RE: [Asterisk-Users] Monitor does not like variable subsitutions
Hello,
It does appear to be an issue with the colon, as I ran
this test:
exten = _9X.,2,SetVar(REC_FILE_NAME=test)
exten = _9X.,3,Monitor(wav|${REC_FILE_NAME}|m)
and it worked fine. Indeed a colon is a valid
filename under Linux. So is this a bug?
Jason
--- Jim Van Meggelen [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
Hello,
There is no colon the filename below.
Exactly. But there *is* (or rather *was*) in the
filename you told it
you wanted to write.
Where'd it go?
--- Jim Van Meggelen [EMAIL PROTECTED] wrote:
You are using illegal characters in your file
name.
See this line in your output?
ast_writefile: No such format
'wav|rec_to_448704386865_at_16022005-16'
It can't get past it because the colon is not a
valid filename
character.
[EMAIL PROTECTED] wrote:
Hello,
I have been attempting to get the Monitor
function to
accept a loal variable substitution in order to
use
the same filename later in the same context.
Monitor
does not appear to like it, as it attempts to
use
wav|filename as the recording type, as opposed
to just wav.
Here is what I get if I just supply a filename
directly (it works fine):
--context-
exten = _9X.,3,Monitor(wav|recording|m)
--context-
--CLI-
-- Executing SetVar(SIP/3004-275c,
REC_FILE_NAME=rec_to_448704386865_at_16022005-16:54:10)
in new stack
-- Executing Monitor(SIP/3004-275c,
wav|recording|m) in new stack
-- Executing AGI(SIP/3004-275c,
outbound.agi) in new stack
--CLI-
Here is what I get when I attempt to to variable
substituion for
the filename:
--context-
exten =
_9X.,2,SetVar(REC_FILE_NAME=rec_to_${EXTEN:1}_at_${DATETIME})
exten = _9X.,3,Monitor(wav|${FILENAME}|m)
--context-
--CLI-
-- Executing SetVar(SIP/3004-da21,
REC_FILE_NAME=rec_to_448704386865_at_16022005-16:56:35)
in new stack
-- Executing Monitor(SIP/3004-da21,
wav|rec_to_448704386865_at_16022005-16:56:35|m) in
new stack
Feb 16 16:56:35 WARNING[17028]: file.c:934
ast_writefile: No such format
'wav|rec_to_448704386865_at_16022005-16'
Feb 16 16:56:35 WARNING[17028]:
res_monitor.c:154
ast_monitor_start: Could not create file
/var/spool/asterisk/monitor/m-in Feb 16 16:56:35
WARNING[17028]: res_monitor.c:300
ast_monitor_change_fname: Cannot change monitor
filename of channel SIP/3004-da21 to m,
monitoring not
started-- Executing AGI(SIP/3004-da21,
outbound.agi) in new stack
--CLI-
I do believe that I had this working before (I
am
running the CVS HEAD from yesterday).
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Re: [Asterisk-Users] chan_sip errors on CVS HEAD
Asterisk wrote: I've got a test * server (hppbx) where I install CVS-HEAD as often as possible, with my extension registered to this, talking through IAX to our production server which then channels out to the PSTN. After completing a call just now, the following appeared on the CLI of hppbx (the 90xxx is a valid number, changed to protect the guilty): == Spawn extension (from-sip, 90xxx, 1) exited non-zero on 'SIP/711-31db' Feb 16 12:42:38 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy Feb 16 12:42:39 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy Feb 16 12:42:41 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy Feb 16 12:42:45 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy Feb 16 12:42:49 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy Feb 16 12:42:53 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy Feb 16 12:42:57 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy hppbx*CLI show version Asterisk CVS-HEAD-02/05/05-09:30:42 built by [EMAIL PROTECTED] on a i686 running Linux Feb 16 12:43:01 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy Feb 16 12:43:05 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy Feb 16 12:43:09 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy hppbx*CLI show channels Channel (ContextExtensionPri ) State Appl. Data 0 active channel(s) There are no more errors after this. Is there a From: header present? Turn on SIP DEBUG and check the SIP packets. /O ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Why Asterisk can't cope with silence suppression?
Peter Svensson wrote: On Wed, 16 Feb 2005, Rob Scott wrote: Why is it that Asterisk can't cope with silence suppression? All the clients seem to be able to but not Asterisk. What would be needed to get it to work with silence suppression? What is the problem? Asterisk clocks outgoing rtp data to a device from the incoming rtp stream from the same device. This is a known limitation and there has been some talk about implementing an internal clocking system. In addition, Asterisk should generate comfort noise when the rtp stream is quiet due to silence supression (which is signalled with a CN packet). Perhaps the new jitter buffer will be able to handle this? Peter, I think we do generate comfort noice in CVS head, even though the clocking is still a major problem. /O ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Asterisk@Home 0.6 Released
Hey! I installed V0.5 and i was suprised: Good job, i love it! Is there a plan to include drivers for HFC-S Cards (zaphfc / bristuff)?? Greets from germany Michael [EMAIL PROTECTED] schrieb: New features include Festival text to speech and a new Web Conferencing GUI. There are also numerous small fixes and enhancements. http://asteriskathome.sourceforge.net/ __ Do you Yahoo!? All your favorites on one personal page Try My Yahoo! http://my.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@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@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
Re: [Asterisk-Users] Sip Notify PAP2-NA?
Chris St Denis wrote: I am using mysql sipfriends and can't seem to get the MWI to work. From what I've read it seems this is not supported with that dynamic system, and probably never will be. In the 1.0 stable release, you can not send MWI for database peers. In CVS head, the base for the future 1.2 stable, there are three ways to load a peer * Static from configuration (conf file and database) * Realtime from database (like MYSQLFRIENDS in 1.0 stable) * Static load at runtime from realtime database - The static peers are loaded at startup and suppot MWI NAT keepalives. - The realtime method still can't support MWI and NAT keepalives. - The new hybrid method, that loads from realtime database when needed and keeps the peer in memory, supports MWI and NAT keepalives. Also, the PAP2-NA has the ability to reboot via a sip notify and I would like to be able to do that. There is support for rebooting via custom SIP notify messages in CVS head. It is time to check the CVS head (v1.1dev) version of Asterisk now, we are heading towards code freeze and production of a new stable release. We do need help testing all new features, finding bugs, reporting them, fixing them. The new realtime architecture is a major improvement and a good platform for a lot of new future technology in Asterisk. We need it tested and proven before we release version 1.2. Thank you for your support in creating a new version of Asterisk -the Open Source PBX! Regards, /Olle ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] ATA's
[...] In the meantime, get a Sipura 2100, supports 2 729 calls and has both WAN/LAN ports. I was told that the Uniden DTA200 also supports 2 g729 calls. I'm buying one to test. Street price around US$ 90. Another one with dual g729 channels is MTA V102. Street price US$ 100. Also will test this one. I'm still looking for other units with dual g729 channels... yoda.com.tw has single, dual and quad channel ATAs, and AFAIK they support all channel codecs individually. roy ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] DTMF inband detection improvement
On Feb 16, 2005, at 10:34 AM, Steve Underwood wrote: BTW, Steve, if you're still reading, what is the RADIO_RELAX option intended to be for in dsp.c? It is something someone else added to the code to make the detection criteria in relaxed mode even more relaxed. If setting that helps, something in your channel must be causing some serious filtering of low frequencies. Can you try logging the audio to a file, and send it to me for analysis? chan_spy, or something like that, should do the job. Actually, it was Florian that posted about this option. I haven't tried it (spent an awful lot of time last week compiling different configurations of stable, head, patches...taking a break this week). This is Florian said: On Feb 15, 2005, at 1:12 PM, Florian Lefeuvre wrote: I find the compilation option RADIO_RELAX. this option change a threshold in DTMF detection (function dtmf_detect in dsp.c) I remark an big improvement in the detection of the dtmf over GSM. have you ever test this option? RADIO is obscur for me, does it mean all wireless device? Florian -mark Hi Steve, I was the one who post a question about the RADIO_RELAX option. In fact when I set it , I remark some better result in the detection of the DTMF... after a few more tests, It appears I was wrong. I did a record of samples used by the DTMF_detect function. I obtain an audio file : PCM , 16 bits signed, big endian, Fs 8kHz. If I compare an audio file of DTMF generated by land line with one generated by GSM phone, I remarks a big difference. for a land line, the shape is very good., amplitude is nearly constant for a dtmf. for a gsm phone, the shape is bad, it can can an amplitude 5 times bigger than a land line. After some tests, I see that lot's of errors occurs when the signal amplitude was too big (saturation). I wonder if I should clip or attenuate the signal or a better detection... if you want I can send you some file Florian . ___ Asterisk-Users mailing list Asterisk-Users@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] change the caller id number
Hello,
I have this configuration
Cisco 2600 SER Asterisk
When I receive a call on asterisk from ser then I dial
2 different extensions a${EXTEN} and b${EXTEN} but I can not set correctly the
caller id number.
When I make a dial asterisk set caller id name and
number to asterisk. I can change the name with SetCIDName but it
is not working for the caller Id number, I have tried to use SetCIDNum.
Laurent
U x.x.x.213:5060 - x.x.x.215:5060
INVITE sip:0244202372@ x.x.x.215
SIP/2.0..Record-Route: sip:0244202372@
x.x.x.213;ftag=4E95FB8E-264A;lr=on..Record-Route: sip:4202372@
x.x.x.213;ftag=4E95FB8E-264A;lr=on..Via: SIP/2.0/
UDP x.x.x.213;branch=z9hG4bK2f25.b2aa926.0..Via:
SIP/2.0/UDP x.x.x.214;branch=z9hG4bK2f25.3c1833e7.0..Via: SIP/2.0/UDP
x.x.x.213;branch=z9hG4bK2f25.a2aa926.0..Via: SIP/2.0/UDP x.x.x.170:5060..From:
sip:[EMAIL PROTECTED];tag=4E95FB8E-264A..To: sip:[EMAIL PROTECTED]..Date:
Tue, 04 May 1993 23:16:59 GMT..Call-ID:
[EMAIL PROTECTED]:
timer,100rel..Min-SE: 1800..Cisco-Guid:
1245359493-1207701964-2264842207-4113283075..User-Agent:
Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK,
COMET, REFER, SUBS
CRIBE, NOTIFY, INFO..CSeq: 101 INVITE..Max-Forwards:
3..Timestamp: 736557419..Contact:
sip:[EMAIL PROTECTED]:5060..Expires: 180..Allow-Events:
telephone-event..Content-Type: application/sdp..C
ontent-Length: 360..P-hint: usrloc
appliedv=0..o=CiscoSystemsSIP-GW-UserAgent 3339 2725 IN IP4
x.x.x.170..s=SIP Call..c=IN IP4 x.x.x.170..t=0 0..m=audio 16916 RTP/AVP 18 4 0
8 101..c=IN I
P4 x.x.x.170..a=rtpmap:18 G729/8000..a=fmtp:18
annexb=no..a=rtpmap:4 G723/8000..a=fmtp:4 annexa=yes..a=rtpmap:0
PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101
0-16..
U x.x.x.215:5060 - x.x.x.213:5060
SIP/2.0 100 Trying..Via: SIP/2.0/UDP
x.x.x.213;branch=z9hG4bK2f25.b2aa926.0..Via: SIP/2.0/UDP
x.x.x.214;branch=z9hG4bK2f25.3c1833e7.0..Via: SIP/2.0/UDP
x.x.x.213;branch=z9hG4bK2f25.a2aa
926.0..Via: SIP/2.0/UDP x.x.x.170:5060..From:
sip:[EMAIL PROTECTED];tag=4E95FB8E-264A..To:
sip:[EMAIL PROTECTED];tag=as1ad1575c..Call-ID:
[EMAIL PROTECTED]: 101 INVITE..User-Agent:
Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Contact:
sip:[EMAIL PROTECTED]..Content-Length: 0
U x.x.x.215:5060 - x.x.x.213:5061
INVITE
sip:[EMAIL PROTECTED]:5060 SIP/2.0..
Via:
SIP/2.0/UDPx.x.x.215:5060;branch=z9hG4bK6d41ce89;rport..
From: asterisk sip:asterisk@x.x.x.215;tag=as68aa6c65..
To:
sip:a0244202372 @10.192.72.197:5060..
Contact:
sip:asterisk@x.x.x.215..
Call-ID: [EMAIL PROTECTED].
CSeq: 102
INVITE..User-Agent: Asterisk PBX..Date: Thu, 17 Feb 2005 10:29:46 GMT..
Allow: INVITE,
ACK, CANCEL, OPTIONS, BYE, REFER..
Content-Type:
application/sdp.
Content-Length:
369...
v=0..o=root
32675 32675 IN IP4 x.x.x.215..s=session..
c=IN IP4
x.x.x.215..
t=0 0..
m=audio 19062 RTP/AVP 0 8
4 18 3 101.
a=rtpmap:0
PCMU/8000.
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Re: [Asterisk-Users] Reccomendation for reliable handsets
I wouldn't recommend the grandstreams, I had very bad experience using the grandstream 102, It kep locking up on me. The buttons are very bad buttons. The sound quality is just as bad. grandstream barbie^H^H^H^H^Hudgettone phones really sucks. they're cheap, and that's it roy ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] 4xHFC-s cards vs 1 quadbri HFC-4S card ?
