Re: [Asterisk-Users] Pattern matching rules for least cost routing
On Wed, 2004-04-21 at 01:03, Fran Boon wrote:
On Tue, 2004-04-20 at 23:21, Mark Elkins wrote:
No matter what is dialled - I always go out on the 'Default' line.
Swapping order makes no difference. If I comment out the 'default' - it
does match the 'Cell' pattern - and works.
Pattern-matching within a context is not done based on order at all.
include = cell
include = default
[cell]
exten = _00[78][234].,1,Playback(posix-cellphone)
exten = _00[78][234].,2,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}})
[default]
exten = _0.,1,Playback(posix-defaultroute)
exten = _0.,2,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}})
Thanks (to the three replies).
Ended up leaving the cell pattern matching where it was and putting just
the default [def-out] in its own context and 'including' that to the end
of the pattern matching with...
include= def-out
Little by little - I get to shape asterisk to the way I want it to
work..
--
. . ___. .__ Posix Systems - Sth Africa
/| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496
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Re: [Asterisk-Users] Repeated Notice: (UN/REACHABLE)
Should this actually attempt more than a single ping before claiming the remote is unreachable? ie, one packet (out of the two - one request + one reply) might be lost or intermittent congestion might be involved. Perhaps a config option for setting number of consecutive ping requests are un-responsive. Also, subsequent requests might be sooner than otherwise queued. ie, successfully answered probes are re-sent every 60 seconds, while after an un-successful probe, we re-send the next probe in 10 seconds Just my 0.02c worth On Wed, 2004-04-21 at 15:03, Robert Hajime Lanning wrote: When you have qualify=yes or some number, then asterisk will poke at the peer, to measure latency. If the peer does not reply or the reply takes to long, you get the UNREACHABLE message, and you will not be able to send/receive calls to/from that channel. When the peer starts replying within the latency threshold, you will get the REACHABLE message, and you will be able to send/receive calls to/from that channel. I get it alot from FWD. Usualy means the peer is to busy (FWD) or something between you and the peer is unstable or over utilized. quote who=Barton Fisher I see repeated over and over the following messages: NOTICE[1142106560]: chan_sip.c:4988 handle_response: Peer '1001' is now REACHABLE then 5 minutes later: NOTICE[1142106560]: chan_sip.c:5958 sip_poke_noanswer: Peer '1001' is now UNREACHABLE both messages repeated over and over Any clue what I can do to fix this? Is there any where I can look up these Notices to find meaning? Thanks Bart ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 4 fedora
Good day all I'm still looking for a SIP client that will work on fedora core 1? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 4 fedora
On Wed, Apr 21, 2004 at 09:20:54AM +0200, Altus Snyman spake thusly: I'm still looking for a SIP client that will work on fedora core 1? Thanks linphone? www.linphone.org -- Tracy Reed The attachment is a digital signature. http://copilotconsulting.com More info: http://copilotconsulting.com/sig pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] sip 4 fedora
Do you have a copy for me,the page seems to be closed and it redirect me to http://swpat.ffii.org/ and I cant read that Thanks On Wed, 2004-04-21 at 09:16, Tracy R Reed wrote: On Wed, Apr 21, 2004 at 09:20:54AM +0200, Altus Snyman spake thusly: I'm still looking for a SIP client that will work on fedora core 1? Thanks linphone? www.linphone.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] uClibc patch?
Hi, I've been searching on an error I'm getting trying to compile against uClibc, related to enum support. I found reference in an earlier thread (http://lists.digium.com/pipermail/asterisk-users/2003-June/014176.html) to a patch adding an Makefile option to remove enum support. Anyone have that diff file lying around? Thanks, Jeremy Jones ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Alsa driver doesn't initialize
---BeginMessage--- ---BeginMessage--- I have just installed the Alsa drivers for my 2.4.18-14 kernel (RH8). I have configured the sound card ok with alsaconf and tested with the aplay , works fine. But when I run asterisk it says.. --- [chan_alsa.so] = (ALSA Console Channel Driver) Apr 20 18:28:34 ERROR[8192]: chan_alsa.c:339 alsa_card_init: snd_pcm_open failed: No such device or address Apr 20 18:28:34 ERROR[8192]: chan_alsa.c:339 alsa_card_init: snd_pcm_open failed: No such device or address Apr 20 18:28:34 ERROR[8192]: chan_alsa.c:474 soundcard_init: Problem opening alsa I/O devices == No sound card detected -- console channel will be unavailable == Turn off ALSA support by adding 'noload=chan_alsa.so' in /etc/asterisk/modules.conf -- earlier when using the OSS, the playback was choppy not smooth, I added some more RAM (total 256 on Intel PIII 600 processor), but the problem was still there so I turned to the Alsa drivers.Asterisk doesn't seem to work with it what might be wrong, any ideas ? ---End Message--- ---End Message---
[Asterisk-Users] Very basic questions
Hi, I am new in asterisk and i've bought a X100p and a TDM400... First of all, how can i verify my config files ? Secondly, when i'm trying to pass a call to the outside, i ve a Notice about appdial.c (l 554) telling me: unable to create channel of type Zap ...and i don't understand... Finally, when i plug my analog phones in RJ45 of my TDM400, there is no tonality ( i'm not sure that it is the right word in english , but i can't hear any tut-tut or any noise...) ... Maybe, it's obvious but i can't succeed... Thx Laurent an hopeless french student ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] About IAX channels
I have been running af Asterisk server Version 0.7.2 for a while now But I I wanted to upgrade my version to the new 0.9.0 or the CVS 1.0 Stable. But when I install one of the new asterisk servers I having lots of troubles with the IAX connection between my servers. When I start the 0.7.2 asterisk server it shows me something lige this == Parsing '/etc/asterisk/iax.conf': Found == Using TOS bits 16 == Registered channel type 'IAX1' (Inter Asterisk eXchange Drver) == Registered channel type 'IAX' (Inter Asterisk eXchange Drver) == IAX Ready and Listening on 0.0.0.0 port 5036 As you can see the Asterisk 0.7.2 registering the Inter Asterisk eXchange Drver But when I start the other 2 versions these lines don't appear. Instead when I try to make a call over the IAX lines the 0.9.0 and 1.0 versions print in the console -- Executing Dial(SIP/302-c0c3, IAX/user:[EMAIL PROTECTED]/201) in new stack Apr 21 09:35:38 WARNING[-1210897488]: channel.c:1676 ast_request: No channel type registered for 'IAX' Apr 21 09:35:38 NOTICE[-1210897488]: app_dial.c:536 dial_exec: Unable to create channel of type 'IAX' == Everyone is busy at this time -- Executing Congestion(SIP/302-c0c3, ) in new stack == Spawn extension (default, 201, 2) exited non-zero on 'SIP/302-c0c3' -- Registered to '192.168.24.100', who sees us as 192.168.24.101:4569 I don't know how to get these IAX lines to work on the 1.0 and 0.9.0 versions do someone know how to do this Thanks for any response I will get Best regards Jan Madsen ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] uClibc patch?
On Wed, 21 Apr 2004, Jeremy Jones wrote: I've been searching on an error I'm getting trying to compile against uClibc, related to enum support. I found reference in an earlier thread (http://lists.digium.com/pipermail/asterisk-users/2003-June/014176.html) to a patch adding an Makefile option to remove enum support. Anyone have that diff file lying around? Its in bugs.digium.com Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] ** WANTED: FreeBSD or OpenBSD programmer
It doesn't look very hard. FreeBSD supports recursive mutexes. It is just a matter of getting the appropriate defines. I'm going to look at this. Tom On Tue, 20 Apr 2004, Olle E. Johansson wrote: The recent addition of recursive mutexes to Asterisk is causing a lot of problems on FreeBSD servers. I need help from someone that knows mutexes on FreeBSD to make it work, otherwise the FreeBSD port of 1.0 will be useless. See bug report http://bugs.digium.com/bug_view_page.php?bug_id=0001411 for more details. Thank you for your help! /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] ** WANTED: FreeBSD or OpenBSD programmer
On Wed, 21 Apr 2004, Tom wrote: It doesn't look very hard. FreeBSD supports recursive mutexes. It is just a matter of getting the appropriate defines. I'm going to look at this. On my Gentoo system I had to add #define _GNU_SOURCE to lock.h just before it #includes pthread.h. That enabled recursive mutexes. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX clients are Unmonitored / UNREACHABLE
Thanks a lot, Which ports U used ? I tried some ... the same error. Only if I comment out the line it works The other problem ist hat all natted IAX clients go Unmonitored (CLIIAX2 show peers) if I disable the Qualify=yes tag in IAX.conf. If I activate qualify all go UNREACHABLE and cannot make or receive calls Mit freundlichen Grüßen, Lars Oertel [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] APPRADIUS ANNOUNCE
We want to inform the asterisk community members that we have released the Appradius project under the GPL License and it is already available for download. It contains app_radius.so and cdr_radius.so applications. The current version of Appradius project supports full RADIUS authorize and account features. The project web site is: http://appradius.minitelecom.org/ MiniTelecom Team -- RADIUS channel for Asterisk PBX http://appradius.minitelecom.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 4 fedora
On Wed, 21 Apr 2004, Altus Snyman wrote: Do you have a copy for me,the page seems to be closed and it redirect me to http://swpat.ffii.org/ and I cant read that You must have missed the link on that page... To enter linphone.org, click here. On the readability of swpat.ffii.org, there are half-a-dozen language flags at the top of the page also... Cheers, Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Milliwatt Quiet terminations
Sorry. 1004 hz...Im forgetting the 4 hertz...you're correct...must of been a while since the function was used. When I called today to see if it was still up, it connected, burped, and stayed up fine from then on. I'll bet there were some transistors that hadn't seen electrons in a few yearsheh.. At 02:00 4/21/2004, you wrote: 1Khz, straight up? If it is, there may be aliasing... Awww what'm I talking about... this is on low bandwidth codecs... of course it's gonna be distorted :) Telco milliwatt is 1004hz to avoid aliasing problems on a T1 James Golovich wrote: On Tue, 20 Apr 2004, tmpm wrote: If you dont mind the call, 716-861-7610 is milliwatt and 716-861-7611 is quiet term. I put them in that Ericsson AXE-10 in 1984 and they're still there. Oh one more thing nobody has pointed out yet. * comes with an app that can do ths as well. -= Info about application 'Milliwatt' =- [Synopsis]: Generate a Constant 1000Hz tone at 0dbm (mu-law) [Description]: Milliwatt(): Generate a Constant 1000Hz tone at 0dbm (mu-law) James ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 ACK // CSeq 0 = ZAP Channel hangup
Szenario:
UA(Grandstream) = PROXY(SER) = GATEWAY(*) = PSTN
After sending the SIP ACK From Gateway (*)
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK86c0bd474ea746b5
From: Me sip:[EMAIL PROTECTED];tag=0f63d269bc25545d
To: sip:[EMAIL PROTECTED];tag=as05df60b5
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 0 ACK
^^^
User-Agent: Grandstream 1.0.4.39
Warning: 399 192.168.0.1 detected firewall/NAT type is full cone
Max-Forwards: 68
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
The * Channels hangs up:
[EMAIL PROTECTED] headers, 0 lines
[EMAIL PROTECTED] ^MNEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active,
peerstate Connect Request
^@ Protocol Discriminator: Q.931 (8) len=9
^@ Call Ref: len= 2 (reference 280/0x118) (Originator)
^@ Message type: DISCONNECT (69)
^@ Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0
Location: Private network serving the local user (1)
^@ Ext: 1 Cause: Normal Clearing (16), class = Normal
Event (1) ]
^@-- Hungup 'Zap/9-1'
Is this a * Problem? Looking into the Sources shows(chan_sip.c):
else if (!strcasecmp(cmd, ACK)) {
/* Uhm, I haven't figured out the point of the ACK yet. Are we
supposed to retransmit responses until we get an ack?
Make sure this is on a valid call */
__sip_ack(p, seqno, 1);
if (strlen(get_header(req, Content-Type))) {
if (process_sdp(p, req))
return -1;
}
if (!p-lastinvite !strlen(p-randdata))
^^
p-needdestroy = 1;
}
Could it be, that * hangs up while getting zero as p-lastresult (CSeq
== 0 )?
Also it looks like a bug in the grandstreamfirmware sending CSeq zero?
would something like this solve the Problem?
if (!p-lastinvite = 0 !strlen(p-randdata))
^
?
Best Regards
Markus
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[Asterisk-Users] h323 oh323 g729 please help !
Hello list, I have many IP hardphone like Siemens 300 basic ( old ) , cisco ata.. etc I need: G711 from old phones must be convert to G729 via asterisk and send to provider I have this problem: oh323 (last version): - asterisk work with this driver ok for old phones, if I only faststart=no . But problem with codec , asterisk can speak with provider ( G729 ) only if I disable all other codec ! ( bug ? ) , but I need minimum 2 - g711 and g729. h323 -- all work ok , but only for new phones ! like cisco ATA .., with this driver old phones don't may speak with asterisk ! So, and last bug.. when I enable 2 codec in both version, I need DTMF inbound ( for g711 ) , but all time error, due g729 enabled. Can I set codec by destination? ( like SIP ) I try use 2 cnannels at the same time, but asterisk down with segmentation fault... Thanks,Serge.
Re: [Asterisk-Users] About IAX channels
IAX was removed on newer versions replace it with IAX2 just make sure to change on your extensions.conf IAX/user:[EMAIL PROTECTED]/201 to IAX2/user:[EMAIL PROTECTED]/201 Good luck Miguel On Wed, 2004-04-21 at 03:44, Jan Madsen wrote: I have been running af Asterisk server Version 0.7.2 for a while now But I I wanted to upgrade my version to the new 0.9.0 or the CVS 1.0 Stable. But when I install one of the new asterisk servers I having lots of troubles with the IAX connection between my servers. When I start the 0.7.2 asterisk server it shows me something lige this == Parsing '/etc/asterisk/iax.conf': Found == Using TOS bits 16 == Registered channel type 'IAX1' (Inter Asterisk eXchange Drver) == Registered channel type 'IAX' (Inter Asterisk eXchange Drver) == IAX Ready and Listening on 0.0.0.0 port 5036 As you can see the Asterisk 0.7.2 registering the Inter Asterisk eXchange Drver But when I start the other 2 versions these lines don't appear. Instead when I try to make a call over the IAX lines the 0.9.0 and 1.0 versions print in the console -- Executing Dial(SIP/302-c0c3, IAX/user:[EMAIL PROTECTED]/201) in new stack Apr 21 09:35:38 WARNING[-1210897488]: channel.c:1676 ast_request: No channel type registered for 'IAX' Apr 21 09:35:38 NOTICE[-1210897488]: app_dial.c:536 dial_exec: Unable to create channel of type 'IAX' == Everyone is busy at this time -- Executing Congestion(SIP/302-c0c3, ) in new stack == Spawn extension (default, 201, 2) exited non-zero on 'SIP/302-c0c3' -- Registered to '192.168.24.100', who sees us as 192.168.24.101:4569 I don't know how to get these IAX lines to work on the 1.0 and 0.9.0 versions do someone know how to do this Thanks for any response I will get Best regards Jan Madsen ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A few questions
Hi, I have a couple of questions about MeetMe and call queues. Im still pretty new to Asterisk, but already having to write a Service Center call manager for it (which I might add, our director has agreed to make open source!). MeetMe: How can I get MeetMe (does it even do this) to ask the user to speak their name first, and play that as the new member announcement. It seems like a common feature in most hardware PBX systems weve used that support Call Conferences. Has anyone found a way of doing this? Is there an alternative to MeetMe that would support this feature (thats as good if not better?). Queues: Im running the 1.0 stable from the cvs server, and Ive added the queue status announcement directives to the queues.conf yet asterisk gives me the following errors: Apr 21 11:22:58 WARNING[950286]: Unknown keyword in queue 'Sales': monitor-format at line 9 of queue.conf Apr 21 11:22:58 WARNING[950286]: Unknown keyword in queue 'Sales': announce-frequency at line 10 of queue.conf Apr 21 11:22:58 WARNING[950286]: Unknown keyword in queue 'Sales': announce-holdtime at line 11 of queue.conf Apr 21 11:22:58 WARNING[950286]: Unknown keyword in queue 'Sales': queue-youarenext at line 12 of queue.conf Apr 21 11:22:58 WARNING[950286]: Unknown keyword in queue 'Sales': queue-thereare at line 13 of queue.conf Apr 21 11:22:58 WARNING[950286]: Unknown keyword in queue 'Sales': queue-callswaiting at line 14 of queue.conf Apr 21 11:22:58 WARNING[950286]: Unknown keyword in queue 'Sales': queue-holdtime at line 15 of queue.conf Apr 21 11:22:58 WARNING[950286]: Unknown keyword in queue 'Sales': queue-minutes at line 16 of queue.conf Apr 21 11:22:58 WARNING[950286]: Unknown keyword in queue 'Sales': queue-thankyou at line 17 of queue.conf These directives I found in the asterisk wiki!
