-2";
> reply-type=response
>
>
> ----- Original Message -----
> From: "Tomasz Chmielewski" <mangoo at mch.one.pl>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Sent: Wednesday, December 01, 2004 4:42 AM
> Subject: [Asterisk-Users] software phones for Asterisk - is there a
> list?
>
>
> > Hello,
> >
> > Is there a list of software phones which will work with Asterisk?
> >
> > For Linux and Windows?
> >
> > I don't have any hardware yet, and before I buy anything I would like
> > to know how Asterisk really works (with software "phones" for
> > example).
> >
>
> I'm sure you didn't search the wiki, did you? There's tons of
> information there on soft phones.
> http://voip-info.org/wiki-VOIP+Phones
>
>
>
> ------------------------------
>
> Message: 2
> Date: Wed, 1 Dec 2004 06:30:17 -0800 (PST)
> From: Girish Gopinath <asterisk_in at yahoo.com>
> Subject: Re: [Asterisk-Users] software phones for Asterisk - is there
> a list?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <20041201143017.95110.qmail at web54209.mail.yahoo.com>
> Content-Type: text/plain; charset=us-ascii
>
> Hello,
>
> --- Tomasz Chmielewski <mangoo at mch.one.pl> wrote:
>
> > Is there a list of software phones which will work with Asterisk?
>
> See the 'SIP Phones (SIP User Agents)' section here:
> http://pernau.at/kd/voip/bookmarks-sip-rtp-ua.html
>
> Regards, Girish
>
>
>
> __________________________________
> Do you Yahoo!?
> The all-new My Yahoo! - What will yours do?
> http://my.yahoo.com
>
>
> ------------------------------
>
> Message: 3
> Date: Wed, 01 Dec 2004 09:33:54 -0500
> From: Sean Cook <scook at kinex.net>
> Subject: Re: [Asterisk-Users] broadvoice and gsm codec
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <1101911634.18099.2.camel at sean.benchmark-systems.com>
> Content-Type: text/plain
>
> Correct me if I am wrong, but G729 is not distributed with asterisk. By
> default it is not available without a license.
>
> http://www.voip-info.org/wiki-ITU+G.729
>
> You have to compile and install the free implementation to test.
> Otherwise it won't work...
>
> Kinda hard to get asterisk to use a codecs that isn't installed ;)
>
>
> Sean
>
> On Tue, 2004-11-30 at 19:36 -0800, Luki wrote:
> > >> I have broadvoice working with ulaw like the example shows.
> > >> I was wondering if another codec like gsm can be used.
> > > We've been told no previously. Wouldn't be hard to prove it though.
> >
> > I just tried it. Looks to me that the answer is no.
> >
> > When forcing G729 I get:
> > Got SIP response 480 "Temporarily unavailable" back from 147.135.0.128
> >
> > With G726 the call "connects" but all you hear is the friendly voice
saying
> > "your call cannot be completed at this time". Changing it back to g711
makes
> > it work again.
> >
> > FYI, sip show channels shows:
> >
> > Peer User/ANR Call ID Seq (Tx/Rx) Format Duration
> > 147.135.0.128 3604000000 5acd3e62478 00102/00000 G726 0:03
> > 164.67.000.000 11 2f1743df-f5 00101/00102 ULAW 0:03
> >
> > --Luki
> >
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> ------------------------------
>
> Message: 4
> Date: Wed, 1 Dec 2004 09:34:22 -0500
> From: mattf <mattf at vicimarketing.com>
> Subject: RE: [Asterisk-Users] Avoided deadlock
> To: 'Paradise Dove' <pardove at gmail.com>, 'Asterisk Users Mailing List
> - Non-Commercial Discussion' <asterisk-users at lists.digium.com>
> Message-ID:
> <DB43F516702AAF4392AA45573F18181950250B at vicimail.vicimarketinggroup.com>
>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hello,
>
> I'd suggest posting a bug if you haven't already and if you have purchased
> any Digium products I would recommend calling them as well. The
> ast_channel_walk_locked error is a rare and hard to diagnose problem and
the
> bug trackers and Digium would be the best people to help you.
>
> It might also help to know what kind of SIP client you are using
(hard/soft
> phone) and what the network you are on is like.