I wonder what makes the difference between inserting 4 HFC-S cards (cca. 120 EUR) and using 1 QuadBRI card (approx. 700 EUR) ? What makes such difference ? Is it possible to do first configuration ? With what drivers ? Is it stable ? 1 HFC-S card - lots of interrupts 4 cards - interrupt havoc 1 QuadBRI - some interrupts, but not too many ___ Asterisk-Users mailing list Asterisk-Users@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] Call termination database
I've been considering doing a web based database system, where you can post your termination offerings or wanted, then search by location, price, minimum volumes, etc. I'd probably make it free, supported by advertising my consulting company, or Google Adwords, or something like that. I've got the design written down, all ready to start coding. I could probably have a prototype up and running in a couple of weeks, consulting work permitting. Would people be interested in such a site? What features would people like to see? In particular, how detailed would you like locations to be? Countries? Regions / States? Individual area codes? Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Michael Welter wrote: Danny N wrote: We don't have a lot of traffic yet. But I am looking for a flat rate of US$0.008 per min to US and Canada termination. Yes, we can work out a prepayment arrangement initially, and extend an 7-14 days term once we establish a good business relationship. You may email me offline for your proposal and coverage. Danny I'm looking as well. Should someone set-up a wiki page for LD vendors? A-Z termination vendors? ___ Asterisk-Biz mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-biz ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] solid-state asterisk pbx?
On 16/02/2005 at 09:00 Michael Graves wrote: Andy Powell has prepared a CF image at www.automated.it/asterisk. I have been able to get this booted on a testbed system. Sadly, I'm a Linux newbie and not skilled at command line administration, thus I'm stuck at the moment. I can get the existing image running, but have not been able to get ssh working, change passwords, load my configs to the CF, etc. If there's someone on-list who could assist in this regard I'd gladly share my experience moving my production server to be CF based. Michael lo, If you are using dhcp for the test box then login and type dhcpcd then to start ssh... sshd to change your password... passwd You need to copy the password files back to the cf so that it'll be copied back at boot. To mount the 3rd partition (where the configs live): mount /dev/hda3 /mnt/cfgs remember to umount it when you're done. There are some example configs on hda3 (hint rename the examples folder to just astlive - AFTER you edit the network stuff etc :D) quick command reference: cd = cd cp = copy rm = delete ls = dir mv = move/rename longer command reference: http://docsrv.sco.com/DOS_others/Going_from_dO_to_u1.html HTH Andy ___ Asterisk-Users mailing list Asterisk-Users@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] SIP address formatting problem for outbound calls going through proxy
Hi,
I have a problem when configuring Asteriskand SER,
using SER as a simple SIP gateway. SER connects to
another third party SIP server. I want to call a user
that is registered in the third party SIP server, from
asterisk. In order to achieve this, I defined a peer
in sip.conf, as follows:
[sip4_out]
type=peer
secret=asterisk1
username=asterisk1
fromuser=asterisk1
insecure=no
context=home
host=10.2.250.151
fromdomain=antero.ssf.pt
port=5070
Then, I defined an outbound rule in extensions.conf:
exten = _77.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
At this point, I am able to dial out SIP addresses,
using my SER and a SIP client. My problem is that ALL
the addresses that I dial are formatted in the SIP
INVITE message with the IP address and port of my SIP
peer (sip4_out), in the TO: field.
For example, if I dial [EMAIL PROTECTED] from my SIP
client, Asterisk will format the TO: field as:
To: sip:[EMAIL PROTECTED]:5070
Is there anything we can do in the configuration files
to make Asterisk format that field with the actual SIP
address that was dialed? That is:
To: sip:[EMAIL PROTECTED]
Many thanks for your help,
Paulo
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[Asterisk-Users] CVS in production env (Attended xfer)
Yesterday I asked about a user manual - ie a user guide to actually using asterisk (now on how to set it up) the doc project (v2) isn't anywhere near complete and is the closest thing I could find. Does anyone know of such a doc? The reason I ask is that while a lot of this may be obvious to many people its not to someone new to asterisk and there is a lot of info to trawl through, most of which is related to configuring asterisk. Again, the question of how to do attended xfers - how? (I now know I need asterisk from CVS) but what key presses? Then there is the issue of CVS in a production environment? I'm guessing people are actually doing this, but it goes against my better judgment. Does a roadmap for asterisk exist anywhere? If there was a roadmap I wouldn't need to ask when the next stable release will be available. By stable I don't mean that I think the CVS code isn't going to be reliable, (but that is a concern) but that the code isn't going to be changing constantly. ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Teles PCI and chan_capi, possible ???
Is there an easier way to cancel the echo ? Is there a way to use chan_capi with Teles cards ? Hi, If your cards are supported by i4l, the odds on support in mISDN are good. mISDN provides a CAPI interface for the cards. Maybe you should check that out. My experience with the echo on i4l is the same and it all went away once switched to CAPI. Good luck. Mr. Michiel, Thank you for setting me in the right direction again. Is there a way to use mISDN in kernel version 2.4.xx ?? Or is it mandatory to use ver. 2.6 ? If so, is there a way to upgrade my (knoppix) Debian with kernel 2.4 to the new kernel 2.6 without erasing and installing everything from scratch ? Yes, I'm still a newbie in linux... :) Like you said, there's a good chance my Teles card can work with mISDN because it uses the cologne chip like the HFC cards do. It doesn't support NT mode but I don't need it anyway... Thanks again Miguel Gonçalves ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] problem : undefined symbol.
Kim Daeyong wrote: I downloaded asterisk to use cvs to checkout the release version. After installing, I would like to load module chan_h323.so but there is some error : *CLI load chan_h323.so Feb 17 15:22:38 WARNING[2865]: loader.c:258 ast_load_resource: /usr/lib/asterisk/m odules/chan_h323.so: undefined symbol: __use_ast_pthread_create_instead__ Unable to load module chan_h323.so *CLI How can I solve that problem? Did you try asterisk-oh323? http://www.inaccessnetworks.com/projects/asterisk-oh323 Michael. ___ Asterisk-Users mailing list Asterisk-Users@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] can't enable trunking :(
I have successfully installed and configured the asterisk, the incoming and the outgoing calls are working fine, its a tremendous solution :) Now i want to enable trunking between two asterisk boxes, in the iax.conf i have put: [karachi] ... ... ... trunk=yes ... ... ... everything seems to work fine but when i load asterisk it says: -- Feb 17 10:59:14 WARNING[18726]: chan_iax2.c:7536 build_user: Unable to support trunking on user 'karachi' without zaptel timingFeb 17 10:59:14 WARNING[18726]: chan_iax2.c:7345 build_peer: Unable to support trunking on peer 'karachi' without zaptel timing -- I tried to install the ztdummy and i succeeded on one of the box but for the other i am having problems :( It was missing the kernel-source rpm. I installed the version that i found but now the first error is still there and when i modprobe ztdummy it gives the following response. --- [EMAIL PROTECTED] asterisk]# modprobe ztdummy/lib/modules/2.4.25-040218/misc/zaptel.o: kernel-module version mismatch /lib/modules/2.4.25-040218/misc/zaptel.o was compiled for kernel version 2.4.20-24.9 while this kernel is version 2.4.25-040218./lib/modules/2.4.25-040218/misc/zaptel.o: insmod /lib/modules/2.4.25-040218/misc/zaptel.o failed/lib/modules/2.4.25-040218/misc/zaptel.o: insmod ztdummy failed[EMAIL PROTECTED] asterisk]# -- Any Kind peace of information will be highly appriciated :) Best Regards Muhammad Muzzamil Luqman ___ Asterisk-Users mailing list Asterisk-Users@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] Voicemail and busy tone
Hi everybody, I have aproblem with voicemail: I have two TDM boards for a total of 5 fxs and 3 fxo. One of thefxo is connectedto the local tel provider and is redirected to a voicemail box. When I call asterisk from outside, I leave my message, but, after hanging on, voicemail continues to record the busy tone that the provider sends. How can I avoid this behaviour? Thx. Thomas. ___ Asterisk-Users mailing list Asterisk-Users@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] Error loading wcfxs module
Hello, I recently instaled an asterisk 1.0.3, libpri 1.0.1 and zaptel 1.0.3 withuot errors. When i try to load the modules, i get, modprobe zaptel - load zaptel without errors. modprobe wcfxs - can't locate wcfxs I search for wcfxs location, and it is on /lib/modules/2.4.20/misc/ like zaptel. Why my system can't find the wcfxs module but it is at the same place that zaptel, that load without errors? Any clue will be wellcome. Thanks. Ismael. ___ Asterisk-Users mailing list Asterisk-Users@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] started asterisk with chan_misdn
hello, i have a problem on started asterisk, when try to start asterisk a get the fowlling error: chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri)) Feb 17 11:34:01 WARNING[3104]: config_old.c:27 ast_load: ast_load is deprecated, use ast_config_load instead! == Parsing '/etc/asterisk/misdn.conf': Found Feb 17 11:34:01 WARNING[3104]: config_old.c:39 ast_destroy: ast_destroy is deprecated, use ast_config_destroy instead! == Registered channel type 'mISDN' (This driver enables the asterisk to use hardware which is supported by the new ) Bidx 0 -- Child 1101 Bidx 1 -- Child 1201 No Upper ID my lsmod: Module Size Used by hfcpci 28716 0 mISDN_dsp 197248 0 l3udss132008 0 mISDN_l2 38272 0 mISDN_l1 10632 0 mISDN_core 77732 5 hfcpci,mISDN_dsp,l3udss1,mISDN_l2,mISDN_l1 md5 4352 1 ipv6 235840 26 parport_pc 25024 1 lp 12396 0 parport42696 2 parport_pc,lp dm_mod 55444 0 uhci_hcd 31896 0 3c59x 36776 0 floppy 59568 0 ext3 116744 2 jbd74904 1 ext3 and i do modprobe hfcpci protocol=0x2 layermask=0xf ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] video conferencing bounty
Good day Dean, I am interested in developing the video conferencing capability. I am going to look over the request during the following two days in order decide defnitively. Can you tell me if your offer still stands? Herman Webley ___ Asterisk-Users mailing list Asterisk-Users@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] asterisk functions without voIP
Dear friends, Can i use the Asterisk functions (call recognition for example), using conventional telephony (in Brazil) ? Thanks in advace Pablo Fernandes ___ Asterisk-Users mailing list Asterisk-Users@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] VoipJet issues?
Whats up to VoipJet.com? Their DNS servers are not reachable. Both primary and secondary are on the same subnet - weird setup. Thanks, David ___ Asterisk-Users mailing list Asterisk-Users@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] (no subject)
The problem was this line at the end of modules.conf alias wcfxs wctdm Ismael. ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] VoipJet issues?
Whats up to VoipJet.com? Their DNS servers are not reachable. Looks like their provider is maybe having problems. AS3728, onr.com, Onramp. Both primary and secondary are on the same subnet - weird setup. While that might be true, it also might not be. 206.55.64.64 and 206.55.64.65 are not on the same network or even the same city, for example (they used to actually be in different states). We use OSPF internally and those addresses are not on any Ethernet network. They're loopback interfaces. They can be moved around. In the case you're talking about, it's *likely* they're on the same network, and that's not good, of course. Those pesky rules about diversity of nameservers exist for a reason. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ Asterisk-Users mailing list Asterisk-Users@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] Re: Voicemail and busy tone
Thomas == Thomas RULMONT [EMAIL PROTECTED] writes: Thomas When I call asterisk from outside, I leave my message, but, Thomas after hanging on, voicemail continues to record the busy tone Thomas that the provider sends. How can I avoid this behaviour? First of all, try to isolate the problem by doing the same experiment without voicemail: have someone call you from outside, you answer the phone, she hangs up while you stay off hook. Look whether asterisk detects that the remote party has hung up or not. Sam -- Samuel Tardieu -- [EMAIL PROTECTED] -- http://www.rfc1149.net/sam ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] chan_sip errors on CVS HEAD
Haven't had it since, so it's hard to try debug :( Julian. Olle E. Johansson wrote: Asterisk wrote: I've got a test * server (hppbx) where I install CVS-HEAD as often as possible, with my extension registered to this, talking through IAX to our production server which then channels out to the PSTN. After completing a call just now, the following appeared on the CLI of hppbx (the 90xxx is a valid number, changed to protect the guilty): == Spawn extension (from-sip, 90xxx, 1) exited non-zero on 'SIP/711-31db' Feb 16 12:42:38 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy Feb 16 12:42:39 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy Feb 16 12:42:41 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy Feb 16 12:42:45 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy Feb 16 12:42:49 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy Feb 16 12:42:53 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy Feb 16 12:42:57 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy hppbx*CLI show version Asterisk CVS-HEAD-02/05/05-09:30:42 built by [EMAIL PROTECTED] on a i686 running Linux Feb 16 12:43:01 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy Feb 16 12:43:05 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy Feb 16 12:43:09 NOTICE[14535]: chan_sip.c:2927 copy_header: No field 'From' present to copy hppbx*CLI show channels Channel (ContextExtensionPri ) State Appl. Data 0 active channel(s) There are no more errors after this. Is there a From: header present? Turn on SIP DEBUG and check the SIP packets. /O ___ Asterisk-Users mailing list Asterisk-Users@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@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
Re: [Asterisk-Users] DTMF inband detection improvement
Florian Lefeuvre wrote: Hi Steve, I was the one who post a question about the RADIO_RELAX option. In fact when I set it , I remark some better result in the detection of the DTMF... after a few more tests, It appears I was wrong. I did a record of samples used by the DTMF_detect function. I obtain an audio file : PCM , 16 bits signed, big endian, Fs 8kHz. If I compare an audio file of DTMF generated by land line with one generated by GSM phone, I remarks a big difference. for a land line, the shape is very good., amplitude is nearly constant for a dtmf. for a gsm phone, the shape is bad, it can can an amplitude 5 times bigger than a land line. After some tests, I see that lot's of errors occurs when the signal amplitude was too big (saturation). I wonder if I should clip or attenuate the signal or a better detection... if you want I can send you some file Florian Send me some data, and I will tak a look. If the signal has overloaded it will not detect, as there will be too much harmonic energy to pass the tests. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Using Hylafax and Digium T100P
Any analog modem (fax or pc) is going to be limited to 9600 baud or slower, and will only achieve that speed if g711 is used through the entire path (including asterisk). If a modem call comes in one T1 (or PRI) and goes out another, asterisk is still handling the pcm packets. The packets don't magically jump across T1 cards (or T1 ports on the same card). Rich, I'm not quite sure what you're concluding here, but we routinely fax hylafax-asterisk-hylafax and hylafax-asterisk-PSTN in a variety of analog and digital configurations, with and without channel banks at speeds up to 33,600. There's no reason Lee's outline: T1 -- TE405P(1) -- Asterisk -- TE405P(2) -- Patton 2977 -- HylaFAX won't work perfectly well. Granted the Patton 2997 is limited to 14,400 maximum, but with other T1 cards such as Eicon Diva Server and Brooktrout TR1034 there's no problem negotiating and sustaining V.34 speeds (14,400 -33,600). Forgive me if I musinderstood your post, and wield your clue bat gently! ;-) In the post that I was responding to, the writer hinted his understanding was that T1 to T1 channel connections didn't involve any asterisk code. His impression seemed to suggest that codec selection, etc, wasn't a factor since the analog fax modem signals were coming in one T1 channel (or PRI channel) and going out another without passing across the * pci bus. (Purhaps I've have read too much into his post though.) If the analog modem signals are transitioning the * pci bus, there is a high likelihood the modem signals will not be accurately handled by * and thus limit the speed at which the fax modem will function. Don't read that as it will, but rather it might. (See many past posts relative to analog modem usage through asterisk, and many other posts where readers didn't understand the significance of g711 usage and analog modems via asterisk.) As a sort of side note, modems that operate at rates higher then about 9600 bits/second actually use encoding techniques (such as trellis encoding) on top of a 9600 baud signal (as an example only), thus achieving 28800 or whatever speed. There is a difference in the terms baud and bits/second. The more sophisticated the encoding technique, the more difficult it is to accurately reproduce analog signals. I'm not a fax modem expert by any stretch, but I'm under the impression that fax modem standards are very old and limited to rather slow speeds (on the wire). I would very quickly defer to Steve Underwood for a more accurate description of that entire topic however. Rich ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Using Hylafax and Digium T100P
On Thu, 17 Feb 2005, Rich Adamson wrote: In the post that I was responding to, the writer hinted his understanding was that T1 to T1 channel connections didn't involve any asterisk code. His impression seemed to suggest that codec selection, etc, wasn't a factor since the analog fax modem signals were coming in one T1 channel (or PRI channel) and going out another without passing across the * pci bus. (Purhaps I've have read too much into his post though.) If the analog modem signals are transitioning the * pci bus, there is a high likelihood the modem signals will not be accurately handled by * and thus limit the speed at which the fax modem will function. Don't read that as it will, but rather it might. (See many past posts relative to analog modem usage through asterisk, and many other posts where readers didn't understand the significance of g711 usage and analog modems via asterisk.) I think the data may get to the zaptel driver on a native bridge. I'm not sure if there is a cross-connect in the actual TE405P card. However, barring missed interrupts there should be no frame slips and no signal degradation when passing a call from one T1 to another T1 on the same TE405P card. They all share the same clocking and there should be no slips, missing data etc. Peter ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Re: Voicemail and busy tone
No, it don't. If i call from outside to an inside phone, when I hang up the outside phone, I hear the busy tone on the inside phone. - Original Message - From: Samuel Tardieu [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, February 17, 2005 1:22 PM Subject: [Asterisk-Users] Re: Voicemail and busy tone Thomas == Thomas RULMONT [EMAIL PROTECTED] writes: Thomas When I call asterisk from outside, I leave my message, but, Thomas after hanging on, voicemail continues to record the busy tone Thomas that the provider sends. How can I avoid this behaviour? First of all, try to isolate the problem by doing the same experiment without voicemail: have someone call you from outside, you answer the phone, she hangs up while you stay off hook. Look whether asterisk detects that the remote party has hung up or not. Sam -- Samuel Tardieu -- [EMAIL PROTECTED] -- http://www.rfc1149.net/sam ___ Asterisk-Users mailing list Asterisk-Users@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@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
Re: [Asterisk-Users] VoipJet issues?