Re: [Asterisk-Users] h323 oh323 g729 please help !
update your crappy hardware :)?? atleast with sip you will be able to allow both codecs. Miguel On Mon, 2004-04-19 at 07:51, Serge wrote: Hello list, I have many IP hardphone like Siemens 300 basic ( old ) , cisco ata.. etc I need: G711 from old phones must be convert to G729 via asterisk and send to provider I have this problem: oh323 (last version): - asterisk work with this driver ok for old phones, if I only faststart=no . But problem with codec , asterisk can speak with provider ( G729 ) only if I disable all other codec ! ( bug ? ) , but I need minimum 2 - g711 and g729. h323 -- all work ok , but only for new phones ! like cisco ATA .., with this driver old phones don't may speak with asterisk ! So, and last bug.. when I enable 2 codec in both version, I need DTMF inbound ( for g711 ) , but all time error, due g729 enabled. Can I set codec by destination? ( like SIP ) I try use 2 cnannels at the same time, but asterisk down with segmentation fault... Thanks, Serge. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asttapi
Hello all, Just to update, Instruction's can be found at www.omniis.com/asttapi, including where to download it from. This is update 0.02, this now includes a little feedback from Asterisk so that when click to dial has occurred then it is indicated at the start and the end of the call. Now working on inbound calls. Any question, please send to me. Regards Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] ANI II/Payphone indication
There is no IE ( information element) in the isdn setup for this indicator. Of course with ISUP(SS7) FGD trunks it is delivered in the OLI ISUP parameter On Tue, 20 Apr 2004, James Sharp wrote: Quickie: Does anyone out there have experience with PRI delivery of ANI II information? Specifically, I want to know if it's possible from within Asterisk to know if the inbound call (which may or may not be to an 800 number) came from a payphone or not. I know with some 800 providers it's possible to block inbound calls from payphones (due to the FCC surcharge etc) but was wondering how accessible that information is once the call hits my box. I'm not sure about PRIs, but when I did it with Feature Group D trunks, the information came in as ANI II info digits prepended to the ANI. I had to modify * a bit, though, because it was stripping off the info digits and throwing them away. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 from scratch
Hi My motto is to connect two computers on the same network with Voip without using any special hardware,i have downloaded Asterisk, I was suggested to use LinPhone as a soft phone as it is very easy to install I have installed Asterisk on my computer and iam using it as a server. And whe i DAIL 1234 at CLI i get the following errors repeatedly Apr 21 17:29:13 WARNING[1167272128]: chan_oss.c:272 sound_thread: Failed to write sound Apr 21 17:29:13 WARNING[1167272128]: chan_oss.c:181 send_sound: Unable to read output space One more doubt i have is after installing a soft phone on the client,how do i configure it to connect to Asterisk. And how do i know,if Asterisk is recognizing the sound card or not Thanking you in Anticipation __ Do you Yahoo!? Yahoo! Tax Center - File online by April 15th http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 4 fedora
Hallo Altus Snyman On Wed, 21 Apr 2004 09:54:42 +0200 you wrote: Do you have a copy for me,the page seems to be closed and it redirect me to http://swpat.ffii.org/ and I cant read that Thanks Try this: http://www.linphone.org/linphone.php?lang=usrubrique=1 On Wed, 2004-04-21 at 09:16, Tracy R Reed wrote: On Wed, Apr 21, 2004 at 09:20:54AM +0200, Altus Snyman spake thusly: I'm still looking for a SIP client that will work on fedora core 1? Thanks linphone? www.linphone.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tho/\/\as ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 4 fedora
On Wed, 21 Apr 2004, Altus Snyman wrote: Do you have a copy for me,the page seems to be closed and it redirect me to http://swpat.ffii.org/ and I cant read that http://www.linphone.org/linphone.php ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] TE410P zaptel Driver Situation
Scott, We have 2 PRI spans on a TE405P, and we're sending faxes out 22 channels concurrently out one span into the other. We were trying to stress our fax application, but I fear we may have been stressing Asterisk (or the TE405P) just a little too much as well. Here's a grep for WARNING from Asterisk's 'messages' log. I have no idea if any of these are serious, but we definitely saw some failing faxes during the test. Not sure yet if they correlate with the timing of these errors. -Darren Apr 20 18:41:50 WARNING[-1329951824]: PRI: Read on 71 failed: Unknown error 500 Apr 20 18:41:50 WARNING[-1340441680]: PRI: Read on 72 failed: Unknown error 500 Apr 20 18:46:09 WARNING[-1329951824]: PRI: Read on 71 failed: Unknown error 500 Apr 20 18:46:10 WARNING[-1340441680]: PRI: Read on 72 failed: Unknown error 500 Apr 20 18:56:58 WARNING[-1329951824]: PRI: Read on 71 failed: Unknown error 500 Apr 20 19:02:22 WARNING[-1340441680]: PRI: Read on 72 failed: Unknown error 500 Apr 20 19:04:33 WARNING[-1329951824]: PRI: !! Got reject for frame 69, retransmitting frame 69 now, updating n_r! Apr 20 19:04:33 WARNING[-1329951824]: PRI: !! Got reject for frame 69, retransmitting frame 70 now, updating n_r! Apr 20 19:04:33 WARNING[-1329951824]: PRI: !! Got reject for frame 69, retransmitting frame 71 now, updating n_r! Apr 20 19:05:37 WARNING[-1329951824]: PRI: Read on 71 failed: Unknown error 500 Apr 20 19:06:42 WARNING[-1329951824]: PRI: !! Got reject for frame 58, retransmitting frame 58 now, updating n_r! Apr 20 19:06:42 WARNING[-1329951824]: PRI: !! Got reject for frame 58, retransmitting frame 59 now, updating n_r! Apr 20 19:07:48 WARNING[-1340441680]: PRI: !! Got reject for frame 30, retransmitting frame 30 now, updating n_r! Apr 20 19:07:48 WARNING[-1340441680]: PRI: !! Got reject for frame 30, retransmitting frame 31 now, updating n_r! Apr 20 19:07:48 WARNING[-1340441680]: PRI: !! Got reject for frame 32, but we have nothing -- resetting! Apr 20 19:14:16 WARNING[-1329951824]: PRI: !! Got reject for frame 111, retransmitting frame 111 now, updating n_r! Apr 20 19:14:16 WARNING[-1329951824]: PRI: !! Got reject for frame 111, retransmitting frame 112 now, updating n_r! Apr 20 19:14:16 WARNING[-1329951824]: PRI: !! Got reject for frame 111, retransmitting frame 113 now, updating n_r! Apr 20 19:14:16 WARNING[-1329951824]: PRI: !! Got reject for frame 111, retransmitting frame 114 now, updating n_r! Apr 20 19:14:16 WARNING[-1340441680]: PRI: !! Got reject for frame 28, retransmitting frame 28 now, updating n_r! Apr 20 19:14:16 WARNING[-1340441680]: PRI: !! Got reject for frame 28, retransmitting frame 29 now, updating n_r! Apr 20 19:15:20 WARNING[-1329951824]: PRI: Read on 71 failed: Unknown error 500 Apr 20 19:16:25 WARNING[-1329951824]: PRI: Read on 71 failed: Unknown error 500 Apr 20 19:16:25 WARNING[-1329951824]: PRI: !! Got reject for frame 32, retransmitting frame 32 now, updating n_r! Apr 20 19:16:25 WARNING[-1329951824]: PRI: !! Got reject for frame 32, retransmitting frame 33 now, updating n_r! Apr 20 19:16:25 WARNING[-1329951824]: PRI: !! Got reject for frame 32, retransmitting frame 34 now, updating n_r! Apr 20 19:16:26 WARNING[-1340441680]: PRI: !! Got reject for frame 43, retransmitting frame 43 now, updating n_r! Apr 20 19:16:26 WARNING[-1340441680]: PRI: !! Got reject for frame 43, retransmitting frame 44 now, updating n_r! Apr 20 19:27:14 WARNING[-1329951824]: PRI: Read on 71 failed: Unknown error 500 Apr 20 19:27:14 WARNING[-1329951824]: PRI: !! Got reject for frame 92, retransmitting frame 92 now, updating n_r! Apr 20 19:27:14 WARNING[-1329951824]: PRI: !! Got reject for frame 92, retransmitting frame 93 now, updating n_r! Apr 20 19:27:14 WARNING[-1329951824]: PRI: !! Got reject for frame 92, retransmitting frame 94 now, updating n_r! Apr 20 19:40:14 WARNING[-1340441680]: PRI: Read on 72 failed: Unknown error 500 Apr 20 19:42:24 WARNING[-1329951824]: PRI: !! Got reject for frame 93, retransmitting frame 93 now, updating n_r! Apr 20 19:42:24 WARNING[-1329951824]: PRI: !! Got reject for frame 93, retransmitting frame 94 now, updating n_r! Apr 20 19:42:24 WARNING[-1329951824]: PRI: !! Got reject for frame 93, retransmitting frame 95 now, updating n_r! Apr 20 19:55:24 WARNING[-1329951824]: PRI: Read on 71 failed: Unknown error 500 Apr 20 20:01:55 WARNING[-1329951824]: PRI: Read on 71 failed: Unknown error 500 Apr 20 20:05:10 WARNING[-1340441680]: PRI: Read on 72 failed: Unknown error 500 Apr 20 20:08:25 WARNING[-1329951824]: PRI: Read on 71 failed: Unknown error 500 Apr 20 20:09:30 WARNING[-1340441680]: PRI: !! Got reject for frame 51, retransmitting frame 51 now, updating n_r! Apr 20 20:09:30 WARNING[-1340441680]: PRI: !! Got reject for frame 51, retransmitting frame 52 now, updating n_r! Apr 20 20:16:00 WARNING[-1329951824]: PRI: Read on 71 failed: Unknown error 500 Apr 20 20:23:36 WARNING[-1329951824]: PRI: Read on 71 failed: Unknown error 500 Apr 20 20:35:32 WARNING[-1329951824]: PRI: !! Got reject for
Re: [Asterisk-Users] Stable from 4/20 launching many processes
On Wed, 21 Apr 2004, Steven Kokinos wrote: which is exactly 15 instances of asterisk. this is certainly a usual way of running for many different applications, but i was not aware asterisk was one of them. i would think there was something haywire going on, however, if i start a single instance of asterisk, then stop it gracefully, all processes do indeed stop. Is this expected behavior, or something unexpected that i should be concerned with? Could it be that you have been using diffrent flags to ps? Check this out: [EMAIL PROTECTED] chris]# ps auxw|grep aste root 1474 0.0 0.0 4196 668 ?SApr13 0:00 /bin/sh /usr/sbin/safe_ast erisk root 1475 3.4 9.3 339372 96004 ? SApr13 399:46 /usr/sbin/asterisk -fg root 500 0.0 0.0 3580 632 pts/0S15:59 0:00 grep aste [EMAIL PROTECTED] chris]# ps auxwm|grep aste root 1474 0.0 0.0 4196 668 ?SApr13 0:00 /bin/sh /usr/sbin/safe_asterisk root 1475 0.0 9.3 339372 96004 ? SApr13 0:00 /usr/sbin/asterisk -fg root 1482 0.0 9.3 339372 96004 ? SApr13 0:00 /usr/sbin/asterisk -fg root 1483 0.0 9.3 339372 96004 ? SApr13 0:02 /usr/sbin/asterisk -fg root 1484 0.0 9.3 339372 96004 ? SApr13 0:00 /usr/sbin/asterisk -fg root 1485 0.0 9.3 339372 96004 ? SApr13 0:00 /usr/sbin/asterisk -fg root 1486 0.0 9.3 339372 96004 ? SApr13 0:20 /usr/sbin/asterisk -fg root 1487 0.0 9.3 339372 96004 ? SApr13 0:10 /usr/sbin/asterisk -fg root 1488 0.0 9.3 339372 96004 ? SApr13 8:21 /usr/sbin/asterisk -fg root 1489 0.0 9.3 339372 96004 ? SApr13 0:32 /usr/sbin/asterisk -fg root 1490 0.0 9.3 339372 96004 ? SApr13 0:15 /usr/sbin/asterisk -fg root 1491 0.0 9.3 339372 96004 ? SApr13 0:05 /usr/sbin/asterisk -fg root 1492 0.0 9.3 339372 96004 ? SApr13 0:05 /usr/sbin/asterisk -fg root 1493 0.0 9.3 339372 96004 ? SApr13 2:54 /usr/sbin/asterisk -fg root 1494 0.0 9.3 339372 96004 ? SApr13 0:00 /usr/sbin/asterisk -fg root 1495 0.0 9.3 339372 96004 ? SApr13 0:49 /usr/sbin/asterisk -fg root 31437 0.1 9.3 339372 96004 ? S13:49 0:08 /usr/sbin/asterisk -fg root 31448 0.1 9.3 339372 96004 ? S13:50 0:09 /usr/sbin/asterisk -fg root 31459 0.1 9.3 339372 96004 ? S13:51 0:08 /usr/sbin/asterisk -fg root 32410 0.1 9.3 339372 96004 ? S15:15 0:02 /usr/sbin/asterisk -fg root 32547 0.1 9.3 339372 96004 ? S15:27 0:02 /usr/sbin/asterisk -fg root 32621 0.1 9.3 339372 96004 ? S15:34 0:01 /usr/sbin/asterisk -fg root 32632 0.1 9.3 339372 96004 ? S15:35 0:02 /usr/sbin/asterisk -fg root 32730 0.1 9.3 339372 96004 ? S15:44 0:01 /usr/sbin/asterisk -fg root 327 0.1 9.3 339372 96004 ? S15:50 0:00 /usr/sbin/asterisk -fg root 384 0.1 9.3 339372 96004 ? S15:55 0:00 /usr/sbin/asterisk -fg root 408 0.1 9.3 339372 96004 ? S15:57 0:00 /usr/sbin/asterisk -fg root 409 0.1 9.3 339372 96004 ? S15:57 0:00 /usr/sbin/asterisk -fg root 494 0.1 9.3 339372 96004 ? S15:59 0:00 /usr/sbin/asterisk -fg root 517 0.0 0.0 3588 656 pts/0S16:00 0:00 grep aste Notice the -m in the last command. The -m tells ps to print out all the diffrent asterisk threads. This is a old behavour that has been around asterisk running on linux for as long as i remember. /Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Repeated Notice: (UN/REACHABLE)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Goryachev Sent: Wednesday, April 21, 2004 2:29 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Repeated Notice: (UN/REACHABLE) Should this actually attempt more than a single ping before claiming the remote is unreachable? ie, one packet (out of the two - one request + one reply) might be lost or intermittent congestion might be involved. Perhaps a config option for setting number of consecutive ping requests are un-responsive. Also, subsequent requests might be sooner than otherwise queued. ie, successfully answered probes are re-sent every 60 seconds, while after an un-successful probe, we re-send the next probe in 10 seconds Just my 0.02c worth On a somewhat related note. I was experiencing some random SIP UN/REACHABLE messages during random points during the day. This would also come hand-in-hand with poor SIP call quality (jitters, stutters, etc). Yesterday I was tryint to debug a choppy SIP phone and it just so happened that I was in my lab , and noticed that we were using Ghostcast server over multicast to send images to some new PCs. On a whim, I cancelled the ghostcast session and the problem immediatly vanished. Must be a misconfig on the switch (Cisco Cat 4500 with all copper 10/100/1000 ports ) cause the switch load was minimal, but somehow the multicast traffic was screwing with the SIP transmission over the wire. Just something for other people to look for. -sb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T100P + Zap Errors
I am having some difficulty getting a T100P card to work with my PRI. When I attempt to make an outbound call via: exten = 1004,1,Dial(Zap/g1/NPANXX) I see the following on the asterisk console: -- Executing Dial(SIP/sbruton-b8ce, Zap/g1/NPANXX) in new stack Apr 21 08:18:48 NOTICE[16401]: app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time both the zaptel and wct1xxp modules are loaded and ztcfg reports no errors: # ztcfg -v Zaptel Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) 24 channels configured. and dmesg looks fine too: Zapata Telephony Interface Registered on major 196 Framer: DS21552, Revision: 3 (T1) Found a Wildcard: Digium Wildcard T100P T1/PRI Registered tone zone 0 (United States / North America) Using ESF/B8ZS coding/framing Calling startup (flags is 4099) my zaptel.conf looks like: span=1,0,0,esf,b8zs bchan=1-23 dchan=24 loadzone=us defaultzone=us and zapata.conf contains: [channels] context=default echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 group=1 switchtype=national signalling=pri_cpe group=1 channel=1-23 Any suggestions are greatly appreciated. -- Sean Bruton Senior Engineer NeoSpire, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ser and Asterisk together
Anybody out there use Ser along with *? Any advantages disadvantages? Is this even a good idea? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help choosing a UK IAX provider
Hi, Currently using voiptalk.org and the quality is getting really bad. I would like a second provider preferably in UK, anyone got any suggestions? Ta.
