[Asterisk-Users] Do Sifira use Asterisk?
Hi, I am looking for what sifira use to provide its services like Callrecorder, Family voicemail..etc Does it uses Asterisk, if yes, for what specific services? Thanks Gurminder ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with queue and remote agents
You should either use Agents (standard or callback) or disable voicemail on the second server, with a straight dial instead of the dial+voicemail macro you'll likely be using. bye l. In data Fri, 23 Sep 2005 17:15:38 +0200, [EMAIL PROTECTED] ha scritto: I all. I have configured a pair of * servers, sip connected each other Mi problem is the following If on the first * i configure a queue containing phone number of the second * (i.e with a round robin strategy) I have non problem as far as all phones are online. If one of the remote phone number is unavailable, when the round-robin strategy touch that phone the call is answered by the voicemail (the extension is onthephone or is unavailable) I think that the problem could be the first * pass the call to the second, and has no way to decide if the remote extension is available or not Could be an improvement to iax interconnect the two asterisk ? Or is there any othe solution ? I already removed static agent from the queue, but the problem is the same if one remote extensions is loggd in but is busy Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Assum est, versa et manduca. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VM low volume - testers needed
The patch is in cvs-head, which has been very stable for me. :) Hi Richard, I am experiencing the same problem. I'd like to test your patch. Thing, is, I don't know which CVS it's in :) ... I checked out 1.2-beta on Tuesday (9/21) and compiled it. When I type 'show application voicemail', it does not describe the g(#) option, so I think my version must not have it. I am using a TDM22B card and voicemails seem very quiet if they are left from in incoming POTS connection. When I enter voicemail by direct dialing a local extension and leave a message from the advanced options menu, the recorded message is much louder. I should qualify, not only are my VMs coming in over POTS, I am actually calling out first through the TDM22B, to Sipura, to VOIP provider, back in via PSTN, to TDM22B, to VM. I'm amazed it works at all :) ... I'm very impressed by Asterisk and especially it's voicemail. I would like to resolve the low volume issue though. If you can tell me which CVS to check out, I can try it. I'd like to stick to the 1.2-beta branch though because I don't want to rework all my config files. On 9/21/05, Rich Adamson [EMAIL PROTECTED] wrote: On Monday 19 September 2005 12:38, Rich Adamson wrote: The g(6) adds a 6 db gain for zap calls that end up recording a Voicemail message. ... * 'g(#)' the specified amount of gain will be requested during message recording (units are whole-number decibels (dB)) How in the hell does that make any sense? are your normal incoming calls quiet too or just voicemail? Yes, see bug 2022 and 2023 for details, as well as http://www.routers.com/asteriskprob/asterisk-config.htm for a very detailed analysis of the problem. I believe one of the more serious issues amounts to: if asterisk is located a fair distance from the central office (-7db in my case), setting the rxgain and/or txgain to any level that would be considered reasonable for that loss (eg, rxgain=5, txgain=5), hugh amounts of echo result that cannot be addressed through zapata.conf echo entris, and changing compile options to agressive, etc, does not help. Its my believe (from working with several TDM users), the further one is from the CO, the bigger the problem. (Or, short pstn cable lengths less then about 4 or 5db can almost always be addressed via parameters.) The above workaround is very usable (assuming it works) when someone calls in via the pstn and leaves a voicemail (which is already at least 7db down plus their own pstn loss), and then I call in via the pstn to retrive the voicemail (now 14db down PLUS the original callers pstn loss), the audio is so faint its difficult to impossible to listen to. In my case, the asterisk box is located about 7db from the central office. As noted in bug 2023 (and 2022), calls from an outside pstn line coming into asterisk incure a 7db pstn loss (which can't be adjusted for with rxgain and txgain as changing those values to something reasonable generates echo). Retrieving that VM message from an outside location creates another 7db loss (now -14db down in total), making it very difficult (if not impossible) to hear the message. (And, yes I've gone through all the recommendations with wav vs gsm files, etc.) I am not sure I understand why the txgain/rxgain isn't fixing it without adding unacceptable echo... this all seems very odd... I mean for a test you should be able to dial an echo() application and have extremely quiet echoed audio... is this the case? As an ex-telco transmission engineer, believe me I've done my homework and some very solid testing with expensive well-calibrated test equipment. As I've mentioned to Kevin, its almost like the TigerJet pci controller on the TDM card is reversing bits six and seven (or something very odd like that). Digium apparently now has a pci engineering type looking at the issues, which I'm told is using a pci logic analyzer, etc. The work around only kicks in if the call comes from a zap channel and ends up in voicemail, adding a 6db gain to that recorded message. No other channel types are impacted by this new parameter. This is a HELL of a band-aid. If you actually follow the logic that was originally stated in 2023, this gain setting is highly useful for those systems that are further away from the CO (as mentioned above). For those closer to the CO, it has zero value. Rich ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com
[Asterisk-Users] wrong password on authentication for INVITE to 'asterisk
Hi list: i tried to send calls through an asterisk box to a voip provider the calls failed and here what i got : *CLI Sep 24 11:09:19 WARNING[23356]: chan_sip.c:6890 handle_response: Forbidden - wrong password on authentication for INVITE to 'asterisk sip:[EMAIL PROTECTED]:5070;tag=as667cb0ae' -- SIP/call-3f73 is circuit-busy == Everyone is busy/congested at this time -- Got SIP response 481 Call Leg Does Not Exist back from 213.61.187.150 but when i have tried to send calls using xlite softphone it worked and the calls passed without any problems. __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wrong password on authentication for INVITE to 'asterisk
i tried to send calls through an asterisk box to a voip provider the calls failed and here what i got : *CLI Sep 24 11:09:19 WARNING[23356]: chan_sip.c:6890 handle_response: Forbidden - wrong password on authentication for INVITE to 'asterisk sip:[EMAIL PROTECTED]:5070;tag=as667cb0ae' -- SIP/call-3f73 is circuit-busy == Everyone is busy/congested at this time -- Got SIP response 481 Call Leg Does Not Exist back from 213.61.187.150 but when i have tried to send calls using xlite softphone it worked and the calls passed without any problems. You've made a hell of a lot of assumptions that we understand your configuration, and we don't. What is 195.112.214.99 and 213.61.187.150? Is your sip phone registered with asterisk? (what does sip show peers indicate?) Is your sip phone or asterisk registering with your sip provider? (what does sip show registry indicate?) Paste the appropriate sections of sip.conf and extensions.conf along with some clue what addresses and extensions are what. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SS7 support ?
Is there any digium card that support E1 with SS7 and does Asterisk support SS7 ??? any 1 who has done this ? Usman ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 support ?
On Sat, 24 Sep 2005, Usman wrote: Is there any digium card that support E1 with SS7 and does Asterisk support SS7 ??? any 1 who has done this ? Maybe google has? http://www.google.nl/search?q=Asterisk+SS7start=0start=0ie=utf-8oe=utf-8client=firefoxrls=org.mozilla:en-US:unofficial He does: http://www.voip-info.org/tiki-index.php?page=Asterisk+SS7 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Seperate siptrunks
Hi all. Is it possible to get * to send calls to different sip trunks depending on what codec the incoming call use? This to avoid transcoding Anders ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wrong password on authentication for INVITE to 'asterisk
i have an asterisk box (195.112.214.99) with this
configuration:
sip.conf
[callshop]
type=peer
host=sip.callshopcompany.com
username=XXX
secret=XX
allow=all
extensions.conf
[call]
exten = _00.,1,Dial,SIP/callshop/${EXTEN}
and when i try to send calls to the voip provider
(callshopcompany 213.61.187.150) i got these
messages:
*CLI dial [EMAIL PROTECTED]
-- Executing Dial(OSS/dsp,
SIP/callshop/0017046872001) in new stack
-- Called callshop/0017046872001
*CLI Sep 24 14:16:45 WARNING[22295]: chan_sip.c:6890
handle_response: Forbidden - wrong password on
authentication for INVITE to 'asterisk
sip:[EMAIL PROTECTED]:5070;tag=as4cda63c2'
-- SIP/callshop-f613 is circuit-busy
== Everyone is busy/congested at this time
-- Got SIP response 481 Call Leg Does Not Exist
back from 213.61.187.150
Sep 24 14:16:58 WARNING[22295]: pbx.c:1949
ast_pbx_run: Timeout, but no rule 't' in context
'call'
Hangup on console
but when ive tried it on xlite in the same
configuration to send calls to the same company it
worked and the calls passed without any problems.
so whats the problem here,why the call goes well using
xlite and fails using asterisk despite they have the
same configuration.