Anyway, they're not reachable since yesterday evening. -D Joe Greco wrote: Whats up to VoipJet.com? Their DNS servers are not reachable. Looks like their provider is maybe having problems. AS3728, onr.com, Onramp. Both primary and secondary are on the same subnet - weird setup. While that might be true, it also might not be. 206.55.64.64 and 206.55.64.65 are not on the same network or even the same city, for example (they used to actually be in different states). We use OSPF internally and those addresses are not on any Ethernet network. They're loopback interfaces. They can be moved around. In the case you're talking about, it's *likely* they're on the same network, and that's not good, of course. Those pesky rules about diversity of nameservers exist for a reason. ... JG ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] can't enable trunking :(
On February 17, 2005 06:08 am, Muhammad Muzzamil Luqman wrote: Feb 17 10:59:14 WARNING[18726]: chan_iax2.c:7536 build_user: Unable to support trunking on user 'karachi' without zaptel timing Feb 17 10:59:14 The answer's pretty simple -- do you have a zaptel timing source? i.e. X100P, T100P, TDM400P, TE410, Zaptel USB device...? If not, you'll need one, or you'll need to try and get ztdummy or ztrtc to work. Trunking requires a hardware timing source that the Zaptel devices provide... -A. ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] [patch] fix libpri problem in Q931_INFORMATION handling
Peter Svensson wrote: What is c-ourcallstate set to at this time? Can you provide a debug log (pri intense debug span xxx) of the call? it's Q931_CALL_STATE_ACTIVE - that's what it should be after a call is established. Asterisk only expects INFORMATION elements when expecting overlap digits (i.e. before CONNECT, PROCEEDING etc). After that it expects digits as inline dtmf. Yep - but ISDN phones normally do not encode inline DTMF. Therefor Keypad information can be sent. According to Q.931 called party address IEs in the INFORMATION message should only be sent during overlap digit transmission, which ends when PROCEEDING is sent. Well I have read the Q.931 specification too but for eg. Siemens HiCom PBXs and phones use keypad IEs within a connected call and no DTMF. This leads me to at least invent a new configuration parameter for ignoring the call state when receiving such IEs. Any other ideas? Deti ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] DIAX 0.9.10d with Eutectics USB phone suport
Hi Dan. ' - audio delay when IAX bridging inside Asterisk Will it cover that problem of long delays that we talked before!? Regards, Denis Galvão. ___ Asterisk-Users mailing list Asterisk-Users@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] Sirrix ISDN Card
How do I test if the card is working or not ? Is there something that I can do to get a response from the card ? Ive put the card in, installed drivers ect but can't dial out and can't see a response when I try dial in from external number. Any ideas ? Thanks Shaun --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.857 / Virus Database: 584 - Release Date: 10/02/2005 ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Using Hylafax and Digium T100P
In the post that I was responding to, the writer hinted his understanding was that T1 to T1 channel connections didn't involve any asterisk code. His impression seemed to suggest that codec selection, etc, wasn't a factor since the analog fax modem signals were coming in one T1 channel (or PRI channel) and going out another without passing across the * pci bus. (Purhaps I've have read too much into his post though.) If the analog modem signals are transitioning the * pci bus, there is a high likelihood the modem signals will not be accurately handled by * and thus limit the speed at which the fax modem will function. Don't read that as it will, but rather it might. (See many past posts relative to analog modem usage through asterisk, and many other posts where readers didn't understand the significance of g711 usage and analog modems via asterisk.) I think the data may get to the zaptel driver on a native bridge. I'm not sure if there is a cross-connect in the actual TE405P card. However, barring missed interrupts there should be no frame slips and no signal degradation when passing a call from one T1 to another T1 on the same TE405P card. They all share the same clocking and there should be no slips, missing data etc. I don't have a 405P card here to test, but I'm fairly certain the card does not have any onboard logic to cross-connect channels, implying all cross-connects happen on the zaptel/asterisk side of the pci bus. Given the track history of missed interrupts (etc), its fair to adjust the expectations from will work to might work. In a recent discussion with technical folks at Supermicro relative to pci latency issues, comments like Latency is the biggest issue since Intel is using I/O hubs on most of their products imply the later chip sets are worse then older ones (for whatever reasons). Their lowest latency motherboard recommendation is actually a two year MB. I would not be a happy camper if I invested in 405P cards, etc, with the expectation that fax will work and then find out later it doesn't, followed by posts that we've all seen relative to get a decent MB. ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] video conferencing bounty
Hi Herman, yes the offer still stands but I really need to see something soon otherwise I'm going to go out and buy the macromedia communications server solution and run it is as a separate standalone application to my Asterisk voice conferencing server. I have had one other email 2 days ago from someone else interested in working on it but like I said before I had 5 people with the best of intentions get involved since I posted the bounty only to never hear back from anyone. Cheers, Dean -Original Message- From: Herman Webley [mailto:[EMAIL PROTECTED] Sent: Thursday, February 17, 2005 6:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; dean collins Subject: Re: [Asterisk-Users] video conferencing bounty Good day Dean, I am interested in developing the video conferencing capability. I am going to look over the request during the following two days in order decide defnitively. Can you tell me if your offer still stands? Herman Webley ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Reccomendation for reliable handsets
Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote: I wouldn't recommend the grandstreams, I had very bad experience using the grandstream 102, It kep locking up on me. The buttons are very bad buttons. The sound quality is just as bad. grandstream barbie^H^H^H^H^Hudgettone phones really sucks. they're cheap, and that's it roy That is very strange. I have one I just received a Grandstream BT-100 last Friday and hooked it up on Saturday. Flashed the firmware up to 1.0.5.22, I think, I know the .22 is correct, and it has been working flawlessly since. No lock ups and great sound quality. The only issue I had was with caller-id and that was my FUBAR as I overlooked removing the setting in sip.conf that manually set it to not be the incoming. Robert ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Call termination database
Sounds very interesting, would providors be willing to insert pricing or would you need to enter all the data? I would suggest a set of rules like pricewatch.com uses to keep people honest. Keep us informed, Cheers, Jonathon On Thu, 17 Feb 2005 10:29:54 +, Alistair Cunningham [EMAIL PROTECTED] wrote: I've been considering doing a web based database system, where you can post your termination offerings or wanted, then search by location, price, minimum volumes, etc. I'd probably make it free, supported by advertising my consulting company, or Google Adwords, or something like that. I've got the design written down, all ready to start coding. I could probably have a prototype up and running in a couple of weeks, consulting work permitting. Would people be interested in such a site? What features would people like to see? In particular, how detailed would you like locations to be? Countries? Regions / States? Individual area codes? Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Michael Welter wrote: Danny N wrote: We don't have a lot of traffic yet. But I am looking for a flat rate of US$0.008 per min to US and Canada termination. Yes, we can work out a prepayment arrangement initially, and extend an 7-14 days term once we establish a good business relationship. You may email me offline for your proposal and coverage. Danny I'm looking as well. Should someone set-up a wiki page for LD vendors? A-Z termination vendors? ___ Asterisk-Biz mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-biz ___ Asterisk-Users mailing list Asterisk-Users@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@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
Re: [Asterisk-Users] Re: Voicemail and busy tone
No, it don't. If i call from outside to an inside phone, when I hang up the outside phone, I hear the busy tone on the inside phone. - Original Message - Thomas When I call asterisk from outside, I leave my message, but, Thomas after hanging on, voicemail continues to record the busy tone Thomas that the provider sends. How can I avoid this behaviour? First of all, try to isolate the problem by doing the same experiment without voicemail: have someone call you from outside, you answer the phone, she hangs up while you stay off hook. Look whether asterisk detects that the remote party has hung up or not. Sounds like your having a problem with pstn disconnect supervision. Not sure how disconnect supervision works in your country, but here in the US it is an opening of the pstn line (no voltage) for something less then a second. Some countries use tones for this purpose. You'll need to find out what the standard is and (hopefully) set asterisk to recognize that method. ___ Asterisk-Users mailing list Asterisk-Users@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] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *
Welcome to the Asterisk users community! Asterisk is the leading Open Source Telephony platform, with support both for classical telephony and IP telephony. Asterisk.org is a fast moving project. New code is added every day. Our community is also growing fast and we're having a lot of interaction, on the IRC and on the mailing lists. It's great to have you participating in this Open Source project - building an Open Source PBX. Here are a few things to know and remember while working with the project. Again, welcome to the Asterisk.org Open Source PBX Project! If you want to get up to speed quickly and plan to visit the Voice on the Net conference in San Jose or live in California, don't miss the Asterisk pavillion where you will meet Digium and Digium partners. Also, on Friday the 11th there will be a one-day Asterisk tutorial called Meet Asterisk - http://www.astricon.net Meet you on the IRC channel :-), the bug tracker or on the mailing list! /oej ** Asterisk version information At this moment we have two current versions of Asterisk, the developer version and the stable version. The stable version is distributed as .tar.gz archives on several servers. The current stable version of Asterisk is 1.0.5. The stable version contains no new functions and only changes when bugs are fixed. The development version is to be used by people that can test new functions and live with bugs and unexpected shortcomings. The development version is branded 1.1 and will be the basis for the next stable version, version 1.2. We will hopefully soon reach a code freeze and start testing the stability of version 1.1, so we will need your help. ** The mailing list is growing Today, we propably have over 10,000 readers on the -users list. This means that everything anyone write to this mailing list, is sent to thousands of mailboxes that are already flowing over with messages. That's why we all need to follow some simple rules on how to use the mailing list and the other tools that are available. ** Think before sending a message, think twice I would like to stress the fact that you have to think before you send a message to such a big list. Do *not* send out personal replies on the list. If you offer services to someone, do *not* CC: or reply to the list, it will annoy more potential customers than get you new customers. If you send out a message by mistake, you don't have to apologize to all of us, we understand you're embarassed. We will get more annoyed by your apology than over your first message. And please do not send out test messages to the list. ** Try finding the answer first, then ask the list The Asterisk Wiki at http://www.voip-info.org is an important knowledge base for the project. Go there to find your answer first, then search the mailing list archives (Google or http://search.voip-forum.com) and then go to the IRC channel. The IRC channel is populated with Asterisk gurus around the clock (literally) and they'll help you move forward. * IRC info: http://www.asterisk.org/index.php?menu=support#irc * There's many links to Asterisk web pages on the documentation page at http://www.asterisk.org * The Asterisk FAQ is found on the wiki http://www.voip-info.org/wiki-Asterisk+FAQ * The Asterisk documentation project (which needs your help) is at http://www.asteriskdocs.org Their handbook The hitchhiker's guide to Asterisk is already well worth reading. Finally, if you don't find the answer elsewhere, try the list. ** Mailing lists For developers, there is a developer's list, asterisk-dev. Do not use this list as a secondary support line if you do not get an answer on the -users list. It is meant for developer discussions, not advanced support. If you need answers, there is a better chance that you will get help on the irc channel. For BSD users (FreeBSD, NetBSD, OpenBSD and OS/X) there's a list called asterisk-bsd. There is also a business list for those that want to ask for commercial services and inform their community about new services (asterisk-biz). You'll find all lists on http://lists.digium.com, which is the site where you manage your subscription to this list as well. Please, do not crosspost the same message to multiple mailing lists. It will not help you, it will only add to the mail flow and get people that read both lists irritated. If you are unsure which list to use, send only to the -users list. Make sure that you remove unnecessary text when you reply, to make it easy to browse the mailing list quickly. And please do not send HTML mail to a mailing list. ** Reporting bugs If you think you have found a bug, report it. We need bug reports. Read this document http://www.digium.com/bugtracker.html and then go to the bugtracker http://bugs.digium.com to file a report. If you are unsure, find a bug marshal on the IRC channel to help you. They're appointed to support you with how to handle bugs. Please check the bugtracker thoroughly before posting a new