Re: [Asterisk-Users] Help choosing a UK IAX provider
On Wed, Apr 21, 2004 at 03:24:06PM +0100, Craig Waddington wrote: Currently using voiptalk.org and the quality is getting really bad. I would like a second provider preferably in UK, anyone got any suggestions? That's the trouble with running VoIP over contended public Internet. Find someone who can offer you connectivity with QoS and then has QoS across their network for VoIP traffic. Or find someone with infinite bandwidth. Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] T100P + Zap Errors
Sean Bruton wrote: I am having some difficulty getting a T100P card to work with my PRI. When I attempt to make an outbound call via: exten = 1004,1,Dial(Zap/g1/NPANXX) There is a real number here, ^^ right? I see the following on the asterisk console: -- Executing Dial(SIP/sbruton-b8ce, Zap/g1/NPANXX) in new stack Apr 21 08:18:48 NOTICE[16401]: app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time both the zaptel and wct1xxp modules are loaded and ztcfg reports no errors: # ztcfg -v Zaptel Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) 24 channels configured. and dmesg looks fine too: Zapata Telephony Interface Registered on major 196 Framer: DS21552, Revision: 3 (T1) Found a Wildcard: Digium Wildcard T100P T1/PRI Registered tone zone 0 (United States / North America) Using ESF/B8ZS coding/framing Calling startup (flags is 4099) my zaptel.conf looks like: span=1,0,0,esf,b8zs Set this ^ zero to 1 so the card will recover timing from the T1 stream. bchan=1-23 dchan=24 loadzone=us defaultzone=us and zapata.conf contains: [channels] context=default echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 group=1 switchtype=national signalling=pri_cpe group=1 channel=1-23 Any suggestions are greatly appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 choosing a UK IAX provider
Yes, but, I am talking about this world. Ive got 2mb up/down with qos, just need another (good) provider. If I can try a few and see which is best. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy Sent: 21 April 2004 15:32 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Help choosing a UK IAX provider On Wed, Apr 21, 2004 at 03:24:06PM +0100, Craig Waddington wrote: Currently using voiptalk.org and the quality is getting really bad. I would like a second provider preferably in UK, anyone got any suggestions? That's the trouble with running VoIP over contended public Internet. Find someone who can offer you connectivity with QoS and then has QoS across their network for VoIP traffic. Or find someone with infinite bandwidth. Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] TE410P zaptel Driver Situation
Hi Darren- In my situation, the frame rejects/retries appear not to cause problems - even thousands of them, however I found that when I got a lot of unknown error 500 messages, they would be associated with stuck channels, ie channels that appeared to be in use when they are not. The stuck channels get cleared periodically because asterisk clears idle channels automatically every once in a while. However, if you get an error 500 during a call, I think the call drops (not sure). All of this, of course, only happens during pretty heavy load. It sounds like formatting a bunch of faxes at the same time would certainly qualify as heavy load, but I'm not sure if actually trasnmitting them would. Would be nice to know if the failed faxes correlate to the TE410 errors. By the way, our theory here from experimentation and looking at the code is that error 500's are actually unhandled errors from the PRI frame driver, something like overrun or underrun. This would be consistent with being caused by heavy load. Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Nickerson Sent: Wednesday, April 21, 2004 6:10 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] TE410P zaptel Driver Situation Scott, We have 2 PRI spans on a TE405P, and we're sending faxes out 22 channels concurrently out one span into the other. We were trying to stress our fax application, but I fear we may have been stressing Asterisk (or the TE405P) just a little too much as well. Here's a grep for WARNING from Asterisk's 'messages' log. I have no idea if any of these are serious, but we definitely saw some failing faxes during the test. Not sure yet if they correlate with the timing of these errors. -Darren Apr 20 18:41:50 WARNING[-1329951824]: PRI: Read on 71 failed: Unknown error 500 Apr 20 18:41:50 WARNING[-1340441680]: PRI: Read on 72 failed: Unknown error 500 Apr 20 18:46:09 WARNING[-1329951824]: PRI: Read on 71 failed: Unknown error 500 Apr 20 18:46:10 WARNING[-1340441680]: PRI: Read on 72 failed: Unknown error 500 Apr 20 18:56:58 WARNING[-1329951824]: PRI: Read on 71 failed: Unknown error 500 Apr 20 19:02:22 WARNING[-1340441680]: PRI: Read on 72 failed: Unknown error 500 Apr 20 19:04:33 WARNING[-1329951824]: PRI: !! Got reject for frame 69, retransmitting frame 69 now, updating n_r! Apr 20 19:04:33 WARNING[-1329951824]: PRI: !! Got reject for frame 69, retransmitting frame 70 now, updating n_r! Apr 20 19:04:33 WARNING[-1329951824]: PRI: !! Got reject for frame 69, retransmitting frame 71 now, updating n_r! Apr 20 19:05:37 WARNING[-1329951824]: PRI: Read on 71 failed: Unknown error 500 Apr 20 19:06:42 WARNING[-1329951824]: PRI: !! Got reject for frame 58, retransmitting frame 58 now, updating n_r! Apr 20 19:06:42 WARNING[-1329951824]: PRI: !! Got reject for frame 58, retransmitting frame 59 now, updating n_r! Apr 20 19:07:48 WARNING[-1340441680]: PRI: !! Got reject for frame 30, retransmitting frame 30 now, updating n_r! Apr 20 19:07:48 WARNING[-1340441680]: PRI: !! Got reject for frame 30, retransmitting frame 31 now, updating n_r! Apr 20 19:07:48 WARNING[-1340441680]: PRI: !! Got reject for frame 32, but we have nothing -- resetting! Apr 20 19:14:16 WARNING[-1329951824]: PRI: !! Got reject for frame 111, retransmitting frame 111 now, updating n_r! Apr 20 19:14:16 WARNING[-1329951824]: PRI: !! Got reject for frame 111, retransmitting frame 112 now, updating n_r! Apr 20 19:14:16 WARNING[-1329951824]: PRI: !! Got reject for frame 111, retransmitting frame 113 now, updating n_r! Apr 20 19:14:16 WARNING[-1329951824]: PRI: !! Got reject for frame 111, retransmitting frame 114 now, updating n_r! Apr 20 19:14:16 WARNING[-1340441680]: PRI: !! Got reject for frame 28, retransmitting frame 28 now, updating n_r! Apr 20 19:14:16 WARNING[-1340441680]: PRI: !! Got reject for frame 28, retransmitting frame 29 now, updating n_r! Apr 20 19:15:20 WARNING[-1329951824]: PRI: Read on 71 failed: Unknown error 500 Apr 20 19:16:25 WARNING[-1329951824]: PRI: Read on 71 failed: Unknown error 500 Apr 20 19:16:25 WARNING[-1329951824]: PRI: !! Got reject for frame 32, retransmitting frame 32 now, updating n_r! Apr 20 19:16:25 WARNING[-1329951824]: PRI: !! Got reject for frame 32, retransmitting frame 33 now, updating n_r! Apr 20 19:16:25 WARNING[-1329951824]: PRI: !! Got reject for frame 32, retransmitting frame 34 now, updating n_r! Apr 20 19:16:26 WARNING[-1340441680]: PRI: !! Got reject for frame 43, retransmitting frame 43 now, updating n_r! Apr 20 19:16:26 WARNING[-1340441680]: PRI: !! Got reject for frame 43, retransmitting frame 44 now, updating n_r! Apr 20 19:27:14 WARNING[-1329951824]: PRI: Read on 71 failed: Unknown error 500 Apr 20 19:27:14 WARNING[-1329951824]: PRI: !! Got reject for frame 92, retransmitting frame 92 now, updating n_r! Apr 20 19:27:14
Re: [Asterisk-Users] Help choosing a UK IAX provider
On Wed, 2004-04-21 at 09:32, Steve Kennedy wrote: That's the trouble with running VoIP over contended public Internet. Find someone who can offer you connectivity with QoS and then has QoS across their network for VoIP traffic. LOL! I've not found any providers that offer QoS on their network other than a small regional ISP that put QoS on their network when we waved enough money at them. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] T100P + Zap Errors
On Wed, Apr 21, 2004 at 08:37:12AM -0600, Michael Welter wrote: Sean Bruton wrote: I am having some difficulty getting a T100P card to work with my PRI. When I attempt to make an outbound call via: exten = 1004,1,Dial(Zap/g1/NPANXX) There is a real number here, ^^ right? of course :) a la 2125551212 I see the following on the asterisk console: -- Executing Dial(SIP/sbruton-b8ce, Zap/g1/NPANXX) in new stack Apr 21 08:18:48 NOTICE[16401]: app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time both the zaptel and wct1xxp modules are loaded and ztcfg reports no errors: # ztcfg -v Zaptel Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) 24 channels configured. and dmesg looks fine too: Zapata Telephony Interface Registered on major 196 Framer: DS21552, Revision: 3 (T1) Found a Wildcard: Digium Wildcard T100P T1/PRI Registered tone zone 0 (United States / North America) Using ESF/B8ZS coding/framing Calling startup (flags is 4099) my zaptel.conf looks like: span=1,0,0,esf,b8zs Set this ^ zero to 1 so the card will recover timing from the T1 stream. bchan=1-23 dchan=24 loadzone=us defaultzone=us and zapata.conf contains: [channels] context=default echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 group=1 switchtype=national signalling=pri_cpe group=1 channel=1-23 Any suggestions are greatly appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] T100P + Zap Errors
On Wed, Apr 21, 2004 at 08:37:12AM -0600, Michael Welter wrote: Sean Bruton wrote: I am having some difficulty getting a T100P card to work with my PRI. When I attempt to make an outbound call via: exten = 1004,1,Dial(Zap/g1/NPANXX) There is a real number here, ^^ right? I see the following on the asterisk console: -- Executing Dial(SIP/sbruton-b8ce, Zap/g1/NPANXX) in new stack Apr 21 08:18:48 NOTICE[16401]: app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time both the zaptel and wct1xxp modules are loaded and ztcfg reports no errors: # ztcfg -v Zaptel Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) 24 channels configured. and dmesg looks fine too: Zapata Telephony Interface Registered on major 196 Framer: DS21552, Revision: 3 (T1) Found a Wildcard: Digium Wildcard T100P T1/PRI Registered tone zone 0 (United States / North America) Using ESF/B8ZS coding/framing Calling startup (flags is 4099) my zaptel.conf looks like: span=1,0,0,esf,b8zs Set this ^ zero to 1 so the card will recover timing from the T1 stream. now reads span=1,1,0,esf,b8zs reloaded modules/*, no change bchan=1-23 dchan=24 loadzone=us defaultzone=us and zapata.conf contains: [channels] context=default echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 group=1 switchtype=national signalling=pri_cpe group=1 channel=1-23 Any suggestions are greatly appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 choosing a UK IAX provider
Craig, 2mb up/down with QoS doesn't mean anything, especially when you hit the Internet. What is better is to look at the exact route of your calls and then determine whether maybe there are some other issues. For instance, we had a customer with Ciscos who was reporting choppy audio. However, this was down to a bug in asterisk (http://bugs.digium.com/bug_view_page.php?bug_id=0001374) and cvs updating fixed the problem. Tan Telappliant.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Waddington Sent: 21 April 2004 15:38 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Help choosing a UK IAX provider Yes, but, I am talking about this world. Ive got 2mb up/down with qos, just need another (good) provider. If I can try a few and see which is best. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy Sent: 21 April 2004 15:32 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Help choosing a UK IAX provider On Wed, Apr 21, 2004 at 03:24:06PM +0100, Craig Waddington wrote: Currently using voiptalk.org and the quality is getting really bad. I would like a second provider preferably in UK, anyone got any suggestions? That's the trouble with running VoIP over contended public Internet. Find someone who can offer you connectivity with QoS and then has QoS across their network for VoIP traffic. Or find someone with infinite bandwidth. Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 choosing a UK IAX provider
In the UK, with the sort of equipment that BT has in its network, you're lucky to even get adsl going through! ISPs can only provide QoS up to a certain boundary. After that it is out of their control! Tan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: 21 April 2004 15:44 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Help choosing a UK IAX provider On Wed, 2004-04-21 at 09:32, Steve Kennedy wrote: That's the trouble with running VoIP over contended public Internet. Find someone who can offer you connectivity with QoS and then has QoS across their network for VoIP traffic. LOL! I've not found any providers that offer QoS on their network other than a small regional ISP that put QoS on their network when we waved enough money at them. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 choosing a UK IAX provider
Wait till DDOS/extortion scams start hitting voip providers! Panny - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, April 21, 2004 3:43 PM Subject: Re: [Asterisk-Users] Help choosing a UK IAX provider On Wed, 2004-04-21 at 09:32, Steve Kennedy wrote: That's the trouble with running VoIP over contended public Internet. Find someone who can offer you connectivity with QoS and then has QoS across their network for VoIP traffic. LOL! I've not found any providers that offer QoS on their network other than a small regional ISP that put QoS on their network when we waved enough money at them. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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: T100P + Zap Errors
On 21/04/04 08:37 -0500, Sean Bruton wrote: I am having some difficulty getting a T100P card to work with my PRI. When I attempt to make an outbound call via: exten = 1004,1,Dial(Zap/g1/NPANXX) I see the following on the asterisk console: -- Executing Dial(SIP/sbruton-b8ce, Zap/g1/NPANXX) in new stack Apr 21 08:18:48 NOTICE[16401]: app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time What is the output of the command zap show channels in the console? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 choosing a UK IAX provider
Steve Kennedy wrote: On Wed, Apr 21, 2004 at 03:24:06PM +0100, Craig Waddington wrote: Currently using voiptalk.org and the quality is getting really bad. I would like a second provider preferably in UK, anyone got any suggestions? That's the trouble with running VoIP over contended public Internet. Find someone who can offer you connectivity with QoS and then has QoS across their network for VoIP traffic. Or find someone with infinite bandwidth. Steve QoS on the internet!! That will be the day.. I can see it now, all the P2P software will set their programs to run with maximum priority and then publish that they have the fastest system.. I know BT is thinking about creating QoS facilities on the DSL but its not available yet and will cost extra.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 choosing a UK IAX provider
On Wed, Apr 21, 2004 at 04:02:21PM +0100, [EMAIL PROTECTED] wrote: In the UK, with the sort of equipment that BT has in its network, you're lucky to even get adsl going through! ISPs can only provide QoS up to a certain boundary. After that it is out of their control! It depends on what you're trying to do. There are various ISP's in the UK that run IP/MPLS networks with metrics suitable for carrying voice traffic. They can run QoS services in and out of their networks to customers utilising leased lines/LES/or SOME forms of DSL. Of course going to another providers network (in the UK) generally goes through LINX and that's a congested exchange with no guarantees. Some networks do have private interconnects and either run QoS across the interconnect or just have enough bandwidth so contention across the interconnect never occurs. Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 choosing a UK IAX provider
Haven't used them, but on my travels have come across: http://www.magrathea-telecom.co.uk Like I said, I don't know anything about them, but seem to remember that they are an IAX provider. Cheers Matt Wait till DDOS/extortion scams start hitting voip providers! Panny - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, April 21, 2004 3:43 PM Subject: Re: [Asterisk-Users] Help choosing a UK IAX provider On Wed, 2004-04-21 at 09:32, Steve Kennedy wrote: That's the trouble with running VoIP over contended public Internet. Find someone who can offer you connectivity with QoS and then has QoS across their network for VoIP traffic. LOL! I've not found any providers that offer QoS on their network other than a small regional ISP that put QoS on their network when we waved enough money at them. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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: T100P + Zap Errors