--- Rich Adamson [EMAIL PROTECTED] wrote:
i tried to send calls through an asterisk box to a
voip provider the calls failed and here what i got
:
*CLI Sep 24 11:09:19 WARNING[23356]:
chan_sip.c:6890
handle_response: Forbidden - wrong password on
authentication for INVITE to 'asterisk
sip:[EMAIL PROTECTED]:5070;tag=as667cb0ae'
-- SIP/call-3f73 is circuit-busy
== Everyone is busy/congested at this time
-- Got SIP response 481 Call Leg Does Not
Exist
back from 213.61.187.150
but when i have tried to send calls using xlite
softphone it worked and the calls passed without
any
problems.
You've made a hell of a lot of assumptions that we
understand
your configuration, and we don't.
What is 195.112.214.99 and 213.61.187.150?
Is your sip phone registered with asterisk? (what
does sip show
peers indicate?)
Is your sip phone or asterisk registering with your
sip provider?
(what does sip show registry indicate?)
Paste the appropriate sections of sip.conf and
extensions.conf
along with some clue what addresses and extensions
are what.
___
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[Asterisk-Users] HP DL360 G4 EM64T and hyperthreading options
Hi all, Just a couple of quick questions. I have a HP DL360 G4 (dual 3.0Ghz processors). The processors are EM64T. I am using a TE411P in the system. 1. Should I run the a x86_64 Linux (CentOS) or just go with the plain old 32 bit version? 2. This being a dual processor system, should I turn on or off hyper thrreading? Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] context question
I briefly looked thru the code and I don't believe there is a way to separate the context or really make them independent. I know exactly what you want to accomplish. I think it could be done with a little trick. For example, every customer on hosted pbx would be given some kind of unique identifier. The back-end would silently place the identifier at the beginning or the end of the context making the new name totally unique. The front-end would hide identifier from users view and just present the name of the context. That way, customers can name their context anything they like and there would be no collision. In that case, Goto would also be local to the context as the real context name will contain customer id. Does that work for you? Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter http://www.0xdecafbad.com Sent: Friday, September 23, 2005 11:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] context question They are aware of each other in 2 senses. First you can goto() them. I wanted to stop the ability of someone to put in a goto() in their dialplan to a context that is someone elses (think asterisk hosting). Second naming collissions. I wanted to stop two people from having the same name and causing grief that way. That is why I made the references about prepending some customer id or something, but I dont think that is the best way to accomplish this (personal preference), so it will either be an AGI to accomplish this or it will be something else that already exists that I havent been able to locate as yet. On Fri, 2005-09-23 at 21:50 -0500, [EMAIL PROTECTED] wrote: I may be missing something, but aren't all contexts unaware of each other be default? If I do the following [contexta] exten = 3200,1,Dial(SIP/3200,5) [contextb] exten = 3300,1,Dial(SIP/3300,5) Each context has a phone and they can't call each other. The are completely isolated. Unless I'm missing what you are trying to do trixter http://www.0xdecafbad.com wrote: Is there any way within asterisk to limit the scope of contexts, basically to make one context totally unaware of another. The application I had in mind involved allowing users to create their own dial plans. To that end I wanted to make it so that a given user could not call a different users dialplan. I could filter everything and prepend a customer id to every context they specify, but that can get ugly fast, especially when the parser misses something. If this doesnt exist I can surely do it with an agi, and that is the road I am headed down right now, but why duplicate an effect that may already exist? Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Will Digium Wildard work with PCI-X or PCI Express
Hi, You stated that Digium is discontinuing the Wildcard series - that would be there whole product line! In particular I am looking at the Wildcard TDM 400P series of cards.. Thanks Matt Roth wrote: Don't bank on it. We were going to use a Wildcard as a timing source on our Dell PowerEdge 6850 and the BIOS didn't see it. Depending on the PCI-X slot I installed it in, sometimes the box wouldn't even boot. For perspective the 6850 has 4 PCI-e slots, and 3 PCI-X slots (one 64-bit 133 MHz, two 32-bit 100 MHz). I believe the timing is only needed for music on hold, IAX trunking, and MeetMe conferencing. We're not doing trunking or conferencing (for now) so we're going with ztdummy. If the timing isn't perfect only our music on hold will suffer, which is no big deal. If we run into other problems, we might try popping our quad-span card in there just to see if it works. Keep in mind that Digium no longer produces Wildcards. I'm not sure why they don't work with our 6850 and the techs at Dell didn't know either. Maybe they are not 100% PCI compliant. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer Kevin Bockman wrote: Chuck Bunn wrote: Does anyone know if the Digium Wildcard will work on a PCI Express or PCI-X motherboard. Specifically I am looking at the Dell 850 1U rack server for use with Asterisk. They will work in PCI-X of course but not PCI Express. They are totally different. You will need the 3.3v cards. Kevin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BT100 can't register
My BT100 won't register with my Asterisk server, it always comes back with a 403. I've included my sip_additional (only one to to have the username 2201) and a portion of the sniffer trace (packets 27 28). This has me puzzled as I have my SPA-3K working (incoming and outgoing). On my BT100 I get no dial tone, I can't call it (asterisk says the extension is busy) but I can call out from my BT100 to other extensions and through the SPA to the POTS line. Don't assume I really know what I'm doing. One minute it all makes sense and the next I'm clueless. Thanks Oh, I trimmed the sniffer trace to only include the SIP decode. == [2201] username = 2201 authuser = 2201 secret= 2201 type = friend host = gs1.uucp ; host = 192.168.24.192 port = 5060 context = from-internal callerid = Grandstream 2201 mailbox = 2201 nat = never dtmfmode = rfc2833 canreinvite = yes qualify = yes ; qualify = no ; outgoinglimit = 2 ; permit only 1 outgoing call at a time ; incominglimit = 1 ; disable callwaiting signal (2nd call to phone) disallow = all ; need to disallow=all before we can use allow= allow = ulaw; Note: In user sections the order of codecs ; listed with allow= does NOT matter! allow = alaw == No. TimeSourceDestination Protocol Info 27 453.810961 gs1.uucp mozart.uucp SIP Request: REGISTER sip:asterisk.uucp(remove all bindings) == Session Initiation Protocol Request-Line: REGISTER sip:asterisk.uucp SIP/2.0 Method: REGISTER Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.24.192;branch=z9hG4bKd57bf9e7269cdaee From: Neil J. Cherry sip:[EMAIL PROTECTED];user=phone;tag=b946eeaaed68b378 SIP Display info: Neil J. Cherry SIP from address: sip:[EMAIL PROTECTED] SIP tag: b946eeaaed68b378 To: sip:[EMAIL PROTECTED];user=phone SIP to address: sip:[EMAIL PROTECTED] Contact: * Supported: replaces Call-ID: [EMAIL PROTECTED] CSeq: 100 REGISTER Expires: 0 User-Agent: Grandstream BT110 1.0.7.11 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 == No. TimeSourceDestination Protocol Info 28 453.811410 mozart.uucp gs1.uucp SIP Status: 403 Forbidden(1 bindings) == Session Initiation Protocol Status-Line: SIP/2.0 403 Forbidden Status-Code: 403 Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.24.192;branch=z9hG4bKd57bf9e7269cdaee From: Neil J. Cherry sip:[EMAIL PROTECTED];user=phone;tag=b946eeaaed68b378 SIP Display info: Neil J. Cherry SIP from address: sip:[EMAIL PROTECTED] SIP tag: b946eeaaed68b378 To: sip:[EMAIL PROTECTED];user=phone;tag=as6453730d SIP to address: sip:[EMAIL PROTECTED] SIP tag: as6453730d Call-ID: [EMAIL PROTECTED] CSeq: 100 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 == -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help!! trying to use an MTA
Hi gang, I've been trying to use asterisk with an MTA device can any one offer some help as to how asterisk can work with the thing. thanks a mil Calvin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk returns 484 ADDRESS INCOMPLETE for incoming SIP calls
I'm new to asterisk and need some help with getting a SIP connection
working.
I am trying to establish a termination point/DID number in another
country. I am currently running Asterisk CVS-HEAD. My foreign provider
uses SIP and authenticates via IP address. I am not required to
register my SIP connection in order to send or receive calls.
Can someone help me with how to understand the error I see below with
receiving incoming calls?
My asterisk box is behind my IPCop firewall. The current configuration
works fine for
outgoing calls, but has problems with receiving incoming ones.
My current configuration looks like:
[general]
context=default
bindaddr=192.168.0.4
srvlookup=no
disallow=all
allow=ulaw
localnet=192.168.0.0/255.255.255.0
externip=65.87.XXX.XXX
nat=no
fromdomain = mydomain.com
[200.XXX.XXX.XXX]
type=peer
secret=asterisk
host=200.XXX.XXX.XXX
allow=ulaw
context=outgoing
dtmfmode=rfc2833
insecure=very
[from-200.XXX.XXX.XXX]
type=user
host=200.XXX.XXX.XXX
allow=ulaw
canreinvite=no
context=outgoing
insecure=very
Outgoing calls seem to work fine, but there is no indication of any
incoming calls in the SIP debug
information when I call the DID number externally. I have all the SIP and
RTP port forwarded to my Asterisk box in my firewall and don't see
anything in the firewall logs.