[Asterisk-Users] Having trouble with extensions in an include file and retrieve_extensions_from_mysql.pl
Folks, I've been running asterisk successfully using the extensions.conf and voicemail.conf. Now that I've got asterisk happily looking up MySQL tables for the VM configuration, I decided to try out the contributed script /usr/src/asterisk/contrib/scripts/retrieve_extensions_from_mysql.pl I edited the script so that its output goes to a separate extensions_from_mysql.conf file. The resulting extensions_from_mysql.conf file looks something like this: [vp_context] exten = 1000,1,Record(/tmp/rec:gsm); exten = 1000,2,Playback(/tmp/rec) ; exten = 1000,3,Background(goodbye) ; exten = 1000,4,Hangup(); I decided to #include this in my main extensions.conf, like so: [main_vp_context] exten = s,1,Answer #include extensions_from_mysql.conf exten = #,1,Background(goodbye) ; Notify caller exten = #,2,Hangup() ; Hang up exten = t,1,Hangup() ; Hang up if timeout exten = i,1,Playback(invalid) ; Play invalid ; extension if caller ; misdials an extension Basically, I expect asterisk to load the two as separate contexts, and I could swear that it used to. In fact, when I set the verbosity higher, asterisk is definitely still loading them as separate contexts. As of yesterday, though, when I have this format, asterisk won't accept incoming calls. It barfs with the message: Feb 16 21:53:14 NOTICE[4330]: chan_iax2.c:5757 socket_read: Rejected connect attempt from 66.234.228.170, request '[EMAIL PROTECTED]' does not exist The only way to get asterisk to receive calls again is to edit the included file to ensure it does not have a context line in it. So I commented out the line where the retrieve_extensions_from_mysql.pl sticks the context information into the created file. Now, it all works fine. But it's no good. What about when I want to have a sip.conf and have a list of extensions that do different things in the sip context? I really like the contributed script for its ability to add multiple context sections. Anyone see a possible reason for the problem? Do you have any ideas how to use an include file which contains multiple contexts? Or will I have to generate multiple include files, one per included context, without the context lines in these files? Thanks for any help! Cheers, Maya __ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Zaptel DACS and FDL
On Feb 16, 2005, at 7:19 PM, Eric Wieling wrote: Jerry wrote: On Feb 16, 2005, at 3:07 PM, Eric Wieling wrote: I have the following configuration: CLEC - T-1 - Asterisk - Adtran Channel Bank - (analog) - Nortel Don't complain that it's ugly. I've already done plenty of that. The CLEC manages their Adtran remotely and needs to be able to continue to do so. I assume they use FDL to do the management. We are using the Zaptel DACS/DACSRBS to cross connect some of the channels directly between the CLEC and the Adtran. These are channels we don't really care about (data, other voice, etc). We are not cross connecting all the channels, just some of them. I'm wondering if/how I can make sure the remote management via FDL continues to work. Does anyone have any information on this or suggestions or anything? What you dropped from your diagram is the T1 from * to the channel bank. FDL is a link level protocol. It is carried in the framing bits of the T1 not within the payload bits. When you use the digium (I presume) card within your * server as a DACS this is connecting the payload, ie timeslot, bits from one port to another. FDL will not work this way. DACSRBS does DACS the robbed bit signalling. I assume this won't help? Ah well. We'll try to find some other way to do Robbed bit refers to robbing a bit from the payload - hence the name - and the reason you can only get a 56k channel on links utilizing this form of signalling - vs 64k when not. ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Sirrix ISDN Card
Hi! How do I test if the card is working or not ? Is there something that I can do to get a response from the card ? Ive put the card in, installed drivers ect but can't dial out and can't see a response when I try dial in from external number. Did you configure groups in sirrix.conf? See http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20sirrix.conf for examples. You need to set the number parameter correctly or simply set number = + to receive any call on any number on that port. IF you have configured ports in sirrix.conf and you startup Asterik that loads chan_sirrix, you should see some output in /var/log/messages like Enabling interrupts from ISACx. When you have configured ports, the LEDs near the RJ45 sockets shall light up as soon as Layer 1 is activated (eg. phone is plugged in). Then you can enable debugging output from Asterisk (logger.conf) and see some details (e.g. when phone requests channel or incoming call is detected), probably you need to start Asterisk in console mode (asterisk -vc). I hope this could help you. If you have further questions about the Sirrix.PCI4S0 feel free to contact me by eMail [EMAIL PROTECTED] Regards, Oskar. ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] DIAX 0.9.10d with Eutectics USB phone suport
Hi Denis, - Original Message - From: Denis Galvão - iSolve [EMAIL PROTECTED] ' - audio delay when IAX bridging inside Asterisk Will it cover that problem of long delays that we talked before!? Yes, with a small remark. In some situations is possible to loose the audio for the first 2-3s of a call. Best regards, Dan ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] asterisk functions without voIP
Pablo Fernandes wrote: Can i use the Asterisk functions (call recognition for example), using conventional telephony (in Brazil) ? Generally, yes. (VOIP is just a cool thing to be into these days.) Can you define call recognition for me? Do you mean CallerID(determining the phone number that is calling you)? You will need hardware that is compatible with your areas telephone network. (Stating that as it is likely different from the US network.) If digital voice circuits(in any form) are available in your area, you'll likely be happier using them than POTS lines. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] [patch] fix libpri problem in Q931_INFORMATION handling
On Thu, 17 Feb 2005, Deti Fliegl wrote: Peter Svensson wrote: Asterisk only expects INFORMATION elements when expecting overlap digits (i.e. before CONNECT, PROCEEDING etc). After that it expects digits as inline dtmf. Yep - but ISDN phones normally do not encode inline DTMF. Therefor Keypad information can be sent. Ok, then INFORMATION with keypad IE needs to be handled differently from IE called number. According to Q.931 called party address IEs in the INFORMATION message should only be sent during overlap digit transmission, which ends when PROCEEDING is sent. Well I have read the Q.931 specification too but for eg. Siemens HiCom PBXs and phones use keypad IEs within a connected call and no DTMF. This leads me to at least invent a new configuration parameter for ignoring the call state when receiving such IEs. I read the ets 300 403 01 spec as well as a more reacent revision of the Q.931 spec. Q.931 allows the keypad IE to signal called party number (during call setup) or to convey supplementary service information (pushed digits similar to dtmf). Q.931 allows the called party number IE to signal called party digits during call setup. The EuroISDN spec. in ets 300 403 01 is stricter - the keypad IE can not be used for overlap digits, only after the call setup. Are the digits you send encoded as Keypad IE? In that case a setting to allow keypad IE digits to always be accepted as dtmf digits may be the best solution. Peter ___ Asterisk-Users mailing list Asterisk-Users@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] Strange MSN issue with HFC-s
Hi, I have two HFC-s boards I configured in NT and TE mode respectively. When I connect the two boards together, I can dial extensions and I see the correct called and caller ID numbers: -- Executing SetCallerID(Zap/2-1, 7516862) in new stack == CDR updated on Zap/2-1 -- Executing Dial(Zap/2-1, Zap/g2/0795025602|30|r) in new stack -- Called g2/0795025602 -- Extension '0795025602' in context 'isdn-local-bus' from '7516862' does not exist. Rejecting call on channel 0/1, span 1 however, when I connect the TE card to my NT2ab, it seems the caller ID number is not passed correctly (?) to the telco, since if when I get the call on my mobile I get the main number for my ISDN connection, not the specified number. With an AVM c4 it works correctly (syntax is CAPI/FROM:TO). Thank you for any help! ___ Asterisk-Users mailing list Asterisk-Users@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] Brand New Digium T100P for sale
We have a brand new T100P that has never been used for sale. We purchased this card from NETXUSA and then decided to use an external VoIP gateway. So I have this unit for sale. Price: $450.00 plus shipping. If interested, please reply off list. Ty Carter, President Strategic Network Consultants, Inc. 524 East 9th Street Washington, NC 27889 252-946-0351 - Voice 252-402-5296 - Cell [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] problem : undefined symbol.
Kim Daeyong wrote: I downloaded asterisk to use cvs to checkout the release version. After installing, I would like to load module chan_h323.so but there is some error : *CLI load chan_h323.so Feb 17 15:22:38 WARNING[2865]: loader.c:258 ast_load_resource: /usr/lib/asterisk/m odules/chan_h323.so: undefined symbol: __use_ast_pthread_create_instead__ Unable to load module chan_h323.so *CLI How can I solve that problem? Exactly which version did you download? (What did you type into your CVS statement?) If asterisk compiled and is runnable other than this error, just log in with asterisk -r and give us the Connected to... line. Using make, there is an option to do a make update, which should download any changes that were tagged to the version of CVS you downloaded. If the problem has been fixed since you downloaded originally, issuing a make update should help. I don't think update recompiles anything, so you will probably have to make install again. I am also not sure if you need to a make clean or anything like that. If that doesn't fix your problem, look in mantis on bugs.digium.com and see if anyone else has reported it. For other readers: I've not had an error like this, so I'm not sure exactly what the protocol is. Should the original poster post next to asterisk-dev, or to mantis? -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ Asterisk-Users mailing list Asterisk-Users@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] Problem with asterisk-addons: libmysqlclient.so.14: cannot open shared object file
Hi, I have compiled asterisk-addons successfully, but when I put res_config_mysql.so in modules directory asterisk fails to load, here is the error: 7:29 WARNING[19097]: loader.c:301 __load_resource: libmysqlclient.so.14: cannot open shared object file: No such file or directory Feb 17 15:17:29 WARNING[19097]: loader.c:509 load_modules: Loading module res_config_mysql.so failed! libmysqlclient is present on the system, should I edit something to point * to the right directory for it or something like ? Tnx for any help! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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] Cyclades-PC300/TE 1 Compatibility?