*CLI zap show channels Chan Extension Context Language MusicOnHold 1default ... (2-22 are the same) 23default On Wed, Apr 21, 2004 at 11:05:53AM -0500, Jason Stewart wrote: On 21/04/04 08:37 -0500, Sean Bruton wrote: I am having some difficulty getting a T100P card to work with my PRI. When I attempt to make an outbound call via: exten = 1004,1,Dial(Zap/g1/NPANXX) I see the following on the asterisk console: -- Executing Dial(SIP/sbruton-b8ce, Zap/g1/NPANXX) in new stack Apr 21 08:18:48 NOTICE[16401]: app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time What is the output of the command zap show channels in the console? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 choosing a UK IAX provider
Thanks Tan. I will look into it my end. Unfortunately it isn't happening from just one location, and a variety of phones. The quality used to be perfect, the odd call would be a little jittery/choppy, but now most are like that, I am running asterisk stable with eicon diva cards. 3.0Ghz dell 2GB ram. 1 1 ms 2 ms 1 ms 10.5.0.1 217 ms14 ms14 ms 195.10.119.94 317 ms14 ms14 ms 195.10.119.158 422 ms14 ms15 ms 217.23.160.1 515 ms15 ms31 ms 217.23.162.122 617 ms15 ms14 ms 217.23.160.85 719 ms18 ms14 ms 217.23.160.186 830 ms26 ms29 ms tier1-1.BUD2.psie.net [154.14.68.113] 931 ms39 ms29 ms linx1.teleglobe.net [195.66.224.51] 1026 ms28 ms30 ms if-0-0-0.bb2.London.Teleglobe.net [195.219.96.81 ] 1159 ms87 ms 108 ms ix-3-1-0-822.bb2.London.Teleglobe.net [195.219.2 .34] 1276 ms54 ms54 ms wi2.westloc.com [82.145.32.2] 13 229 ms 239 ms 187 ms wc3-10.westloc.com [82.145.32.73] Trace complete. I don't know if asterisk is reporting this right, but all day on the console I am seeing voiptalk unreachable, then 5 secs later reachable? IAX.conf allow=ulaw allow=alaw jitterbuffer=500 maxexcessbuffer=300 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 21 April 2004 16:01 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Help choosing a UK IAX provider Craig, 2mb up/down with QoS doesn't mean anything, especially when you hit the Internet. What is better is to look at the exact route of your calls and then determine whether maybe there are some other issues. For instance, we had a customer with Ciscos who was reporting choppy audio. However, this was down to a bug in asterisk (http://bugs.digium.com/bug_view_page.php?bug_id=0001374) and cvs updating fixed the problem. Tan Telappliant.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Waddington Sent: 21 April 2004 15:38 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Help choosing a UK IAX provider Yes, but, I am talking about this world. Ive got 2mb up/down with qos, just need another (good) provider. If I can try a few and see which is best. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy Sent: 21 April 2004 15:32 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Help choosing a UK IAX provider On Wed, Apr 21, 2004 at 03:24:06PM +0100, Craig Waddington wrote: Currently using voiptalk.org and the quality is getting really bad. I would like a second provider preferably in UK, anyone got any suggestions? That's the trouble with running VoIP over contended public Internet. Find someone who can offer you connectivity with QoS and then has QoS across their network for VoIP traffic. Or find someone with infinite bandwidth. Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TxFax/SpanDSP problems
I'm getting the following when sending to a specific fax machine. Any ideas? File name is '/var/spool/asterisk/email2fax/7F2SOeYJiU.tif' Changed from phase 0 to 2 Slow carrier up Slow carrier down Slow carrier up NSF: 20 00 00 11 80 00 8a 49 10 53 54 49 52 4c 49 4e 47 20 43 4f 56 49 4e 47 54 00 67 00 80 80 80 0c 01 02 NSF without final frame tag The remote is made by 'Canon' DIS: 80 20 ee 88 c4 80 95 80 80 80 38 DIS with final frame tag In state 10 DIS: V.8 capable Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter, V.29 and V.17 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) and B4 (364mm) Receiver's minimum scan line time: 20ms at 3.85 l/mm: T7.7 = T3.85 Error correction mode T.6 coding R8x15.4lines/mm OK R16x15.4lines/mm and/or 400x400pels/25.4 mm OK Metric-based resolution preferred Minimum scan line time for higher resolutions: T15.4 = T7.7 North American Letter (215.9mm x 279.4mm) North American Legal (215.9mm x 355.6mm) Single-progression sequential coding (Rec. T.85) basic DCS: Selected data signalling rate: V.29, 9600bps 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 20ms Start sending document Start tx document - compression 2 Fine mode Changed from phase 2 to 4 Sending ident TSI: 43 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DCS: 83 00 c4 00 HDLC underflow in state 3 Changed from phase 4 to 6 Changed from phase 6 to 3 Slow carrier up XCN: fa XCN with final frame tag In state 4 Disconnecting Changed from phase 3 to 7 Changed from phase 7 to 8 -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 choosing a UK IAX provider
Hahahhaaa your right there Tan. List, don't get me wrong, voiptalk are very good, service, support, price, I am just having some issues which may be my end. I was just wanting to try some iax providers out to see what worked best for us. Hopefully will get sorted. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 21 April 2004 16:02 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Help choosing a UK IAX provider In the UK, with the sort of equipment that BT has in its network, you're lucky to even get adsl going through! ISPs can only provide QoS up to a certain boundary. After that it is out of their control! Tan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: 21 April 2004 15:44 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Help choosing a UK IAX provider On Wed, 2004-04-21 at 09:32, Steve Kennedy wrote: That's the trouble with running VoIP over contended public Internet. Find someone who can offer you connectivity with QoS and then has QoS across their network for VoIP traffic. LOL! I've not found any providers that offer QoS on their network other than a small regional ISP that put QoS on their network when we waved enough money at them. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 choosing a UK IAX provider
On Wednesday 21 April 2004 16:12, Matt wrote: Haven't used them, but on my travels have come across: http://www.magrathea-telecom.co.uk Like I said, I don't know anything about them, but seem to remember that they are an IAX provider. I haven't used Magrathea for anything 'production' yet, but the initial tests I did were good, and Linus is a very friendly helpful bloke who will be happy to 'hook you up' =) Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Webvmail
I am having trouble locating webvmail on my * server. Is this a seprate porgram or does it come with *. I am running version asterick*CLI show version Asterisk CVS-03/26/04-17:08:20 built by [EMAIL PROTECTED] on a i686 running Linux asterick*CLI Thanks Kurt __ Do you Yahoo!? Yahoo! Photos: High-quality 4x6 digital prints for 25¢ http://photos.yahoo.com/ph/print_splash ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Webvmail
make webvmail from your source directory. Then, point your browser to: http://your_ip/cgi-bin/vmail.cgi Regards, Gus -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Kurt Enviado el: Miercoles, 21 de Abril de 2004 12:36 p.m. Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] Webvmail I am having trouble locating webvmail on my * server. Is this a seprate porgram or does it come with *. I am running version asterick*CLI show version Asterisk CVS-03/26/04-17:08:20 built by [EMAIL PROTECTED] on a i686 running Linux asterick*CLI Thanks Kurt __ Do you Yahoo!? Yahoo! Photos: High-quality 4x6 digital prints for 25 http://photos.yahoo.com/ph/print_splash ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 choosing a UK IAX provider
That's the trouble with running VoIP over contended public Internet. Find someone who can offer you connectivity with QoS and then has QoS across their network for VoIP traffic. LOL! I've not found any providers that offer QoS on their network other than a small regional ISP that put QoS on their network when we waved enough money at them. FWIW, we were recently engaged to identify a VoIP problem associated with a DS3 trade show link provided by a major Internet provider. The reported problem by our client was essentially: a DSL circuit at the trade show is providing rock solid voip service, however a dedicated DS3 is providing very poor voip quality with a single workstation. The client wanted to demo a bunch of workstations running voip, etc, so a single DSL was not going to cut it. The problem turned out to be QoS had been implemented on the Internet-based DS3 from the Washington DC area to Nebraska (not requested, not expected). The client had ordered the DS3 with a certain CIR which was believed to have been mostly a billing approach (not a technical implementation). The solution actually ended up being one of implementing QoS on their XP demo workstations (simple checkmark in IP definitions), and the DS3 nicely handled several voip sessions very reliably. Surprised: Yes!!! The point of that is there are some backbone providers that have done something in terms of QoS even though its not openly discussed or advertised. Could it be some form of pre-sales technical testing or whatever? Sure. Pure guess: I'd suspect some major ISPs are playing/testing/evaluating approaches, or, may have implemented something technically that enforces a CIR on an ordinary DS3. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] One, två, tre, quatre, cinq ... International numbers in say.c
I did a quick test with the danish numbers in say.c patch (04-20-04 02:11) and found this.. *1 -- Executing SayNumber(SIP/1000-497f, 1) in new stack -- Playing 'digits/1' (language 'da') *2 -- Executing SayNumber(SIP/1000-497f, 100) in new stack -- Playing 'digits/1' (language 'da') -- Playing 'digits/hundred' (language 'da') *3 -- Executing SayNumber(SIP/1000-497f, 101) in new stack -- Playing 'digits/1' (language 'da') -- Playing 'digits/hundred' (language 'da') -- Playing 'digits/1' (language 'da') *4 -- Executing SayNumber(SIP/1000-497f, 1000) in new stack -- Playing 'digits/1' (language 'da') -- Playing 'digits/thousand' (language 'da') -- Playing 'digits/and' (language 'da') -- Executing SayNumber(SIP/1000-497f, 1001) in new stack -- Playing 'digits/1' (language 'da') -- Playing 'digits/thousand' (language 'da') -- Playing 'digits/and' (language 'da') -- Playing 'digits/1' (language 'da') *5 -- Executing SayNumber(SIP/1000-497f, 100) in new stack -- Playing 'digits/1' (language 'da') -- Playing 'digits/million' (language 'da') -- Playing 'digits/and' (language 'da') -- Executing SayNumber(SIP/1000-497f, 101) in new stack -- Playing 'digits/1' (language 'da') -- Playing 'digits/million' (language 'da') -- Playing 'digits/and' (language 'da') -- Playing 'digits/1' (language 'da') *1)pronounced en, not an issue in itself but see next point. *2)pronounced et + hundrede, different digit 1 et. *3)pronounced et + hundrede + og + en, there is an og missing. *4)pronounced et + tusinde, no need for the og *5)pronounced en + million, no need for the og A few pointers to how it is done... (Last time I translated VoiceMail software was 7 years ago and the biggest problem was making our vendor understand that we needed two different 1's. ) (1,2,3...99) en, to, tre...ni|og|halvfems (100,101...199) et|hundrede, et|hundrede|og|en, ..., et|hundrede|og|ni|og|halvfems (1000,1001...1099) et|tusinde, et|tusinde|og|en, ..., et|tusinde|og|ni|og|halvfems (1100,1101...1999) et|tusinde|et|hundrede, et|tusinde|et|hundrede|og|en, ..., et|tusinde|ni|hundrede|ni|og|halvfems (100,101...199) en|million, en|million|og|en, ..., en|million|og|ni|og|halvfems (1000100...199) en|million|et|hundrede, ..., en|million|ni|hundrede|ni|og|halvfems|tusinde|ni|hundrede|ni|og|ni|og|halvfe ms (200...X) to|millioner...X -- Soren - Original Message - From: Olle E. Johansson [EMAIL PROTECTED] To: Users Asterisk [EMAIL PROTECTED] Sent: Monday, April 19, 2004 9:53 PM Subject: [Asterisk-Users] One, två, tre, quatre, cinq ... International numbers in say.c http://bugs.digium.com/bug_view_page.php?bug_id=0001429 [SNIP] * If we all work on this together quickly, we may have a working say.c in the CVS soon. But to even ask a committer for support, I need test results up there on the bug tracker. * Thank you for your support! /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] TxFax/SpanDSP problems
Hi Eric ! I have the same problem with Canon fax mashine as you have. I have wrote an email to Steve (developer of spandsp) some weeks ago and got no answer how to fix the problem :( Looks like connection was dropped by fax mashine without any reason - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, April 21, 2004 6:19 PM Subject: [Asterisk-Users] TxFax/SpanDSP problems I'm getting the following when sending to a specific fax machine. Any ideas? File name is '/var/spool/asterisk/email2fax/7F2SOeYJiU.tif' Changed from phase 0 to 2 Slow carrier up Slow carrier down Slow carrier up NSF: 20 00 00 11 80 00 8a 49 10 53 54 49 52 4c 49 4e 47 20 43 4f 56 49 4e 47 54 00 67 00 80 80 80 0c 01 02 NSF without final frame tag The remote is made by 'Canon' DIS: 80 20 ee 88 c4 80 95 80 80 80 38 DIS with final frame tag In state 10 DIS: V.8 capable Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter, V.29 and V.17 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) and B4 (364mm) Receiver's minimum scan line time: 20ms at 3.85 l/mm: T7.7 = T3.85 Error correction mode T.6 coding R8x15.4lines/mm OK R16x15.4lines/mm and/or 400x400pels/25.4 mm OK Metric-based resolution preferred Minimum scan line time for higher resolutions: T15.4 = T7.7 North American Letter (215.9mm x 279.4mm) North American Legal (215.9mm x 355.6mm) Single-progression sequential coding (Rec. T.85) basic DCS: Selected data signalling rate: V.29, 9600bps 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 20ms Start sending document Start tx document - compression 2 Fine mode Changed from phase 2 to 4 Sending ident TSI: 43 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DCS: 83 00 c4 00 HDLC underflow in state 3 Changed from phase 4 to 6 Changed from phase 6 to 3 Slow carrier up XCN: fa XCN with final frame tag In state 4 Disconnecting Changed from phase 3 to 7 Changed from phase 7 to 8 -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] TE410P zaptel Driver Situation