I do see the following 2 entries back-to-back in an ethereal dump. I
don't know enough about
SIP to know if the DID side is sending a bad INVITE or if Asterisk is not
handling the INVITE
correctly. I cannot tell if the DID side is not responding back with more
address detail or if my Asterisk box is dropping the connection right
after the 484 response.
Can someone help?
Thanks,
Frank
No. TimeSourceDestination Protocol
Info
2497 21.504651 XXX-IPA.155.115.200.in-addr.arpa lyla.mydomain.com
SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session
description
Frame 2497 (1088 bytes on wire, 1088 bytes captured)
Arrival Time: Sep 22, 2005 23:19:50.962763000
Time delta from previous packet: 0.003659000 seconds
Time since reference or first frame: 21.504651000 seconds
Frame Number: 2497
Packet Length: 1088 bytes
Capture Length: 1088 bytes
Ethernet II, Src: 00:04:e2:bc:76:80, Dst: 00:0e:0c:62:cb:08
Destination: 00:0e:0c:62:cb:08 (lyla.mydomain.com)
Source: 00:04:e2:bc:76:80 (SmcNetwo_bc:76:80)
Type: IP (0x0800)
Internet Protocol, Src Addr: XXX-IPA.155.115.200.in-addr.arpa
(200.115.155.XXX), Dst Addr: lyla.mydomain.com (192.168.0.4)
Version: 4
Header length: 20 bytes
Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00)
00.. = Differentiated Services Codepoint: Default (0x00)
..0. = ECN-Capable Transport (ECT): 0
...0 = ECN-CE: 0
Total Length: 1074
Identification: 0x (0)
Flags: 0x04 (Don't Fragment)
0... = Reserved bit: Not set
.1.. = Don't fragment: Set
..0. = More fragments: Not set
Fragment offset: 0
Time to live: 44
Protocol: UDP (0x11)
Header checksum: 0x2632 (correct)
Source: XXX-IPA.155.115.200.in-addr.arpa (200.115.155.XXX)
Destination: lyla.mydomain.com (192.168.0.4)
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
Source port: 5060 (5060)
Destination port: 5060 (5060)
Length: XXX4
Checksum: 0x3933 (correct)
Session Initiation Protocol
Request-Line: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Method: INVITE
Resent Packet: False
Message Header
Via: SIP/2.0/UDP 200.115.155.XXX:5060
Via: SIP/2.0/UDP 200.115.155.XXX:5061;branch=z9hG4bK-e4907aa1
From: office1 sip:[EMAIL PROTECTED];tag=bc58fe6c90fb9969o1
SIP Display info: office1
SIP from address: sip:[EMAIL PROTECTED]
SIP tag: bc58fe6c90fb9969o1
To: sip:[EMAIL PROTECTED]
SIP to address: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
Max-Forwards: 69
Contact: office1 sip:[EMAIL PROTECTED]:5060
Expires: 240
User-agent: Sipura/SPA3000-2.0.10(GWf)
Content-Length: 432
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
Record-Route: sip:200.115.155.XXX:5060;lr
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): - 13054566 13054566 IN IP4
200.115.155.XXX
Owner Username: -
Session ID: 13054566
Session Version: 13054566
Owner Network Type: IN
Owner Address Type: IP4
Owner Address: 200.115.155.XXX
Session Name (s): -
Connection Information (c): IN IP4 200.115.155.XXX
Connection Network Type: IN
Connection Address Type: IP4
Connection Address: 200.115.155.XXX
Re: [Asterisk-Users] dial (iax/Xsip/y) get y fraction earlier
exten = _06.,1,Dial(IAX2/X/${EXTEN},30,rSIP/[EMAIL PROTECTED])
Even that's incorrect. It should be:
exten = _06.,1,Dial(IAX2/X/${EXTEN}SIP/[EMAIL PROTECTED],30,r)
See:
[Description]
Dial(Technology/resource[Technology2/resource2...][|timeout][|options][|URL]):
Also, see the Wiki or this mailing list regarding the use of the 'r' option.
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Re: [Asterisk-Users] VM low volume - testers needed
Hmmm...
I checked out CVS-HEAD, built and installed it this morning. Most
testing was going well, but then I found out the behavior of
ChanIsAvail has changed (is broken?)
In my Dial Plan, if a call comes in on the PSTN line, and is not
answered by the extension (or if the extension is busy), ChanIsAvail
checks to see of the outgoing VOIP line is available. If so, it
forwards the call to the VOIP voice mail. If not, it forwards the call
to the Asterisk Voicemail.
With 1.2-beta, ChanIsAvail works for me. With CVS-HEAD, it hangs up on the caller.
Here is the relevant portion of my extensions.conf:
exten = s,7,Dial(${PHONE1},15)
exten = s,8,Goto(108)
exten = s,108,ChanIsAvail(${VOIP1})
exten = s,109,Dial(${VOIP1}/${VOIPNUM})
exten = s,209,VoiceMail(123|sbg(6))
In the globals section, VOIP1 is set equal to Zap/4
With 1.2-beta, -vvv logs show this, which is successful:
-- Executing ChanIsAvail(Zap/3-1, Zap/4) in new stack
-- Executing VoiceMail(Zap/3-1, 123|sbg(6)) in new stack
-- Playing '/var/spool/asterisk/voicemail/default/123/busy' (language 'en')
With CVS-HEAD -vvv logs show this, which is unsuccessful:
-- Executing ChanIsAvail(Zap/3-1, Zap/4) in new stack
== Spawn extension (incoming-pstn, s, 208) exited non-zero on 'Zap/3-1'
-- Hungup 'Zap/3-1'
Is there another list or someone I should mention this to? Asterisk should not hangup Zap/3-1 at this point.
On 9/24/05, Rich Adamson [EMAIL PROTECTED] wrote:
The patch is in cvs-head, which has been very stable for me. :) Hi Richard, I am experiencing the same problem. I'd like to test your patch. Thing, is, I don't know which
CVS it's in:) ... I checked out 1.2-beta on Tuesday (9/21) and compiled it. When I type 'show applicationvoicemail', it does not describe the g(#) option, so I think my version must not have it.
I am using a TDM22B card and voicemails seem very quiet if they are left from in incoming POTSconnection. When I enter voicemail by direct dialing a local extension and leave a message from the advanced options
menu, the recorded message is much louder. I should qualify, not only are my VMs coming in over POTS, I am actually calling out firstthrough the TDM22B, to Sipura, to VOIP provider, back in via PSTN, to TDM22B, to VM. I'm amazed it works at all:)... I'm very
impressed by Asterisk and especially it's voicemail. I would like to resolve the low volume issue though. If you can tell me which CVS to check out, I can try it. I'd like to stick to the 1.2-beta
branch though because I don't want to rework all my config files. On 9/21/05, Rich Adamson [EMAIL PROTECTED] wrote: On Monday 19 September 2005 12:38, Rich Adamson wrote:
The g(6) adds a 6 db gain for zap calls that end up recording a Voicemail message. ...* 'g(#)' the specified amount of gain will be requested during message
recording
(units are whole-number decibels (dB)) How in the hell does that make any sense?are your normal incoming calls quiet too or just voicemail? Yes, see bug 2022 and 2023 for details, as well as
http://www.routers.com/asteriskprob/asterisk-config.htm for a very detailed analysis of the problem. I believe one of the more serious issues amounts to: if asterisk is
located a fair distance from the central office (-7db in my case), setting the rxgain and/or txgain to any level that would be considered reasonable for that loss (eg, rxgain=5, txgain=5), hugh amounts of echo result that
cannot be addressed through zapata.conf echo entris, and changing compile options to agressive, etc, does not help. Its my believe (from working with several TDM users), the further one is from the CO,
the bigger the problem. (Or, short pstn cable lengths less then about 4 or 5db can almost always be addressed via parameters.) The above workaround is very usable (assuming it works) when someone
calls in via the pstn and leaves a voicemail (which is already at least 7db down plus their own pstn loss), and then I call in via the pstn to retrive the voicemail (now 14db down PLUS the original callers
pstn loss), the audio is so faint its difficult to impossible to listen to. In my case, the asterisk box is located about 7db from the central office. As noted in bug 2023 (and 2022), calls from an outside pstn
line coming into asterisk incure a 7db pstn loss (which can't be adjusted for with rxgain and txgain as changing those values to something reasonable generates echo).Retrieving that VM message from an outside
location creates another 7db loss (now -14db down in total), making it very difficult (if not impossible) to hear the message. (And, yes I've gone through all the recommendations with wav vs gsm files, etc.)