Title: Cyclades-PC300/TE 1 Compatibility? Hello, Has anyone on this list tried the Cyclades PC300 card with asterisk? Thanks, Brad. ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Having trouble with extensions in an include file and retrieve_extensions_from_mysql.pl
beonice wrote: The resulting extensions_from_mysql.conf file looks something like this: [vp_context] exten = 1000,1,Record(/tmp/rec:gsm); exten = 1000,2,Playback(/tmp/rec) ; exten = 1000,3,Background(goodbye) ; exten = 1000,4,Hangup(); I decided to #include this in my main extensions.conf, like so: [main_vp_context] exten = s,1,Answer #include extensions_from_mysql.conf exten = #,1,Background(goodbye) ; Notify caller exten = #,2,Hangup() ; Hang up exten = t,1,Hangup() ; Hang up if timeout exten = i,1,Playback(invalid) ; Play invalid ; extension if caller ; misdials an extension snip Anyone see a possible reason for the problem? Do you have any ideas how to use an include file which contains multiple contexts? Or will I have to generate multiple include files, one per included context, without the context lines in these files? The only thing that seems out of place to me is your #include in [main_vp_context]. It looks to me like you intend for the s, #, t, and i extensions to be in [main_vp_context]. The way you layed out this example, that's not what is happenning. I think you wanted this: Your extensions_from_mysql.conf should still look like: [vp_context] exten = 1000,1,Record(/tmp/rec:gsm); exten = 1000,2,Playback(/tmp/rec) ; exten = 1000,3,Background(goodbye) ; exten = 1000,4,Hangup(); Then, in extensions.conf: #include extensions_from_mysql.conf [main_vp_context] exten = s,1,Answer exten = #,1,Background(goodbye) ; Notify caller exten = #,2,Hangup() ; Hang up exten = t,1,Hangup() ; Hang up if timeout exten = i,1,Playback(invalid) ; Play invalid ; extension if caller ; misdials an extension include = vp_context This way, you define both contexts, and include the extensions that were defined in [vp_context] into [main_vp_context]. I don't know if this will resolve your other problem, but I believe this is the dialplan you were trying to build. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] asterisk functions without voIP
Hi, If digital voice circuits(in any form) are available in your area, you'll likely be happier using them than POTS lines. yes, here is available Digital voice circuits. You will need hardware that is compatible with your areas telephone network. (Stating that as it is likely different from the US network.) But, how can i known about this? There is some hardware list? i need to buy that hardware to make CallerID (external calls) and to repass to a PABX. Wich hardware i need? Can you tell me the specifications (for Brazil network or US network). Thanks very much in advace Pablo Fernandes Andrew Thompson wrote: Pablo Fernandes wrote: Can i use the Asterisk functions (caller id for example), using conventional telephony (in Brazil) ? Generally, yes. (VOIP is just a cool thing to be into these days.) Can you define call recognition for me? Do you mean CallerID(determining the phone number that is calling you)? You will need hardware that is compatible with your areas telephone network. (Stating that as it is likely different from the US network.) If digital voice circuits(in any form) are available in your area, you'll likely be happier using them than POTS lines. ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] asterisk functions without voIP
Pablo, Brazil uses normal PRI (primary rate ISDN) over E1, so the Digium TE cards will definitely work. As always with PRI, you will need to get the correct settings for framing, line coding, and so on. I would imagine that BRI (basic rate ISDN) would also be normal in Brazil, but have not tested myself. Both of these can handle callerid if your provider gives you the information. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Pablo Fernandes wrote: Hi, If digital voice circuits(in any form) are available in your area, you'll likely be happier using them than POTS lines. yes, here is available Digital voice circuits. You will need hardware that is compatible with your areas telephone network. (Stating that as it is likely different from the US network.) But, how can i known about this? There is some hardware list? i need to buy that hardware to make CallerID (external calls) and to repass to a PABX. Wich hardware i need? Can you tell me the specifications (for Brazil network or US network). Thanks very much in advace Pablo Fernandes Andrew Thompson wrote: Pablo Fernandes wrote: Can i use the Asterisk functions (caller id for example), using conventional telephony (in Brazil) ? Generally, yes. (VOIP is just a cool thing to be into these days.) Can you define call recognition for me? Do you mean CallerID(determining the phone number that is calling you)? You will need hardware that is compatible with your areas telephone network. (Stating that as it is likely different from the US network.) If digital voice circuits(in any form) are available in your area, you'll likely be happier using them than POTS lines. ___ Asterisk-Users mailing list Asterisk-Users@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@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] Sangoma A104 - D-Channel problem
Hello, I have following problem with Sangoma A104 card: CLI pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 NOTICE[16509]: chan_zap.c:7494 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 NOTICE[16509]: chan_zap.c:7494 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 NOTICE[16509]: chan_zap.c:7494 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 NOTICE[16509]: chan_zap.c:7494 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Any idea how to fix it? My configs: zaptel.conf: span=1,1,1,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 zapata.conf [channels] switchtype=euroisdn pridialplan=unknown overlapdial=no usecallerid=yes hidecallerid=yes callwaiting=no callwaitingcallerid=no threewaycalling=no transfer=no cancallforward=no callreturn=no jitterbuffers=4 echocancel=yes echocancelwhenbridged=no echotraining=yes relaxdtmf=yes rxgain=0.5 txgain=0.9 immediate=no amaflags=billing adsi=no busydetect=no callprogress=no context = zapline1 switchtype = euroisdn group=1 signalling = pri_cpe channel = 1-15 channel = 17-31 cat /proc/zaptel/1 : Span 1: WPE1/0 wanpipe1 card 0 HDB3//CRC4 1 WPE1/0/1 Clear (In use) 2 WPE1/0/2 Clear (In use) 3 WPE1/0/3 Clear (In use) 4 WPE1/0/4 Clear (In use) 5 WPE1/0/5 Clear (In use) 6 WPE1/0/6 Clear (In use) 7 WPE1/0/7 Clear (In use) 8 WPE1/0/8 Clear (In use) 9 WPE1/0/9 Clear (In use) 10 WPE1/0/10 Clear (In use) 11 WPE1/0/11 Clear (In use) 12 WPE1/0/12 Clear (In use) 13 WPE1/0/13 Clear (In use) 14 WPE1/0/14 Clear (In use) 15 WPE1/0/15 Clear (In use) 16 WPE1/0/16 HDLCFCS (In use) 17 WPE1/0/17 Clear (In use) 18 WPE1/0/18 Clear (In use) 19 WPE1/0/19 Clear (In use) 20 WPE1/0/20 Clear (In use) 21 WPE1/0/21 Clear (In use) 22 WPE1/0/22 Clear (In use) 23 WPE1/0/23 Clear (In use) 24 WPE1/0/24 Clear (In use) 25 WPE1/0/25 Clear (In use) 26 WPE1/0/26 Clear (In use) 27 WPE1/0/27 Clear (In use) 28 WPE1/0/28 Clear (In use) 29 WPE1/0/29 Clear (In use) 30 WPE1/0/30 Clear (In use) 31 WPE1/0/31 Clear (In use) Information from dmesg: Processing WAN device wanpipe1... wanpipe1: Locating: A104 card, CPU A, PciSlot=8, PciBus=0 wanpipe1: Found: A104 card, CPU A, PciSlot=8, PciBus=0, Port=0 PCI: Found IRQ 10 for device 00:08.0 wanpipe1: AFT PCI memory at 0xEE00 wanpipe1: IRQ 10 allocated to the AFT PCI card wanpipe1: Initializing for SMP wanpipe1: Starting AFT Quad Hardware Init. wanpipe1: Enabling front end link monitor wanpipe1: Global Chip Configuration: used=1 wanpipe1: Global Front End Configuraton! wanpipe1: T1/E1/J1 Global configuration! wanpipe1: AFT Data Mux Bit Map: 0x76543210 wanpipe1: Setting E1 configuration (Port 1)! wanpipe1: All channels enabled wanpipe1: Front end successful wanpipe1: AFT Security: UnChannelised wanpipe1: Configuring Device :wanpipe1 FrmVr=8 wanpipe1:Global MTU = 1500 wanpipe1:Global MRU = 1500 wanpipe1:Data Mux Map = 0x76543210 wanpipe1: Configuring Interface: w1g1 wanpipe1:w1g1: Running in TDM Voice mode. wanpipe1: AFT Fifo Level Map: 0x02082082 wanpipe1: Registering interface to Zaptel span # 1! wanpipe1:MRU :248 wanpipe1:MTU :248 wanpipe1:HDLC Eng :Off (Transparent) wanpipe1:Data Mux Ctrl :On wanpipe1:w1g1: Active channels = 0xFFFE wanpipe1:w1g1: Setting first time slot to 1 wanpipe1:w1g1: Config for Transparent mode: Idle=0 Len=248 wanpipe1:w1g1: Allocating 65 dma skb len=256 Chaining=Off ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Problem with asterisk-addons:libmysqlclient.so.14: cannot open shared object file
Run this from inside asterisk-addons: make clean; cvs update; make; make install then try again. be sure you have v1.7 of res_config_mysql The Makefile seems to check most places for mysql libraries but check it again to make sure. Also make sure your mysql lib path is in ld.so.config then rerun ldconfig. (Oh..do that before you do the above commands) -Matthew - Original Message - From: Alessio Focardi [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, February 17, 2005 8:21 AM Subject: [Asterisk-Users] Problem with asterisk-addons:libmysqlclient.so.14: cannot open shared object file Hi, I have compiled asterisk-addons successfully, but when I put res_config_mysql.so in modules directory asterisk fails to load, here is the error: 7:29 WARNING[19097]: loader.c:301 __load_resource: libmysqlclient.so.14: cannot open shared object file: No such file or directory Feb 17 15:17:29 WARNING[19097]: loader.c:509 load_modules: Loading module res_config_mysql.so failed! libmysqlclient is present on the system, should I edit something to point * to the right directory for it or something like ? Tnx for any help! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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@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
Re: [Asterisk-Users] SIP peer registration interval
On Thu, 17 Feb 2005 15:04:50 +0100 Stefan Gofferje [EMAIL PROTECTED] wrote: Hi folks, I'm registered with sipgate, a German SIP provider. Configs works fine so far. Trouble is, after a while, it seems, my registration is dropped by sipgate. How do I tell * the interval for * registering with a provider? I suppose, the re-registration interval is to long... Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Linux is like a Wigwam - No gates, no windows, Apache inside defaultexpirey=120 :Default length of incoming/outoing registration I believe that is the correct option. This site is your friend. Try searching... http://www.voip-info.org/wiki-Asterisk+config+sip.conf ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] SIP peer registration interval
Hi folks, I'm registered with sipgate, a German SIP provider. Configs works fine so far. Trouble is, after a while, it seems, my registration is dropped by sipgate. How do I tell * the interval for * registering with a provider? I suppose, the re-registration interval is to long... Regards, Stefan If you're behind NAT try enabling qualify in sip.conf. Either qualify=yes or try a specific value, for example qualify=1000. More info on the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20qualify -nathan ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Zap/g0/ to a Telstra Mobile
Howard Lowndes wrote: On Thu, 2005-02-17 at 15:51, [EMAIL PROTECTED] wrote: I've installed a TDM400. Having a go with AMP. I would like incoming calls to be put throuhg to an extension (at my desk) and a mobile (cell phone) at the same time. Whichever picks up, gets the call.. This should be possible with AMP (call groups, 200|201|0*0408xx), but it didn't work, so I have created a custom-incoming in extensions-custom.conf What is happening is, The extension rings for about 5 secs (as long as it takes the TDM400 to dial the mobile number), then just the telstra mobile rings.. From asterisk -vvvr -- Goto (custom-incoming,s,1) -- Executing Dial(SIP/202-b424, Zap/g0/0408xxSip/200|30|t) in new stack -- Called g0/0408xx -- Called 200 -- SIP/200-fece is ringing -- SIP/200-fece is ringing -- SIP/200-fece is ringing -- SIP/200-fece is ringing -- Zap/2-1 answered SIP/202-b424 This tend to indicate to me that the mobile system has picked up the call request on the zap channel and that * therefore thinks that the zap channel has picked up the call and will then bridge the zap channel to the sip 202 channel and kill off the ringing on the sip 200 channel. I don't know that there is much you can do about this as basically you are trying to get interaction on two different systems. No. Analog ports are always considered ANSWERED as soon as Asterisk finishes dialing. This is covered over and over and over again in the mailing list archives. There are a few very ugly hacks to work around the problem. ___ Asterisk-Users mailing list Asterisk-Users@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] RDIS board for gatewaying
Hi all I want to connect Asterisk with my Siemens HiPath PBX, to use it as a Gateway to the PSTN. I already have a RDIS entry in the Siemens HiPath, but the PC with Asterisk doesnt have any RDIS board, can someone tell me about good and cheap PCI RDIS boards that supports QSIG? The Eicon boards are very expensive... a BRI costs 630 Euros... thats a lot And what is the best protocol to use between them? Siemens supports QSIG and Cornet (siemens proprietary) maybe QSIG is the best choice Thanks Joao ___ Asterisk-Users mailing list Asterisk-Users@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] The 'sipfriends' table is obsolete - ????