Dear Scott i have notice the same type of warning plus another with my TE410P with not very high load 30 IN/OUT line, i use only two span in this moment. What is only warning? or not? Thanks in advance Dimitri Apr 21 10:27:35 WARNING[966674]: Unable to forward voice Apr 21 10:28:41 WARNING[999442]: Unable to forward frame Apr 21 10:30:11 WARNING[1048594]: Unable to forward frame Apr 21 10:32:02 WARNING[1130515]: Unable to forward frame Apr 21 10:34:56 WARNING[1212434]: Unable to forward frame Apr 21 10:35:45 WARNING[1245202]: Unable to forward frame Apr 21 10:36:34 WARNING[1277970]: Unable to forward frame Apr 21 10:37:02 WARNING[1294354]: Unable to forward frame Apr 21 10:40:03 WARNING[1359891]: Unable to forward frame Apr 21 10:45:51 WARNING[1441811]: Unable to forward frame Apr 21 10:47:15 WARNING[1474579]: Unable to forward frame Apr 21 10:47:21 WARNING[1490963]: Unable to forward frame Apr 21 10:49:57 WARNING[1540115]: Unable to forward frame Apr 21 10:50:26 WARNING[1572883]: Unable to forward frame Apr 21 10:51:25 WARNING[1622035]: Unable to forward voice Apr 21 11:11:42 WARNING[1851410]: Unable to forward frame Apr 21 11:19:20 WARNING[2097170]: Unable to forward frame Apr 21 11:29:17 WARNING[2228243]: Unable to forward voice Apr 21 11:31:39 WARNING[2375700]: Unable to forward frame Apr 21 11:45:30 WARNING[2588692]: Unable to forward frame Apr 21 11:53:00 WARNING[2686995]: Unable to forward frame Apr 21 11:56:14 WARNING[2752531]: Unable to forward frame Apr 21 11:58:31 WARNING[2850837]: Unable to forward voice Apr 21 11:59:13 WARNING[2867219]: Unable to forward voice Apr 21 12:01:41 WARNING[3047443]: Unable to forward frame Apr 21 12:02:21 WARNING[3112979]: Unable to forward voice Apr 21 12:03:27 WARNING[3145747]: Unable to forward voice Apr 21 12:07:49 WARNING[3325970]: Unable to forward frame Apr 21 12:13:49 WARNING[3506198]: Unable to forward frame Apr 21 12:18:32 WARNING[3686422]: Unable to forward frame Apr 21 12:27:03 WARNING[3768339]: Unable to forward frame Apr 21 12:29:35 WARNING[3899413]: Unable to forward frame Apr 21 12:29:40 WARNING[3932181]: Unable to forward voice Apr 21 12:43:15 WARNING[4177942]: Unable to forward voice Apr 21 12:45:21 WARNING[4194322]: Unable to forward frame Apr 21 12:46:34 WARNING[4227090]: Unable to forward frame Apr 21 12:47:24 WARNING[4243474]: Unable to forward frame Apr 21 13:05:48 WARNING[4653074]: Unable to forward frame Apr 21 13:06:20 WARNING[4685843]: Unable to forward frame Apr 21 13:21:57 WARNING[5062678]: Unable to forward frame Apr 21 13:36:04 WARNING[5390360]: Unable to forward frame Apr 21 13:38:55 WARNING[5521428]: Unable to forward frame Apr 21 13:42:15 WARNING[5783576]: Unable to forward frame - Apr 20 17:42:45 WARNING[163851]: PRI: !! Got reject for frame 112, retransmitting frame 11 2 now, updating n_r! Apr 20 17:42:45 WARNING[163851]: PRI: !! Got reject for frame 112, retransmitting frame 11 3 now, updating n_r! Apr 20 18:38:48 WARNING[40583199]: Unable to forward frame Apr 20 18:39:02 WARNING[163851]: PRI: !! Got reject for frame 44, retransmitting frame 44 now, updating n_r! Apr 20 18:39:02 WARNING[163851]: PRI: !! Got reject for frame 44, retransmitting frame 45 now, updating n_r! Apr 20 18:39:40 WARNING[40812585]: Unable to forward voice Apr 20 18:40:08 WARNING[40878111]: Unable to forward frame Apr 20 18:41:26 WARNING[41025576]: Unable to forward voice Apr 20 18:41:30 WARNING[41091112]: Unable to forward frame Apr 20 18:41:55 WARNING[147466]: PRI: !! Got reject for frame 76, retransmitting frame 76 now, updating n_r! Apr 20 18:41:55 WARNING[147466]: PRI: !! Got reject for frame 76, retransmitting frame 77 now, updating n_r! -- On Wednesday 21 April 2004 04:40 pm, Scott Stingel wrote: Hi Darren- In my situation, the frame rejects/retries appear not to cause problems - even thousands of them, however I found that when I got a lot of unknown error 500 messages, they would be associated with stuck channels, ie channels that appeared to be in use when they are not. The stuck channels get cleared periodically because asterisk clears idle channels automatically every once in a while. However, if you get an error 500 during a call, I think the call drops (not sure). All of this, of course, only happens during pretty heavy load. It sounds like formatting a bunch of faxes at the same time would certainly qualify as heavy load, but I'm not sure if actually trasnmitting them would. Would be nice to know if the failed faxes correlate to the TE410 errors. By the way, our theory here from experimentation and looking at the code is that error 500's are actually unhandled errors from the PRI frame driver, something like overrun or underrun. This would be consistent with being caused by heavy load. Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto
[Asterisk-Users] re: webvmail
Next question: After doing your rerecommendation was able to get to the main web page. I trtriedogging in using one of the vmvmailccounts (I am to assume that the login and password is what I have set up in vovoicemailoconfor mail boxes) and I got login incorrect. Do i need to change permission on any of the files etc... Kurt __ Do you Yahoo!? Yahoo! Photos: High-quality 4x6 digital prints for 25¢ http://photos.yahoo.com/ph/print_splash ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Questions about alarm reporting in Asterisk
I am currently helping a friend build an Asterisk PBX that spans several cities using anything from T1s to DSL connections to link remote SIP phones, IAX gateways, etc. to a central Asterisk PBX server that serves up voicemail, features, etc. The biggest problem that I have had with this system appears to be the leading problem that my day job company finds with their VOIP deployments: Most common problems are on the infrastructure network but are reported as phone system problems because that is the piece that the customer directly interacts with. I'm interested in hearing success stories in tying things like Asterisk YELLOW and RED alarms and network problems into a central alarm reporting solution. The most common problems that I have found are: 1. Someone unplugs a X100P from the Dmarc and nobody knows until people complain that calls are not coming in. 2. A network span goes down and nobody knows until they can't send or receive calls on that span. Here are some ideas that I have thought about so far: 1. Installing a basic SNMP agent on each Linux box and using a central SNMP manager to monitor each node. This would give notice when a remote node became isolated from the monitoring network. 2. Rolling in Asterisk alarm logs into a syslog server or even as SNMP traps. Any good ideas would be appreciated! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fw: Interconnecting to an Altigen PBX?
Has anyone got Asterisk talking successfully to an Altigen PBX using h323? I can successfully make calls from Asterisk to Altigen, but calls from Altigen to Asterisk get a fast busy. There seems to be a lack of h323 example files (or maybe I'm looking in the wrong places) as well as a severe lack of h323 documentation from Altigen. Any pointers would be greatly appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Questions about alarm reporting in Asterisk
On Wed, 21 Apr 2004, Clif Jones wrote: I am currently helping a friend build an Asterisk PBX that spans several cities using anything from T1s to DSL connections to link remote SIP phones, IAX gateways, etc. to a central Asterisk PBX server that serves up voicemail, features, etc. The biggest problem that I have had with this system appears to be the leading problem that my day job company finds with their VOIP deployments: Most common problems are on the infrastructure network but are reported as phone system problems because that is the piece that the customer directly interacts with. I'm interested in hearing success stories in tying things like Asterisk YELLOW and RED alarms and network problems into a central alarm reporting solution. The most common problems that I have found are: 1. Someone unplugs a X100P from the Dmarc and nobody knows until people complain that calls are not coming in. 2. A network span goes down and nobody knows until they can't send or receive calls on that span. Here are some ideas that I have thought about so far: 1. Installing a basic SNMP agent on each Linux box and using a central SNMP manager to monitor each node. This would give notice when a remote node became isolated from the monitoring network. 2. Rolling in Asterisk alarm logs into a syslog server or even as SNMP traps. The manager interface sends events when a channel/span goes into alarm. A simple app collecting this data should be able to handle this for you James ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Repeated Notice: (UN/REACHABLE)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Goryachev Sent: Wednesday, April 21, 2004 2:29 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Repeated Notice: (UN/REACHABLE) Should this actually attempt more than a single ping before claiming the remote is unreachable? ie, one packet (out of the two - one request + one reply) might be lost or intermittent congestion might be involved. Perhaps a config option for setting number of consecutive ping requests are un-responsive. Also, subsequent requests might be sooner than otherwise queued. ie, successfully answered probes are re-sent every 60 seconds, while after an un-successful probe, we re-send the next probe in 10 seconds Just my 0.02c worth That would be more robust/quicker to recover. You do have to remember that the RTP session (when you make a call) does not try to recover. So, usually when the SIP poke fails, the RTP would be of bad quality. quote who=Bisker, Scott (7805) On a somewhat related note. I was experiencing some random SIP UN/REACHABLE messages during random points during the day. This would also come hand-in-hand with poor SIP call quality (jitters, stutters, etc). Yesterday I was tryint to debug a choppy SIP phone and it just so happened that I was in my lab , and noticed that we were using Ghostcast server over multicast to send images to some new PCs. On a whim, I cancelled the ghostcast session and the problem immediatly vanished. Must be a misconfig on the switch (Cisco Cat 4500 with all copper 10/100/1000 ports ) cause the switch load was minimal, but somehow the multicast traffic was screwing with the SIP transmission over the wire. Just something for other people to look for. You would need to configure the switch for IGMP snooping and the ghost clients need to send multicast group membership requests, that the switch will be able to snoop. Otherwise multicast traffic is broadcast to every active port. So, it is not the switch that is being overrun, it is your SIP endpoints, that are flooded with the ghost traffic. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] One, två, tre, quatre, cinq ... International numbers in say.c
Sören! Tusen Tack :-) I'll add your input and will see what I can do to fix this. Does the other danish people agree? For the rest of you - please add your input to the bugtracker. For those of you who have earlier contributed with patches, answer my e-mails! If I don't get disclaimers from the french, portuguese, spanish or danish contributors, we might have to rip those languages out of the patch again. If there are other people out there that have written the code to support one of these languages, please contribute your code with a disclaimer so I can replace the current non-disclaimed code with your code. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Questions about alarm reporting in Asterisk
Any good ideas would be appreciated! We use a package called Nagios to monitor our servers, which works quite well. It has the ability to track service and host dependencies so you don't get flooded with a bunch of service down alerts when the real cause is a bad switch (or similar). It would seem logical for someone (hah!) to write a res_snmp.c for asterisk that would expose a lot of asterisk's internal data. This would seem a logical step toward writing fully functional monitoring applications as well. The module would allow clients to add themselves to the list and receive traps, as well as check for the current status of various variables. brainstorming Okay, this may be over the top, but here goes. Write an asterisk application that sends (and receives) status information to another box over the PSTN. My idea is not only to use this as a way to verify that * is running, but as a way to RELIABLY tell that a remote * box is actively accepting incoming calls. It wouldn't have to be anything complicated, just a heartbeat and some basic details to let the caller know that yes, I'm alive and accepting calls over this line. Simplified protocol: 1) Monitoring box calls up and says (in DTMF): #my CallerID#extension I am trying to reach#I'm a machine, so reply in DTMF instead of voice#the secret code is# 2) The remote box says #your CallerID#Your DNIS#yes I will accept a call to that number# 3) Monitoring box acknowledges and disconnects 4) Remote box disconnects 5) Monitoring box decides whether it likes the answers it received and performs actions accordingly. /brainstorming --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Clif Jones Sent: Wednesday, April 21, 2004 10:16 AM To: asterisk users Subject: [Asterisk-Users] Questions about alarm reporting in Asterisk I am currently helping a friend build an Asterisk PBX that spans several cities using anything from T1s to DSL connections to link remote SIP phones, IAX gateways, etc. to a central Asterisk PBX server that serves up voicemail, features, etc. The biggest problem that I have had with this system appears to be the leading problem that my day job company finds with their VOIP deployments: Most common problems are on the infrastructure network but are reported as phone system problems because that is the piece that the customer directly interacts with. I'm interested in hearing success stories in tying things like Asterisk YELLOW and RED alarms and network problems into a central alarm reporting solution. The most common problems that I have found are: 1. Someone unplugs a X100P from the Dmarc and nobody knows until people complain that calls are not coming in. 2. A network span goes down and nobody knows until they can't send or receive calls on that span. Here are some ideas that I have thought about so far: 1. Installing a basic SNMP agent on each Linux box and using a central SNMP manager to monitor each node. This would give notice when a remote node became isolated from the monitoring network. 2. Rolling in Asterisk alarm logs into a syslog server or even as SNMP traps. Any good ideas would be appreciated! ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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: webvmail
No, you don't need to change permissions. Check in your voicemail.conf the user password for accounts. I don't know how vmail.cgi works with multiple contexts, or if you have mysql/pgsql support with app_voicemail. See http://www.voip-info.org/wiki-Asterisk+gui+vmail.cgi for more details. Regards, Gus -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Kurt Enviado el: Miercoles, 21 de Abril de 2004 02:05 p.m. Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] re: webvmail Next question: After doing your rerecommendation was able to get to the main web page. I trtriedogging in using one of the vmvmailccounts (I am to assume that the login and password is what I have set up in vovoicemailoconfor mail boxes) and I got login incorrect. Do i need to change permission on any of the files etc... Kurt __ Do you Yahoo!? Yahoo! Photos: High-quality 4x6 digital prints for 25 http://photos.yahoo.com/ph/print_splash ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] TE410P zaptel Driver Situation
Dimitri- I'm not sure about the message Unable to forward voice - I don't get that one on my systems. The frame reject messages are the same ones I was talking about - these are load related, I'm pretty sure. Regards, Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of reseaux Sent: Wednesday, April 21, 2004 9:48 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] TE410P zaptel Driver Situation Dear Scott i have notice the same type of warning plus another with my TE410P with not very high load 30 IN/OUT line, i use only two span in this moment. What is only warning? or not? Thanks in advance Dimitri Apr 21 10:27:35 WARNING[966674]: Unable to forward voice Apr 21 10:28:41 WARNING[999442]: Unable to forward frame Apr 21 10:30:11 WARNING[1048594]: Unable to forward frame Apr 21 10:32:02 WARNING[1130515]: Unable to forward frame Apr 21 10:34:56 WARNING[1212434]: Unable to forward frame Apr 21 10:35:45 WARNING[1245202]: Unable to forward frame Apr 21 10:36:34 WARNING[1277970]: Unable to forward frame Apr 21 10:37:02 WARNING[1294354]: Unable to forward frame Apr 21 10:40:03 WARNING[1359891]: Unable to forward frame Apr 21 10:45:51 WARNING[1441811]: Unable to forward frame Apr 21 10:47:15 WARNING[1474579]: Unable to forward frame Apr 21 10:47:21 WARNING[1490963]: Unable to forward frame Apr 21 10:49:57 WARNING[1540115]: Unable to forward frame Apr 21 10:50:26 WARNING[1572883]: Unable to forward frame Apr 21 10:51:25 WARNING[1622035]: Unable to forward voice Apr 21 11:11:42 WARNING[1851410]: Unable to forward frame Apr 21 11:19:20 WARNING[2097170]: Unable to forward frame Apr 21 11:29:17 WARNING[2228243]: Unable to forward voice Apr 21 11:31:39 WARNING[2375700]: Unable to forward frame Apr 21 11:45:30 WARNING[2588692]: Unable to forward frame Apr 21 11:53:00 WARNING[2686995]: Unable to forward frame Apr 21 11:56:14 WARNING[2752531]: Unable to forward frame Apr 21 11:58:31 WARNING[2850837]: Unable to forward voice Apr 21 11:59:13 WARNING[2867219]: Unable to forward voice Apr 21 12:01:41 WARNING[3047443]: Unable to forward frame Apr 21 12:02:21 WARNING[3112979]: Unable to forward voice Apr 21 12:03:27 WARNING[3145747]: Unable to forward voice Apr 21 12:07:49 WARNING[3325970]: Unable to forward frame Apr 21 12:13:49 WARNING[3506198]: Unable to forward frame Apr 21 12:18:32 WARNING[3686422]: Unable to forward frame Apr 21 12:27:03 WARNING[3768339]: Unable to forward frame Apr 21 12:29:35 WARNING[3899413]: Unable to forward frame Apr 21 12:29:40 WARNING[3932181]: Unable to forward voice Apr 21 12:43:15 WARNING[4177942]: Unable to forward voice Apr 21 12:45:21 WARNING[4194322]: Unable to forward frame Apr 21 12:46:34 WARNING[4227090]: Unable to forward frame Apr 21 12:47:24 WARNING[4243474]: Unable to forward frame Apr 21 13:05:48 WARNING[4653074]: Unable to forward frame Apr 21 13:06:20 WARNING[4685843]: Unable to forward frame Apr 21 13:21:57 WARNING[5062678]: Unable to forward frame Apr 21 13:36:04 WARNING[5390360]: Unable to forward frame Apr 21 13:38:55 WARNING[5521428]: Unable to forward frame Apr 21 13:42:15 WARNING[5783576]: Unable to forward frame - Apr 20 17:42:45 WARNING[163851]: PRI: !! Got reject for frame 112, retransmitting frame 11 2 now, updating n_r! Apr 20 17:42:45 WARNING[163851]: PRI: !! Got reject for frame 112, retransmitting frame 11 3 now, updating n_r! Apr 20 18:38:48 WARNING[40583199]: Unable to forward frame Apr 20 18:39:02 WARNING[163851]: PRI: !! Got reject for frame 44, retransmitting frame 44 now, updating n_r! Apr 20 18:39:02 WARNING[163851]: PRI: !! Got reject for frame 44, retransmitting frame 45 now, updating n_r! Apr 20 18:39:40 WARNING[40812585]: Unable to forward voice Apr 20 18:40:08 WARNING[40878111]: Unable to forward frame Apr 20 18:41:26 WARNING[41025576]: Unable to forward voice Apr 20 18:41:30 WARNING[41091112]: Unable to forward frame Apr 20 18:41:55 WARNING[147466]: PRI: !! Got reject for frame 76, retransmitting frame 76 now, updating n_r! Apr 20 18:41:55 WARNING[147466]: PRI: !! Got reject for frame 76, retransmitting frame 77 now, updating n_r! -- On Wednesday 21 April 2004 04:40 pm, Scott Stingel wrote: Hi Darren- In my situation, the frame rejects/retries appear not to cause problems - even thousands of them, however I found that when I got a lot of unknown error 500 messages, they would be associated with stuck channels, ie channels that appeared to be in use when they are not. The stuck channels get cleared periodically because asterisk clears idle channels automatically every once in a while. However, if you get an error 500 during a call, I think the call drops (not sure). All of this, of course, only happens during pretty heavy load. It sounds like