I am not sure I understand why the txgain/rxgain isn't fixing it without adding unacceptable echo...this all seems very odd...I mean for a test you should be able to dial an echo() application and have extremely quiet
echoed audio... is this the case? As an ex-telco transmission engineer,
Re: [Asterisk-Users] VM low volume - testers needed
Under 1.2 the +101 jumping is not enabled by default. There is a
variable returned showing the status of the application. You need to add
a j flag or put priorityjumping=yes in extensions.conf
Julian.
Brian McEntire wrote:
Hmmm...
I checked out CVS-HEAD, built and installed it this morning. Most testing
was going well, but then I found out the behavior of ChanIsAvail has changed
(is broken?)
In my Dial Plan, if a call comes in on the PSTN line, and is not answered by
the extension (or if the extension is busy), ChanIsAvail checks to see of
the outgoing VOIP line is available. If so, it forwards the call to the VOIP
voice mail. If not, it forwards the call to the Asterisk Voicemail.
With 1.2-beta, ChanIsAvail works for me. With CVS-HEAD, it hangs up on the
caller.
Here is the relevant portion of my extensions.conf:
exten = s,7,Dial(${PHONE1},15)
exten = s,8,Goto(108)
exten = s,108,ChanIsAvail(${VOIP1})
exten = s,109,Dial(${VOIP1}/${VOIPNUM})
exten = s,209,VoiceMail(123|sbg(6))
In the globals section, VOIP1 is set equal to Zap/4
With 1.2-beta, -vvv logs show this, which is successful:
-- Executing ChanIsAvail(Zap/3-1, Zap/4) in new stack
-- Executing VoiceMail(Zap/3-1, 123|sbg(6)) in new stack
-- Playing '/var/spool/asterisk/voicemail/default/123/busy' (language 'en')
With CVS-HEAD -vvv logs show this, which is unsuccessful:
-- Executing ChanIsAvail(Zap/3-1, Zap/4) in new stack
== Spawn extension (incoming-pstn, s, 208) exited non-zero on 'Zap/3-1'
-- Hungup 'Zap/3-1'
Is there another list or someone I should mention this to? Asterisk should
not hangup Zap/3-1 at this point.
On 9/24/05, Rich Adamson [EMAIL PROTECTED] wrote:
The patch is in cvs-head, which has been very stable for me. :)
Hi Richard,
I am experiencing the same problem. I'd like to test your patch. Thing,
is, I don't know which
CVS it's in :)
... I checked out 1.2-beta on Tuesday (9/21) and compiled it. When I
type 'show application
voicemail', it does not describe the
g(#) option, so I think my version must not have it.
I am using a TDM22B card and voicemails seem very quiet if they are left
from in incoming POTS
connection. When I enter
voicemail by direct dialing a local extension and leave a message from
the advanced options
menu, the recorded message is much
louder.
I should qualify, not only are my VMs coming in over POTS, I am actually
calling out first
through the TDM22B, to Sipura, to
VOIP provider, back in via PSTN, to TDM22B, to VM. I'm amazed it works
at all :) ... I'm very
impressed by Asterisk and
especially it's voicemail. I would like to resolve the low volume issue
though.
If you can tell me which CVS to check out, I can try it. I'd like to
stick to the 1.2-beta
branch though because I don't want to
rework all my config files.
On 9/21/05, Rich Adamson [EMAIL PROTECTED] wrote:
On Monday 19 September 2005 12:38, Rich Adamson wrote:
The g(6) adds a 6 db gain for zap calls that end up recording a
Voicemail
message.
...
* 'g(#)' the specified amount of gain will be requested during
message
recording (units are whole-number decibels (dB))
How in the hell does that make any sense? are your normal incoming
calls
quiet too or just voicemail?
Yes, see bug 2022 and 2023 for details, as well as
http://www.routers.com/asteriskprob/asterisk-config.htm
for a very detailed analysis of the problem.
I believe one of the more serious issues amounts to: if asterisk is
located a fair distance from the central office (-7db in my case),
setting
the rxgain and/or txgain to any level that would be considered
reasonable
for that loss (eg, rxgain=5, txgain=5), hugh amounts of echo result that
cannot be addressed through zapata.conf echo entris, and changing
compile options to agressive, etc, does not help. Its my believe
(from working with several TDM users), the further one is from the CO,
the bigger the problem. (Or, short pstn cable lengths less then about
4 or 5db can almost always be addressed via parameters.)
The above workaround is very usable (assuming it works) when someone
calls in via the pstn and leaves a voicemail (which is already at
least 7db down plus their own pstn loss), and then I call in via the
pstn to retrive the voicemail (now 14db down PLUS the original callers
pstn loss), the audio is so faint its difficult to impossible to
listen to.
In my case, the asterisk box is located about 7db from the central
office. As noted in bug 2023 (and 2022), calls from an outside pstn
line coming into asterisk incure a 7db pstn loss (which can't be
adjusted
for with rxgain and txgain as changing those values to something
reasonable generates echo). Retrieving that VM message from an
outside
location creates another 7db loss (now -14db down in total), making
it
very difficult (if not impossible) to hear the message. (And, yes
I've
gone through all the recommendations with wav vs gsm files, etc.)
I am
[Asterisk-Users] unable to use misdn group dial
I have set up a * box with two hfc ISDN pci cards
using mISDN both in TE mode with PmP mode.
(using $MODPROBE hfcpci protocol=0x2,0x2
layermask=0xf,0xf)
I have no problem dialing out by explicitly naming the
mISDN port, ex: Dial(mISND/1/${EXTEN},60)
or Dial(mISDN/2/${EXTEN},60)
But it does NOT work when specifying the mISDN group:
exten =
_(outpattern),1,Dial(mISDN/g:TEmode/${EXTEN},60)
exten = _(outpattern),2,Congestion
I get a message such as, When dialing out from Zap/1
channel:
Executing
Dial(Zap/1-1,mISDN/g:incoming/2107253178|60) in
new stack
Checking Availbl. Chan in Group: incoming
-- * NEW CHANNEL dad: oad:2107253178 ctx:
* CALL: g:TEmode/2107253178
-- Group Call group: TEmode
def_l1:-1, portup:0
-- ! No free channel chan 0x81afc20 even after Group
Call
-- SEND: State Down
-- Couldn't call g:TEmode/2107253178
== Everyone is busy/congested at this time (0:0/0/0)
-- Executing Congestion(Zap/1-1, ) in new
stack
Both cards are in the same group, in misdn.conf:
[TEmode]
ports=1,2
immediate=yes
context=incoming
callerid=Some CallerId
msns=*
the misdn device is set up with:
mknod /dev/mISDN c 46 0
I don't if one mISDN device is enough for two cards,
so I also created another one:
mknod /dev/mISDN1 c 46 1
but it made no difference...
Any suggestions?