>
> MATT---
>
>
> -----Original Message-----
> From: Paradise Dove [mailto:pardove at gmail.com]
> Sent: Wednesday, December 01, 2004 7:56 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Avoided deadlock
>
>
> but i have already an UltraWide 320 Scsi HardDisk installed on my * box.
> seems that this won't be the cause of my problem at least.
> i think that it should be something betweeen these two errors:
>
> - NOTICE[6189]: Disconnecting call 'SIP/2502-6303' for lack of RTP
> activity in 4794 seconds
> - WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided
> deadlock for 'SIP/2502-6303', 10 retries!
>
> I have these two lines in my sip.conf
>
> rtptimeout=300
> rtpholdtimeout=480
>
> it seems that these options don't work as expected.
>
> Paradise Dove
>
>
> On Wed, 1 Dec 2004 07:11:53 -0500, mattf <mattf at vicimarketing.com> wrote:
> > Hello,
> >
> > I had this problem a few months ago on a machine that I did a lot of
> > recording on. It was caused by slow disk access time. Asterisk would
wait
> > for something to write to disk and basically freeze everything. It would
> > always eventually happen to the same machine no matter if I wiped it
> > completely and did a full reinstall. I fixed it by buying 4 new 320-SCSI
> > drives and a new 320SCSI RAID card, no problems since then(6 months).
> >
> > I did report this to Digium several times, they were even in my machine
a
> > few times to monitor it, they had no clue what was causing it, and the
> > ast_channel_walk_locked bug has no documentation anywhere about it(not
> much
> > help to look in the code either).
> >
> > Hope this helps.
> >
> > MATT---
> >
> >
> >
> >
> > -----Original Message-----
> > From: Bartosz Jozwiak [mailto:bartek at cq-link.sr]
> > Sent: Wednesday, December 01, 2004 6:40 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Avoided deadlock
> >
> > > at the same time I have also this notice log.
> > > this makes my problem more meaningful.
> > > i think it might be a bug inside *. (am i right?)
> > >
> > > Dec 1 12:44:46 NOTICE[6189]: Disconnecting call 'SIP/2502-6303' for
> > > lack of RTP activity in 4794 seconds
> > > Dec 1 12:44:47 NOTICE[6189]: Disconnecting call 'SIP/2502-6303' for
> > > lack of RTP activity in 4795 seconds
> > > Dec 1 12:44:47 NOTICE[6189]: Disconnecting call 'SIP/2502-6303' for
> > > lack of RTP activity in 4795 seconds
> > >
> > > Paradise Dove
> > >
> > > > Dec 1 12:08:43 WARNING[6189]: channel.c:495
ast_channel_walk_locked:
> > > > Avoided deadlock for 'SIP/2502-6303', 10 retries!
> > > > Dec 1 12:08:44 WARNING[6189]: channel.c:495
ast_channel_walk_locked:
> > > > Avoided deadlock for 'SIP/2502-6303', 10 retries!
> > > > Dec 1 12:08:44 WARNING[6189]: channel.c:495
ast_channel_walk_locked:
> > > > Avoided deadlock for 'SIP/2502-6303', 10 retries!
> > > >
> > > > what does this warning really mean? I have tones of them!!!!!
> > > > does it have any side effect on my * box? 'cose I've recently had
> > > > random seg. faults on my box.
> > > > I'm using latest CVS -r v1-0
> >
> > I have the same problem as you. But so far did not find an asnwer for
it.
> >
> > Bartosz
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> ------------------------------
>
> Message: 5
> Date: Wed, 1 Dec 2004 14:35:55 +0000 (GMT)
> From: Ronan Mullally <ronan at iol.ie>
> Subject: Re: [Asterisk-Users] kernel: Out of storage space while 900
> MB free?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <Pine.LNX.4.60.0412011435380.1772 at localhost>
> Content-Type: TEXT/PLAIN; charset=US-ASCII
>
> "df -i" - you're out of inodes on that filesystem.
>
>
> -Ronan
>
> On Wed, 1 Dec 2004, Roger Schreiter wrote:
>
> > Hi,
> >
> > after loading the zaptel driver wct4xxp I have strange
> > log lines in the syslog:
> >
> > Out of storage space.
> >
> > free tells, that more than 900 MB are still free.