After updating to the latest CVS Feb 17 15:20:03 WARNING[15317]: config.c:819 read_config_maps: The 'sipfriends' table is obsolete, update your config to use sipusers and sippeers, though they can point to the same table. == Binding sipusers to mysql/asterisk/sip == Binding sippeers to mysql/asterisk/sip Feb 17 15:20:03 WARNING[15317]: config.c:823 read_config_maps: The 'iaxfriends' table is obsolete, update your config to use iaxusers and iaxpeers, though they can point to the same table. == Binding iaxusers to mysql/asterisk/iax == Binding iaxpeers to mysql/asterisk/iax IS Anything changed?? Missed something? How should the iaxpeers and sippeers tables look like then? ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] CVS in production env (Attended xfer)
Mark Benson wrote: Yesterday I asked about a user manual - ie a user guide to actually using asterisk (now on how to set it up) the doc project (v2) isn't anywhere near complete and is the closest thing I could find. Does anyone know of such a doc? The reason I ask is that while a lot of this may be obvious to many people its not to someone new to asterisk and there is a lot of info to trawl through, most of which is related to configuring asterisk. Again, the question of how to do attended xfers - how? (I now know I need asterisk from CVS) but what key presses? Then there is the issue of CVS in a production environment? I'm guessing people are actually doing this, but it goes against my better judgment. Does a roadmap for asterisk exist anywhere? If there was a roadmap I wouldn't need to ask when the next stable release will be available. By stable I don't mean that I think the CVS code isn't going to be reliable, (but that is a concern) but that the code isn't going to be changing constantly. Asterisk lacks good documentation. The documentation that is available is fragmented. This is bad. Fortunately, we are seeing a slow consolidation of documentation. SineApps is now syndicating the updates information on asteriskdocs.org. I closed down MY small web site with AGI script examples and sample configs and donated the whole site to the Asterisk Docs project. I want to encourage everyone that has Asterisk related documentation web pages/sites to donate the information to the Asterisk Docs project. I also want to encourage everyone to participate in both the Asterisk Docs project and the Asterisk Wiki. ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] can't enable trunking :(
Muhammad Muzzamil Luqman wrote: I have successfully installed and configured the asterisk, the incoming and the outgoing calls are working fine, its a tremendous solution :) Now i want to enable trunking between two asterisk boxes, in the iax.conf i have put: [karachi] ... ... ... trunk=yes ... ... ... everything seems to work fine but when i load asterisk it says: -- Feb 17 10:59:14 WARNING[18726]: chan_iax2.c:7536 build_user: Unable to support trunking on user 'karachi' without zaptel timing Feb 17 10:59:14 WARNING[18726]: chan_iax2.c:7345 build_peer: Unable to support trunking on peer 'karachi' without zaptel timing -- I tried to install the ztdummy and i succeeded on one of the box but for the other i am having problems :( If you can't install Zaptel (a real driver, ztdummy, zaprtc, etc) then you can't use trunking. Remember trunking is only really useful when you have 3 or more calls at the same time between the same two Asterisk systems. Trunking with only one call actually uses MORE bandwidth. ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] HELP!!!!!!!! {Scanned}
If your X-ten phones are on the same lan as asterisk then try nat=no. David On Thu, 2005-02-17 at 07:28 +0300, Julius Kidubuka wrote: My sip.conf file; [luke] type=friend host=dynamic username=luke secret=luke ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info dtmfmode=rfc2833 mailbox=202 ; Mailbox for message waiting indicator allow=all context=sip callerid=luke 2123 nat=yes [mike] type=friend host=dynamic username=mike secret=badru ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info dtmfmode=rfc2833 mailbox=203 ; Mailbox for message waiting indicator context=sip callerid=mike 2125 nat=yes [juki] type=friend host=dynamic username=juki secret=juki dtmfmode=rfc2833 mailbox=204 ; Mailbox for message waiting indicator allow=all context=sip callerid=juki 2125 nat=yes plus my extensions.conf file; exten = 202,1,Dial(SIP/luke,20,tr) exten = 202,2,VoiceMail,u202 exten = 202,102,VoiceMail,b202 exten = 203,1,Dial(SIP/mike,20,tr) exten = 203,2,VoiceMail,u203 exten = 203,102,VoiceMail,b203 exten = 204,1,Dial(SIP/juki,20,tr) exten = 204,2,VoiceMail,u204 exten = 204,102,VoiceMail,b204 Hope this provides a little bit more info. I new to this as will. But add more info like your sip.conf file. David On Wed, 2005-02-16 at 18:04 +0300, Julius Kidubuka wrote: Hi, I have installed two X-Lite phones and theyre able to login successfully. The two phones plus the Asterisk system are all on the same LAN with private addresses assigned to each of them. When a call is initiated and is picked up on the other end, there is completely no sound at all (as in the line goes dead). The codecs set in the softphones are g711u, g711a, GSM, iLBC and SPX. From the Asterisk CLI I see the following errors; i)Unknown RTP codec 72 received ii) RFC3389 support incomplete Anyone got ideas on how I can go about this? Thanks in advance. Julius Kidubuka When you do the common things in life in an uncommon way, you will command the attention of the world -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. Plase contact [EMAIL PROTECTED] if you have questions about this email. ___ Asterisk-Users mailing list Asterisk-Users@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 -- David Shaw [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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 -- Rgds, Julius Kidubuka. My advice to you is get married: if you find a good wife you'll be happy; if not, you'll become a philosopher. ___ Asterisk-Users mailing list Asterisk-Users@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 Spam detection software, running on the system zeus.avanzada7.com, has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: My sip.conf file; [luke] type=friend host=dynamic username=luke secret=luke ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info dtmfmode=rfc2833 mailbox=202 ; Mailbox for message waiting indicator allow=all context=sip callerid=luke 2123 nat=yes [...] Content analysis details: (0.5 points, 5.0 required) pts rule name description -- -- 0.1 FORGED_RCVD_HELO Received: contains a forged HELO 0.4 PLING_PLINGSubject has lots of exclamation marks -- David Shaw [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Voicemail and busy tone
Thomas RULMONT wrote: Hi everybody, I have a problem with voicemail: I have two TDM boards for a total of 5 fxs and 3 fxo. One of the fxo is connected to the local tel provider and is redirected to a voicemail box. When I call asterisk from outside, I leave my message, but, after hanging on, voicemail continues to record the busy tone that the provider sends. How can I avoid this behaviour? You need the telco to signal when the calling party hangs up. You need this signaling to be compatable with Asterisk. I have no idea how telco lines in .be signal calling party disconnect. ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Having trouble with extensions in an include file and retrieve_extensions_from_mysql.pl
--- Andrew Thompson [EMAIL PROTECTED] wrote: --- snip --- The only thing that seems out of place to me is your #include in [main_vp_context]. It looks to me like you intend for the s, #, t, and i extensions to be in [main_vp_context]. The way you layed out this example, that's not what is happenning. I think you wanted this: Your extensions_from_mysql.conf should still look like: [vp_context] exten = 1000,1,Record(/tmp/rec:gsm); exten = 1000,2,Playback(/tmp/rec) ; exten = 1000,3,Background(goodbye) ; exten = 1000,4,Hangup(); Then, in extensions.conf: #include extensions_from_mysql.conf [main_vp_context] exten = s,1,Answer exten = #,1,Background(goodbye) ; Notify caller exten = #,2,Hangup() ; Hang up exten = t,1,Hangup() ; Hang up if timeout exten = i,1,Playback(invalid) ; Play invalid ; extension if caller ; misdials an extension include = vp_context This way, you define both contexts, and include the extensions that were defined in [vp_context] into [main_vp_context]. I don't know if this will resolve your other problem, but I believe this is the dialplan you were trying to build. Hi, Andrew. Yes, I see what you are saying. This sounds backwards, but it's actually doing what I _want_ it to do. :) From what I see in the dialplan, what asterisk does is, it loads the handlers for '#', 't' and 'i' as part of vp_context, not as part of main_vp_context. That actually happens to be as I wanted it. main_vp_context is simply a place-holder for when I am testing without the include file, and in those cases, I simply comment out my include file and voila, those handlers now handle the main_vp_context incoming cases. I know, I'm weird. :) I'm seriously concerned that my problem may be caused by some interaction between asterisk and voicepulse: at the time of writing this, even with a simple extensions.conf that has no included files at all, I cannot dial in to the asterisk box ... all calls are being rejected. Now I've spent a few minutes on (non-toll-free) hold with Voicepulse, sent them copies of my extensions.conf and iax.conf and am waiting for a response. Life really is exciting on the bleeding edge. Cheers, Maya __ Do you Yahoo!? Yahoo! Mail - now with 250MB free storage. Learn more. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] HELP!!!!!!!! {Scanned}
When I do apply nat=no, the X-ten phones don't login at all! If your X-ten phones are on the same lan as asterisk then try nat=no. David On Thu, 2005-02-17 at 07:28 +0300, Julius Kidubuka wrote: My sip.conf file; [luke] type=friend host=dynamic username=luke secret=luke ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info dtmfmode=rfc2833 mailbox=202 ; Mailbox for message waiting indicator allow=all context=sip callerid=luke 2123 nat=yes [mike] type=friend host=dynamic username=mike secret=badru ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info dtmfmode=rfc2833 mailbox=203 ; Mailbox for message waiting indicator context=sip callerid=mike 2125 nat=yes [juki] type=friend host=dynamic username=juki secret=juki dtmfmode=rfc2833 mailbox=204 ; Mailbox for message waiting indicator allow=all context=sip callerid=juki 2125 nat=yes plus my extensions.conf file; exten = 202,1,Dial(SIP/luke,20,tr) exten = 202,2,VoiceMail,u202 exten = 202,102,VoiceMail,b202 exten = 203,1,Dial(SIP/mike,20,tr) exten = 203,2,VoiceMail,u203 exten = 203,102,VoiceMail,b203 exten = 204,1,Dial(SIP/juki,20,tr) exten = 204,2,VoiceMail,u204 exten = 204,102,VoiceMail,b204 Hope this provides a little bit more info. I new to this as will. But add more info like your sip.conf file. David On Wed, 2005-02-16 at 18:04 +0300, Julius Kidubuka wrote: Hi, I have installed two X-Lite phones and they’re able to login successfully. The two phones plus the Asterisk system are all on the same LAN with private addresses assigned to each of them. When a call is initiated and is picked up on the other end, there is completely no sound at all (as in the line goes dead). The codecs set in the softphones are g711u, g711a, GSM, iLBC and SPX. From the Asterisk CLI I see the following errors; i)Unknown RTP codec 72 received ii) RFC3389 support incomplete Anyone got ideas on how I can go about this? Thanks in advance. Julius Kidubuka When you do the common things in life in an uncommon way, you will command the attention of the world -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. Plase contact [EMAIL PROTECTED] if you have questions about this email. ___ Asterisk-Users mailing list Asterisk-Users@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 -- David Shaw [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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 -- Rgds, Julius Kidubuka. My advice to you is get married: if you find a good wife you'll be happy; if not, you'll become a philosopher. ___ Asterisk-Users mailing list Asterisk-Users@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 Spam detection software, running on the system zeus.avanzada7.com, has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: My sip.conf file; [luke] type=friend host=dynamic username=luke secret=luke ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info dtmfmode=rfc2833 mailbox=202 ; Mailbox for message waiting indicator allow=all context=sip callerid=luke 2123 nat=yes [...] Content analysis details: (0.5 points, 5.0 required) pts rule name description -- -- 0.1 FORGED_RCVD_HELO Received: contains a forged HELO 0.4 PLING_PLINGSubject has lots of exclamation marks -- David Shaw [EMAIL PROTECTED] -- Rgds, Julius Kidubuka. My advice to you is get married: if you find a good wife you'll be happy; if not, you'll become a philosopher. ___ Asterisk-Users mailing list Asterisk-Users@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
Re[2]: [Asterisk-Users] Problem with asterisk-addons:libmysqlclient.so.14: cannot open shared object file
MB The Makefile seems to check most places for mysql libraries but check it MB again to make sure. Also make sure your mysql lib path is in ld.so.config MB then rerun ldconfig. (Oh..do that before you do the above commands) That was the problem, tnx ! P.S. Any skill in realtime ? I'm struggling to get it working with the BRISTUFFED version of * -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] [patch] fix libpri problem in Q931_INFORMATION handling
Peter Svensson wrote: Ok, then INFORMATION with keypad IE needs to be handled differently from IE called number. This is what it looks like with pri intense debug enabled: Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 N(S): 116 0: 0 N(R): 126 P: 0 8 bytes of data -- ACKing all packets from 125 to (but not including) 126 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: Q.931 (8) len=8 Call Ref: len= 2 (reference 7/0x7) (Originator) Message type: INFORMATION (123) [2c 01 31] Keypad Facility (len= 3) [ 1 ] Feb 16 11:42:25 VERBOSE[2975]: [ 02 01 e8 fc 08 02 00 07 7b 2c 01 31 ] I read the ets 300 403 01 spec as well as a more reacent revision of the Q.931 spec. Q.931 allows the keypad IE to signal called party number (during call setup) or to convey supplementary service information (pushed digits similar to dtmf). Q.931 allows the called party number IE to signal called party digits during call setup. The EuroISDN spec. in ets 300 403 01 is stricter - the keypad IE can not be used for overlap digits, only after the call setup. Are the digits you send encoded as Keypad IE? In that case a setting to allow keypad IE digits to always be accepted as dtmf digits may be the best solution. see trace above. It's definitely a Keypad IE and its sent not as called party digits but instead of DTMF tones. This is imho the only way to make a Siemens HiCom PBX work with Asterisk Voicemail or IVR menus. I guess there are a couple of ISDN devices out there that act the same. Deti ___ Asterisk-Users mailing list Asterisk-Users@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] Sipura to dial extension automatically
Has anyone figured out how to make a Sipura to dial an extension automatically as soon as you pick the the handset? I need to make all my users go thorugh a menu to place a call. Users should not be able to dial directly, only through the menu. Any ideas? O.A. ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] RTP Stream on Multicast
As far as I am aware there isnt a way for * to receive/send audio to a multicast group. There needs to be a way for Asterisk to tell the phone which ip multicast group to join in order to receive the page. This method varies by vendor. I know that with Cisco ip phone multicast paging they send a SCCP message to the phone to instruct it to send the appropriate IGMP join for the multicast group. Another way would be to advertise the ip multicast groups with SDP. Do you know how the Zultys phone knows which ip multicast group to join? I hope they arent statically tying themselves to 224.0.0.1 as this is a reserved ip multicast address always has a TTL of 1. If they are using this address have them read RFC 1112. The reason? I have found a method to paging on Zultys ZIP2 and ZIP4x4 handsets. Basically it involves sending a stream of RTP data to port 3771 to multicast address 224.0.0.1. ___ Asterisk-Users mailing list Asterisk-Users@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] ISDN board for gatewaying
Hi all I want to connect Asterisk with my Siemens HiPath PBX, to use it as a Gateway to the PSTN. I already have a ISDN entry in the Siemens HiPath, but the PC with Asterisk doesnt have any ISDN board, can someone tell me about good and cheap PCI ISDN boards that supports QSIG? The Eicon boards are very expensive... a BRI costs 630 Euros... thats a lot And what is the best protocol to use between them? Siemens supports QSIG and Cornet (siemens proprietary) maybe QSIG is the best choice Thanks Joao PS: sorry to send it twice, but I forgot that RDIS is portuguese, but it means ISDN ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] The 'sipfriends' table is obsolete - ????
[EMAIL PROTECTED] wrote: IS Anything changed?? Missed something? You're running head and not watching -dev? How should the iaxpeers and sippeers tables look like then? This message was posted to asterisk-dev recently: http://lists.digium.com/pipermail/asterisk-dev/2005-February/009445.html -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] fax with asterisk
I'm not using any Digium cards. I'm actually using SpanDSP and app_rxfax to process incoming faxes. After drilling into it for about 8 hours yesterday I come to realize that there is a lot more to it than the asterisk upgrade. I patched my FC3 box, which means libtiff is now 3.6.1 which according to Steven (spandsp) is broken for this application. First, i compiled the latest spandsp which didn't make a difference. I re-compiled the rxfax and txfax apps, got nowhere. Finally started trying to revert to an older libtiff, but apparently went too far back (3.5.7) and ended up getting core dumps. I am need to downgrade libtiff to 3.6.0, and everything dependant on libtiff to try again. simply downgrading libtiff on its own doesn't work well.. :-( Thats what I get for patching.. :-) This setup worked nearly flawless until I upgraded, so I'm pretty sure I can get it back again after i downgrade the right stuff in the right order. If anyone has that list, please share!! On Wed, 16 Feb 2005 19:45:51 -0700, Keith Burns [EMAIL PROTECTED] wrote: Are you both using Digium cards? Do you know if you are using G3 (standard) or SuperG3 (like a modem) fax machines? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Justin Richards Sent: Wednesday, February 16, 2005 4:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] fax with asterisk I'm getting a lot of this too :-( my fax stuff worked great under 1.0 but after upgrading to 1.0.5 i've been broken.. Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 496 (got 912, expected 1728). Fax3Decode2D: (FakeInput): Bad code word at scanline 497 (x 470). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 497 (got 470, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 498 (got 3195, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 500 (got 2471, expected 1728). Fax3Decode2D: (FakeInput): Bad code word at scanline 501 (x 1595). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 501 (got 1595, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 503 (got 2583, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 504 (got 1877, expected 1728). Fax3Decode2D: (FakeInput): Bad code word at scanline 506 (x 22). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 506 (got 22, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 507 (got 1818, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 509 (got 1729, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 510 (got 1738, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 511 (got 2228, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 514 (got 1824, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 515 (got 2466, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 516 (got 1730, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 517 (got 2534, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 518 (got 1949, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 519 (got 1830, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 521 (got 3401, expected 1728). Fax3Decode2D: (FakeInput): Bad code word at scanline 522 (x 775). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 522 (got 775, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 523 (got 2413, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 524 (got 2279, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 526 (got 2015, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 530 (got 2677, expected 1728). Fax3Decode2D: (FakeInput): Bad code word at scanline 531 (x 1220). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 531 (got 1220, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 532 (got 1729, expected 1728). Fax3Decode2D: (FakeInput): Bad code word at scanline 534 (x 0). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 534 (got 0, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 536 (got 2454, expected 1728). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 537 (got 0, expected 1728). Page 4 of
Re: [Asterisk-Users] capiECT problem
On Wed, Feb 16, 2005 at 08:58:41PM +0100, Robert Rozman wrote:
Hi,
I'm trying to get capiECT working. I'd like to transfer call to another ISDN
router connected extension and free channel from router to Asterisk.