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[Asterisk-Users] Need Help on Areski Calling Card Solution plz
Can someone share its working files experience on areskicc with me. I got it installed but my sip user and iax could not get registered talkless of making call and all the include directives instructed in the idiot guide were followed. Can someone share its experience with me on this? Aruna -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of CM Rahman Jr. Sent: Tuesday, July 19, 2005 8:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Comments on Areski Calling Card Solution plz I am using it. I liked it. The guy did a good job. He doesn't have the agent module yet. But I think that is on its way. Thanks Quoting Arnd Vehling [EMAIL PROTECTED]: Hi, can anyone who has the Areski Calling Card solution on Asterisk working comment on it? Is is stable enough for a production system? Any pros and cons? thx, Arnd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CM Rahman Jr. CTO CCS Internet www.ccsi.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] context question
On Sat, 2005-09-24 at 09:10 -0400, Alex Vishnev wrote: I briefly looked thru the code and I don't believe there is a way to separate the context or really make them independent. I know exactly what you want to accomplish. I think it could be done with a little trick. For example, every customer on hosted pbx would be given some kind of unique identifier. The back-end would silently place the identifier at the beginning or the end of the context making the new name totally unique. The front-end would hide identifier from users view and just present the name of the context. That way, customers can name their context anything they like and there would be no collision. In that case, Goto would also be local to the context as the real context name will contain customer id. Does that work for you? no, because as I stated I didnt like that for personal reasons. That sounds exactly what I was thyinking too, prepending some customer specific identifier. If that is the only way to do this, then I think I will just have to run everything through an AGI, which can differentiate between customers since none of the 'dialplan' is in extensions.conf :) Thanks though, at least its confirmed that this doesnt exist (yet anyway). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC on Fedora 4 and MySQL 4.1.12
I just installed the CVS 9-22 and am trying to get ASTCC up and running. I was able to get the web interface config running and it made the database but when I go to the brands page it says there is a problem with the table. Also when I save the config file through the intraface it wont save it to any location. I want to set up a small CC application so if there is a better product to use please let me know. Thanks, Scott ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BRI Hunting, using both channels on one msn
On Fri, 23 Sep 2005 [EMAIL PROTECTED] wrote: Hello Armin, I tried your new version of chan capi and it works well. I did have one question about capi.conf. I have a bri with 2 spids, but I want to have the second go to a zap fax channel. Right now I can direct it, but the echo canceller is setting up. Do you know a way to cancel it? Fax works, but I suspect it would work better with EC off on large faxes. I don't know enough about the spid stuff, but you should be able to create two interfaces in capi.conf (instead of one), which devices=1 for each. So you should be able to set different settings for each channel. When using fax via capi with Eicon cards, the echo canceler should not be used automatically. I tried the capi fax receive, but the images came out with the wrong dimensions(on .05). Maybe a problem with setting fine/normal resolution? Also, Is there a way to split each msn into a different call group in capi.conf? I tried a few combinations but no luck. I was thinking I could disable the EC for the line in general. See above. Oh as per the hunt, I had verizon program a hunt into the line and it seems to work now. It is funny though, since I think my usrobotics modem can also do it, I just don't know exactly how it is handled. capi just causes the line to report a busy. And there is a new eicon driver that works on 2.6. Which one do you mean? Armin Thanks, Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 16, 2005 4:49 AM To: Gregory Wiktor - ADCom Corp. Subject: RE: [Asterisk-Users] BRI Hunting, using both channels on one msn Hi Greg, now I understand. You use NI-1 with spids. I'm sorry, I don't know anything about this protocol. ETSI does not have this 'channel-problem'. Maybe it can be solved with some load parameters for the BRI card. You should use the latest driver and divactrl (possibly the SRPM from Eicon). regards, Armin On Tue, 16 Aug 2005 [EMAIL PROTECTED] wrote: Hello Armin, My setup is as follows: I have 1 bri with 2 spid's, or msn's. 2781980 and 2781984. If a call comes in to 2781980, and is active, and another call comes in to 2781980, the second call will be busy. A call to 278-1984 will proceed while the 1980 is busy. The telco tells me though that the bri should be capable of hunting on it's own. I did this in the past with modem banks, but they were on top of centrex. What I would like to do is put an 800 number to point to the 278-1980, and for the most part not use the 278-1984 except for maybe a disa. The eiconctrl monitor app is aware that the line is busy, and I do not believe it is notifying asterisk of the issue. I am trying to move some lines to bri since my audio quality on pots has been horrible. The isdn is great, especially since you told me of the ulaw modification I needed to make... I got lucky with this one, since they really could not install it without doing special construction, which I managed to avoid paying the big bucks for because the csr was nice about the 3 month delay. I set it up through a panasonic dbs so the secretary can just hit a button, and I get immediate rings on 4 sip phones and my cell. I would love a PRI, but only need 4 channels max which is why I went with the bri. Compared to pots, the isdn is way better. I also find it much more stable than IP, to the point where it is worth the 1c/minute to use. Thanks for the help. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 16, 2005 2:06 AM To: Gregory Wiktor - ADCom Corp. Subject: RE: [Asterisk-Users] BRI Hunting, using both channels on one msn On ISDN, the second channel is automatically used if the first channel is busy. Normaly you never get a busy signal, just because ONE channel is busy. Only if there is no application/phone available for that MSN, then you get busy. Or maybe I just don't understand what you are doing... Armin On Mon, 15 Aug 2005 [EMAIL PROTECTED] wrote: Well, I want to direct a toll free to my first msn. The problem is, if the line is busy a busy signal is returned. I want the line to hunt to the next channel, so it can be answered on the first msn. Regards, Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Monday, August 15, 2005 3:53 PM To: Gregory Wiktor - ADCom Corp. Cc: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] BRI Hunting, using both channels on one msn On Mon, 15 Aug 2005 [EMAIL PROTECTED] wrote: Hello All, Has anyone configured bri to answer for only one msn? In essence, when the primary is busy I want to have channel 2 ring. I am using an eicon diva server bri I know I saw it in
Re: [Asterisk-Users] ASTCC on Fedora 4 and MySQL 4.1.12
I fought with this one for hours last night. I have to get it yet but I'm not sure what the problem is. The permissions are all fine. Any comments anyone? Darren Wiebe [EMAIL PROTECTED] Scott Wolfe wrote: I just installed the CVS 9-22 and am trying to get ASTCC up and running. I was able to get the web interface config running and it made the database but when I go to the brands page it says there is a problem with the table. Also when I save the config file through the intraface it wont save it to any location. I want to set up a small CC application so if there is a better product to use please let me know. Thanks, Scott ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID issue
Adam Moffett wrote: Hello. I'm having trouble with callerid on outgoing calls. The recipient of the call only sees unknown rather than the number I'm specifying. If I set callerid info when calling an internal extension then I see the callerid name and number when I call that extension. I did that thusly: Reverting back to 1.09 Saturday brought my caller-id back. Doug ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Send DTMF after call bridge
Hello everyone. Let me first begin by explaining what I'm trying to do... I have a calling card that has an access number and requires a PIN to be entered and then the number you want to dial, like normal calling cards. So what I have done is assign a local DID which when called, initiates a Dial to the access number of the calling card. Now, I'm having a hard time figuring out how to send DTMF tones via the dialplan once the call has been bridged. So far I've tried using 'w' in the Dial string to specify the wait period before dialing the digits that follow. I've also tried using the D(digits) option for the Dial application but it clearly says that it will only send the digits once the channel is answered and before it is bridged. So how in the world can you send DTMF via the dialplan to a bridged call? Is it even possible? Thanks in advance for any help. Regards, Mohammed Salim ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VM low volume - testers needed
Hmm. Thanks for the heads up, but I'm not sure that's it.
It's jumping to 208 rather than 209, so it looks more like an off-by-one error.
I tried changing to priorityjumping=yes in
/etc/asterisk/extensions.conf and reinstalled the CVS-HEAD version, but
it still jumps to 208 whereas it used to jump to 209.
On 9/24/05, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
Under 1.2 the +101 jumping is not enabled by default. There is avariable returned showing the status of the application. You need to adda j flag or put priorityjumping=yes in extensions.confJulian.
Brian McEntire wrote: Hmmm... I checked out CVS-HEAD, built and installed it this morning. Most testing was going well, but then I found out the behavior of ChanIsAvail has changed
(is broken?) In my Dial Plan, if a call comes in on the PSTN line, and is not answered by the extension (or if the extension is busy), ChanIsAvail checks to see of the outgoing VOIP line is available. If so, it forwards the call to the VOIP
voice mail. If not, it forwards the call to the Asterisk Voicemail. With 1.2-beta, ChanIsAvail works for me. With CVS-HEAD, it hangs up on the caller. Here is the relevant portion of my
extensions.conf: exten = s,7,Dial(${PHONE1},15) exten = s,8,Goto(108) exten = s,108,ChanIsAvail(${VOIP1}) exten = s,109,Dial(${VOIP1}/${VOIPNUM}) exten = s,209,VoiceMail(123|sbg(6))
In the globals section, VOIP1 is set equal to Zap/4 With 1.2-beta, -vvv logs show this, which is successful: -- Executing ChanIsAvail(Zap/3-1, Zap/4) in new stack
-- Executing VoiceMail(Zap/3-1, 123|sbg(6)) in new stack -- Playing '/var/spool/asterisk/voicemail/default/123/busy' (language 'en') With CVS-HEAD -vvv logs show this, which is unsuccessful:
-- Executing ChanIsAvail(Zap/3-1, Zap/4) in new stack == Spawn extension (incoming-pstn, s, 208) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1'
Is there another list or someone I should mention this to? Asterisk should not hangup Zap/3-1 at this point. On 9/24/05, Rich Adamson
[EMAIL PROTECTED] wrote:The patch is in cvs-head, which has been very stable for me. :)Hi Richard,
I am experiencing the same problem. I'd like to test your patch. Thing,is, I don't know whichCVS it's in :)... I checked out 1.2-beta on Tuesday (9/21) and compiled it. When I
type 'show applicationvoicemail', it does not describe theg(#) option, so I think my version must not have it.I am using a TDM22B card and voicemails seem very quiet if they are left
from in incoming POTSconnection. When I entervoicemail by direct dialing a local extension and leave a message fromthe advanced options
menu, the recorded message is muchlouder.I should qualify, not only are my VMs coming in over POTS, I am actuallycalling out first
through the TDM22B, to Sipura, toVOIP provider, back in via PSTN, to TDM22B, to VM. I'm amazed it worksat all :) ... I'm veryimpressed by Asterisk and
especially it's voicemail. I would like to resolve the low volume issuethough.If you can tell me which CVS to check out, I can try it. I'd like to
stick to the 1.2-betabranch though because I don't want torework all my config files.On 9/21/05, Rich Adamson
[EMAIL PROTECTED] wrote:On Monday 19 September 2005 12:38, Rich Adamson wrote:The g(6) adds a 6 db gain for zap calls that end up recording a
Voicemailmessage* 'g(#)' the specified amount of gain will be requested during
messagerecording (units are whole-number decibels (dB))How in the hell does that make any sense? are your normal incoming
callsquiet too or just voicemail?Yes, see bug 2022 and 2023 for details, as well as
http://www.routers.com/asteriskprob/asterisk-config.htmfor a very detailed analysis of the problem.I believe one of the more serious issues amounts to: if asterisk is
located a fair distance from the central office (-7db in my case),settingthe rxgain and/or txgain to any level that would be consideredreasonable
for that loss (eg, rxgain=5, txgain=5), hugh amounts of echo result thatcannot be addressed through zapata.conf echo entris, and changingcompile options to agressive, etc, does not help. Its my believe
(from working with several TDM users), the further one is from the CO,the bigger the problem. (Or, short pstn cable lengths less then about4 or 5db can almost always be addressed via parameters.)