> > Disk space is also available.
> >
> > I'm using a dual opteron in 64 bit mode.
> >
> > Any ideas?
> >
> >
> > Roger.
> >
> >
> > Syslog:
> >
> > Dec 1 02:18:37 ipphone4 kernel: TE410P: Launching card: 0
> > Dec 1 02:18:37 ipphone4 kernel: TE410P: Setting up global serial
parameters
> > Dec 1 02:18:38 ipphone4 kernel: Found a Wildcard: Wildcard
TE410P-Xilinx
> > Dec 1 02:18:38 ipphone4 kernel: Registered tone zone 2 (France)
> > Dec 1 02:18:38 ipphone4 kernel: TE410P: Span 1 configured for
CCS/HDB3/CRC4
> > Dec 1 02:18:38 ipphone4 kernel: SPAN 1: Primary Sync Source
> > Dec 1 02:18:38 ipphone4 kernel: TE410P: Span 2 configured for
CCS/HDB3/CRC4
> > Dec 1 02:18:38 ipphone4 kernel: TE410P: Span 3 configured for
CCS/HDB3/CRC4
> > Dec 1 02:18:38 ipphone4 kernel: TE410P: Span 4 configured for
CCS/HDB3/CRC4
> > Dec 1 02:18:38 ipphone4 kernel: SPAN 4: Quaternary Sync Source
> > Dec 1 02:18:38 ipphone4 kernel: Zaptel: Master changed to TE4/0/2
> > Dec 1 02:18:38 ipphone4 kernel: Zaptel: Master changed to TE4/0/3
> > Dec 1 02:18:38 ipphone4 kernel: Zaptel: Master changed to TE4/0/4
> > Dec 1 02:18:38 ipphone4 kernel: Zaptel: Master changed to TE4/0/3
> > Dec 1 02:18:38 ipphone4 kernel: Zaptel: Master changed to TE4/0/1
> > Dec 1 02:18:48 ipphone4 kernel: Out of storage space
> > Dec 1 02:18:48 ipphone4 last message repeated 121 times
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
>
>
> ------------------------------
>
> Message: 6
> Date: Wed, 01 Dec 2004 22:36:22 +0800
> From: "HengWee Chin" <chinhw at hotmail.com>
> Subject: [Asterisk-Users] SIP expiry time
> To: asterisk-users at lists.digium.com
> Message-ID: <BAY22-F3092A724859DC5FD7BF215DCBF0 at phx.gbl>
> Content-Type: text/plain; format=flowed
>
> Hi,
>
> I notice that SJPhone is registering to asterisk with an expires of 120
> secs. However, when I invoke the command "sip show peer [sip id]". I
notice
> that the output indicates the expires 427 and the expiry is 900. Can
someone
> explain these numbers to me?
>
> I also notice that just before SJPhone re-register, when I try to make a
> call to the SJPhone, asterisk will complain that it is not able to find
the
> URL to sip. When I issue the command "sip show peers", I can still see the
> URL in the list of sip registered. After SJPhone re-register, asterisk is
> able to call again. Any Ideas how come there is this behaviour? A bug?
>
> Regards,
> Chin
>
>
>
>
> ------------------------------
>
> Message: 7
> Date: Wed, 01 Dec 2004 16:41:58 +0200
> From: Thorsten Neumann <TNeumann at gt247.com>
> Subject: [Asterisk-Users] CallerID on X100P in South Africa
> To: asterisk-users at lists.digium.com
> Message-ID: <1101912118.6347.1819.camel at localhost.localdomain>
> Content-Type: text/plain; charset="us-ascii"
>
> Heya
>
> I have my * box connected to the Telkom PSTN, and an analogy line with
> callerID subscription (yes we get charged extra :).
>
> When i call the line, it rings once, a short pause, and then the
> continued ringing of the phone. Using an external callerID device, it
> shows the number of the call initiator.
>
> However, when * answers the line, it does not pick up the initiator, but
> the destination number?? e.g. 793 1486 calls 787 0107. For some bizarre
> reason, it shows my number being dialled. Why?
>
> I wanted to ask how this could be, and if it requires me to make changes
> to
> #define DEFAULT_CIDRINGS 2 or usecallerid=uk.