I get incoming call on CAPI and would liek to transfer it to dialed local
extension - 400 in this case:
[outbound-capi-local]
exten = _4XX,1,NoOp(Transferring to local PBX ISDN number ${EXTEN} on msn
CAPI/${CALLERIDNUM})
exten = _4XX,2,capiHOLD
exten = _4XX,3,capiECT,${CALLERIDNUM:1}:${EXTEN}
When I dial 400, another extension rings, shows right callerid (1st argument
to capiECT), but incoming call gets constant sound and obviously loses
connection. But capi channel is freed. When I lift handset of 400 extension,
asterisk s starts to anounce number that was sent as callerid ...
Any help, hint or working example for capiECT ?
Try to Answer the call first.
--
Tho/\/\as
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Re: [Asterisk-Users] Strange MSN issue with HFC-s
On Thu, Feb 17, 2005 at 03:02:07PM +0100, Marc SCHAEFER wrote: Hi, I have two HFC-s boards I configured in NT and TE mode respectively. When I connect the two boards together, I can dial extensions and I see the correct called and caller ID numbers: -- Executing SetCallerID(Zap/2-1, 7516862) in new stack == CDR updated on Zap/2-1 -- Executing Dial(Zap/2-1, Zap/g2/0795025602|30|r) in new stack -- Called g2/0795025602 -- Extension '0795025602' in context 'isdn-local-bus' from '7516862' does not exist. Rejecting call on channel 0/1, span 1 however, when I connect the TE card to my NT2ab, it seems the caller ID number is not passed correctly (?) to the telco, since if when I get the call on my mobile I get the main number for my ISDN connection, not the specified number. With an AVM c4 it works correctly (syntax is CAPI/FROM:TO). Hm, do you have the right settings in zapata.conf? (switchtype, pridialplan...) -- Tho/\/\as ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Having trouble with extensions in an include file and retrieve_extensions_from_mysql.pl
beonice wrote: Yes, I see what you are saying. This sounds backwards, but it's actually doing what I _want_ it to do. :) From what I see in the dialplan, what asterisk does is, it loads the handlers for '#', 't' and 'i' as part of vp_context, not as part of main_vp_context. That actually happens to be as I wanted it. main_vp_context is simply a place-holder for when I am testing without the include file, and in those cases, I simply comment out my include file and voila, those handlers now handle the main_vp_context incoming cases. I know, I'm weird. :) Not necessarily... I'm thinking other words... ;) Back to your original post... As of yesterday, though, when I have this format, asterisk won't accept incoming calls. It barfs with the message: Feb 16 21:53:14 NOTICE[4330]: chan_iax2.c:5757 socket_read: Rejected connect attempt from 66.234.228.170, request '[EMAIL PROTECTED]' does not exist So, where is this voicepulse_connect_context context? -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ Asterisk-Users mailing list Asterisk-Users@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] UIP-200, registers, 4 seconds pass, then #1 disconnected
No kidding, every time. I know I have the config via tftp working. Funny story - I was getting nowhere with it and then decided to tcpdump on the tftpd box, and wow! The UIP-200 tftp client was looking for the unidenmac.txt in lower-case! Hah! That was easy to fix. Now the config is transferred to the UIP-200 at startup. It registers to the * server. The phone displays time and station name. Watch the clock tick 4 times and bingo, it says #1 DISCONNECTED. If I power up the phone, then pick up the handset before the 4 seconds and dial, the call proceeds nicely. The instant I hang up, I get #1 DISCONNECTED. If I power up the phone, then pick up the handset before the 4 seconds and hang it up, I get #1 DISCONNECTED immediately. I have 2 Sipura handsets configured and working with the * server all on the same network. Relevant info follows: asterisk 1.0.5 - Gentoo/sparc64 UIP-200 firmware version - BS4.63 Relevant sip.conf: [rob_office] type=friend host=dynamic defaultip=192.168.100.203 ;username=blah ;secret=blah dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info mailbox=1000 ; Mailbox for message waiting indicator context=sip callerid=Rob's Office 2122 ;nat=route nat=never ;qualify=yes qualify=no canreinvite=yes ;canreinvite=no ;port=5060 Any help is appreciated... please CC my email too. I *did* subscribe to the list, but strangely I have not yet received any reply email from the subscription bot. Thanks, Rob __ Do you Yahoo!? The all-new My Yahoo! - Get yours free! http://my.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Anyone having trouble with VoicePulse Connect?
Look at an EXTENSIONS RELOAD and make sure the include is being parsed -- and not throwing file not found errors. I broke my include functionality last week by reMAKEing and not paying attention to a known bug in the #INCLUDE function that existed in non-HEAD versions. /rg On Feb 16, 2005, at 10:41 PM, beonice wrote: I was doing some testing and it seems to be related to my extensions.conf. I have a #include extensions_from_mysql.conf that was working fine yesterday: ___ Asterisk-Users mailing list Asterisk-Users@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] ISDN board for gatewaying
Hi all I want to connect Asterisk with my Siemens HiPath PBX, to use it as a Gateway to the PSTN. I already have a ISDN entry in the Siemens HiPath, but the PC with Asterisk doesnt have any ISDN board, can someone tell me about good and cheap PCI ISDN boards that supports QSIG? The Eicon boards are very expensive... a BRI costs 630 Euros... thats a lot And what is the best protocol to use between them? Siemens supports QSIG and Cornet (siemens proprietary) maybe QSIG is the best choice Thanks Joao ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] [patch] fix libpri problem in Q931_INFORMATION handling
On Thu, 17 Feb 2005, Deti Fliegl wrote: Protocol Discriminator: Q.931 (8) len=8 Call Ref: len= 2 (reference 7/0x7) (Originator) Message type: INFORMATION (123) [2c 01 31] Keypad Facility (len= 3) [ 1 ] Feb 16 11:42:25 VERBOSE[2975]: [ 02 01 e8 fc 08 02 00 07 7b 2c 01 31 ] see trace above. It's definitely a Keypad IE and its sent not as called party digits but instead of DTMF tones. This is imho the only way to make a Siemens HiCom PBX work with Asterisk Voicemail or IVR menus. I guess there are a couple of ISDN devices out there that act the same. I think we agree that keypad elements may/should be passed as digits. Since this may or may not be desireable always and is a change from earlier behaviour then perhaps it should be an option? I *think* it would be ok to always pass keypad elements - it is the responsibility of an isdn device not to send both keypad and inband tones. I think it is best to only allow keypad elements always and leave called party elements disabled except during call setup. Peter ___ Asterisk-Users mailing list Asterisk-Users@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] Packet 8
I remember reading some people were talking about being able to use packet 8 without the ATA (I currently connect via an X100P card). Did this ever get anywhere? The wiki doesnt have any information on this lots of referrals but thats it. ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Sipura to dial extension automatically
On Thu, 17 Feb 2005, Oswaldo Arratia wrote: Has anyone figured out how to make a Sipura to dial an extension automatically as soon as you pick the the handset? I need to make all my users go thorugh a menu to place a call. Users should not be able to dial directly, only through the menu. You can get the manual for a Sipura from their web site. If you read it, specifically the section on the dial plan, you'll find that you can use pattern substitution with a zero delay to effect a hotline function. You could even search the pdf for that word (hotline) and that should get you to the right page quickly. Greg ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Sipura to dial extension automatically
Oswaldo Arratia wrote: Has anyone figured out how to make a Sipura to dial an extension automatically as soon as you pick the the handset? Go to google and type: sipura hotline Read the first three links. Test. Send us a note telling what worked for you. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Packet 8
dean collins wrote: I remember reading some people were talking about being able to use packet 8 without the ATA (I currently connect via an X100P card). Did this ever get anywhere? Packet8 made changes at least a year ago that prevents this. Just like Vonage did. ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Zaptel DACS and FDL
Jerry wrote: On Feb 16, 2005, at 7:19 PM, Eric Wieling wrote: Jerry wrote: On Feb 16, 2005, at 3:07 PM, Eric Wieling wrote: I have the following configuration: CLEC - T-1 - Asterisk - Adtran Channel Bank - (analog) - Nortel Don't complain that it's ugly. I've already done plenty of that. The CLEC manages their Adtran remotely and needs to be able to continue to do so. I assume they use FDL to do the management. We are using the Zaptel DACS/DACSRBS to cross connect some of the channels directly between the CLEC and the Adtran. These are channels we don't really care about (data, other voice, etc). We are not cross connecting all the channels, just some of them. I'm wondering if/how I can make sure the remote management via FDL continues to work. Does anyone have any information on this or suggestions or anything? What you dropped from your diagram is the T1 from * to the channel bank. FDL is a link level protocol. It is carried in the framing bits of the T1 not within the payload bits. When you use the digium (I presume) card within your * server as a DACS this is connecting the payload, ie timeslot, bits from one port to another. FDL will not work this way. DACSRBS does DACS the robbed bit signalling. I assume this won't help? Ah well. We'll try to find some other way to do Robbed bit refers to robbing a bit from the payload - hence the name - and the reason you can only get a 56k channel on links utilizing this form of signalling - vs 64k when not. Yes, but does FDL run over the per channel robbed bit signalling or does it run over the T-1 signaling. i.e. Does FDL run using ESF (applies to the whole T-1) or does it run on the robbed bit signaling for specific channels? ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Help Please!!!!