The above workaround is very usable (assuming it works) when someonecalls in via the pstn and leaves a voicemail (which is already atleast 7db down plus their own pstn loss), and then I call in via the
pstn to retrive the voicemail (now 14db down PLUS the original callerspstn loss), the audio is so faint its difficult to impossible tolisten to.
In my case, the asterisk box is located about 7db from the centraloffice. As noted in bug 2023 (and 2022), calls from an outside pstnline coming into asterisk incure a 7db pstn loss (which can't be
adjustedfor with rxgain and txgain as changing those values to somethingreasonable generates echo). Retrieving that VM message from an
outsidelocation
Re: [Asterisk-Users] wrong password on authentication for INVITE to 'asterisk
i have an asterisk box (195.112.214.99) with this
configuration:
sip.conf
[callshop]
type=peer
host=sip.callshopcompany.com
username=XXX
secret=XX
allow=all
extensions.conf
[call]
exten = _00.,1,Dial,SIP/callshop/${EXTEN}
and when i try to send calls to the voip provider
(callshopcompany 213.61.187.150) i got these
messages:
*CLI dial [EMAIL PROTECTED]
-- Executing Dial(OSS/dsp,
SIP/callshop/0017046872001) in new stack
-- Called callshop/0017046872001
*CLI Sep 24 14:16:45 WARNING[22295]: chan_sip.c:6890
handle_response: Forbidden - wrong password on
authentication for INVITE to 'asterisk
Sure looks like an authentication problem. If you are absolutely
positive you have no typos in the username/secret, then have you
tried the suggestions from
/usr/src/asterisk/configs/sip.conf.sample
that suggests:
;[sip_proxy-out]
;type=peer ; we only want to call out, not be called
;secret=guessit
;username=yourusername ; Authentication user for outbound proxies
;fromuser=yourusername ; Many SIP providers require this!
;fromdomain=provider.sip.domain
;host=box.provider.com
;usereqphone=yes; This provider requires ;user=phone on URI
;call-limit=5 ; permit only 5 simultaneous outgoing calls to
this peer
Looks like a couple more parameters might be needed.
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Re: [Asterisk-Users] wrong password on authentication for INVITE to 'asterisk
hi:
no , i dont think it an authentication problem
,because once i have experienced this problem with
another voip provider and when i told him the problem
he fix the problem at his side ,so i think it an
invitation problem at his side.
--- Rich Adamson [EMAIL PROTECTED] wrote:
i have an asterisk box (195.112.214.99) with this
configuration:
sip.conf
[callshop]
type=peer
host=sip.callshopcompany.com
username=XXX
secret=XX
allow=all
extensions.conf
[call]
exten = _00.,1,Dial,SIP/callshop/${EXTEN}
and when i try to send calls to the voip provider
(callshopcompany 213.61.187.150) i got these
messages:
*CLI dial [EMAIL PROTECTED]
-- Executing Dial(OSS/dsp,
SIP/callshop/0017046872001) in new stack
-- Called callshop/0017046872001
*CLI Sep 24 14:16:45 WARNING[22295]:
chan_sip.c:6890
handle_response: Forbidden - wrong password on
authentication for INVITE to 'asterisk
Sure looks like an authentication problem. If you
are absolutely
positive you have no typos in the username/secret,
then have you
tried the suggestions from
/usr/src/asterisk/configs/sip.conf.sample
that suggests:
;[sip_proxy-out]
;type=peer ; we only want to
call out, not be called
;secret=guessit
;username=yourusername ; Authentication
user for outbound proxies
;fromuser=yourusername ; Many SIP providers
require this!
;fromdomain=provider.sip.domain
;host=box.provider.com
;usereqphone=yes; This provider
requires ;user=phone on URI
;call-limit=5 ; permit only 5
simultaneous outgoing calls to this peer
Looks like a couple more parameters might be needed.
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Re: [Asterisk-Users] Play sound on connect
Thanks for your answer. This is not what the customer wants, they answer +500 calls a day, and dont want to say Welcome to BigCorp every time. They want a personal welcome file to be played to the caller every time they pick up the ringing phone. Michael 2005/9/24, Mathew McKernan [EMAIL PROTECTED]: Hi Mir, You would need to put a Play command in before the Dial command. For example: Exten = 108,1,Play(108-greeting) Exten = 108,2,Dial(SIP/108) Etc. This however, will play on _every_ attempted call to 108. If 108 is offline or unreachable the caller will still hear the message. Thanks Matty Mathew McKernan | Support Engineer | Digital World Computers | * +61 3 9318 6022 | * mat at dwonline.com.au or visit www.dwonline.com.au This email is intended solely for the use of the addressee and may contain information that is confidential or subject to legal professional privilege. If you receive this email in error please immediately notify the sender and delete the email. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mir Sent: Saturday, 24 September 2005 6:39 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Play sound on connect Hello A calls B, on connect I want B's greeting to be played to caller A. I can see it is possible to play a sound to B on connect (DIAL(SIP/123 ,A(hello)), but I cant se how to play a sound to A, is this possible? Thank you Michael ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Send DTMF after call bridge
On Saturday 24 September 2005 21:21, Mohammed Salim wrote:
Hello everyone.
Let me first begin by explaining what I'm trying to do...
I have a calling card that has an access number and requires a PIN to be
entered and then the number you want to dial, like normal calling cards. So
what I have done is assign a local DID which when called, initiates a Dial
to the access number of the calling card.
I'm in the same case and you, and the D(digits) options of Dial command works
fine for me. I use it this way:
exten = _9.,1,Dial(SIP/[EMAIL PROTECTED]||D(w6969w${EXTEN}))
Where 6969 is my calling card number (or whatever).
--
Alvaro Gamez Machado.
[EMAIL PROTECTED]
Hazent Systems, S.L.
http://www.hazent.com
C/Rio Cañamares 2, Oficina ocho
28804 Alcalá de Henares
Madrid
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[Asterisk-Users] Directed pickup syntax?
What's the proper syntax for implementing directed call pickup?
Running cvs-head from today (9/24/05 including Mark's fixes), and
tried:
exten = *99,1,Pickup(${EXTEN:3})
but that does not seem to work, and there isn't an example in the
configs directory. 'show application pickup' suggests the above
should work with our sip phones, but apparently I'm missing
something.
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[Asterisk-Users] Falsh Panel in Xorcom Rapid
I have a clean install of rapid 1.1 installed. I have installed the Flash Operator Panel from the Install Other Software menu. I am able to log into the panel from another computer on my network but all I see is the Conference Room 300. There are no extensions or any other options on the panel. Clearly I am missing something. Can anyone giude me as to how to get the Panel going so that I can see the extensions and other options. Thanks in advance. Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VM low volume - testers needed
Oops, I didn't cc the list. Julian suggested I should try the older version of app_chanisavail.c and that worked out well. I can now use the g(#) switch and that works very well. On 9/24/05, Brian McEntire [EMAIL PROTECTED] wrote: That fixes it! Thanks. So I can run CVS HEAD but I need to check out -r 1.17 asterisk/apps/app_chanisavail.c to revert just that file to the old version. I guess it could still be a prob with the new app_chanisavail.c but it also looks like whatever provides ast_goto_if_exists could be at fault. - - - To Rich: The new gain g(#) switch works great! I have to bump mine up to g(12) which seems rediculously high, but then again I'm going out voip and back in PSTN and perhaps the VOIP is quieting the signal too. Anway, with g(12), voicemail messages are recorded at a very acceptable volume and sound good too. Thanks! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Directed pickup syntax?