>
> Thanks,
> Thorsten
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> ------------------------------
>
> Message: 8
> Date: Wed, 01 Dec 2004 17:46:47 +0200
> From: Michael Manousos <manousos at inaccessnetworks.com>
> Subject: Re: [Asterisk-Users] Asterisk Process Stop After few hours
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <41ADE767.9030603 at inaccessnetworks.com>
> Content-Type: text/plain; charset=us-ascii; format=flowed
>
> Daniel Eboa wrote:
> > Hello to all,
> >
> > I have a strange behavior of my asterisk box. I'm running asterisk with
> > asterisk-oh323 channel driver and everything works very well.
> > But after few hours, my asterisk stop running and I have to restart it
> > by typing "asterisk -vvvc". Most of the time I connect to my asterisk
> > with a remote host so I don't know exactly which error causes my box to
> > stop, but I found on the console this message: "Segmentation Fault". Did
> > any one has experience this problem?? what is the solution?
>
> What versions of Asterisk/asterisk-oh323 do you run?
> Please provide a backtrace of the core file dumped.
>
> >
> > I use Cisco ATA 186 Boxes with my asterisk.
> >
> > Thank In advance.
> >
> > Daniel.
> >
>
>
> Michael.
>
>
>
>
> ------------------------------
>
> Message: 9
> Date: Wed, 01 Dec 2004 15:47:45 +0100
> From: Tomasz Chmielewski <mangoo at mch.one.pl>
> Subject: Re: [Asterisk-Users] software phones for Asterisk - is there
> a list?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <41ADD991.2060307 at mch.one.pl>
> Content-Type: text/plain; charset=ISO-8859-2; format=flowed
>
> Roger Hanson wrote:
> >
> > ----- Original Message ----- From: "Tomasz Chmielewski"
<mangoo at mch.one.pl>
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > <asterisk-users at lists.digium.com>
> > Sent: Wednesday, December 01, 2004 4:42 AM
> > Subject: [Asterisk-Users] software phones for Asterisk - is there a
list?
> >
> >
> >> Hello,
> >>
> >> Is there a list of software phones which will work with Asterisk?
> >>
> >> For Linux and Windows?
> >>
> >> I don't have any hardware yet, and before I buy anything I would like
> >> to know how Asterisk really works (with software "phones" for example).
> >>
> >
> > I'm sure you didn't search the wiki, did you? There's tons of
> > information there on soft phones.
> > http://voip-info.org/wiki-VOIP+Phones
>
> nopez, didn't really know there is one :)
>
> OK, so I found kphone, installed on two linuxes, configured the way the
> wiki says, registered with asterisk, but they won't connect to each
> other... will start a new post if I don't find anything useful in the
> wiki and lists.digium.com... :)
>
> Tomek
>
>
> ------------------------------
>
> Message: 10
> Date: Wed, 1 Dec 2004 08:37:27 -0600
> From: Rich Adamson <radamson at routers.com>
> Subject: Re: [Asterisk-Users] Advantage of IAX2 to SIP?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <Chameleon.1101912442.adar0 at vegas>
> Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1
>
> > Some - few - providers are using IAX2 as a protocol. Most are using SIP.
> > I know that there are advantages of IAX2 regarding multiple connections.
> > But beside this I'm asking myself (and you all) why I should prefer IAX2
> > when my SIP connection is working.
> >
> > Are there differences in the performance?
>
> Generic answer: most of the sip providers require g711 which consumes
about
> 80kb/s of bandwidth, and a fair number of those are non-asterisk based
> systems that don't have iax2 support. Most/all iax2 providers are asterisk
> based and typically support other codecs besides g711. Iax2 with gsm
> consumes roughly 30kb/s, which on a typical dsl broadband connection, is
> preferred by most asterisk users.
>
> There are several other technical advantages for iax2, one of which is its
> ability to carry multiple calls within a single packet reducing the IP
> overhead on the net. (Bigger bang for the broadband buck.)
>
> The downside of iax2 is that its not yet a standard (rfc or otherwise)
> and even though its gpl'ed, few non-asterisk systems have bothered to
> implement it as yet.