Re: [Asterisk-Users] TxFax/SpanDSP problems
Almost all of our fax machines are Canon, so it's kinda tough to test TxFax. On Wed, 2004-04-21 at 11:35, Serge Oleinikov wrote: Hi Eric ! I have the same problem with Canon fax mashine as you have. I have wrote an email to Steve (developer of spandsp) some weeks ago and got no answer how to fix the problem :( Looks like connection was dropped by fax mashine without any reason - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, April 21, 2004 6:19 PM Subject: [Asterisk-Users] TxFax/SpanDSP problems I'm getting the following when sending to a specific fax machine. Any ideas? File name is '/var/spool/asterisk/email2fax/7F2SOeYJiU.tif' Changed from phase 0 to 2 Slow carrier up Slow carrier down Slow carrier up NSF: 20 00 00 11 80 00 8a 49 10 53 54 49 52 4c 49 4e 47 20 43 4f 56 49 4e 47 54 00 67 00 80 80 80 0c 01 02 NSF without final frame tag The remote is made by 'Canon' DIS: 80 20 ee 88 c4 80 95 80 80 80 38 DIS with final frame tag In state 10 DIS: V.8 capable Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter, V.29 and V.17 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) and B4 (364mm) Receiver's minimum scan line time: 20ms at 3.85 l/mm: T7.7 = T3.85 Error correction mode T.6 coding R8x15.4lines/mm OK R16x15.4lines/mm and/or 400x400pels/25.4 mm OK Metric-based resolution preferred Minimum scan line time for higher resolutions: T15.4 = T7.7 North American Letter (215.9mm x 279.4mm) North American Legal (215.9mm x 355.6mm) Single-progression sequential coding (Rec. T.85) basic DCS: Selected data signalling rate: V.29, 9600bps 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 20ms Start sending document Start tx document - compression 2 Fine mode Changed from phase 2 to 4 Sending ident TSI: 43 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DCS: 83 00 c4 00 HDLC underflow in state 3 Changed from phase 4 to 6 Changed from phase 6 to 3 Slow carrier up XCN: fa XCN with final frame tag In state 4 Disconnecting Changed from phase 3 to 7 Changed from phase 7 to 8 -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: Auto Answering PSTN -- Asterisk using X 100PCard
On Tue, 2004-04-20 at 18:07, [EMAIL PROTECTED] wrote: worked came to one ring only now. Thank you very much. If I use TE410 or TE405 instead of X100P. do it make that first ring disappear? Excuse me if already answered, but broken threads means I don't know till I am done reading all my mail if it was answered. If you go to a digital interface that is ISDN based like most E1 circuits and PRI on T1, then you can even make the first ring disappear. This is because signalling is done in the D channel as part of the endpoints negotiating which B channel to place the call through on. If you are on a channelized T1 though, you are just using digital versions of analog signalling, and the first ring may still be required depending on the signalling type. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 20, 2004 12:27 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Auto Answering PSTN -- Asterisk using X100PCard In article [EMAIL PROTECTED], [EMAIL PROTECTED] wrote: How can I remove callerid functionality? That was mentioned on this list only a couple of days ago, and will be in the mailing list archives. In zapata.conf you need to include the line usecallerid=no. Cheers, Tony -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: webvmail
On Wed, 2004-04-21 at 18:40, CW_ASN wrote:
No, you don't need to change permissions. Check in your voicemail.conf the
user password for accounts.
I don't know how vmail.cgi works with multiple contexts, or if you have
mysql/pgsql support with app_voicemail.
You need to log in with [EMAIL PROTECTED]
i.e. [EMAIL PROTECTED] or whatever.
You could edit vmail.cgi to change the default context. Look for the
line:
if (!$context) {
$context = default;
}
and change default to the default context you want it to be.
Regards,
--
-Barry Flanagan
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Re: [Asterisk-Users] Questions about alarm reporting in Asterisk
I'm interested in hearing success stories in tying things like Asterisk YELLOW and RED alarms and network problems into a central alarm reporting solution. The most common problems that I have found are: 1. Someone unplugs a X100P from the Dmarc and nobody knows until people complain that calls are not coming in. 2. A network span goes down and nobody knows until they can't send or receive calls on that span. Here are some ideas that I have thought about so far: 1. Installing a basic SNMP agent on each Linux box and using a central SNMP manager to monitor each node. This would give notice when a remote node became isolated from the monitoring network. 2. Rolling in Asterisk alarm logs into a syslog server or even as SNMP traps. Any good ideas would be appreciated! There are lots of different ways to sense problems including those you've mentioned. Others include: - writing a small app that simply interrogates those interfaces that are important to the operation (iax2/udp, sip/udp, etc, send a crafted pkt and interpret the returned result. Port not open is obvious, no response is obvious, incorrect response is not so obvious) - test call to an outside number once per five minutes, hourly, or whatever trips your trigger (outside number only needs to respond with something that is predictable, doesn't have to be a person or company) - monitoring logs looking for keywords (may take some time to identify the appropriate keywords) There are open source apps available that already address some of those, but weren't written specifically for *. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FWD SIP Asterisk IAX Firefly
Hello, In my sip.conf I have: ;Register and forward FWD numbers to internal extensions register = FWDNUMBER:[EMAIL PROTECTED]/9500 Which should register Asterisk at FWD and then when any calls are made to FWDNUMBER those calls should be forwarded to extension 9500 as specified in the extensions.conf. What I am getting is it is trying to dial the 9500 (IAX Firefly) client twice when a call is made to FWDNUMBER. The output from the CLI from Asterisk is this: -- Registered '9500' (AUTHENTICATED) at 10.10.20.5:4569 -- Executing Macro(SIP/-080fbd10, stdExt|IAX2|9500) in new stack -- Executing Dial(SIP/-080fbd10, IAX2/9500|15) in new stack -- Called 9500 -- Call accepted by 10.10.20.5 (format GSM) -- Format for call is GSM -- IAX2[9500]/3 is ringing -- Executing Macro(SIP/-0814f210, stdExt|IAX2|9500) in new stack -- Executing Dial(SIP/-0814f210, IAX2/9500|15) in new stack -- Called 9500 Apr 21 11:14:37 WARNING[1158883648]: chan_iax2.c:4898 socket_read: Call rejected by 10.10.20.5: In call -- Hungup 'IAX2[9500]/4' == No one is available to answer at this time -- Executing VoiceMail(SIP/-0814f210, u9500) in new stack -- Playing 'vm-theperson' (language 'en') -- Hungup 'IAX2[9500]/3' == Spawn extension (macro-stdExt, s, 1) exited non-zero on 'SIP/-080fbd10' in macro 'stdExt' == Spawn extension (default, 9500, 1) exited non-zero on 'SIP/-080fbd10' -- Playing 'digits/9' (language 'en') -- Playing 'digits/5' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'vm-isunavail' (language 'en') -- Playing 'vm-intro' (language 'en') -- Playing 'beep' (language 'en') Apr 21 11:14:49 WARNING[1142106688]: chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call 56da75f25ec [EMAIL PROTECTED] for seqno 104 (Response) == Spawn extension (macro-stdExt, s, 2) exited non-zero on 'SIP/-0814f210' in macro 'stdExt' == Spawn extension (default, 9500, 1) exited non-zero on 'SIP/-0814f210' Does anyone know why it is doing this? Have any suggestions? It worked yesterday well and then today I have been getting the above where when it dials the second time it goes to voicemail because the 9500 extension is in use. Thanks for the help! Darrin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] one-way audio and isdn4linux
Hi,
Apologies in advance for the lengthy email.
I'm new to asterisk and have trouble with isdn4linux.
The setup is very basic like this:
winxp --- asterisk winxp
x-lite | x-lite
|
pstn
The hardware involved is:
Compaq EVO with RH9/kernel 2.4.20-30.9.
Fritz!Card PCI v2
Asterisk CVS-04/17/04-21:36:18
Basically I run:
modprobe -a hisax_fcpcipnp
before I start asterisk with -vgc
The configs are like this:
MODEM.CONF:
[interfaces]
context=default
driver=aopen
driver=i4l
language=en
type=autodetect
stripmsd=0
dialtype=tone
mode=immediate
device = /dev/ttyI0
device = /dev/ttyI1
group=1
msn=0208080808
incomingmsn=*
device = /dev/ttyI0
device = /dev/ttyI1
EXTENSIONS.CONF:
[general]
static=yes
writeprotect=no
[globals]
TRUNK=Modem/g1
TRUNKMSD=0
[default]
exten = _9001,1,Dial(${TRUNK}:0208080809,60,tr)
Firewalls are disabled on all machines (doublechecked).
Sip calls between the two x-lite clients work consistently.
After I boot the asterisk server I can make one call succesfull to the
pstn (two-way voice). Subsequent calls from x-lite to pstn only give me
one-way voice; the direction that works is pstn-sip: a call is
cleary established, I just cannot understand why voice is lacking
sip-pstn or why it works the first time after a reboot.
I had a look on the sip-clients and on the server with ethereal, and
two-way traffic is happening consistently.
Show channels within the CLI saw 'frames in' and 'frames out' augmenting
on all channels.
Looking at /dev/isdnctrl I see slightly different isdn messages, but in
both cases I got a: L3DC ChangeState ST_L3_LC_ESTAB
I have tried to compile chan_capi but ran into the same issue as
described on a posting on april 12th:
[Asterisk-Users] Trouble compiling chan_capi on Suse 9.0
As I had two servers and two fritz cards I tried both with identical
software and both behave the same.
I hope I have made a stupid mistake and somebody on the list can point
it out to me. Every suggestion is of course most welcome.
I'm a bit unsure about the machines though: both machines crashed one or
more times completely (froze up) during a call to the pstn and dumped
core. Basically a hardware reset was all that I could do. One of these
times I was looking at the console and wrote down this:
Code: 8b 38 8b 58 04 8b 68 10 c7 44 24 18 00 00 00 00 ff 4c 24 40
0Kernel Panic: Aiee, killing interrupt handler!
In interrupt handler - not syncing
Thanks, andre
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Re: [Asterisk-Users] Limiting incoming SIP calls OriginalCallerID on transfer
OK,
I've fixed the '#' transfer problem. We setup a macro for dialing staff
extensions, however, it was missing the 'tr' options on the Dial
application:
[macro-staff-extension]
; Macro for Staff Extensions
exten = s,1,Dial(${ARG2},20,tr) --
exten = s,2,Voicemail(u${ARG1})
exten = s,102,Voicemail(b${ARG1})
exten = s,103,Hangup
I added the 'tr' and we can now perform call transfers while preserving
the correct CallerID information.
Thanks,
--
Erik Barker
Sr. Systems Engineer
NetNation Communications Inc.
http://www.netnation.com | http://www.domainpeople.com
On Tue, 2004-04-20 at 03:16, David Liu wrote:
Hi Erik,
Can you post your dial plan from incoming PSTN to the receptionist?
David
- Original Message -
From: Erik Barker [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, April 21, 2004 4:37 AM
Subject: Re: [Asterisk-Users] Limiting incoming SIP calls OriginalCallerID
on transfer
Thanks for the info David,
I'll look at getting the '#' transfer option working again I had it
working at some point where we used it to park calls, however, it does
not appear to work anymore.
--
Erik Barker
On Mon, 2004-04-19 at 11:13, David Liu wrote:
Hi Erik,
From my experience with Polycom phones, I can answer you on your
TRANSFER
and Caller ID issue. For Polycom, the transfer behavior is consultation
transfer. In consultation transfer mode, the caller ID of the
transferer is
passed to the ringing extension. To actually pass the caller ID of the
incoming caller on the PSTN, you would want to do a blind transfer. So
far,
I have only figured to use the Asterisk transfer option # to do blind
transfer. And this assumes you have the t option enabled on the dial
plan
to the receptionist.
Hope this helps.
David
- Original Message -
From: Erik Barker [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, April 20, 2004 6:19 PM
Subject: [Asterisk-Users] Limiting incoming SIP calls Original
CallerID on
transfer
I have 2 issues which I need to resolve on our production Asterisk
server:
We are currently using Polycom IP600 VOIP phones for our office which
are capable of handling 2 calls per SIP registration. What we're
finding
is when staff are on the phone, Asterisk will pass them a second call
which will show up on their display, and an audible beep is heard over
the phone (regular call waiting). I would like to limit the number of
calls sent to each phone to 1 call only; otherwise respond as being
busy. I have looked at trying to accomplish this in the sip.conf by
using the 'incominglimit' and 'outgoinglimit' parameters, however, the
only one that *seems* to work is the 'incominglimit'. This prevents
further calls from reaching the phones, rings busy, but does not allow
our phones to initiate a 2nd call OR transfer their existing call. The
'outgoinglimit' parameter does not seem to have any effect on limiting
whatsoever. Is there a way to limit calls passed to the phones from
Asterisk and also allow each phone to initiate 2 calls or transfer
calls
(disable call waiting)??
I have also looked at the WIKI for the parameters listed above and it
*appears* that 'outgoinglimit' should do what I want, however it also
states that this function has been disabled??
The _outgoinglimit__ is currently disabled in the source code of the
SIP channel.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20incominglimit
My second problem is that when external calls are transferred by our
receptionist to other staff members, the CallerID of course changes to
her Name instead of the original caller. Is there a way (in the
extensions logic or other) to preserve this CallerID information so
that
staff members receive calls with the proper CallerID information?
Thanks,
--
Erik Barker
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Re: [Asterisk-Users] Fw: Interconnecting to an Altigen PBX?