Try:
exten = *99,1,Pickup(${EXTEN:[EMAIL PROTECTED])
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Rich Adamson
Sent: Saturday, September 24, 2005 6:29 PM
To: Asterisk-users-list
Subject: [Asterisk-Users] Directed pickup syntax?
What's the proper syntax for implementing directed call pickup?
Running cvs-head from today (9/24/05 including Mark's fixes), and
tried:
exten = *99,1,Pickup(${EXTEN:3})
but that does not seem to work, and there isn't an example in
the configs directory. 'show application pickup' suggests the
above should work with our sip phones, but apparently I'm
missing something.
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RE: [Asterisk-Users] Play sound on connect
Would using an IVR that ends with connecting .. do it - or do you have to have the call answered by someone who will wait until the recording plays in every call ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mir Sent: Sunday, September 25, 2005 00:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Play sound on connect Thanks for your answer. This is not what the customer wants, they answer +500 calls a day, and dont want to say Welcome to BigCorp every time. They want a personal welcome file to be played to the caller every time they pick up the ringing phone. Michael 2005/9/24, Mathew McKernan [EMAIL PROTECTED]: Hi Mir, You would need to put a Play command in before the Dial command. For example: Exten = 108,1,Play(108-greeting) Exten = 108,2,Dial(SIP/108) Etc. This however, will play on _every_ attempted call to 108. If 108 is offline or unreachable the caller will still hear the message. Thanks Matty Mathew McKernan | Support Engineer | Digital World Computers | * +61 3 9318 6022 | * mat at dwonline.com.au or visit www.dwonline.com.au This email is intended solely for the use of the addressee and may contain information that is confidential or subject to legal professional privilege. If you receive this email in error please immediately notify the sender and delete the email. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mir Sent: Saturday, 24 September 2005 6:39 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Play sound on connect Hello A calls B, on connect I want B's greeting to be played to caller A. I can see it is possible to play a sound to B on connect (DIAL(SIP/123 ,A(hello)), but I cant se how to play a sound to A, is this possible? Thank you Michael ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pictures from VON Fall 2005 Digium/Asterisk booth
Enjoy! http://www.asterisk.org/vonfall2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC on Fedora 4 and MySQL 4.1.12
Okay, after spending 12 hours on it I checked the thing that has bit me before. Turn SElinux off. OUCH!! :-) Darren Wiebe [EMAIL PROTECTED] Darren Wiebe wrote: I fought with this one for hours last night. I have to get it yet but I'm not sure what the problem is. The permissions are all fine. Any comments anyone? Darren Wiebe [EMAIL PROTECTED] Scott Wolfe wrote: I just installed the CVS 9-22 and am trying to get ASTCC up and running. I was able to get the web interface config running and it made the database but when I go to the brands page it says there is a problem with the table. Also when I save the config file through the intraface it wont save it to any location. I want to set up a small CC application so if there is a better product to use please let me know. Thanks, Scott ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Directed pickup syntax?
You have to tell it the extension you want to pick up, it's not psychic.
Doing what you're doing now would give the application no extension.
Exten = _*99.,1,Pickup(${EXTEN:3}) should work, with usage being
*99extension to pickup
Joshua Colp
On 9/24/05 7:28 PM, Rich Adamson [EMAIL PROTECTED] wrote:
What's the proper syntax for implementing directed call pickup?
Running cvs-head from today (9/24/05 including Mark's fixes), and
tried:
exten = *99,1,Pickup(${EXTEN:3})
but that does not seem to work, and there isn't an example in the
configs directory. 'show application pickup' suggests the above
should work with our sip phones, but apparently I'm missing
something.
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[Asterisk-Users] PA1688 Phones using IAX MWI
Anybody have these working with Asterisk? I have an AT-320. Brett ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IBM x306
Hi, This is a little off-topic,but if someone has any info, it could help me a LOT!, I am building an asterisk pbx (1.0.9) on an IBM x306 SCSI machine,my problem is that the BIOS assigns the same IRQ to the SCSI controller, and the TDM400P, i have tried several options of making the bios change the IRQ, but it will always move them together, anyone with some info about my options ? Thanks, Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IBM x306
Can you try a different slot on the PCI bus?? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marco Supino Sent: Saturday, September 24, 2005 8:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] IBM x306 Hi, This is a little off-topic,but if someone has any info, it could help me a LOT!, I am building an asterisk pbx (1.0.9) on an IBM x306 SCSI machine,my problem is that the BIOS assigns the same IRQ to the SCSI controller, and the TDM400P, i have tried several options of making the bios change the IRQ, but it will always move them together, anyone with some info about my options ? Thanks, Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM x306
On Sun, 25 Sep 2005, Marco Supino wrote: I am building an asterisk pbx (1.0.9) on an IBM x306 SCSI machine,my problem is that the BIOS assigns the same IRQ to the SCSI controller, and the TDM400P, i have tried several options of making the bios change the IRQ, but it will always move them together, anyone with some info about my options ? Linux usually don't care about Bios settings, you could try kernel cmdline parameters. Acpi and IRQ are google terms for it. Stefan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM x306
Only one PCI slot can hold the full size card like the TDM400P , the other slot has a smaller opening on the case. Marco. Alexander Lopez wrote: Can you try a different slot on the PCI bus?? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marco Supino Sent: Saturday, September 24, 2005 8:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] IBM x306 Hi, This is a little off-topic,but if someone has any info, it could help me a LOT!, I am building an asterisk pbx (1.0.9) on an IBM x306 SCSI machine,my problem is that the BIOS assigns the same IRQ to the SCSI controller, and the TDM400P, i have tried several options of making the bios change the IRQ, but it will always move them together, anyone with some info about my options ? Thanks, Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM x306
Hi, I tried setpci INTERRUPT_LEVEL (or something similar, cant remmeber now), and also setpci seems like it changed the IRQ, lspci -v still shows the old IRQ Marco. Stefan de Konink wrote: On Sun, 25 Sep 2005, Marco Supino wrote: I am building an asterisk pbx (1.0.9) on an IBM x306 SCSI machine,my problem is that the BIOS assigns the same IRQ to the SCSI controller, and the TDM400P, i have tried several options of making the bios change the IRQ, but it will always move them together, anyone with some info about my options ? Linux usually don't care about Bios settings, you could try kernel cmdline parameters. Acpi and IRQ are google terms for it. Stefan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need good explanation on contexts and extensions
Hello:
My Asterisk book is on its way, so please bear with me.
Based on what I have read and my actual Asterisk experiences, I am not
too clear on the context-extension relationship. I am not sure if some
of the error messages (Not Found) are a result of a bug or a feature.
My experience so far is limited to sip.conf and extensions.conf, as I
don't have a hardware board yet.
First: It seems like an extension can be part of more than one context?
If you have something like this:
sip.conf:
---
[general]
port = 5060
context = incoming
[1234]
context = internal
---
What's the meaning of the above? Is this like a programming language
where the global variable 'context' is shadowed by the local variable
with the same name?
Or is the extension '1234' somehow related to both contexts ('incoming'
and 'internal')?
Thanks,
-Ramon F Herrera
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[Asterisk-Users] Software to generate an SRTP key pair?
I have been looking all over for software to generate the keys needed to have secure calls with my Sipura. The only one that I have found is on-line and thus not so secure: http://voxilla.com/certrequest.php Any pointers? Thx, -RFH ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cheap Time sources which is best?