Thanks, I will begin my testing Erick - Original Message - From: Race Vanderdecken [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, February 16, 2005 8:18 PM Subject: RE: [Asterisk-Users] Help Please Greetings Mr. Weber, Remember the rule in mathematics that is much easier to solve for one variable. You stateed you are having a problem with the 1088 extension. If look like you are trying to make a call from the 404 extension to the 1088 extension. 1. If you have 6 ATA's running shut 5 of them off. Test each one separately. Then turn one on at a time and see the problem can be traced to one ATA 2. You are getting sent an authorization request from asterisk to the 1088 extension. WWW-Authenticate: Digest realm=asterisk, nonce=0711b1d6 Make sure you don't have any of the secret= or the md5secret= stuff set in the sip.conf, until you can get each phone to talk in the open. Then change, one, 1, uno, phone at a time. 3. If you have a SIP phone that is not an ATA then set it up and try to dial the 1088 and see if you get the same thing. 4. Do a sip show users to make sure the 1088 is registered with asterisk. 5. Do the normal, things don't work dance, by unplugging the phone and reconnecting a different phone to the ata. Change the power suplly with another ata. Change the RJ45 patch cable. Try a different port in the switch or wall. Swap one of the known working ATA and change it to the 1088 ata. 6. Go to lunch and have a beer. Find a new job and settle down with a good woman. Leave telecom and go into organic farming. Race The Tyrant Vanderdecken [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erick Weber V. Sent: Wednesday, February 16, 2005 2:34 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Help Please Importance: High I have a asterisk server with 6 Cisco ATA connected in SIP. My problem is that one of them is dropping calls an I can't figure out what is the problem; I had made a SIP DEBUG PEER 1088 that is the peer with the problem. Any help will be appreciate Thanks Erick Weber VoIP*CLI sip debug peer 1088 SIP Debugging Enabled for IP: 201.133.170.82:5060 Peer RTP is at port 192.168.1.69:0 Peer RTP is at port 192.168.1.69:0 -- Executing Dial(SIP/404-cbc9, SIP/1088|60|tr) in new stack We're at XXX.XXX.XXX.XXX port 17506 Answering/Requesting with root capability 256 12 headers, 8 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK78f35612;rport From: Weber Automundo sip:[EMAIL PROTECTED];tag=as4da46cda To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 16 Feb 2005 00:43:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 164 v=0 o=root 1679 1679 IN IP4 XXX.XXX.XXX.XXX s=session c=IN IP4 XXX.XXX.XXX.XXX t=0 0 m=audio 17506 RTP/AVP 18 a=rtpmap:18 G729/8000 a=silenceSupp:off - - - - (NAT) to 201.133.170.82:5060 -- Called 1088 -- SIP/1088-ec82 is ringing Found RTP audio format 18 Found RTP audio format 101 Peer RTP is at port 192.168.1.2:0 Found description format G729 Found description format telephone-event Capabilities: us - 0x100(G729A), peer - audio=0x100(G729A)/video=0x0(EMPTY), combined - 0x100(G729A) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) list_route: hop: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp set_destination: Parsing sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp for address/port to send to set_destination: set destination to 192.168.1.2, port 5060 Transmitting: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK642900c4;rport From: Weber Automundo sip:[EMAIL PROTECTED];tag=as4da46cda To: sip:[EMAIL PROTECTED];tag=939809556 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 201.133.170.82:5060 -- SIP/1088-ec82 answered SIP/404-cbc9 -- Attempting native bridge of SIP/404-cbc9 and SIP/1088-ec82 -- Attempting native bridge of SIP/404-cbc9 and SIP/1088-ec82 Using latest request as basis request Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: sip:[EMAIL PROTECTED];user=phone;tag=3858230914 To: sip:[EMAIL PROTECTED];user=phone;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 201.133.170.82:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: sip:[EMAIL PROTECTED];user=phone;tag=3858230914 To: sip:[EMAIL PROTECTED];user=phone;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 1
Re: [Asterisk-Users] Re: Cisco 7970 Won't boot after factory reset
Hi Keith, I have a TFTP server set up with the proper files on it, but after a factory reset, how does the phone know where to find the TFTP server..? I cannot get into it to set the TFTP server IP address. Thanks On Wed, 16 Feb 2005 20:02:22 -0500 Keith O'Brien [EMAIL PROTECTED] wrote: It is trying to download its firmware. You need to setup a TFTP Server. Also be aware that the 7970 only supports SCCP not SIP. Further, the * implementation of SCCP doesn't support the latest version of SCCP which is required for the 7970. I don't see how it would work at all with *. Hi Everyone - I just got my hands on a Cisco 7970 and was told that I should do a factory reset before trying to configure it to work with Asterisk. After the factory reset, it will not boot at all, instead sits with the line button lights flashing one at a time in sequence. I have had no luck trying to figure it out - anyone run into the same problem that can lend a hand..? ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] Packet 8
Thanks for the headsup and saving my time. It's a great service, still highly recommended I use 2 of them here Guess I'll just have to stick with running connections to the ATA's via X100P Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Thursday, February 17, 2005 11:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Packet 8 dean collins wrote: I remember reading some people were talking about being able to use packet 8 without the ATA (I currently connect via an X100P card). Did this ever get anywhere? Packet8 made changes at least a year ago that prevents this. Just like Vonage did. ___ Asterisk-Users mailing list Asterisk-Users@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@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
RE: [Asterisk-Users] Re: Cisco 7970 Won't boot after factory rese t
how does the phone know where to find the TFTP server..? Dude, option 150 in your DHCP server: http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186 a00800942f4.shtml We use the same option for our Mitel phones. HTH. ___ Asterisk-Users mailing list Asterisk-Users@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] PRI and echocancel
Hello, I have a crossover PRI(Asterisk server to PBX) and a regular telco PRI T1 line and currently have echocancel=yes and echocancelwhenbridged=yes on those spans in zapata.conf. I was discussing CPU load with another Asterisk user and he mentioned that PRIs don't need echo cancelation and that turning it off will reduced CPU load on the server. I checked many sample configs and the archives and noticed that half of the people have echocancel on for PRIs and half do not. I checked the Digium site and indeed in the FAQ they say: There should also be no echo on PRI connections. Does it really matter that much in terms of CPU usage and will it hurt at all if I turn it off for the crossover PBX connection or the telco PRI that I have? Thanks, MATT--- ___ Asterisk-Users mailing list Asterisk-Users@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] Digium TDM 400P and Dell 1750
Has anyone figured out how to power a Digium TDM 400P card in a Dell 1750 server? I opened the server and noticed that there is no access to 4 pin power to power the card. Is there some sort of adapter that I need to power the Digium card in a Dell Server? I see that the 1750 is listed on the Wiki. How have others powered the TDM400P in a Dell 1750? http://www.voip-info.org/tiki-print.php?page=Asterisk+hardware Thanks Keith ___ Asterisk-Users mailing list Asterisk-Users@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] IAXy Provisioning Using Windows
For anyone playing around with IAXy(S100i) devices, I am making the following available: Windows IAXy Provision v1.00 This is a from-the-ground-up development of a means of provisioning IAXy devices using a Windows environment. For some users, being bound to Linux for IAXy provisioning is not viable or convenient in some cases. This application provides a GUI data entry for the various IAXy parameters and communicates the new parameters to the selected IAXy. You are free to do with this application as you wish. It is provided as-is with the hope that it will make someone's day a little easier. A download package is available at: http://dacosta.dynip.com/asterisk ...Tony da Costa ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] CVS in production env (Attended xfer)
On Thu, 17 Feb 2005 09:42:54 -0600, Eric Wieling [EMAIL PROTECTED] wrote: Asterisk lacks good documentation. The documentation that is available is fragmented. This is bad. Fortunately, we are seeing a slow consolidation of documentation. SineApps is now syndicating the updates information on asteriskdocs.org. I closed down MY small web site with AGI script examples and sample configs and donated the whole site to the Asterisk Docs project. Which I am still in the process of trying to digest and figure out the best way to add it to the site. I expect more things of this nature to be donated at some point in the future (hopefully!) so I am trying to figure out a good way of adding it to Xoops (the backend we use for the Asterisk Documentation Project website). If anyone is a Xoops guru and can give me a hand, please contact me OFF-LIST. I want to encourage everyone that has Asterisk related documentation web pages/sites to donate the information to the Asterisk Docs project. I also would like encourage this. I believe centralizing Asterisk documentation is a step in the right direction. While documentation *is* fragmented and spread throughout the Internet, at the very least having links to those sites from the Asterisk Docs website is a step in the right direction. I have to thank Eric for donating his fantastic site to the docs project and I promise to get the information parsed and placed on the docs project in due time. I also want to encourage everyone to participate in both the Asterisk Docs project and the Asterisk Wiki. Documentation is created by people as they figure out how to things. Unfortunately the documentation project is a volunteer effort and the majority of documentation written for Asterisk has been done by a small group of people. If you find something wrong or missing from the Wiki or Docs project, please contribute! Thanks, Leif Madsen http://www.leifmadsen.com http://www.asteriskdocs.org ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Digium TDM 400P and Dell 1750
Keith O'Brien wrote: Has anyone figured out how to power a Digium TDM 400P card in a Dell 1750 server? I opened the server and noticed that there is no access to 4 pin power to power the card. Is there some sort of adapter that I need to power the Digium card in a Dell Server? I see that the 1750 is listed on the Wiki. How have others powered the TDM400P in a Dell 1750? Have you no way to use one of those power "splitters"? One female to two male cables that can be found almost everywhere. Unplug a pwer connector to a HD or CD and insert. John Novack ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Call forwarding
Wow. This list is high traffic Just to add to the noise, here's
some of my extensions.conf that implements what you are talking about.
In particular, the macro featureexten takes an argument that is the
same as the context the user uses for outbound dialing. The result
being that whatever number the user puts in the CFIM database is
dialed in the same way that it would be if the user dialed it directly
from their telephone. It is used like this:
exten = 5551212,1,Macro(featureexten,usercontext,SIP/whatever,${EXTEN})
Normally [usercontext] will include = [features]
Of course beware that this is only accessible from lines that have their
callerid forced to something reasonable, else you can steal lines...
[features]
; Unconditional Call Forward
exten = _*21X.,1,Answer()
exten = _*21X.,2,DBput(CFIM/${CALLERIDNUM}=${EXTEN:3})
exten = _*21X.,3,Playback(unconditional)
exten = _*21X.,4,Playback(call-forwarding)
exten = _*21X.,5,SayDigits(${EXTEN:3})
exten = _*21X.,6,Hangup()
exten = #21,1,Answer()
exten = #21,2,DBdel(CFIM/${CALLERIDNUM})
exten = #21,3,Playback(unconditional)
exten = #21,4,Playback(call-forwarding)
exten = #21,5,Playback(disabled)
exten = #21,6,Hangup()
; Call Forward on Busy or Unavailable
exten = _*61X.,1,Answer()
exten = _*61X.,2,DBput(CFBS/${CALLERIDNUM}=${EXTEN:3})
exten = _*61X.,3,Playback(call-forwarding)
exten = _*61X.,4,Playback(on-busy)
exten = _*61X.,5,SayDigits(${EXTEN:3})
exten = _*61X.,6,Hangup
exten = #61,1,Answer()
exten = #61,2,DBdel(CFBS/${CALLERIDNUM})
exten = #61,3,Playback(call-forwarding)
exten = #61,4,Playback(on-busy)
exten = #61,5,Playback(disabled)
exten = #61,6,Hangup()
; Hide Caller-ID
exten = _*67X.,1,SetCIDNum()
exten = _*67X.,2,SetCIDName(UNKNOWN)
exten = _*67X.,3,Goto(${EXTEN:3},1)
; Last Number
exten = *69,1,Answer()
exten = *69,2,DBget(temp=LCN/${CALLERIDNUM})
exten = *69,3,Playback(last-num-to-call)
exten = *69,4,SayDigits(${temp})
exten = *69,5,Hangup
exten = *69,103,Playback(im-sorry)
exten = *69,104,Playback(num-not-in-db)
exten = *69,105,Hangup
; Call-Centre
exten = *66,1,AgentLogin(${CALLERIDNUM})
; Voicemail
exten = *98,1,VoiceMailMain(${CALLERIDNUM})
[macro-featureexten]
;
; Standard extension macro (with call forwarding):
; ${ARG1} - forwarding context/dialplan
; ${ARG2} - Device(s) to ring
; ${ARG3} - voicemailbox
;
exten = s,1,DBPut(LCN/${MACRO_EXTEN}=${CALLERIDNUM})
exten = s,2,DBget(temp=CFIM/${MACRO_EXTEN}) ; Get CFIM key, if not existing,
goto 102
exten = s,3,Goto(${ARG1},${temp},1) ; unconditional forward
exten = s,4,ChanisAvail({ARG2})
exten = s,5,Dial(${AVAILORIGCHAN},30)
exten = s,6,DBget(temp=CFBS/${MACRO_EXTEN}) ; Get CFBS key, if not existing,
goto 105
exten = s,7,Goto(${ARG1},${temp},1) ; Forward on busy or unavailable
; No CFIM key
exten = s,103,Goto(s,4)
; No available channels
exten = s,105,Goto(s,107)
; No CFBS key - voicemail ?
exten = s,107,VoiceMail(u${ARG3})
On Fri, Feb 04, 2005 at 09:22:21AM -0500, Adam Robins wrote:
I've written a macro that allows users to dynamically change their call
forwarding destination. The purpose is to set up a follow me process
where a user can get calls on their cell, at home, etc., based on the
forwarding number they enter. Using the CFIM database, I have the setup
portion working great. Now, I want to actually use that information to
forward a call.
Here is my issue: The forwarded number saved in CFIM could be another
extension, a local number or an LD number, each of which would be dialed
using a different technology (internal, SIP-provider, Zap, etc.). I
want to avoid having to check the number and code all of the logic for
each method - because I already have all of this set up in the dialplan
for callers who would have dialed this forwarded number directly.
What I would like to do is take the variable containing the number
retrieved from CFIM, place it on the stack as the called number, and
have it reenter the dial plan, similar to the WAITEXTEN command.
Any ideas are appreciated!
For those interested, here is the Forwarding Setup macro:
;
; Call forwarding Macro
;
[macro-forwarding]
exten = s,1,Answer
exten = s,2,Wait(1)
exten = s,3,DigitTimeout(3)
exten = s,4,ResponseTimeout(10)
exten = s,5,Read(fwext,fw-extension,2); ask
extension (2 digits)
exten = s,6,Authenticate(/etc/asterisk/authFWD) ; only
authorized individuals
exten = s,7,Playback(fw-extension); repeat back
extension
exten = s,8,SayNumber(${fwext},f)
exten = s,9,DBget(fwnum=CFIM/${fwext}); check if
already forwarded
; ext is forwarded
exten = s,10,Playback(fw-is-forwarded-to) ; play
forwarded number from database
exten = s,11,SayDigits(${fwnum})
exten = s,12,Read(resp,fw-cancel-1-change-2,1); 1 to cancel
fwd, 2 to change #
exten = s,13,GotoIf($[${resp} = 1]?17:14) ; 1 entered,
goto delete
exten = s,14,GotoIf($[${resp} =
Re: [Asterisk-Users] can't enable trunking :(
On Thu, 17 Feb 2005 16:08:26 +0500, Muhammad Muzzamil Luqman [EMAIL PROTECTED] wrote: It was missing the kernel-source rpm. I installed the version that i found but now the first error is still there and when i modprobe ztdummy it gives the following response. --- [EMAIL PROTECTED] asterisk]# modprobe ztdummy /lib/modules/2.4.25-040218/misc/zaptel.o: kernel-module version mismatch /lib/modules/2.4.25-040218/misc/zaptel.o was compiled for kernel version 2.4.20-24.9 while this kernel is version 2.4.25-040218. /lib/modules/2.4.25-040218/misc/zaptel.o: insmod /lib/modules/2.4.25-040218/misc/zaptel.o failed /lib/modules/2.4.25-040218/misc/zaptel.o: insmod ztdummy failed [EMAIL PROTECTED] asterisk]# -- Doesn't sound like you updated your /usr/src/linux-2.4 symlink to point to the new kernel sources. Fix the symlink and verify it points to the new kernel sources, then perform a make clean ; make install in the zaptel directory and try reloading the driver. Without a timing source, trunking simply will not work. HTH, Leif Madsen http://www.leifmadsen.com ___ Asterisk-Users mailing list Asterisk-Users@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] Re: Strange MSN issue with HFC-s
Hm, do you have the right settings in zapata.conf? (switchtype, pridialplan...) So, in Switzerland, I assume switchtype = euroisdn now, for the pridialplan, am I right that the pridialplan configures the way the phone number to be dialed (called ID) is sent, and that the prilocaldialplan denotes the way the caller ID is sent to the telco ? The fact is that anything else than pridialplan = local and prilocaldialplan = local prevent any dialing out. I haven't yet understood how this impacts the way the ID is sent out. At least on CAPI, I need to set the MSN without the prefix (e.g. 7516862 and not 0327516862). This is what I tried. Also, am I right that `callerid=asreceived' tells the received caller ID to Asterisk when a call comes in ? And has nothing to do with the way caller ID is sent to the telco ? [ BRI interface with HFC-s in TE mode ] probably I will need to add the patches for the ISDN analyzer to see what happens. ___ Asterisk-Users mailing list Asterisk-Users@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