Ian McLaughlin wrote: Has anyone got Asterisk talking successfully to an Altigen PBX using h323? I can successfully make calls from Asterisk to Altigen, but calls from Altigen to Asterisk get a fast busy. There seems to be a lack of h323 example files (or maybe I'm looking in the wrong places) as well as a severe lack of h323 documentation from Altigen. Any pointers would be greatly appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have successfully got my Altigen talking h.323 to my asterisk server. There is an option when setting up h.323 connections in Altigen that essentally tells the Altigen to dial after so many digits have been entered.I had to disable that option on the Altigen side of things as it would not let me dial a full 11 digit number, it would send the first 4 digits or so. I am able to make calls from the * server to internal extensions on the Altigen just fiine, but when I try and grab an outside line it fails.have you been able to grab one of your Altigens outside lines via Asterisk? Isaac ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] TE410P zaptel Driver Situation
Dear Scott the reject warning is a bug? I must put in bug track? Thanks in advance Dimitri On Wednesday 21 April 2004 07:44 pm, Scott Stingel wrote: Dimitri- I'm not sure about the message Unable to forward voice - I don't get that one on my systems. The frame reject messages are the same ones I was talking about - these are load related, I'm pretty sure. Regards, Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of reseaux Sent: Wednesday, April 21, 2004 9:48 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] TE410P zaptel Driver Situation Dear Scott i have notice the same type of warning plus another with my TE410P with not very high load 30 IN/OUT line, i use only two span in this moment. What is only warning? or not? Thanks in advance Dimitri Apr 21 10:27:35 WARNING[966674]: Unable to forward voice Apr 21 10:28:41 WARNING[999442]: Unable to forward frame Apr 21 10:30:11 WARNING[1048594]: Unable to forward frame Apr 21 10:32:02 WARNING[1130515]: Unable to forward frame Apr 21 10:34:56 WARNING[1212434]: Unable to forward frame Apr 21 10:35:45 WARNING[1245202]: Unable to forward frame Apr 21 10:36:34 WARNING[1277970]: Unable to forward frame Apr 21 10:37:02 WARNING[1294354]: Unable to forward frame Apr 21 10:40:03 WARNING[1359891]: Unable to forward frame Apr 21 10:45:51 WARNING[1441811]: Unable to forward frame Apr 21 10:47:15 WARNING[1474579]: Unable to forward frame Apr 21 10:47:21 WARNING[1490963]: Unable to forward frame Apr 21 10:49:57 WARNING[1540115]: Unable to forward frame Apr 21 10:50:26 WARNING[1572883]: Unable to forward frame Apr 21 10:51:25 WARNING[1622035]: Unable to forward voice Apr 21 11:11:42 WARNING[1851410]: Unable to forward frame Apr 21 11:19:20 WARNING[2097170]: Unable to forward frame Apr 21 11:29:17 WARNING[2228243]: Unable to forward voice Apr 21 11:31:39 WARNING[2375700]: Unable to forward frame Apr 21 11:45:30 WARNING[2588692]: Unable to forward frame Apr 21 11:53:00 WARNING[2686995]: Unable to forward frame Apr 21 11:56:14 WARNING[2752531]: Unable to forward frame Apr 21 11:58:31 WARNING[2850837]: Unable to forward voice Apr 21 11:59:13 WARNING[2867219]: Unable to forward voice Apr 21 12:01:41 WARNING[3047443]: Unable to forward frame Apr 21 12:02:21 WARNING[3112979]: Unable to forward voice Apr 21 12:03:27 WARNING[3145747]: Unable to forward voice Apr 21 12:07:49 WARNING[3325970]: Unable to forward frame Apr 21 12:13:49 WARNING[3506198]: Unable to forward frame Apr 21 12:18:32 WARNING[3686422]: Unable to forward frame Apr 21 12:27:03 WARNING[3768339]: Unable to forward frame Apr 21 12:29:35 WARNING[3899413]: Unable to forward frame Apr 21 12:29:40 WARNING[3932181]: Unable to forward voice Apr 21 12:43:15 WARNING[4177942]: Unable to forward voice Apr 21 12:45:21 WARNING[4194322]: Unable to forward frame Apr 21 12:46:34 WARNING[4227090]: Unable to forward frame Apr 21 12:47:24 WARNING[4243474]: Unable to forward frame Apr 21 13:05:48 WARNING[4653074]: Unable to forward frame Apr 21 13:06:20 WARNING[4685843]: Unable to forward frame Apr 21 13:21:57 WARNING[5062678]: Unable to forward frame Apr 21 13:36:04 WARNING[5390360]: Unable to forward frame Apr 21 13:38:55 WARNING[5521428]: Unable to forward frame Apr 21 13:42:15 WARNING[5783576]: Unable to forward frame - Apr 20 17:42:45 WARNING[163851]: PRI: !! Got reject for frame 112, retransmitting frame 11 2 now, updating n_r! Apr 20 17:42:45 WARNING[163851]: PRI: !! Got reject for frame 112, retransmitting frame 11 3 now, updating n_r! Apr 20 18:38:48 WARNING[40583199]: Unable to forward frame Apr 20 18:39:02 WARNING[163851]: PRI: !! Got reject for frame 44, retransmitting frame 44 now, updating n_r! Apr 20 18:39:02 WARNING[163851]: PRI: !! Got reject for frame 44, retransmitting frame 45 now, updating n_r! Apr 20 18:39:40 WARNING[40812585]: Unable to forward voice Apr 20 18:40:08 WARNING[40878111]: Unable to forward frame Apr 20 18:41:26 WARNING[41025576]: Unable to forward voice Apr 20 18:41:30 WARNING[41091112]: Unable to forward frame Apr 20 18:41:55 WARNING[147466]: PRI: !! Got reject for frame 76, retransmitting frame 76 now, updating n_r! Apr 20 18:41:55 WARNING[147466]: PRI: !! Got reject for frame 76, retransmitting frame 77 now, updating n_r! -- On Wednesday 21 April 2004 04:40 pm, Scott Stingel wrote: Hi Darren- In my situation, the frame rejects/retries appear not to cause problems - even thousands of them, however I found that when I got a lot of unknown error 500 messages, they would be associated with stuck channels, ie channels that appeared to be in use when they are not. The stuck channels get cleared periodically because asterisk
[Asterisk-Users] g729 problem HELP!
Dear i have buy two license of G729 codec and i have install/registered as documented but after i start Asterisk -vvvcng i notice this warning and if i made call the CLI say No compatible codec! How can i solve this problem? Thanks in advance Dimitri -- [app_datetime.so] = (Date and Time) == Registered application 'DateTime' [codec_g729b.so] = (Annex B (floating point) G.729/PCM16 Codec Translator) == Detected 2 licensed G.729 transcoders Apr 21 20:52:15 WARNING[16384]: translate.c:213 calc_cost: Translator 'g729tolinb' does n t produce sample frames. == Registered translator 'g729tolinb' from format G729A to SLINR, cost 9 == Registered translator 'lintog729b' from format SLINR to G729A, cost 43 == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI Apr 21 20:52:17 NOTICE[131081]: chan_sip.c:5880 sip_poke_noanswer: Peer 'santoext' s now UNREACHABLE! Apr 21 20:52:36 WARNING[131081]: chan_sip.c:2113 process_sdp: No compatible codecs! Apr 21 20:52:38 NOTICE[131081]: chan_sip.c:5337 handle_request: Failed to authenticate us r sip:[EMAIL PROTECTED]:5060;tag=c398050c4b92f090 *CLI -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] TxFax/SpanDSP problems
It would be VERY nice if TxFax exited with a non-zero return code if the fax was not actually sent. In the case of the problem with sending to the Canon fax machine, this always returns 0: $status = $AGI-exec(TxFAX, $fax_filename|caller); On Wed, 2004-04-21 at 12:46, Eric Wieling wrote: Almost all of our fax machines are Canon, so it's kinda tough to test TxFax. On Wed, 2004-04-21 at 11:35, Serge Oleinikov wrote: Hi Eric ! I have the same problem with Canon fax mashine as you have. I have wrote an email to Steve (developer of spandsp) some weeks ago and got no answer how to fix the problem :( Looks like connection was dropped by fax mashine without any reason - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, April 21, 2004 6:19 PM Subject: [Asterisk-Users] TxFax/SpanDSP problems I'm getting the following when sending to a specific fax machine. Any ideas? File name is '/var/spool/asterisk/email2fax/7F2SOeYJiU.tif' Changed from phase 0 to 2 Slow carrier up Slow carrier down Slow carrier up NSF: 20 00 00 11 80 00 8a 49 10 53 54 49 52 4c 49 4e 47 20 43 4f 56 49 4e 47 54 00 67 00 80 80 80 0c 01 02 NSF without final frame tag The remote is made by 'Canon' DIS: 80 20 ee 88 c4 80 95 80 80 80 38 DIS with final frame tag In state 10 DIS: V.8 capable Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter, V.29 and V.17 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) and B4 (364mm) Receiver's minimum scan line time: 20ms at 3.85 l/mm: T7.7 = T3.85 Error correction mode T.6 coding R8x15.4lines/mm OK R16x15.4lines/mm and/or 400x400pels/25.4 mm OK Metric-based resolution preferred Minimum scan line time for higher resolutions: T15.4 = T7.7 North American Letter (215.9mm x 279.4mm) North American Legal (215.9mm x 355.6mm) Single-progression sequential coding (Rec. T.85) basic DCS: Selected data signalling rate: V.29, 9600bps 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 20ms Start sending document Start tx document - compression 2 Fine mode Changed from phase 2 to 4 Sending ident TSI: 43 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DCS: 83 00 c4 00 HDLC underflow in state 3 Changed from phase 4 to 6 Changed from phase 6 to 3 Slow carrier up XCN: fa XCN with final frame tag In state 4 Disconnecting Changed from phase 3 to 7 Changed from phase 7 to 8 -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Ser and Asterisk together
We are using ser together with *. Ser is used as a SIP proxy/registrar, * is used as a sip - pstn gateway and voicemail/forward/conference server. Advanteges - scalable, very large number of sip clients with easier radius/database user management, advanced sip logic/routing options, better sip interoperability disadvantages - you've got two boxes, no iax on ser so you still have to manage iax users on asterisk In my opinion, if you plan to deploy large number of sip clients - it's a good idea Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of AJ Grinnell Sent: Wednesday, April 21, 2004 3:42 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Ser and Asterisk together Anybody out there use Ser along with *? Any advantages disadvantages? Is this even a good idea? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7940/7960 SIP functionality questions
Hello, I'm considering using Asterisk with some type of Cisco phone, and currently considering either the 7940 or 7960 because of its more-complete functionality (compared to the 7905). I'm currently wondering: Do all the expected functions (transfer, conference, voice mail, message waiting indicator, etc.) work normally with Asterisk over SIP? What caveats are known about using these phones with SIP, as opposed to Cisco's proprietary SCCP? If an SCCP module is available for Asterisk, how functional is it? How customizable are the phone menus while using SIP (or if a SCCP module is available, using SCCP)? Cisco doesn't seem to have much documentation online about using these phones in SIP mode, so if anyone is using these phones now, I'd appreciate hearing about your experiences. Thanks! -- David Carter ** [EMAIL PROTECTED] ** [EMAIL PROTECTED] PGP Key 581CBE61: E07EE199C767C752 8A8B1A9F015BF2EA Key available at www.keyserver.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 choosing a UK IAX provider
Yeah, primarily fired at them from large telcos with infinite bandwidth... At 11:01 4/21/2004, you wrote: Wait till DDOS/extortion scams start hitting voip providers! Panny ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * and CCM Voicemail questions
I am using a CallManager Server at work and all my phones register with it. I have built an Asterisk server and am connecting to the CCM with OH323. This is working great and am going to get more features working as time permits. I have calls going both ways between these servers. The question that still keeps popping up from my users is 1. How do I get the Cisco phone to light the MWI light on the handset? 2. How do I get the Voicemail Messages button to work? I am really looking for info on where I can get this information as I am trying to learn as much as I can but time is the biggest problem for me right now. If only I didn't sleep. Thanks in advance Keith
Re: [Asterisk-Users] Cisco 7940/7960 SIP functionality questions
On Apr 21, 2004, at 12:20 PM, David Carter wrote: Hello, I'm considering using Asterisk with some type of Cisco phone, and currently considering either the 7940 or 7960 because of its more-complete functionality (compared to the 7905). I'm currently wondering: Do all the expected functions (transfer, conference, voice mail, message waiting indicator, etc.) work normally with Asterisk over SIP? I've tested transfer, voicemail, and MWI. Conference should work have I haven't tested it. What caveats are known about using these phones with SIP, as opposed to Cisco's proprietary SCCP? If an SCCP module is available for Asterisk, how functional is it? There are two SCCP modules, but I haven't heard about anyone using 7940/60s with SCCP and Asterisk. The phone works very well with SIP. How customizable are the phone menus while using SIP (or if a SCCP module is available, using SCCP)? To the best of my knowledge, the phone's internal menus aren't customizable at all. However, the 'services' button on the front panel works just like a menu button and fetches XML content over HTTP. You can get a good overview of the syntax and possibilities from http://www.cisco.com/univercd/cc/td/doc/product/voice/vpdd/cdd/3_0/ phsvcdev.pdf In addition to the service button, you can also change some of the content reached from the directory button. With SIP firmware v6.3, there are 5 menus available from the directory button: missed calls incoming calls outgoing calls personal directory external directory (I may have the wording and order slightly wrong, I'm not in front of my phone right now). The first 3 menus are built into the phone and are maintained automatically. You can copy names and numbers from these three into the personal directory. I assume that the personal directory is limited to ~32 entries, but I haven't tested it. Finally, the external directory option fetches XML from a web server, just like the services button. You can use it to implement a shared phone directory. Cisco doesn't seem to have much documentation online about using these phones in SIP mode, so if anyone is using these phones now, I'd appreciate hearing about your experiences. Yeah, the documentation is kind of sparse. Google can help. Here are a couple useful pages that I've tracked down: http://www.cisco.com/global/FR/documents/pdfs/ciscotheque/ journee_developpeurs/ 03_Cisco_CallManager_Version_3_3_IP_Phone_Services_nchrisso.pdf http://www-106.ibm.com/developerworks/wireless/library/wi-voip/ Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Ser and Asterisk together
Thanks, those are the advantages I needed to hear. Is there any special config I need to do to either * or SER? Do I just set SER as a friend in sip.conf? Still looking for documentation on using the two together. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dawid Mielnik Sent: Wednesday, April 21, 2004 3:15 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Ser and Asterisk together We are using ser together with *. Ser is used as a SIP proxy/registrar, * is used as a sip - pstn gateway and voicemail/forward/conference server. Advanteges - scalable, very large number of sip clients with easier radius/database user management, advanced sip logic/routing options, better sip interoperability disadvantages - you've got two boxes, no iax on ser so you still have to manage iax users on asterisk In my opinion, if you plan to deploy large number of sip clients - it's a good idea Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of AJ Grinnell Sent: Wednesday, April 21, 2004 3:42 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Ser and Asterisk together Anybody out there use Ser along with *? Any advantages disadvantages? Is this even a good idea? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] TE410P zaptel Driver Situation
Dear Scott the reject warning is a bug? I must put in bug track? Thanks in advance Dimitri No, not a bug I don't think. A warning that the framer driver was not able to keep up with the PRI bit stream. Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England URL:www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Questions about alarm reporting in Asterisk
On Wed, 2004-04-21 at 18:41, Ernest W. Lessenger wrote: We use a package called Nagios to monitor our servers, which works quite well. It has the ability to track service and host dependencies so you don't get flooded with a bunch of service down alerts when the real cause is a bad switch (or similar). Nagios is great :) Here is some basic info on integration with Asterisk: http://www.voip-info.org/tiki-index.php?page=Asterisk+monitoring It would seem logical for someone (hah!) to write a res_snmp.c for asterisk that would expose a lot of asterisk's internal data. This would seem a logical step toward writing fully functional monitoring applications as well. The module would allow clients to add themselves to the list and receive traps, as well as check for the current status of various variables. brainstorming Okay, this may be over the top, but here goes. Write an asterisk application that sends (and receives) status information to another box over the PSTN. My idea is not only to use this as a way to verify that * is running, but as a way to RELIABLY tell that a remote * box is actively accepting incoming calls. It wouldn't have to be anything complicated, just a heartbeat and some basic details to let the caller know that yes, I'm alive and accepting calls over this line. Simplified protocol: 1) Monitoring box calls up and says (in DTMF): #my CallerID#extension I am trying to reach#I'm a machine, so reply in DTMF instead of voice#the secret code is# 2) The remote box says #your CallerID#Your DNIS#yes I will accept a call to that number# 3) Monitoring box acknowledges and disconnects 4) Remote box disconnects 5) Monitoring box decides whether it likes the answers it received and performs actions accordingly. /brainstorming Great stuff - I've added this the other comments to the Wiki page :) - please keep adding stuff there as it's an important area where we could benefit from sharing ideas ( implementations!) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Ser and Asterisk together