On the same P2 450Mhz box. I have tried both UHCI usb on a 2.4 kernel and enhanced RTC on a 2.6 kernel. Have not tried UHCI USB on a 2.6 kernel as of yet. Both seem to work GREAT. I have read in many places to be sure to use a digium card as a time source and not to reply on the cheap solutions. However I have regular meetme sessions of 5 and 6 people at the same time that frequently go on for an hour or so and we have not noticed any problems yet. I'd like to know 'what is better' the enhanced RTC or UHCI-USB. and at what point do you start to need a digium board for timing? Seems that for even a low volume production system that I have not needed a recommended timing source yet. Thank you for your insight! Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR problem
Hi to All, I've an Asterisk CVS Head working with Mysql. My problem is that instead of ANSWERED or something like, into the CDR database records, I find only numbers. This is also a problem to let ASTPP works, infact I receive an error: ERROR - ERROR - ERROR - ERROR - ERROR DISPOSITION NOT MATCHED and the call has no cost. Any suggestions? Thanks -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR problem
Could you post an example of you cdr output. The ASTPP question would be better put on astpp-users. Visit http://aleph.aleph-com.net/mailman/listinfo/astpp-users to subscribe. Darren Wiebe [EMAIL PROTECTED] FaberK wrote: Hi to All, I've an Asterisk CVS Head working with Mysql. My problem is that instead of ANSWERED or something like, into the CDR database records, I find only numbers. This is also a problem to let ASTPP works, infact I receive an error: ERROR - ERROR - ERROR - ERROR - ERROR DISPOSITION NOT MATCHED and the call has no cost. Any suggestions? Thanks -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SoundPoint IP Attendant Console
So the IP 601 is the 600 with a few extras? Looks like Polycom dropped the ball again -- yet another pretty phone with NO BACK LIGHT. Does the design team at Polycom have their brains unscrewed? I've been playing with some Aastra phones lately, with limited success on working properly. The only motivations for checking out these other phones were the two things that polycom has been lacking from the get-go...REAL LEDs and Backlit displays. PBX and KSU telephones have had backlights for more than a decade now. If I wanted a phone I can't see in the dark, I'd use the Panasonic D1232 I already had... I've been using a set of 300s, 500s and one 600 for 4 months now, and have been very happy with the results. As soon as Polycom pulls its head out of its you-know-what and makes LEDs, backlights, and maybe a usable administrator's guide, I'll consider them for my customers in the future. So far, looks like Aastra is winning the bid...(barely) Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, September 21, 2005 6:36 PM Subject: Re: [Asterisk-Users] SoundPoint IP Attendant Console Bartosz Jozwiak wrote: Does anybody use SoundPoint IP Attendant Console for Polycom IP 601 with asterisk ? Is it going to work with hints in dial plan ? Since it is not even shipping yet (it was just announced two days ago), the answer is no. However, we have had a test unit for some time (and we have one in our booth at VON), and yes, it works just like the built-in buttons on the phone. The only issue today with displaying hint status is an artificial limit of eight (8) 'buddies' in the Contact Directory to watch. Once Polycom has released the final firmware for the phone with support for a larger number of watched contacts, the expansion module will be fully usable with Asterisk. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM x306
On Sun, Sep 25, 2005 at 03:04:31AM +0200, Stefan de Konink wrote: On Sun, 25 Sep 2005, Marco Supino wrote: I am building an asterisk pbx (1.0.9) on an IBM x306 SCSI machine,my problem is that the BIOS assigns the same IRQ to the SCSI controller, and the TDM400P, i have tried several options of making the bios change the IRQ, but it will always move them together, anyone with some info about my options ? Linux usually don't care about Bios settings, you could try kernel cmdline parameters. Acpi and IRQ are google terms for it. But it's the hardware (bios? interrupt controller) that sends the interrupts in the first place. Right? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cheap Time sources which is best?
On Sat, Sep 24, 2005 at 10:23:42PM -0400, Steve Gladden wrote: On the same P2 450Mhz box. I have tried both UHCI usb on a 2.4 kernel and enhanced RTC on a 2.6 kernel. Have not tried UHCI USB on a 2.6 kernel as of yet. Both seem to work GREAT. I have read in many places to be sure to use a digium card as a time source and not to reply on the cheap solutions. What does zttest tell you? How good is it as a diagnostic tool? What can it tell me? However I have regular meetme sessions of 5 and 6 people at the same time that frequently go on for an hour or so and we have not noticed any problems yet. I'd like to know 'what is better' the enhanced RTC or UHCI-USB. RTC, I figure. The USB code is just a hack. However the 2.6 kernel has some on-going work regarding timers. and at what point do you start to need a digium board for timing? Seems that for even a low volume production system that I have not needed a recommended timing source yet. Thank you for your insight! -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IPSpeedDial has just been released
IPSpeedDial creates speed dial numbers for Asterisk. Download from: http://ipsoftware.thorben.dk Use this to create speed dial numbers that can be used by all extensions on your Asterisk server. This program will create entries in the asterisk database which you then can lookup in you dial plan get the number to dial. All extensions can now dial these speed dial number. IPManager has this build into the dial plan and will work without further changes. This is also supported by IPSwitchBoard so you can search the names in the call box of IPSwitchBoard. The program can import .txt or .csv files containing your speed dials. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC on Fedora 4 and MySQL 4.1.12
Thanks for this. Interface works as it should now. -Scott - Original Message - From: Darren Wiebe [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, September 24, 2005 5:07 PM Subject: Re: [Asterisk-Users] ASTCC on Fedora 4 and MySQL 4.1.12 Okay, after spending 12 hours on it I checked the thing that has bit me before. Turn SElinux off. OUCH!! :-) Darren Wiebe [EMAIL PROTECTED] Darren Wiebe wrote: I fought with this one for hours last night. I have to get it yet but I'm not sure what the problem is. The permissions are all fine. Any comments anyone? Darren Wiebe [EMAIL PROTECTED] Scott Wolfe wrote: I just installed the CVS 9-22 and am trying to get ASTCC up and running. I was able to get the web interface config running and it made the database but when I go to the brands page it says there is a problem with the table. Also when I save the config file through the intraface it wont save it to any location. I want to set up a small CC application so if there is a better product to use please let me know. Thanks, Scott ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] didgium card in india
where can i buy the digium or any other card to work with asterisk in india and what is the cost like __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dialplan game
Has anyone built a game with the dialplan? I would think this would most easily be managed by an AGI, but its possible with realtime extensions. The game would be like 'adventure' that I first played on a prime in 1979. Or any of the infocom games (ie zork). Infact since the infocom spec is known it might be possible to plug in the data files directly from an AGI. If anyone has done this I would love to hear about it. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] didgium card in india
Such hardware I believe incurs a stock standard duty of 35% plus some other charges. All up, AFAIK it will cost you $2300USD to import the card (based on the $1495 price for a 4 E1 card). You can try guys like Drishti in Delhi, they can help out. Regards, Sahil Gupta VoiceValley On Sat, 24 Sep 2005, Capt MS wrote: where can i buy the digium or any other card to work with asterisk in india and what is the cost like __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: goiax expanded with free us domestic calling
Can I ask how you are providing calls to us domestic numbers for free? goiax.com is backed by TxLink [www.txlink.net]. We terminate a lot of minutes. Matt: That first logo ( companylogo / www.webaddresshere.com ) on the website could use some work :) but the service works great!! Thanks! -Andy ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] didgium card in india
thanks for the reply Is Digium card compatible with EPABX standards available in india , further how much does a card with three FXS and one FXO interface cost, Do u have any experience of implenting the same , I am in army what we lookin at is voice gateway to interface our PBX with the data network so that we have one underlying network to handle , any suggestions on how to implement in a cost effective manner. --- Sahil Gupta [EMAIL PROTECTED] wrote: Such hardware I believe incurs a stock standard duty of 35% plus some other charges. All up, AFAIK it will cost you $2300USD to import the card (based on the $1495 price for a 4 E1 card). You can try guys like Drishti in Delhi, they can help out. Regards, Sahil Gupta VoiceValley On Sat, 24 Sep 2005, Capt MS wrote: where can i buy the digium or any other card to work with asterisk in india and what is the cost like __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Extension Mobility (roaming) Cisco 7960
Hi, I was wondering if it is possible to setup with Asterisk a Cisco 7960 to use extension mobility / roaming. Meaning that a user logs into a phone and his profile moves with him / her. I have a network of ~75 Cisco 7960 phones, running SIP 7.5 distributed across 2 asterisk servers in 2 cities. I would like to enable the users to have the capability to move from one phone to another one with the extension moving with them. Also have the capability that when a user uses a soft phone like Eyebeam, they dont need to have 2 extensions setup with a ring group in order for it to work. Anyone have any ideas on how to make this work? Please let me know Thanks Sascha ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dialplan game
On Sat, Sep 24, 2005 at 10:08:19PM -0700, trixter http://www.0xdecafbad.com wrote: Has anyone built a game with the dialplan? I would think this would most easily be managed by an AGI, but its possible with realtime extensions. The game would be like 'adventure' that I first played on a prime in 1979. Or any of the infocom games (ie zork). Infact since the infocom spec is known it might be possible to plug in the data files directly from an AGI. If anyone has done this I would love to hear about it. Such a game requires the player to keep a lot of state information in the head. Why not start with something simpler? I forgot the English name of the game, but it's the game where one player randomly chooses 4 digits and the other player has to guess them and their right order, and for each guess the reply is the total number of correct digits and the total number of digits that are also in the right place. Also, looking at the package bsdgames, some games are command-line based and thus could be adapted to a dialplan control. There are some adventure-type games. And there is also monop (monopoly). Though frankly, I'm not sure those would be of any atraction to any user. You do need the game to sing a little bit, as it can't dance. But singing will become annoying after a while if there's no simple way of skipping it. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