>
>
>
>
> ------------------------------
>
> Message: 11
> Date: Wed, 1 Dec 2004 14:56:26 -0000
> From: "Craig Waddington" <craig at barony.com>
> Subject: [Asterisk-Users] Asterisk Call Monitor and soxmix error
> To: <asterisk-users at lists.digium.com>
> Message-ID:
> <463B387E4425E444AF9FB66135142DA2125663 at bar01.seaside.co.im>
> Content-Type: text/plain; charset="us-ascii"
>
> Asterisk Monitor seems to be working fine. Though the problem I am
> having is the two files (in & out) muxing.
>
>
>
> I added ,m to the string, yet the call records two files still, and I
> get the resulting error, at the bottom.
>
>
>
> monitor executing ( nice -n 19 soxmix
> /var/spool/asterisk/monitor/rec_fr_1624672199_to_621950_at_01122004-13:4
> 8:23-in.gsm
> /var/spool/asterisk/monitor/rec_fr_1624672199_to_621950_at_01122004-13:4
> 8:23-out.gsm
> /var/spool/asterisk/monitor/rec_fr_1624672199_to_621950_at_01122004-13:4
> 8:23.wav && rm -f
> /var/spool/asterisk/monitor/rec_fr_1624672199_to_621950_at_01122004-13:4
> 8:23-* ) &
>
> nice: soxmix: No such file or directory
>
>
>
> soxmix exists
>
>
>
> exten => _8.,2,Monitor(gsm,${CALLFILENAME},m)
>
>
>
> Path to soxmix = /usr/bin/soxmix
>
>
>
> Asterisk seems to be looking in the wrong place for it?
>
>
>
> Is there a command line for soxmix to test muxing two .gsm files ?
>
>
>
>
>
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> ------------------------------
>
> Message: 12
> Date: Wed, 1 Dec 2004 08:58:40 -0600
> From: "Brian West" <brian at bkw.org>
> Subject: RE: [Asterisk-Users] Avoided deadlock
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID: <auto-000026727887 at cgp1.tulsaconnect.com>
> Content-Type: text/plain; charset="us-ascii"
>
> Repeat after me... WARNING != ERROR. This is just letting you know that
it
> walked the channel list and did avoid a dead lock by not trying to grab a
> lock on a channel that's already locked.
>
> if (ast_mutex_trylock(&l->lock)) {
> if (retries < 10)
> ast_log(LOG_DEBUG, "Avoiding initial deadlock for
> '%s'\n", l->name);
> else
> ast_log(LOG_WARNING, "Avoided initial deadlock for
'%s',
> %d retries!\n", l->name, retries);
>
>
> Read the code it tells you... channel.c
>
> bkw
>
> > I'd suggest posting a bug if you haven't already and if you have
purchased
> > any Digium products I would recommend calling them as well. The
> > ast_channel_walk_locked error is a rare and hard to diagnose problem and
> > the
> > bug trackers and Digium would be the best people to help you.
>
>
>
>
> ------------------------------
>
> Message: 13
> Date: Wed, 1 Dec 2004 17:02:10 +0200
> From: "Doug Reid - Stormcorp" <doug at stormcorp.co.za>
> Subject: RE: [Asterisk-Users] CallerID on X100P in South Africa
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Cc: TNeumann at gt247.com
> Message-ID: <FNEMKMMOKGIILBOPJAMNKEPLCMAA.doug at stormcorp.co.za>
> Content-Type: text/plain; charset="utf-8"
>
> Hi Thorston
>
> It could be the ver of Asterisk or the card driver, we have not used that
particular card but have had
> issues with that and found that the driver was the problem.
>
> Doug
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Thorsten
Neumann
> Sent: Wednesday, December 01, 2004 4:42 PM
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] CallerID on X100P in South Africa
>
>
> Heya
>
> I have my * box connected to the Telkom PSTN, and an analogy line with
callerID subscription (yes we get charged extra :).
>
> When i call the line, it rings once, a short pause, and then the
continued ringing of the phone. Using an external callerID device, it shows
the number of the call initiator.
>
> However, when * answers the line, it does not pick up the initiator, but
the destination number?? e.g. 793 1486 calls 787 0107. For some bizarre
reason, it shows my number being dialled. Why?
>
> I wanted to ask how this could be, and if it requires me to make changes
to
> #define DEFAULT_CIDRINGS 2 or usecallerid=uk.
>
> Thanks,
> Thorsten
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