Hello, From: Dawid Mielnik [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Ser and Asterisk together Date: Wed, 21 Apr 2004 21:15:21 +0200 We are using ser together with *. Ser is used as a SIP proxy/registrar, * is used as a sip - pstn gateway and voicemail/forward/conference server. Advanteges - scalable, very large number of sip clients with easier radius/database user management, advanced sip logic/routing options, better sip interoperability Our's is only a SIP based system and we use SER in front of Asterisk as SIP Proxy/Registrar. Asterisk is mainly used as a Media Server that plays IVR and for voice mails. Yes, it is highly scalable and database management is easy. We dont have SIP peers for Asterisk. It just plays the IVR and routes the call back to SER when it receives dtmf. As told by OEJ in a mail off the list, it is nothing but just a SIP call for Asterisk. Currently our system is under testing. disadvantages - you've got two boxes, no iax on ser so you still have to manage iax users on asterisk We are using only one box, SER is on 5060 and Asterisk on a different port. In my opinion, if you plan to deploy large number of sip clients - it's a good idea Very true. Dave Regards, Girish _ Marriage? http://www.bharatmatrimony.com/cgi-bin/bmclicks1.cgi?72 Join BharatMatrimony.com for free. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Ser and Asterisk together
On Wed, 2004-04-21 at 21:02, AJ Grinnell wrote: Thanks, those are the advantages I needed to hear. FWD SipGate apparently have this config: http://www.voip-info.org/wiki-Asterisk+at+large Is there any special config I need to do to either * or SER? Do I just set SER as a friend in sip.conf? Still looking for documentation on using the two together. http://www.voip-info.org/wiki-Asterisk+config+sip.conf (See Example 2) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] TE410P zaptel Driver Situation
On Wed, Apr 21, 2004 at 01:17:23PM -0700, Scott Stingel wrote: Dear Scott the reject warning is a bug? I must put in bug track? Thanks in advance Dimitri No, not a bug I don't think. A warning that the framer driver was not able to keep up with the PRI bit stream. Zaptel hardware handles HDLC framing in software. Thus, if for some reason a Zaptel frame (8 bytes) is lost, a full HDLC frame will be lost, and it will be detected on the next one, thus the error message. The main reason it can happen is probably because of the server load. The driver can't poll the zaptel device fast enough. However, it's not clear to me how it can happen on busmaster devices such as T405P and T410P. Maybe a double buffering issue, or PCI bus load. Or maybe an interrupt fault; the Zaptel hardware provides a kHz interrupt to the driver for polling, and an interrupt might get lost (particularly if the IRQ line is shared). However, it seems to be detected by the driver (and it should print a warning). -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 choosing a UK IAX provider
On Wed, Apr 21, 2004 at 04:18:51PM +0100, Craig Waddington wrote: 1 1 ms 2 ms 1 ms 10.5.0.1 217 ms14 ms14 ms 195.10.119.94 317 ms14 ms14 ms 195.10.119.158 422 ms14 ms15 ms 217.23.160.1 515 ms15 ms31 ms 217.23.162.122 617 ms15 ms14 ms 217.23.160.85 719 ms18 ms14 ms 217.23.160.186 830 ms26 ms29 ms tier1-1.BUD2.psie.net [154.14.68.113] 931 ms39 ms29 ms linx1.teleglobe.net [195.66.224.51] 1026 ms28 ms30 ms if-0-0-0.bb2.London.Teleglobe.net [195.219.96.81 ] 1159 ms87 ms 108 ms ix-3-1-0-822.bb2.London.Teleglobe.net [195.219.2 .34] 1276 ms54 ms54 ms wi2.westloc.com [82.145.32.2] 13 229 ms 239 ms 187 ms wc3-10.westloc.com [82.145.32.73] This last hop may be the source of your problem. Since I believe it's not a trans-continent link, it's either : - a very congestioned link - a router with serious problems at hop 13 (or maybe 12). You should contact whoever manages westloc.com -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MWI forwarding
As far as I understand how voicemail is integrated into Asterisk, it seems that SIP channels poll MWI directly from the filesystem. Is it possible (feasible?) to have something like : - a central voicemail server - which has an IAX peer with a mailbox= line with tens of VM boxes - this peer has itself tens of SIP phones connected to it. And it would alert the SIP phones when it receives MWI over the IAX channel from the central server. -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ok, Im confused
Im totally a newbee at * Im confused. Ive got a FWD account, and it works on the winboxen. Ive got * up and can do the echotest etc, so its working. I want to get FWD working, and all the pages ive seen on setup are most confusing. Is FWD setup like IAXTEL? Do i plug in my FWD info in the same places as the IAXTEL stuff? Ive been trying for a week now, and Im more lost than before. Ive got a Internet phonejack card in the penguin, phone0, and all I want to do at this point is make and receive calls thru FWD using that jackIll plug the house in later...Ill learn the other stuff later. No voicemail, no BS, no dial thru least cost routing, or nightlines just make it work as a phone. Im either more stupid than I think, or Im missing something major here. Ive got to the point the CLI shows me connected to FWD fine.(I think) Sip show users UsernameSecret Authen Def. Contexta/c fwd.pulver.com secret md5,plaintext default no Need some basic, stupidly simple scripts here...I dont need or want to dial 1-700 or *9 or any other crap, just make it work like the stupid winbox phone for now...Ill read the wiki for a couple years, and then maybe I can do voicemail or whatever... frustrated...and I know its showing...sri ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Make an H323 phone act like a SIP ohone
I have some Grandstream BT101 SIP phones. Work great (so far). I have some Planet VIP-101T H323 phones... how do I make them look/feel/act like a SIP phone I can dial to them from both Trunk + SIP's (ie - I've added 'oh323' libraries) What config do I add so that if I dial the * IP - they then at least act as an extension? Ideally I'd like to just pick up the handset, and dial a number - just like the SIP phones... Pointers please? -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 signature.asc Description: This is a digitally signed message part
[Asterisk-Users] weird IAX2 things going on
Hi all, I have 3 * boxes all running the same OS and software version. Machine A has an X100P card, machines B and C do not. They all have the same dialplan. I can dial from machine A to either of the other 2 with no problem. I can dial from either of the other 2 to machine A with no problem. I cannot dial from B to C or vice versa. What's really wierd is that I can dial from machine B through machine A to machine C. The IAX2 session then drops machine A from the picture and continues directly between B and C. The same happens in reverse. Why can I not dial directly from B to C? Do B and C require X100P cards before IAX2 will work correctly? I don't think this is the case because the call can be passed to them after it has been setup via A. Its really bugging me. Any ideas? G7LTT/KC2ENI Mark Phillips ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] weird IAX2 things going on
On Wed, 2004-04-21 at 17:31, Mark Phillips wrote: Hi all, I have 3 * boxes all running the same OS and software version. Machine A has an X100P card, machines B and C do not. They all have the same dialplan. I can dial from machine A to either of the other 2 with no problem. I can dial from either of the other 2 to machine A with no problem. I cannot dial from B to C or vice versa. What's really wierd is that I can dial from machine B through machine A to machine C. The IAX2 session then drops machine A from the picture and continues directly between B and C. The same happens in reverse. Why can I not dial directly from B to C? Do B and C require X100P cards before IAX2 will work correctly? I don't think this is the case because the call can be passed to them after it has been setup via A. Its really bugging me. Any ideas? Is B and C registering to each other like they are to A? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Ok, Im confused
Look here: http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: tmpm [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, April 21, 2004 11:50 AM Subject: [Asterisk-Users] Ok, Im confused Im totally a newbee at * Im confused. Ive got a FWD account, and it works on the winboxen. Ive got * up and can do the echotest etc, so its working. I want to get FWD working, and all the pages ive seen on setup are most confusing. Is FWD setup like IAXTEL? Do i plug in my FWD info in the same places as the IAXTEL stuff? Ive been trying for a week now, and Im more lost than before. Ive got a Internet phonejack card in the penguin, phone0, and all I want to do at this point is make and receive calls thru FWD using that jackIll plug the house in later...Ill learn the other stuff later. No voicemail, no BS, no dial thru least cost routing, or nightlines just make it work as a phone. Im either more stupid than I think, or Im missing something major here. Ive got to the point the CLI shows me connected to FWD fine.(I think) Sip show users Username Secret Authen Def. Context a/c fwd.pulver.com secret md5,plaintext default no Need some basic, stupidly simple scripts here...I dont need or want to dial 1-700 or *9 or any other crap, just make it work like the stupid winbox phone for now...Ill read the wiki for a couple years, and then maybe I can do voicemail or whatever... frustrated...and I know its showing...sri ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 Operator Unallocated number message
We have set up an Asterisk PBX managing a EuroPRI in Italy. We have conneccted to the asterisk PBX some Cisco IP Phones and a Panasonic PBX with 10 analogic phones. If we dial an unassigned telephone number we are not able to listen to PSTN Operator message telling that the subscriber does not exist both on IP phones and analogic ones. Asterisk simply hungs up. We have repeated the test using a X100P card and everything works fine. Can someone help us? This is the pri debug trace of the call. -- Making new call for cr 35670 Protocol Discriminator: Q.931 (8) len=32 Call Ref: len= 2 (reference 2902/0xB56) (Originator) Message type: SETUP (5) Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Unknown (67) '' ] Called Number (len=13) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '0816283509' ] -- Called g1/0816283509 Protocol Discriminator: Q.931 (8) len=14 Call Ref: len= 2 (reference 35670/0x8B56) (Terminator) Message type: SETUP ACKNOWLEDGE (13) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Processing IE 24 (Channel Identification) -- Processing IE 30 (Progress Indicator) Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 35670/0x8B56) (Terminator) Message type: DISCONNECT (69) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Processing IE 8 (Cause) -- Processing IE 30 (Progress Indicator) -- Channel 1, span 1 got hangup NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 2902/0xB56) (Originator) Message type: RELEASE (77) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/1-1' == No one is available to answer at this time Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 35670/0x8B56) (Terminator) Message type: RELEASE COMPLETE (90) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null Apr 20 23:14:03 WARNING[50937879]: pbx.c:1836 ast_pbx_run: Timeout, but no rule 't' in context 'sip' ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] ANI II/Payphone indication
Quickie: Does anyone out there have experience with PRI delivery of ANI II information? Our carrier appends it to the DNIS. For instance, if I call from my cellphone, we get: 877852000263 Where 8778520002 is the dialed number, and 63 are the info digits. So your carrier provides you with 12 digit DNIS? Is this something special, achieved after a bit of nudging and winking at your friendly CO technician, or is it a standard service offering? Any magic words/incantations used to get this feature? I'm curious as to your provider too.. Looks exactly like what I'm looking for though - thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Ok, Im confused
Thanks Jim, But that page started my trip off to confusionbeen theretried it 10 different ways...still no joy. I'll go through it once again, maybe Im missing something, I dont know. Im about ready to boot the penguin to the curb... I know its in there...I think Ive got it all configured, and I dial the outbound strings, and get a fast busy...I know one stinking letter off, and its whacked... HOW for example do I specify my one and only extension is the Internet phone jack? Phone0? Somehow theres got to be a tie-in...I cant find it. been thru extensions.conf, phones.conf, sip.conf..etc. groan.. At 18:40 4/21/2004, you wrote: Look here: http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: tmpm [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, April 21, 2004 11:50 AM Subject: [Asterisk-Users] Ok, Im confused Im totally a newbee at * Im confused. Ive got a FWD account, and it works on the winboxen. Ive got * up and can do the echotest etc, so its working. I want to get FWD working, and all the pages ive seen on setup are most confusing. Is FWD setup like IAXTEL? Do i plug in my FWD info in the same places as the IAXTEL stuff? Ive been trying for a week now, and Im more lost than before. Ive got a Internet phonejack card in the penguin, phone0, and all I want to do at this point is make and receive calls thru FWD using that jackIll plug the house in later...Ill learn the other stuff later. No voicemail, no BS, no dial thru least cost routing, or nightlines just make it work as a phone. Im either more stupid than I think, or Im missing something major here. Ive got to the point the CLI shows me connected to FWD fine.(I think) Sip show users Username Secret Authen Def. Context a/c fwd.pulver.com secret md5,plaintext default no Need some basic, stupidly simple scripts here...I dont need or want to dial 1-700 or *9 or any other crap, just make it work like the stupid winbox phone for now...Ill read the wiki for a couple years, and then maybe I can do voicemail or whatever... frustrated...and I know its showing...sri ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Questions about alarm reporting in Asterisk
At 9:16 PM +0100 on 4/21/04, Fran Boon wrote: On Wed, 2004-04-21 at 18:41, Ernest W. Lessenger wrote: We use a package called Nagios to monitor our servers, which works quite well. It has the ability to track service and host dependencies so you don't get flooded with a bunch of service down alerts when the real cause is a bad switch (or similar). Nagios is great :) Here is some basic info on integration with Asterisk: http://www.voip-info.org/tiki-index.php?page=Asterisk+monitoring It would seem logical for someone (hah!) to write a res_snmp.c for asterisk that would expose a lot of asterisk's internal data. This would seem a logical step toward writing fully functional monitoring applications as well. The module would allow clients to add themselves to the list and receive traps, as well as check for the current status of various variables. brainstorming Okay, this may be over the top, but here goes. Write an asterisk application that sends (and receives) status information to another box over the PSTN. My idea is not only to use this as a way to verify that * is running, but as a way to RELIABLY tell that a remote * box is actively accepting incoming calls. It wouldn't have to be anything complicated, just a heartbeat and some basic details to let the caller know that yes, I'm alive and accepting calls over this line. Simplified protocol: 1) Monitoring box calls up and says (in DTMF): #my CallerID#extension I am trying to reach#I'm a machine, so reply in DTMF instead of voice#the secret code is# 2) The remote box says #your CallerID#Your DNIS#yes I will accept a call to that number# 3) Monitoring box acknowledges and disconnects 4) Remote box disconnects 5) Monitoring box decides whether it likes the answers it received and performs actions accordingly. /brainstorming Great stuff - I've added this the other comments to the Wiki page :) - please keep adding stuff there as it's an important area where we could benefit from sharing ideas ( implementations!) F There is an SNMP module for Asterisk already, but it is apparently not widely used. I've added this to the Wiki as well in response to questions about SNMP for Asterisk: http://www.faino.it/en/ast-ax-snmpd.html I took a look through this code, and it seemed to have quite a few nice things that could easily be monitored through the interface provided. Instead of traps (yuck!) it allows polls, which could then be graphed with RRDTool or monitored with Nagios easily enough... Lots of work yet to be done on it, I'm sure, but I'm sure there are good coders out there who could submit patches and updates to it. If I had time I'd implement SNMPv3 into all my * servers for monitoring... JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] ANI II/Payphone indication
Quickie: Does anyone out there have experience with PRI delivery of ANI II information? There is no IE ( information element) in the isdn setup for this indicator. Of course with ISUP(SS7) FGD trunks it is delivered in the OLI ISUP parameter Of course! ;-) But we're not quite there with SS7 straight in to Asterisk just yet are we? And that's before we get to the regulatory/interconnect agreements etc.. So.. Maybe I can get it on PRI by having it appended to the DNIS (as per Ryan Tucker's reply).. or maybe through some other kind of T1 arrangement that isn't PRI? Excuse my ignorance here but back in Europe it's pretty much primary rate ISDN or nothing, the whole 17 flavours of T1 is a bit of an unknown.. Cheers Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
